Sarvam Ucs v1r5 System Manual

SARVAM UCS
System Manual
SARVAM UCS
The Unified Communication Server
System Manual
Documentation Disclaimer
Matrix Comsec reserves the right to make changes in the design or components of the product as engineering and
manufacturing may warrant. Specifications are subject to change without notice.
This is a general documentation of the product. The product may not support all the features and facilities
described in the documentation.
Information in this documentation may change from time to time. Matrix Comsec reserves the right to revise
information in this publication for any reason without prior notice. Matrix Comsec makes no warranties with respect
to this documentation and disclaims any implied warranties. While every precaution has been taken in the
preparation of this system manual, Matrix Comsec assumes no responsibility for errors or omissions. Neither is any
liability assumed for damages resulting from the use of the information contained herein.
Neither Matrix Comsec nor its affiliates shall be liable to the purchaser of this product or third parties for damages,
losses, costs or expenses incurred by the purchaser or third parties as a result of: accident, misuse or abuse of this
product or unauthorized modifications, repairs or alterations to this product or failure to strictly comply with Matrix
Comsec's operating and maintenance instructions.
Copyright
All rights reserved. No part of this system manual may be copied or reproduced in any form or by any means
without the prior written consent of Matrix Comsec.
Version 1
Release date: October 10, 2017
Contents
Overview.......................................................................................................................................................... 1
About the Product ............................................................................................................................................... 1
Introduction..................................................................................................................................................... 5
Welcome ............................................................................................................................................................. 5
About this System Manual .................................................................................................................................. 5
Know Your SARVAM UCS............................................................................................................................ 11

Introduction ....................................................................................................................................................... 11
Applications of SARVAM UCS .......................................................................................................................... 15
Hardware Overview ........................................................................................................................................... 21
The Interfaces ................................................................................................................................................... 25
Before You Start ........................................................................................................................................... 35

Protecting the System and Yourself .................................................................................................................. 43
Installing ETERNITY LENX........................................................................................................................... 47

The Power Supply Card .................................................................................................................................... 52
The CPU Card ................................................................................................................................................... 59
The Single Line Telephone Card ...................................................................................................................... 71
The Intercom Line Card .................................................................................................................................... 83
The Digital Key Phone Card .............................................................................................................................. 93
The CO Card ................................................................................................................................................... 116
The BRI Card .................................................................................................................................................. 122
The T1E1PRI Card .......................................................................................................................................... 132
The E1FO Card ............................................................................................................................................... 139
The E&M Card ................................................................................................................................................ 146
The Mobile Card .............................................................................................................................................. 154
The Magneto Card .......................................................................................................................................... 158
The Radio Card ............................................................................................................................................... 162
The Data Card ................................................................................................................................................. 165
SIP Extensions ................................................................................................................................................ 167
Starting Up ETERNITY LENX ......................................................................................................................... 209
Installing ETERNITY MENX........................................................................................................................ 211

The Power Supply Card .................................................................................................................................. 216
The CPU Card ................................................................................................................................................. 220
The Single Line Telephone Card .................................................................................................................... 232
The Intercom Line Card .................................................................................................................................. 240
Table of Contents i
The Digital Key Phone Card ............................................................................................................................ 243
The CO Card ................................................................................................................................................... 266
The BRI Card .................................................................................................................................................. 272
The T1E1PRI Card .......................................................................................................................................... 282
The E1FO Card ............................................................................................................................................... 289
The E&M Card ................................................................................................................................................ 296
The Mobile Card .............................................................................................................................................. 304
The Magneto Card .......................................................................................................................................... 309
The Radio Card ............................................................................................................................................... 313
The Data Card ................................................................................................................................................. 316
SIP Extensions ................................................................................................................................................ 318
Starting Up ETERNITY MENX ........................................................................................................................ 360
Installing ETERNITY GENX ........................................................................................................................ 361

The Power Supply Card .................................................................................................................................. 365
The CPU Card ................................................................................................................................................. 369
The Single Line Telephone Card .................................................................................................................... 380
The Intercom Line Card .................................................................................................................................. 389
The Digital Key Phone Card ............................................................................................................................ 391
The CO Card ................................................................................................................................................... 414
The BRI Card .................................................................................................................................................. 421
The T1E1PRI Card .......................................................................................................................................... 430
The E1FO Card ............................................................................................................................................... 436
The E&M Card ................................................................................................................................................ 442
The Mobile Card .............................................................................................................................................. 448
The Magneto Card .......................................................................................................................................... 452
The Radio Card ............................................................................................................................................... 455
The Data Card ................................................................................................................................................. 457
SIP Extensions ................................................................................................................................................ 458
Starting Up ETERNITY GENX ........................................................................................................................ 500
Configuring ETERNITY GENX ................................................................................................................... 501
Configuring SARVAM UCS ........................................................................................................................ 507

Configuring using Web-based GUI: Jeeves .................................................................................................... 507
Configuring using a Telephone ....................................................................................................................... 513
Using Quick Installation Wizard - Standard PBX ............................................................................................ 518
Using Configuration ......................................................................................................................................... 562
Configuring System Pre-requisites .................................................................................................................. 564
Configuring ‘Region’ ........................................................................................................................................ 567
Configuring Network Parameters .................................................................................................................... 572
Configuring Maintenance Port Parameters ..................................................................................................... 589
Configuring Redundancy Parameters ............................................................................................................. 592
Configuring Extensions ................................................................................................................................... 594
Configuring SIP Extensions ............................................................................................................................ 623
Configuring Matrix SPARSH VP248 ............................................................................................................... 627
Configuring Matrix VARTA WIN200 UC Client ................................................................................................ 697
Configuring Matrix VARTA ADR100/AMP100 UC Clients .............................................................................. 708
VARTA License Management ......................................................................................................................... 722
Configuring Open SIP Phones ........................................................................................................................ 724
Third Party IP-Phone General Parameters ..................................................................................................... 763
Device Management ....................................................................................................................................... 765
Black List IP Address - SIP Extensions ........................................................................................................... 767
ii Table of Contents
Configuring DKP Extensions ........................................................................................................................... 769
DSS Keys Programming ................................................................................................................................. 787
Configuring ISDN Terminals ........................................................................................................................... 838
Configuring SLT Extensions ............................................................................................................................ 845
Configuring Radio Interface ............................................................................................................................ 854
Configuring Magneto Interface ........................................................................................................................ 862
Configuring 'Operator' ..................................................................................................................................... 871
Extension Search ............................................................................................................................................ 876
Configuring Trunks .......................................................................................................................................... 877
Configuring VoIP Parameters ......................................................................................................................... 932
Configuring SIP Trunks ................................................................................................................................... 938
Configuring PRI Trunks ................................................................................................................................... 954
Configuring T1 Trunks ..................................................................................................................................... 959
T1 RBS Parameters ...................................................................................................................................... 1002
Configuring E1 Trunks .................................................................................................................................. 1011
Configuring Data Parameters ........................................................................................................................ 1077
Configuring BRI Trunks ................................................................................................................................. 1081
Configuring Mobile Trunks ............................................................................................................................ 1120
Configuring CO Trunks ................................................................................................................................. 1142
Configuring E&M Lines ................................................................................................................................. 1153
Configuring LCR ............................................................................................................................................ 1161
Configuring Emergency Number Dialing ....................................................................................................... 1175
Configuring Voice Mail System ............................................................................................................... 1183

Configuring VMS General Parameters .......................................................................................................... 1184
General Mailbox Settings .............................................................................................................................. 1191
Extension Voice Mail Settings ....................................................................................................................... 1193
Voice Mail Auto-Attendant Menu ................................................................................................................... 1201
Language Selection Profile ........................................................................................................................... 1221
Message Profile ............................................................................................................................................ 1225
Call Transfer Profile ...................................................................................................................................... 1238
Extensions Over Q-SIG ................................................................................................................................. 1266
Notification via Call-Profile ............................................................................................................................ 1268
Mailbox Menu ................................................................................................................................................ 1271
Distribution List .............................................................................................................................................. 1279
Recording Voice Messages .......................................................................................................................... 1281
Prompts Management ................................................................................................................................... 1284
Mailbox Status ............................................................................................................................................... 1294
Voicemail Backup .......................................................................................................................................... 1297
VMS Debug ................................................................................................................................................... 1299
SMS Server................................................................................................................................................ 1301

Bulk SMS ...................................................................................................................................................... 1306
SMS Routing ................................................................................................................................................. 1317
SMS Server - Mail Settings ........................................................................................................................... 1323
SMS Server Reports ..................................................................................................................................... 1326
SMS over IP .................................................................................................................................................. 1333
SMS/Email Group ......................................................................................................................................... 1338
Computer Telephony Integration (CTI) ................................................................................................... 1341
Incoming and Outgoing Call Routing ..................................................................................................... 1345

Incoming Call Routing ................................................................................................................................... 1345
Outgoing Call Routing ................................................................................................................................... 1355
Features and Facilities ............................................................................................................................. 1359

Abbreviated Dialing ....................................................................................................................................... 1359
Table of Contents iii

Alternate Number Dialing for EON74 ............................................................................................................ 1387
Access Codes ............................................................................................................................................... 1389
Account Codes .............................................................................................................................................. 1397
AC Impedance Test ...................................................................................................................................... 1405
Alarms ........................................................................................................................................................... 1413
Alternate Number Dialing .............................................................................................................................. 1428
Authority Codes ............................................................................................................................................. 1435
Apple Push Notification Service Support ...................................................................................................... 1442
Auto Attendant .............................................................................................................................................. 1444
Auto Answer .................................................................................................................................................. 1451
Auto Call Back (ACB) .................................................................................................................................... 1455
Auto Redial .................................................................................................................................................... 1461
Automatic Number Translation ...................................................................................................................... 1466
Barge-In ........................................................................................................................................................ 1471
BCCH Selection ............................................................................................................................................ 1475
Behind the System Application ..................................................................................................................... 1479
Building Intercom .......................................................................................................................................... 1483
Busy Lamp Field for Trunks .......................................................................................................................... 1489
Call Back on Trunk Ports .............................................................................................................................. 1492
Call Budget on Extension .............................................................................................................................. 1498
Call Budget on Trunk .................................................................................................................................... 1504
Call Chaining ................................................................................................................................................. 1515
Call Cost Calculation (CCC) .......................................................................................................................... 1517
Call Cost Display ........................................................................................................................................... 1549
Call Duration Control (CDC) .......................................................................................................................... 1551
Call Duration Display ..................................................................................................................................... 1561
Call Forward .................................................................................................................................................. 1562
Call Forward-Remote .................................................................................................................................... 1569
Call Forward-Scheduled ................................................................................................................................ 1574
Call Forward-When Not Registered .............................................................................................................. 1582
Call Hold ........................................................................................................................................................ 1588
Call Park ........................................................................................................................................................ 1594
Call Logs ....................................................................................................................................................... 1597
Call Pick Up ................................................................................................................................................... 1602
Call Progress Tones ...................................................................................................................................... 1608
Call Restriction based on IP Address ............................................................................................................ 1619
Call Taping .................................................................................................................................................... 1622
Call Toggle .................................................................................................................................................... 1631
Call Traffic ..................................................................................................................................................... 1634
Call Transfer .................................................................................................................................................. 1638
Calling Line Identification and Presentation (CLIP) ....................................................................................... 1646
Calling Line Identification Restriction (CLIR) ................................................................................................. 1652
Cancel All Station Features ........................................................................................................................... 1655
Class of Service (COS) ................................................................................................................................. 1657
CLI Based Routing ........................................................................................................................................ 1666
Clock Synchronization ................................................................................................................................... 1671
Closed User Group (CUG) ............................................................................................................................ 1675
Closed User Group-With Exchange ID ......................................................................................................... 1681
Communication Ports .................................................................................................................................... 1685
Conference-3 Party ....................................................................................................................................... 1690
Conference-Multiparty ................................................................................................................................... 1694
Conference Dial-In ........................................................................................................................................ 1702
Conflict Dialing .............................................................................................................................................. 1710
Conversation Recording ................................................................................................................................ 1713
Customer Name ............................................................................................................................................ 1717
COSEC Integration ....................................................................................................................................... 1719
iv Table of Contents
Day Night Mode ............................................................................................................................................ 1722
Daylight Saving Time (DST) .......................................................................................................................... 1725
Department Call ............................................................................................................................................ 1733
Dial By Name ................................................................................................................................................ 1747
Dialed Number Directory ............................................................................................................................... 1750
Dial Plan for SIP Extension ........................................................................................................................... 1752
Digest Authentication .................................................................................................................................... 1755
Digital Key Phone-Operation ......................................................................................................................... 1757
Direct Dialing-In (DDI) ................................................................................................................................... 1786
Direct Inward System Access (DISA) ............................................................................................................ 1797
Direct Station Selection Console ................................................................................................................... 1806
Distinctive Rings ............................................................................................................................................ 1809
Do Not Disturb (DND) ................................................................................................................................... 1815
DSS Call Pick-Up .......................................................................................................................................... 1825
Dynamic Lock ................................................................................................................................................ 1827
E1/T1 Maintenance ....................................................................................................................................... 1835
E&M Connectivity .......................................................................................................................................... 1842
Emergency Calls (911) - Reporting to PSAP ................................................................................................ 1846
Emergency Conference ................................................................................................................................ 1848
Emergency Detection and Reporting ............................................................................................................ 1852
Emergency Dialing ........................................................................................................................................ 1854
Extended IP Phone/VARTA UC Client - Operation ....................................................................................... 1856
Flash Timer ................................................................................................................................................... 1894
Flashing on Trunks (Continued Dialing) ........................................................................................................ 1895
Flexible Numbers .......................................................................................................................................... 1896
Floor Service ................................................................................................................................................. 1899
Follow Me ...................................................................................................................................................... 1905
Forced Answer .............................................................................................................................................. 1907
Forced Call Disconnection ............................................................................................................................ 1909
Gain Settings ................................................................................................................................................. 1911
Gateway Application-Answer Signaling ........................................................................................................ 1914
GPAX Application .......................................................................................................................................... 1917
GPAX Billing .................................................................................................................................................. 1920
Handover and Handoff .................................................................................................................................. 1921
Help Desk ...................................................................................................................................................... 1922
Holiday Table ................................................................................................................................................ 1923
Hot Desking ................................................................................................................................................... 1927
Hotline ........................................................................................................................................................... 1929
Incoming CLI Modification ............................................................................................................................. 1934
Intercom ........................................................................................................................................................ 1938
Internal Call Restriction ................................................................................................................................. 1940
Interrupt Request (IR) ................................................................................................................................... 1942
Last Caller Recall .......................................................................................................................................... 1944
Last Number Redial ...................................................................................................................................... 1945
Least Cost Routing-Carrier Pre-Selection ..................................................................................................... 1946
License Management .................................................................................................................................... 1948
Live Call Screening ....................................................................................................................................... 1968
Live Call Supervision ..................................................................................................................................... 1970
Logical Partition ............................................................................................................................................. 1971
Loop Back Tests ........................................................................................................................................... 1976
Macros .......................................................................................................................................................... 1985
Meet Me Paging ............................................................................................................................................ 1989
Message Wait ............................................................................................................................................... 1991
Mobility Extension ......................................................................................................................................... 1996
Multi-Stage Dialing ........................................................................................................................................ 2003
Music on Hold (MOH) .................................................................................................................................... 2006
Table of Contents v
Mute .............................................................................................................................................................. 2007
Number Lists ................................................................................................................................................. 2009
OFF-Hook Alert ............................................................................................................................................. 2014
One Touch Transfer ...................................................................................................................................... 2016
Outgoing Trunk Bundle ................................................................................................................................. 2019
OG Trunk Bundle Group ............................................................................................................................... 2022
Paging ........................................................................................................................................................... 2028
Peer-to-Peer Calling ...................................................................................................................................... 2033
PIN Dialing .................................................................................................................................................... 2039
PLCC-An Introduction ................................................................................................................................... 2043
Presence ....................................................................................................................................................... 2051
Preset Call Forward ...................................................................................................................................... 2059
Priority Calls in E&M MFCR2 Signaling ........................................................................................................ 2062
Priority ........................................................................................................................................................... 2066
Privacy .......................................................................................................................................................... 2069
Power Fail Transfer ....................................................................................................................................... 2071
QSIG ............................................................................................................................................................. 2074
Quick Dial ...................................................................................................................................................... 2087
Raid ............................................................................................................................................................... 2089
RCOC (Return Call to Original Caller) .......................................................................................................... 2091
Real Time Clock (RTC) ................................................................................................................................. 2096
Reminder ....................................................................................................................................................... 2102
Remote Programming ................................................................................................................................... 2110
Room Monitor ................................................................................................................................................ 2112
Response Mapping ....................................................................................................................................... 2114
Routing Group ............................................................................................................................................... 2116
Selective Port Access ................................................................................................................................... 2121
Security Settings ........................................................................................................................................... 2123
Self Ring Test ................................................................................................................................................ 2128
Shared Call Appearance ............................................................................................................................... 2129
SIM Card Balance and Recharging ............................................................................................................... 2131
SMS Gateway ............................................................................................................................................... 2134
SMS on No Reply .......................................................................................................................................... 2138
SMTP Settings .............................................................................................................................................. 2141
Simple Network Management Protocol (SNMP) ........................................................................................... 2144
Simple Network Time Protocol - SNTP ......................................................................................................... 2153
Software Port and Hardware ID .................................................................................................................... 2155
Static Routing Table ...................................................................................................................................... 2160
Station Message Detail Recording (SMDR) .................................................................................................. 2164
Station Message Detail Recording-Online .................................................................................................... 2165
Station Message Detail Recording-Posting ................................................................................................... 2185
Station Message Detail Recording-Report .................................................................................................... 2223
Station Message Detail Recording-Storage .................................................................................................. 2248
System Activity Log ....................................................................................................................................... 2257
System Activity Log Display .......................................................................................................................... 2268
System Fault Log .......................................................................................................................................... 2269
System Fault Log Display ............................................................................................................................. 2279
System Log Notification ................................................................................................................................ 2280
System Parameters ....................................................................................................................................... 2282
System Timers and Counts ........................................................................................................................... 2305
System Security ............................................................................................................................................ 2317
Time Tables .................................................................................................................................................. 2327
Time Zone Display ........................................................................................................................................ 2331
Toll Control .................................................................................................................................................... 2332
Trunk Auto Answer ........................................................................................................................................ 2345
Trunk Call Waiting ......................................................................................................................................... 2348
vi Table of Contents
Trunk Landing Group (TLG) .......................................................................................................................... 2349
Trunk Reservation ......................................................................................................................................... 2352
User Absent/Present ..................................................................................................................................... 2354
User Password .............................................................................................................................................. 2356
Video Call ...................................................................................................................................................... 2358
Virtual Extension ........................................................................................................................................... 2361
Voice Help ..................................................................................................................................................... 2366
Voice Message Applications ......................................................................................................................... 2367
Walk-In Class of Service ............................................................................................................................... 2380
Voice Mail Features .................................................................................................................................. 2385

Accessing your Mailbox ................................................................................................................................ 2385
Alarms and Reminders .................................................................................................................................. 2387
VMS DISA Login ........................................................................................................................................... 2393
Sending Messages ........................................................................................................................................ 2396
Redirecting Messages ................................................................................................................................... 2398
Recording Personal Greetings ...................................................................................................................... 2401
Recording Conditional Greetings .................................................................................................................. 2402
Message Verification ..................................................................................................................................... 2403
Message Notification ..................................................................................................................................... 2405
Mailbox Settings ............................................................................................................................................ 2406
Listening to Messages .................................................................................................................................. 2408
Leaving a Message ....................................................................................................................................... 2409
Accessing the General Mailbox ..................................................................................................................... 2410
Forwarding Messages ................................................................................................................................... 2412
Email Based Notification ............................................................................................................................... 2414
Message Wait Notification via Call ................................................................................................................ 2415
Dial By Name ................................................................................................................................................ 2417
Dial by Extension Number ............................................................................................................................. 2418
Call Transfer Types ....................................................................................................................................... 2419
Broadcast Message ...................................................................................................................................... 2420
System Maintenance ................................................................................................................................ 2421

Configuration Backup/Restore ...................................................................................................................... 2421
Firmware Management ................................................................................................................................. 2425
Auto Update .................................................................................................................................................. 2427
Uploading Custom MoH ................................................................................................................................ 2429
Default Settings ............................................................................................................................................. 2430
PCAP Trace .................................................................................................................................................. 2437
Network Diagnosis ........................................................................................................................................ 2445
Network Drive Settings .................................................................................................................................. 2447
Redundancy Debug ...................................................................................................................................... 2449
Redundancy Process .................................................................................................................................... 2451
Restart the Card ............................................................................................................................................ 2453
Restart the System ........................................................................................................................................ 2454
SIP Extension ................................................................................................................................................ 2455
System Debug ............................................................................................................................................... 2456
VoIP Debug ................................................................................................................................................... 2463
System Details .............................................................................................................................................. 2465
System Usage ............................................................................................................................................... 2467
Appendix ................................................................................................................................................... 2469

Technical Specifications - ETERNITY LENX ................................................................................................ 2469
Technical Specifications - ETERNITY MENX ............................................................................................... 2470
Technical Specifications - ETERNITY GENX ............................................................................................... 2471
Technical Specifications - SARVAM UCS ..................................................................................................... 2472
Table of Contents vii

Packing List ................................................................................................................................................... 2489
How to Make the Telecom Earth ................................................................................................................... 2502
VMS Prompts ................................................................................................................................................ 2505
SARVAM UCS Features Supported in Terminals ......................................................................................... 2518
Features at a Glance ..................................................................................................................................... 2521
SA Commands .............................................................................................................................................. 2529
System Commands ....................................................................................................................................... 2537
SARVAM UCS Features tested on IP Phones of different Brands ............................................................... 2579
SARVAM UCS Features supported with RTP/Direct RTP ............................................................................ 2581
Troubleshooting ............................................................................................................................................ 2582
Acronyms ...................................................................................................................................................... 2584
Warranty Statement ...................................................................................................................................... 2590
Disposal of Products/Components after End-Of-Life .................................................................................... 2591
Regulatory Information .................................................................................................................................. 2592
Regulatory Information for Terminals ............................................................................................................ 2596
Open Source Licensing Terms and Conditions ............................................................................................. 2605
viii Table of Contents

CHAPTER 1 Overview
About the Product

Matrix provides different platforms to run the SARVAM Application.
• The ETERNITY GENX platform - This is the common hardware platform for SARVAM UCS SME and
SARVAM UMG Application.
• The ETERNITY LENX/MENX platform - This is the hardware platform for SARVAM UCS ENT Application.
ETERNITY GENX Platform

The ETERNITY GENX is the common hardware platform for SARVAM UCS and SARVAM UMG Application.
You can use the ETERNITY GENX as the Unified Communication Server or the Universal Media Gateway
depending upon the Application License you purchase.
To run the ETERNITY GENX as the Unified Communication Server, you need to purchase the SARVAM UCS SME
Application license and to run it as the Universal Media Gateway, you need to purchase the SARVAM UMG SME
Application license.
UCS Universal Media Gateway
SARVAM UCS SME SARVAM UMG

Application Application
Matrix SARVAM UCS System Manual 1

ETERNITY GENX as the Universal Media Gateway acts as an IP-based system providing value added voice
services. The system enables you to route calls from the Source port to the Destination port using Destination
Number determination and Destination Port determination methods. It also offers a robust SIP Stack which gives a
thorough interoperability and usability with various SIP trunk providers and IP-PBX. The system provides high
performance and high reliability in a compact, modular design.
ETERNITY GENX as the Unified Communication Server acts as a fully hosted and managed Unified
Communication Server. It delivers the convergence of voice, data, wired communications for small and medium
sized businesses. It also offers UC features, Voice over IP Integration, Voice Mail, Computer Telephony Integration
and Switching functions. The system provides reliable, efficient and unrestricted simultaneous communication
(incoming and outgoing) by all users.
SARVAM UCS SME

Application
ETERNITY LENX/MENX Platform

The ETERNITY LENX/MENX is the hardware platform for SARVAM UCS ENT Application.
To run the ETERNITY LENX/MENX as the Unified Communication Server, you need to purchase the SARVAM
UCS ENT Application license.
ETERNITY LENX/MENX as the Unified Communication Server acts as a fully hosted and managed Unified
Communication Server. It delivers the convergence of voice, data, wired communications for small and medium
sized businesses. It also offers UC features, Voice over IP Integration, Voice Mail, Computer Telephony Integration
and Switching functions. The system provides reliable, efficient and unrestricted simultaneous communication
(incoming and outgoing) by all users.
2 Matrix SARVAM UCS System Manual

The manual henceforth will describe in detail — the application, installation, configuration and features of SARVAM
UCS.

CHAPTER 2 Introduction
Welcome
Thank you for choosing our product SARVAM UCS! We hope you will make optimum use of this intelligent, fully
hosted and managed Unified Communication Server. Please read this document carefully to get acquainted with
the product before installing and operating it.
About this System Manual

This System Manual provides information and instructions for installing, configuring and using the SARVAM UCS.
You may also refer to the SARVAM UCS Quick Start shipped with the system for quick installation.
For instructions on using the features of the SARVAM UCS by the clients, refer to the respective User Guides. The
documentation can be found at https://www.matrixtelesol.com/product-manuals.html
For product registration and warranty related details, please visit https://www.matrixcomsec.com/product-
registration-form.html
This is a common documentation for all the platforms and the SARVAM UCS Application. This document is
written with reference to the ETERNITY GENX platform and SARVAM UCS SME Application.
The configuration commands are written with respect to ETERNITY MENX.
Intended Audience
This System Manual is aimed at:

• System Engineers, who will install, maintain and support the system. System Engineers are persons who
customize the system configuration to meet the requirements of the organization/users. It is assumed that
they are experienced in installing the Unified Communication Server, are familiar with telecom wiring
technology, how it works, and the various technical terms and functions associated with it. The SE must
have undergone training in configuring the SARVAM UCS.
No one, other than the System Engineer is permitted to make any alterations to the configuration of the
SARVAM UCS.

• System Administrators, who are the people administering the system. Generally an operator/receptionist
in an organization, or the staff manning the reception or front desk area of the establishment are selected
as System Administrators.
It is assumed that the System Administrators have some previous experience in administering the system
and its Terminals and Consoles. The System Administrators are not expected to install and program the
Unified Communication Server, but only the routine jobs and features that are specific to them like
generating SMDR reports, Setting report filters, configuring the features, Setting Alarms, reminders, etc.
• Users, people/organizations who will use the resources of the SARVAM UCS. They may be executives,
include personnel of small and medium businesses, large enterprises, front desk and service staff of
Hotels/Motels, hospitals, and other commercial and public organizations/institutions.
Organization of this Document

This system manual is broadly divided into the following:
• Know Your SARVAM UCS - introduces the system features and benefits, describes the application
scenarios, hardware overview and the interfaces supported.
• Installing the ETERNITY LENX - gives step-by-step instructions for preparing for and installing the
ETERNITY LENX Platform in general, like setting up the main distribution frames for the wiring, the safety
measures for protecting the system and persons handling the installation and maintenance. It also
provides step-by-step instructions for installation, inserting the cards, connecting the cables and powering
the system.
• Installing the ETERNITY MENX - gives step-by-step instructions for preparing for and installing the
ETERNITY MENX Platform in general, like setting up the main distribution frames for the wiring, the safety
the system.
• Installing the ETERNITY GENX - gives step-by-step instructions for preparing for and installing the
ETERNITY GENX Platform in general, like setting up the main distribution frames for the wiring, the safety
the system.
• Configuring ETERNITY GENX - describes the basic configuration to select the application you wish to run
on the ETERNITY GENX platform.
• Configuring SARVAM UCS - contains description of the different tools and options available to configure
SARVAM UCS. It provides detailed description of how to configure the various extension and trunk port
types - SIP Extensions, VoIP-SIP, T1E1PRI, E1FO, CO, Radio, Data, Mobile, BRI, E&M, SLT, DKP, ISDN
Terminal and Magneto, Virtual Extensions - supported by SARVAM UCS.
• Features and Facilities - describes in detail, each feature and facility offered by the SARVAM UCS. This
includes a description of the feature/facility, how it works, and how to program the feature/facility.
The feature description is arranged alphabetically by Feature Name to make it easy for you to locate the
description you want to look up.
• System Maintenance - provides instructions for back-up, generating reports and debugging.

How to Read this System Manual
This document is organized in a manner to help you get familiar with the ETERNITY GENX Platform and SARVAM
UCS SME Application, learn how to install the platform, load application on platform, connect it to the various
networks and interfaces, start the system and use the features.
This System Manual is presented in a manner that will help you find the information you need easily and quickly.
You may use the table of contents and the Index to navigate through this document to the relevant topic or
information you want to look up.
Cross-references are provided in blue font with hyperlinks. You can look up the source by clicking the links.
Instructions
The instructions in this document are written in a numbered, step-by-step format, as follows. Each step, its outcome
and indication/notification, wherever they occur, have been described.
Access Codes
Access codes are strings of digits dialed by an extension to
• Call another extension, Department Group,

• Grab a trunk line
• Use a Feature, like Call Block, Call Forward.
The Access Codes provided in the instructions throughout this document, are default access codes. It is possible to
change the Access Codes according to user requirement and preferences. Verify with the Installer/System
Engineer, if the default Access Codes have been changed, and use the codes programmed by the System
Engineer. For more information, read the topic “Access Codes” in this document.
Notices
The following symbols have been used for notices to draw your attention to important items.
Important: to indicate something that requires your special attention or to remind you of something you
might need to do when you are using the system.
Caution: to indicate an action or condition that is likely to result in malfunction or damage to the system or
your property.
Warning: to indicate a hazard or an action that will cause damage to the system and or cause bodily harm
to the user.
Tip: to indicate a helpful hint giving you an alternative way to operate the system or carry out a procedure,
or use a feature more efficiently.
Terminology used in this System Manual

The technical terms and Acronyms used in this Manual are standard terms, commonly used in the
telecommunications and data communications industry. Considering the broad group of intended users of this
manual, wherever possible, use of jargon has been avoided.

Acronyms have been defined in the text and a list of the same is appended.
Some of the terms specific to this Manual that you will encounter are defined below:
The words 'UCS', ‘SARVAM UCS’ and ‘System’ are used interchangeably and synonymously to mean Unified
Communication Server.
• IP Phone: a phone that uses Voice over IP (VoIP) Technologies for placing and transmitting telephone
calls over an IP network, such as the Internet.
• SIP Extensions: any SIP-enabled device — a Matrix UC Client (Matrix VARTA WIN200, Matrix VARTA
ADR100, Matrix VARTA AMP100), an IP-phone, an Extended IP-phone, a Soft phone or an Analog Phone
Adapter — registered with the system, from which you can make/receive calls to any extension or
external number.
• SIP Trunking: VoIP or streaming media service based on Session Initiation Protocol (SIP) by which
Internet Telephony Initiation Protocols (ITSPs) deliver telephonic services and Unified communications to
the customers equipped with SIP-based System.
• Port: the physical interfaces on the cards for trunk lines and extension lines.
• Extension: it is the port of the system to which a telephone instrument (DKP/SLT/ISDN/SIP) is connected.
• Mobile Extension: a mobile/landline phone used as a remote extension of SARVAM UCS. You can
access all the features of an extension of SARVAM UCS from the mobile/landline phone.
• RUIM Card: Removable User Identity Module is a card that supports the CDMA services just like the SIM
card that supports GSM services. In this manual, consider SIM as RUIM, if CDMA Mobile Card is installed
in your system.
• Station: same as extension.
• Service Provider: the providers of telecom network lines/Internet - POTS, PSTN, CDMA, GSM, ISDN
PRI, ISDN BRI, and Internet Telephony Service Providers (ITSP).
• CO trunks: two-wire trunks, that is, analog trunk lines from the POTS network.
• System Administrator Commands/SA Commands: number strings dialed from the System
Administrator access/mode to operate features or set/cancel features for other extensions.
• System Commands/SE Commands: number strings dialed from the System Engineer access/mode to
program the system features/functions.
• CO Network: the public telephone exchange.
• CO Lines: the lines subscribed from the CO Network. These may be Two-wire Trunk Lines, ISDN BRI,
ISDN PRI, etc.
• Digital Key Phone (DKP): refers to EON, the proprietary digital key phone of Matrix that can be
connected with the system. The term 'Digital Key Phone' refers to all models of EON.
• Single Line Telephone (SLT): any standard two-wire telephone attached as extensions of the system.

• Called party/Callee: the person to whom the call is made.
• Calling party/Caller: the person who makes a call.
• Enterprise Application/Features: pertaining to the general and special telephone and call management
features required by business establishments, public and private organizations.
• Internal numbers: same as extension numbers.
• External Numbers: numbers of parties/individuals outside the UCS network. The unique number string
given to subscribers of PSTN, PLMN, ITSP, etc.
• Internal Calls: calls made from and received by one extension to another extension of the SARVAM UCS.
• External Calls: calls made by users of SARVAM UCS to subscribers of PSTN, PLMN, ITSPs, etc.
• Hospitality Application/ Features: pertaining to the special telephone and guest/patient management
features required by accommodation establishments like hotels and hospitals.
Using this Manual, we hope, you will be able to set up, operate and make optimum use of this feature packed
Unified Communication Server. If you encounter any technical problems, please contact your Dealer/reseller or the
Matrix Customer Care.

CHAPTER 3 Know Your SARVAM UCS
Introduction
When ETERNITY GENX platform is used with SARVAM UCS application, it acts as a fully hosted and managed
Unified Communication Server. It delivers the convergence of voice, data, wired and wireless communications for
small and medium sized businesses.
SARVAM UCS offers UC features, Voice over IP Integration, wireless communications, voice mail, computer
telephony integration and switching functions.
This Unified Communication Server creates a collaborative environment by:
• Integrating UC Features like unified messaging, video conferencing etc.

• Addition of more features and functions
• Simplifying management
• Providing services and support
• Centralizing troubleshooting to help solve problems quickly
• Securing IP communications for the mobile workforce
Features and Benefits

Features - UC Features like video calling, instant messaging, IM to SMS, Email to SMS etc, Caller ID, Call
Forwarding, Pushkit support, Conference calling and voice messaging.
SIP Trunking - SIP Trunking to Internet Telephony Service Providers. SARVAM UCS allows users to make or
receive SIP Trunk calls.
Trunk Interfaces - A variety of network trunk interfaces including CO/Two-wired trunk, ISDN BRI, ISDN T1E1PRI,
E&M, Mobile and SIP Trunks.
Extensions - Provides support for a range of extensions — Digital Extensions, Analog Extensions, Radio
Extensions, Virtual Extensions and SIP Extensions using Voice over IP technology to provide sophisticated voice
performance for new and growing businesses.
UC Clients - Provides support for Matrix UC Clients — VARTA WIN200 Desktop UC Client, VARTA ADR100 and
VARTA AMP100 Mobile UC Clients.
Telephones - A variety of telephones including analog, digital, IP hard and soft phones (wired and wireless) to
provide an appropriate desktop or device telephone for every need.

Advanced Call Routing - Various options for routing incoming and outgoing calls on trunks are provided.
Q-Sig Networking - With Q-Sig, you can network SARVAM UCS with another UCS or any other ISDN QSIG
supported system to expand the system resources. This provides the feature transparency between the systems to
function as a single unit.
Universal Connectivity
SARVAM UCS supports UC communication which includes Instant Messaging, Call Management, Smart Directory
and Video Calling using a variety of IP phones such as the Open SIP Phones and Matrix UC Clients — VARTA
WIN200 Desktop UC Client, VARTA ADR100 and VARTA AMP100 Mobile UC Clients.
SARVAM UCS offers Universal Connectivity, working with all major telecom interfaces: VoIP, POTS, ISDN BRI and
PRI (T1/E1), GSM/3G/CDMA, E&M, Radio and Magneto. So, you have access to multiple telecom networks on a
single platform. The system's intelligent Least Cost Routing logic diverts your calls through the appropriate network,
ensuring least possible call cost.
SARVAM UCS also supports Video Conferencing over IP using any standard SIP Video Conferencing device.
Video Conferencing and data connectivity is also supported over ISDN interface.
Flexibility and Scalability

SARVAM UCS is designed to support both pure IP and Hybrid IP solution. Non-IP interfaces can be added by
inserting non-IP homogeneous and Mix Port Slave Cards in Universal Slots. The Hybrid Cards offer all possible
communication interface options to match your communication requirement.
The system is also designed to provide very high level of flexibility and scalability to meet your future
communication needs. The flexible licensing allows you to start with desired number of SIP Users, VoIP Channels
and VARTA UC Users. If required, you may increase the number of users or channels supported by purchasing the
respective licenses.
The Universal Slot platform and the modular design of the Cards allow you to start with the minimum required
configuration and expand the system capacity later, by adding more Cards to the Universal Slots. So, you can
invest progressively in scaling up the system as the communication needs of your organization grow.
Cost Saving and Productivity Enhancing Features

Key features like Multi-party Conference, Auto Attendant, Remote programming, SMDR Buffer with a large
capacity, that normally warrant additional investment in most other brands, are built into the system.
Intelligent features like Auto Attendant, CLI/DDI based Routing and Dial by Name ensure efficient call management
and prompt response to callers.
Least Cost Routing and Call Budgeting help reduce communication cost and enhance productivity.
The system can route a VoIP call to GSM, CO or T1E1PRI port. In the same way, a call on a T1E1PRI port can be
routed to VoIP, GSM or CO ports. Further, you can select Fixed or Least Cost Routing to route outgoing calls.
SARVAM UCS can handle calls on all ports simultaneously, allowing full traffic on all ports.
SARVAM UCS can work as an adjunct to your existing telephony infrastructure, as a Gateway, saving you the cost
of equipment replacement, wiring and installation, while giving you Universal Connectivity and a host of intelligent
features.

Redundancy
To reduce down time and provide uninterrupted communication, the ETERNITY LENX/MENX supports redundancy
option for the two cards that are critical to its functioning: The Power Supply Card and the CPU Card. There are two
cards of each on the platform. When the active card fails, the standby card takes over.
Hot Swap
With the Hot Swap feature you can remove a card and insert it back without switching off the system. So, you can
replace a faulty card with a functional one without affecting the functioning of the system. The ETERNITY LENX/
MENX supports Hot Swap.
UC Features
• Set/View Presence
• Presence based call controlling (make call/reject call)
• Video Calling
• IM using VARTA WIN200, VARTA ADR100 and VARTA AMP100
• IM using third party SIP Phones/soft clients
• IM to SMS and vice versa using SMS Server
• SMS campaigning using SMS Server and SMS Gateway(in collaboration with third party SMPP
application)
• Smart Handover from extension to mobile
• Smart directory access using VARTA WIN200, VARTA ADR100 and VARTA AMP100 for easy and quick
access to the extensions and other contacts
• Mobile and Remote workers support
• Outlook integration using CTI Interface
• Auto-attendant with configurable call-flow
Other Key Features

• Account Codes
• Auto Black List IP Address for Web access, VoIP access, etc.
• Automatic Call Distribution
• Call Accounting Software interfaces supported
• Call Billing
• CAS Interface
• Class of Service
• CLI Based Routing
• Closed User Group
• Conference upto 21 party
• Emergency Call Detection and Reporting
• Firmware and Configuration zip file upload using browser
• Hospitality features with Front Desk Web Interface (for Hotels and Hospitals)
• Hot Desking
• Internal SMDR Report buffer
• Least Cost Routing
• Logical Partitioning for restricting toll bypass
• PMS Interface
• Priority
• Q-Sig
• Remote(ISDN) Programming
• SNMP interface for error/event notification and status check
• Syslog (for easy maintenance)

• System Activity Log
• System Fault Log
• Toll Control
• Trusted IP Address to avoid unauthorized access
• Universal Slots
• Voice Applications
• Voice Mail
• Web-based Programming(HTTP/HTTPS)
• White List IP Address

Applications of SARVAM UCS
The Matrix SARVAM UCS can be deployed in small to large enterprises and institutions: manufacturing units,
corporate offices, banking and financial institutions, software firms, shopping malls, hospitals, hotels-motels, in
power line carrier communication of electric utilities, call centers, in institutions and, power line carrier
communication (PLCC) networks.
Illustrated in the following are various scenarios where the SARVAM UCS finds application.
Enterprise Application
UC Applications
If connecting to the Public IP Network,
Plug one end of the RJ45 Ethernet cable into the WAN Port of the system and the other end into the Broadband
Router/Modem.

If connecting to a Private Network (Behind a NAT Router),
Plug one end of the RJ45 Ethernet cable into the WAN Port of the system and the other end into the LAN Switch/
Hub.

Video Conferencing
Mobility

Hotel Application
PSTN
Administration
Phones
Front Desk
User Wizard
Hotel Reports
SARVAM UCS
SMDR-Posting
CAS
Guest Room
Phones
Activity Log
PSTN
Administration
Phones
PMS Interface PMS

SARVAM UCS
Hotel
Reports
SMDR - Posting
Guest Room
Phones
CAS
Activity Log

Closed User Group Application
Power Line Carrier Communication Application

Universal Gateway Application

Hardware Overview
ETERNITY LENX
The Enclosure
The enclosure of ETERNITY LENX consists of fixed and universal slots. The fixed slots are occupied by specific
cards - Power Supply Cards and CPU Cards - and cannot be changed, whereas in the universal slots, you can
install any of the various card.
The slot connectors are located on the motherboard on the backplane of the enclosure. Each slot has guide rails for
inserting the cards.
ETERNITY LENX has two racks and a total of 27 Universal slots. The first rack has a total of 15 slots. The first slot
from the left is a fixed slot and the last two slots to the right are fixed slots and the remaining 12 are universal slots.
The second rack has a total of 16 slots. The first slot from the left is a fixed slot and the remaining 15 are universal
slots.
ETERNITY LENX can be wall mounted, rack mounted or placed on a table. You can also affix wheels to the
system; to move it like a trolley. After you have inserted the required cards, you must affix the front top cover.
Illustrated below are design of the enclosure and the position of the slots in ETERNITY LENX.
The Cards
ETERNITY LENX houses the following Cards:
1. CPU Card
2. Power Supply Card - PS48V
3. SLT Card
4. CO Card
5. CO+SLT Card
6. DKP Card
7. E&M Card
8. BRI Card

9. T1E1PRI Card
10. GSM/3G Card
11. Magneto Card
12. Radio Card
13. Data Card
ETERNITY MENX
The Enclosure
The enclosure of ETERNITY MENX has fixed and universal slots. The fixed slots are occupied by specific Cards -
Power Card and the CPU Card - and cannot be changed, whereas in the universal slots you can install various
Slave Cards. ETERNITY MENX has 16 universal slots.
Inside the enclosure of ETERNITY MENX are slot connectors located on the motherboard on the backplane of the
enclosure. Each slot has guide rails for inserting the Cards.
Illustrated below are the design of the enclosure and the slots positioning in ETERNITY MENX.
Power Supply CPU Card

Card
Power Supply CPU Card

Card
ts
al Slo
nivers
16 U
The Cards
ETERNITY MENX houses the following Cards:
1. CPU Card
2. Power Supply Card - PS48V
3. SLT Card
4. CO Card
5. CO+SLT Card
6. DKP Card
7. E&M Card

8. BRI Card
9. T1E1PRI Card
10. GSM/3G/CDMA Card
11. Magneto Card
12. Radio Card
13. Data Card
ETERNITY GENX
The Enclosure
The enclosure of ETERNITY GENX has fixed and universal slots. The fixed slots are occupied by specific Cards -
Power Card and the CPU Card - and cannot be changed, whereas in the universal slots you can install various
Slave Cards. ETERNITY GENX has 12 universal slots.
Inside the enclosure of ETERNITY GENX are slot connectors located on the motherboard on the backplane of the
enclosure. Each slot has guide rails for inserting the Cards.
Illustrated below are the design of the enclosure and the slots positioning in ETERNITY GENX.
ETERNITY GENX
The first slot from the left is reserved for the Power Supply Card and the first slot from the right is reserved for the
CPU Card.
The Cards
The ETERNITY GENX houses the following types of cards:
1. Power Supply Card - AC Input Card or DC Input Card
2. CPU Card
3. SLT Card
4. CO Card
5. DKP Card
6. CO+SLT(Combo) Card
7. DKP+SLT(Combo) Card
8. CO+DKP+SLT(Combo) Card
9. Intercom Line Card
10. E&M Card
11. E1FOPRI Single Card
12. BRI Card
13. T1E1PRI Single Card

14. Magneto Card
15. Radio Interface Card
16. GSM/3G/CDMA Card
For an at-a-glance view of the Cards available with ETERNITY GENX Platform when used with SARVAM UCS
Application, refer “Technical Specifications - SARVAM UCS”.

The Interfaces
The SARVAM UCS supports the following interfaces for connecting with different telecom networks, digital key
phones, standard telephones and other external devices.
The VoIP Interface

The Voice-over-IP (VoIP) Interface connects SARVAM UCS with the internet or the private IP network. It enables
the SARVAM UCS to handle incoming and outgoing calls over the IP network. It offers great flexibility and mobility
to users for making or receiving calls across the world.
The VoIP Interface supports UC Clients, SIP Extensions and SIP Trunks.
With SIP Trunks users can make IP calls using the SIP Server of the Internet Telephony Service Providers (ITSPs).
Make sure the NX DBM VOCODER64 module is installed and the channel licenses — SARVAM
VOCODER CHNL4 or SARVAM VOCODER CHNL16 — purchased as per your requirement.
The in-skin Registrar Server allows any SIP enabled device like a UC/SIP Soft Client or an IP-Phone to be
registered with it and function as the 'SIP Extension' of SARVAM UCS. The SIP Extension users can make and
receive calls to any extension user of the SARVAM UCS as well as any external numbers over VoIP, PSTN, GSM
and other trunks. With SIP Extensions, organizations can communicate and stay connected at the lowest cost
without any geographical restrictions.
The VoIP Interface supports adaptive jitter buffer for reducing delay and improving speech quality.
The key features of the VoIP Interface are:
• Up to 99 SIP Trunks - for Proxy as well as Peer-to-Peer (non-Proxy) calls.
• Up to 9991 SIP Extensions.
• Upto 5502 simultaneous IP-to-IP calls (with Direct RTP/ RTP Relay).
• 64 simultaneous IP-to-IP call legs / IP-to-TDM calls (with Transcoding) for one NX DBM VOCODER64
module installed.
• 128 simultaneous IP-to-IP call legs/ IP-to-TDM calls (with Transcoding) for two NX DBM VOCODER64
modules installed.
• Selectable Network Assignment (Connection Type) - Static IP, DHCP, PPPoE.
• Selectable DNS - Automatic and Static.
• Dynamic DNS for SIP devices in public network.
• STUN.
1. ETERNITY LENX/MENX supports 2000 SIP Extensions.

2. This is the total capacity supported by SARVAM UCS system.

• TCP and UDP NAT Keep Alive.
• VLAN
• Symmetric RTP Selection.
• MAC Address Cloning option.
• Fax over IP - T.38 (UDPTL), T.38 (RTP) and Pass Through.
• Send CLI Option for outgoing calls
• Selectable DTMF - RTP (RFC 2833), SIP Info, InBand
• Flash Detection using SIP INFO and RFC2833.
• Broad Voice Codec Selection: G.723, G.729 AB, GSM FR, iLBC - 30 ms, iLBC - 20 ms, G. 711 µ-Law, and
G. 711 A-Law.
• Quality of Service - SIP DiffServe/ToS, RTP DiffServe/ToS
• VoIP Silence Detection and Disconnection.
• Voice Mail Subscription on SIP Extensions.
• Busy Lamp Field Subscription on SIP Extensions and SIP Trunks.
• Upto 10 Call Appearances on SIP Extensions.
• Registration of SIP Extensions from 3 different locations and Shared Call Appearance.
• Sharing the same SIP Extension number among a maximum of 3 different SIP devices using Share Call
Appearance functionality.
• Programmable Loop Current
SIP Trunks
The ETERNITY GENX3 platform when used with SARVAM UCS application supports a maximum of 99 SIP
Trunks, allowing you to subscribe to as many as 99 different Internet Telephony Service Providers (ITSP) or peer-
to-peer connectivity with multiple end SIP devices.
Make sure the NX DBM VOCODER64 module is installed.
SIP Extensions
ETERNITY GENX platform when used with SARVAM UCS application supports 9994 SIP Extensions which can be
registered with the in-skin SIP B2BUA of CPU Card.
3. ETERNITY LENX/MENX also supports 99 SIP Trunks.

4. ETERNITY LENX/MENX supports 2000 SIP Extensions.

Any SIP-enabled device like an IP-phone, a Softphone, analog phone adapter, can be registered with the CPU
Card and function as the 'SIP Extension' of the ETERNITY GENX platform when used with SARVAM UCS
application.
The SIP Extensions function in the same way as other extensions of the SARVAM UCS. SIP Extension users can
make and receive calls from and to other extensions of System and external numbers over PSTN, GSM, VoIP and
E&M lines5. You can also connect the Standard and Extended IP Phones offered by Matrix as SIP Extensions.
Make sure the NX DBM VOCODER64 module is installed and the channel licenses — SARVAM
VOCODER CHNL4 or SARVAM VOCODER CHNL16 — purchased as per your requirement.
A SIP Extension can be registered with SARVAM UCS from three different locations. This helps organizations
overcome geographical distances and reduce call costs.
Five SIP Extensions are free, for additional extensions you require an IP Subscriber license. To know more
about Licensing requirements and how to acquire and activate a license key, see the topic “License
Management”.
You can connect ETERNITY GENX platform when used with SARVAM UCS application to the IP network, which
may be Public Internet or a LAN.
The ISDN PRI Interface

The ISDN PRI Interface enables SARVAM UCS to be connected to digital networks over T16 and E17 carrier lines.
On T1 carrier lines, the Interface supports the following signaling types:
• PRI
• Robbed Bit Signaling (RBS)
• Q-Signaling (QSIG)
• E&M
On E1 carrier lines, SARVAM UCS supports the following signaling types:
• PRI
• Channel Associated Signaling (CAS)
• E&M
The ISDN T1E1PRI Interface supports the following features:
• Terminal (TE) mode and Network (NT) Mode

• Voice Calling
• Video conferencing and data connectivity
5. Only if there are no restrictions on calls from VoIP to other Public Networks in your country. If the telecom regulations of your coun-
try prohibit call traffic between the public telephony networks and IP networks, you must configure Logical Partition in your system.
To know more, see “Logical Partition”.
6. T1 PRI (T-Carrier) offers 23 Bearer Channels and one Signaling Channel (23B+D). It is used in North America and Korea.
7. E1 PRI (E-Carrier) offers 30 Bearer Channels and two Signaling Channels (30B+D). It is used in all countries, except North
America, Japan and Korea.

ISDN BRI
The ISDN BRI Interface enables SARVAM UCS to be connected to ISDN BRI Lines and connect ISDN BRI
compatible devices with the SARVAM UCS.
The ISDN BRI Interface has the following features:
• Signaling types - 2B+D Signaling

• Terminal (TE) mode
• Video conferencing and data connectivity
Depending on the requirement, each BRI Port can be configured in the TE/NT mode.
It is possible to feed power from the SARVAM UCS to the terminal equipment connected to the SARVAM UCS (on
its BRI port configured as NT).
The Mobile Interface

The Mobile Interface enables the SARVAM UCS to be connected to GSM/3G/CDMA network operators worldwide.
For example, the Frequency Band supported by the GSM networks varies across countries. In some countries, the
network operators may use 850MHz while network operators in some countries may use 1900MHz, in some
countries network operators may use both frequency bands.
SARVAM UCS's Mobile Interface supports full Quad-Band Operation (GSM850, 900, 1800, 1900MHz) for world-
wide use, for Global, Inter and Intra country roaming.
The Mobile Interface supports the following features:
• CDMA, GSM 2G, GSM 3G network support.

• Selectable GSM Frequency Bands - 900, 1800, 1900, 850 + 1900, 900+1800 MHz.
• Selectable CDMA Frequency Bands - 800MHz CDMA cellular; 1900MHz PCS.
• Programmable Network Selection for GSM - Manual and Automatic. For CDMA, only Automatic is
applicable.
• Programmable Network Operator Codes in order of priority (from 1 to 9) in case of Manual Network
Selection.
• Programmable Speech Tx Gain
• Programmable Speech Rx Gain
• Selectable Incoming Call Modes - Allow, Ignore, Reject.
• SIM Card protection with a Personal Identification Number (PIN)8.
• Single Rooftop (RT) antenna for 'High Gain'
If CDMA module is installed in your system, following features and facilities will not be applicable.
• Band Selection
• CLIR
• BCCH Selection
• Emergency Number
• Network Selection
• Preferred Network Mode
• RCOC (Route Call to Original Caller)
8. Not applicable for CDMA Mobile Card.

• SIM PIN
• SIM Card Balance and Recharging
• SIM Recharge
• SMSC
• SMS Parameters
• SMS on No Reply
• SMS Server
• SMS Gateway
If CDMA module is installed in your system, DTMF Detection might not work properly.
The SARVAM UCS Mobile Interface does not offer GPRS features, Fax and Data services, and network
supported services, except CLIR and USSD.
The CO Interface
The CO Interface enables the SARVAM UCS to be connected to the POTS Network. The POTS Networks across
the world support various standards and differ in features. For example, some networks support Caller ID
Presentation using DTMF signaling, while some support Caller ID Presentation using FSK signaling; some
networks offer 600 Ohms Impedance, while others offer complex impedance.
SARVAM UCS's versatile architecture allows it to be connected to such networks differing in their characteristics.
The CO Interface supports following features:
• Programmable AC Impedance - 600, 900 and various complex impedances

• Answer Supervision/Polarity Reversal
• Selectable Disconnect Supervision - Polarity Reversal, Open Loop Disconnect
• Selectable Caller ID Presentation - DTMF, FSK
• Programmable Dialing method - Pulse/Tone (with programmable Pulse Ratio/DTMF On-Off period)
• Programmable Speech Tx Gain
• Programmable Speech Rx Gain
• Programmable Disconnect Tone Sensing
• Programmable Flash Timer
The Radio Port Interface

The Radio Interface Card offers connectivity to the Radio transceiver devices/Combat-Net Radio devices. The
Radio Interface Card (RIC) adds the two-way Radio functionality in SARVAM UCS, wherein the speech can be
transmitted as well as received by the radio device.
Radio Transceiver or CNR radio devices interfaced with the Radio port can be operated on High Frequency (HF),
Very High Frequency (VHF) or Ultra High Frequency (UHF).
Using the Radio Interface, the user can:

• Set the frequencies for receiving and sending audio messages.
• Playback the audio received over the air through the Speaker.
• Send the audio message over the air through the Microphone.
• Activate radio transmission over air using Push-to-Talk (PTT) button.
The Radio Transceivers remains in receiving (Rx) mode, so that the audio messages broadcasted over the air can
be heard.

The E&M Interface
The E&M Interface of the SARVAM UCS supports analog trunking to connect various communication equipment
telephone switches, Routers, Leased Lines, etc.
Often, E&M connectivity is used to expand the system capacity (by connecting a second system with the main
system) or to connect two or more remotely located Systems, forming a network of Systems.
The E&M Interface can be used for the following applications:
• Power Line Carrier Communication (PLCC) Networks, where several EPAXs are connected with each
other through E&M tie lines. Refer “PLCC-An Introduction” to know more.
• Closed User Groups, where several Systems are connected with each other through E&M tie lines9.
• System expansion, where two Systems are connected with each other with E&M tie lines.
• Connecting remote Systems/Satellite equipment over E&M tie lines
Also, refer the topics “E&M Connectivity” and “E&M Feature Template” to know more.
The E&M Interface can be programmed to provide Trunk Interface, a Subscriber (Station) Interface or both, as a
Tie Line with the dual personality of a Trunk and a Subscriber.
The E&M Interface of the SARVAM UCS has the following features:
• Programmable Orientation Type - trunk, station, tie-line.

• Selectable E&M Interface Types - IV and V.
• Selectable Speech Interface (Audio Interface)10 - Two-wire and four-wire.
• Selectable E&M Trunk Seizure Type11 - Immediate, Immediate + Wink, Seizure Pulse, Seizure Pulse +
Wink, Express, and Compander Control Signal.
• Selectable Address Signaling - Pulse dial (Pulse 10PPS, Pulse 20PPS) and Tone Dial (DTMF).
The Magneto Interface

Magneto Telephones12 can be connected to the Magneto Interface of SARVAM UCS. Magneto telephones are
widely used by the defense establishments as field phones in front lines, and by other establishments such as
railroad companies (signaling emergencies, crossings, etc.), electric utilities, pipeline companies, who need to have
their networks at places that are too remote to be serviced by public telephone networks.
SARVAM UCS can land calls from magneto field telephones on the extensions (SIP, SLT, DKP, ISDN Terminal) of
the SARVAM UCS and place calls from the extensions of the SARVAM UCS on magneto telephones.
To know more about how the Magneto Card works, refer the topic “Configuring Magneto Interface”.
9. The Systems in a Closed User Group can be connected over ISDN T1/E1 Lines as well. Refer the topic “Closed User Group
(CUG)” to know more.
10. The number of wires used to transmit audio signals.
11. This is the line protocol that defines how the equipment seizes the E&M trunk. Also, referred to as Start Dial Supervision Signaling
Protocol.
12. A magneto telephone is a local battery telephone set, in which signaling current is provided by a magneto hand generator. The
hand generator, commonly referred to as 'crank', is located on the right hand side of the telephone set and is turned to produce
energy to ring other phones or to signal the CO. The magneto, also called the generator, is used to convert the mechanical motion
via the crank to produce sufficient energy to ring other phones or to signal the CO.

The Digital Key Phone Interface
The Digital Key Phone Interface of SARVAM UCS is for the purpose of connecting the proprietary Digital Key
Phones of Matrix Comsec of the EON series, the proprietary Direct Station Selection (DSS) Consoles and the PC-
based phone, EONSOFT, and the proprietary digital turret, EON7413, with SARVAM UCS.
The Digital Key Phone Interface has the following features:
• Programmable Call Capacity - up to 10 Call Appearances/Call Loops on a single DKP.

• Selectable Headset and Speaker (Hands-free) modes.
• Selectable Auto Answer mode.
• Programmable Headset Speech Tx and Rx Gains.
• Programmable Hands-free (Speaker) Speech Tx and Rx Gains.
• Programmable Key Click Volume levels.
• Programmable Key Maps for Operator, Executive, Hotel Attendant and Guests.
• 4-Level Adjustable Back light and Contrast Controls
• Selectable Ring Types.
• Ringer Volume Control.
• Programmable Ringer Modes - Ring Immediately, Ring if Idle, Ring after Delay, Ring Off.
• Ring On Speakerphone and Ring on Headset options.
• Language Support - English, French, German, Portuguese, Spanish.
The Single Line Telephone Interface

The Single Line Telephone (SLT) Interface allows any standard, two-wire, analog single line telephone instrument -
rotary, pulse-tone, cordless, feature phones with or without Calling Line Identification - to be connected to the
SARVAM UCS as extension phone.
The SLT Interface has the following features:

• Selectable Caller ID Presentation - DTMF, FSK
• Programmable CLIP Digit Pad Count (0-9).
• Programmable Ring Type - Trapezoidal, Sinusoidal, Low Trapezoidal, Low Sinusoidal.
• Programmable AC Impedance - 600, 900, 350 + (100 || 0.21uF), 220+ (820 || 120nF), 270 +
(750 || 150 nF)
• Selectable Answer Signaling.
• Selectable Disconnect Signaling - Polarity Reversal, Open Loop Disconnect.
• Programmable Speech Rx Gain.
• Programmable Speech Tx Gain.
• Programmable Flash Timer.
• Programmable Loop Current.
• Fax machine connectivity.
Voice Mail System (VMS)

The SARVAM UCS application supports a full-fledged, in-skin Voice Mail System module to provide mailbox facility
to all its extensions users. The Voice Mail System also forms the basis of other features like Conversation
Recording and Call Taping.
Each Mailbox has the capacity of storing 15,000 voice messages. The maximum size of each Mailbox is 60,000
minutes. By default, the size of each Mailbox is set to 5 minutes. The maximum Message Length for each Mailbox
13. Check for availability.

is 9,999 seconds. By default, the Maximum Message Length for each Mailbox is set to 15 seconds.
The VMS Module in the SARVAM UCS is an optional module. It must be purchased separately. The VMS in
ETERNITY GENX is delivered as a module. The default factory fitted 8GB Pen Drive on the CPU Card contains the
VMS configuration files, voice messages for prompts and greetings. The Pen Drive is also the storage device for
mailbox messages.
If required, you may use a Pen Drive of upto 64GB by replacing the factory fitted pendrive with a new one.
The key Voice Mail and Auto Attendant features supported by SARVAM UCS's Voice Mail System are:
• Programmable Mailbox Size.

• Programmable Message Length.
• Welcome Greetings according to the time of the day.
• Different voice greetings for different time zones.
• Special greetings for holidays.
• Five call transfer types: none, blind, wait for ring, wait for answer, and screened.
• Dial by extension.
• Dial by name.
• Personalized greetings for each mailbox.
• Individual mailbox size.
• Call forward to voice mail.
• Message forwarding.
• Distribution lists.
• Broadcast message.
• Message Wait Notification.
• Redirecting messages.
• Message Wait Notification via Email and Call.
SARVAM UCS's Voice Mail System also forms the basis of other features like:
• Conversation Recording
• Call Taping
• Voice-guided Wake-up Calls and Reminders
• Message Wait Notification
• Call Transfer to Mailbox
• Call Forward to Voice Mail
• Department Calls - Mailbox for Department Groups
The Data Port Interface

The Data Card supports four Ethernet 10/100mbps interfaces. Ethernet data coming to ports can be mapped to
2Mb streams. It can map data to 60 channels of E1 interface. The remaining channels of E1 can be used for voice
applications. The Data Card has a 4-port Ethernet switch on board, which can aggregate multiple data streams to
PCM streams of the system.
Computer
You can connect a standalone computer with the SARVAM UCS over the RS232 serial Communication Port/s14 of
the system.
14. ETERNITY GENX supports two Communication Ports — COM and USB to COM. ETERNITY MENX/LENX supports only one
communication port —USB to COM. For details, see “Communication Ports”.

You can also connect SARVAM UCS to a standalone computer or to a LAN Switch over the LAN/WAN Port of the
SARVAM UCS.
PC connectivity is required to:
• access the web-based configuration interface JEEVES.

• set up and run software applications such as Property Management Software (PMS), Call Accounting
Software (CAS) in the Hotel Application from Web JEEVES interface.
• to run the Front Desk User Wizard of the Hotel Application.
• capture and download Station Message Detail Records (SMDR): SMDR reports, SMDR Online and SMDR
Posting.
• capture and download System Activity Log and System Fault Log, and Hotel Motel Activity Log in the Hotel
Application.

CHAPTER 4 Before You Start
Before you begin the installation of the system, make sure that the required telecom wiring has been done.
The number of extensions you require and their location will determine the type of cabling you require on your
premises.
We recommend that you plan the wiring and the installation of the system according to your current and expected
future requirements.
Before you begin to install and set up the hardware of the system, make sure you have the following items:
• A Main Distribution Frame (MDF)

• A suitable location to install the Main Distribution Frame and the system.
• Cables for trunk lines and extensions.
• The Cards supported.
• One or more Single Line Telephone or Digital Key Phones for testing.
• Power supply.
• One or more active Two-wire trunk lines for test calls.
• A modem for the ISDN T1E1PRI line.
• An NT1 termination device for the ISDN BRI line.
• Appropriate cables and connectors to set up and test the Ethernet interface of the system and the LAN
connection.
• A standalone PC or a PC connected in a LAN that can PING the system’s IP Address.
• A SIM card to test mobile network connectivity.
• A SIP Account to test VoIP connectivity.
Well begun is half done; plan your hardware installation well.
The Main Distribution Frame (MDF)

The MDF connects outside telephone lines coming from the local exchange, on one side and the internal (System)
lines on the other.
In simple form, the MDF is a special metallic frame designed and constructed with columns of receptacles to firmly
hold the termination modules for the trunk and extension cables.
The cables or trunk lines to/from the Public Telephone Exchange terminate on the line side and cross connections
(jumpers) run to the opposite (System) side of the MDF. From those terminals, a multi-core cable runs from a
second set of terminals into the System.

A multi-core cable runs from the System into the MDF. From the distribution frame, the smaller cables run into each
individual extension telephone outlet or socket (RJ11 or RJ45).
In a multi-storied building or on a widely spread out premises, it is common to have more than one distribution
frame, called the Intermediate Distribution Frame (IDF) on each floor, to provide the connection between the MDF
and the individual telephone wiring. IDFs function as wiring points to gather and distribute wiring. IDFs are used
when a large number of extensions are to be connected and the wire runs extend over hundreds of feet; hence the
distance is too great to economically terminate every extension individually to the MDF.
• Select a suitable MDF (and IDF, if required) with the standard lead-in cable termination KRONE modules.
• Ensure that the MDF complies with the local building telecom wiring Guidelines, Rules and Regulations.
• Select an appropriate site to install the MDF.
Location of the MDF

• The MDF is normally installed inside the building in a location and position which is free from the ingress of
dust and moisture, and which is not subject to damp or humid conditions.
• This also applies to MDF installed outside the building. It must be protected from exposure to weather
conditions, dust, dampness and humidity
• Do not install the MDF in any of the following locations:
• In washing or toilet facilities, boiler/plant/machine rooms or any area subject to corrosive fumes and
fluids;
• In fire escape stairways;
• Within a cupboard containing a fire hose reel;
• Within any refrigeration room or sauna heater room;
• Near any water feature or water body like fountains, sprinklers, a bath, shower or other fixed water
container, a swimming pool, paddling pool, spa pool or tub; or any area where hosing down operations
are carried out.
• In a high voltage electrical switch room or near a heavy voltage transformer.
• The MDF should be robust and securely attached to a permanent building element such as a wall, floor or
column. Do not mount the MDF on movable elements such as hinged panels or wheeled trolleys.
• Provide adequate lighting in the place where the MDF is located.
• Provide adequate space around the MDF where any person is required to pass to enable safe and
convenient access to the MDF and ready escape from the vicinity under emergency conditions.
• Any room containing the MDF must not require the use of a tool, key, card, number pad or the like to exit
the room. Ensure a quick hurdle-free exit from such a room.
• The MDF or the enclosure in which it is located should have the provision for securing with a key, lock or
tool. External MDF should be adequately secured against vandalism and access by children or
unauthorized persons.
• The MDF enclosure should be designed so as to prevent access to live parts by unqualified persons and
should be free of exposed sharp edges.
• The MDF enclosure should be protected from insects and rodents.

Selecting the Installation Site for the system
The system maybe be mounted on a table or wall. Refer to the mechanical dimensions of the model you have,
when selecting the site for installation, and deciding whether to mount the system on a table top, or on the wall, or
on a rack.
Select an appropriate site to install the system taking into consideration the following recommendations and
precautions:
• The site of installation should be well-ventilated, moisture and dust free, and not exposed to direct sunlight,
heat, excessive cold or humidity.
• The site should be equidistant from all the extensions to simplify cabling network and reduce cabling costs.
• The system should be installed at a height of at least 3.5 feet from the ground. Installation at this height
makes preventive or corrective maintenance tasks easy.
• The system should be installed away from any source of electromagnetic noise such as any radio
equipment, heavy transformers, faulty electric chokes of tube-lights, any device having faulty coil, etc.
Read “Protecting the System and Yourself”.
Selecting Cables
• Select standard good quality telephone cables with 0.5 mm conductor diameter for the internal as well as
over-head cabling.
• Use twisted pair wires to reduce interference.
• Use separate cable conduits for electrical and telephone cables.
• The length of the cables must not be too long. They must have minimum number of joints. This will help
you detect cable faults easily.
• Maintain cable records so that cables and cross-connections on the MDF can be correctly identified and
connected. The records should be in a clear, legible and updatable format.
Selecting Extension Telephones

Select appropriate telephone instruments to be connected as extension phones. You may connect:
• any of the models of the proprietary Digital Key Phone (DKP) of the EON series.
• Any standard telephone instrument like rotary phone, Pulse/tone switchable push-button phone, Feature
phone or Cordless phone. So, you can also use your existing telephone instruments.
You can connect DKPs with DSS of the EON series for Operator/Receptionist/ Front Desk/Senior
management extensions.

Providing Power Supply Source
• ETERNITY LENX operates on 48VDC (-10% to +20%).
• The ETERNITY GENX/ MENX work with input voltages ranging between 100-240VAC or 48VDC (250W
for GENX and 500W for MENX).
• Arrange for a separate power point and switch, close to the system.
• Power supply for the system must be separate from other heavy electrical loads like Air-conditioners,
heaters, welding machines, electrical motors, etc.
Terminating Trunk and Extension Cables on the MDF

• Terminate the CO Trunk Line cables from the CO (public telephone exchange) and E&M cables into the
'Trunk Lines' side of the MDF using the punch tool for Krone modules.
• Terminate all the extension cables (connected to the wall sockets/outlets) into the 'Station Lines' side of
the MDF using the punch tool for Krone modules.
• Label the trunk and extension line cables for easy identification and keep a record of the trunk and
extension lines in an updatable format.
Where multiple wiring cabinets/distribution frames are used, label each frame and reference its number on
the corresponding outlet.
• Install Primary Protection modules with Gas Discharge Tubes (GDT) and fuses on entry points for all trunk
lines. This is to protect the system from heavy voltages from trunk lines and overhead stations.
The product warranty does not cover damages resulting from lack of primary protection on trunk lines.
• It is recommended that you also install Primary Protection modules with GDT and fuses on all Extension
lines, particularly off-premise extensions, and E&M ports.
For this, you are recommended to use the Primary Protection Module (PPM4) supplied by Matrix.

• A typical connection between a System and the MDF is illustrated in the figure below.
You are recommended to use the “Primary Protection Module - PPM4” supplied by Matrix.
MAIN BUILDING
PPM4
CO Lines
Distribution Frame
CO Lines
Extensions ETERNITY GENX

EARTH
ADJACENT BUILDING
PPM4
EARTH

Connecting Primary Protection Modules to Trunk and Overhead/Off-
Premises Lines
It is necessary to protect the system from heavy voltages entering the system from the Trunk Lines and overhead
stations.
The protection can be in the form of surge suppressor devices like Gas Discharge Tubes (GDT), MOVs, Fuses, etc.
Input Protection Modules (IPM)

Install IPM on the Krone Modules of the MDF. Input protection modules are for analog input channels to protect
against over-voltages that may be applied between any two input connectors or between an input connector and
the ground.
Primary Protection Module - PPM4

Matrix provides Primary Protection Modules (PPM) consisting of four PPM circuits. The PPM4 contains Gas
Discharge Tubes and Fuses.
The Gas Discharge Tube is an over voltage protection device. It has three terminals. It is connected parallel to the
CO Line or the overhead station cable. The third terminal is connected to a telecom earth. When the voltage
between any of the two terminals exceeds the permissible limit (general 150V), the gas in the device begins to
conduct and the terminals with the earth terminal. Heavy voltage passes to the earth instead of entering the
system, thereby protecting the system.
The Fuses in the PPM4 are an over current protection device. Whenever the current builds up beyond the
permissible limit, (generally 100mA), the fuse opens to protect the circuit ahead.
PPM4 must be properly earthed to work well. It is recommended that PPM4 be connected to a separate telecom
earth (ground).
Telecom earth is a dedicated earth (ground) only for the System. A dedicated earth greatly reduces the risk of back
voltage.

Installing PPM4
Refer the block diagram above for the location of the PPM4.
1. Unpack the PPM and check the package contents.
2. Select an appropriate location for the PPM4. Refer the block diagram above when deciding where to place
the PPM4. Also, take into consideration the length of the cables of the PPM4
.
3. Use the Mounting Template supplied with the PPM4 to drill holes on the wall to fix the PPM in the selected
location. Fix the screws supplied with the PPM4 into the drilled holes, with their heads protruding from the
wall.
4. You may mount the PPM4 first and connect the cables OR you may connect the cables first and then
mount the PPM4.
5. To connect cables, press the snap fits on both sides of the PPM4 to release the cover. Remove the cover.
6. Connect the Earth wire (green wire) to the Telecom Earth.
7. Now connect the CO Trunk wires from the CO side into the PPM4 port connectors marked as P1, P2, P3
and P4.
8. To do so, strip off about half a centimeter of the insulation of the wire ends of the first pair of CO Trunk you
want to connect to the PPM4.
9. Push back the (orange-color) levers of the connector of port P1, using a blunt pin or a small flat screw
driver or your thumbnail.
10. Insert the stripped ends of the two wires into the two (green-color) openings of the connector, with one wire
in each opening.
11. Release pressure on the levers. Both wires will be held in place by spring clamp action.
To remove the wires,

• push back the levers.
• pull out the wires gently.
• release pressure on the levers.

12. Now, repeat the above steps to connect the other CO Trunk wires from the CO side into the connectors of
the ports P2, P3, and P4.
13. Now, terminate the wire pairs emerging from the PPM4 multi-pair cable into the 'Trunk Lines' side of the
MDF using the punch tool for Krone Modules.
Refer the following table for connection details of the wires:
PPM4 Port Color
P1 Blue and White
P2 Orange and White
P3 Green and White
P4 Brown and White
14. Replace the cover of the PPM4 by pressing back the snap fits on both sides.
15. Mount the PPM, if not done already.

Protecting the System and Yourself
The System is an electronic device. When you handle any electrical or electronic equipment, you are in a situation
that could cause you bodily harm, besides damage to the product. When handling any electronic equipment, you
must be aware of the safety hazards involved in electrical circuitry and the standard practices for accident
prevention.
When using any telephone equipment, take every safety precaution to reduce the risk of fire, electric shock and
injury to persons. Read and understand the precautions, dos and don'ts of handling this product listed below.
These instructions are by no means exhaustive. So, take all the necessary precautions for handling electronic and
electrical appliances. Your safety and that of the others lies in your hands.
Location
• Do not place this product in any of the following locations:
• Near a water source like a wash bowl, kitchen sink, laundry tub, near a swimming pool, or in a wet
basement.
• In places where dust, oil, corrosive fumes may come in contact with the system.
• Any area where it is exposed to direct sunlight, heat, excessive cold or humidity.
• On movable or unstable surfaces, which may cause the product to fall and get damaged.
• Any area where shocks or vibration are frequent or strong.
• Near High-Frequency generating devices such as Electric Welder, Sewing Machine or and Microwave
Oven.
• Do not leave cables exposed on the ground where they may be trampled upon, or get damaged by
entangling with feet or pressure from other heavy objects.
Power Supply
• This product should be operated with proper supply voltage. If you are not sure about supply voltage,
contact authorized dealer.
• The System does not work in isolation from the environment. Power is fed to the system for functioning of
the system. It has several interfaces like trunk lines and extensions, PC interface, etc. So there are
chances of heavy voltages entering the system through these interfaces. Also, static charges could find
their way through the system components.
Protect the system from heavy voltages from the mains

• The System is designed to work with input voltages ranging between 100-240VAC. The Power Card of
System have a 'switch mode' design to support such a wide range of operating voltage.

Protect the system from heavy voltages on the trunk lines and the
overhead stations
• The System may be damaged by heavy voltages entering the system from trunk lines or from overhead
stations. These heavy voltages may enter the trunk lines and from overhead stations due to:
• Heavy voltage line falling on the CO line or on the overhead stations cable. A dangerous surge can
occur if a telephone line comes in contact with a power line.
• Lightning/Thunderbolts.
• To protect System from these voltages, use Primary Protection/Surge Protectors on the trunk and long
distance extension lines to protect the system from lightning and electrical surges.
• Install any standard Input Protection (punch down protection) on the Krone Modules of the MDF or the
“Primary Protection Module - PPM4” supplied by Matrix at entry points for all CO trunks lines and all
overhead stations. The product Warranty does not cover damages resulting from heavy voltage on CO
lines and overhead stations!
• It is recommended that you install the PPM on the MDF, as MDF cables from the CO are terminated on the
System MDF.
Protect the system from Lightning

• To protect System from extremely high voltage currents associated with lightning strikes, install a lightning
protector on an outside (CO) line.
Protect the system from static charges

• Every person carries some static charge in his/her body depending upon body composition and the
environment around them. Most of the times, this charge finds its way to the earth when the person
touches any object which is grounded, or when the person is barefoot.
• Generally, persons installing or handling electronic and electrical equipment take precaution to wear
appropriate footwear to get protection from electric shocks. Doing so, the static charge accumulates in his/
her body and does not find its way to the ground. But when such a person touches any of the electronic
cards, the static charge finds its way through the electronic components thereby causing damage to the
cards.
• So, the person installing or servicing the system must provide a path to the static charges, by wearing an
antistatic belt, which is properly earthed.
Protect the system from heavy voltage on the communication cable

• The Communication Port (COM Port) is provided on the System for connecting a PC.
• If an electrical wire carrying heavy voltage accidentally shorts with this cable, heavy voltages can damage
the communication port.
• It is recommended that the communication cable (connecting System and the PC) be run through the
conduit carrying telephone cables or through a separate conduit.

Telecom Earth (Ground)
• The Earth (Ground) is the most important safety procedure to prevent electrical shocks and fires. It
protects from lightning strikes, electrical transients, static discharges, electromagnetic interference and
electrical hazards.
• Ensure that a proper electrical earth and a telecom earth are in place for the safety of people and the
system. Telecom earth is a dedicated earth for the System/any other telecom equipment.
• Provide a separate Telecom Earth (Ground) to the system installation. Providing a separate earth to the
telecom equipment eliminates the possibility of any back-voltage on the earth.
• Refer “How to Make the Telecom Earth” for instructions on making the perfect earth (ground).
Shock and Fire Hazard

• Always wear a properly earthed, electrostatic discharge preventive wrist strap/belt while handling the
system and its cards to prevent damage to the system and harm to yourself.
• Do not open the system in power On condition.
• Slots and openings in the cabinet and the back or bottom are provided for ventilation, to protect the system
from overheating. These openings must not be blocked or covered.
• Never insert or push objects of any kind into this product through the cabinet slots as they may touch
dangerous voltage points or short out parts which may result in fire or electric shock.
• Do not allow anything to rest on the power cord. Do not locate this product where the cord will be trampled
upon or get entangled.
• This product is equipped with a plug having a third (ground) pin, which fits only into a grounding-type
outlet. This is a safety feature. So, if the existing outlet is not a three-pin and or if you are unable to insert
the plug into the outlet, have the outlet replaced by the electrician.
• Do not overload wall outlets and extension cords as this can result in the risk of fire or electric shock.
• Do not disassemble this product. Opening or removing covers may expose you to dangerous voltages or
other risks. Incorrect reassembly may cause electric shock when the appliance is used. Take the product
to a qualified technician when service or repair work is required.
• Avoid using a telephone (other than a cordless type) during a storm, to prevent electric shock from
lightning.
• Do not use the telephone to report a gas leak in the vicinity of the leak so as to prevent the risk of fire.
External Devices
• When you connect external devices like headset, sensors, paging devices, telephone instruments, cables,
connectors, etc., ensure that they are of standard make and good quality, so that the functioning of the
system is not affected.
• Matrix does not guarantee the performance of external devices that are not supplied by it.

Cleaning and maintenance
• Unplug this product from the wall outlet before cleaning.
• Do not use liquid cleaners or aerosol cleaners. Never spill liquid of any kind on the product.
• Use a dry cloth for cleaning.
Service and Repair

• Unplug this product from the wall outlet and refer servicing to a qualified service person under the following
conditions:
• When the power supply cord or plug is damaged or frayed.

• If liquid has been spilled into the product.
• If the product has been exposed to rain or water.
• If the product has been dropped or the cabinet has been damaged.
• If the product exhibits a distinct change in performance.
Battery
System contains a 3VDC/15mAh Manganese Lithium Coin Battery (ML 1220 - Rechargeable) of diameter 12.5mm
and height 2.0mm. The Battery is located on the CPU Card. The Battery should be replaced only by authorized
dealers of Matrix. End Users must not attempt to replace it.
Caution: There is risk of explosion if the Battery is replaced in an incorrect manner. Please dispose-off
used Batteries.
Disposal
• This product must be disposed according to the national laws and regulations prevailing in the country
where it is installed.
Warning for RF Safety

The product complies with RF exposure guidelines as per standard FCC 47 CFR part 2. However, please observe
the following precautions.
• Make sure that the RF Antenna is installed at least 20 cm away from other electronic and radio
transmission devices.
• Make sure that the RF Antenna is installed at a place at 20 cm away from people's vicinity.
• Do not place magnetic storage media near the product.
• People carrying medical implants like cardiac pacemakers are advised to maintain appropriate distance
from the system. They are also advised to avoid being in the vicinity of the product for a long time.

CHAPTER 5 Installing ETERNITY LENX
• The Matrix ETERNITY LENX is to be installed by persons trained and experienced in telecom wiring.
• The person installing the ETERNITY LENX must be familiar with trunks, physical wiring of the MDF on
both the exchange (System) side and the line side (CO).
• When installing any equipment, make sure that you take all the necessary precautions for handling
electronic and electrical appliances. Follow proper procedures for static electricity, while handling the
• Use a grounding mat and wear an anti-static strap/belt. Read the do's and don'ts listed in “Protecting
the System and Yourself”.
• If you have complied with the requirements and instructions described in “Before You Start”, you may
now begin the installation of your ETERNITY LENX.
Firmware Version V1R2 and earlier

system; to move it like a trolley.
slots.
The Matrix ETERNITY LENX is shipped factory fitted with the Power Supply Card, the CPU Card in their respective
fixed slots (refer the section “Know Your SARVAM UCS”). VoIP and VMS are in-skin to the CPU Card. Hence,
separate VMS and VoIP Cards are not required.
The Slave Cards - BRI, T1E1PRI, GSM, DKP, CO, SLT, E&M, Radio, Data, Magneto, E1FO - are shipped
separately as per the order placed by individual customers. These Cards are installed in any of the Universal slots.
ETERNITY LENX supports ETERNITY LE PS48VDC card, ETERNITY MENX-LENX CPU card, ETERNITY LE
SLT48 card, ETERNITY LE ILC48 card and all other cards of ETERNITY ME (except ETERNITY ME Power Supply
card.)

If you upgrade the system firmware to V1R3 and later, the Expansion Slots license will be applicable for the
universal slots. No universal slots will be functional by default. You must purchase the SARVAM EXP4 ENT license
to activate the universal slots as required.
For details, see “Expansion Slots” under “License Management”.
Firmware Version V1R3 and Later

system; to move it like a trolley.
slots.
If you have upgraded the system firmware to V1R3 and later in the old ETERNITY LENX system, the Expansion
Slots license will be applicable for the universal slots. No universal slots will be functional by default. You must
purchase the SARVAM EXP4 ENT license to activate the universal slots as required.
If you have purchased the new ETERNITY LENX system with the firmware V1R3 and later, the Expansion Slots
license will be applicable for the universal slots. The first eight universal slots after the power supply card in the first
rack will be functional by default. If you require more functional universal slots, you must purchase the SARVAM
EXP4 ENT license.
Each SARVAM EXP4 ENT license will provide the activation for next four universal slots in sequence.
The Matrix ETERNITY LENX is shipped factory fitted with the Power Supply Card, the CPU Card in their respective
The Slave Cards - BRI, T1E1PRI, GSM, DKP, CO, SLT, E&M, Radio, Data, Magneto, E1FO - are shipped
separately as per the order placed by individual customers. These Cards are installed in any of the Universal slots.
ETERNITY LENX supports ETERNITY LE PS48VDC card, ETERNITY MENX-LENX CPU card, ETERNITY LE
SLT48 card, ETERNITY LE ILC48 card and all other cards of ETERNITY ME (except ETERNITY ME Power Supply
card.)
Illustrated below is the position of the fixed and universal slots in ETERNITY LENX.

ETERNITY LENX
Rack 1
Power Supply Card
CPU Card
CPU Card
12 Universal Slots
Rack 2
Power Supply Card
15 Universal Slots
The extreme left slot of each rack is reserved for the Power Supply Cards and the two extreme right slots of the first
rack are reserved for the CPU Cards.
Follow the installation instructions for Cards described here also when you expand the system (add more Cards) or
remove or swap Cards for maintenance and repair.
1. Unpack the box. Check the package contents (see “Packing List”). Contact your Dealer/Distributor if any of
the items is missing, faulty or damaged. Do not discard the packaging material.
Mounting the System

2. Decide where to mount the ETERNITY LENX - on a table or wall - taking into consideration the mechanical
dimensions and the weight. If mounting the system on a wall, you may refer the mechanical dimensions
and the Mounting Template for drilling holes at appropriate places on the wall. Make sure the system
orientation is horizontal.
3. When installing the system in a rack, allow adequate space between the system and other units for air
circulation. Make sure the system orientation is horizontal.
4. Mount the system at the selected site. Make sure that the system is placed such that you have full access
to the front and back panels. The holes in the panels are provided for ventilation; Make sure that these are
not blocked, to prevent overheating.

Connecting Input Power Supply
5. Ensure that a proper electrical earth and telecom earth are in place.
6. Check the voltage at the power point from where the supply is to be given to the system. It should be as
per the specifications. Earth the system properly. (Refer “How to Make the Telecom Earth”).
Inserting Cards
7. Make sure that the ETERNITY LENX power is off and the power cord is unplugged.
8. Select a free slot from the universal slots.
9. Unscrew and remove the filler bracket that covers the card-slot opening of the slot you intend to use.
10. Hold the card with the connectors facing you. Do not grab the card from both ends.
11. Slide the card into the slot, along the guide rails provided for each slot at the top and bottom planes.
12. Ensure that the cards are inserted deep enough for all the connector pins on the cards make complete
contact with those of the motherboard on the backplane.
Do not force the card into the slot. Doing so can damage the card or the slot connector.
13. When the card is firmly seated in the connector, push down the levers on the card mounting bracket and
secure the card with the screw provided.

14. Tighten the screws on either side of the bracket.
15. Following the above steps, install each card into the universal slots.
Detailed installing instructions are provided for each card - DKP, SLT, CO, ISDN BRI, ISDN T1E1PRI,
GSM, E&M, E1FO, Magneto, Radio - later in this section. Refer to them when installing each card type.
16. To remove a card:
• Switch off power supply, unplug the power cord.

• Disconnect any cables connected to the card.
• Remove the screws from the card-mounting bracket.
• Lift the levers on the mounting bracket to release the card.
• The card will emerge out of the slot.
• Grasp the card by its mounting bracket, and ease it out of its slot.
• If you are removing the card permanently or for a certain period of time, install a filler bracket over the
empty card opening in the chassis.
• Installing filler brackets over empty card-slot openings is necessary to protect the system from dust,
dirt, insects and damage.
17. Using the cables supplied with each card, and terminate the cables in the Main Distribution Frame (SLT,
DKP, CO, and E&M lines), the NT1 device (ISDN BRI lines), ISDN Modem (ISDN PRI Lines), as
applicable.
Lead the cables neatly and tangle-free into the MDF.
18. After you have completed inserting and connecting the cards, power ON the system and observe the
Reset cycle and the LED pattern of each card, where applicable.

The Power Supply Card
Matrix ETERNITY LENX supports PS48V Card with 48VDC (-10 to +20%) as Input DC Power Supply Voltage. A
Float cum Boost Charger (FCBC) or 48VDC power supply is required to feed 48VDC power (Emerson Power
Supply supplied by Matrix or any other) to the card. The FCBC works on input AC mains. This card is available in
DC to DC 1000W power.
The card has two LEDs, a power ON/OFF Switch, protection MCB and a 3-way termination block for connecting the
power cord. The PS48V Card provide DC output voltages as: +3.5V, +5.0V, -27V and -85V. These are indicated by
LEDs.
The ETERNITY provides Redundancy and Hot Swap option for the Power Supply card.
The ETERNITY LENX supports two PS48V power supply cards. Whenever there is a fault in one, the other takes
over the control, providing uninterrupted communication.
Installing the Power Supply Card

The Power Supply Card is located in a fixed slot. No other card can be inserted in this slot.
The Power Supply Card is delivered factory fitted, when you buy the system. However, if you want to remove the
card for the purpose of maintenance or replace it with a new one, please follow the instructions below:
1. Unpack the Power Supply Card and verify the package contents.
If already installed, switch OFF power supply, unplug the power cord. Remove the screws securing the
card. Lift the levers on the mounting bracket to release the card. As the card emerges from the slot, ease it
out of the slot.
2. Insert the Power Supply card into the guide rails into the slot designated for the Power Supply Card. Make
sure that the card is inserted deep enough to make perfect contact with the connectors on the
motherboard at the backplane.
3. Now, press down the levers on the card mounting bracket to secure the card in its slot.
4. Secure the card in the slot by screwing the bracket on both ends.
5. If installing the PS48V card, connect the Float cum Boost Charger (FCBC) or Emerson PS or any 48VDC
power source. Terminate the power cord from the FCBC output or the Emerson PS into the 3-way
termination block (female connector) on the PS48V card.
Polarity is critical. Ensure that the wires are connected with the correct polarity. Follow the standard color
codes used by FCBC manufacturers:
Color Signal
Red +48VDC
Black GND
Yellow-Green Earth

If connecting the Emerson PS, refer “Connecting an Emerson Power Supply to the ETERNITY LENX”.
It is recommended that you measure the voltage before connecting the power cable to the power supply
card. Ensure that the earth is connected.
Connecting an Emerson Power Supply to the ETERNITY LENX

• Unpack the Emerson PS and the Emerson PS Tray. Check the package contents. Refer “ETERNITY
LENX”.
• Affix the Emerson PS unit onto the Emerson PS tray supplied by Matrix with the help of three screws.

• Now insert the tray into the last slot of the system rack and affix the tray onto the rack with the help of
screws.

• The Emerson PS has an Input Terminal and an Output Terminal. The Input Terminal is used to connect
the Emerson PC to the Mains Power Supply. The Output Terminal is used to connect the Emerson PS to
the 48VDC Power Supply Card.
• The Input Terminal has three screws. Unfasten and remove the screws. Use the Power cable (not supplied
by Matrix) to connect the Emerson PS to the Mains. The cable has three wires — Red, Black and Yellow-
Green. At one end of the cable a plug is connected and the wires are free on the other end. A lug is affixed
to each wire.
• Insert the lugs onto the screws as per the color codes mentioned below and fix the screws back.
Color Label on the Emerson PS
Red L (Line)
Black N (Neutral)
Yellow-Green (Earth)
• Insert the Plug into the Main power source.
• The Output Terminal has four screws — two to carry V+ signaling and two to carry V- signaling as the
system supports two 48VDC PS cards.
• A cable is supplied by Matrix with each 48VDC Power Supply card to connect the card to the Emerson PS.
The cable has three wires — Red, Black and Yellow-Green. At one end of the cable a male connector is
connected and the wires are free on the other end. A lug is affixed to each wire. A separate screw is placed
on the top cover of the Emerson PS for earthing.
• To connect one card, unfasten and remove two screws, one that carries V+ signal and the other one that
carries V- signal. Similarly, also remove the screw provided for earthing. Insert the lugs onto the screw as
per the color code given below:
Color Label on the Emerson PS
Red V+
Black V-
Yellow-Green Earth

• Fix the screws back.
• Follow the same steps to connect another 48VDC Power Supply card with the Emerson PS.
• Now insert the male connector into the female connector (3-way termination block) on the card.
Connecting FCBC to the ETERNITY LENX

AC Mains Supply
FCBC ETERNITY LE
230VAC +/- 41 to 56V
25A Card PS48VDC
10%@50Hz
48V Battery
6. Connect Battery back up to the FCBC15.
Battery backup time depends upon the total load. The total load is the sum of system's load and load of
active extensions.
The Battery back up time depends on the 'Ah' rating of the battery connected to the FCBC. If 48V/26Ah
batteries are connected to the FCBC for the system then backup time of 2.5 to 316 hrs can be ensured.
The FCBC uses the constant voltage charging method. So, the batteries get charged faster if less power is
consumed by the system when in mains mode.
7. Switch on power supply, after completing all other installation.
LED Pattern
LED L1 is a dual color LED for Power.
LED 2 is a single color LED that indicates Reverse Input Voltage.
Condition LED Color Cadence
All modules operating inside

L1 (PWR) GREEN Continuously ON
the card are OK
Any one or more modules

operating inside the card are L1 (PWR) RED Continuously ON
faulty
Reverse Input L2 (REV) RED Continuously ON
15. When the batteries are drained, the FCBC goes into the boost mode and begins to charge the batteries at higher current. When
the batteries reach a preset voltage level (typically set to 56.0 volts), the FCBC goes to float mode. In the float mode the FCBC
keeps charging the battery but at lower current. The FCBC monitors the voltage level of the batteries. As soon as the battery volt-
age goes below preset voltage (typically set to 50.4 volts), FCBC goes from float mode to boost mode. The change over from
mains to battery and vice-versa is automatic. The advantage of using an FCBC is that batteries get charged faster, since the bat-
teries are charged with higher current initially.
16. This is calculated taking into consideration 25% SLT load of the System.

The CPU Card
The ETERNITY MENX-LENX CPU Card hosts the SARVAM UCS Application. It supports four VOCODER modules
and one VMS module. Both the modules — NX DBM VOCODER64 and NX DBM VMS64 are optional and can be
purchased separately.
This card hosts Communication Manager, Feature Server, VoIP Server, VMS Server and other important servers
and modules which controls all other slave cards (SLT, DKP, CO+SLT, DKP+SLT, E&M, BRI, T1E1, E1FO, GSM
etc.). All the configuration and programming information is stored on this card.
ETERNITY LENX supports redundancy along with Hot Swap for the CPU Card. An additional CPU Card can be
installed for redundancy. If the main CPU Card fails, the other card takes over. However, during this process all the
ongoing calls will be disconnected.
The CPU Card occupies fixed slots, the first two slots from the extreme right of the first Rack, with a unique
arrangement of connectors. So no other card can be inserted in the slots of the CPU Card.
The CPU Card has a WAN Port, LAN Port and USB Port on the front panel. It also has an Internal USB Port with a
factory fitted pendrive.

Ports and Connectors:
Port Connector Description
Used for connecting the Ethernet cable into LAN Port to connect to a PC or a
LAN RJ45
LAN Switch.
Used for connecting the Ethernet cable into WAN Port to connect to a Broadband
WAN RJ45
Router/Modem.

The External USB can be used as COM Port by connecting the USB to COM
Converter.
USB to COM The USB to COM Port can be used to:

USB • set up and run software applications — PMS and CAS.
Converter (Optional)
• capture System Activity Log, System Fault log and Hotel Motel
Activity logs.
• generate SMDR reports.
If you buy a spare CPU Card separately, the default pendrive will not be provided along with it.
LAN Interface
The LAN Port is provided to connect:
• the system to a PC or a LAN. This port is used for operating the web-based programming software Jeeves.
• the CPU Card to the Local Area Network to register SIP extensions through the LAN Port.
• set up and run software applications such as PMS and CAS on any PC on the LAN.
• generate Station Message Detail Record (SMDR) Reports on any PC on the LAN.
• capture “System Activity Log”, “System Fault Log” and Hotel Motel Activity Log.
WAN Interface
The WAN Port is provided to connect:
• a LAN Switch/Hub/Router/Modem.
• the CPU Card to the public network over a Router/Modem. Any user on the public network can be
registered as SIP Extension through the WAN Port.
VoIP Interface
The CPU Card supports four NX DBM VOCODER64 modules. You must purchase the module separately for VoIP
functionality.
VOCODER Channels
The system supports four NX DBM VOCODER64 Modules. Each module supports 64 VOCODER Channels17. You
must purchase the modules separately. The system provides 4 pre-activated VOCODER channels by default which
can be used after installing NX DBM VOCODER64 module. If you require more channels, you can purchase the
licenses accordingly. Matrix provides two licenses — SARVAM VOCODER CHNL4 and SARVAM VOCODER
CHNL16.
If you require more than 64 VOCODER channels, you can install another NX DBM VOCODER64 Module.
17. The number of VOCODER channels that will be supported would be as per the license you purchase.

A call made from a SIP Extension or SIP Trunk to another SIP Extension or SIP Trunk will consume two
VOCODER channels, whereas a call made from a SLT or DKP extension to a SIP Extension or SIP Trunk
will consume one VOCODER channel. Thus, the number of speech paths available to make simultaneous
calls will depend not only on the number of VOCODER channels, but also on the number of channels
consumed by such SIP-to-SIP and Analog/Digital extension to SIP Trunk/SIP Extension calls.
VMS Interface
The system supports a full-fledged, 'in-skin' Voice Mail System module to provide mailbox facility to all its
extensions users. The Voice Mail System also forms the basis of other features like Conversation Recording and
Call Taping.
The NX DBM VMS64 Module is an optional module. It must be purchased separately. The factory fitted Pen Drive
provided also contains the VMS configuration files, and voice messages for prompts and greetings along with the
SARVAM UCS ENT Application. The Pen Drive is also the storage device for mailbox messages.
The system supports a maximum of 64 channels out of which 4 channels are provided by default. If you require
more channels, you can purchase the licenses accordingly. Matrix provides two licenses — SARVAM VMS CHNL4
and SARVAM VMS CHNL16.
Internal USB Port

The CPU Card has an Internal USB Port with a pendrive inserted into it.
The pendrive supports FAT32 file format. It contains the SARVAM UCS Application, VMS greetings, messages,
Matrix Extended IP phone firmwares and SMS Server firmware.

Do not remove the pendrive.
When you select the SARVAM UCS ENT Application, the system fetches the application from the pendrive.
External USB Port (Device Port) 3.0

The CPU Card has an External USB Port on the fascia. This can be used as a COM Port by connecting the USB to
COM Converter.
The USB to COM Converter will not be provided by Matrix.
The following USB to COM Converters are supported:
• Prolific PL2303 by BAFO
• CH341 by Winchiphead
If you use any other USB to COM Converter, it may not function properly.
The USB to COM Port has a DB-9 connector.
The port allows you to connect a PC to the system, so that you can install and operate the following features:
• capture “System Activity Log” and “System Fault Log”, Hotel Motel Activity Log.
LED
The CPU Card has three dual color LEDs:
• Heart beat (L1)
• Layer Communication indications (L2)
• Slave Switch indications (L3)
LED for both Active and Standby Card:
State Color Cadence
At Power ON
L1 and L3 – RED Continuous ON for 15 sec

(approximately)
L1 and L3 OFF for 30 sec (approximately)
When L1 and L3 has been OFF Continuously ON for 3 sec

L2 – ORANGE
for 10 sec
L2 – OFF OFF
LED for Active Card:
State Color Cadence
L1 Pattern

State Color Cadence
During initialization process ORANGE Continuous ON

After initialization process - OFF
In normal condition GREEN Toggle
L2 Pattern
500 ms GREEN – 500 ms RED – 500

During initialization process GREEN, RED, ORANGE
ms ORANGE
In normal condition - OFF
L3 Pattern
When sync is received by

RED Continuous ON
Active card from Standby Card
When Sync is completed - OFF
When Data transfer done by

RED Continuous ON
Active card
LED for Standby Card:
State Color Cadence
L1 Pattern

In normal condition RED 1 sec ON – 1 sec OFF
L2 Pattern
500 ms GREEN – 500 ms RED – 500

ms ORANGE
PLL Lock (standby card clock

GREEN Continuous ON
is in sync with active card)
PLL is not lock (standby card

clock is not in sync with active - OFF
card)
L3 Pattern
When sync is sent by Standby

RED Continuous ON
card to Active card

The CPU Card has a Password IP Default Switch (SW1) on the board:
Switch Position Function
SW1 ON Password IP Default
OFF (default) Normal
Installing the VOCODER Module

To install,
• Unpack the NX DBM VOCODER64 Module.
• If the CPU Card is already installed, switch off power supply, unplug the power cord. Remove the screws
securing the card. Lift the levers on the mounting bracket to release the card. As the card emerges from
the slot, ease it out of the slot.
• Place the card carefully on a table with some packing underneath it. Avoid any physical contact with the
PCB part of the card as this could cause Electrostatic discharge (ESD) and may damage the hardware.

• There are 5 PCI Connectors (4 for VoIP and 1 for VMS) and five pairs of studs on the CPU board.
• Locate the VOIP #1 label on the CPU board. The first NX DBM VOCODER64 Module should be mounted
here.
• Remove the screws on the studs for the module and keep them aside.
• Carefully hold the NX DBM VOCODER64 Module from the edges. Make sure you do not touch the PCB
area.

• Insert the NX DBM VOCODER64 Module into the PCI Connector socket.
• Press the Module with a finger and match the mounting holes perfectly with the stud holes. Make sure you
do not touch the PCB area of the module except the yellow line provided for grounding at the front end of
the module.
Do not apply excessive pressure.

• Secure the module with the screws on the studs.
You may install the second NX DBM VOCODER64 Module by locating the VOIP #2 label on the CPU board and
follow the same steps as above. Similarly, install the other modules if required.
Removing the VOCODER Module

• Locate the VOCODER Module you want to remove from the CPU Card.
• Remove the screws from the module and keep them aside.

• Firmly hold the module and ease it out of the PCI connector carefully.
• Replace the screws on the studs.
Installing the VMS Module

To install the NX DBM VMS64 Module, locate the VMS MOD label on the CPU board and follow the same steps as
described for “Installing the VOCODER Module”.

Make sure you use the plastic screws provided to affix the VMS Module over the studs.
• The pendrive which is provided to you by default contains VMS data and VMS firmware. You will be able to
use the VMS features once you activate the VMS License.
If you want to store more voice mail messages or greetings then you will need more space to store the
same. You can replace this default pendrive with a new one having more space.
To do so, you need to format your new pendrive with FAT32 file format and then copy all the contents of
the factory fitted pendrive into the new pendrive.
Switch-off the system and then replace the pendrive. The system will not detect the new pendrive if you do
not restart the system after replacement.
Make sure while replacing the pendrive, you insert it in the same USB Port where the factory fitted
pendrive was inserted.
• If you have no other modules to install, insert the card back into the ETERNITY LENX.
• Connect a computer to the LAN/WAN Port of the system with the ethernet cable supplied for the port.
• Open a Web browser on the computer to access the embedded Web server, Jeeves.
• Activate the VMS License Voucher. See “License Management” for instructions.
• Configure VMS. For detailed instructions, see “Configuring Voice Mail System”.
For removing the VMS Module, follow the same steps as described for removing the VOCODER Module. See
“Removing the VOCODER Module”.

The Single Line Telephone Card
The Single Line Telephone (SLT) Card provides the interface to connect as extension phones, any standard, two-
wire, analog single line telephone instrument - rotary, pulse-tone, cordless, feature phones with or without Calling
Line Identification.
The SLT Card is available in the following configurations. SLT interface also is available in combination with Two-
wire trunk interfaces on a single card.
SLT Cards for ETERNITY LENX
Card Name Configuration and Application
ETERNITY LE Card SLT48 48-port card to connect 48 Single Line Telephones
ETERNITY ME Card SLT32 32-port card to connect 32 Single Line Telephones
Combination card, with 8-ports to connect to 8 Two-wire Analog trunk lines and
24 Single Line Telephones
ETERNITY ME Card
CO8+SLT24
This Card supports Power Fail Transfer. To know more, see “Power Fail
Transfer”.
The maximum number of SLT ports supported are 129618.
Connectors
The SLT Cards have RJ45 connectors, with each connector having 4 SLT ports. A multi-pair, MDF cable is supplied
for each connector.
Only the SLT48 card has a 50-pin Centronics connector for the ports.
LEDs
The SLT cards have a single, tri-color LED to indicate:
• the health of the card during the Reset Cycle.

• the status of any one extension during normal functioning of the system.
You may monitor any of the SLT Extension ports by assigning the LED to that port19.
18. Using commands you can configure upto 999 ports. The remaining ports can be configured from Jeeves only.
19. To do this, enter SE mode, and dial the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot
in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its functioning. LED
Number is the number of the LED on the card, which will monitor the port.

Installing Single Line Telephones
To be able to connect Single Line Telephones as Extensions to your system, you must install at least one of the
aforementioned SLT Cards in the System.
1. Decide the number of SLT extensions required and arrange for as many telephone instruments.
You may use any standard telephone instrument like a rotary phone, a pulse-tone switchable push-button
phone, a feature phone or a cordless phone.
Use SLTs equipped with a 'Flash' key, as several of the features and facilities of the system require you to
press Flash. If any of the SLTs you have selected does not have a Flash key, tap the Hook switch of the
phone to dial Flash.
2. Unpack the SLT Card and check the package contents. Ensure that the power supply is switched off,
before you begin the installation of the card. Always wear an electrostatic discharge prevention wrist strap/
belt and use a grounding mat.
3. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Do not
discard the filler bracket! You may require it at a later stage.
4. Insert the SLT Card into the guide rails of the free slot you selected for the card.
Make sure that the connectors on the card make perfect contact with those on the motherboard on the
backplane.
5. Press down the levers on the mounting bracket to secure the card in its slot. Now, secure the mounting
bracket with the two screws provided.
If you are installing more than one SLT Card, you can install the second card in any other free slot. It is not
necessary to install the second/third card in the subsequent slots.
6. Use the cables supplied with the SLT Card to connect the SLT wires with the Main Distribution Frame.
For each connector on the SLT Card, there is a separate 4-pair cable with an RJ45 jack on one end and
free at the other end.
Refer the illustrations below for pin out details of each connector.

ETERNITY LE Card SLT48
Connector 1
Port Type Port Number Pin Number Signalling Wire Colour
Port 1 1 Tip Blue
26 Ring White
Port 2 2 Tip Orange
27 Ring White
Port 3 3 Tip Green
28 Ring White
Port 4 4 Tip Brown
29 Ring White
Port 5 5 Tip Gray
30 Ring White
Port 6 6 Tip Blue
31 Ring Red
Port 7 7 Tip Orange
32 Ring Red
Port 8 8 Tip Green
33 Ring Red
SLT
Port 9 9 Tip Brown
34 Ring Red
Port 10 10 Tip Gray
35 Ring Red
Port 11 11 Tip Blue
36 Ring Black
Port 12 12 Tip Orange
37 Ring Black
Port 13 13 Tip Green
38 Ring Black
Port 14 14 Tip Brown
39 Ring Black
Port 15 15 Tip Gray
40 Ring Black
Port 16 16 Tip Blue
41 Ring Yellow

42 Ring Yellow
43 Ring Yellow
44 Ring Yellow
Port 20 20 Tip Gray
45 Ring Yellow
Port 21 21 Tip Blue

SLT
46 Ring Violet
47 Ring Violet
48 Ring Violet
49 Ring Violet
Port 25 25 NC Gray
50 NC Violet

Connector 2
1 Tip Blue
Port 25
26 Ring White
27 Ring White
Port 27 3 Tip Green
28 Ring White
Port 28 4 Tip Brown
29 Ring White
Port 29 5 Tip Gray
30 Ring White
Port 30 6 Tip Blue
31 Ring Red
SLT
32 Ring Red
Port 32 8 Tip Green
33 Ring Red
Port 33 9 Tip Brown
34 Ring Red
Port 34 10 Tip Gray
35 Ring Red
Port 35 11 Tip Blue
36 Ring Black
37 Ring Black

38 Ring Black
39 Ring Black
Port 39 15 Tip Gray
40 Ring Black
Port 40 16 Tip Blue
41 Ring Yellow
42 Ring Yellow
43 Ring Yellow

SLT
44 Ring Yellow
Port 44 20 Tip Gray
45 Ring Yellow
Port 45 21 Tip Blue
46 Ring Violet
47 Ring Violet
48 Ring Violet
49 Ring Violet
25 NC Gray
50 NC Violet
Jumpers on the Main Board (SLT48)
Jumper Number Position Function
J1 AB (default) Normal Operation
BC For uploading software using COM Port
J2 & J3 AB (default) Normal Operation

ETERNITY ME Card SLT32
L1
Connector Color Connection H/w Port Offset
RJ45-1 Blue - (Blue & White) SLT 01
Orange - (Orange & White) SLT 02
1 Green - (Green & White) SLT
Brown - (Brown & White) SLT 04

L1

L1
Blue - (Blue & White) SLT 01
1 Orange - (Orange & White) SLT 02
Green - (Green & White) SLT

ETERNITY ME Card CO8+SLT24
L1
RJ45-7 Blue - (Blue & White) TWT 01
Orange - (Orange & White) TWT 02
7 Green - (Green & White) TWT
Brown - (Brown & White) TWT 04
7. Plug in the RJ45 end of the MDF cables supplied with the card into the respective connectors. Refer to the
pinout details of the connectors of each SLT Card type illustrated above.
8. Terminate the open end of the cables into the punch down blocks of the Krone modules designated for
'Station Lines' in the “The Main Distribution Frame (MDF)”.
Each wire-pair from the SLT Port must be terminated to the bottom of the Krone Connector, while the wire-
pair of the extension line to be connected to this port must be terminated on the top of the Krone
connector. Refer the topic “The Main Distribution Frame (MDF)” for illustration.
9. Repeat the same steps to install another SLT Card.
10. If you have completed all installation tasks, power ON the system, observe the Reset Cycle and the LED
Pattern of the SLT Card.

LED Pattern of the SLT Card
Stage LED Color Cadence
Auto Upgradationa
Card waiting for application RED ON-200ms-OFF 200ms
Card is up, loaded with new application GREEN ON-200ms-OFF 200ms
Initialization
RED ON 500ms-OFF 500ms
GREEN ON 500ms - OFF 500ms
ORANGE ON 500ms - OFF 500ms
Stand-by task ORANGE 1 sec Orange -1 sec Green
Errors
Flash Failure None None
RAM Failure None None
a. Done by the Boot Loader Application.
Status of Selected Port
PORT Status LED Color LED Cadence
Selected port data transmitted to Master Card RED Togglea
Selected port data received from Master Card RED Toggle
a. The current LED state will remain the same until the next command is re-
ceived from the application on the SLT Port. For example, if the current LED
state is Green/Red ON, on the next command received, the LED will be
turned OFF. It will remain OFF until the next command is received. When the
next command is received it will be turned Green/Red ON again. This pro-
cess continues.
Jumpers on the Main Board
Connecting SLT instruments

11. Connect the SLT instruments you have arranged for. Plug in the SLTs into the wall socket/outlets.
• For the purpose of testing, you may connect one or two Single Line Telephone instruments by plugging
in the phone cables into the RJ45 connectors on the card.

• When you plug the RJ11 connector of SLT into an RJ45 connector on the SLT Card, the SLT will be
connected on the first port on the connector.

The Intercom Line Card
For the Building Intercom application, the system supports the Intercom Line Card (ILC).
You can connect any standard, two-wire, analog single line telephone instrument - rotary, pulse-tone, cordless,
feature phones with or without Calling Line Identification to the Intercom Line Card.
ILC Cards for ETERNITY LENX
ETERNITY LE Card ILC48 48-port card to connect 48 Single Line Telephones
ETERNITY ME Card ILC32 32-port card to connect 32 Single Line Telephones
Choose an ILC Card with the configuration that meets your requirement for intercom ports. Also, consider the
maximum Port capacity of the system you are installing. The maximum number of intercom ports supported are
1296.
Connectors
The ILC32 Cards have RJ45 connectors with four ports on each connector. A multi-pair, MDF cable is supplied for
each connector.
Only the ILC48 card has a 50-pin Centronics connector for the ports.
LEDs
The ILC cards have a single, tri-color LED to indicate the health of the card during the Reset Cycle.
Installing Intercom Telephones

To be able to connect intercom telephones to the system, you must install at least one of the aforementioned
intercom line cards in the system.
1. Decide the number of intercom extensions required and arrange for as many telephone instruments.
2. Ensure that the extension wiring is completed according to your requirements. The extension cables from
the wall jack are terminated in the Main Distribution Frame and the telephones are connected to the wall
jacks.
3. Always wear an electrostatic discharge prevention wrist strap/belt and use a grounding mat to prevent
damage to the components of the card.
4. Unpack the ILC card and check the package contents. You may switch off power supply before you install
the card. Since, ETERNITY LENX supports Hot Swap, you can install the card in power on condition.
5. Unscrew and remove the filler card mount bracket of any of the free (empty) Universal Slots. Keep the filler
bracket for future use.
6. Insert the ILC Card into the guide rails of the free slot you selected for the card.

backplane.
8. Repeat these steps to install another card.
9. Now, use the cables supplied with the ILC Card to connect the card to the Main Distribution Frame to
which the intercom phones are connected.
For each connector on the card, there is a separate 4-pair cable with an RJ45 jack on one end and free at
the other end. Refer the illustrations of the pinout of the intercom cards to connect the wires.
ETERNITY LE Card ILC48
Connector 1
Port Type Port Number Pin Number Signaling Wire Colour
Port 1 1 Tip White
26 Ring Blue
Port 2 2 Tip White
27 Ring Orange
Port 3 3 Tip White
28 Ring Green
Port 4 4 Tip White
29 Ring Brown
Port 5 5 Tip White
30 Ring Slate
SLT
Port 6 6 Tip Red
31 Ring Blue
Port 7 7 Tip Red
32 Ring Orange
Port 8 8 Tip Red
33 Ring Green
Port 9 9 Tip Red
34 Ring Brown
Port 10 10 Tip Red
35 Ring Slate

Port Type Port Number Pin Number Signaling Wire Colour
Port 11 11 Tip Black
36 Ring Blue
37 Ring Orange
38 Ring Green
39 Ring Brown
40 Ring Slate
41 Ring Orange
42 Ring Brown

SLT
43 Ring Slate
Port 19 19 Tip Blue
44 Ring Orange
Port 20 20 Tip Blue
45 Ring Green
Port 21 21 Tip Blue
46 Ring Brown
Port 22 22 Tip Blue
47 Ring Slate
48 Ring Brown
49 Ring Slate
Port 25 25 NC Slate
50 NC Brown

Connector 2
Port 25 1 Tip White
26 Ring Blue
Port 26 2 Tip White
27 Ring Orange
Port 27 3 Tip White
28 Ring Green
Port 28 4 Tip White
29 Ring Brown
Port 29 5 Tip White
30 Ring Slate
Port 30 6 Tip Red
31 Ring Blue
Port 31 7 Tip Red
32 Ring Orange
Port 32 8 Tip Red
33 Ring Green
Port 33 9 Tip Red

SLT
34 Ring Brown
Port 34 10 Tip Red
35 Ring Slate
36 Ring Blue
37 Ring Orange
38 Ring Green
39 Ring Brown
40 Ring Slate
41 Ring Orange
42 Ring Brown

43 Ring Slate
Port 43 19 Tip Blue
44 Ring Orange
Port 44 20 Tip Blue
45 Ring Green
Port 45 21 Tip Blue
46 Ring Brown
SLT
Port 46 22 Tip Blue
47 Ring Slate
48 Ring Brown
49 Ring Slate
25 NC Slate
50 NC Brown
Connector 1
Port 1 1 Tip Blue
26 Ring White
Port 2 2 Tip Orange
27 Ring White
Port 3 3 Tip Green
28 Ring White
Port 4 4 Tip Brown
29 Ring White
SLT
Port 5 5 Tip Gray
30 Ring White
Port 6 6 Tip Blue
31 Ring Red
Port 7 7 Tip Orange
32 Ring Red
Port 8 8 Tip Green
33 Ring Red

Port 9 9 Tip Brown
34 Ring Red
Port 10 10 Tip Gray
35 Ring Red
Port 11 11 Tip Blue
36 Ring Black
37 Ring Black
38 Ring Black
39 Ring Black
Port 15 15 Tip Gray
40 Ring Black
Port 16 16 Tip Blue
41 Ring Yellow

SLT
42 Ring Yellow
43 Ring Yellow
44 Ring Yellow
Port 20 20 Tip Gray
45 Ring Yellow
Port 21 21 Tip Blue
46 Ring Violet
47 Ring Violet
48 Ring Violet
49 Ring Violet
Port 25 25 NC Gray
50 NC Violet

Connector 2
Port 25 1 Tip Blue
26 Ring White
27 Ring White
Port 27 3 Tip Green
28 Ring White
Port 28 4 Tip Brown
29 Ring White
Port 29 5 Tip Gray
30 Ring White
Port 30 6 Tip Blue
31 Ring Red
32 Ring Red
Port 32 8 Tip Green
33 Ring Red
Port 33 9 Tip Brown

SLT
34 Ring Red
Port 34 10 Tip Gray
35 Ring Red
Port 35 11 Tip Blue
36 Ring Black
37 Ring Black
38 Ring Black
39 Ring Black
Port 39 15 Tip Gray
40 Ring Black
Port 40 16 Tip Blue
41 Ring Yellow
42 Ring Yellow

43 Ring Yellow
44 Ring Yellow
Port 44 20 Tip Gray
45 Ring Yellow
Port 45 21 Tip Blue
46 Ring Violet
SLT
47 Ring Violet
48 Ring Violet
49 Ring Violet
25 NC Gray
50 NC Violet
Jumpers on the Main Board (ILC48)

ETERNITY ME Card ILC32
L1
10. If you have completed all other installation tasks, power ON the system, observe the Reset Cycle and the
LED indication of the card.
LED Indication of the ILC Card
Auto Upgradationa
Initialization
ORANGE,
Stand-by task 1 sec Orange -1 sec Green
GREEN

Errors
cess continues.

The Digital Key Phone Card
The Digital Key Phone (DKP) Card provides the interface to connect the proprietary digital key phones of the EON
series, the proprietary PC-based phone EONSOFT, the Direct Station Selection (DSS) Consoles, with the system.
DKP Cards for ETERNITY LENX
ETERNITY ME DKP32 32-port card to connect 32 DKP/DSS Consoles
16-port card to connect 16 DKP/DSS Consoles and the proprietary Digital Turret,
ETERNITY ME DKP16
EON74a.
a. EON74, the proprietary Digital Turret of EON74 is currently supported only on ETERNITY ME DKP16
Card and ETERNITY ME DKP8 Card.
Select a DKP Card with the configuration that meets your requirement for DKP Ports.To connect the proprietary
digital key phones with the system, you must have at least one of the above mentioned DKP Cards installed in the
system.
The maximum number of DKP Ports supported by the system are 128.
Connectors
The DKP Cards have RJ45 connectors, with each connector having 4 DKP ports. A multi-pair MDF cable is
supplied for each connector on the card.
LEDs
The DKP cards have a single, tri-color LED to indicate:

• the status of any one of the ports during normal functioning of the system. By default, it is assigned to DKP
Port 1.
You may monitor any of the DKP ports by assigning the LED to that port20.
Installing the Digital Key Phone Card

Decide the number of DKP extensions and DSS Consoles required and arrange for as many EON, EONSOFT and
DSS Consoles.
Decide the locations of the DKP extensions and make sure that the necessary wiring for the DKP extensions, from
the wall jack to the MDF, is done.
20. You can do this from the SE mode, by dialing the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the uni-
versal slot in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its function-
ing. LED Number is the number of the LED on the card, which will monitor the port.

1. Unpack the DKP Card and check the package contents21. Before handling the card, make sure that power
supply is switched off and you are wearing an antistatic-wrist strap/belt and have a grounding mat.
discard the filler bracket, keep for future use to cover empty slots.
3. Insert the DKP Card into the guide rails of the free slot you have selected for the card. All the pins on the
connector of the card should make perfect contact with those on the connector of the slot on the backplane
motherboard.
4. Press down the levers on the mounting bracket to secure the card in its slot. Now, fix the card in its slot
with the two screws provided.
If you are installing more than one DKP Card, it is not necessary to install the next card in the subsequent
slot.
5. Using the MDF Cables supplied with the DKP Card connect the DKP ports to the Main Distribution Frame.
Refer the connector pin details for each DKP Card type given in the following.
21. See “ETERNITY GENX Cards” under ‘Packing List’ of Appendix topic.

ETERNITY ME Card DKP32
L1
RJ45-1 Blue - (Blue & White) DKP 01
Orange - (Orange & White) DKP 02
1 Green - (Green & White) DKP
Brown - (Brown & White) DKP 04

L1

L1
1 Orange - (Orange & White) DKP 02
Green - (Green & White) DKP
6. Plug in the RJ45 end of the MDF cables provided with the DKP card into the respective connectors.
7. Terminate the free end of the cables into the punch down blocks of the Krone modules designated for
'Station Lines' in the Main Distribution Frame (MDF).
Each wire-pair from the DKP Port must be terminated to the bottom of the Krone Connector, while the wire-
8. Connect the Digital Key Phones to the wall jacks at their respective locations. Detailed installations
instructions for EON, EONSOFT are provided separately. Installation instructions for EON74 are provided
in the EON74 User Guide.
If you have completed all installation tasks, power on the system and observe the Reset Cycle and the LED Pattern
of the DKP Card.

LED Pattern DKP Card
Auto Upgradation
Initialization
GREEN ON 500ms-OFF 500ms
ORANGE ON 500ms-OFF 500ms
ORANGE, 1 sec Orange-1 sec

Stand-by task
GREEN Green
Errors
Status of Selected DKP Port
Selected DKP's data are transmitted to CPU Card RED Togglea on each event
Selected DKP's data are received from CPU Card RED Toggleb on each request from Master
a. The current LED state will remain the same until the next event is received from the application on the DKP Port.
For example, if the current LED state is Green/Red ON, on the next event, the LED will be turned OFF. It will
remain OFF until the next event occurs. When the next event is received it will be turned Green/Red ON again.
This process continues.
b. Same as the above note.
J2 & J4 AB NA
BC (default) Normal Operation/ Debug

Installing EON48
• Unpack the box and verify the package contents.
• You can mount the phone on a wall or on desk.
Mount the phone on a Wall
• Use the mounting template to drill holes of appropriate size and distance.
• Fix the screw grips in the holes you drilled.
• Fix two screws in the holes on the wall, ensuring that they are aligned with the Keyhole Slots 1 and 2. The
screws should protrude from the wall to fit into the keyhole slots.
• Now, mount the phone with the screws fitting into the keyhole slots.
• Reverse the handset wall mount tab to make sure that the handset remains intact when you mount the
phone. Push the handset wall mount tab upwards to remove it from the slot. Rotate it 180 degrees
clockwise and push it downwards into the slot.

Mount the phone on the Desk
• You can attach the Foot Stand in the following ways—at an angle of 20° Angle or at 30° Angle or at 50°
Angle.
• If you attach the Foot Stand at 50°, the phone will be placed in an almost upright position on your desk.
• Decide which of these positions would work for you best and accordingly attach the Foot Stand.
• Connect the handset of the EON48 to the phone body using the spring cord.

• To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
connector into the headset jack with the symbol on the left side panel of the phone.
Headset
You may also plug in a stereo headset with an RJ9 connector into the headset port at the bottom of the
phone, marked with the symbol .
• Plug one end of the RJ11 cable supplied with the phone into the RJ11 connector of the phone labeled as
‘LINE’ and the other end into the wall jack/DKP Port.

• When the system is powered ON, the EON will reset. The EON communicates with the system. The
handshaking lasts for 5-6 seconds. The EON model, version and revision number, along with the message
'Please wait'… appear on the LCD display.
M AT R I X E O N 4 8 - S V 2 R 2
PLEASE WAI T .. .
• After successful handshaking and reset cycle, if the DKP Parameters have been configured, the LCD
display of the EON will show the extension number and the extension name in one line and the day, date
and time and the time zone in the other line.
202 Reception
M on 2 4 A U G 1 2 : 0 0
• You may adjust the LCD for brightness, contrast and backlight. Refer the topic, “Digital Key Phone-
Operation” for instructions.
Installing EON310

• You can attach the Foot Stand in the following ways—at an angle of 35° Angle or at 50° Angle.

You may also plug in a stereo headset with an RJ9 connector into the headset port marked with the
symbol , on the left side panel of the phone.

• When the system is powered ON, the EON will get reset and the message “Welcome to Matrix.
Booting”…appears on the LCD display.
• The EON communicates with the system. The handshaking lasts for 5-6 seconds. The EON model,
version and revision number, along with the message “Please Wait”…appears on the LCD display.

• After successful handshaking and reset cycle, the extension number, day, date and time will appear on the
LCD of the phone. If you have already assigned extension number and name, in the DKP Parameters,
these will appear on the LCD.
Operation”.
Installing EON510
Keyhole Keyhole
Slot 1 Slot 2

If you are unable to remove the wall mount tab, you may use a tool like a minus screw driver to remove it.
Stand attached at
45 degree angle

Stand attached at
55 degree angle
AUX
Handset
• Plug one end of the RJ11 cable supplied with the phone into the RJ11 connector of the phone marked with
the symbol and the other end into the wall jack/DKP Port.
• To connect DSS532 with the phone, plug one end of the RJ11 cable into the AUX Port of the phone and
the other end into the IN Port of the DSS532. For installation, see “Installing DSS532 with EON510”.

• When the system is powered ON, the EON will get reset.and the message 'Phone is Booting; Please
wait…’ appear on the LCD display.
• The EON communicates with the system. The handshaking lasts for 5-6 seconds. The message 'Loading
Firmware Version-Revision; Please wait…’ appear on the LCD display.
these will appear, as illustrated below.
You may adjust the LCD for brightness, contrast and backlight. Refer the topic, “Digital Key Phone-
Operation”.
Installing DSS Consoles
Installing DSS64
Once you have installed EON48/310 with SARVAM UCS, installing the DSS Consoles can be done in a few simple
steps, very much similar to those involved in the installation of EON.
1. Unpack the box and verify the package contents22.
2. Place the DSS Console next to the DKP to which it is to be attached.
3. Decide which DKP Ports on the DKP Card are to be assigned to the DSS Consoles. You may select any
free (unused) port on the card for DSS Consoles. It is not necessary for the DSS Console ports to be in a
sequence with the DKP ports to which they are attached.
22. See “Packing List” of Appendix topic.

For example: you have connected DKP1 to Port 1 on the first RJ45 connector of the DKP8 card. You want
to attach two DSS Consoles to DKP1. The two DSS Consoles may be connected to any port on the
second connector of the card, not necessarily to Port 2 and Port 3 on the first connector.
4. The wire-pairs from the DKP Ports designated for DSS Consoles should be terminated on the bottom of
the Krone Connector (of 'Station Lines' on the MDF).
5. The wire-pairs of the DSS Consoles should be terminated into the top of the Krone Connector (of 'Station
Lines' on the MDF). Refer the topic “The Main Distribution Frame (MDF)” for illustration.
You can connect maximum two DSS64 with a single EON48/310.
6. The system automatically detects the DSS Console you connect and it will be will appear under
Unassigned DSS64 in “DSS Status”. You must first assign these DSS Consoles to the respective DKP
Ports and thereafter you will be able to configure the DSS Keys.
7. To assign the DSS Consoles, see “DSS Status” and to configure the DSS Keys, see “Programming DSS
Console Keys”.
Installing DSS532 with EON510

The instructions for installing DSS532 with EON510 or SPARSH VP510 are same. For detailed instructions, refer to
“Installing DSS532 with SPARSH VP510”.
Installing EONSOFT
To install EONSOFT, you must have a computer with Windows as the operating system. The EONSOFT is
compatible with the following Operating Systems of Windows:
• Windows 98
• Windows XP
• Windows NT
• Windows 2003
• Windows Vista
• Windows 2007
23. See “EONSOFT” under ‘Packing List’ of Appendix topic.

2. Connect the Handset to the dongle in the handset jack. If using a headset, connect the microphone and
the speaker connectors into the dongle.
3. Connect one end of the Communication cable to the COM port of the dongle. Connect the other end of the
communication cable into the COM port of the computer.
4. Connect a wire-pair of a DKP port to the RJ11 port marked 'DKP' on the dongle.
5. Switch ON the computer. The computer must have Windows Operating System installed on it.
6. Copy the EONSOFT Application Software provided by the Support Team onto your PC and install the
application.
7. After the program has been installed and run, a shortcut will be automatically created and appear on your
desktop.

8. Click the shortcut to open the program. The EONSOFT window will open:
9. Click Options at the top left of the window. A drop down menu will appear.

10. Click the option COM Port.
11. Select the COM Port to which the communication cable is connected.

12. EONSOFT is now connected. If you have already configured the DKP parameters like Access Code and
Name for the port to which EONSOFT is connected, these will appear.
• If this window does not appear after you have selected the COM Port Option, test the COM Port for
data transfer.
• If the wrong COM port has been selected, a dialog box will pop up on your screen with the message:
"COMx is invalid or busy, please select another COM Port". Select the correct COM Port.
Test the functioning of the COM Port of the PC and the communication cable, before you install the
EONSOFT.
Testing the COM Port

• From the drop down menu of Options, select the COM Port to which you have connected the
communication cable.
• The COM Port Settings dialog box will open.
• Connect the communication cable to the COM Port of the PC.
• Short pin2 and pin3 of the DB-9 connector at the free end of the cable.
• Click the button labeled Start Test in the COM Port Settings dialog box.
• After clicking this button, observe the Test Result section on the dialog box.

• The Error Count value shows zero, if both the communication cable and the COM port are working.
The above screen shows that the COM Port/communication cable is working.
• If the Error Count shows a value other than zero, it means that either the communication cable or the
COM port of the PC is faulty.
Above screen shows the faulty COM Port/Communication Cable.
• Remove the communication cable from the COM Port of the PC.
• Short pin2 and pin3 of the communication port of the computer and click 'Start Test' in the COM Port
Settings dialog box.
• Now, if the error count is zero, please check the Communication Cable.
• If the error count is not a zero, the COM Port of the PC is faulty. Try another communication port.

The CO Card
The CO Card provides the interface to connect the system with the Two-Wire Analog Trunk lines from the CO
Network. The CO Card supports the different standards and features of CO Networks across the world.
The CO Card is available in the following configurations. CO interface is also available in combination with SLT
ports on a single card.
CO Cards for ETERNITY LENX
ETERNITY ME Card CO16 16-port card to connect 16 Two-wire Trunk lines from the CO network
ETERNITY ME Card CO8+SLT24 Combination card, with 8 CO ports to connect 8 CO analog trunk lines
and 24 SLT ports to connect 24 Single Line Telephones
Transfer”.
Choose a CO Card with the configuration that meets your requirement for CO trunk ports, keeping in mind the
maximum CO Trunk Port capacity of the system you are installing.
The maximum CO Trunk Ports supported are 128.
Connectors
The CO Card has RJ45 connectors, with 4 CO ports on each connector. A multi-pair, MDF cable is supplied for
each connector on the card.
LED
The CO Cards have a single tri-color LED to indicate:

• the status of a selected Trunk port during normal functioning of the system.
You can assign the LED to any CO port on the card which you want to monitor24.
Installing the CO Card

For CO connectivity, you must install at least one of the above mentioned CO Cards in the system.
1. Take all the necessary precautions prescribed for handling the cards and electronic equipment. Make sure
that power supply is turned off before you begin the installation of the card. Put on an electrostatic-
discharge preventive wrist strap/belt and use a grounding mat.
24. To assign the LED to a selected port for monitoring its functioning, you must enter SE mode and dial the SE Command 7902-Slot-
LED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to
which the LED is to be assigned to monitor its functioning. LED Number is the number of the LED on the card, which will monitor
the port.

2. Unpack the CO card and check the package contents.
3. Select any free (empty) slot from the Universal Slots. Unscrew and remove the filler bracket of the empty
slot. Preserve the filler bracket for future use!
4. Insert the CO Card into the guide rails of the free slot you selected for the card. The connectors on the card
should make perfect contact with those of the slot on the backplane motherboard.
5. Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
in place with the two screws provided.
If installing more than one CO Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots.
6. Use the cables supplied for each connector on the CO Card to connect the Trunk Lines with the Main
Distribution Frame.
Refer the illustrations below for the pinout details of the connectors on each card.

ETERNITY ME Card CO16
L1
RJ45-1 Blue - (Blue & White) CO 01
Orange - (Orange & White) CO 02
1 Green - (Green & White) CO
Brown - (Brown & White) CO 04

L1
1 Orange - (Orange & White) CO 02
Green - (Green & White) CO
RJ45-3 Unused
3
RJ45-4 Unused
4

L1
7. Plug in the RJ45 end of the Trunk Card cables into the respective connectors.
8. Terminate the free end of the CO Card cable into the punch down blocks of the Krone modules designated
for 'Trunk Lines' on “The Main Distribution Frame (MDF)”.
Trunk cables from the system are to be connected with the Trunk Lines from the PSTN/CO terminated on
the MDF. Each wire-pair form the CO Port must be terminated on the bottom of the Krone Connector, while
the wire-pair of the trunk line from the CO Network to be connected to this port must be terminated on the
top of the Krone Connector.
Refer the topics “The Main Distribution Frame (MDF)” and “Terminating Trunk and Extension Cables on
the MDF”.
9. Repeat these steps to install other CO Cards, if applicable.
10. If you have completed all other installation tasks, power ON the system.

LED Pattern of the CO Card
Auto Upgradationa
Initialization
1 sec GREEN - 1Sec

Stand-by task ORANGE
Orange
Errors
a. Done by the boot loader application.
Status of Selected CO Port
Selected CO's data are transmitted RED Togglea on each event

to CPU Card
Selected CO's data are received from RED Toggleb on each request from Master
CPU Card
a. The current LED state will remain the same until the next event is received from the application on
the CO Port. For example, if the current LED state is Green/Red ON, on the next event, the LED
will be turned OFF. It will remain OFF until the next event occurs. When the next event is received
it will be turned Green/Red ON again. This process continues.
b. Same as above note.
J2 & J4 AB NA

The BRI Card
The BRI Card provides the interface to connect system with ISDN BRI Lines. The BRI lines may be from a public
ISDN exchange, a private ISDN exchange.
BRI Cards of ETERNITY LENX
ETERNITY ME Card BRI8 8-Port card to connect 8 ISDN BRI Lines or ISDN Compatible Devices
The maximum number of BRI lines supported are 32.
Connectors
The BRI cards have RJ45 connectors. The ETERNITY ME BRI8 card has 8 RJ45 connectors for 8 BRI ports.
The ETERNITY ME BRI4 card has 4 RJ45 connectors for 4 BRI ports. A separate cable is supplied for each
connector.
LEDs
The ETERNITY ME BRI8 and ETERNITY ME BRI4 has 4 LEDs.
ISDN BRI - Installation Scenarios

Most ISDN Service Providers also provide the NT1 device along with the BRI line. The BRI Line from the ISDN
central office is terminated on the NT1 on the Customer's Premises, as illustrated below.
Where,
• U Interface = between the NT1 equipment and the ISDN central office.
• S/T Interface = between the ISDN user equipment, in this case, ETERNITY LENX and the Network
Interface Equipment (NT1).
The BRI line is terminated on the NT1. The S/T interface of the NT1 is connected to BRI port of the ETERNITY
LENX.

TE and NT Modes
In this illustration, the BRI line from ISDN Service Provider is directly connected to BRI port of the ETERNITY LENX
via the NT1 device. Here, the ETERNITY LENX is the Terminal Equipment, so the BRI Port must be programmed
to work in the TE mode.
When an ISDN Phone is to be connected to the BRI port of ETERNITY LENX, the BRI port must be programmed to
work in NT mode.
When a BRI port of another ISDN System is to be connected to the BRI port of the ETERNITY LENX, in such a
configuration, you may configure
• the BRI port of the other ISDN System in the TE mode and the BRI Port of the ETERNITY LENX in the NT
mode.
OR
• the BRI port of the other ISDN System in the NT mode and the BRI Port of the ETERNITY LENX in the TE
mode
Also refer the topic “Configuring BRI Trunks” to know more.
Types of BRI Configuration

There are two types of configurations in BRI: Point-to-Point Configuration and Point-to-Multipoint Configuration.
Each of these is discussed below.
Point-to-Point Configuration
The maximum distance between the NT (Network Termination, NT1 or NT2) and a single Terminal Equipment, in
this case ETERNITY LENX, can be up to 1 kilometer.
Point-to-Multipoint Configuration
A maximum of 8 ISDN equipment can be connected on a single BRI Bus line in a Point-to-Multipoint configuration.
Further, two configurations are possible in a Point-to-Multipoint configuration:
a. Short Passive Bus Configuration
b. Extended Passive Bus Configuration

Short Passive Bus Configuration
Where,
TE = Terminal Equipment or ISDN device (End user device)
NT = Network Termination provided by the ISDN Service Provider
d = distance from NT to the last TE equipment.
In a Short Passive Bus Configuration,
• A maximum of 8 TEs or ISDN devices can be connected to a single NT on a bus up to 200 meters from the
NT.
• 100 Terminal Resistance is required to be inserted at the NT side as well as the last TE Equipment as
shown in the figure.
• Using this configuration, any subscriber from ETERNITY LENX can access a BRI line and can make
outgoing calls. At the same time, another subscriber from ETERNITY LENX or any ISDN phone shown in
the figure can make outgoing call from the same BRI. In the same way, incoming calls are possible on the
same BRI.
• Only two simultaneous speech paths can be established, as BRI supports 2 voice channels only.
• This configuration is useful on the smaller premises, where a single BRI line and multiple ISDN devices are
used.

Extended Passive Bus Configuration
Where,
TE = Terminal equipment of any ISDN Equipment
NT = Network Termination provided by Service Provider
TR Terminal Resistance 100
d = distance from NT to the last TE Equipment
d1 = the total distance from first TE equipment and the last TE equipment.
In an Extended Passive Bus Configuration,
• You can connect only 3 Terminal Equipment or ISDN devices. These devices are grouped together at one
end of the bus, with may extend to a distance of up to 1 kilometer from the NT.
• However, all the 3 Terminal Equipment/ISDN devices must be located within a range of 30 meters, as
• Using this configuration, any subscriber from ETERNITY LENX can access the BRI line and make
outgoing calls. At the same time, another subscriber from the ETERNITY LENX or any ISDN phone shown
in the figure can make outgoing calls from the same BRI. In the same way, incoming calls are possible on
the same BRI.
• This configuration is useful on large premises where a limited number of ISDN devices (maximum 3) are to
be used within a range of 30 meters.
Installing the BRI Card

1. Take all the necessary precautions prescribed for handling the cards and electronic equipment: turn off
power supply, always wear an electrostatic-discharge preventive wrist strap/belt and use a grounding mat.
2. Unpack the BRI Card and check the package contents.
slot. Do not discard the filler bracket! Preserve it for future use!

Setting Orientation Type of BRI Port
4. The BRI Ports can be configured for different applications and can be interfaced directly with the BRI
Network with Terminal Equipment like an ISDN Phone, with an ISDN-System.
To connect the BRI Port to the public network, BRI Port must configured in the TE mode.
To connect ISDN phones, an ISDN System or any ISDN equipment, the BRI Port must be configured in
the NT mode.
By default, BRI Ports are configured in the TE mode.
To set Orientation Type of the BRI Port, under Configuration, open the BRI Configuration link, and
under BRI Parameters, set the Orientation Type.
Inserting Termination Resistance

5. Termination of 100 should be inserted in the following cases:
• When the BRI port is configured in the TE mode and connected in a Point-to-Point configuration as
shown below.
• When the BRI port is configured in the TE mode in a Point-to-Multipoint configuration as shown below.
100 Termination is required on the last Terminal connected on the S0 bus to terminate calls properly.

In a Point-to-Multipoint configuration, 100 termination can be provided on either of the following:
• Last TE equipment
• Last point of the bus bar where the last TE equipment is connected.
• When BRI port is configured in the NT mode.
• If the S0 bus itself supports Terminating resistors, Termination Resistance need not be inserted when
• BRI Port is configured as TE and connected in a Point-to-Point Configuration as illustrated above.
• BRI Port is configured as NT.
• Termination need not be inserted if the BRI port of ETERNITY LENX (configured in TE mode) is
connected as any terminal other than the last terminal on the S0 bus (in a Multi-point configuration).
Termination in TE Equipment (BRI Port)

6. To set the 100 termination on the BRI port set the Jumpers on the BRI Module (daughter-board) to
the position described below:
Module 2 (M2) Module 3 (M3)
BRI Port 1 BRI Port 2 BRI Port 3 BRI Port 4

Function
Jumper Position Jumper Position Jumper Position Jumper Position
J6 J8 J7 J9 J6 J8 J7 J9
To insert 100
AB AB AB AB AB AB AB AB
termination
To remove 100
BC BC BC BC BC BC BC BC
termination

Function
J6 J8 J7 J9 J6 J8 J7 J9
To insert 100
termination
To remove 100
termination
By default, Termination Resistance of 100 is set on the BRI port (the Jumpers are in AB position).

Termination in the Bus Bar
7. 100 termination resister can be connected between TX and RX, between pin number 3-4 and 5-6 in the
RJ45 connector as illustrated below.
1 RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
Tx 3
100
Rx 4
Rx 5
100
Tx 6
As shown in the application diagrams for Point-to-Multipoint connectivity, each ISDN TE device is
connected in a Bus Bar, which may be Short Passive Bus Bar configuration or an Extended Passive Bus
Bar configuration.
Illustrated below is the connection diagram of two ports connected with each other on the same BRI bus
bar.
1 1 RJ45 Connector
ports on BRI Bus
Bar to which the
3 3
ISDN TE
4 4 Equipment is
connected
5 5
6 6
8 8
• The above figure shows the connection details of two ports on the BRI Bus Bar. Similarly, you can
connect 8 ports on the Bus Bar, keeping in mind the Termination Resister for the NT and the Last TE
on the Bus bar.
• Pin number 3, 4, 5 and 6 of the RJ45 connector are used for connectivity.
• Pin number 3 and 6 are used for Transmit (Tx) and pin number 4 and 5 are used for Receive (Rx) from
the ISDN TE side.
• Pin number 3 and 6 are used for Receive (Rx) and pin number 4 and 5 are used for Transmit (Tx) from
the NT side.
Feeding Power to the Terminal

8. When the BRI Port of the ETERNITY LENX is used as BRI-NT, you can feed power to the terminal
equipment connected to the BRI-NT Port from the ETERNITY LENX.
To do this,
• Enable Feed Power on the BRI Port. For instructions see Power Feed under “Configuring BRI Trunks”.

• By default, the Jumpers are set in AB position to feed power through Tx and Rx wires (Phantom
Power). If you want to feed power through a separate pair of wires, you may change the position of the
Jumpers on the BRI module as mentioned in the table below.

Function
J4 J5 J2 J3 J4 J5 J2 J3
To feed power on Tx and Rx

wires (Phantom Power) AB AB AB AB AB AB AB AB
To feed power on separate

pair of wires

Function
J4 J5 J2 J3 J4 J5 J2 J3


pair of wires
• The maximum power that can be fed to a single BRI port is 50mA.
• From signaling point of view, a maximum of 8 terminal equipment can be connected on the BRI port
configured in the NT mode.
• The number of ISDN Terminals that can be connected on the BRI port configured in the NT mode
depends on the power consumed by the ISDN terminals.
9. Insert the BRI Card into the guide rails of the free slot you selected for the card. The connectors on the
card should make perfect contact with those of the slot on the backplane motherboard.
Press down the lever on the card mounting brackets to secure the card in its slot. Fix the mounting bracket
If installing more than one BRI Card, it is not necessary to insert the other cards in subsequent slots. Any
card can be inserted in any of the Universal Slots. Remember to set the Orientation Type, Termination
Resistance and Power Feed, as required.
10. Use the straight cables supplied for each connector on the BRI card to connect the BRI Ports to the NT1
device supplied by your ISDN service provider. Refer the configuration and pinout details given below for
guidance.

Configuration details of the U interface (RJ-45) at NT1
Pin Number Pin Details
4 Tx
5 Rx
Configuration details of the S/T interface (RJ-45) on NT1
3 Rx1
4 Tx1
5 Tx2
6 Rx2
Pinout and Cable Details of BRI4 Port in TE Mode
Pin Color Connection
1 Orange-White Not connected
2 Orange Not connected
3 Green-White TxA
4 Blue RxA
5 Blue-White RxB
6 Green TxB
7 Brown-White V-
8 Brown V+
Pinout and Cable Details of BRI4 Port in NT Mode
3 Green-White RxA
4 Blue TxA
5 Blue-White TxB
6 Green RxB
7 Brown-White V-
8 Brown V+

The following diagram shows how to connect a BRI Line to the BRI port in the TE mode.
11. If you have completed all other installation tasks, you may turn ON the system and observe the Reset
Cycle and the LED pattern of the BRI Card.
LED Pattern of the BRI Card
• The BRI8 Card has 8 LEDs: L125, L2, L3, L4, L526, L6, L7 and L8. These display the status of each port.
• The BRI4 Card has 4 LEDs: L127, L2, L3 and L4. These display the status of each port.
The LEDs show the Status of the Ports as summarized in the table below:
Port Status LED Color LED Cadence
Port is not active RED Continuously ON
Port is active GREEN Continuously ON
25. This LED keeps blinking. It displays the system heart bits, at the rate of 100ms. It will remain OFF for 100ms and will show the sta-
tus of port 1 for the next 100ms.

The T1E1PRI Card
The ETERNITY ME T1E1PRI Card provides the interface to connect the system to ISDN Network.
When connected to T1 carrier lines, the Card supports the following signaling types:
• PRI
• Robbed Bit Signaling
• E&M
When connected to E1 carrier lines, the Card supports the following signaling types:
• PRI
• E&M
The T1E1PRI Card is available in the following configurations:
T1E1PRI Card for ETERNITY LENX
ETERNITY ME Card T1E1PRI 2-Port card with QSIG support to connect 2 ISDN T1/E1 PRI Lines or ISDN
Dual Compatible Devices
ETERNITY ME Card T1E1PRI 1-Port card with QSIG support to connect 1 ISDN T1/E1 PRI Line or ISDN
Single Compatible Device
The maximum number of PRI Lines supported are 24.
Connectors
The T1E1PRI card has an RJ45 Connector for each port. The ETERNITY ME T1E1PRI Dual card has 2 RJ45
Connectors for the two ports, while the ETERNITY ME T1E1PRI Single card has a single RJ45 Connector.
A cable with RJ45 plugs on both ends is supplied for each connector.
LEDs
The ETERNITY ME T1E1PRI Dual Card has four LEDs: L1, L2, L3 and L4.
The ETERNITY ME T1E1PRI Single Card has two LEDs L1 and L2.
Installing the T1E1PRI Card

1. Before installing the card, take the necessary precautions prescribed for handling the cards. Always wear
an electrostatic-discharge preventive wrist strap and use a grounding mat. Make sure the power supply is
turned off.
2. Unpack the T1E1PRI Card and check the package contents.

slot. Do not discard the filler bracket.
Setting Line Termination Resistor

4. The default positions of SW3, SW4, SW6, SW7 switches should be as follows:
Pin1 Pin2 Pin3 Pin4 Termination Resistance ()
OFF OFF OFF ON 0 (Default)
Do Not Change the positions of any of these switches!
5. By default, termination resistance of PRI port is set as 120, which is for E1 connectivity.
• To use the PRI Port for T1 connectivity, termination resistance must be changed to 100.
• Use DIP Switch SW5 to change the Termination Resistance of PRI Port 1. Set the Pins of SW5 as
shown below:
Pin-1 Pin-2 Pin-3 Pin-4 Resistance
OFF OFF ON OFF 120 (for E1)
OFF ON OFF OFF 100 (for T1)
• If using the ETERNITY ME T1E1PRI Dual Card, use DIP Switch SW2 to change the Termination
Resistance of PRI Port 2. Set the Pins of SW2 as shown below:
6. Insert the T1E1PRI Card into the guide rails of the free slot you selected for the card. Make sure that the
connectors on the card make perfect contact with those of the slot on the backplane motherboard.
7. Now, press down the levers on the card mounting brackets to secure the card in its slot. Fix the card in
place with the two screws provided.
Connecting ISDN T1/E1 PRI Lines

8. Use the cable supplied with the T1E1PRI Card to connect the system to the T1/E1 PRI network interface
equipment (modem), which is usually supplied by your ISDN Service Provider along with the PRI line.

The diagram below illustrates this.
• Most Service Providers insist on connecting an ISDN modem at both the ends of the PRI line—one at
the Local Exchange and other at the Customer's Premises.
• At the Customer's Premises, the PRI line is terminated on the HDSL interface of the modem.
• The DTE interface of the modem is to be connected to the PRI port (RJ-45 connector on the
ETERNITY ME T1E1PRI Card).
9. Plug in one end of the RJ45 cable supplied with the card into the card's connector. Plug the other end of
the RJ45 cable into the Network Termination Unit.
10. Refer the following pin details for connecting the Network Termination Unit with the system.
Pin details of HDSL Interface of the G.703 Modem. (HDSL Network Termination Unit)
1 Line A
2 Line A
3 Not used
4 Line B
5 Line B
6 Not used
7 Not used
8 Not used
Pin details of DTE Interface of G.703 Modem. (HDSL Network Interface Unit)
1 TX1 (Tip)
2 TX2 (Ring)
3 Not used
4 RX1 (Ring)

5 RX2 (Tip)
6 Not used
7 Not used
8 Not used
Most of the HDSL Network Termination Unit manufacturers use these connectors. But you are advised to
read the installation guide of the HDSL Network Termination Unit being used by you.
Pin details of the System’s T1E1PR1 Port

The T1E1PRI Port of the system terminates in an 8-pin RJ45, female connector and is wired according to the table
below.
Tx1 (Ring) Tx2 (Tip)

4 5
NC NC
3 6
Rx2 (Tip) NC
2 7
Rx1 (Ring) NC
1 8
The cable wires may have to be crossed depending on the pinout of the DTE Interface of the modem.
11. Repeat the same steps to install another card. It is not necessary to install the other T1E1PRI Cards in a
sequence. Any card can be installed in any of the slots.
12. If you have completed all other installation tasks. Power the system. After the Reset Cycle, observe the
LED patterns of the T1E1PRI Card.
LED Patterns
• L1 and L2 are assigned to PRI Port 1 (PRI#1)

• L3 and L4 are assigned to PRI Port 2 (PR1#2)
• L128 shows Card Heart Bit as well as status of Port 1.
The ETERNITY ME T1E1PRI Single Card has two LEDs: L1 and L2.
Given below are the LED Patterns defined for indicating port states in the signaling types supported by the
ETERNITY LENX.
28. This LED keeps blinking. It displays the system heart bits, at the rate of one second. It will remain OFF for one second and will
show the status of port 1 for the next one second.

Port Active Mode
Signaling Type: E1-PRI
LED1/LED3 Pattern:
Port Status Color Cadence
Layer 1 established successfully GREEN Continuous ON
CRC4 Alarm GREEN 100ms ON-100 ms OFF
BFA Alarm RED 500ms ON-500 ms OFF
LOS Alarm RED Continuous ON
LED2/LED4 Pattern:
RAI Alarm RED 500ms ON-500 ms OFF
AIS or LOS Alarm RED Continuous ON
Signaling Type: E1-CAS
LED1/LED3 Pattern:
MFA Alarm RED 100ms ON-100 ms OFF
BFA Alarm RED 500ms ON-500ms OFF
LED2/LED4 Pattern:
Y-Bit Alarm GREEN 100ms ON-100 ms OFF
AIS16 Alarm RED 100ms ON-100 ms OFF

Signaling Type: T1-RBS or T1-PRI
LED1/LED3 Pattern:
No Alarm GREEN Continuous ON
BFA Alarm or MFA Alarm RED 500ms ON-500 ms OFF
AIS Alarm RED 100ms ON-100 ms OFF
LED2/LED4 Pattern:
RAI or LOS Alarm RED Continuous ON
2. Port Maintenance Mode

LED1/LED3 Pattern:
Maintenance Mode RED -GREEN 500 ms RED-500 ms GREEN
LED2/LED4 Pattern:
Near end loop back wait before activate RED 100ms ON-100 ms OFF
Near end loop active RED Continuous ON
Near end loop back wait before deactivate RED 500ms ON-500 ms OFF
Far end loop back wait after activate GREEN 100ms ON-100 ms OFF
Far end loop active GREEN Continuous ON
Far end loop back wait after deactivate GREEN 500ms ON-500 ms OFF
3. Port Disable Mode

LED1/LED3 Pattern:
Port Disable RED Continuous ON
LED2/LED4 Pattern:
Port Disabled OFF OFF

J5a AB Dual T1E1
BC Single T1E1
J6 AB JTAG Mode
BC (default) Embedded ICE
J7 AB Internal Boot
BC (default) External Boot
J8 & J12 AB (default) Debug
BC UART
J9, J10 & J11 BC (default) Normal
AB Boot
a. In T1E1 PRI Dual Card, default jumper position will be AB. In

T1E1 PRI Single Card, default jumper position will be BC.

The E1FO Card
The ETERNITY LE E1FO Card provides the interface to connect the ETERNITY to the ISDN PRI Network. For E1
carrier lines the card supports the signaling types PRI, Q-Sig and CAS. For T1 carrier lines the card supports the
signaling types PRI, Q-Sig and Robbed Bit Signaling. The T1/E1 ports can be set in Terminal or Network mode.
The E1FO Card supports Copper and Fiber Optic (FO) interfaces. At a time, either the Copper interface or the FO
interface can be used. The FO interface supports only Single-mode (Mono mode) fiber connectivity and will work
within a range of 30km.
E1 connectivity is supported over, the Copper interface as well as the Fiber Optic interface. The T1 connectivity is
supported over the Copper interface only.
The E1FO Card is available in the following configurations:
E1FO Card for ETERNITY LENX
ETERNITY ME Card E1FO Dual 2-Port card to connect 2 ISDN T1/E1 Lines.
ETERNITY ME Card E1FO Single 1-Port card to connect 1 ISDN T1/E1 Line.
Connectors
The E1FO Dual card has two RJ45 Connectors and two Fiber Optic connectors. The E1FO Single card has one
RJ45 Connector and one Fiber Optic connector.
LEDs
The ETERNITY ME E1FO Dual Card has four LEDs: L1, L2, L3 and L4.
The ETERNITY ME E1FO Single Card has two LEDs L1 and L2.
Installing the E1FO Card

turned off.
2. Unpack the E1FO Card and check the package contents.

shown below:
• If using the ETERNITY ME E1FO Dual Card, use DIP Switch SW2 to change the Termination
6. Insert the E1FO Card into the guide rails of the free slot you selected for the card. Make sure that the
Connecting the T1/E1 Lines
Copper Interface Connectivity

8. Use the cable to connect the T1/E1 Port to the T1/E1 network interface equipment (modem), which is
usually supplied by your ISDN Service Provider along with the PRI line.

• The DTE interface of the modem is to be connected to the PRI port (RJ45 connector on the Matrix
ETERNITY ME CARD E1FOPRI Single).
9. Plug in one end of the RJ45 cable into the card's connector. Plug the other end of the RJ45 cable into the
Network Termination Unit.
1 Line A
2 Line A
3 Not used
4 Line B
5 Line B
6 Not used
7 Not used
8 Not used
1 TX1 (Tip)
2 TX2 (Ring)
3 Not used
4 RX1 (Ring)

5 RX2 (Tip)
6 Not used
7 Not used
8 Not used
Pin details of the T1/E1 PRI Interface
The T1/E1 Interface of the ETERNITY ME CARD E1FOPRI Single terminates in an 8-pin RJ45, female connector
and is wired according to the table below.

4 5
NC NC
3 6
Rx2 (Tip) NC
2 7
Rx1 (Ring) NC
1 8
Fiber Optic Interface Connectivity
• Fiber Optic interface supports E1 connectivity only.
• The Fiber Option (FO) interface supports only Single-mode (Mono mode) fiber connectivity and will
work within a range of 30km.

11. If your service provider provides Fiber Optic connectivity or you have an existing Fiber Optic infrastructure,
you can connect the FO line to the E1FO interface of the E1FO Card.
To use Fiber Optic interface, make sure you select Line Coding Mechanism as NRZ (Fiber Optic). For
details, see “Configuring E1 Trunks”.
12. Repeat the same steps to install another card. It is not necessary to install the other E1FO Cards in a
LED patterns of the E1FO Card.
LED Patterns

The ETERNITY ME E1FO Single Card has two LEDs: L1 and L2.
ETERNITY LENX.
1. Port Active Mode

LED1/LED3 Pattern:
LED2/LED4 Pattern:

LED1/LED3 Pattern:
LED2/LED4 Pattern:
LED1/LED3 Pattern:
LED2/LED4 Pattern:

LED1/LED3 Pattern:

LED2/LED4 Pattern:

LED1/LED3 Pattern:
LED2/LED4 Pattern:

The E&M Card
The E&M Card of the system provides the interface for analog trunking to connect various communication
equipment telephone switches, Routers, Leased Lines, etc. using Tie-Lines.
The E&M Card is required for the following applications:
• Power Line Carrier Communication (PLCC) Networks, where several systems are connected with each
• “Closed User Group (CUG)”, where several systems are connected with each other through E&M tie
lines30.
• System expansion, where two systems are connected with each other with E&M tie lines.
• Connecting remote systems over E&M tie lines.
Refer the topic “E&M Connectivity” to know more.
An E&M Port can be programmed to behave as a Trunk Interface, a Subscriber (Station) Interface or both, as a Tie
Line with the dual personality of a Trunk and a Subscriber.
The E&M Card supports

• E&M Interface - Types IV and V
• Speech Interface - Two-wire and four-wire.
• E&M Trunk Seizure Type31: Immediate, Immediate + Wink, Immediate with Ack, Immediate with
Ack+Wink, Seizure Pulse, Seizure Pulse + Wink, Express, and Compander Control Signal.
• Address Signaling: Pulse dial (Pulse 10PPS, Pulse 20PPS) and Tone Dial (DTMF).
The E&M Card is available in the following configurations:
E&M Card for ETERNITY LENX
ETERNITY ME Card E&M8 8-port card to connect 8 E&M Tie Lines
The maximum number of E&M ports supported are 64.
Connectors
The E&M Card has RJ45 Connectors. A separate MDF cable is supplied for each connector.
30. The Systems in a “Closed User Group (CUG)” can be connected over ISDN T1E1PRI Lines as well. Refer the topic Closed User
Groups to know more.
31. This is the line protocol that defines how the equipment seizes the E&M trunk. Also referred to as Start Dial Supervision Signaling
Protocol.

LEDs
The ETERNITY ME Card E&M8 has eight tri-color LEDs. The ETERNITY ME Card E&M4 has 4 LEDs, to indicate
the functioning of the ports.
Installing the E&M Card

An E&M port can be programmed to take on the function of:
• a Station - works like an extension interface, receiving incoming calls.
OR
• a Trunk - works like a trunk interface when any of the extensions of the system makes an outgoing call
through it.
OR
• a Tie Line - takes on a dual personality: functioning as both as an extension and a trunk. The E&M port
works like an extension interface for incoming calls. It works like a trunk interface when any extension
makes an outgoing call through it.
This dual function is used in systems that are used as Transit Exchanges as in a PLCC Network. Read
“PLCC-An Introduction” to know more.
You cannot connect a trunk line or an SLT / DKP to an E&M port.
1. Have the necessary wiring for the E&M Analog trunk in place. Take the necessary safety precautions
before you begin handling the card; switch off power supply and always wear an antistatic wrist strap and
use a grounding mat.
2. Unpack the E&M Card and check the package contents.
3. The E&M Card supports E&M Interface Type IV and Type V connection. To select the appropriate
Interface Type out of the two, you need to change the Jumper Settings.
Refer the table below to select the desired Interface Type and Speech Interface.
J1 and J2 AB Type IV E&M Interface
BC Type V E&M Interface
• By default all the E&M Ports are set to support Type-IV.
• To select the Type-V connection for the E&M Port, set Jumpers J1 and J2 (located on the E&M
module) in BC Position.

4. Select the speech interface - 2-wire speech or 4-wire speech - as required, by changing the jumper
settings. Refer the table below.
J3 and J4 AB 4-wire speech interface
BC 2-wire speech interface
• By default all the E&M Ports are set to support 2-wire Speech Interface.
• To select 2-wire speech interface for the E&M Port, set Jumpers J3 and J4 (given on E&M module) to
BC Position.
• To select 4-wire speech interface for the E&M Port, set Jumpers J3 and J4 on E&M module to AB
Position.
Keep Jumper number J5 in BC position and Jumper number J6 in BC/Open position.
5. Now, select a free slot for the E&M Card. Unscrew and remove the filler bracket by pushing up the levers
on the bracket. Preserve the filler bracket for future use.
6. Insert the E&M Card into guide rails of the empty slot. Make sure the connectors on the card make perfect
contact with those on the backplane motherboard. Secure the card by pressing down the levers and fix the
bracket with the screws provided with the card.
7. Connect the cables supplied with the E&M Card into the RJ45 connectors on the E&M Card.
8. Connect the free ends of the cables into the E&M Ports of the other System/PBX/Router/Tie Line
equipment by appropriate crossing of the wires.
Refer the following pin-out details for the E&M Card and for each Interface and Speech Interface Type.

Pinout details of ETERNITY ME Card E&M8 - RJ45 Connectors
01 Open Gray 01
02 SB Green-White
03 M OUT Green
04 RX SPCH A Orange-White
05 SPCH A Blue
06 SPCH B Blue-White
01 Open Gray 02 07 RX SPCH B Orange
02 SB Green-White 08 E IN Brown-White
03 M OUT Green 09 BGND Brown
05 SPCH A Blue
07 RX SPCH B Orange
08 E IN Brown-White
09 BGND Brown 01 Open Gray 03
02 SB Green-White
03 M OUT Green
05 SPCH A Blue
05 SPCH A Blue
07 RX SPCH B Orange
08 E IN Brown-White
09 BGND Brown
01 Open Gray 05
02 SB Green-White
03 M OUT Green
05 SPCH A Blue
05 SPCH A Blue
07 RX SPCH B Orange
08 E IN Brown-White
02 SB Green-White
03 M OUT Green
05 SPCH A Blue
05 SPCH A Blue
07 RX SPCH B Orange
08 E IN Brown-White
09 BGND Brown

L1 L3
L2 L4
Pin No. Connection Colour H/w Port Offset
01 Open Gray 01
02 SB Green-White
03 M OUT Green
1 RJ45-1 05 SPCH A Blue
Pin No. Connection Colour H/w Port Offset 06 SPCH B Blue-White
07 RX SPCH B Orange
01 Open Gray 02
08 E IN Brown-White
02 SB Green-White
09 BGND Brown
03 M OUT Green
10 CCC Gray-White
05 SPCH A Blue
06 SPCH B Blue-White RJ45-2 2
07 RX SPCH B Orange Pin No. Connection Colour H/w Port Offset
08 E IN Brown-White
10 CCC Gray-White 02 SB Green-White
03 M OUT Green
07 RX SPCH B Orange
01 Open Gray 04
08 E IN Brown-White
02 SB Green-White
09 BGND Brown
03 M OUT Green
10 CCC Gray-White
05 SPCH A Blue RJ45-4
06 SPCH B Blue-White 4
07 RX SPCH B Orange
08 E IN Brown-White
09 BGND Brown
10 CCC Gray-White

2 Wire, Type IV E&M Connection
2 Wire / Type IV E&M Connection
M M
SB SB
E E
SG SG
SP(RxA) NC NC SP(RxA)
SP(RxB) NC NC SP(RxB)
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS
One E&M Port of System-A One E&M Port of System-B
2 Wire, Type V E&M Connection

2 Wire / Type V E&M Connection
M M
SB NC NC SB
E E
SG SG
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS

M M
SB SB
E E
SG SG
SP(RxA) SP(RxA)
SP(RxB) SP(RxB)
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS

M M
SB NC NC SB
E E
SG SG
SP(RxA) SP(RxA)
SP(RxB) SP(RxB)
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS
9. If you are connecting two PLCC EPAX in a Power Line Carrier Communication Network Compander
Control Signal (CCS) Connection should be made as illustrated in the block diagram below for any of the
four combinations of E&M and Speech Interfaces illustrated in the previous step.
M NC NC M
SB NC NC SB
E NC NC E
SG NC NC SG
SP(A) NC NC SP(A)
SP(B) NC NC SP(B)
CCS CCS
PLCC PLCC
One E&M Port of System-A Panel Panel One E&M Port of System-B
Compander Control Signal (CCS) is a special type of signal used by Power Line Carrier Communication
Networks to improve quality of speech transmission. The PLCC network expects this signal from the
system when speech is established and the E&M Card supports this facility. The system sends CCS signal
to the PLCC panel.
• When the E&M port is used as an Endpoint; the system sends a CCS to the PLCC panel while making
an outgoing call through the E&M port or when a call is received at the E&M port.
• When the E&M port is used for Transit Exchange; the system sends a CCS to the PLCC panel while
there is a Transit call through the E&M port.
pattern of the E&M Card.

LED Pattern of the E&M Card
Stage LED Color LED Cadence
Initialization
At Power ON RED L1 to L8 glow one after the other in a sequence,

ON 100ms -OFF.
L1 ON 100ms-OFF,
L2 100ms ON-OFF,
L3 100ms ON-OFF,
L4 100ms ON-OFF……L8 100ms ON-OFF
GREEN L1 to L8 glow one after the other in a sequence,

ON 100ms -OFF.
L1 ON 100ms-OFF,
L2 100ms ON-OFF,
L3 100ms ON-OFF,
Stand-Bya GREEN, ORANGE L1 toggles GREEN ON 1 sec,

ORANGE ON 1 sec
Normal (Port Event)
M-Wire High GREEN LED of the Port continuously ON
M-Wire Low LED of the Port continuously OFF
E-Wire High RED LED of the Port continuously ON
E-Wire and M-Wire High ORANGE LED of the Port continuously ON
Errors
Controller RAM failure ORANGE All LEDs Toggle at 1 sec
External RAM failure ORANGE All LEDs Toggle at 2 sec
Eprom failure ORANGE All LEDs Toggle at 3 sec
Invalid Slot detected ORANGE All LEDs Toggle at 6 sec
a. Waiting to be detected by Master Card.

The Mobile Card
The Mobile Card interfaces the system with GSM/3G networks. It routes calls made and received over mobile
networks, like a mobile handset.
The Mobile Cards are available in 2G and 3G variants.
The Mobile Card does not support GPRS features, Fax and Data services, network supported services,
except CLIR and USSD.
The Mobile Card for ETERNITY LENX
ETERNITY ME Card GSM8 8-port card to connect to 8 GSM networks (8 SIM Cards can be installed)
ETERNITY ME Card GSM4 4-port card to connect to 4 GSM networks (4 SIM Cards can be installed)
ETERNITY ME Card GSM8 8-port card to connect to 8 GSM networks with 3G support (8 SIM Cards can be
3G installed)
ETERNITY ME Card GSM4 4-port card to connect to 4 GSM networks with 3G support (4 SIM Cards can be
3G installed)
Just like mobile handsets, each Mobile Port has a unique IMEI (International Mobile Equipment Identity) number,
pasted on the mobile engine.
The maximum Mobile ports supported are 64.
SIM cards from different service providers can be used.
Antenna
There is a single rooftop (RT) antenna for four GSM ports. A splitter connects all the four ports on the card into a
single antenna. An antenna cable is also provided, giving you the flexibility to move the antenna to another position
(in case of weak signal).
Personal Identification Number (PIN)

The SIM cards can be protected from unauthorized use by programming a Personal Identification Number (PIN) on
the SIM. If the wrong SIM PIN is entered thrice in a row, by a user, the SIM card suspects the user and asks for the
Personal Unlock Keyword (PUK).
LEDs
There is a tri-color LED for each mobile port on the card to indicate the functioning of the card and the status of the
ports.
Installing the Mobile Card

To be able to connect the system to GSM/3G networks, you must have one of the above mobile cards installed in
the system.

1. To install the Mobile Card,
• If using a GSM/3G card, get the SIM Card from the GSM/3G service provider of your choice ready. Use
SIM PIN protection, if required.
Disable Call Waiting in the SIM.
2. Make sure that the system is installed at a location where sufficient network coverage is available. The
power supply should be turned off, and you must be wearing an electrostatic discharge preventive wrist
strap and must have a grounding mat, before you begin handling the card.
3. Unpack the Mobile Card and verify the package contents.
ETERNITY ME Card GSM4/GSM4 3G

Enabling PIN Protection on SIM

4. For the GSM/3G Card, enable SIM PIN before installing the SIM card in the system.
• insert the SIM into a mobile handset first.

• enable PIN Protection from the mobile handset.
• change the SIM PIN to 1234 (this is the default PIN for all SIM cards used in the system). Changing the
SIM PIN to '1234' enables you to change the SIM PIN from the Jeeves later (Refer SIM PIN under
“Configuring Mobile Trunks” for instructions).
• remove the SIM from the mobile handset.
If you do not want to use PIN protection, insert the SIM in the mobile handset and disable PIN protection.
Remove the SIM Card from the mobile handset.
5. Insert the SIM card (PIN changed to 1234), with its connector side down into the SIM holder on the Mobile
card. You can insert multiple SIM cards of the same GSM service provider or of different service providers.
6. Insert the Mobile Card into the guide rails of the Universal Slot you have selected for this card. Make sure
that the card is inserted deep enough to make perfect contact with the connectors in the backplane. Now,
press down the levers on the card mount bracket to secure the card in its slot.
7. Connect the antenna provided with the card on the splitter connector on the front panel of the card. You
may also use the antenna cable to place the antenna at another position.
8. Repeat Steps 1-7 to insert another Mobile Card.
9. If you have completed all installations tasks, power the system.
10. Wait for the system to register with the Mobile network. By default, the Mobile ports are set to select and
register with the Mobile networks automatically. Now, observe the LED Patterns of the Mobile Ports.

• At every power up of the system, it takes about 3 minutes for the Mobile ports to get registered with the
network. Once registration with the GSM network is completed, the mobile port can be used.
• Each time the Mobile Port sends a request, such as Registration Request, the system waits for the
duration of the Network Response Timer. This Timer signifies the time for which the Mobile Port waits
for a response from the Mobile network. It is fixed for 150 seconds for all Mobile ports.
LED Pattern of Mobile Ports

The number of tri-color LEDs on the GSM card corresponds with the number of mobile ports on the card.
At Power On: All LEDs will blink 1 second ON and 1 second OFF in the color sequence: Red-Green-Orange until
the Reset cycle is complete.
In the Stand-by state: All LEDs will glow Orange for a second and turn Green for a second, repeatedly.
During normal functioning: The LEDs will various events on the Mobile port in the color and cadence described
in the table below:
Event Color Cadence in msec (1 cadence is of 3000 msec)
Port disabled - LED OFF
Port idle - LED OFF
Port Active (All States other Red Continuous ON

than Ring and Speech)
Ring Event Green 400ms ON-200ms OFF400ms ON-200ms OFF
Speech Green Continuous ON
GSM initialization Orange 200ms ON-200ms OFF-200ms ON-200ms OFF-200ms ON-200ms

OFF-200ms ON-200ms OFF-200ms ON-1200ms OFF (5 blinks)
PUK required Orange 200ms ON-200ms OFF-200ms ON-200ms OFF-200ms ON-200ms

OFF-200ms ON-1600ms OFF-
SIM PIN faulty Orange 200ms ON-200ms OFF-200ms ON-200ms OFF-200ms ON-2000ms
OFF (3 blinks)
SIM Absent Orange 200ms ON-200ms OFF-200ms ON-2400ms OFF (2 blinks)
Network Link Down Orange 200ms ON-2800ms OFF

(Absence of GSM Network)

The Magneto Card
The Magneto Card is used for connecting the system to Magneto Telephones32, which are widely used by the
defense establishments as field phones in front lines, and by other establishments such as railroad companies
(signaling emergencies, crossings, etc.), electric utilities, pipeline companies, who need to have their networks at
places that are too remote to be serviced by public telephone networks.
The Magneto Card lands calls from magneto field telephones on the extensions of the system and places calls from
the extensions of the system on magneto telephones.
Magneto Card for ETERNITY LENX
ETERNITY ME Card Magneto8 8-port card to connect 8 Magneto Phones
The maximum number of magneto ports supported are 16.
Connectors
The Magneto Card has RJ45 connectors. A multi-pair cable is provided for each connector.
LED
The ETERNITY Magneto8 has 8 LEDs for each magneto port supported by the card.
The LEDs indicate the health of the cards during the Reset Cycle and the status of the ports during the normal
functioning of the system.
Installing the Magneto Card

1. Have the necessary wiring for the Magneto Ports in place.
You may install an MDF to connect the Magneto Ports with the Field Telephone wires.
OR
You may connect the wires from the Magneto Field Telephones directly to the Magneto Port.
You are advised to use a separate set of Krone Modules for connecting the Magneto phones to the
Magneto ports of the system
2. Prepare for the card installation by switching off power supply and wearing an electrostatic discharge
preventive wrist strap and use a grounding mat.
3. Unpack the Magneto Card and check the package contents.
32. A magneto telephone is a local battery telephone set, in which signaling current is provided by a magneto hand generator, usually
a magneto. The hand generator, commonly referred to as 'crank', is located on the right hand side of the telephone set and is
turned to produce energy to ring other phones or to signal the CO. The magneto, also called the generator, is used to convert the
mechanical motion via the crank to produce sufficient energy to ring other phones or to signal the CO.

4. Select any universal slot to insert the card. Unscrew the filler bracket and remove it by pushing up the
levers on the bracket.
5. Insert the Magneto card into the guide rails of the free slot. The card's connectors must make perfect
contact with the connectors on the backplane motherboard. Press down the levers of the mounting bracket
to secure the card in its slot and fix the two screws provided with the card on the mounting bracket.
6. Now, plug in the cables supplied with the Magneto Card into the connectors on the card. Terminate the
free ends of the cables into the MDF, if applicable.
Refer to the following block diagram for terminating the cables from the Magneto Card and the wires from
the Magneto Field Telephones.
Connecting Magneto Telephones to the Magneto Card

7. Connect the pairs of wires from the Magneto Field Phones to the appropriate pairs emerging from the
Magneto Card on the MDF. Refer the cable diagram below.
8. Repeat the same steps to install the other Magneto Cards.
9. If you do not have any other Card to insert and have completed the installation procedures, power on the
system and observe the Reset Cycle and the LED Pattern of the Magneto Card.

LED Pattern of the Magneto Card
Auto Upgradation / Auto Downgradation

Card waiting for application RED L1 turned ON for 200ms, turned OFF
simultaneously.
Card is up, loaded with new GREEN L1 turned ON for 200ms, turned OFF
application simultaneously.
Initialization
RED L1 to L8 turned ON 500ms-OFF 500ms
GREEN L1 to L8 turned ON 500ms-OFF 500ms
ORANGE L1 to L8 turned ON 500ms-OFF 500ms
Stand-Bya ORANGE, L1 toggles 1 sec Orange and RED

GREEN,
Normal (Port Event)
Ring (incoming/outgoing call) RED LED of the Port continuously ON
Port Disabled LED of the Port continuously OFF
OFF-Hook (in Speech) GREEN LED of the Port continuously ON
For Outgoing call from Magneto:

When called extension answers the call
For Incoming call to Magneto:

When stopping Ring on Magneto using
MRE key or #
Port Idle LED of the Port continuously OFF
Errors
Invalid Card Configuration Jumper ORANGE All LEDs Flash (250ms ON-250ms OFF) twice,
OFF 3 sec.
Invalid Slot detectionb ORANGE LEDs Flash (250ms ON-250ms OFF) 6 times,
OFF 3 sec.
a. Waiting to be detected by CPU Card.

b. Waiting to be detected by CPU Card.
J1 & J2 AB NA
BC (default) Normal Operation
J3, J5 & J6 AB (default) Normal Operation

The Radio Card
The Radio Interface Card (RIC) adds the Two-way Radio functionality in the system. In Two-way radio, the speech
can be transmitted as well as received by the radio devices such as Radio Phone, Radio Repeater. Such devices
are called Radio Transceivers. The Two-way radio works on High Frequency (HF), Very High Frequency (VHF) or
Ultra High Frequency (UHF).
Radio Cards for ETERNITY LENX
ETERNITY ME Card Radio8 8-port card to connect 8 Radio devices.
The maximum number of radio ported supported are 16.
Connectors
The Radio Card has RJ45 connectors. A multi-pair cable is provided for each connector.
LED
The Radio Card has one tri color LED.
Installing the Radio Card

1. Have the necessary wiring for the Radio Ports in place.
You may install an MDF to connect the Radio Ports with the Radio device wires.
OR
You may connect the wires from the Radio device directly to the Radio Port.
You are advised to use a separate set of Krone Modules for connecting the Radio devices to the Radio
ports of the system.
3. Unpack the Radio Card and check the package contents.
5. Insert the Radio Card into the guide rails of the free slot. The card's connectors must make perfect contact
with the connectors on the backplane motherboard. Press down the levers of the mounting bracket to
secure the card in its slot and fix the two screws provided with the card on the mounting bracket.

6. Now, plug in the cables supplied with the Radio Card into the connectors on the card. Terminate the free
ends of the cables into the MDF, if applicable.
7. Connect the pairs of wires from the Radio devices to the appropriate pairs emerging from the Radio Card
of the system on the MDF. For more details, see “The Main Distribution Frame (MDF)”.
Refer to the Pin-out details given below.
ETERNITY ME Card Radio8
H/w Port
Connector Color Pin Number Signaling
Offset
RJ45-1 Orange & White 1 PTT 01 to 08

to
Orange 2 PTT_RTN
RJ45-8
Green & White 3 Rx-
Blue 4 Tx+
Blue & White 5 Tx-
Green 6 Rx+
Brown & White 7 Unused
Brown 8 Unused
H/w Port
Offset

to
Orange 2 PTT_RTN
RJ45-4
Green & White 3 Rx-
Blue 4 Tx+
Blue & White 5 Tx-
Green 6 Rx+
Brown 8 Unused
8. Repeat the same steps to install the other Radio Cards.
system. Observe the Reset Cycle and the LED Pattern of the Radio Card.
LED Pattern
Card waiting for application RED 200ms ON - 200ms OFF

Card is up, loaded with new

GREEN 200ms ON - 200ms OFF
application
Initialization
RED 500ms ON - 500ms OFF
ORANGE 500ms ON - 500ms OFF
ORANGE,
Stand-By Toggles 1 sec Orange 1 sec Green
GREEN
Port Status
Selected Port's data are transmitted

RED Toggle on each event
to master card
Selected Port's data are received

GREEN Toggle on each event from master
from master card
J4 and J5 AB (default) Normal Operation
In PCB-P-200-80-01-01, for 4-wire speech the Jumpers J1 and J2 on the modules must always be set in
AB position.
In PCB-P-200-80-01-02 onwards only 4-wire speech is possible therefore there are no Jumpers on the
modules.

The Data Card
2Mb streams. Each data port can be mapped to one 2Mb - E1 stream, that is 30 channels. The remaining channels
of E1 can be used for voice applications. The Data Card has a 4-port Ethernet switch on board, which can
aggregate multiple data streams to PCM streams of the system.
The Data Card can be installed in any of the Universal Slots of the system.
Ports and Connectors

The Data Card has an RJ45 Connector for each port. Use the cables supplied with the card for connectivity.
You can connect the cables from the LAN switch to these connectors.
LEDs
The Data Card has one dual color LED.
Installing the Data Card

1. Unpack the Data Card and check the package contents. It is recommended that you switch off the power
supply, before you begin the installation of the card. Always wear an electrostatic discharge prevention
wrist strap/belt and use a grounding mat.
3. Insert the Data Card into the guide rails of the free slot you selected for the card. The connectors on the
4. Use the cable supplied with the card to connect the Data Ports to the Ethernet Network (Switch/PC).
Cycle and the LED pattern of the Data Card.
LED Pattern of the Data Card
Status Color Cadence
Waiting for Master to start Up-gradation

Red Blinking 200ms ON - 200ms OFF
procedure (waits for 10 seconds)
Up-gradation in progress Green Blinking 200ms ON - 200ms OFF
Application run error after up-gradation

Red Continuous ON
procedure
Auto upgradation completed successfully

Green Continuous ON
and in process to start new application

Main application is being executed Red Blinking 1 sec ON - 1 sec OFF
Jumpers
J9 AB (default) External Boot - Normal
BC Internal Boot

SIP Extensions
SARVAM UCS supports up to 2000 SIP/UC Users. The SIP/UC Users function in the same way as DKP/SLT
extensions of the system. SIP/UC Users can make and receive calls to any extension user of the system and to
external numbers over various telecom networks like CO, Mobile, ISDN PRI, BRI, and VoIP33.
You may register any SIP-enabled device — a Matrix UC Client, an IP-phone, a Soft phone, Analog Phone Adapter
— as the SIP User of the system.
The Matrix UC Clients also offer UC functionalities in addition to the SIP functionalities.
The SIP Users register with the CPU Card of the system. Five free SIP Users are provided by default. You may
register any of the SIP-enabled devices except the Matrix UC Clients with these free SIP Users. For registering the
Matrix UC Clients, you must purchase the Matrix VARTA User License. If you require additional SIP Extensions you
must purchase the IP Subscribers License.
The system supports two NX DBM VOCODER64 Modules. You must purchase the module separately. Each NX
DBM VOCODER64 module supports a maximum of 64 VOCODER channels. The Vocoder channels are required
for — VoIP to Non-VoIP calls, VoIP to VMS calls and VoIP to VoIP calls — where transcoding is required.
The system provides 4 pre-activated VOCODER channels by default. To use these channels make sure you have
installed atleast one NX DBM VOCODER64 module. If you require more channels, you can purchase the channel
licenses according to your requirement.
For more information on Licenses — Matrix VARTA User License, IP Subscribers License and VOCODER Channel
License, see “License Management”.
You may connect any Standard Phone or Extended IP Phones of Matrix as SIP Users.
Matrix VARTA WIN200, VARTA ADR100 and VARTA AMP100 can be registered as SIP Users, also offering the
support for UC functionalities.
You may also connect/register the following as SIP Extensions of the system:
• Connect SPARSH VP248, the Extended IP Phone. For instructions, see “Connecting SPARSH VP248 as
Extended SIP Extension”.
• Connect SPARSH VP330, the Touch Screen Extended IP Phone. For instruction,“Connecting SPARSH
VP330 as Extended SIP Extension”.
• Connect SPARSH VP510, the Premium IP Phone. For instruction, “Connecting SPARSH VP510 as
You can register following UC Clients as SIP Users of the system:

• Matrix VARTA WIN200, Unified Communication Client for Windows. For instruction, refer to the MATRIX
VARTA WIN200 User Guide.
• Matrix Mobile UC Clients, as given below:
• Matrix VARTA AMP100, the Mobile UC Client for iPhones. For instruction, refer to the Matrix VARTA
AMP100 User Guide.
33. Calls between VoIP, Public and Private Networks may be subject to Regulation in your country. You may have to configure your
system to allow or restrict call traffic between networks to comply with the telecom regulations of your country. To know more, read
“Logical Partition”.

• Matrix VARTA ADR100, the Mobile UC Client for Android Smartphones/Tablets. For instruction, refer
to the Matrix VARTA ADR100 User Guide.
Refer to “SARVAM UCS Features Supported in Terminals” to know the features supported in each client.
The SIP Users may be registered over WAN or over LAN according to your preference and your IP network
installation scenario. Extended SIP Phones and UC Clients can be registered with SARVAM UCS using IPv4
Addresses only.
You can register the same SIP User from three different locations.
If you register the Extended IP Phone outside the Region/Country selected for SARVAM UCS, the time
and Time Zone dependant features, such as Alarms, Reminders, Time Zone Display, of the phone at each
location will operate according to the Real Time Clock of SARVAM UCS. Also, Access Codes and
Emergency Numbers will work according to the Region/Country selected for SARVAM UCS.
• Connect the Extended IP Phone, or any Open IP Phone to the LAN Switch.
• Register any SIP device (Extended IP phone/ Soft clients or Open IP phone) on the public network as SIP
Extension.
• When you register the Matrix Extended IP Phone with the system, the WAN/LAN port is used for Auto
Configuration as well for Registration of the Extended IP Phones.
• When you register a SIP device other than the Matrix Extended IP Phone on the public network as SIP
Extension, do the following:
• In this SIP device configure the following:
• the Registrar Server Address of SARVAM UCS
• the Registrar Server Port
• the SIP ID
• Authentication ID and Password.
• Configure Port Forwarding for the WAN Port of SARVAM UCS on the Router.

If the SARVAM UCS is connected to a Public Network,
• Connect the Matrix VARTA WIN200, Extended IP Phone, or any Open SIP device to the LAN Switch.
• Register any SIP device (Matrix VARTA UC Clients, Extended IP phone or Open SIP phone) on the public
network as SIP extension.
SARVAM UCS
LAN Switch/ Hub
SIP Extn.
SIP Extn.
Matrix VARTA
WIN200
IP
Router SIP Extn.
If the SARVAM UCS is connected to a Private Network (Behind the NAT),
SARVAM UCS
SIP Extn.
LAN Switch/ Hub
IP
Router SIP Extn.
SIP Extn.
Matrix VARTA
WIN200
• Connect Matrix VARTA WIN200, Extended IP Phones or Open SIP Phones to the LAN Switch
• You may also register any SIP device (Matrix VARTA UC Clients, Extended IP Phone or Open SIP phone)
on the public network as SIP Extension.
When you register the Matrix Extended IP Phone with SARVAM UCS, configure Port Forwarding for the
WAN port of the CPU Card on the Router. The WAN Port is used for Auto Configuration of the Extended
IP Phones.

Connecting SPARSH VP248 as Extended SIP Extension
You are recommended to complete the following steps before connecting the Matrix Extended IP Phone to
SARVAM UCS:
• Decide the location of the Extended IP Phone, whether within the same network or outside, according to
your installation scenario.
If you want to use the DHCP Server for assigning IP Address to the Extended IP Phone, select DHCP
option 224 and Data Type as ‘String’ and program the LAN or WAN IP Address /Domain Name of
SARVAM UCS and SPARSH Port in the format “IP_Address:Port” in your DHCP Server as per your
installation scenario.
• Log in to Jeeves. For instructions, read the topic “Configuring SARVAM UCS”.
• Assign SIP User ID (will work as an extension number) to the Extended IP Phone. For instructions on
assigning SIP ID, see “Configuring SIP Extensions”.
For the SIP User ID you assigned to the Extended IP Phone, you must configure the necessary
parameters in SARVAM UCS so that Extended IP Phone can register as a SIP Extension. For instructions,
see the topic “Configuring SIP Extension Settings as per the Extended Phone Type” under Configuring SIP
Extensions.
Now, follow the steps described below to install the Extended IP Phone. The instructions are common for all models
of the SPARSH VP248. For the purpose of illustration, the premium model, SPARSH VP248P, has been used.
1. Unpack the SPARSH VP248 box and verify package contents.
2. Mount the phone on a desk or wall at a location convenient to you.
• When mounting the phone on the wall,
• Use the mounting template to drill holes of appropriate size and distance. Fix the screw grips in the
holes you drilled.
• Fix two screws in the holes on the wall, ensuring that they are aligned with the Keyhole Slots 1 and
2.
• Use wall plugs, if required, to fix the screws. Leave the screw heads protruding from the wall to fit
into the Keyholes.
• Now, mount the phone on the wall, with the screws fitting into the Keyhole slots.

• Reverse the handset wall mount tab to make sure the handset remains intact when you mount the
• When you mount the phone on a desk,
• You can attach the Foot Stand in two ways as illustrated in the following.
Foot Stand attached at 30° Angle
If you attach the Foot Stand at 50°, the phone will be placed in an almost upright position on your
desk.
3. Connect the Handset to the Phone body.
• Plug the long straightened end of the phone cord into the handset jack at the bottom of the phone
marked with the handset symbol.

• Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
4. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 2.5 mm single
connector into the headset jack headset jack with the symbol on the left side panel of the phone, as
illustrated in the figure below.
Headset
OR

You may plug a headset with an RJ9 connector into the headset port at the bottom of the phone, marked
with the symbol , as illustrated in the figure below.
Foot Stand
Keyhole Keyhole
Slot 1 Slot 2
Headset
Handset
5. Connect the LAN Port of SPARSH VP248 to the LAN Switch/Hub or a Router, according to your installation
scenario.
6. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port of the phone to the LAN Port of the computer.
7. Plug the connector of the Power Adapter in to the power jack at the back of the phone34. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
8. Plug the Power Adapter into a power outlet.
9. Switch ON power supply.
When you power the phone, the boot process will be initiated in the following sequence.
• All keys with LED, including the Speaker key, and the Ringer LED, will glow.
• The LCD display will light up and booting message appears.
• As soon as the ‘Loading...’ message appears on the phone display, press # key.
• Select the firmware Extended - IP Phone. Move the cursor by pressing the DOWN navigation key V.
34. The SPARSH VP248 does not have a power switch.

• When the cursor is placed under the Extended IP Phone, press Enter key.
Select the fir mwar e

Standar d S IP
Extended S I P
• The phone will start loading the Extended IP Phone Firmware. It will display current firmware being loaded.
• After loading the firmware, the phone will prompt you to change Network settings.
Change Network Settings?

Yes
No
• Wait for a few seconds.
If you want to change the Network Settings or Server Settings, press the Enter key. Detailed instructions
for changing the Network Settings of the phone are provided at the end of this topic. See “Network
Settings”at the end of this topic.
• The phone makes DHCP Discovery and fetches its IP Address and Server Address from the DHCP
Server.
On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
configuration files from SARVAM UCS.
• As the phone downloads the configuration files, the file names will appear one by one.
• On successful download of all configuration files, the phone attempts to register with SARVAM UCS.
• On successful registration, the phone will display the current day, date and time, the extension number and
name assigned to the Extended IP Phone.
Network Settings
You can change the network settings of the Extended IP Phone by accessing the Local Menu of the phone. To
move the cursor and scroll through the menu and submenu options, use the following touch sense navigation keys
on your phone.
• The Enter key to make a selection or to complete an action.

• The Up key to move up the Menu.
• The Down key move down the Menu.

• The Forward key move the cursor one character.
• The Back key to move the cursor one character and to return from the submenu to the main menu.
The cursor is a non-blinking underscore that appears under the first letter of the first option in the menu. To make a
selection in the menu, you must move the cursor in the desired direction using the Up, Down, Forward and Back
key. When the cursor is at the desired position, press Enter key to make a selection.
Accessing Network Settings

You can access the Network Settings of the Extended IP Phone in any of the following stages:
1. During start-up, when the phone prompts you to change the network settings after loading the firmware.

Yes
No
You must press the Enter Key to select Yes and access network settings.
2. When the phone is making Network discovery, downloading configuration files, attempting registration.
You must press the Enter Key to access network settings,
3. When the phone is in idle state. You must press the DSS key assigned to ‘Local Menu’.
M on 10 M AY 1 5: 40
2 00 1 Re ce pt i on
Voice Mail Call Back Canc el Mute Conference Transfer
Forw ard DN D Names Redial Release Hold
Local Menu
1 2 abc 3 def
4 ghi 5 jkl 6 mno
7 pqrs 8 tuv 9 wxyz

CA04
0 #
CA03 *
CA02
CA01

4. When you press the Local Menu DSS Key (in idle state) or when you press the Enter key during any
process, the Local Menu appears on your phone display.
LO CAL ME NU
N e t wo r k P a r a m e t e r s
N e t wo r k S t a t u s
You can configure Network Parameters and view Network status from the Local Menu.
Configuring Network Parameters
• In the Local Menu of the phone, select Network Parameters by pressing the Enter Key.
• The Network Parameters submenu appears.
N E T W O R K PA R A M E T E R S
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
C o n n e c t i o n Ty p e
I P A d d r e ss
S u b n e t Ma s k
G a t e w ay A d d r es s
• Use the Down/Up key to reach the desired network parameter and press Enter key to select and change
the settings.
• You can configure all network parameters described below, except the MAC Address.
• To enter a dot in the editable fields — IP Address, Subnet Mask, Gateway Address, DNS Address,
Server Address — press * (Star) key.
• If you want to clear a single digit/character, move the cursor to the right of the digit/character you wish
to clear. Now press the Cancel key. The digit to the left of the cursor will be deleted. If the cursor is to
the extreme left and you press the Cancel key, you will go to the previous menu.
Connection Type
• Select the Connection Type as DHCP, PPPoE or Static according to the IP Addressing scheme of your
network.
If you select DHCP or PPPoE, the phone will be assigned IP Address, Subnet Mask and Gateway
Address, DNS Address Server Address, automatically by the DHCP/PPPoE server.
For PPPoE Connection Type, you must configure the PPPoE User ID and Password provided by the
Internet Service Provider.
If you select Static, you must assign the IP Address, Subnet Mask and Gateway Address to the phone.

IP Address
• If you select Static as Connection Type, enter the static IP Address to be assigned to the phone.
Enter the desired Static IP Address by pressing the digit keys.
To enter the dot/period in the IP Address, press the Star ‘*’ key.
Subnet Mask
• If you select Static as Connection Type, enter the Subnet Mask to be applied on the phone by pressing the
digit keys.
Gateway Address
• If you select Static as Connection Type, enter the Gateway Address here. This is the IP Address of the
LAN Port of the Router.
DNS Server
• If you select Static as Connection Type, select the DNS Server option Static and configure the DNS
Address.
• If you select DHCP or PPPoE as Connection Type and your Internet Service Provider provides DNS
Address, select the DNS Server option Automatic. However, if your Internet Service Provider does not
provide DNS Address, select Static and configure the DNS Address.
DNS Address
• If you select DNS Server as Static, enter the DNS Address here.
To enter dot/period in the IP Address, press the Star ‘*’ key.
DNS Domain Name
• If you select DNS Server as Static, enter the DNS Domain Name here. DNS Domain Name is optional.
PPPoE User ID
• If you have selected PPPoE as Connection Type, you must enter the User ID provided to you by your
PPPoE Password
• This is the password provided by your Internet Service Provider for the PPPoE User ID. If you have
selected PPPoE as Connection Type, you must enter the password provided by your Internet Service
provider here.
PPPoE Service Name
• If your Internet Service Provider has provided a Service Name, enter the Service Name here. If your
Internet Service Provider has not provided a Service Name, do not configure this parameter.

Server Address
• The system works as the Auto Configuration Server for the phone. Enter the LAN or WAN IP Address/
Domain Name of SARVAM UCS here. Default: blank. The phone sends the request for configuration files
to this Server Address.
If you have selected DHCP as Connection Type, the phone will get the Server Address and Port
automatically from the DHCP Server. For this, use DHCP option 224 and Data Type as ‘String’ to provide
Server Address and Port from the DHCP Server.
For PPPoE and Static Connection Types, you need to enter the Server Address.
Server Port
• Enter the SPARSH Port of SARVAM UCS here. The phone sends the request for configuration files to this
port.
Valid range of the port is: 80 or 1025–65535. Default: 80.
VLAN Setting
If your phone is connected to a virtual LAN, you need to configure VLAN Settings.
To enable the VLAN switch to correctly route packets generated by the phone and the computers (on the LAN) to
each other, the packets must be tagged with a VLAN header.
The VLAN header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of traffic35.
The meaning of CoS bits with respect to traffic type is as follows:
CoS Traffic Type
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
• Select Phone VLAN/COS to add VLAN header to the packets generated by the phone, and add VLAN
header to the packets relayed from the PC to its LAN port (packets generated by the PC connected to its
PC port).
• To configure Phone VLAN/COS, select Enable?. The VLAN ID will be tagged on all packets generated
by the phone (SIP, RTP, DNS, ARP, etc.). Default: Disabled.
35. The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), that is, better han-
dling of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred
to as Class of Service (CoS) which is defined by IEEE 802.1P.

• Select VLAN ID and enter the VLAN ID that you have assigned to the VLAN in which the IP Phones are
connected. Valid range: 0-4094. Default: 1.
• Select SIP CoS and define the CoS (priority) bits in all SIP packets. Valid range: 0-7. Default: 3
• Select RTP CoS and define the CoS (priority) bits in all RTP packets. Valid range: 0-7. Default: 6.
• Select PC/VLAN CoS to add VLAN header to all packets entering the PC Port and leaving the LAN port of
the phone. Default: Disabled.
• To configure PC VLAN/COS, select Enable?.
• Select VLAN ID and enter the same ID as you have assigned to the VLAN in which the computers are
• Select CoS and define the Layer 2 CoS (priority) bits. Valid range: 0-7. Default: 0.
PCAP
To capture packets sent and received from and by the phone for monitoring and troubleshooting, you can enable
PCAP on the phone. The phone captures up to 1 MB of packets. For more information and for instructions on how
to use PCAP Trace on the phone, see “Using PCAP Trace for Matrix SPARSH VP248 Extended IP Phone”, under
PCAP Trace.
When you change the Network Settings, the phone will restart.
MAC Cloning
If you require cloning of the MAC Address, configure the following:
• Select Enable?.
• In Enter Clone MAC Address, enter the address you wish to clone.
802.1x Authentication
If you want to restrict unauthorized clients from connecting to your LAN, you need to enable 802.1x Authentication.
Using EAP MD5 protocol the PC connected to the LAN port of the IP Phone is first authenticated and then it gets
connected to LAN.
You need to configure the following 802.1x Authentication parameters:
• Select Enable?.
• Enter the 802.1x Authentication Identity provided by you network administrator.
• Enter the 802.1x Authentication MD5 Password associated with identity provided by your network
administrator.
Viewing Network Status
• In the Local Menu of the phone, place the cursor on Network Status and press the Enter key.

• The Network Status submenu appears.
N E T W O R K S TAT U S
MAC: 0 0:1 b:0 9:0 0:9 a:a 7
IP : 1 92. 16 8. 2 0 1 .2 0 5
MASK: 2 5 5 . 25 5 . 2 5 5 .0
G W: 1 9 2 . 16 8 . 2 0 1 .3
DNS:
Use the Down/Up key to view the status of the various network parameters. The status of the following
parameters appear on your display as you scroll.
• MAC: This is the MAC Address of the phone.
• IP: The current IP Address assigned to the phone.
• MASK: The current Subnet mask assigned to the phone.
• GW: The current Gateway IP Address assigned to the phone.
• DNS: The Domain Name Server address assigned to the phone.
• S. ADD: The LAN or WAN IP Address / Domain Name of the SARVAM UCS.
• S. PORT: The SPARSH Port SARVAM UCS.
• DOMAIN: The Domain Name assigned to the phone.
• 802.1x Authentication: The 802.1x authentication status is displayed—Success, Failure,

Authenticating or Disabled.
• FIRM: The version of the current Firmware of the phone.
• UBOOT: The UBOOT release date.
• KERNEL: The KERNEL release date.

SARVAM UCS:

• Assign an extension number (SIP ID) to the Extended IP Phone. For instructions on assigning SIP ID, see
“Configuring SIP Extensions”.
For the SIP extension number you assigned to the Extended IP Phone, you must configure the necessary
Extensions.
Now, follow the steps described below to install the Extended IP Phone.
2. You can mount the phone on a wall or on the desk.
3. When you mount SPARSH VP310 on a wall,
• Fix two screws in the holes on the wall, ensuring that they are aligned with the Keyhole Slots 1 and 2 of
SPARSH VP310. The screws should protrude from the wall to fit into the Keyhole Slots.
• Now, mount the phone with the screws fitting into the Keyhole Slot.

If you attach the Foot Stand at 50°, the phone will be placed in an almost upright position on your desk.
• Plug the long straightened end of the phone cord into the handset jack on the left side panel of the
phone marked with the handset symbol .


illustrated in the figure above.
OR
You may also plug in a headset with RJ9 connector into the headset port on the left side panel of the
scenario.
an 802.3af connection on the LAN Port of the phone. In this case you need not connect the Power Adapter.
If both the power options, that is, PoE as well as Power Adapter are available to the phone, then the phone
will derive power from the PoE enabled LAN Switch.

• The LCD display will light up and the booting message appears.
• Then the ‘Loading...’ message appears on the phone display.
Settings” at the end of this topic.
Server.
Network Settings
You can change the network settings of the Extended IP Phone. Press the Down key when the phone is in idle
state. To move the cursor and scroll through the menu and submenu options, use the following navigation keys on
your phone.
• The Cancel key to exit a menu.


You must press the Enter key to select Yes and access network settings.
You must press the Down key to access network settings.
3. When the phone is in idle state. You must press the Down key to access the Network Settings.
• When the phone is in idle state. You must press the Down key to access the Network Settings.
• Press Enter key to select Network Parameters.
the settings.
Connection Type
network.
Address, DNS Address, Server Address, automatically by the DHCP/PPPoE server.

IP Address
Subnet Mask
digit keys.
Gateway Address
DNS Server
Address.
DNS Address
DNS Domain Name
PPPoE User ID
PPPoE Password
provider here.
PPPoE Service Name

Server Address
Server Port
port.
VLAN Setting
CoS Traffic Type
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
PC port).

PCAP
to use PCAP Trace on the phone, see “Using PCAP Trace for Matrix SPARSH VP310 Matrix Extended IP Phone”,
under PCAP Trace.
MAC Cloning
• Select Enable? and press the Enter key. Select Yes to enable.
connected to LAN.
administrator.
• Again press Down key to select Network Status and press the Enter key .

• S. PORT: The SPARSH Port of SARVAM UCS.


You are recommended to complete the following steps before connecting the Matrix SPARSH VP330 to SARVAM
UCS:
• Decide the location where you want to place SPARSH VP330 within your LAN.
• By Default, in SPARSH VP330, the Connection Type selected is DHCP.
• If you want to use the DHCP Server for assigning IP Address to the Extended IP Phone, select DHCP
SARVAM UCS and SPARSH Port in the format “IP_Address:Port” in your LAN DHCP Server as per your
• You must configure the necessary parameters in SARVAM UCS so that SPARSH VP330 can register as a
SIP Extension. For instructions, see “Configuring Matrix SPARSH VP330”.
Now, follow the steps described below to install SPARSH VP330.

holes you drilled.
2.
into the Keyholes.
3. When you mount the phone on a desk, you can attach the Foot Stand in two ways at 35° Angle or at 50°
Angle.

phone marked with the handset symbol.
connector into the headset jack headset jack with the symbol on the left side panel of the phone.
OR
You may plug a headset with an RJ9 connector into the headset port on the side panel of the phone,
marked with the symbol .
6. Connect the LAN Port of SPARSH VP330 to the IP Network — A Router or LAN Switch — using the
Ethernet Cable.
OR

Connect the Wi-Fi USB Adapter into the USB port of the phone.
You can purchase the Wi-Fi USB Adapter from Matrix as an optional peripheral device to support wireless
network.
connect the PC Port at the bottom of the phone to the LAN Port of the computer.
• While loading the application then the loading message appears on the phone display,
Server.
If you want to change the Network Settings/Server Settings or want to use Wi-Fi for connectivity, press
Settings .
Refer to the SPARSH VP330 User Guide, for detailed instructions:
• To change the Network Settings of the phone and configure the network parameters.
• To use Wi-Fi for connectivity and configure its parameters.
• On getting the IP Address and Server Address, the phone initiates Auto Configuration to download the
• On successful registration, the Home screen appears.

The phone will register successfully, only if the SIP Extension parameters in SARVAM UCS have been
correctly configured as per your installation scenario.
Refer to the SPARSH VP330 User Guide to know more.

You are recommended to complete the following steps before connecting the Matrix Extended IP Phone to the
system when used with SARVAM UCS application:
option 224 and Data Type as ‘String’ and program the LAN or WAN Port IP Address /Domain Name and
SPARSH Port in the format “IP_Address:Port” in your DHCP Server as per your installation scenario.
Extensions.
Now, follow the steps described below to install the Extended IP Phone:
• Use the mounting template to drill holed of appropriate size and distance.
Keyhole Keyhole
Slot 1 Slot 2

• Now, mount the phone with the screws into the Keyhole Slots.
If you are unable to remove the wall mount tab, you may use a tool like a minus screw-driver to remove it.
• You can attach the Foot Stand in the following ways — at an angle of 45 degrees or 55 degrees
Stand attached at
45 degree angle
Stand attached at
55 degree angle

Refer to the diagram below for connectivity.
AUX
Handset
4. Connect the Handset.
• Plug the long straightened end of the Spring Cord into the handset jack at the bottom of the phone,
marked with the handset symbol .
• Plug the other (short straight) end of the Spring Cord into the jack at the bottom of the handset.
5. Connect the Headset (not supplied by Matrix).
connector into the headset audio jack at the bottom of the phone, marked with the symbol .
OR
You may also plug in a headset with an RJ9 connector into the headset port at the bottom of the
6. Connect to the IP Network.
• Plug one end of the Ethernet Cable into the LAN Port at the bottom of the phone, marked with the
symbol and the other end to the IP Network — A Router or LAN Switch.
7. Connect a PC to the Phone.

• Plug one end of the Ethernet Cable into the PC Port at the bottom of the phone, marked with the
symbol and the other end into the LAN Port of your PC/LAN Switch.
8. Connect DSS532 with the Phone.
• To connect DSS532 with the phone, plug one end of the RJ11 cable into the AUX Port of the phone
and the other end into the IN Port of the DSS532. For installation, see “Installing DSS532 with SPARSH
VP510”.
9. Connect the Power Supply.
• Plug the connector of the Power Adapter in to the power jack at the back of the phone, marked with the
symbol . Use only the adapter provided with the phone to prevent any damages that may
arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power
through an 802.3af connection on the LAN Port of the phone. In this case you need not connect the
Power Adapter.
• The phone will start loading the Extended IP Phone Firmware. It will display current firmware being
loaded.
If you want to change the Network Settings or Server Settings, press Yes key. Detailed instructions for
changing the Network Settings of the phone are provided at the end of this topic. See “Network Settings” at
the end of this topic.
Server.

Network Settings
state.
To navigate the menu,

• Press the Menu Key when the phone is idle.
• Scroll by pressing the Up/Down Navigation Key to reach the desired Menu option.
• Press the Select / OK Key to select the desired Menu option.
• Scroll by pressing the Up/Down Navigation Key to reach the desired sub-menu option.
• Press the Select / OK Key to select the desired sub-menu option.
To exit menu,
• Press Cancel Key.
or
Go ON-Hook.

You must press Yes key and access network settings.

You must press the Menu key to access network settings.
3. When the phone is in idle state, press the Down key .
• Use the Down/Up key to reach the desired network parameter and press Enter key to select. Change the
settings as per your requirements.
• Press Save key, to save the changes you make.
to clear. Now press the Delete key. The digit to the left of the cursor will be deleted.

Before you start configuring the Network Parameters, get acquainted with following context keys:
Context Keys Description
Enter/OK To select a particular parameter
Save To save the changes
Back To move a step backwards without saving the changes
Delete To delete previous characters from the cursor position
2Ab - Alphanumeric Mode

2Ab/123
123 - Numeric Mode
Connection Type
network.
IP Address
Subnet Mask
digit keys.
Gateway Address
DNS Server
Address.

DNS Address
DNS Domain Name
PPPoE User ID
PPPoE Password
provider here.
PPPoE Service Name
Server Address
• The system works as the Auto Configuration Server for the phone. Enter the IP Address or the Dynamic
DNS Domain Name of the LAN/WAN Port of SARVAM UCS here. Default: blank. The phone sends the
request for configuration files to this Server Address.
Server Port
• Enter the SPARSH Port of the SARVAM UCS here. The phone sends the request for configuration files to
this port.
VLAN Setting

CoS Traffic Type
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
PC port).
PCAP
PCAP on the phone. The phone captures up to 1 MB of packets. For more information and instructions on how to
use PCAP Trace on the phone, refer to the EON510_SPARSH VP510 User Guide.
MAC Cloning

connected to LAN.
administrator.
• Again press Down key to select Network Status and press the Enter key.

Refer to the EON510_SPARSH VP510 User Guide to know more.

Installing DSS532 with SPARSH VP510
Once you have installed SPARSH VP510 with SARVAM UCS, you can install the DSS532 by following the steps
given below:
2. Four clamps are provided with the phone — 2 DSS-Phone Clamps and 2 DSS-DSS Clamp.
Connecting the First DSS532
Connecting the Extender

3. Turn the phone upside down on the table and place the inverted DSS532 adjacent to it.
4. To attach the DSS532 with the phone, place the DSS Extender as illustrated below.

5. Insert the hooks on the Extender into the slots provided on the phone and the DSS532.
6. Firmly slide the DSS Extender upwards to lock them in place.
Attaching the Clamps

7. Now attach the clamps. To do so,
• Remove the screws to attach the clamps.
• Place the DSS-Phone Clamps between the DSS532 and the phone.
• Insert the screws back to fix the clamps.

Attaching the Footstand
8. You can mount the DSS532 with the phone on the desk at two angles — 45 degrees or 55 degrees by
attaching the Foot Stand.
Make sure both, the DSS532 and phone are mounted at the same angle.
Connecting the Cables

9. To connect the DSS532 with phone, plug one end of RJ11 Cable into Auxiliary(AUX) Port of the phone
and the other end into the IN Port of the DSS532.
Connecting Multiple DSS532
Remove the Foot Stand

10. Remove the Foot Stand of attached DSS532. To do so,
• Firmly slide the Foot Stand of the attached DSS532 downward to unlock.
• Now, slide down the attached DSS Extender in downward direction.

Attach the second DSS Extender
11. To attach the second DSS Extender,
• Place another inverted DSS532 adjacent to the existing assembly.
• Place the DSS Extender as illustrated in the diagram below.
• Insert the hooks on the Extender into the slots provided on both the DSS532.
12. Firmly slide both the DSS Extenders upward consecutively (attach the second extender first followed by
the existing one attached to the phone) and lock them in place.

Attach the Clamps
13. Attach the DSS-DSS Clamps between both the DSS532.
Attach the Foot Stand

14. Attach the Foot Stand of both the DSS532.
Make sure both, the DSS532 and the phone are mounted at the same angle.
Connect the second DSS532 to the existing assembly

15. Plug one end of the RJ11 Cable into the OUT Port of the existing DSS532 (already connected with the
phone) and the other end into the IN Port of the second DSS532.

You can install a maximum of four DSS532 with a phone.
16. After you have connected the DSS532 with the phone, you can configure the DSS Keys. For instructions,
see “Programming DSS Console Keys”.

Starting Up ETERNITY LENX
Power ON
• If you have completed all the installation tasks, switch on power supply. Keep the MCB Switch ON of your
PS48V card installed in the system and power the FCBC.
• Observe the Reset Cycle.
Reset Cycle
• Reset Cycle (Power-ON Self Test) takes about 2 minutes to finish.
• All the LEDs of the system, the cards and the keys of the DKP/SIP devices attached to the System are
turned on.
Interpreting LEDs
The functioning of the LEDs of the system and the various cards and their meaning are summarized at the
end of the installation instructions for each Card Type.
Refer to the LED Patterns described for each Card Type to verify if the system is operating properly and locate
faults, where they occur.
When the reset cycle is successful, the default Extension “Access Codes” loaded by the system and the date
and time of the “Real Time Clock (RTC)” of the system will appear on the LCD display of the DKPs/ IP Phones
you have connected with the system.

CHAPTER 6 Installing ETERNITY MENX
• The Matrix ETERNITY MENX is to be installed by persons trained and experienced in telecom wiring.
• The person installing the ETERNITY MENX must be familiar with trunks, physical wiring of the MDF on
now begin the installation of your ETERNITY MENX.

ETERNITY MENX can be wall mounted, rack mounted or placed on a table.
ETERNITY MENX has a total of 16 Universal slots.
The Matrix ETERNITY MENX is shipped factory fitted with the Power Supply Card and the CPU Card in their
respective fixed slots (refer the section “Know Your SARVAM UCS”). VoIP and VMS are in-skin to the CPU Card.
Hence, separate VMS and VoIP Cards are not required.
ETERNITY LENX has a total of 16 Universal slots. The cards - BRI, T1E1PRI, GSM/CDMA, DKP, CO, SLT, E&M,
Magneto - are shipped separately as per the order placed by individual customers. These cards can be installed in
any of the Universal slots.
universal slots. No universal slots will be functional by default. You must purchase the SARVAM EXP4 ENT license

Firmware Version V1R3 and later
ETERNITY MENX can be wall mounted, rack mounted or placed on a table.
The Matrix ETERNITY MENX is shipped factory fitted with the Power Supply Card and the CPU Card in their
respective fixed slots (refer the section “Know Your SARVAM UCS”). VoIP and VMS are in-skin to the CPU Card.
Hence, separate VMS and VoIP Cards are not required.
ETERNITY LENX has a total of 16 Universal slots. The cards - BRI, T1E1PRI, GSM/CDMA, DKP, CO, SLT, E&M,
Magneto - are shipped separately as per the order placed by individual customers. These cards can be installed in
any of the Universal slots.
If you have upgraded the system firmware to V1R3 and later in the old ETERNITY MENX system, the Expansion
purchase the SARVAM EXP4 ENT license to activate the universal slots as required.
If you have purchased the new ETERNITY MENX system with the firmware V1R3 and later, the Expansion Slots
license will be applicable for the universal slots. The first eight universal slots after the power supply card will be
functional by default. If you require more functional universal slots, you must purchase the SARVAM EXP4 ENT
license.
Each SARVAM EXP4 ENT license will provide the activation for next four universal slots in sequence.
Illustrated below is the position of the fixed and universal slots in ETERNITY MENX.
ETERNITY MENX
Power Supply Card
Power Supply Card
CPU Card
CPU Card
Slot#16
The two extreme left slots are reserved for the Power Supply Cards and the two extreme right slots are reserved for
the CPU Cards.
Mounting the System

2. Decide where to mount the ETERNITY MENX - on a table or wall - taking into consideration the
mechanical dimensions and the weight. If mounting the system on a wall, you may refer the mechanical
dimensions and the Mounting Template for drilling holes at appropriate places on the wall. Make sure the
system orientation is horizontal.

Inserting Cards
7. Make sure that the ETERNITY MENX power is off and the power cord is unplugged.

GSM/CDMA, E&M, E1FO, Magneto, Radio - later in this section. Refer to them when installing each card
type.

applicable.


Two types of Power Supply Cards are supported by the Matrix ETERNITY MENX: PS UNI Card and PS48V Card.
• PS UNI Card with 100-240VAC, 47-63Hz Mains as Input AC Voltage Power Supply.
This card is designed on the SMPS scheme. As this card does not have any provision for battery backup,
it is recommended that a UPS be connected to keep the system powered during outages.
This card has four LEDs, a Mains Switch, and a Socket assembly for connecting the mains cord.
• PS48V Card with 48VDC as Input DC Power Supply Voltage. A Float cum Boost Charger (FCBC) is
required to feed 48VDC power to the card. The FCBC works on input AC mains.
This card is available in one variant - DC to DC 500 W.
The card has four LEDs, an MCB Switch, a power ON/OFF Switch, and a 3-way termination block for
connecting the power cord.
Both, the PS UNI card and the PS48V Card provide DC output voltages as: +3.5V, +5.0V, -27V and -85V.
These are indicated by LEDs.
• Redundancy option is provided for the Power Supply Card and the CPU Card.
• The ETERNITY MENX supports two power supply cards. Whenever there is a fault in one, the other
takes over the control, providing uninterrupted communication.
• It also support 'Hot Swap' for the Power Supply Card.
• The maximum number of ports supported by the GSM, SLT and DKP Cards may vary according to the
type of Power Supply used. Refer the following table for maximum ports supported with Universal
Power (PSUNI) and DC Power Supply.
ETERNITY MENX - Universal Power Supply (PSUNI)
Type of Port Maximum Number of Ports Supported
GSM 32 ports in Talk mode
SLT 128 ports in Talk mode for short loop

10 extensions in Talk mode for long loop
472 extensions permanently connected
DKP 96 ports maximum
Analog SLT ports supported for Short Loop with Loop Current programmed
Loop Current Programmed 25mA 30mA 35mA 40mA
Number of ports supported in talk mode (OFF-Hook short loop) 200 172 150 128
according to loop current programmed

ETERNITY MENX DC to DC 500W Power Supply
Type of Port Maximum Number of Ports Supported
GSM 112 ports in Talk mode
SLT 256 ports in Talk mode for short loop
20 extensions Talk mode for long loop
512 extensions permanently connected
DKP 128 ports maximum

out of the slot.
2. Insert the Power Supply card into the guide rails of the first slot on the extreme left, designated for the
Power Supply Card. Make sure that the card is inserted deep enough to make perfect contact with the
connectors on the motherboard at the backplane.
To install a second card for redundancy insert the second card on the next slot.
5. If installing the PSUNI card, connect the three-pin power cord into the socket of the PS UNI card and plug
in the cord into the mains supply.
You may connect the PSUNI Card to a UPS to keep the system live during power outages.
Select a UPS considering the typical power consumption of ETERNITY MENX presented in the table
below:
Model Power Consumption (Typical)
ETERNITY MENX16S 100 watts
6. If installing the PS48V card, connect the Float cum Boost Charger (FCBC). Terminate the power cord from
the FCBC output into the 3-way termination block on the PS48V card.
Color Signal
Red +48VDC

Color Signal
Black GND
Green Earth
Connecting FCBC to the ETERNITY MENX
-85VDC (LED L1)
230VAC +/- FCBC ETERNITY ME -27VDC (LED L2)

41 to 56V
10%@50Hz 10A Card PS48VDC +5VDC (LED L3)
+3.5VDC (LED L4)
48V Battery
If two PS48V cards are installed for redundancy, each must be connected to a separate FCBC and each
FCBC must be connected to a separate source of power supply.
7. Connect Battery back up to the FCBC41.

Battery backup time depends upon the total load. The total load is the sum of system's load and load of
active extensions. The power consumed by the variants of ETERNITY MENX is given in the table below:
Model Power Consumption (Typical)
ETERNITY
100 watts
MENX16S
41. When the batteries are drained, the FCBC goes into the boost mode and begins to charge the batteries at higher current. When
the batteries reach a preset voltage level (typically set to 56.0 volts), the FCBC goes to float mode. In the float mode the FCBC
keeps charging the battery but at lower current. The FCBC monitors the voltage level of the batteries. As soon as the battery volt-
age goes below preset voltage (typically set to 50.4 volts), FCBC goes from float mode to boost mode. The change over from
mains to battery and vice-versa is automatic. The advantage of using an FCBC is that batteries get charged faster, since the bat-
teries are charged with higher current initially.

The Battery back up time depends on the 'Ah' rating of the battery connected to the FCBC. The FCBC
uses the constant voltage charging method. So, the batteries get charged faster if less power is consumed
by the system when in mains mode.
8. Switch on power supply, after completing all other installation.

The CPU Card
The ETERNITY MENX-LENX CPU Card hosts the SARVAM UCS Application. It supports four VOCODER modules
and one VMS module. Both the modules — NX DBM VOCODER64 and NX DBM VMS64 are optional and can be
and modules which controls all other slave cards (SLT, DKP, CO+SLT, DKP+SLT, E&M, BRI, T1E1, E1FO, GSM
etc.). All the configuration and programming information is stored on this card.
ETERNITY MENX supports redundancy along with Hot Swap for the CPU Card. An additional CPU Card can be
installed for redundancy. If the main CPU Card fails, the other card takes over. However, during this process all the
ongoing calls will be disconnected.
The CPU Card occupies fixed slots, the first two slots from the extreme right of the first Rack, with a unique
arrangement of connectors. So no other card can be inserted in the slots of the CPU Card.
The CPU Card has a WAN Port, LAN Port and USB Port on the front panel. It also has an Internal USB Port with a
factory fitted pendrive.

LAN RJ45
LAN Switch.
WAN RJ45
Router/Modem.

Converter.
USB to COM The USB to COM Port can be used to:

Activity logs.
LAN Interface
WAN Interface
VoIP Interface
The CPU Card supports four NX DBM VOCODER64 modules. You must purchase the module separately for VoIP
functionality.
VOCODER Channels
The system supports four NX DBM VOCODER64 Modules. Each module supports 64 VOCODER Channels42. You
CHNL16.

VMS Interface
Call Taping.
SARVAM UCS ENT Application. The Pen Drive is also the storage device for mailbox messages.
Internal USB Port

Matrix Extended IP phone firmwares and SMS Server firmware.

When you select the SARVAM UCS ENT Application, the system fetches the application from the pendrive.

COM Converter.
If you use any other USB to COM Converter, it may not function properly.
LED
The CPU Card has three dual color LEDs:
• Heart beat (L1)
• Layer Communication indications (L2)
• Slave Switch indications (L3)
LED for both Active and Standby Card:
State Color Cadence
At Power ON
L1 and L3 – RED Continuous ON for 15 sec

(approximately)
L1 and L3 OFF for 30 sec (approximately)
When L1 and L3 has been OFF Continuously ON for 3 sec

L2 – ORANGE
for 10 sec
L2 – OFF OFF
LED for Active Card:
State Color Cadence
L1 Pattern

State Color Cadence

In normal condition GREEN Toggle
L2 Pattern
500 ms GREEN – 500 ms RED – 500

ms ORANGE
In normal condition - OFF
L3 Pattern
When sync is received by

RED Continuous ON
Active card from Standby Card
When Data transfer done by

RED Continuous ON
Active card
LED for Standby Card:
State Color Cadence
L1 Pattern

In normal condition RED 1 sec ON – 1 sec OFF
L2 Pattern
500 ms GREEN – 500 ms RED – 500

ms ORANGE
PLL Lock (standby card clock

GREEN Continuous ON
is in sync with active card)
PLL is not lock (standby card

clock is not in sync with active - OFF
card)
L3 Pattern
When sync is sent by Standby

RED Continuous ON
card to Active card

The CPU Card has a Password IP Default Switch (SW1) on the board:
Switch Position Function
SW1 ON Password IP Default
OFF (default) Normal

To install,

• There are 5 PCI Connectors (4 for VoIP and 1 for VMS) and five pairs of studs on the CPU board.
• Locate the VOIP #1 label on the CPU board. The first NX DBM VOCODER64 Module should be mounted
here.
• Remove the screws on the studs for the module and keep them aside.
area.

• Press the Module with a finger and match the mounting holes perfectly with the stud holes. Make sure you
do not touch the PCB area of the module except the yellow line provided for grounding at the front end of
the module.
Do not apply excessive pressure.

• Secure the module with the screws on the studs.
You may install the second NX DBM VOCODER64 Module by locating the VOIP #2 label on the CPU board and
follow the same steps as above. Similarly, install the other modules if required.

• Remove the screws from the module and keep them aside.

• Replace the screws on the studs.

To install the NX DBM VMS64 Module, locate the VMS MOD label on the CPU board and follow the same steps as
described for “Installing the VOCODER Module”.

Make sure you use the plastic screws provided to affix the VMS Module over the studs.
Switch-off the system and then replace the pendrive. The system will not detect the new pendrive if you do
not restart the system after replacement.
Make sure while replacing the pendrive, you insert it in the same USB Port where the factory fitted
pendrive was inserted.
• If you have no other modules to install, insert the card back into the ETERNITY MENX.

wire trunk interfaces on a single card.
SLT Cards for ETERNITY MENX
Combination card, with 8-ports to connect to 8 Two-wire Analog trunk lines and
24 Single Line Telephones
ETERNITY ME Card
CO8+SLT24
Transfer”.
Connectors
for each connector.
Only the SLT48 card has a 50-pin Centronics connector for the ports.
LEDs
The SLT cards have a single, tri-color LED to indicate:

• the status of any one extension during normal functioning of the system.
You may monitor any of the SLT Extension ports by assigning the LED to that port43.

43. To do this, enter SE mode, and dial the SE Command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot
in which the card is installed and Port is the port on the card to which the LED is to be assigned to monitor its functioning. LED
Number is the number of the LED on the card, which will monitor the port.

backplane.

L1

L1

L1
Blue - (Blue & White) SLT 01

L1
Pattern of the SLT Card.

LED Pattern of the SLT Card
Auto Upgradationa
Initialization
Stand-by task ORANGE 1 sec Orange -1 sec Green
Errors
cess continues.



ILC Cards for ETERNITY MENX
Choose an ILC Card with the configuration that meets your requirement for intercom ports. Also, consider the
maximum Port capacity of the system you are installing. The maximum number of intercom ports supported are
288.
Connectors
The ILC32 Cards have RJ45 connectors with four ports on each connector. A multi-pair, MDF cable is supplied for
each connector.
LEDs
The ILC cards have a single, tri-color LED to indicate the health of the card during the Reset Cycle.
Installing Intercom Telephones

jacks.
4. Unpack the ILC card and check the package contents. You may switch off power supply before you install
the card. Since, ETERNITY MENX supports Hot Swap, you can install the card in power on condition.
backplane.

L1
10. If you have completed all other installation tasks, power ON the system, observe the Reset Cycle and the
LED indication of the card.

LED Indication of the ILC Card
Auto Upgradationa
Initialization
ORANGE,
Stand-by task 1 sec Orange -1 sec Green
GREEN
Errors
cess continues.

series, the proprietary PC-based phone EONSOFT, the Direct Station Selection (DSS) Consoles, and the
proprietary Digital Turret, EON7444, with the system.
DKP Cards for ETERNITY MENX
16-port card to connect 16 DKP/DSS Consoles and the proprietary Digital Turret,
ETERNITY ME DKP16
EON74a.
a. EON74, the proprietary Digital Turret of EON74 is currently supported only on ETERNITY ME DKP16
Card and ETERNITY ME DKP8 Card with firmware version-revision V04R01 and onwards.
Select a DKP Card with the configuration that meets your requirement for DKP Ports.To connect the proprietary
digital key phones with the system, you must have at least one of the above mentioned DKP Cards installed in the
system.
Connectors
LEDs
The DKP cards for ETERNITY ME have a single, tri-color LED to indicate:

• the status of any one of the ports during normal functioning of the system. By default, it is assigned to DKP
Port 1.
You may monitor any of the DKP ports by assigning the LED to that port45.

DSS Consoles.
44. Check for availability.


motherboard.
slot.

L1

L1

L1
6. Plug in the RJ45 end of the MDF cables provided with the DKP card into the respective connectors.
instructions for EON, EONSOFT are provided separately. Installation instructions for EON74 are provided
in the EON74 User Guide.
of the DKP Card.

Auto Upgradation
Initialization
GREEN ON 500ms-OFF 500ms
ORANGE ON 500ms-OFF 500ms
ORANGE, 1 sec Orange-1 sec

Stand-by task
GREEN Green
Errors
Status of Selected DKP Port
Selected DKP's data are transmitted to CPU Card RED Togglea on each event
Selected DKP's data are received from CPU Card RED Toggleb on each request from Master
a. The current LED state will remain the same until the next event is received from the application on the DKP Port.
For example, if the current LED state is Green/Red ON, on the next event, the LED will be turned OFF. It will
remain OFF until the next event occurs. When the next event is received it will be turned Green/Red ON again.
This process continues.
b. Same as the above note.
J2 & J4 AB NA

Installing EON48

Angle.

Headset

M AT R I X E O N 4 8 - S V 2 R 2
PLEASE WAI T .. .
202 Reception
M on 2 4 A U G 1 2 : 0 0
Installing EON310




Operation”.
Installing EON510
Keyhole Keyhole
Slot 1 Slot 2

Stand attached at
45 degree angle

Stand attached at
55 degree angle
AUX
Handset

Operation”.
Installing DSS64

Console Keys”.

The instructions for installing DSS532 with EON510 or SPARSH VP510 are same. For detailed instructions, refer to
“Installing DSS532 with SPARSH VP510”.
Installing EONSOFT
• Windows 98
• Windows XP
• Windows NT
• Windows 2003
• Windows Vista
• Windows 2007

application.
desktop.



data transfer.
EONSOFT.



The CO Card
The CO Card is available in the following configurations. CO interface is also available in combination with SLT
ports on a single card.
CO Cards for ETERNITY MENX
ETERNITY ME Card CO8+SLT24 Combination card, with 8 CO ports to connect 8 CO analog trunk lines
and 24 SLT ports to connect 24 Single Line Telephones
Transfer”.
Choose a CO Card with the configuration that meets your requirement for CO trunk ports, keeping in mind the
maximum CO Trunk Port capacity of the system you are installing.
Connectors
LED
The CO Cards have a single tri-color LED to indicate:

• the status of a selected Trunk port during normal functioning of the system.

LED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to
the port.

Distribution Frame.

L1

L1
RJ45-3 Unused
3
RJ45-4 Unused
4

L1
7. Plug in the RJ45 end of the Trunk Card cables into the respective connectors.
the MDF”.
10. If you have completed all other installation tasks, power ON the system.

LED Pattern of the CO Card
Auto Upgradationa
Initialization
1 sec GREEN - 1Sec

Stand-by task ORANGE
Orange
Errors
a. Done by the boot loader application.
Status of Selected CO Port
Selected CO's data are transmitted RED Togglea on each event

to CPU Card
Selected CO's data are received from RED Toggleb on each request from Master
CPU Card
a. The current LED state will remain the same until the next event is received from the application on
the CO Port. For example, if the current LED state is Green/Red ON, on the next event, the LED
will be turned OFF. It will remain OFF until the next event occurs. When the next event is received
it will be turned Green/Red ON again. This process continues.
J2 & J4 AB NA

The BRI Card
BRI Cards of ETERNITY MENX
Connectors
The BRI cards have RJ45 connectors. The ETERNITY ME BRI8 card has 8 RJ45 connectors for 8 BRI ports.
The ETERNITY ME BRI4 card has 4 RJ45 connectors for 4 BRI ports. A separate cable is supplied for each
connector.
LEDs
The ETERNITY ME BRI8 and ETERNITY ME BRI4 has 4 LEDs.

Where,
• S/T Interface = between the ISDN user equipment, in this case, ETERNITY MENX and the Network
MENX.

TE and NT Modes
In this illustration, the BRI line from ISDN Service Provider is directly connected to BRI port of the ETERNITY
MENX via the NT1 device. Here, the ETERNITY MENX is the Terminal Equipment, so the BRI Port must be
programmed to work in the TE mode.
When an ISDN Phone is to be connected to the BRI port of ETERNITY MENX, the BRI port must be programmed
to work in NT mode.
When a BRI port of another ISDN System is to be connected to the BRI port of the ETERNITY MENX, in such a
• the BRI port of the other ISDN System in the TE mode and the BRI Port of the ETERNITY MENX in the NT
mode.
OR
• the BRI port of the other ISDN System in the NT mode and the BRI Port of the ETERNITY MENX in the TE
mode

this case ETERNITY MENX, can be up to 1 kilometer.

Where,
NT.
• Using this configuration, any subscriber from ETERNITY MENX can access a BRI line and can make
outgoing calls. At the same time, another subscriber from ETERNITY MENX or any ISDN phone shown in
same BRI.
used.

Where,
• Using this configuration, any subscriber from ETERNITY MENX can access the BRI line and make
outgoing calls. At the same time, another subscriber from the ETERNITY MENX or any ISDN phone
shown in the figure can make outgoing calls from the same BRI. In the same way, incoming calls are
possible on the same BRI.


the NT mode.

shown below.

• Termination need not be inserted if the BRI port of ETERNITY MENX (configured in TE mode) is

6. To set the 100 termination on the BRI port set the Jumpers on the BRI Module (daughter-board) to
the position described below:

Function
J6 J8 J7 J9 J6 J8 J7 J9
To insert 100
termination
To remove 100
termination

Function
J6 J8 J7 J9 J6 J8 J7 J9
To insert 100
termination
To remove 100
termination
By default, Termination Resistance of 100 is set on the BRI port (the Jumpers are in AB position).

1 RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
Tx 3
100
Rx 4
Rx 5
100
Tx 6
Bar configuration.
bar.
1 1 RJ45 Connector
ports on BRI Bus
Bar to which the
3 3
ISDN TE
4 4 Equipment is
connected
5 5
6 6
8 8
on the Bus bar.
the ISDN TE side.
the NT side.

8. When the BRI Port of the ETERNITY MENX is used as BRI-NT, you can feed power to the terminal
equipment connected to the BRI-NT Port from the ETERNITY MENX.
To do this,


Function
J4 J5 J2 J3 J4 J5 J2 J3


pair of wires

Function
J4 J5 J2 J3 J4 J5 J2 J3


pair of wires
guidance.

Configuration details of the U interface (RJ-45) on NT1
4 Tx
5 Rx
3 Rx1
4 Tx1
5 Tx2
6 Rx2
3 Green-White TxA
4 Blue RxA
5 Blue-White RxB
6 Green TxB
7 Brown-White V-
8 Brown V+
3 Green-White RxA
4 Blue TxA
5 Blue-White TxB
6 Green RxB
7 Brown-White V-
8 Brown V+

• The BRI8 Card has 8 LEDs: L150, L2, L3, L4, L551, L6, L7 and L8. These display the status of each port.
• The BRI4 Card has 4 LEDs: L152, L2, L3 and L4. These display the status of each port.

The T1E1PRI Card
The ETERNITY ME T1E1PRI Card provides the interface to connect the system to ISDN Network.
• PRI
• E&M
• PRI
• E&M
The T1E1PRI Card is available in the following configurations:
T1E1PRI Card for ETERNITY MENX
ETERNITY ME Card T1E1PRI 2-Port card with QSIG support to connect 2 ISDN T1/E1 PRI Lines or ISDN
Dual Compatible Devices
ETERNITY ME Card T1E1PRI 1-Port card with QSIG support to connect 1 ISDN T1/E1 PRI Line or ISDN
Connectors
The T1E1PRI card has an RJ45 Connector for each port. The ETERNITY ME T1E1PRI Dual card has 2 RJ45
Connectors for the two ports, while the ETERNITY ME T1E1PRI Single card has a single RJ45 Connector.
A cable with RJ45 plugs on both ends is supplied for each connector.
LEDs
The ETERNITY ME T1E1PRI Single Card has two LEDs L1 and L2.

turned off.


shown below:
• If using the ETERNITY ME T1E1PRI Dual Card, use DIP Switch SW2 to change the Termination


ETERNITY ME T1E1PRI Card).
1 Line A
2 Line A
3 Not used
4 Line B
5 Line B
6 Not used
7 Not used
8 Not used
1 TX1 (Tip)
2 TX2 (Ring)
3 Not used
4 RX1 (Ring)

5 RX2 (Tip)
6 Not used
7 Not used
8 Not used
Pin details of ETERNITY MENX T1E1PR1 Port

below.

4 5
NC NC
3 6
Rx2 (Tip) NC
2 7
Rx1 (Ring) NC
1 8
LED Patterns

The ETERNITY ME T1E1PRI Single Card has two LEDs: L1 and L2.
ETERNITY MENX.

Port Active Mode
LED1/LED3 Pattern:
LED2/LED4 Pattern:
LED1/LED3 Pattern:
LED2/LED4 Pattern:

LED1/LED3 Pattern:
LED2/LED4 Pattern:

LED1/LED3 Pattern:
LED2/LED4 Pattern:

LED1/LED3 Pattern:
LED2/LED4 Pattern:

J5a AB Dual T1E1
BC Single T1E1
J6 AB JTAG Mode
BC (default) Embedded ICE
J7 AB Internal Boot
BC (default) External Boot
J8 & J12 AB (default) Debug
BC UART
J9, J10 & J11 BC (default) Normal
AB Boot
a. In T1E1 PRI Dual Card, default jumper position will be AB. In T1E1 PRI
Single Card, default jumper position will be BC.

The E1FO Card
The ETERNITY LE E1FO Card provides the interface to connect the ETERNITY to the ISDN PRI Network. For E1
The E1FO Card is available in the following configurations:
E1FO Card for ETERNITY MENX
ETERNITY ME Card E1FO Dual 2-Port card to connect 2 ISDN T1/E1 Lines.
ETERNITY ME Card E1FO Single 1-Port card to connect 1 ISDN T1/E1 Line.
Connectors
The E1FO Dual card has two RJ45 Connectors and two Fiber Optic connectors. The E1FO Single card has one
RJ45 Connector and one Fiber Optic connector.
LEDs
The ETERNITY ME E1FO Single Card has two LEDs L1 and L2.

turned off.

shown below:
• If using the ETERNITY ME E1FO Dual Card, use DIP Switch SW2 to change the Termination


ETERNITY ME CARD E1FOPRI Single).
1 Line A
2 Line A
3 Not used
4 Line B
5 Line B
6 Not used
7 Not used
8 Not used
1 TX1 (Tip)
2 TX2 (Ring)
3 Not used
4 RX1 (Ring)

5 RX2 (Tip)
6 Not used
7 Not used
8 Not used
The T1/E1 Interface of the ETERNITY ME CARD E1FOPRI Single terminates in an 8-pin RJ45, female connector

4 5
NC NC
3 6
Rx2 (Tip) NC
2 7
Rx1 (Ring) NC
1 8

LED Patterns

The ETERNITY ME E1FO Single Card has two LEDs: L1 and L2.
ETERNITY MENX.
1. Port Active Mode

LED1/LED3 Pattern:
LED2/LED4 Pattern:

LED1/LED3 Pattern:
LED2/LED4 Pattern:
LED1/LED3 Pattern:
LED2/LED4 Pattern:

LED1/LED3 Pattern:

LED2/LED4 Pattern:

LED1/LED3 Pattern:
LED2/LED4 Pattern:

The E&M Card
lines55.

The E&M Card is available in the following configurations:
E&M Card for ETERNITY MENX
Connectors
Protocol.

LEDs
The ETERNITY ME Card E&M8 has eight tri-color LEDs. The ETERNITY ME Card E&M4 has 4 LEDs, to indicate
the functioning of the ports.

OR
through it.
OR

BC Position.
Position.

01 Open Gray 01
02 SB Green-White
03 M OUT Green
05 SPCH A Blue
05 SPCH A Blue
07 RX SPCH B Orange
08 E IN Brown-White
02 SB Green-White
03 M OUT Green
05 SPCH A Blue
05 SPCH A Blue
07 RX SPCH B Orange
08 E IN Brown-White
09 BGND Brown
01 Open Gray 05
02 SB Green-White
03 M OUT Green
05 SPCH A Blue
05 SPCH A Blue
07 RX SPCH B Orange
08 E IN Brown-White
02 SB Green-White
03 M OUT Green
05 SPCH A Blue
05 SPCH A Blue
07 RX SPCH B Orange
08 E IN Brown-White
09 BGND Brown

L1 L3
L2 L4
Pin No. Connection Colour H/w Port Offset
01 Open Gray 01
02 SB Green-White
03 M OUT Green
07 RX SPCH B Orange
01 Open Gray 02
08 E IN Brown-White
02 SB Green-White
09 BGND Brown
03 M OUT Green
10 CCC Gray-White
05 SPCH A Blue
06 SPCH B Blue-White RJ45-2 2
07 RX SPCH B Orange Pin No. Connection Colour H/w Port Offset
08 E IN Brown-White
10 CCC Gray-White 02 SB Green-White
03 M OUT Green
07 RX SPCH B Orange
01 Open Gray 04
08 E IN Brown-White
02 SB Green-White
09 BGND Brown
03 M OUT Green
10 CCC Gray-White
05 SPCH A Blue RJ45-4
06 SPCH B Blue-White 4
07 RX SPCH B Orange
08 E IN Brown-White
09 BGND Brown
10 CCC Gray-White

M M
SB SB
E E
SG SG
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS

M M
SB NC NC SB
E E
SG SG
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS

M M
SB SB
E E
SG SG
SP(RxA) SP(RxA)
SP(RxB) SP(RxB)
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS

M M
SB NC NC SB
E E
SG SG
SP(RxA) SP(RxA)
SP(RxB) SP(RxB)
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS
M NC NC M
SB NC NC SB
E NC NC E
SG NC NC SG
SP(A) NC NC SP(A)
SP(B) NC NC SP(B)
CCS CCS
PLCC PLCC
to the PLCC panel.

LED Pattern of the E&M Card
Initialization
At Power ON RED L1 to L8 glow one after the other in a sequence,

ON 100ms -OFF.
L1 ON 100ms-OFF,
L2 100ms ON-OFF,
L3 100ms ON-OFF,
GREEN L1 to L8 glow one after the other in a sequence,

ON 100ms -OFF.
L1 ON 100ms-OFF,
L2 100ms ON-OFF,
L3 100ms ON-OFF,
Stand-Bya GREEN, ORANGE L1 toggles GREEN ON 1 sec,

ORANGE ON 1 sec
Normal (Port Event)
M-Wire High GREEN LED of the Port continuously ON
E-Wire High RED LED of the Port continuously ON
E-Wire and M-Wire High ORANGE LED of the Port continuously ON
Errors
Controller RAM failure ORANGE All LEDs Toggle at 1 sec
External RAM failure ORANGE All LEDs Toggle at 2 sec
Eprom failure ORANGE All LEDs Toggle at 3 sec
Invalid Slot detected ORANGE All LEDs Toggle at 6 sec
a. Waiting to be detected by Master Card.

The Mobile Card
The Mobile Card interfaces the system with GSM/3G/CDMA networks. It routes calls made and received over
mobile networks, like a mobile handset.
The Mobile Cards are available in CDMA, 2G and 3G variants.
The Mobile Card for ETERNITY MENX
8-port card to connect to 8 GSM networks (8 SIM Cards can be

ETERNITY ME Card GSM8
installed)
4-port card to connect to 4 GSM networks (4 SIM Cards can be

installed)
8-port card to connect to 8 GSM networks with 3G support (8 SIM Cards

ETERNITY ME Card GSM8 3G
can be installed)
4-port card to connect to 4 GSM networks with 3G support (4 SIM Cards

ETERNITY ME Card GSM4 3G
can be installed)
ETERNITY GE CARD CDMA4 4-port card to connect to 4 CDMA networks (4 RUIM Cards can be
installed)
If you are installing CDMA Mobile Card in your system, it is recommended to avoid using features that
support DTMF Detection.
The features where the caller is asked to dial digits and the system has to detect it, for example, DID,
Voice Mail Auto Attendant etc might not work efficiently.
Antenna
There is a single rooftop (RT) antenna for four GSM ports. A splitter connects all the four ports on the card into a
single antenna. An antenna cable is also provided, giving you the flexibility to move the antenna to another position
(in case of weak signal).

Personal Identification Number (PIN)57
LEDs
There is a tri-color LED for each mobile port on the card to indicate the functioning of the card and the status of the
ports.

To be able to connect the system to GSM/3G/CDMA networks, you must have one of the above mobile cards
installed in the system.
• If using a GSM/3G/CDMA card, get the SIM Card from the GSM/3G/CDMA service provider of your
choice ready. Use SIM PIN protection58, if required.
57. Not applicable if you are installing CDMA Mobile Card in your system.
58. SIM PIN Protection is not applicable if you are installing CDMA Mobile Card.

ETERNITY ME Card GSM4/GSM4 3G/CDMA459
Enabling PIN Protection on SIM60

59. Insert RUIM Card instead of SIM Card if you are installing CDMA Mobile Card.
60. Disable PIN Protection on RUIM if you are installing CDMA Mobile Card in your system.


The number of tri-color LEDs on the Mobile card corresponds with the number of mobile ports on the card.
At Power On: All LEDs will blink 1 second ON and 1 second OFF in the color sequence: Red-Green-Orange until
the Reset cycle is complete.
In the Stand-by state: All LEDs will glow Orange for a second and turn Green for a second, repeatedly.
During normal functioning: The LEDs will various events on the Mobile port in the color and cadence described
in the table below:
Port disabled - LED OFF
Port idle - LED OFF




OFF (3 blinks)


The Magneto Card
Magneto Card for ETERNITY MENX
ETERNITY ME Card Magneto8 8-port card to connect 8 Magneto Phones
Connectors
LED
The ETERNITY Magneto8 has 8 LEDs for each magneto port supported by the card.
The LEDs indicate the health of the cards during the Reset Cycle and the status of the ports during the normal
functioning of the system.

OR


Magneto Card on the MDF. Refer the cable diagram below.
system and observe the Reset Cycle and the LED Pattern of the Magneto Card.

LED Pattern of the Magneto Card

Card waiting for application RED L1 turned ON for 200ms, turned OFF
simultaneously.
Card is up, loaded with new GREEN L1 turned ON for 200ms, turned OFF
application simultaneously.
Initialization
RED L1 to L8 turned ON 500ms-OFF 500ms
GREEN L1 to L8 turned ON 500ms-OFF 500ms
ORANGE L1 to L8 turned ON 500ms-OFF 500ms
Stand-Bya ORANGE, L1 toggles 1 sec Orange and RED

GREEN,
Normal (Port Event)
Ring (incoming/outgoing call) RED LED of the Port continuously ON
Port Disabled LED of the Port continuously OFF
OFF-Hook (in Speech) GREEN LED of the Port continuously ON
For Outgoing call from Magneto:

When called extension answers the call
For Incoming call to Magneto:

When stopping Ring on Magneto using
MRE key or #
Port Idle LED of the Port continuously OFF
Errors
Invalid Card Configuration Jumper ORANGE All LEDs Flash (250ms ON-250ms OFF) twice,
OFF 3 sec.
Invalid Slot detectionb ORANGE LEDs Flash (250ms ON-250ms OFF) 6 times,
OFF 3 sec.
a. Waiting to be detected by CPU Card.

b. Waiting to be detected by CPU Card.
J1 & J2 AB NA
BC (default) Normal Operation
J3, J5 & J6 AB (default) Normal Operation

The Radio Card
Radio Cards for ETERNITY MENX
Connectors
LED
The Radio Card has one tri color LED.

OR

H/w Port
Offset

to
Orange 2 PTT_RTN
RJ45-8
Green & White 3 Rx-
Blue 4 Tx+
Blue & White 5 Tx-
Green 6 Rx+
Brown 8 Unused
H/w Port
Offset

to
Orange 2 PTT_RTN
RJ45-4
Green & White 3 Rx-
Blue 4 Tx+
Blue & White 5 Tx-
Green 6 Rx+
Brown 8 Unused
system. Observe the Reset Cycle and the LED Pattern of the Radio Card.
LED Pattern
Card waiting for application RED 200ms ON - 200ms OFF

Card is up, loaded with new

application
Initialization
RED 500ms ON - 500ms OFF
ORANGE 500ms ON - 500ms OFF
ORANGE,
Stand-By Toggles 1 sec Orange 1 sec Green
GREEN
Port Status
Selected Port's data are transmitted

RED Toggle on each event
to master card
Selected Port's data are received

GREEN Toggle on each event from master
from master card
J4 and J5 AB (default) Normal Operation
In PCB-P-200-80-01-01, for 4-wire speech the Jumpers J1 and J2 on the modules must always be set in
AB position.
In PCB-P-200-80-01-02 onwards only 4-wire speech is possible therefore there are no Jumpers on the
modules.

The Data Card

LEDs
The Data Card has one dual color LED.

Cycle and the LED pattern of the Data Card.
LED Pattern of the Data Card
Waiting for Master to start Up-gradation

Red Blinking 200ms ON - 200ms OFF
procedure (waits for 10 seconds)
Up-gradation in progress Green Blinking 200ms ON - 200ms OFF
Application run error after up-gradation

Red Continuous ON
procedure
Auto upgradation completed successfully

Green Continuous ON
and in process to start new application

Main application is being executed Red Blinking 1 sec ON - 1 sec OFF
Jumpers
BC Internal Boot

SIP Extensions

AMP100 User Guide.

Addresses only.
Extension.
• the SIP ID

SARVAM UCS
LAN Switch/ Hub
SIP Extn.
SIP Extn.
Matrix VARTA
WIN200
IP
Router SIP Extn.
SARVAM UCS
SIP Extn.
LAN Switch/ Hub
IP
Router SIP Extn.
SIP Extn.
Matrix VARTA
WIN200
• You may also register any SIP device (Matrix VARTA UC Clients, Extended IP Phone or Open SIP phone)
IP Phones.

SARVAM UCS:
Extensions.
holes you drilled.
2.
into the Keyholes.

desk.

Headset
OR

Foot Stand
Keyhole Keyhole
Slot 1 Slot 2
Headset
Handset
scenario.


Standar d S IP
Extended S I P

Yes
No
Server.
Network Settings
on your phone.




Yes
No
M on 10 M AY 1 5: 40
Local Menu
1 2 abc 3 def
4 ghi 5 jkl 6 mno
7 pqrs 8 tuv 9 wxyz

CA04
0 #
CA03 *
CA02
CA01

LO CAL ME NU
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
I P A d d r e ss
S u b n e t Ma s k
the settings.
Connection Type
network.

IP Address
Subnet Mask
digit keys.
Gateway Address
DNS Server
Address.
DNS Address
DNS Domain Name
PPPoE User ID
PPPoE Password
provider here.
PPPoE Service Name

Server Address
Server Port
port.
VLAN Setting
CoS Traffic Type
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
PC port).

PCAP
PCAP Trace.
MAC Cloning
• Select Enable?.
connected to LAN.
• Select Enable?.
administrator.

MAC: 0 0:1 b:0 9:0 0:9 a:a 7
IP : 1 92. 16 8. 2 0 1 .2 0 5
MASK: 2 5 5 . 25 5 . 2 5 5 .0
G W: 1 9 2 . 16 8 . 2 0 1 .3
DNS:


SARVAM UCS:

Extensions.



OR
scenario.

Server.
Network Settings
your phone.


the settings.
Connection Type
network.

IP Address
Subnet Mask
digit keys.
Gateway Address
DNS Server
Address.
DNS Address
DNS Domain Name
PPPoE User ID
PPPoE Password
provider here.
PPPoE Service Name

Server Address
Server Port
port.
VLAN Setting
CoS Traffic Type
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
PC port).

PCAP
under PCAP Trace.
MAC Cloning
connected to LAN.
administrator.



UCS:

holes you drilled.
2.
into the Keyholes.
Angle.

OR
Ethernet Cable.
OR

network.
Server.
Settings .


Extensions.
Keyhole Keyhole
Slot 1 Slot 2

Stand attached at
45 degree angle
Stand attached at
55 degree angle

AUX
Handset
OR

VP510”.
Power Adapter.
loaded.
Server.

Network Settings
state.

To exit menu,
or
Go ON-Hook.




2Ab/123
123 - Numeric Mode
Connection Type
network.
IP Address
Subnet Mask
digit keys.
Gateway Address
DNS Server
Address.

DNS Address
DNS Domain Name
PPPoE User ID
PPPoE Password
provider here.
PPPoE Service Name
Server Address
Server Port
this port.
VLAN Setting

CoS Traffic Type
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
PC port).
PCAP
MAC Cloning

connected to LAN.
administrator.


given below:








Attach the Clamps




Starting Up ETERNITY MENX
Power ON
• If you have completed all the installation tasks, switch on power supply. Keep the MCB Switch ON of your
PS48V card installed in the system and power the FCBC.
• Observe the Reset Cycle.
Reset Cycle
turned on.
Interpreting LEDs

CHAPTER 7 Installing ETERNITY GENX
• The Matrix ETERNITY GENX is to be installed by persons trained and experienced in telecom wiring.
• The person installing the ETERNITY GENX must be familiar with trunks, physical wiring of the MDF on
now begin the installation of your ETERNITY GENX.

The Matrix ETERNITY GENX is shipped factory fitted with the Power Supply Card, the CPU Card in their respective
ETERNITY GENX has a total of 12 Universal slots. The Slave Cards - BRI, T1E1PRI, GSM/CDMA, DKP, CO, SLT,
E&M, Radio, Data, Magneto, E1FO - are shipped separately as per the order placed by individual customers.
These Cards are installed in any of the Universal slots.
universal slots. No universal slots will be functional by default. You must purchase the SARVAM EXP4 SME license
Firmware Version V1R3 and later

The Matrix ETERNITY GENX is shipped factory fitted with the Power Supply Card, the CPU Card in their respective

ETERNITY GENX has a total of 12 Universal slots. The Slave Cards - BRI, T1E1PRI, GSM/CDMA, DKP, CO, SLT,
E&M, Radio, Data, Magneto, E1FO - are shipped separately as per the order placed by individual customers.
These Cards are installed in any of the Universal slots.
If you have upgraded the system firmware to V1R3 and later in the old ETERNITY GENX system, the Expansion
purchase the SARVAM EXP4 SME license to activate the universal slots as required.
If you have purchased the new ETERNITY GENX system with the firmware V1R3, the Expansion Slots license will
be applicable for the universal slots. The first four universal slots after the power supply card will be functional by
default. If you require more functional universal slots, you must purchase the SARVAM EXP4 SME license.
Each SARVAM EXP4 SME license will provide the activation for next four universal slots in sequence.
Illustrated below is the position of the fixed and universal slots in ETERNITY GENX.
ETERNITY GENX
The extreme left slot is reserved for the Power Supply Card and the extreme right slot is reserved for the CPU
Card.
Mounting the System

2. Decide where to mount the ETERNITY GENX - on a table or wall - taking into consideration the
mechanical dimensions and the weight. If mounting the system on a wall, you may refer the mechanical
dimensions and the Mounting Template for drilling holes at appropriate places on the wall. Make sure the
system orientation is horizontal.

Inserting Cards
7. Make sure that the ETERNITY GENX power is off and the power cord is unplugged.
GSM/CDMA, E&M, E1FO, Magneto, Radio - later in this section. Refer to them when installing each card
type.


applicable.

Two types of Power Supply Cards are supported by the system: ETERNITY GE CARD PSUNI (250 W) and
ETERNITY GE CARD PS48VDC (250W).
• ETERNITY GE CARD PSUNI (250W) with 100-240VAC, 47-63Hz Mains as Input AC Voltage Power
Supply and output 250W.
This card is designed on the SMPS scheme. As this card does not have any provision for battery backup,
it is recommended that a UPS be connected to keep the system powered during outages.
This card has four Green LEDs, a Mains Switch, and a Socket assembly for connecting the mains cord.
• ETERNITY GE CARD PS48VDC (250W) with 48VDC as Input DC Power Supply Voltage and output as
250W. A Float cum Boost Charger (FCBC) is required to feed 48VDC power to the card. The FCBC works
on input AC mains.
The card has four Green LEDs (for four different voltages), one Red LED (for input reverse indication, an
MCB Switch, a power ON/OFF Switch, and a 3-way termination block for connecting the power cord.
Both, the PSUNI card and the PS48V Card provide DC output voltages as: +3.5V, +5.0V, -30V and -85V.
These are indicated by LEDs.

out of the slot.
AC Power Supply Card must be removed from the platform three minutes after the power supply is
switched off.
Make sure you do not place the Power Supply Card on any conductive surface.
2. Insert the Power Supply card into the guide rails of the first slot on the extreme left, designated for the
Power Supply Card. Make sure that the card is inserted deep enough to make perfect contact with the
connectors on the motherboard at the backplane.
5. If installing the PSUNI card, connect the three-pin power cord into the socket of the PSUNI card and plug
in the cord into the mains supply.

You may connect the PSUNI Card to a UPS to keep the system live during power outages.
Select a UPS considering the typical power consumption of the system. See “Power Consumption Table”.
6. If installing the Eternity GE CARD PS48VDC (250W), connect the Float cum Boost Charger (FCBC) or
AC-DC Power Supply with 48VDC Output. Terminate the power cord from the FCBC output into the 3-way
termination block on the PS48V card.
Color Signal Symbol
Red +48VDC +
Black GND -
Green Earth

Connecting an FCBC to the System
As PS48VDC Card works with FCBC, the input voltage working range has hysterisis.The power supply turns ON at
Voltage greater than 45VDC and it remains ON till 40VDC during battery discharge condition. Moreover, the input
voltage for the card must be from 40VDC to 60VDC.
ETERNITY GE CARD PS48VDC
FCBC
BATTERY LOAD
48VDC
ETERNITY GENX
48VDC BATTERY
+3.5VDC (LED L1)
AC MAINS SUPPLY
230VA C +/-
FCBC ETERNITY GE
10% @ 50 Hz 10A CARD PS48VDC
48V BATTERY
7. Connect the Battery Backup70. Battery backup time depends upon the total load. The total load is the sum
of system's load and load of active extensions.
8. Switch on power supply, after completing all other installation tasks.
70. When the batteries are drained, the FCBC goes into the charge mode and begins to charge the batteries at higher current. When the
batteries reach a preset voltage level (typically set to 56.0 volts), the FCBC goes to float mode. In the float mode the FCBC keeps
charging the battery but at lower current. The FCBC monitors the voltage level of the batteries. As soon as the battery voltage goes
below preset voltage (typically set to 50.4 volts), FCBC goes from float mode to charge mode. The change over from mains to battery
and vice-versa is automatic. The advantage of using an FCBC is that batteries get charged faster, since the batteries are charged
with higher current initially.

Power Consumption Table
Power Consumptiona Battery Backupb

Platform Number of Ports Off - Hook
30%
30%
ETERNITY GENX 227 W (AC)

240 5.5hr(b)
(System) 183 W (DC)
a. Power Consumption is in Watts, if you want it in BTU per Hour then use the following
relation:
1Watt=3.412 BTU/Hr
b. This Backup detail shown is considered with respect to 48V | 25Ah Battery.
The Power Supply unit can prove hazardous due to high Voltage Power it carries.
Make sure you get it serviced by a trained service person only.

The CPU Card
The ETERNITY GENX CPU Card hosts the SARVAM UCS Application. It supports two VOCODER modules and
one VMS module. Both the modules — NX DBM VOCODER64 and NX DBM VMS64 are optional and can be
and modules which controls all other slave cards (SLT, DKP, CO+SLT, DKP+SLT, E&M, BRI, T1E1,E1FO, GSM/
CDMA etc.). All the configuration and programming information is stored on this card.
The CPU Card occupies a fixed slot, the first slot on the extreme right, with a unique arrangement of connectors. So
no other card can be inserted in the slot of the CPU Card.
The CPU Card has a WAN Port, LAN Port, USB Port and COM Port on the front panel. It also has an Internal USB
Port with a factory fitted pendrive.

L1 L2
LAN
WAN
COM 10101
USB
LAN RJ45
LAN Switch.
WAN RJ45
Router/Modem.
Converter.
USB to COM The USB be COM Port can be used to:

Activity logs.
Used to:
• set up and run software applications — PMS and CAS.
COM DB-9 • capture System Activity Log, System Fault log and Hotel Motel
Activity logs.

LAN Interface
WAN Interface
VoIP Interface
The CPU Card supports two NX DBM VOCODER64 modules. You must purchase the module separately for VoIP
functionality.
VOCODER Channels
The system supports two NX DBM VOCODER64 Modules. Each module supports 64 VOCODER Channels71. You
CHNL16.
VMS Interface
Call Taping.

SARVAM UCS SME Application. The Pen Drive is also the storage device for mailbox messages.
Internal USB Port

Internal USB Port
mail boxes, Matrix Extended IP phone firmwares and SMS Server firmware.
When you select the SARVAM UCS SME Application, the system fetches the application from the pendrive.

COM Converter.
If you use any other USB to COM Converter, Matrix does not guarantee it’s proper functioning.

Communication Port (COM Port)

There is a asynchronous, serial, full-duplex RS-232C Communication (COM) Port, labeled as COM10101. The
COM Port has a DB-9 connector.
The COM port allows you to connect a PC to the system, so that you can install and operate the following features:
LED
The CPU Card has two dual color (Green and Red) LEDs.
• LED 1 - L1 works as a Heart Bit of CPU Card. In Normal Condition, L1 will be turned ON Green for 1 sec
and OFF for 1 sec.
• LED 2 - L2 indicates the Layer Application status. In Normal condition, L2 will be turned ON Orange and
will blink very fast.
• Both L1 and L2 also indicate Application status.
Case 1: L1 is steady GREEN/OFF and LED2 is OFF. This means the Application is hung and there is
some problem in the Application code.
Case 2: LED1 is steady GREEN/OFF and LED2 is GREEN/RED/ORANGE. This means the Layer is hung
and there is some problem with the Layer code.
Jumpers
The position and function of the Jumpers on the CPU Card are:
J1 AB Default SE Password.
BC (default) Normal.

To install,
• The NX DBM VOCODER64 Module is to be mounted adjacent to the fan on the CPU board.
PCI PCI
Connector Latch
• Locate the PCI Connector and PCI Latch on the CPU board.

area.

• Press the Module with a finger to fix the latches perfectly into the mounting holes. Make sure you do not
touch the PCB area of the module except the yellow line provided for grounding at the front end of the
module.
Do not apply excessive pressure. Follow the same steps if you wish to install another module.

NX DBM VOCODER64
Module

• Press both the latches together.
Make sure you support the base of the latches from behind with your forefinger.

To install,
• Locate the PCI connector for NX DBM VMS64 Module on the CPU card.
PCI PCI
Connector Latch
• Follow the same steps as described in installing the NX DBM VOCODER64 Module. See “Installing the
VOCODER Module”.
NX DBM VMS64 Module
Make sure you switch-off the system to replace the pendrive. The system will not detect the new pendrive
if you do not restart the system after replacement.

• If you have no other modules to install, insert the card back into the ETERNITY GENX.

wire trunk and digital key phone interfaces on a single card.
SLT Cards for ETERNITY GENX
ETERNITY GE CARD SLT20 20-port card to connect 20 Single Line Telephones
ETERNITY GE CARD DKP4+SLT16 Combination card, with 4-ports to connect to 4 Digital Key Phones and
16 ports to connect 16 Single Line Telephones
ETERNITY GE CARD CO4+SLT16 Combination card with 4 ports to connect 4 Two-wire Trunk lines, and
16 ports to connect 16 Single Line Telephones
Combination card, with 4 ports to connect 4 Two-wire Trunk lines, 2

ports to connect 2 Digital Key Phones and 12 ports to connect 12
ETERNITY GE CARD Single Line Telephones.
CO4+DKP2+SLT12
This Card supports Power Fail Transfer. To know more, see “Power
Fail Transfer”.
Combination card, with 4 ports to connect 4 Two-wire Trunk lines, 2

ports to connect 2 Digital Key Phones and 8 ports to connect 8 Single
ETERNITY GE CARD Line Telephones.
CO4+DKP2+SLT8
This Card supports Power Fail Transfer. To know more, see “Power
Fail Transfer”.
ETERNITY GE CARD Combination card, with 2 ports to connect 2 Two-wire Trunk lines, 2
CO2+DKP2+SLT16 ports to connect 2 Digital Key Phones, and 16 ports to connect 16
Single Line Telephones
Connectors
for each connector.
LEDs
The Cards SLT8 and SLT16 have 2 LEDs, while SLT20 has no LED:
• The LEDs indicate the health of the card during the Reset Cycle.
• the status of any one of the extension ports during normal functioning of the system.

You can monitor any of the SLT Extension ports by assigning the LED to that port72.
LED Pattern of SLT Card
LED 2 (L2)
Commands from Application to SLT Port. GREEN Togglea at each command
Events to Application from SLT Port. RED Toggleb at each event
a. The current LED state will remain the same until the next command is received from the ap-
plication on the SLT Port. For example, if the current LED state is Green/Red ON, on the
next command received, the LED will be turned OFF. It will remain OFF until the next com-
mand is received. When the next command is received it will be turned Green/Red ON
again. This process continues.

backplane.
72. To do this, enter SE mode, enter the programming mode from any extension connected to the system, by dialing 1#91-SE Pass-
word. Dial the command 7902-Slot-LED Number-Port, where Slot is the number of the universal slot in which the card is installed
and Port is the port on the card to which the LED is to be assigned to monitor its functioning. LED Number is the number of the
LED on the card, which will monitor the port. Exit programming mode by dialing '00'.

ETERNITY GE CARD SLT20

Green - (Green & White) SLT 03
Green - (Green & White) - -
Brown - (Brown & White) - -



ETERNITY GE CARD DKP4+SLT16

ETERNITY GE CARD CO2+DKP2+SLT16


ETERNITY GE CARD CO4+SLT16

ILC Cards for ETERNITY GENX
ETERNITY GE CARD ILC20 20-port card to connect 20 Single Line Telephones
The maximum number of intercom ports supported are 240.
Connectors
The ILC Cards have RJ45 connectors, with each connector having 4 ports. A multi-pair, MDF cable is supplied for
each connector.
LEDs
The Card ILC 20 has no LED:
• The LEDs indicate the health of the card during the Reset Cycle.
• the status of any one of the extension ports during normal functioning of the system.

jacks.
4. Unpack the ILC Card and check the package contents. Switch off power supply before you install the card.
backplane.

ETERNITY GE CARD ILC20

10. If you have completed all other installation tasks, power ON the system, observe the Reset Cycle.

series, the proprietary PC-based phone EONSOFT, the Direct Station Selection (DSS) Consoles, with the system.
The DKP Card is available in the following configurations:
DKP Cards for ETERNITY GENX
ETERNITY GE CARD
16-port card to connect 16 DKP/DSS Consoles
DKP16
ETERNITY GE CARD
8-port card to connect 8 DKP/DSS Consoles
DKP8
ETERNITY GE CARD Combination card, with 4-ports to connect to 4 Digital Key Phones and 16 ports to
DKP4+SLT16 connect 16 Single Line Telephones
ETERNITY GE CARD Combination card, with 2 ports to connect 2 Two-wire Trunk lines, 2 ports to connect 2
CO2+DKP2+SLT16 Digital Key Phones, and 16 ports to connect 16 Single Line Telephones
Combination card, with 4 ports to connect 4 Two-wire Trunk lines, 2 ports to connect 2
ETERNITY GE CARD Digital Key Phones and 12 ports to connect 12 Single Line Telephones.
CO4+DKP2+SLT12
This Card supports Power Fail Transfer. To know more, see “Power Fail Transfer”.
Combination card, with 4 ports to connect 4 Two-wire Trunk lines, 2 ports to connect 2
ETERNITY GE CARD Digital Key Phones and 8 ports to connect 8 Single Line Telephones.
CO4+DKP2+SLT8
This Card supports Power Fail Transfer. To know more, see “Power Fail Transfer”.
To connect the proprietary digital key phones with the system, you must have at least one of the above mentioned
DKP Cards installed in the system.
Connectors
LEDs
The DKP16 and DKP8 Cards have two dual color LEDs:
• LED1 indicates the health of the card during the Reset Cycle.
• LED2 monitors the status of any one of the extension ports during normal functioning of the system.
LED2 can be assigned to any DKP port to monitor the status of that port73.

DSS Consoles.
motherboard.
slot.

ETERNITY GE CARD DKP16



6. Plug in the RJ45 end of the MDF cables provided with the DKP Card into the respective connectors.
instructions for EON and EONSOFT are provided separately.
of the DKP Card.
LED 2 (L2)
Commands from Application to DKP Port. GREEN Togglea

Events to Application from DKP Port. RED

a. The current LED state will remain the same until the next command is received
from the application on the DKP Port. For example, if the current LED state is
Green/Red ON, on the next command received, the LED will be turned OFF. It
will remain OFF until the next command is received. When the next command
is received it will be turned Green/Red ON again. This process continues.
Installing EON48

Angle.

Headset

M AT R I X E O N 4 8 - S V 2 R 2
PLEASE WAI T .. .
202 Reception
M on 2 4 A U G 1 2 : 0 0
Installing EON310




Operation”.
Installing EON510

Keyhole Keyhole
Slot 1 Slot 2

Stand attached at
45 degree angle
Stand attached at
55 degree angle
AUX
Handset

Operation”.

Installing DSS64
Console Keys”.

The instructions for installting DSS532 with EON510 or SPARSH VP510 are same. For detailed instructions, refer
to “Installing DSS532 with SPARSH VP510”.
Installing EONSOFT
• Windows 98
• Windows XP
• Windows NT
• Windows 2003
• Windows Vista
• Windows 2007

application.
desktop.



data transfer.
EONSOFT.



The CO Card
The CO Card is available in the following configurations. CO interface is also available in combination with SLT and
DKP ports on a single card.
CO Cards for ETERNITY GENX
ETERNITY GE CARD CO16 16-port card to connect 16 Two-wire Trunk lines from the CO network
ETERNITY GE CARD CO8 8-port card to connect 8 Two-wire Trunk lines from the CO network
ETERNITY GE CARD Combination card, with 4 CO ports to connect 4 CO analog trunk lines and 16
CO4+SLT16 SLT ports to connect 16 Single Line Telephones
ETERNITY GE CARD Combination card, with 2 CO ports to connect 2 Two-wire Trunk lines, 2 DKP
CO2+DKP2+SLT16 ports to connect 2 Digital Key Phones, and 16 SLT ports to connect 16 Single
Line Telephones
Combination card, with 4 ports to connect 4 Two-wire Trunk lines, 2 ports to

connect 2 Digital Key Phones and 12 ports to connect 12 Single Line
ETERNITY GE CARD Telephones.
CO4+DKP2+SLT12
Transfer”.
Combination card, with 4 ports to connect 4 Two-wire Trunk lines, 2 ports to

connect 2 Digital Key Phones and 8 ports to connect 8 Single Line Telephones.
ETERNITY GE CARD
CO4+DKP2+SLT8
Transfer”.
Connectors
LED
The CO16 and CO8 Cards have two LEDs to indicate:
• the health of the card during the Reset Cycle

• the status of a selected Trunk port during normal functioning of the system
-LED Number-Port, where Slot is the number of the universal slot in which the card is installed and Port is the port on the card to
the port.

Among the combination cards CO2+DKP2+SLT16, CO2+DKP2+SLT8, CO4+CO16, CO4+DKP2+SLT8
have no LEDs. The combination card CO8+SLT8 has two LEDs.
LED Pattern of CO Card
LED 2 (L2)
Commands from Application to CO Port. GREEN Togglea

Events to Application from CO Port. RED
a. The current LED state will remain the same until the next command is received
from the application on the CO Port. For example, if the current LED state is
Green/Red ON, on the next command received, the LED will be turned OFF. It
will remain OFF until the next command is received. When the next command
is received it will be turned Green/Red ON again. This process continues.

Distribution Frame.

ETERNITY GE CARD CO16
L1 L2
Green - (Green & White) CO 03
3 Green - (Green & White) CO 11
4 Green - (Green & White) CO 15
L1 L2
RJ45-3 Unused
3
RJ45-4 Unused
4

L1 L2
3 Green - (Green & White) SLT 11
4 Green - (Green & White) SLT 15
5

6



7. Plug in the RJ45 end of the Trunk Card cables into the respective connectors. Refer to the connector
diagrams illustrated above for each CO Card type.
the MDF”.

The BRI Card
BRI Cards of ETERNITY GENX
ETERNITY GE CARD
4-Port card to connect 4 ISDN BRI Lines or ISDN Compatible Devices
BRI4
Connectors
The BRI Card has 4 RJ45 Connectors. A separate cable is supplied for each connector.

ISDN
Network NT BRI Port
Power
ETERNITY GENX
U-Interface S/T Interface
(2-wire)
Customer Premises
Where,
• S/T Interface = between the ISDN user equipment, in this case, ETERNITY GENX and the Network
GENX.
TE and NT Modes
In this illustration, the BRI line from ISDN Service Provider is directly connected to BRI port of the ETERNITY
GENX via the NT1 device. Here, the ETERNITY GENX is the Terminal Equipment, so the BRI Port must be
programmed to work in the TE mode.

When an ISDN Phone is to be connected to the BRI port of ETERNITY GENX, the BRI port must be programmed
to work in NT mode.
When a BRI port of another ISDN System is to be connected to the BRI port of the ETERNITY GENX, in such a
• the BRI port of the other ISDN System in the TE mode and the BRI Port of the ETERNITY GENX in the NT
mode.
OR
• the BRI port of the other ISDN System in the NT mode and the BRI Port of the ETERNITY GENX in the TE
mode

BRI Line
NT BRI Port
(TE Mode)
ISDN
Network
ETERNITY GENX
Upto 1 Km
this case ETERNITY GENX, can be up to 1 kilometer.

d < 200 meters
ISDN NT
Network BRI Bus Bar
Terminal
BRI Port Resistance 100
(TE Mode)
Terminal
Resistance 100
ETERNITY GENX ISDN Phone ISDN Phone ISDN Phone
Terminal 1 Terminal 2 Terminal 3 Terminal 8
Where,
NT.
• Using this configuration, any subscriber from ETERNITY GENX can access a BRI line and can make
outgoing calls. At the same time, another subscriber from ETERNITY GENX or any ISDN phone shown in
same BRI.
used.

d < 1 Km
d < 30 m
ISDN NT
Network BRI Bus Bar
Terminal
BRI Port Resistance 100
(TE Mode)
Terminal
Resistance 100
ETERNITY GENX ISDN Phone ISDN Phone
Terminal 1 Terminal 2 Terminal 3
Where,
• Using this configuration, any subscriber from ETERNITY GENX can access the BRI line and make
outgoing calls. At the same time, another subscriber from the ETERNITY GENX or any ISDN phone
shown in the figure can make outgoing calls from the same BRI. In the same way, incoming calls are
possible on the same BRI.


the NT mode.

shown below.
BRI Line
ISDN
Network
NT
100
ETERNITY GENX
ISDN BRI Line

NT
Network
100
BRI TE BRI TE BRI TE
Other ISDN Other ISDN

Equipment Equipment ETERNITY GENX

• Termination need not be inserted if the BRI port of ETERNITY GENX (configured in TE mode) is

6. To set the 100 termination on the BRI port set the Jumpers on the BRI module of each port as given in
the table below.

Function
J6 J8 J7 J9 J6 J8 J7 J9
To insert 100
termination
To remove 100
termination
By default, Termination Resistance of 100 is set on the BRI port (the Jumpers are in AB position)

1 RJ45 Connector on
Bus Bar at the Last
TE ISDN Equipment
Tx 3
100
Rx 4
Rx 5
100
Tx 6
Bar configuration.

bar.
1 1 RJ45 Connector
ports on BRI Bus
Bar to which the
3 3
ISDN TE
4 4 Equipment is
connected
5 5
6 6
8 8
on the Bus bar.
the ISDN TE side.
the NT side.

8. When the BRI Port of the ETERNITY GENX is used as BRI-NT, you can feed power to the terminal
equipment connected to the BRI-NT Port from the ETERNITY GENX.
To do this,

Function
J4 J5 J2 J3 J4 J5 J2 J3


pair of wires

guidance.
Configuration details of the U interface (RJ-45) at NT1
4 Tx
5 Rx
3 Rx1
4 Tx1
5 Tx2
6 Rx2
3 Green-White TxA
4 Blue RxA
5 Blue-White RxB
6 Green TxB

7 Brown-White V-
8 Brown V+
3 Green-White RxA
4 Blue TxA
5 Blue-White TxB
6 Green RxB
7 Brown-White V-
8 Brown V+
NT-1 ETERNITY GENX

3
Rx1 TxA
4 4
Tx Tx1 RxA
ISDN
Network 5 5
Rx Tx2 RxB
6
Rx2 TxB
NC V-
Power NC V+
U-Interface S/T Interface BRI port

(2-wire) (4-wire)

The BRI4 Card has an LED for each port: L1, L2, L3, L4.

The T1E1PRI Card
The ETERNITY GE CARD T1E1PRI Single provides the interface to connect the system to ISDN PRI Network.
• PRI
• E&M
• PRI
• E&M
T1E1PRI Card for ETERNITY GENX
ETERNITY GE CARD T1E1PRI 1-Port card with QSIG support to connect 1 ISDN T1/E1 PRI Line or ISDN
Connectors
The T1E1PRI Card has an RJ45 Connector. A cable with RJ45 plugs on both ends is supplied for the connector.
LEDs
The ETERNITY GE CARD T1E1PRI Single has 2 LEDs - L1 and L2 - for indicating the port states.

turned off.

4. To set the Line Termination Resistor for the PRI Port for T1 or E1 Connectivity, you must change the
position of the jumper J5. Refer the table below.
Jumper Position Meaning
J5 BC To set termination resistance of 120 for E1 connectivity
J5 AB To set termination resistance of 100 for T1 connectivity
• By default J5 is set to BC position to provide 120 termination resistance for E1 connectivity.
• To set 100 termination resistance for T1 connectivity, set jumper J5 to AB position.

ETERNITY GENX
4-wire 4-wire
PRI Port
ISDN
HDSL DTE
(RJ-45 Connector) (RJ-45 Connector)
Network
G.703
Modem
G.703
Modem
Power
ETERNITY GE T1E1PRI Card).

1 Line A
2 Line A
3 Not used
4 Line B
5 Line B
6 Not used
7 Not used
8 Not used
1 TX1 (Tip)
2 TX2 (Ring)
3 Not used
4 RX1 (Ring)
5 RX2 (Tip)
6 Not used
7 Not used
8 Not used

Pin details of ETERNITY GENX T1E1PR1 Port
below.

4 5
NC NC
3 6
Rx2 (Tip) NC
2 7
Rx1 (Ring) NC
1 8
LED Patterns
The ETERNITY GE CARD T1E1PRI has 2 LEDs: L1 and L2. Given below are the LED Patterns defined for each
port state in the different signaling types supported by the system.
1. Port Active Mode

LED1 Pattern:
LED2 Pattern:

LED1 Pattern:
LED2 Pattern:
LED1 Pattern:
TFA Alarm or MFA Alarm RED 500ms ON-500 ms OFF
LED2 Pattern:
LED1 Pattern:

LED2 Pattern:
Near end loop back wait Before activate RED 100ms ON-100 ms OFF
Near end loop back wait Before deactivate RED 500ms ON-500 ms OFF
LED1 Pattern:
LED2 Pattern:

The E1FO Card
The ETERNITY GE CARD E1FO provides the interface to connect the system to the ISDN PRI Network. For E1
E1FO Card for ETERNITY GENX
ETERNITY GE CARD E1FO 1-Port card to connect 1 ISDN T1/E1 Line on copper interface or E1 line on
Single Fiber Optic (FO) interface.
Connectors
The E1FO Card has one RJ45 Connector and one Fiber Optic connector.
LEDs
The ETERNITY GE E1FO Card has 2 LEDs - L1 and L2 - for indicating the port states.

turned off.

4. To set the Line Termination Resistor for the Port for T1 or E1 Connectivity, you must change the position of
the jumper J5. Refer the table below.
Jumper Position Meaning
J5 BC To set termination resistance of 120 for E1 connectivity
J5 AB To set termination resistance of 100 for T1 connectivity
• By default J5 is set to BC position to provide 120 termination resistance for E1 connectivity.

• To set 100 termination resistance for T1 connectivity, set jumper J5 to AB position.

ETERNITY GENX
4-wire 4-wire
T1/E1 Interface
ISDN
HDSL DTE
(RJ-45 Connector) (RJ-45 Connector)
Network
G.703
Modem
G.703
Modem
Power
ETERNITY GE CARD E1FOPRI Single).
1 Line A
2 Line A
3 Not used
4 Line B
5 Line B

6 Not used
7 Not used
8 Not used
1 TX1 (Tip)
2 TX2 (Ring)
3 Not used
4 RX1 (Ring)
5 RX2 (Tip)
6 Not used
7 Not used
8 Not used
The T1/E1 Interface of the ETERNITY GE CARD E1FOPRI Single terminates in an 8-pin RJ45, female connector

4 5
NC NC
3 6
Rx2 (Tip) NC
2 7
Rx1 (Ring) NC
1 8

ETERNITY GENX
ISDN FO
E1FO Interface
Infrastructure
Network
LED Patterns
The ETERNITY GE E1FO Card has 2 LEDs: L1 and L2. Given below are the LED Patterns defined for each port
state in the different signaling types supported.
1. Port Active Mode

LED1 Pattern:

LED2 Pattern:
LED1 Pattern:
LED2 Pattern:
LED1 Pattern:
TFA Alarm or MFA Alarm RED 500ms ON-500 ms OFF
LED2 Pattern:

LED1 Pattern:
LED2 Pattern:
Near end loop back wait Before activate RED 100ms ON-100 ms OFF
Near end loop back wait Before deactivate RED 500ms ON-500 ms OFF
LED1 Pattern:
LED2 Pattern:

The E&M Card
lines78.

E&M Card for ETERNITY GENX
ETERNITY GE CARD E&M4 4-port card to connect 4 E&M Tie Lines
Connectors
LEDs
The ETERNITY GE E&M4 Card has 4 LEDs to indicate the functioning of the ports.
Protocol.

OR
through it.
OR

BC Position.
Position.
Pinout details of ETERNITY GE CARD E&M4

M M
SB SB
E E
SG SG
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS

M M
SB NC NC SB
E E
SG SG
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS

M M
SB SB
E E
SG SG
SP(RxA) SP(RxA)
SP(RxB) SP(RxB)
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS

M M
SB NC NC SB
E E
SG SG
SP(RxA) SP(RxA)
SP(RxB) SP(RxB)
SP(A) SP(A)
SP(B) SP(B)
CCS NC NC CCS
M NC NC M
SB NC NC SB
E NC NC E
SG NC NC SG
SP(A) NC NC SP(A)
SP(B) NC NC SP(B)
CCS CCS
PLCC PLCC
to the PLCC panel.
LED Pattern of E&M Card
At Power ON LED OFF
After 30-60 seconds LED OFF

After 60-90 seconds RED L1, L2, L3, L4 ON 500ms - L1, L2, L3, L4 OFF
GREEN L1, L2, L3, L4 ON 500ms - L1, L2, L3, L4 OFF
After 65-95 seconds RED L1, L2 L3, L4 ON 500ms - L1, L2, L3, L4 OFF
GREEN L1, L2 L3, L4 ON 500ms - L1, L2, L3, L4 OFF
Normal (Port Event)
M-Wire High Green LED of the Port continuously ON
E-Wire High Red LED of the Port continuously ON
E-Wire Low LED of the Port continuously OFF
E-Wire and M-Wire High Orange LED of the Port continuously ON

The Mobile Card
The Mobile Card interfaces the system with GSM/3G/CDMA networks. It routes calls made and received over
mobile networks, like a mobile handset.
The Mobile Cards are available in CDMA, 2G and 3G variants.
The Mobile Card for ETERNITY GENX
ETERNITY GE CARD GSM4 4-port card to connect to 4 GSM networks (4 SIM Cards can be
installed)
ETERNITY GE CARD GSM4 3G 4-port card to connect to 4 GSM networks with 3G support (4 SIM
Cards can be installed)
ETERNITY GE CARD CDMA4 4-port card to connect to 4 CDMA networks (4 RUIM Cards can be
installed)
If you are installing CDMA Mobile Card in your system, it is recommended to avoid using features that
Antenna
ETERNITY GE CARD GSM4 has a single antenna for the four ports. A splitter connects all the four ports on the
card into a single antenna. An antenna cable is also provided, giving you the flexibility to move the antenna to
another position (in case of weak signal).
Personal Identification Number (PIN)80

80. Not applicable if you are installing CDMA Mobile Card in your system.

To be able to connect the system to GSM/3G/CDMA networks, you must have one of the above mobile cards
installed in the system.
• If using a GSM/3G/CDMA card, get the SIM Card from the GSM/3G/CDMA service provider of your
choice ready. Use SIM PIN protection81, if required.
ETERNITY GE CARD GSM4/CDMA482
Enabling PIN Protection on SIM83


81. SIM PIN Protection is not applicable if you are installing CDMA Mobile Card.
82. Insert RUIM Card instead of SIM Card if you are installing CDMA Mobile Card.
83. Disable PIN Protection on RUIM if you are installing CDMA Mobile Card in your system.


The number of LEDs on the Mobile Card corresponds with the number of mobile ports on the card.
After the Reset cycle is complete, during normal functioning, the LEDs color and cadence is described in the table
below for various events on the Mobile port:
Port idle - LED OFF




OFF (3 blinks)


The Magneto Card
Magneto Card for ETERNITY GENX
ETERNITY GE CARD
4-port card to connect 4 Magneto Phones
MAGNETO4
Connectors
LED
The ETERNITY GE CARD MAGNETO4 does not support any LED.

OR


ETERNITY GE CARD MAGNETO4 on the MDF. Refer the cable diagram below.
1 RJ45-1 Blue - (Blue & White) Magneto 01

(Blue) Orange - (Orange & White) - -
RJ45-1 Blue - (Blue & White) Magneto 02
2 (Orange) Orange - (Orange & White) - -
3 (Green) Orange - (Orange & White) - -
4 (Brown) Orange - (Orange & White) - -
system.

The Radio Card
Radio Cards for ETERNITY GENX
ETERNITY GE CARD RADIO4 4-port card to connect 4 Radio devices
Connectors
LED
The ETERNITY GE CARD RADIO4 does not support any LED.

OR

ETERNITY GENX Card Radio4
H/w Port
Offset

to
Orange 2 PTT_RTN
RJ45-4
Green & White 3 Rx-
Blue 4 Tx+
Blue & White 5 Tx-
Green 6 Rx+
Brown 8 Unused
system.

The Data Card

LEDs
The Data Card does not have any LED.

5. If you have completed all other installation tasks, you may turn ON the system.
Jumpers
BC Internal Boot

SIP Extensions

AMP100 User Guide.

Addresses only.
Extension.
• the SIP ID

SARVAM UCS
LAN Switch/ Hub
SIP Extn.
SIP Extn.
Matrix VARTA
WIN200
IP
Router SIP Extn.
SARVAM UCS
SIP Extn.
LAN Switch/ Hub
IP
Router SIP Extn.
SIP Extn.
Matrix VARTA
WIN200
• You may also register any SIP device (Matrix VARTA UC Clients, Extended IP Phone or open SIP phone)
IP Phones.

SARVAM UCS:
Extensions.
holes you drilled.
2.
into the Keyholes.

desk.

Headset
OR

Foot Stand
Keyhole Keyhole
Slot 1 Slot 2
Headset
Handset
scenario.


Standar d S IP
Extended S I P

Yes
No
Server.
Network Settings
on your phone.




Yes
No
M on 10 M AY 1 5: 40
Local Menu
1 2 abc 3 def
4 ghi 5 jkl 6 mno
7 pqrs 8 tuv 9 wxyz

CA04
0 #
CA03 *
CA02
CA01

LO CAL ME NU
M A C : 0 0 : 1 b : 09 : 00 : 9a : a 7
I P A d d r e ss
S u b n e t Ma s k
the settings.
Connection Type
network.

IP Address
Subnet Mask
digit keys.
Gateway Address
DNS Server
Address.
DNS Address
DNS Domain Name
PPPoE User ID
PPPoE Password
provider here.
PPPoE Service Name

Server Address
Server Port
port.
VLAN Setting
CoS Traffic Type
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
PC port).

PCAP
PCAP Trace.
MAC Cloning
• Select Enable?.
connected to LAN.
• Select Enable?.
administrator.

MAC: 0 0:1 b:0 9:0 0:9 a:a 7
IP : 1 92. 16 8. 2 0 1 .2 0 5
MASK: 2 5 5 . 25 5 . 2 5 5 .0
G W: 1 9 2 . 16 8 . 2 0 1 .3
DNS:


SARVAM UCS:

Extensions.


OR
scenario.

Server.
Network Settings
your phone.


the settings.
Connection Type
network.

IP Address
Subnet Mask
digit keys.
Gateway Address
DNS Server
Address.
DNS Address
DNS Domain Name
PPPoE User ID
PPPoE Password
provider here.
PPPoE Service Name

Server Address
Server Port
port.
VLAN Setting
CoS Traffic Type
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
PC port).

PCAP
under PCAP Trace.
MAC Cloning
connected to LAN.
administrator.



UCS:

holes you drilled.
2.
into the Keyholes.
Angle.

OR
Ethernet Cable.
OR

network.
Server.
Settings .


Extensions.
Keyhole Keyhole
Slot 1 Slot 2

Stand attached at
45 degree angle
Stand attached at
55 degree angle

AUX
Handset
OR

VP510”.
Power Adapter.
loaded.
Server.

Network Settings
state.

To exit menu,
or
Go ON-Hook.



2Ab/123
123 - Numeric Mode
Connection Type

network.
IP Address
Subnet Mask
digit keys.
Gateway Address
DNS Server
Address.
DNS Address
DNS Domain Name
PPPoE User ID

PPPoE Password
provider here.
PPPoE Service Name
Server Address
Server Port
this port.
VLAN Setting
CoS Traffic Type
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice

CoS Traffic Type
7 Network Control
PC port).
PCAP
MAC Cloning
connected to LAN.

administrator.


given below:









Attach the Clamps




Starting Up ETERNITY GENX
Power ON
1. If you have completed all the installation tasks, switch on power supply.
• For PSUNI Card installed in the system, connect the three-prong plug of the power cord from the
system into the AC outlet, and switch on power supply.
• For PS48V Card installed in the system, keep the MCB Switch ON and power the FCBC.
2. Observe the Reset Cycle.
Reset Cycle
turned on.
Interpreting LEDs

CHAPTER 8 Configuring ETERNITY
GENX
ETERNITY GENX provides a Graphic User Interface (GUI), Jeeves, the proprietary web-based configuration
software of Matrix. The built-in web server Jeeves allows you to select the Application you want to run on the
ETERNITY GENX Platform.
The accessibility to the Web-based GUI is secured by a password. This password cannot be used to configure the
system using commands.
To be able to access Jeeves,
• the LAN/WAN Port of the system must be connected with a stand-alone PC or in a LAN.
• a web-browser, either Internet Explorer 7 or later or Mozilla Firefox 3.5.1 or later, must be installed on the
PC.
If the computer for accessing Jeeves is connected in a LAN Switch and the WAN Port of the system is
connected behind a NAT router, make sure that both the LAN and WAN connections are in different
Subnet Masks.
To login,
• Open the browser (Internet Explorer/Mozilla Firefox) on the PC (Standalone or LAN PC) to which the
system is connected.
• Make sure the IP Address of the computer and the LAN Port of the system do not conflict, and that both
are in the same Subnet.
The default IP Address of the LAN Port is: 192.168.2.100
The default Subnet Mask of the LAN Port is: 255.255.255.000
Change the Subnet of the computer, if necessary.
• In the address bar of the browser, enter https://192.168.2.100.

If you enter the IP Address 192.168.2.100 directly, you will be redirected to the HTTPS protocol for secure
access. Click the https://192.168.2.100 link provided.
• The Login page will open.
• In Login Password, enter 1234, the default Password.
• Click the Login button.
• On successful login, the Home page of Jeeves opens.
The left navigation bar displays the links — Application Selection, License Information, Status,
Firmware Management and Debug.
Application Selection enables you to select the application you wish to run on the ETERNITY GENX
platform. You may select — SARVAM UCS SME or SARVAM UMG.
License Information displays the License key along with the License details of the applications.
Status displays the system details and the status of all the ports.
Firmware Management enables you to manage the upgradation of the system software with a click of a
button.

Debug allows you to enable and configure the debug settings.
Application Selection
Through Application Selection, you can select the application you wish to run on the ETERNITY GENX Platform.
• In Select an Application, you may select the required application — SARVAM UCS SME or SARVAM
UMG. Default: SARVAM UCS SME.
• Select the Always start the selected application when system restarts check box, if you want the
selected application to start whenever the system restarts.
Keep the check box disabled only if you want to select the application to be run on the ETERNITY GENX
platform everytime the system restarts. Default: Disabled.
To run the selected application directly whenever the system restarts, Enable the Always start the
selected application when system restarts check box.
• Now, click on the Next button to proceed to the SARVAM UCS SME Configuration.
When you click the Next button, you will be redirected to the SARVAM UCS SME Application. To return
back to this page, see “Switch Application” in “System Details”.
Once you are redirected to the SARVAM UCS SME Application,
• If you have purchased the SARVAM UCS SME License, you must activate it. For detailed instructions,
refer “How to activate your License”.
• If you have not purchased the license and you wish to use the features on trial basis, you can use the
Demo Provision. Demo Provision enables you to use the SARVAM UCS SME application, free of cost for a
period of 60 days.
During the Demo Provision you can access and use all the features and functionalities93 supported by the
application. For detailed instructions, refer “Demo Provision”.
93. The number of Vocoder channels that will be supported in demo period will be as per the license you purchase.

• If you do not have the license for the SARVAM UCS SME Application and you do not start the Demo
Period, the system will disconnect all the connected calls94 from any port after 60 seconds.
License Information
License Information displays the License key along with the License details of the applications. Whenever you
want to activate license for any software application, you can check the current license key from here.
To activate the license, refer topic “How to activate your License”.
To know more regarding the licenses, refer topic “License Management”.
94. Connected calls means where speech is connected between the calling party port and the called party port even if the called party
port is not matured.

Status
Status displays the system details, that is, Product name and the Available Software Application, Network status,
WAN Port and LAN Port status.
Firmware Management
To upgrade firmware of ETERNITY GENX,
• In Upgrade Firmware from PC, click the Browse button to reach the location on the local disk where the
firmware files are stored in your PC. Make sure that the file is a zip file with .zip extension.
• After selecting the required firmware zip file from the PC, click the Upgrade button.
The system starts the upgradation process. After successful upgradation and validating the file, the system
restarts with the upgraded firmware.
If you select a file other than zip file, an Alert Message is displayed when you click the Upgrade button.

Debug
ETERNITY GENX supports Syslog Client for debugging. Debug messages are sent to the remote Syslog Server.
• Select the Enable Debug check box to enable system debug. Default: Disabled.
You will be able to configure the Debug Settings only after you enable this check box.
• In Syslog Server IP Address, enter the remote Syslog Server IP Address. Default: Blank.
In Syslog Server Port, enter the port number. The range of the server port is 514, 1024 to 65535. Default:
514.

CHAPTER 9 Configuring SARVAM UCS
The SARVAM UCS can be configured using the following tools:
• A web-based graphic user interface (GUI), called Jeeves.
• A telephone
• Serial Communication Port
Each of these is explained in detail in this chapter.
Configuring using Web-based GUI: Jeeves

SARVAM UCS provides a Graphic User Interface (GUI), Jeeves, the proprietary web-based configuration software
of Matrix. It is a Web server built into the SARVAM UCS.
Jeeves allows system configuration at three levels: System Engineer, System Administrator, and the Front Desk
User.
A distinct set of features and facilities can be configured at each of these levels. The accessibility to each level is
secured by a password. These passwords cannot be used to configure the system using commands.
These passwords can be changed using Jeeves only.
It is possible to configure the SARVAM UCS from any location using Jeeves. You can use Jeeves to
configure the system On-site (where it is installed) and Off-site, from a remote location.
1. System Engineer Mode

At the System Engineer level, the entire system configuration, all configurable features and facilities of the
SARVAM UCS can be changed to match user requirements and preferences.
To be able to do this, the System Engineer must enter the System Engineer (SE) configuration mode, by
logging into Jeeves as System Engineer.
Only the System Engineer, who installs, configures and maintains the is allowed access to this mode. Hence
access to the SE mode is protected by means of a password, referred throughout this document as the SE
Password.

To login into the SE mode, on the login page in Login as select System Engineer.
• In Password, enter the default SE password, 1234.
• Click Login.
• You are prompted to change the default password.
• In Current Password, enter the default SE password.
• Enter the New Password. The new password must be:

• a minimum of 6 characters to a maximum of 12 characters.
• all ASCII characters (except Percentage %, Hash #, Equal to =, Plus +, And &, Backslash \, Less than
<, Greater than >, Apostrophe ’, Double Quote " and Space) and digits 0 to 9 are allowed.
• include atleast one upper-case, one lower-case, one number and one special character.

• In Confirm New Password, re-enter the new password to confirm.
• Click Submit. You will be re-directed to the Login page again.
• Now, in Login as select System Engineer and in Password enter the new password.
As this password is meant for restricting access to the SE mode, we strongly recommend you to:
• Keep the password secret.
• Select a complex password that cannot be easily guessed.
• Change the password regularly. See “System Security”.
• Not use the “Remember Password” property of your Web Browser.
• Each login session into the SE Mode is set to 60 minutes by default. So, the login session will expire at
the end of 60 minutes. The duration of the login session can be extended or shortened according to
your preference by changing the settings for the 'Web Configuration Timer'. Refer the topic “Changing
Login Session Time Out of Jeeves”.
• It is possible for four users to simultaneously log into the System Engineer Mode of Jeeves. It is also
possible to log out all of these users at once or log out any of these users selectively. Refer the topic
“Logging Out Users from Jeeves”.
Quick Installation Wizards

For the ease of installation, as well as to simplify and speed up the process of setting up the SARVAM UCS, the
Jeeves offers two Wizards, namely:
• Quick Installation Wizard - Standard PBX: This wizard helps the Installer/System Engineer to quickly
set-up the SARVAM UCS for the standard UC (Enterprise) Application.
Using this Wizard, the Installer/System Engineer can configure as much as 80 percent of the system
configuration, covering all the parameters necessary for the functioning of the system. For advanced
configuration of features and facilities, the Installer/System Engineer must use the “Using Configuration”
mode.
Detailed information on this Wizard and instructions for using it for system configuration are given later in
this chapter. Refer the topic “Using Quick Installation Wizard - Standard PBX”.
• Quick Installation Wizard - Hotel: This is a special wizard for the Hospitality Application of the SARVAM
UCS. The Hotel Installation Wizard simplifies the configuration process and helps the Installer/System
Engineer to configure the system for the special telephone and guest/patient management facilities and
features for hotels and hospitals.
You may use this Wizard if you have deployed the SARVAM UCS as a Hospitality Application. The Hotel
Installation Wizard is recommended to be used when configuring the SARVAM UCS for the first time.
Refer the separate SARVAM UCS Hospitality System Manual to know more.
Configuration
While the Quick Installation Wizard provides a fast-track way for system configuration, detailed and advanced
configuration of the system can be done only from the links under Configuration of Jeeves.

The Configuration, as the title itself suggests, enables the Installer to change the values of all configurable
parameters of the system, including those not covered by the Wizard.
Refer the topic “Using Configuration” for more information and instructions.
As many as four System Engineers can simultaneously login into SE Mode and configure the system via
the Configuration and the Quick Installation Wizard-Standard PBX. However, it is recommended that
multiple login be avoided when using the Quick Installation Wizard and the use of the Wizard be restricted
to a single person only.
2. System Administrator Mode

At the System Administrator level, the settings of the features for the Extension users can be changed, and
various system activity logs and reports such as fault log reports, Station Message Detail Recording reports,
and Hotel reports, can be captured and printed.
For this, the System Administrator, who is usually the operator or receptionist, must login as System
Administrator into Jeeves.
• To login into the SA mode, on the login page in Login as select System Administrator.
• In Password, enter the default SA password, 1111.
• Click Login.

• You are prompted to change the default password.
• In Current Password, enter the default SA password.
• Enter the New Password. The new password must be:

• Click Submit. You will be re-directed to the Login page again.
• Now, in Login as select System Administrator and in Password enter the new password.
As this password is meant for restricting access to the SA mode, we strongly recommend you to:
• Change the password regularly. See “System Security”
Four persons can simultaneously login and change system settings from the SA mode.
3. Front Desk User Mode

The Front Desk Mode is relevant only for the Hospitality Application of the SARVAM UCS.
This mode is meant for the personnel at the Front Desk of the Hotels/Motels, allowing them access to and
operation of the hospitality features of SARVAM UCS, for example: Check-In/Out of guests, Changing Room
Occupancy and Clean Status, setting Call Budgets for guests, setting Wake-up calls, Reminders, Do Not
Disturb for guests, printing Call Reports and Hotel reports, and several others.
The Front Desk User mode too is password protected.

Refer the SARVAM UCS Hospitality System Manual to know more.
• If you select the language on the 'Welcome' page or on any of the Login pages, it is valid for the
session only. The default language will be applied on next login.
• When you select the “Configuring ‘Region’” for the country in which SARVAM UCS is being installed,
the system will load the country-specific default settings and automatically select the local language of
the country, if the local language is among the above listed languages. The default local language set
on selecting the Region will be applied for every login session, unless you select another language as
the default local language.
• The default local language set on selecting the Region can also be changed from the “System
Parameters” page of Jeeves.
The system can be configured both, 'on site' and from a remote location (refer “Remote Programming”), and from
multiple extensions simultaneously, without its functioning being affected.

Configuring using a Telephone
The SARVAM UCS can be configured by dialing the relevant command strings from a telephone. The telephone
may be any Single Line Telephone (SLT), the Matrix proprietary Digital Key Phone (DKP), EON, or the Matrix
Extended IP Phone connected as an extension of the SARVAM UCS.
For the ease of operation, you may use EON. Using EON has the advantage as,
• you can view the command strings that you have keyed in on the LCD display of EON.
• you will get prompts and confirmatory messages on the phone's LCD display, in addition to confirmation
tone played to you.
• you can dial alphanumeric command strings.
All these facilities will not be available to you on an SLT.
System Configuration using a Telephone can be done at two levels: System Engineer Level and System
Administrator Level.
A distinct set of features and facilities can programmed at each of these levels.
It is possible to configure the SARVAM UCS from any location using a Telephone. You can use a
Telephone connected as an extension of the SARVAM UCS to configure the system On-site (where it is
installed). You can also configure the system using a Telephone Off-site, from a Remote location, using
the “Direct Inward System Access (DISA)” feature of the SARVAM UCS. You can access both the System
Engineer mode as well as the System Administrator mode from the remote location. Refer the topic
'Remote Configuration using a Telephone' later in this chapter for instructions.
1. System Engineer Mode

At the System Engineer level, the entire system configuration, all programmable features and facilities of the
SARVAM UCS can be configured by dialing command strings referred to in this document as SE Commands.
The System Engineer has Full Programming Access when configuring the system using a Telephone. In other
words, the System Engineer can configure the entire system, nearly all the programmable parameters, using a
telephone.
Refer the topic “Using Configuration” for more information and instructions.
SE Commands
These are number strings to be dialed by the System Engineer on entering the SE mode via a telephone. SE
Commands are unique to the feature/facility being programmed. Hence SE Commands for configuring a particular
feature/facility are presented in description of that feature/facility in this manual.
For a complete List of all SE Commands, refer “System Commands”.

SE Password for programming from extensions
The SE password is a code used to restrict unauthorized access to the SE Mode. The password can be a minimum
of 4 digits to a maximum of 12 digits. The valid digits are from 0 to 9. The default SE Password is 1234. To avoid
unauthorized access, we recommend you to change the password. Make sure it is strong and is kept confidential.
Refer the topic “System Security” for instructions on how to change the SE password. In case the System Engineer
forgets the password, it can be restored to the default password and changed again. Read the section “Default
Settings” for instructions on restoring the default SE password.
There is no restriction on the number of persons who can simultaneously enter SE mode from a Telephone
and configure the system.
Entering the SE mode using a Telephone

The telephone that you use to enter the SE programming mode may be a DKP or an SLT connected as extension
phone of the SARVAM UCS.
To enter the SE mode via a DKP/SLT (for all countries except New Zealand):
• Dial 1#91-SE Password
• You get programming tone to indicate entry into the SE mode.
• Now, dial the desired SE Command to configure.
• You get confirmation tone and message on the LCD if using EON.
To enter the SE mode via a DKP/SLT (for New Zealand):

• Dial *1#91-SE Password
• You get programming tone.
• Dial the desired SE Command to configure.
• You get confirmation tone and message on the LCD if using EON.
To exit the SE mode:

• Dial 00.
• You will hear the dial tone of the SARVAM UCS.
• If the SE Password you entered is incorrect, you will be played an Error Tone and an Error Message on
EON.
• The system accepts and executes the command immediately, but it takes approximately 2 minutes to
save a command. So, it is advisable that you do not turn OFF the system for 2 to 3 minutes after
entering the last command.
2. System Administrator Mode

At the System Administrator level, you can set/cancel features settings for extensions, capture and print various
system activity logs and reports.
The Operator or Receptionist, who usually administer the system, must enter the System Administrator (SA) mode
and then after dial command strings referred to as SA Commands from a telephone.

SA Commands
SA commands consist of a prefix string 1072, followed by the Command string. For example: the SA command for
setting Do Not Disturb for an extension is 1072-001-extension number-1, where 1072 is the prefix string and 001
is the command string.
The Prefix string in the SA Command (1072) can be changed by the installer/System Engineer. However, the
command strings of the SA Command (001 in the above example) cannot be changed.
The command for entering SA mode is also non-programmable. Refer “Access Codes” in the chapter Features and
Facilities.
To know how to use or change feature settings with SA Commands, please refer the description of individual
features under “Features and Facilities”.
For a complete list of SA Commands, refer “SA Commands”.
SA Password for programming from extensions

The access to SA mode may be protected by means of an SA password.
The SA password is code for preventing unauthorized access to the SA mode. The password can be a minimum of
4 digits to a maximum of 12 digits. The valid digits are from 0 to 9. The default, SA Password is 1111. To avoid
unauthorized access, we recommend you to change the password. Make sure it is strong and is kept confidential. It
can be changed and reset by the System Engineer.
Refer the topic “System Security” for instructions on how to change and reset the SA Password using SE
commands.
• When the SARVAM UCS is used in the Standard Application, you can enter SA mode only from
extensions which have the features 'SA Mode' and/or 'SA Extension’ enabled in their Class of Service.
• When the feature ‘SA Extension’ is enabled in the Class of Service of an extension, the extension will
always be in SA mode. You do not need to enter SA mode by dialing the SA password. You can enter
the SA mode by dialing the SA command prefix string.
• When the feature 'SA Mode' is enabled in the Class of Service of an extension, dialing of the SA
Password is required to enter the SA mode. SA Commands can be dialed only after successfully
entering the SA Mode.
• There is no restriction on the number of persons who can simultaneously enter and operate from the
SA mode using a telephone.
Entering SA Mode using a Telephone

You may use a DKP or an SLT extension of the SARVAM UCS to enter the SA Mode.
To enter SA mode via an extension phone (for all countries except New Zealand),
• Dial 1#92-SA Password95
• You get a confirmation Tone of the SARVAM UCS to indicate successful entry into the SA mode.
• Dial SA Command strings: 1072-Feature Access Code.
• You get a confirmatory tone and text message on the phone display (if using EON).

• Replace handset to exit SA mode.
OR
• Dial 1#92
To enter SA mode from an extension phone (for New Zealand),

• Dial *1#92-SA Password96
• You get a confirmation Tone.
• Dial SA Command strings: 1072-Feature Access Code.
• You get a confirmation tone and text message on the phone display (if using EON).
• Replace handset to exit SA mode.
OR
• Dial *1#92
You can exit from the SA mode automatically or manually. To exit the SA mode automatically, you must
configure the SA Mode Timer. On the expiry of the set time, the system disconnects the extension phone
from the SA mode. This Timer is loaded automatically every time a new SA command is issued.
To program the SA Mode Timer:
• Dial 1072-016-SA Mode Timer

Where,
SA Mode Timer = 000 to 255 minutes.
• You can also exit the SA mode before the SA Mode Timer expires by dialing this command. If the SA
Mode Timer is set to 000 minutes, you can exit the SA mode only by dialing the command.
You may change the SA Mode Timer from the SA mode of Jeeves also.
Changing Login Session Time Out of Jeeves

As mentioned earlier, each login session of Jeeves has been set to 60 minutes by default. The Login session will
expire at the end of 60 minutes. The duration of the Login session can be changed as per your preference by
changing the settings of the 'Web Configuration Time Out Timer'.
To do this:
• Dial 1#91-SE Password to enter SE mode from a DKP/SLT.
• Dial 2118-Time
Where,
Time is from 001 to 255 minutes.
For example, to set log out time to 45 minutes, dial 2118-045.
• Press 'Enter' key to save setting, if using EON.
95. If the password is less than 12 digits, you must dial #* after the new password to indicate end of dialing. When you dial this com-
mand, the system will check if the facility 'SA Mode' is enabled in the Class of Service allowed to the extension from which you
have dialed this command. If the SA mode is not allowed, an Error Tone will be played. The Error Tone will be played also when
the SA password is entered incorrectly.
If the facility 'SA Extension' is enabled in the Class of Service of the extension you are using, you can skip this step and directly dial
the SA Command (1072-<Command String>).
96. When you dial this command, the system will check if the facility 'SA Mode' is enabled in the Class of Service allowed to the exten-
sion from which you have dialed this command. If the SA mode is not allowed, an Error Tone will be played. The Error Tone will be
played also when the SA password is entered incorrectly.
If the facility 'SA Extension' is enabled in the Class of Service of the extension you are using, you can skip this step and directly dial
the SA Command (1072-<Command String>).

Logging Out Users from Jeeves
It is possible for four users to simultaneously log in as System Engineers and use Jeeves. It is also possible to log
out all these users at once. To do this,
• Dial 1#91-SE Password to enter SE mode from a DKP/SLT.

• Dial 2188 to log out all users.
• Press 'Enter' key to save setting, if using EON.

Using Quick Installation Wizard - Standard PBX
The Quick Installation Wizard-Standard PBX helps you with the basic configuration of the system in easy steps. It is
designed to break down the complexities of programming and can cover as much as 80% of your basic installation
requirements.
The advantage of using the Wizard is that it speeds up system configuration, as you configure parameters that are
specific to the SARVAM UCS and show only those Trunks and Extensions that are connected to the system for
configuration. For instance, if you are installing only 12 out of the 64 CO trunks supported by SARVAM UCS,12
DKP extensions and 240 SLT extensions, the Wizard will show only these many trunks and extension ports for
configuration. Thus, the Wizard makes the entire system configuration very focused.
Further, the Wizard can be used for system configuration for first time installation as well as any time post-
installation to make modifications in the system configuration.
Structure of the Wizard

The Wizard stores its own set of configuration files and once the installation is completed, it updates the system
configuration files with the data you entered. However, the reverse does not work. The changes you make in the
system configurations under Configuration will not be reflected in the Quick Installation Wizard.
The Quick Installation Wizard - Standard PBX offers the configuration of the following parameters:
1. Region
2. Pre-requisites
3. Extension Numbers in Range
4. Extension Numbering Plan
5. Trunks
6. Day-Night time
7. Number Patterns
8. Operator
9. Extensions
10. Least Cost Routing (LCR)
11. Call Pickup Group
12. CO Trunks
13. BRI Trunks
14. T1E1 PRI Trunks
15. Mobile Trunks
16. VoIP Network
17. SIP Trunks
18. Emergency Numbers
Help Text
There is a help text to explain the parameters on each screen. You may use this help text to guide you in entering
the information.

The Buttons
The Wizard has the following buttons:
• Next: Clicking this button will cause the existing values of the parameters in a page to be submitted and
takes you to the next page of the Wizard.
• Submit: Clicking this button will cause the parameters configured in the page to be submitted, but will not
take you to the next page. The 'Submit' button is to be used when there are multiple pages for configuring
a facility, for example, configuring extensions, CO trunks. Instead of navigating all the pages, you can
access the desired page, make changes and submit them. Similarly, if you want to make any changes
post-installation, instead of navigating through all pages of the Wizard and clicking Done to effect the
changes, you can reach the desired page, make the necessary changes and submit the changes.
To navigate further, you must click the Next button.
• Skip: Clicking this button will take you to the next page, without making any changes to the current page.
The Skip button is to be used when the Wizard is used post-installation to reach a page which is to be
modified, without changing the existing configuration settings on other pages.
• Undo: Clicking this button refreshes the page with the parameters configured in the system. You may use
this button if you are not sure about the values you entered or have entered incorrect values and wish to
start all over again. This button is available only on select pages of the Wizard.
• Help: Clicking this button on a page opens a new window, containing Help Text for that page. The window
can be maximized.
• Default: Clicking this button will populate all the fields of the page with the default values. Use this button
when you want to assign default values to parameters.
• Clear: Clicking this button will cause the values of all the fields on the page to be cleared. Use this button
to make corrections or start entering the values all over again.
• Exit: Clicking this button will exit the Jeeves amidst of an activity.
The changes you make in the system configuration under Configuration will not be reflected in the Quick
Installation Wizard.
You are advised to either use the Wizard only during installation and to make modifications post
installation OR use the Wizard only for basic configuration (first time installation) and for making
subsequent changes click the desired link under Configuration.
Entering the Wizard

• Open Jeeves (see “Configuring SARVAM UCS” for instructions). The Login page will open.

• In Login as select System Engineer.
• In Password enter the SE password.
• Click Login.
• On successful login, the Home page opens.

• Click the Wizard icon on top right corner of the browser.
• To configure the Basic Settings, select the Use Quick Installation Wizard-Standard PBX.
• The Quick Installation Wizard opens.
Jeeves allows 4 persons to simultaneously log in as System Engineer. When you are logged into the
Wizard, it is recommended that no other person logs into Jeeves as System Engineer.

Configuring Parameters
The Wizard gives you two options for configuring the system:
• Selective Configuration - This option allows you to choose what you want to configure and click the
related links on the left navigation bar of the Wizard page. You are recommended to use this option for
making the desired changes post-installation.
• Navigate Wizard - In this option, the Wizard will lead you logically, step-by-step through the configuration
of the desired parameters. Click the Next button to navigate through the Wizard. You are recommended to
use this option when installing the system for the first time.
Region
• In Select the Region, select the country where the system is installed, and then click Next.
Selecting the Region will instruct the system to load the default values of features/facilities according to
the country specific requirements.
The default Region is India.
System Pre-requisites

• Enter the “Customer Name”. You may enter the name (and address, if desired) of the organization/
enterprise. For example: Prudent Investment, 701 Sunshine Boulevard, Bannerghatta, Bangalore. The
Customer Name can be a maximum of 80 characters.
• If you enable the On Site configuration check box, the configuration GUI of SARVAM UCS, Jeeves, will
show the pages for only those trunks and extension port types that are on board in the system, that is,
detected by the system at Power-On.
• The Model Type displays the name of the model that you are configuring — SARVAM UCS SME or
SARVAM UCS ENT.
• Select the number of ports to be used for each Port Type (CO, DKP, SLT, Mobile, T1E1 PRI, BRI, SIP
Trunks, SIP Extensions) from the respective boxes. For example, if you want 8 CO Trunks, 24 DKP
extensions, and 128 SLT extensions to be used, select the same numbers from the box.
• Click Submit to save the changes.
If you have selected the System Pre-requisites, navigate to the next page of the Wizard by clicking the
Next button.
• You can set the Wizard to display only those trunk and extension port types which are present in the
system. This can be done by enabling the 'On Site Configuration' check box.
• When the On Site Configuration check box is enabled, the Wizard will display the only those ports
that are present in the system (detected by the system at Power-ON) on this page.
For example, the system detects SLT20 Card at power-on, so the maximum number of SLT ports in the
box will be 20. If you want only 18 SLTs to be used, select 18 in the box.
The Wizard will now consider that there are only 18 SLTs in the system and will modify the relevant
pages accordingly.
The fields for port types which are not available on-board (detected at Power-On) are displayed as non-
editable.
• To enable the On site Configuration check box you must login as System Engineer.
• The Wizard does not provide for the port types Magneto, Radio, Data, ILC and E&M as these are less
commonly used. If your system has Magneto Card and/or E&M Card installed, you must configure the
related trunk parameters under Configuration.
It is recommended that you enable the On Site Configuration check box when you are configuring the
system at the installation site.
Extension Numbers in Range

You can assign Access Codes to a range of Extensions — SLT, DKP, ISDN, SIP. If you are assigning a range of
Extension Numbers (Access Codes) to the desired ports, and a match is found for the same extension numbers,
the system will clear these extension numbers from the existing database. The new extension numbers will be
assigned according to the given range.

• Against each Index configure the following:
• Extension Type: Select the Extension Type. You can select SLT,DKP, SIP, ISDN Terminal.
• Start S/W Port Number: Enter the Software Port Number from which you want the system to start
assigning the desired extension numbers.
• Define the range of Station Access Codes/Extension Numbers (Flexible Number/Access Code) that
you wish to assign to the Extension Type you selected in Start Extension Number and End
Extension Number.
For the given range of extension numbers, the system will assign extension numbers from the software
port number specified in Start S/W Port Number for this particular entry. The range of extension
numbers will be assigned in ascending order of the Software Port Number.
For example:
Extension Type you selected is SLT
Start Software Port Number is 1
Start Extension Number is 2001
End Extension Number is 2100
The system will assign Extension Number 2001 to Software Port Number 1, 2002 to Software Port
Number 2 and so on. The system will assign the last Extension Number 2100 to Software Port Number
100.
If you want to assign access codes starting with # or * to a range of extensions, make sure both Start and
End extension numbers begin with # or *.
• Click Submit.

• If you do not want to assign access codes to range of extensions, click Skip.
• To assign the extension numbers to a range all over again, click the Clear Extn. No. button on this page.
Extension Numbering Plan

• Click the respective tab to assign extension numbers and names for the SLT, DKP and SIP Extension as
desired on this page.
You can also configure Access Codes from this page.
The desired Extension Number can be 6 digits long. The digits 0 to 9, # and * are allowed.
The desired Extension Name may be the name of the person who will use the extension. The name can be
a maximum of 18 characters.
When you change the extension numbers, make sure that they do not clash with any other Feature Access
Code in the dialing phase. To know more, refer the topic “Access Codes”.
The Wizard pops up an alert about clashing extension numbers and prompts you to resolve the conflicting
numbers.
• If you click the SIP tab, in Authentication Password, enter the password to be used by the system to
authenticate the SIP messages received from the SIP Extension. To avoid unauthorized access, we
recommend you to change the password regularly. Make sure it is strong and is kept confidential. The
password must:
• be of minimum 6 characters and can be a maximum of 12 characters.

Default: Blank.
The Wizard will display only those ports available in the system and the number of ports you have defined
earlier in the System Pre-requisites page. The Wizard automatically detects the Hardware Slot and Port
Offset of the ports and assigns them to Software Ports. The Wizard also assigns the default extension
numbers to the ports, but leaves the extension names blank.
The default extension numbers for the above port types are:
2001 to 2240 for SLTs 001 to 240.
3001 to 3128 for DKPs 001 to 96.
Blank for SIP Extensions 001 to 999.
Blank for ISDN Terminals 01 to 64.
The Hardware Slot and Port Offset is not applicable for SIP Trunks.
• If you have finished assigning numbers and names, click Submit.
• To assign the extension numbers and names all over again, click the Clear button on this page.
• To default all extension numbers, click the Default button on this page.
Access Codes
• Change the Feature Access Codes, if required. Feature Access Codes may consist of single digits or a
sequence of a maximum of 6 digits. Digits 0 to 9,# and * are allowed. The default Access Codes that
appear on the screen are country-dependant. Also, refer “Access Codes” to know more.
• Click Submit and click Next to move to the next page.
If you assign the same Access Code to more than one feature, the Wizard will pop up a Total Conflict
message and ask you to resolve the conflicting codes. It will not allow you to submit until you have
resolved the conflict. If you assign an Access Code which has the same first digit as another Access Code,
the Wizard will pop up an alert about the clashing numbers. You must resolve the clashing numbers by
clicking OK.
You may choose to resolve the clashing number by clicking ‘Cancel’ or you may ignore the alert and
continue programming by clicking ‘OK’.
Trunks
• Configure names for Trunk ports for easy identification of the trunks. The name may consist of a maximum
of 18 alphanumeric characters.
The Wizard automatically detects the Hardware Slot and Port Offset of the trunk ports and assigns them to
Software Ports. The Wizard also assigns the default trunk port names along with their respective software
port numbers. For example, if a Two-wire Trunk is assigned software port number 001; the name will be
displayed as CO -001. Thus CO trunks are named as CO-001 to CO-128, Mobile Ports as MOB-001 to
064, BRI Ports as BRI-001 to BRI-032, and so on.

The Wizard will display only those Trunk port types available in the system and the number of Trunk ports
you have defined for each trunk port type earlier on the System Pre-requisites page.
The Hardware Slot and Port Offset is not applicable for SIP Trunks.
• Click Submit and click Next to navigate the page further.
Day-Night Time
• Ask your customer about their working days and working hours (24 hours format) and program the
parameters accordingly. The Wizard considers the working hours you have selected as Daytime and the
remaining hours as Night Time (non-working hours). The default working hours are from 09:00 to 18:00.
The default Working Days are Monday to Saturday.

• Day-Night Time is assigned to Trunks and Extensions, so that they behave differently according to the
Time of the day. For example, the customer may want the system to route trunk calls to the security
personnel when the office is closed, or deny certain extensions access to outgoing long distance calls
during non-working hours and days, or to play a different greeting message to callers on holidays.
• This parameter is based on Time Table 197, which is assigned to Trunks and Extensions by default.
• The Wizard simplifies the assignment of Time Tables to trunks and extensions, requiring you to
configure only the working hours and working days, instead of prompting you to define the non-working
hours and break hours. The Wizard automatically applies the working hours and days you have
programmed to time table-dependent facilities and features such as Class of Service, Toll Control,
Outgoing Trunk Access, etc.
• Skip this page if you feel that the requirements of your customer are not served by this parameter98.
Configure the Time Tables and related features like Class of Service, Toll Control, Outgoing Trunk
Access, etc. under Configuration.
• If you want to change the working hours and days at a later stage while navigating the Wizard, you may
use the Back button of your browser to return to this page and make the changes.
• Click the Next button to navigate the Wizard further.
97. A Time Table is a schedule of the three time zones (working hours, break hours, non-working hours) for a week. There are 8 differ-
ent Time Table templates to select from. Different Time Tables can be assigned to different trunks and extensions. Refer the sec-
tion “Time Tables” to know more.
98. For instance, the working hours are not the same throughout the week.

Number Patterns
This parameter is related to the “Toll Control” feature, which allows you to define a particular calling permission for
each extension, referred to as Call Privilege. A Call Privilege allows the extension to call certain areas and restricts
it from calling others. The extension can also be restricted from the dialing of specific telephone numbers. The
SARVAM UCS supports different types of Call Privileges, these are: No Calls, All Calls, Local Calls, Regional Calls,
National Calls and International Calls.
• On this page, you are required to define the number strings which the system should consider as Local
Numbers, Regional Numbers, National Numbers and International Numbers.
In Numbers starting with, you may enter only the first digit of the number string, or a part of the string, or
the complete number string.
In except, enter the number strings which you want to restrict from being dialed out.
Each number string you enter must not exceed 16 characters. Separate number strings with comma.
Do not provide <space> between commas and numbers.
• Click the Next button to navigate the Wizard further.
• The Call Privilege Type No Calls and All Calls does not require any number pattern programming.
• The Call Privilege Type Limited Calls allows the dialing of only specific telephone numbers. It can be
programmed only under Configuration. To know more, refer the feature description for “Toll Control”.
Operator
• Select the extensions which are to be used as Operator.

When you double click the Operator field, a multiple selection box opens.
The box on the left displays the extension names and numbers you configured or the default access
codes. Place your cursor on the desired extension and click the Select>> button. The selected extensions
will appear in the box on the right.
It is possible to change the sequence of extensions on the right side box using the Up/Down Arrow.

To remove/delete any extension number from the right box, select the extension number and press the
delete key from the keyboard.
A maximum of 16 extensions can be selected as Operator extension.
You will be shown an alert if you configure more than 16 extensions. You may select and delete the excess
extensions or choose to ignore the alert by clicking 'OK' or closing the alert dialog box. Regardless of this,
the system will consider only 16 extensions. You can change the sequence of the extensions on the right
hand side using the Up and Down arrows.
When you click OK, all the extensions you selected will appear sequentially, separated by comma in the
order you selected the extensions.
• Define the Class of Service for the Operator extension. Select the features to be allowed to the Operator
extension during the day and at night by selecting the check boxes of the features listed in the table.
• Configure the Toll Control for the Operator extension for day time and night time. Select the type of calls to
be allowed during the day, and the type of calls to be allowed during the night. By default, all types of calls
are allowed during the day and at night.
• Select the Outgoing Trunks for the Day Time (the trunks through which calls are to be routed during the
day). If you want the system to use Least Cost Routing, select the check box.
When you double click this field, the multiple selection box opens. Select the outgoing trunks from the left
side box.
As you can see, the same trunk types are arranged sequentially, regardless of their hardware port location.
If you select trunks of the same type in sequential order, for example: CO-001, CO-002, CO-003, BRI-001,
BRI-002 and MOB-002, the same trunk type will be grouped in one OG Trunk Bundle: CO-001, CO-002,
CO-003 will be OGTB #1, BRI-001 and BRI-002 will be grouped as OGTB#2, and MOB-002 as OGTB#3.
These are default trunk names. If you have changed the trunk names (see Naming of Trunks), the name
will appear here, instead of the default trunk names.
If you select the trunks of the same type in a non-sequential order, for example, CO-001,
MOB-001, BRI-001 and CO-002, four OGTBs will be formed with CO-001 as member of OGTB#1, MOB-
001 as member of OGTB#2, BRI-001 in OGTB#3 and CO-002 in OGTB#4. So, despite two same trunk
types being selected (CO-001 and CO-002) they are grouped in separate OGTBs.

A maximum of 8 OGTB are allowed. If you exceed the number, the Wizard will show an alert indicating that
the system is short of resources.
It is possible to change the sequence of the trunks on the right side box using the Up/Down Arrow.
To remove/delete any trunk number from the right box, select the trunk number and press the delete key
from the keyboard.
• Select the Outgoing Trunks for the Night Time (the trunks through which calls are to be routed during the
night). If you want the system to use Least Cost Routing, select the check box.
Follow the same instructions for selecting the trunks from the multiple selection box as described in the
previous step.

• Set the “Priority”99 for the Operator extension, by selecting the desired Priority Level from 1-9 from the box.
By default the priority level for the Operator extension is set to level 5.
• Click Next to navigate to the next page.
For the parameters of the Operator extension, the Wizard uses the following resources:
• Class of Service Groups 18 to 19.
• Outgoing Trunk Bundle Groups 18 and 19
• Outgoing Trunk Bundles 69 to 76.
When configuring the system from the links under Configuration, do not modify the settings of these
parameters. It will affect the settings made by the Wizard.
Extensions
• Create a User Profile for extensions. A User Profile consists of Class of Service (COS), Toll Control and
Trunk Access to be assigned to an extension during day time and night time.
99. Each extension of the SARVAM UCS is assigned a Priority Level starting from 1, 2, 3, 4...to 9. With 1 being the lowest priority and
9 being the highest priority. The calls from an extension with higher priority have preference in call landing. When an extension with
higher priority calls another with lower priority, a triple ring is placed on the called extension, and the call will land first on the exten-
sion when there are multiple incoming calls on the extension with lower priority.

The Wizard makes configuration of the extensions easy with User Profiles. Instead of configuring each
extension individually, you can group together extensions that are to be allowed the same COS group, Toll
Control, Trunk Access (Outgoing Trunk Bundle and Outgoing Trunk Bundle Group) and Priority in a single
User Profile.
User Profiles meet the requirements of organizations that desire to assign a different set of features to their
personnel according to their position in the organization, like senior managers, field executives,
administrative assistants etc. In such cases, each of these groups of users can be assigned a different
User Profile.
As many as 8 different User Profiles can be programmed using the Wizard.
You can name each User Profile such that it reflects the extension user group to which it is assigned. For
example, you may rename User Profile-1 created for managers as 'Manager', User Profile-2 created for
field executives as 'Executive', User Profile-3 created for administrative assistants as 'Admin'.
• To change the label of the User Profile, click the 'Pencil' icon.
• A prompt will pop up, asking you if you want to rename the User Profile Name. Enter the desired name in
the field. You can enter a maximum of 18 characters as name. Click 'OK'.

• The name you entered will appear in place of 'User Profile' number.
• To assign the User Profile to the desired extensions, double click the box.
The Wizard will display the configured extension numbers on the left box. Place your cursor on the desired
extension numbers and click Select >>. The selected extension numbers will appear on the right box.
You can select the extensions for the User Profile after configuring the other parameters of the User
Profile.
• Define the Class of Service for the User Profile for day time and night time by selecting the check boxes of
the features you want to allow in the Class of Service.
• Define the Toll Control for the User Profile for the day time and night time by selecting the desired Toll
Control level in the box.
The Toll Control levels on this page are based on the number lists you have programmed earlier on the
Number Patterns page of the Wizard.
• Select the Outgoing Trunks for the Day Time (the trunks through which calls are to be routed during the
day). If you want to the system to use Least Cost Routing, select the check box.
When you double click the field, the multiple selection box opens. Select the outgoing trunks from the left
box.
As you can see, the same trunk types are arranged sequentially, regardless of their hardware port location.
If you select trunks of the same type in sequential order, for example, CO-001, CO-002, CO-003, BRI-001,
BRI-002 and MOB-002, the same trunk type will be grouped in one OG Trunk Bundle: CO-001, CO-002,
CO-003 will be OGTB #1, BRI-001 and BRI-002 will be grouped as OGTB#2, and MOB-002 as OGTB#3.
If you select the trunks of the same type in a non-sequential order, such as: CO-001, MOB-001, BRI-001
and CO-002, four OGTBs will be formed with CO-001 as member of OGTB#1, MOB-001 as member of
OGTB#2, BRI-001 in OGTB#3 and CO-002 in OGTB#4. So, despite two same trunk types being selected
(CO001 and CO002) they are grouped in separate OGTBs.

It is possible to change the sequence of trunks on the right side box using the Up/Down Arrow. You can
also delete a particular trunk from the right box.
• Select the Outgoing Trunks for the Night Time (the trunks through which calls are to be routed during the
night). If you want to the system to use Least Cost Routing, select the check box.
Follow the same instructions for selecting the trunks from the box as described in the previous step.
• Set the “Priority”100 for the extension, by selecting the desired Priority Level from 1-9 in the box. By default
the priority level for the extension is set to level 5.
You may set a different Priority Level in each User Profile, depending on the requirement of the extension
users, whose extension the User Profile is to be assigned. For example, you may set a higher Priority level
to the User Profile to be assigned to Managers.
• Click Submit to save the changes in the User Profile.
• To program another User Profile, click the label.
• Rename the label, if required, by clicking the 'Pencil' icon. Repeat all the steps described above to define
Class of Service, Toll Control, Trunk Access and Priority.
• Click Submit to save changes. Repeat the same steps to program another User Profile.
• When you have finished programming the desired number of User Profiles, click Next button to navigate
the Wizard further.
• The User Profiles 1-8 use the following resources in the system:
• Station Basic Feature Template number 2 to 9
• Outgoing Trunk Bundle Groups 2 to 17
• Do not use them when configuring the system from Configuration.
100. Each extension of the SARVAM UCS is assigned a Priority Level starting from 1, 2, 3, 4...to 9. With 1 being the lowest priority and
9 being the highest priority. The calls from an extension with higher priority has preference in call landing. When an extension with
higher priority calls another with lower priority, a triple ring is placed on the called extension, and the call will land first on the exten-
sion when there are multiple incoming calls on the extension with lower priority. Refer the feature description for “Priority” to know
more.

Least Cost Routing (LCR)
• Assign Cost Factor to the trunks. This parameter is of relevance only if 'Least Cost Routing' feature is
applied on the CO Trunk port.
• On this page, the Wizard displays the number of trunk ports you selected in the System Pre-Requisites
page.
• Cost Factor is a number assigned to each trunk for identification. This number also serves as a preference
number for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same preference must be
assigned the same Cost Factor. Different trunk types can also be assigned the same Cost Factor. These
trunks are used for routing calls.
• Now configure the number of or part of the number and the preferred trunk to route that number. The
number can be a maximum of 16 digits.
• Select the cost factor trunk applicable for each number in the order of preference. Select the foremost
preferred Cost Factor trunk in Preference 1, the second most preferred Cost Factor trunk in Preference 2,
the third preferred Cost Factor trunk in Preference 3 and the least preferred Cost Factor trunk in
Preference 4.
• Click Next to navigate the Wizard further.

SARVAM UCS supports four different types of LCR, but the Wizard supports only Number based LCR. To
know more, refer “Configuring LCR”.
Call Pickup Group

• Assign extensions to Call Pick-Up (CPU) groups.
Call Pick-Up allows extension users to answer (internal and trunk) calls ringing on other extensions from
their own extension; without physically going to the ringing extensions. For this extensions must be
assigned to CPU Groups.
• While you can create as many as 01 to 99 CPU Groups, and assign extensions to these groups, the
Wizard allows you to create and assign extensions to 01 to 16 CPU Groups. By default, all extensions are
assigned to CPU Group number 99.
• To create a CPU group, click the CPU group number, for example CPU 01, and double click to select the
extensions to be assigned to this group from the box.
• Click Submit to save the settings of the CPU group you created. Now, repeat the same steps to create
another group.
• If you have finished programming CPU groups, click Next to navigate to the next page of the Wizard.
To know more about this feature, refer the topic “Call Pick Up”.

CO Trunks
• Configure features of the Two-wire Trunks (CO) connected to the SARVAM UCS on this page.
The Wizard makes configuration of the CO Trunks easy with CO Profiles. Instead of configuring each CO
Trunk individually, you can group together trunks that are to be assigned the same features - Calling Line
Identification, Trunk Landing Group - in a single CO Profile.
You can configure as many as 8 different CO Profiles using the Wizard.
It is also possible to name each CO Profile by the service provider. For example, you may rename CO
Profile-1 created as 'BSNL', CO Profile-2 as 'Reliance' and so on.

• To change the label of the CO Profile, click the Pencil icon.
• A prompt will pop up, asking you if you want to rename the CO Profile Name. Enter the desired name in
the field. You can enter a maximum of 18 characters as name. Click OK.
• The name you entered will appear in place of CO Profile number.

• To assign the CO Profile to the desired trunks, double click the field Select Trunks/Apply to Trunks.
The left side box will display the number of CO trunks (along their names) you configured in the System
Pre-requisites page. Place your cursor on the desired trunks and click Select >>. The selected CO trunks
will appear on the right side box.
You can select the trunks for the CO Profile after configuring the other parameters of the CO Profile.
• Define the Calling Line Identification Format for the CO Profile by selecting the desired option in the box.
You are advised to consult the service provider in this regard.
• In the field Route Calls during day to, select the landing destination for calls on trunks in the CO Profile
during day time. You may select any option: Operator extension, other extensions, Built-In Auto Attendant
or the Voice Mail Auto Attendant (if available) from the box.
• If you select Built-In Auto Attendant, the default Voice Modules 02 to 13 containing the default Voice
Messages (Morning, Afternoon and Evening Greetings, Built-In Auto Attendant Greeting and Guidance
Messages) will be applied. Refer “Voice Message Applications” to know more.
• If you select Voice Mail Auto Attendant, the voice messages recorded and stored in Voice Mail module
will be played.
• If you select Extension/s as the landing destination, select the extensions in the corresponding field.
Double click the field, the multiple selection box will open. Select the extensions from the left box.
A maximum of 16 extensions can be selected. If you exceed the number, the Wizard will prompt you to
make the selection again. You can delete the excess extensions from those you selected. You can also
change the order of the selected landing extensions in the right side box using the Up/Down Arrow.
The selected extension numbers will appear in the field, separated by comma.
• In the field Route Calls during Night to, select the landing destination for calls on trunks in the CO Profile
during night time. Follow the same instruction as in the previous step.
• Click Submit to save your settings in the CO Profile. Repeat these steps to create another CO Profile.
• If you have finished creating CO profiles and assigning them to trunks, click Next to navigate further.
• If there is any trunk you have not assigned a CO Profile, the Wizard will pop-up an alert informing you
about the trunk you have not programmed.

You must complete configuring the parameters for this CO trunk and only then will the wizard allow you to
configure other parameters.
The Wizard uses the following resources for this page:

• Trunk Feature Templates 34 to 41.
• Routing Groups 34 to 49.
• CO Hardware Templates 02 to 09.
Do not make any modifications to them when configuring the system from Configuration.
BRI Trunks
• You will reach this page only if your system has detected the presence of BRI Trunks or if you have
specified the number of BRI Trunks to be used in the System Pre-requisites page.
• The SARVAM UCS supports a maximum of 32 BRI Trunks.
• Using the Wizard, you can configure only the first four BRI Trunks in your system. You can configure
the remaining BRI Trunks from Configuration.
• The Wizard will display the name of first BRI Trunk.
The name will be displayed only if you have programmed names in the Trunks page of the Wizard. If you
have not named the trunk on that page, you may change the label; select the desired BRI Trunk tab and
click the Pencil icon.
A prompt will pop up, asking you if you want to rename the BRI Trunk. Enter the desired name in the field.
The name you entered will appear on the screen.
If required, you may change the Hardware slot and port number of the BRI trunk displayed here.

• Select the mode for routing incoming calls. Incoming calls may be routed according to MSN Numbers, DDI
Numbers or Port-wise.
• Route Incoming Calls MSN Number wise: Select this option if you want to route incoming calls
according to MSN numbers101. You can program maximum 6 MSN numbers.
• Route Incoming Calls DDI Number wise: Select this option if you want to route incoming calls
according to DDI Numbers102. Select this option only if your extensions are arranged sequentially. If
the extensions are not arranged sequentially (for example, DDI numbers 2630551 to 2630559 are to be
routed to extensions 3001 to 3009) you are recommended to program this parameter from the Full
Programming Access mode. You can program maximum 12 DDI numbers.
• Route Incoming Calls Port wise: Select this option if you want to route all incoming calls on the BRI
trunk port to extensions, irrespective of dialed MSN/DDI number.
• If you select MSN Number-wise routing, configure the following parameters:
• MSN Number: Enter the MSN Number provided by your Service Provider. This number is used to
route the incoming calls and sent to the ISDN Exchange when making an outgoing call. You can enter
up to 6 MSN numbers using the Wizard.
• Route calls during day to: Select where you want to the route the calls during day time. You can route
the calls to the Operator or to a group of extensions or to Built-In Auto Attendant or to the Voice Mail
Auto Attendant103.
If you click the option Built-In Auto Attendant, the default Voice Modules 02 to 13 containing the
default Voice Messages (Morning, Afternoon and Evening Greetings, Built-In Auto Attendant Greeting
and Guidance Messages) will be applied by the Wizard. Refer “Voice Message Applications” to know
more.
If you select Voice Mail Auto Attendant, the voice messages recorded and stored in Voice Mail
module will be played.
If you select Extensions, double click the empty field, the multiple selection box will open. Select the
extensions from the left side box.
change the order of the selected landing extensions in the right side box using drag and drop option.
• When NR?: Select an option for what action the system should take when there is no response (NR),
that is, the incoming call is not answered within the DDI Ring Timer (default: 45 sec.) by the destination.
By default, it is set to 'Disconnect'.
• When Busy?: Select an option for what action the system should take when the landing destination is
busy. By default, it is set to 'Disconnect'.
101. MSN numbers is a set of numbers with no logical connection between the numbers themselves. For example, MSN numbers for a
BRI connection could be 2630555, 2634872, 2635098, etc. Up to 8 MSN numbers are provided on a single BRI connection.
102. DDI numbers are a set of numbers arranged sequentially, for example DDI numbers for a BRI connection could be 2630551 to
2630559.
103. The in-built Auto Attendant offers 9 simultaneous calls and Voice Mail's Auto Attendant offers 64 simultaneous calls.

• Route calls during night to: Select where you want to the route the calls during night time. You can
route the calls to the Operator or to a group of extensions or to Built-In Auto Attendant or to the Voice
Mail Auto Attendant.
If you select Extensions, you must now select the extensions by clicking the field provided for it. A
maximum of 16 extensions can be selected. Follow the same procedure described above for selection
of extensions for calls during day time.
busy. By default, it is set to 'Disconnect'.
• The Wizard allows you to select the desired landing extension (Operator/ extensions/ Built-In Auto
Attendant/Voice Mail's Auto Attendant) only for the first 3 MSN numbers. The remaining 3 MSN
numbers can be routed to a single extension only.
• So, if you want to route incoming calls to a particular extension during the day time and night time,
enter the MSN numbers in MSN number 4 to MSN number 6.
• Configure the same parameters for MSN number 2 to 6, as applicable.
• If you select DDI Number-wise routing, configure the following parameters for each DDI#:
• Root DDI #: Enter the Root DDI number. This number is used to send the DDI number to the ISDN
Exchange. This number is also sent to the Exchange when an outgoing call is made by an extension
which is not assigned DDI number. This is the same as the Pilot Number.
• Start DDI #: Enter the first DDI number provided by your Service Provider. More often than not, the
Start DDI# is the same as the Root DDI#.
• Total DDIs: Enter the total DDI numbers provided by your Service Provider.
• Start Extension#: Enter the number of the first extension from which the DDI number assignment is to
be done.
For example, your Service Provider has given you DDI numbers 2630550 to 2630560. These are to be
assigned to extension numbers 2001 to 2010. In this case 2630550 will be the Root DDI# as well as
Start DDI#. As 10 DDI numbers are provided to you, the Total DDI# would be 10, and the Start
Extension# would be 2001.
busy, during DDI Ring Timer (default: 45 sec.). By default, it is set to 'Disconnect'.
• Now, configure the same parameters for DDI#2 to #12.

• If you select BRI port-wise routing, configure the following parameters:
• Route Calls during Day to: Select the landing destination for calls on BRI trunk during day time. You
may select any option: Operator extension, other extensions, Built-In Auto Attendant or the Auto
Attendant of the Voice Mail (if available).
If you select Built-In Auto Attendant, the default Voice Modules 02 to 13 containing the default Voice
Messages) will be applied by the Wizard. Refer “Voice Message Applications” to know more.
If you select Extensions as landing destination, you must select the extensions. Double click the field
and select the extensions from the left box.
• Route Calls during Night to: Select the landing destination for calls on BRI trunk during night time.
Follow the same procedure for selecting extensions as described above.
• Click Submit to save changes. Repeat the same steps to program the other BRI Trunks.
• If you have finished configuring the BRI Trunks, click Next to navigate further.
• If there is a BRI trunk you have not configured, the Wizard will pop-up an alert informing you about the
trunk you have not programmed.
You must complete configuring the parameters for this trunk and only then will the wizard allow you to
For the BRI parameters, the Wizard uses the following resources:
• Entries 01 to 48 of the Incoming Call (IC) Reference Table.
• Entries 01 to 48 of the Outgoing Reference Table.
• Routing Groups 01 to 25
• Trunk Feature Template 01 to 25.
When programming the system from Configuration, do not modify the settings of these resources. It will
affect the settings made by the Wizard.
T1E1 PRI Trunks
• The SARVAM UCS supports a maximum of 8 T1E1PRI Trunks.
• However, using the Wizard you can configure only the first 2 PRI Trunks in the system. You may
configure the remaining Trunks, if applicable, from Configuration.

• Change the label of the T1E1PRI Trunk, if required. Click the Pencil icon. A prompt will pop-up. Enter the
desired name in the field of the prompt and click OK. The name you programmed will appear.
You may enter the name of the Service Provider to make the identification of the Trunk easy.
• The Wizard will display the Hardware Slot-Port Number of the first T1E1 PRI Trunk the system has
detected in this field.
• Select the carrier type as E1 or T1. By default E1 is selected.
• Select the Signaling Type as applicable: from PRI, RBS, QSIG, and E&M. These are the signaling types
supported by the SARVAM UCS. By default PRI selected.
• Select the Framing Mode, as applicable. By default CEPT1 MF (Auto CRC) is selected as Framing Mode.
• Select the Line Coding as applicable. By default HDB3 is selected.
• Select the ISDN Switch variant. By default ETSI NET5 is selected as the variant.
• Select the mode for routing incoming calls. Incoming calls may be routed according to DDI Numbers or
Port-wise.
• Route Incoming Calls Port wise: Select this option if you want to route all incoming calls on the
T1E1PRI trunk port to groups of extensions, without identifying the DDI number.

• Route Incoming Calls DDI Number wise: Select this option if you want to route incoming calls
according to DDI numbers. Select this option only if your extension numbers are arranged sequentially.
If the extensions are not arranged sequentially you are recommended to program this parameter from
the Full Programming Access mode.
If your installation uses multiple DDI blocks on the same T1E1PRI connection, you can configure only 2
such DDI blocks using the Wizard. You can configure more DDI blocks on a single T1E1PRI
connection from Configuration.
• If you select DDI Number-wise routing, configure the following parameters:
• Root DDI #1: enter the Root DDI # provided by your Service Provider. The Root DDI # is the main
number assigned to the T1E1PRI trunk. It is also known as MSN Number, Pilot Number or Main
Number104.
If your exchange requires area code to be sent with the DDI number, program the root number with
area code. For example: Root DDI # is 2630555 and area code where the PBX is installed is 265, enter
2652630555.
• Route calls during day to: Select where you want to the route the calls during day time. You can route
the calls to the Operator or to a group of extensions or to Built-In Auto Attendant or to the Voice Mail
Auto Attendant105.
If you select Extensions, then select the extensions by clicking the empty field. A multiple selection
box will open. Select the extensions from the left box.
change the order of the selected landing extensions in the right box using drag and drop option.
• When NR?: Select an option for what action the system should take when there is no response (the
incoming call is not answered within the DDI Ring Timer, set by default to 45 seconds) by the
destination. By default, it is set to 'Disconnect'.
• When Busy?: Select an option for what action the system should take when the landing destination
is busy (for the duration of the DDI Ring Timer; default: 45 seconds). By default, it is set to
'Disconnect'.
104. This number will be used to prepare the DDI number to the Exchange (Reverse DDI) when an Outgoing call is made from the SAR-
VAM UCS by the Extension assigned DDI Number. Also, this number will be sent to the Exchange without any modification when
an Outgoing call will be made by an extension which is not assigned DDI number.
105. The Built-In Attendant offers 5 simultaneous calls, whereas Voice Mail's Auto Attendant offers 16 simultaneous calls.

• Route calls during night to: Select where you want to the route the calls during night time from the
options: Operator or to a group of extensions or to Built-In Auto Attendant or to the Voice Mail Auto
Attendant.
If you select Extensions, you must now select the extensions by clicking the field provided for it. A
maximum of 16 extensions can be selected. Follow the same procedure described above for selection
of extensions for calls during day time.
• When NR?: same as described above for calls during day time.
• When Busy?: same as described above for calls during day time.
• Start DDI#: Enter the first DDI number given by the Service Provider. The Start DDI Number may be
the Root DDI number, as seen in most of the cases.
• Total DDIs: Enter the Total DDI numbers. Ask your Service Provider.
• Start Extension#: Enter the number of the first extension from which the DDI assignment is to be
done.
For example, your Service Provider has given you DDI numbers 2630551 to 2630560. These numbers
are to be routed to extensions 2001 to 2010. In this case, the Root DDI number as well as the Start
DDI# will be 2630550. As 10 DDI numbers are used, enter Total DDI numbers =10 and Start Extension
# = 2001.
• When NR?: same as described above for calls during day time.
• When Busy?: same as described above for calls during day time.
Now, configure the same parameters for Root DDI#2
• If you select Port-wise routing, configure the following parameters:
• Route Calls during Day to: Select the landing destination for calls on T1E1PRI trunks during day
time. You may select any option: Operator extension, other extensions, Built-In Auto Attendant or the
Auto Attendant of the Voice Mail (if available).
If you select Extensions as landing destination, you must select the extensions. Double click the field
and select the extensions from the left box.

• Route Calls during Night to: Select the landing destination for calls on T1E1PRI trunks during night
time. Follow the same procedure for selecting extensions as described above.
• Click Submit to save changes. Repeat the same steps to program the second T1E1PRI Trunk.
• If you have finished configuring the T1E1 PRI Trunks, click Next to navigate further.
Mobile Trunks
• You will reach this page only if your system has detected the presence of the Mobile Card in the system
(as the On-Site Configuration check box is enabled) or if you have specified the number of Mobile
Ports Used earlier in the System Pre-requisites page of the Wizard.
• SARVAM UCS supports up to 40 Mobile Ports.
• The Wizard makes configuration of the Mobile Trunks easy with Mobile Profiles. Instead of configuring
each Mobile Trunk individually, you can group together trunks that are to be assigned the same
features in a single Mobile Profile.
You can configure as many as 4 different Mobile Profiles using the Wizard.
• Change the label of the Mobile Profile, if required. Select the desired Mobile Trunk tab and click the Pencil
icon. A prompt will pop-up. Enter the desired name in the field of the prompt and click OK. The name you
programmed will appear. For example rename Mobile Profile 1 as 'Vodafone'. Naming the profile with the
Service Provider's name makes identification of the Trunk on which this profile is applied easy.

• Select the mobile trunks on which this Mobile Profile is to be applied. It is also possible to configure all the
other parameters and then select the mobile trunks.
To select the desired mobile trunks, double click the box. The Wizard will display the number of Mobile
Trunks you specified in the System Pre-requisites page on the left side box. All the Mobile Trunks appear
in this box and are arranged sequentially in the increasing order of their software port number.
Place your cursor on the desired trunks and click Select>> button. The selected Mobile Trunks will appear
on the right side box. It is also possible to select a range of trunks at a time pressing the 'SHIFT and 'Down'
arrow keys.
• Change the SIM PIN for the mobile profile.
Recall that you have changed the SIM PIN of the SIM Card using a Handset before installing it in the
system to the default 1234. Now, you may change it to a desired SIM PIN.
Since you are changing the SIM PIN for a Mobile Profile, this SIM PIN will be applied for all the Mobile
Trunks you selected for this Mobile Profile.
You must click ‘Submit’ after you enter the new PIN. Wait for 5 seconds, and then refresh this page to view
the new SIM PIN.
• Select the extensions to which the incoming calls on the Mobile Trunks should be routed.
• In Route Calls during Day to, select the landing destination for calls on Mobile trunks during day time.
You may select any option: Operator extension, other extensions, Built-In Auto Attendant or the Auto
Attendant of the Voice Mail (if available).

If you select Voice Mail Auto Attendant, the voice messages recorded and stored in Voice Mail module
will be played.
If you select Extensions as the landing destination, select the extensions in the corresponding field.
Double click this field, the multiple selection box will open. Select the extensions from the left side box.
The selected extension numbers will appear in the field, separated by a comma.
• In Route Calls during Night to, select the landing destination for calls on Mobile trunks during night time.
Follow the same instruction as in the previous step to select extensions as the landing destination.
• Click Submit to save your settings in the Mobile Profile. Repeat these steps to configure another Mobile
Profile.
• If you have finished configuring the Mobile Profiles, click Next to navigate further.
• If there is any mobile trunk you have not assigned a Mobile Profile, the Wizard will pop-up an alert
informing you about the trunk you have not programmed. You must complete configuring the parameters
for this trunk and only then will the wizard allow you to configure other parameters.

VoIP Network
• To configure the VoIP Parameters, click VoIP Network. The VoIP Parameters page opens.
VoIP Server Domain

Click VoIP Server Domain to expand.
• VoIP Server Domain: This parameter is of relevance only if you are configuring SIP Extensions on the
system.
The system is capable of maintaining a domain for registering SIP/UC clients (any SIP-enabled device) as
SIP Extensions.
Configure the Server Domain if you want SIP/UC clients to register with the Registrar Server of the system
using the domain handled by the system106. The Domain Name can be a maximum of 40 characters.
Default: Blank.
If you configure Server Domain for registration of SIP clients, you must also map the Domain Name and
the IP Address of the WAN Port to the DNS Server in the network.
106. SIP clients can be registered with the CPU either using the domain handled by the Vocoder module or using the WAN or LAN Port
IP Address.
If domain is programmed, the CPU will listen for the SIP message which is redirected to the programmed domain only. It will also
listen for SIP messages on the WAN IP address and LAN IP address.
But if domain is not programmed, the CPU will listen for SIP messages only on the WAN IP Address and LAN IP address.

RTP Mode
Click RTP Mode to expand.
• RTP Mode: Select the desired RTP mode using which you want SARVAM UCS to route SIP to SIP calls.
You can select from the following:
• Transcoding: When this option is selected, RTP packets will be routed through the Vocoder module
and Vocoder channels107 will be used for SIP to SIP calls. SIP/UC Users (Open/Extended SIP Devices
and Matrix VARTA UC Clients) will be able to access all the features of SARVAM UCS. This option
uses two Vocoder channels for SIP to SIP calls. Thus the maximum number of SIP to SIP calls per
Vocoder Module is equal to the number of Vocoder channels divide by 2.
• RTP Relay: When this option is selected, RTP packets will be routed through the Vocoder module but
no Vocoder channel will be used for SIP to SIP calls. The system will use Vocoder channels to route
SIP to TDM calls and vice versa. System will use Vocoder channels for some features and Open SIP
Clients will be able to use limited features of SARVAM UCS. Refer to “SARVAM UCS Features
supported with RTP/Direct RTP” for more details.
• Direct RTP: When this option is selected, no Vocoder channel will be used for SIP to SIP calls and
RTP packets will be sent to and fro directly between SIP end points. If transfer of RTP packets is not
possible between SIP end points, system will use RTP Relay as the fallback option. The system will
use Vocoder channels to route SIP to TDM calls and vice versa. System will use Vocoder channels for
some features and Open SIP Clients will be able to use limited features of SARVAM UCS. Refer to
“SARVAM UCS Features supported with RTP/Direct RTP” for more details.
• Direct RTP is not supported on SIP Trunks and Matrix VARTA UC Clients. System will use RTP Relay
for SIP Trunks and Matrix VARTA UC Clients.
• If RTP Relay or Direct RTP is selected,

• Maximum 550 Audio calls or 55 Video calls are supported in SARVAM UCS.
• Certain voice messages will not be played. See “Voice Message Applications” for more details.
• If you change the RTP Mode, the system will release all ongoing calls.
• MoH Vocoder Preference: If RTP Mode is set as RTP Relay or Direct RTP, the system will play the MoH
through the Vocoder module when the system holds any SIP end point. The MoH is played as per the
configured MoH Vocoder Preference.
Select the Vocoders in the order of their preference, for MoH Vocoder Preference 1, MoH Vocoder
Preference 2 and MoH Vocoder Preference 3.
If required, you can customize the MoH to be played as per your requirement. For detailed instructions,
see “Uploading Custom MoH”.
107. The number of Vocoder channels that will be supported would be as per the license you purchase.

Quality of Service (QoS)
Click Quality of Service (QoS) to expand.
• Quality of Service (QoS): QoS refers to priority of IP packets on network layer. It can be programmed for
both signaling (SIP) and media (Voice, Video and Fax). Configure the following types QoS:
• SIP DiffServe/ToS: The system sends all the SIP signaling messages with this QoS setting. This field
defines the priority bits for SIP messages. The Valid DiffServe range is from 00-63, default: 26
• Voice DiffServe/ToS: The system sends all the Voice packets with this QoS setting. This field defines
the priority bits for Voice packet. The Valid DiffServe range is from 00-63, default: 46
• Video DiffServe/ToS: The system sends all the Video packets with this QoS setting. This field defines
the priority bits for Video packet. The Valid DiffServe range is from 00-63, default: 46
• Fax DiffServe/ToS: The system sends all the Fax packets with this QoS setting. This field defines the
priority bits for Fax packet. The Valid DiffServe range is from 00-63, default: 46
QoS parameters are applicable for all packets (SIP/ Media) leaving both LAN and WAN port as well as
TCP connection.
Channel Reservation for SIP Trunks

Click Channel Reservation for SIP Trunks to expand.
• Channels Reserved for SIP Trunks: The system supports up to 128 voice channels, which can be used
by SIP Extensions and SIP trunks.
It may happen that SIP Extension users use up most of the channels of the Vocoder module, leaving too
few or none for making/receiving SIP Trunk calls.
This can be avoided by reserving some voice channels exclusively for SIP trunk calls.
Specify the minimum number of voice channels you want to reserve for SIP Trunk calls. By default, no
channel is reserved.

SIP 100rel
Click SIP 100rel to expand.
• SIP 100rel: This parameter is to be configured if you want to support reliable transmission of (SIP)
provisional responses. Enable 100rel by selecting the check box, if you want CPU to use 100rel for reliable
transmission of SIP provisional responses and to use PRACK (Provisional Acknowledgement). Default:
Disabled.
SIP Over TCP

Click SIP Over TCP to expand.
• SIP Over TCP: SARVAM UCS supports transporting of SIP messages over User Datagram Protocol
(UDP), Transfer Control Protocol (TCP) as well as Transport Layer Security connection (TLS). Despite the
advantages that SIP over TCP and SIP over TLS offer, it is more common to use UDP to transport SIP
messages.
By default, SIP Over TCP is enabled. To be able to send SIP messages over TCP, you must configure
‘TCP’ as the ‘Default Transport for Outgoing Messages’ on the ‘SIP Trunk Parameters’ page.
If you do not want to transport SIP messages using TCP, clear the SIP Over TCP check box.
SIP Over TLS

Click SIP Over TLS to expand.
• SIP Over TLS: SARVAM UCS supports transporting of SIP messages over Transport Layer Security. TLS
offers secure SIP signaling.
By default, SIP Over TLS is enabled. To be able to send SIP messages over TLS, you must configure
‘TLS’ as the ‘Default Transport for Outgoing Messages’ on the ‘SIP Trunk Parameters’ page.
If you do not want SIP messages to be transported using TLS, clear the SIP Over TLS check box.

SIP/RTP Ports
Click SIP/RTP Ports to expand.
• SIP UDP Port: This port defines the port on which SARVAM UCS listens for SIP messages transported
over UDP. This port is also used as the source port for sending SIP messages to the remote peer. The
valid range for this port is 1025-65535. Default: 05060.
• SIP TCP Port: This port defines the port on which SARVAM UCS listens for SIP messages transported
over TCP. This port is also used as the source port for sending SIP messages to the remote peer. The valid
range for this port is 1025-65535. The default SIP TCP Port is 05060.
• SIP TLS Port: This port defines the port on which SARVAM UCS listens for SIP messages transported
over TLS. This port is also used as the source port for sending SIP messages to the remote peer. The valid
range for this port is 1025-65535. The default SIP TLS Port is 05061.
• RTP Listening Port: This port defines the port on which SARVAM UCS listens for RTP Packets. This port
is also used as the source port for sending RTP packets to the remote peer. The valid range for this port is
1025-65278. The default RTP Listening Port is 08000.
Timers
Click Timers to expand.
• SIP Invite Timer (sec): This is the time in seconds that the Vocoder module waits for a response from the
called party after ending INVITE message. This timer starts after sending INVITE message to the called
party and stops on receipt of the provisional response or the final response or when the user disconnects
the call. On expiry of the timer, the call process is terminated by the SARVAM UCS and an error tone is
played to the user. The range of the SIP Invite Timer is 010-180 seconds. Default: 30 seconds.
• SIP Provisional Timer (sec): This is the time in seconds that the Vocoder module waits for the final
response after receiving the provisional response from the called party. This timer starts on the receipt of
the provisional response from the called party and stops on receipt of the final response from the called
party or when the user disconnects the call. On expiry of the timer, the SARVAM UCS terminates the call
process and plays error tone to the user. The range of SIP provisional Timer is 010-180. Default: 60
seconds.
• General Request Timer (sec): The time in seconds for which the Vocoder module waits for response for
a transaction request. This timer starts on the initiation of a transaction. This timer stops on receipt of a
response for the request. On expiry of the timer, the Vocoder module clears the transaction. This timer is
used for Registration request, etc. The range of the General Request Timer is 10-60 seconds. Default: 20
seconds.

• Click Submit to save.
• Click 'Next' to navigate to further.
SIP Trunks
• You will reach this page only if you have specified Number of SIP Trunks Used on the System Pre-
requisites page.
• SIP Trunks are to be configured only if you are using Internet Telephony Service Providers for VoIP
calls.
• The Wizard however, allows you to configure only 4 SIP Trunks (even if you have specified more than
4 SIP Trunks being used on the System Pre-Requisites page). You can program the remaining SIP
Trunks from Configuration.
• You may change the label of the SIP Trunk (rename 'SIP Trunk 1'), if required. You may use the name of
the ITSP to make identification of the SIP Trunk easy.
Click the Pencil icon. A prompt will pop up, asking you if you want to change the name of this SIP Trunk.
Enter the desired name in the blank field. Click OK. The new name will appear instead of the SIP Trunk
number.

• Select the Enable SIP Trunk check box to enable the SIP Trunk.
• Enter the SIP ID provided by the ITSP. This is the ID which callers will use to call this SIP Trunk. The SIP
ID may be a number or text.
• Enter the Proxy/Registrar Server Address and the Registrar Server Port provided by the ITSP. The
Registrar Server Address can be an IP Address or domain. The Registrar Server Listening Port ranges
from 1025 to 65535. The default Registrar Server Port is 5060.
• Enter the Authentication ID (User ID) and Password provided by the ITSP.
• Select the Enable Outbound Proxy check box, if the ITSP who provided this SIP Trunk has a SIP
outbound server to handle voice calls. By default Outbound Proxy is disabled.
• If you have enabled Outbound Proxy, enter the Outbound Proxy Server Address and the Server Port
provided by the ITSP. This can be an IP Address or Domain name.
• Set the desired Vocoder Preference for this SIP Trunk. Vocoders are the various Voice Codecs used to
compress the data in RTP packets for optimum use of bandwidth and for ensuring voice quality. You can
set 7 Vocoders options in the order of preference for this SIP account.
• Select the DTMF Option. The DTMF option you select will determine how the DTMF digits will be sent
over the IP Network, when a DTMF digit is pressed. The system supports three DTMF options: RTP (RFC
2833), SIP Info, and InBand. Select the appropriate option. By default RTP (RFC 2833) is selected.
• RFC2833 Payload Type: If you have selected RTP (RFC 2833) as the DTMF Type, you must configure
the value of RFC2833 Payload Type. The RTP packets will be tagged as DTMF as per the set value. The
value of RFC2833 Payload Type can be set from 96 to 124.
• Select the appropriate Fax Type. The system supports T.38 (UDPTL), T.39 (RTP) and Pass-Through.
• Select the Source Port IP Address.
• Select Use Ethernet Port IP Address, if the WAN port of the system is connected directly to the public
internet.
• Select Use IP Address Fetched Using STUN, if the WAN port of the system is located behind a NAT
router other than Symmetric.
• Select Use Router's Public IP Address, if the WAN port of the system is located behind a NAT Router
(any type).
• In No. of Simultaneous Calls, select the number of simultaneous calls you want to allow on this SIP
Trunk.
The number of simultaneous SIP calls depends on the number of simultaneous calls allowed by the ITSP
with whom you have subscribed this SIP Trunk and the number of simultaneous calls supported by
SARVAM UCS.
• In Route Calls during Day to, select the landing destination for calls on SIP trunks during day time. You
may select any option: Operator extension, other extensions, Built-In Auto Attendant or the Auto Attendant
of the Voice Mail (if available).

If you select Voice Mail Auto Attendant, the voice messages recorded and stored in Voice Mail module
will be played.
If you select Extensions as the landing destination, select the extensions in the corresponding field.
Double click the field, the multiple selection box will open. Select the extensions from the left box.
change the order of the selected landing extensions in the right side box using Up/Down Arrow.
• In Route Calls during Night to, select the landing destination for calls on SIP trunks during night time.
Follow the same instruction as in the previous step to select extensions as the landing destination.
• Click Submit to save your settings in the SIP Trunk. Repeat these steps to configure another SIP Trunk.
• If you have finished configuring the SIP Trunks, click Next to navigate further.
• If there is any SIP Trunk you have not configured, the Wizard will pop-up an alert informing you about the
trunk you have not programmed.
You must complete configuring the parameters for this SIP Trunk and only then will the wizard allow you to
For the SIP Trunks, the Wizard uses the following resources:
• Routing Groups 67-74.
• Trunk Feature Template 46-49.
• SIP Hardware Template 2-5.
• Do not change the these resources when using Configuration.
Emergency Numbers
• This page displays the default Emergency Numbers according to the Region selected for the system.
In other words, it displays the Emergency Numbers specific to your country where the SARVAM UCS is
installed.
• If there are no Emergency Numbers defined as per the selected region, the system displays the
message ‘ No Emergency Number programmed.’
• The Emergency Numbers on this page are non-editable. All you need to do is to select the Outgoing
Trunk Bundle Group (OGTB) for each Emergency Number. For example, '112' is the default
Emergency Number for the mobile network. So, you may select the Mobile Trunk for dialing this
number.
• Make sure that the trunks configured by default for each Emergency number route the Emergency call
to the correct network.

• For each default Emergency number, select the trunks to be used to route this call. Double click the
Through field. The box will open, displaying the trunk types present in the system in the left box.
All the Trunk types present in the system are arranged by their type, in the sequence of their software port,
irrespective of their hardware location.
• Double click the desired trunk type and click Select>>. The selected trunk type will appear on the box on
the right.
If you select trunks of the same type in sequential order, like: CO-001, CO-002, CO-003, BRI-001, BRI-002
and MOB-002, the same trunk type will be grouped in one OG Trunk Bundle; CO-001, CO-002, CO-003
will be OGTB #1, BRI-001 and BRI-002 will be grouped as OGTB#2, and MOB-002 as OGTB#3.
If you select the trunks of the same type in a non-sequential order, for example: CO-001, MOB-001, BRI-
001 and CO-002, four OGTBs will be formed with CO-001 as member of OGTB#1, MOB-001 as member
of OGTB#2, BRI-001 in OGTB#3 and CO-002 in OGTB#4. So, despite two same trunk types being
selected (CO-001 and CO-002) they are grouped in separate OGTBs.
It is possible to change the sequence of trunks on the right side box using the Up/Down Arrow. You can
also delete a particular trunk from the right box.
• Click OK. All the trunks you selected for a particular Emergency Number (in the right-hand side box) will
appear on the 'Through' field, sequentially separated by comma in the order of selection made.
• Click Submit to save the values entered on this page.
• Click Next to navigate further.

You may program the Emergency Numbers of your country from Configuration.
Disclaimer:
• Matrix Comsec will not be responsible for incorrect programming of Emergency Numbers.
Done
You have now reached the last screen of the Wizard. Click Done to submit the configuration you have done using
the Wizard.
After completing the validation of the configuration, the Wizard will inform you about successful configuration of the
system with the message: “Congratulations! System is configured as per your wish".
Exiting the Wizard

Each page of the Wizard also has an Exit button, which you may click to exit the Wizard.
However, the changes you made till the page you exited from will not be applied to the system configuration files.
The Standard PBX Wizard has been designed keeping a very broad user base in mind and is limited to the
configuration of the parameters most commonly required by a broad user group. So, it does not cover the
configuration of all the features and facilities of the system. You can configure other features “Configuring
SARVAM UCS”.

Using Configuration
The Configuration allows configuration of all programmable parameters of the system.
You can program all the parameters using Jeeves or an extension Telephone (DKP or SLT) of the SARVAM UCS.
You are recommended to use Jeeves. If you choose to use a telephone, you are recommended to use a DKP for
ease of operation.
To configure the Configuration parameters, you must log into Jeeves via System Engineer Login and click the
Configuration link.

Also, refer the topic “Configuring SARVAM UCS”.
To program the parameters using a Telephone, enter into SE mode from a DKP/SLT by dialing 1#91 followed by
the SE password.
For detailed instructions, refer the topic “Entering the SE mode using a Telephone”.
Avoid modifications or use of the following system resources, when configuring the system from the
Configuration.
• Station Basic Feature Template number 2 to 10.
• Outgoing Trunk Bundle Groups 2 to 27
The system uses these resources for the Quick Installation Wizard-Standard PBX. Changes made in them
will not be updated in the Wizard.

Configuring System Pre-requisites
System Pre-requisites for SARVAM UCS

This displays the System resources (number of trunk and extension ports supported) of SARVAM UCS. It is quite
common for users to utilize the system resources below its capacity, especially when they begin using a new
system.
To make the task of configuring for such users easier, SARVAM UCS allows you to specify the number of trunks
and extension ports you want to configure. Accordingly all the relevant pages of Jeeves will show only as many
trunk and extension ports that you have specified, instead of showing all the ports supported by SARVAM UCS.
To be able to do this, you must configure the System Pre-requisites using Jeeves or a Telephone.
On-site Configuration
SARVAM UCS makes configuration even more focused by making it possible to configure only those trunk and
extension ports which are actually present in the system.
This can be done by enabling the On Site Configuration flag.
When On Site Configuration flag is enabled, SARVAM UCS will detect all the different trunk and extension port
types present in the system (at Power-ON). Accordingly, all the relevant pages of Jeeves will show only those ports
detected by the system for configuration.
The system will detect the presence of ports at each Power ON/Reset. Whenever a new card is found, the range of
ports is updated and displayed on Jeeves. You can then define the number of ports to be used.
By default, the On-site Configuration flag is disabled.
When the On Site Configuration flag is enabled, the Quick Installation Wizard - Standard PBX will also
display only the ports and port types that are on-board. Enable this flag if you want the Wizard to display
only the port types present in the system.
It is recommended that you enable the On-Site Configuration flag when you are configuring the system at
the installation site.
Defining System Pre-requisites using Jeeves

• Log into Jeeves as System Engineer.

• Under Configuration, click System Pre-requisites to open the page.
• Configure the following parameters:
• Customer Name: You can assign the name of the enterprise/organization that is using SARVAM UCS
as the Customer Name. The Customer Name may contain up to 80 characters. You may enter the
address of organization/enterprise along with the name.
The Customer Name you assign will appear on the various System Reports generated and printed by
the SARVAM UCS.
• On Site Configuration: Click the On-site Configuration check box to enable.
Click Submit at the bottom of the page.
The fields for Number of Ports Used, will be populated with exactly the number the system has
detected.
The fields for port types which are not available on-board are displayed as non-editable fields.
For example, the system has detected SLT20 Card at power-on, so the maximum number of SLT ports
will be 20. If you want only 18 SLTs to be used, select 18.
• If the system has detected BRI ports, the fields BRI Trunks and ISDN Terminals will be editable.
• By default, the number of BRI Trunks will be equal to the Number of BRI ports used.
• The number of ISDN terminals used by default will be zero. Only when the System Engineer changes
the value of the Number of BRI Trunks to be used will the number of ISDN Terminals change to the

number of available BRI ports x 8. In other words, the number of ISDN terminals will be: Number of BRI
Ports used minus the Number of BRI Trunks used multiplied by 8.
• Model Type: This displays the name of the model that you are configuring — ETERNITY LENX,
ETERNITY MENX or ETERNITY GENX.
• Number of Ports Used: Define the number of ports to be used for each Port Type (CO, DKP, SLT,
Mobile, T1E1 PRI, BRI, SIP) in the respective boxes.
For example, if you want 8 CO Trunks, 24 DKP extensions, and 128 SLT extensions to be used, select
the same numbers in the respective boxes.
• Click Submit at the bottom of the page to save changes.

Configuring ‘Region’
The SARVAM UCS is a versatile system that can operate anywhere in the world, meeting the diverse customer
requirements worldwide.
To speed up the process of system configuration, SARVAM UCS is supplied with factory-set values for the system
and feature settings, referred to as “Default Settings”. These factory-set values are loaded when the system is
installed and are sufficient for getting the system into operation. However, users may alter or customize the Default
Settings to match their exact requirement.
SARVAM UCS provides Default Settings to match country/region-specific requirements of users around the world.
The system is designed to work efficiently in any country with these default settings.
To load the country-specific Default Settings, users must select the Region that is the country in which the system
is installed.
Certain countries are divided into various regions. If you select only a different region in the same country the DST
and Date and Time Settings only will be change as per the selected region. The other parameters are country-
specific.
India is selected as the default Region. So, if you are installing SARVAM UCS in a country other than India, change
the Region.
Changing Region using Jeeves

• Under Configuration, click Regional Settings.
• Click Region Selection to open the page.
• In the Region list, select the country where the system is installed.

Changing Region using a Telephone
• Enter SE mode from a DKP/SLT.
To select Region, dial:

• 5301-Region Code
Where,
Region Code is the code for the country in which SARVAM UCS is installed.
Refer the following table and select the appropriate Region Code.
Region Code Country Name

001 Afghanistan
002 Algeria
003 Antigua and Barbuda
004 Argentina
005 Australia (Perth)*
006 Australia (Adelaide)*
007 Australia (Brisbane, Canberra, Melbourne, Sydney)*
008 Austria
009 Bahamas
010 Bahrain
011 Bangladesh
012 Belarus
013 Belgium
014 Bhutan
015 Bolivia
016 Bosnia and Herzegovina
017 Botswana
018 Brunei
019 Brazil (Fernando De Noronha)*
020 Brazil (Brasilia, Rio de Janeiro, Sao Paulo)*
021 Brazil (Manaus)*
022 Brazil (Acre)*
023 Bulgaria
024 Cambodia
025 Cameroon
026 Canada (St. John's)*
027 Canada (Halifax)*
028 Canada (Montreal, Ottawa, Toronto)*
029 Canada (Winnipeg)*
030 Canada (Calgary)*
031 Canada (Vancouver)*
032 Chile
033 China
034 Colombia
035 Costa Rica
036 Croatia
037 Cuba
038 Cyprus

039 Czech Republic
040 Denmark
041 Egypt
042 Fiji
043 Finland
044 France
045 Germany
046 Greece
047 Guyana
048 Hong Kong
049 Hungary
050 India
051 Indonesia
052 Iran
053 Iraq
054 Ireland
055 Israel
056 Italy
057 Japan
058 Jordan
059 Kazakhstan
060 Kenya
061 Korea - North
062 Korea - South
063 Kuwait
064 Kyrgyzstan
065 Lebanon
066 Libya
067 Malaysia
068 Maldives
069 Mauritius
070 Mexico (Mexico City)*
071 Mexico (Chihuahua)*
072 Mexico (Tijuana)*
073 Mongolia
074 Mozambique
075 Myanmar
076 Namibia
077 Nepal
078 Netherlands
079 New Zealand
080 Nigeria
081 Norway
082 Oman
083 Pakistan
084 Paraguay
085 Peru

086 Philippines
087 Poland
088 Portugal
089 Qatar
090 Romania
091 Russia (Moscow, St. Petersburg)*
092 Russia (Novosibirsk)*
093 Russia (Vladivostok)*
094 Singapore
095 Slovakia
096 South Africa
097 Spain
098 Sri Lanka
099 Sudan
100 Sweden
101 Switzerland
102 Syria
103 Taiwan
104 Tajikistan
105 Thailand
106 Turkey
107 Uganda
108 Ukraine
109 United Arab Emirates
110 United Kingdom
111 United States (Atlanta, Augusta, Boston, Charlotte, Columbus, Detroit, Indiapolis,
Miami, NY, Philadelphia, Washington)*
112 United States (Chicago, Dallas, Des Moines, Memphis, Minneapolis, New Orleans,
Oklahoma, Omaha, St. Louis)*
113 United States (Albuquerque, Boise, Cheyenne, Denver, Salt Lake City)*
114 United States (Las Vegas, Los Angeles, Phoenix, San Francisco, Seattle)*
115 United States (Juneau)*
116 United States (Hawaii)*
117 Uzbekistan
118 Venezuela
119 Vietnam
120 Yemen
121 Yugoslavia
122 Zambia
123 Zimbabwe
124 Saudi Arabia
125 Cote d'Ivoire
* These countries are divided into several regions hence only Date and Time Settings and DST will change as per
the region you select. Other parameters will be country-specific.

By default, the region code is 050 (India).
For example, if SARVAM UCS is installed in the Philippines, dial 5301-086
• Exit SE mode.

SARVAM UCS has a LAN and WAN Port.The network parameters must be configured according to your network
scenario.
SARVAM UCS may be installed typically, in a Public IP Network or in a Private network, behind a NAT Router.
When the SARVAM UCS is installed in a Public IP Network,

• the WAN Port of the card is connected to a Broadband Router/Modem.
• Public IP is assigned to the WAN Port.
• the LAN Port of the card is connected to a LAN Switch/Hub to which SIP devices are connected.
When the SARVAM UCS is installed in a Private Network, behind a NAT Router,
• the WAN Port of the card is connected to the LAN Switch/Hub.
• Private IP is assigned to the WAN Port.
• SIP devices within the LAN can get registered with the card.
When your system is installed in a Private Network, you may have to change the IP Address and Subnet
Mask of the WAN Port, before connecting it to the LAN Switch/Hub. However, this will not be necessary, if
there is a DHCP server on the LAN which will automatically assign an IP Address that does not conflict
with any other device on the LAN.
If you need to change the IP Address and the Subnet Mask of the LAN Port and the WAN Port of SARVAM
UCS, you may do so by dialing System Commands from the telephone connected to the SLT Port. It is also
possible to view the current IP Address and Subnet Mask of the LAN Port and the WAN Port by dialing
System Commands from the telephone connected to the SLT Port. For instructions, see “System
Commands”.
Depending on your installation scenario, configure the Network Parameters using Jeeves or dialing commands
from a Telephone.
Configuring Network Parameters using Jeeves

• Login as System Engineer.
• Under Configuration, click Network Parameters.
• Click Parameters to open the page.

Network Port
Click Network Port to expand.
• IP Addressing mode: Select the IP version you want the system to use. You may select — IPv4 only or
IPv4 and IPv6. Default: IPv4 only.
If you select IPv4 only, you can configure the IPv4 parameters only.
If you select IPv4 and IPv6, you can configure both IPv4 and IPv6 parameters.
• Preferred DNS Server: If you select IPv4 and IPv6 as the IP Addressing mode, you must select the
Preferred DNS Server — IPv4 or IPv6. Default: IPv4
LAN Port (IPv4 settings)

Click LAN Port (IPv4 settings) to expand.

• IP Address: Enter the IP Address to be assigned to the LAN Port. The default IP Address is
192.168.002.100. You can assign only Static IP to the LAN Port.
• Subnet Mask: Enter the Subnet Mask to be assigned to the LAN Port. The default Subnet Mask is
255.255.255.0
WAN Port (IPv4 settings)

Click WAN Port (IPv4 settings) to expand.
• Connection Type: Select the appropriate Connection Type for the WAN port, according to the IP
Addressing scheme of your installation scenario. Consult your System Administrator also in this regard.
Default: Static.
• Static: Select this option if the connection type is Static. When you select this option, you must:
• assign an IP Address to the WAN Port.
• change the Subnet mask of the WAN Port as appropriate.
• configure the Router's LAN Interface IP Address as the Gateway IP Address.
• Configure the DNS Address/Domain Name provided by your ISP or ask your LAN Administrator for
the DNS Address and Domain Name.
• DHCP: Select this option if the connection type DHCP. As the DHCP Server will automatically assign IP
Address, Subnet Mask, Gateway Address to the WAN Port, you need not configure any of these.
• PPPoE: Select this option if the connection type is PPPoE. As the PPPoE server will automatically
assign the IP Address, Subnet Mask and Gateway Address to the WAN Port, you need not change any
of these. You must program the User ID, Password and PPPoE Service Name as provided by your ISP.
Program the Service Name only if it has been provided. You must set DNS address.
Configure the following PPPoE parameters:
• User ID: Enter the User ID provided by the Internet Service Provider. The User ID may be a
maximum of 64 characters.
• Password: Enter the User Password provided by the Internet Service Provider. The password may
be a maximum of 64 characters.
• Service Name: Enter the PPPoE Service Name, if provided by your Internet Service Provider. The
Service Name may consist of a maximum of 64 characters. If Service Name is not required, leave
this field blank.
• IP Address: If you have selected 'Static' as the Connection Type, the default IP is 192.168.001.100.

If you configure the PPPoE User ID, Password and Service Name using commands, you can configure a
maximum of 16 characters only.
If you have selected DHCP or PPPoE as the Connection Type, the IP Address will be assigned by the
DHCP/PPPoE server.
• Subnet Mask: You must enter the Subnet Mask, only if you have selected 'Static' as the Connection Type.
If you have selected DHCP/PPPoE as the Connection Type, the Subnet Mask will be assigned by the
DHCP/PPPoE server.
• Default Gateway: You must enter the Gateway IP Address, only if you have selected 'Static' as the
Connection Type.
If you have selected DHCP/PPPoE as the Connection Type, the Gateway IP Address will be assigned by
the DHCP/PPPoE server.
Domain Name Server (DNS) for IPv4

Click Domain Name Server (DNS) for IPv4 to expand.
• Configure the following DNS Connection settings for the WAN Port:
• DNS Address Assignment: If you have selected 'Static' as your network Connection Type (IP
Addressing), you can select only 'Static' as the DNS Address Assignment.
If you have selected DHCP as your network Connection Type, and the DHCP server provides DNS
Address, set the DNS Address Assignment to 'Auto'. If the DHCP server does not provide DNS
Address, set DNS Address Assignment as 'Static' and configure the DNS Server Address provided by
your ISP.
If you have selected PPPoE as your network Connection Type, and the PPPoE server provides DNS
Address, set the DNS Address Assignment as 'Auto'. If the PPPoE server does not provide DNS
Address, set the DNS Address Assignment as 'Static' and configure the DNS Server Address provided
by your ISP.
• DNS Address: This field will be editable only if you have selected DNS Address Assignment as 'Static'.
Enter the DNS Address here.
If you have selected DNS Address Assignment as 'Auto', the DNS Address will be assigned by the
DHCP/PPPoE server.
• DNS Domain Name: Configure the DNS Domain Name if provided by your ITSP/LAN Administrator.
Otherwise, keep it blank. The Domain Name may be a maximum of 40 characters. Default: Blank.

LAN Port (IPv6 settings)
Click LAN Port (IPv6 settings) to expand.
• IPv6 Addressing using: You can select — Complete Address or Prefix. Default: Complete Address.
If you select Complete Address,
• Configure the IPv6 Address and the Prefix Length. The IP Address configured will be considered as
the complete IPv6 address.
For example: 2001:0:3238:DFE1:63::FEFB
The Prefix Length is a decimal value that indicates how many of the high-order contiguous bits of the
address comprise the prefix (the network portion of the address).
Valid Range of the IPv6 Address is A to F, a to f, 0 to 9,:(colon). It can be a maximum of 39 characters.

Default: Blank.
The Prefix Length range is from 1 to 128 bits. Default: Blank.
If you select Prefix,
• Configure the IPv6 Prefix. The system will consider the configured value as 64 bit Prefix of the IPv6
Address. Then the system will generate the complete IPv6 Address from it. Default: Blank.
Valid characters 0 to 9, a to f, A to F and : (colon). It can be a maximum of 21 characters.
WAN Port (IPv6 Settings)

Click WAN Port (IPv6 Settings) to expand.
• IPv6 Connection Type: Select the appropriate Connection Type for the WAN port, according to the IP
Addressing scheme of your installation scenario.
You can select — Static, Statefull DHCPv6, Stateless Auto-Configuration, PPPoE. Default: Static.
• Static: Select this option if the connection type is Static.

• IPv6 Addressing using: You can select — Complete Address or Prefix.
If you select Complete Address,
• Configure the IPv6 Address and the Prefix Length. The IP Address configured will be
considered as the complete IPv6 Address.
The Prefix Length is a decimal value that indicates how many of the high-order contiguous bits
of the address comprise the prefix (the network portion of the address).
Valid Range of the IPv6 Address is A to F, a to f, 0 to 9,:(colon). It can be a maximum of 39

characters. Default: Blank.
If you select Prefix,
• Configure the IPv6 Prefix. The system will consider the configured value as 64 bit Prefix of the
IPv6 Address. Then the system will generate the complete IPv6 Address from it. Default: Blank.
Valid characters 0 to 9, a to f, A to F and : (colon). It can be a maximum of 21 characters.
• Default Gateway: Configure the Gateway IP Address for the WAN Port. It can be a maximum of 39
characters.
• PPPoE: Select this option if the connection type is PPPoE. As the PPPoE server will automatically assign
the IP Address, Subnet Mask and Gateway Address to the WAN Port, you need not change any of these.
Configure the following PPPoE parameters:
• IPv6 Scope Preference: IPv6 includes support of Global as well as Non-Global Addresses(Unique).
Select the scope of preference — Global or Unique. Default: Global.
• User ID: Enter the User ID provided by the Internet Service Provider. The User ID may be a maximum
of 64 characters.
• Password: Enter the User Password provided by the Internet Service Provider. The password may be
a maximum of 64 characters.
• Service Name: Enter the PPPoE Service Name, if provided by your Internet Service Provider. The
Service Name may consist of a maximum of 64 characters. If Service Name is not required, leave this
blank.
• Prefix Length: Configure the Prefix Length. Valid Range: 1 to 128 bits. Default: 064.
• Statefull DHCPv6: Select this option as the connection type, if your network uses DHCP to obtain various
necessary parameters from DHCP Servers so the DHCP clients can operate in an Internet Protocol (IP)

network. Statefull DHCP is centrally managed on a DHCP server(s); and the DHCP clients use Statefull
DHCP to obtain an IP address(es) and other useful configuration information from the DHCP server(s).
• Stateless Auto-Configuration: Select this option as the connection type, if your network uses DHCP to
obtain various necessary parameters from DHCP Servers so the DHCP clients can operate in an Internet
Protocol (IP) network. DHCPv6 for stateless configuration parameters allows a stateless or statefull
DHCPv6 client to export configuration parameters (DHCPv6 options) to a local DHCPv6 server pool. The
local DHCPv6 server can then provide the imported configuration parameters to other DHCPv6 clients.
• IPv6 Scope Preference: IPv6 includes support of Global as well as Non-Global Addresses. Select the
scope of preference — Global or Unique. Default: Global.
Domain Name Server (DNS) for IPv6

Click Domain Name Server (DNS) for IPv6 to expand.
• Configure the following DNS Connection settings for the WAN Port:
• DNS Address Assignment: If you selected 'Static' as your IPv6 connection type, you can select only
'Static' as the DNS Address Assignment.
If you have selected PPPoE as your IPv6 connection type, and the PPPoE server provides DNS
Address, set the DNS Address Assignment as 'Auto'. If the PPPoE server does not provide DNS
Address, set the DNS Address Assignment as 'Static' and configure the DNS Server Address provided
by your ISP.
If you have selected Statefull DHCP or Stateless Auto-Configuration as your network Connection Type,
and the a server provides DNS Address, set the DNS Address Assignment as 'Auto'. If the server does
not provide DNS Address, set the DNS Address Assignment as 'Static' and configure the DNS Server
Address provided by your ISP
• DNS Address: This field will be editable only if you have selected DNS Address Assignment as 'Static'.
Enter the DNS Address here. The DNS Address can be a maximum of 39 characters.
If you selected DNS Address Assignment as 'Auto', the DNS Address will be assigned by the Server.

MAC Address
Click MAC Address to expand.
• LAN Port MAC Address: This non-editable field displays the MAC Address of the LAN port.
• WAN Port MAC Address: This non-editable field displays the MAC Address of the WAN port.
• Use MAC Cloning: MAC Cloning is required when you want the WAN Port to use a MAC Address other
than its own unique MAC Address as source MAC Address.
When MAC Address Cloning is disabled, the WAN Port will use its unique MAC Address as the source
MAC Address on all Ethernet Frames. When MAC Cloning is enabled, the WAN Port will use the cloned
MAC Address on all Ethernet frames.
Select the check box to enable cloning of the MAC Address of the WAN Port. Default: Disabled.
• Clone MAC Address: If you have enabled MAC Cloning, enter the MAC Address to be cloned here.
Dynamic DNS (DynDns.org)

Click Dynamic DNS (DynDns.org) to expand.
• Dynamic DNS (DynDNS.org): This parameter is applicable only when you are going to configure the SIP
Extensions.
When the WAN port is assigned dynamic IP Address using DHCP or PPPoE, SIP-enabled devices
registered with the system as SIP Extensions need to change their configuration whenever a new IP
Address is assigned to the WAN port. Dynamic DNS resolves this.
SARVAM UCS supports Dynamic DNS Server client of the Service Provider Dynamic DNS.org.
If you want to use the DNS Service of DynDNS.org, configure these parameters:

• Enable Dynamic DNS: If you have taken the services of DynDNS.org, you must enable this check
box. Default: Disabled.
• Update IP Address at Power ON: When your WAN port is registered with the DynDNS.org, the
Dynamic DNS server stores the mapping between hostname and IP Address, which can be updated
periodically. However, if the system frequently sends IP Address update request to the DDNS server,
the server is likely to block the hostname in its database and terminate the DDNS services provided to
you.
So, if you restart the SARVAM UCS frequently, there is a great chance that DDNS server will block the
hostname configured in the system. This will in-turn affect the ability of the system to receive the calls
using DDNS host name since the entry (mapping between host name and IP Address) in the DNS
server will be deleted during those scenarios.
The system allows you a control over whether the system should update the IP Address in the DDNS
server at each Power ON or not. It also allows you to update the IP Address at any time, as required.
If you want to update the IP Address in the DDNS Server at each Power ON, select the check box.
Default: Disabled.
• User ID: Enter the User ID created by you with DynDNS.org here. A maximum of 40 characters,
including all ASCII characters are allowed. Default: Blank.
• Password: Enter the Password created by you for your User ID with DynDNS.org here. The password
may be not more than 24 characters long. Default: Blank.
• Host Name: Enter the Host Name created by you with DynDNS.org here. A maximum of 40
characters. All ASCII characters except < > and “ (double quote) are allowed.Default: Blank.
• Retry Trials: This count defines the number of attempts that the system should make to send the IP
Address Update Request to the Dynamic DNS Server. The Retry Count may be set from 1 to 9. By
default the count is set to 1. Default: 5.
• Update IP Address Now?: Click the Click to Update IP Address button, if you want to update the IP
Address in the DDNS server at any point of time. Default: Disabled.
This option is useful if you have not enabled the option Update IP Address during each Power ON. You
can update the IP Address in the DDNS server whenever required.
Router's Public IP Address

Click Router's Public IP Address to expand.
• Router's Public IP Address: This parameter is of relevance if the WAN port of the system is located
behind a NAT Router and SIP Messages are to be forwarded to the public internet.
Router's Public IP Address specifies the fixed IP Address of your NAT router required for NAT Traversal in
SIP messages.

Make sure you select the Use Router/STUN’s IP Address option as the Registrar Server Address, when
you configure SIP Extensions.
You can also use STUN as an alternative to the Router's Public IP Address as NAT Traversal mechanism.
Ask your Network Administrator about the NAT Traversal mechanism that suits best for your voice network
and program this parameter.
Simple Traversal of UDP through NATs (STUN)

Click Simple Traversal of UDP through NATs (STUN) to expand.
• Simple Traversal of UDPs through NATs (STUN):This parameter is to be configured only if the WAN
port is located behind a NAT Router and SIP Messages need to be forwarded to the public internet.
Simple Traversal of UDP through NAT (STUN) specifies the mechanism required for NAT traversal in SIP
messages. The STUN Server facilitates traversing through most NATs, except symmetric NATs. If your
router has symmetric NAT, do not program this parameter. If your router as asymmetric NAT, configure the
following STUN parameters:
• STUN Server Address: Enter the STUN Server Address, a maximum of 40 characters. Default: Blank.
• STUN Port: Enter the Listening Port of the STUN Server. The valid range for this field is from 1025-
65535. The default STUN Port is 03478.
• Use SIP Port fetched using STUN: By default, this check box is selected (enabled), to allow SIP Port
Number to be fetched using STUN in the SIP message. Clear this check box, if you are using Port-
Forwarding in the Router for SIP messages.
• Use RTP Port fetched using STUN: By default, this check box is selected (enabled), to allow RTP
Port Number to be fetched using STUN in the SIP message. Clear this check box, if you are using Port-
Forwarding in the Router for SIP messages.
You also need to select the 'Use IP Address fetched using STUN'’ check box in Network Parameters and
select the same as the Registrar Server Address, when you configure SIP Extensions.
Since STUN does not work with symmetric NAT, as an alternative to STUN you can use the Router's Public
IP Address as NAT Traversal mechanism. Ask your Network Administrator about the NAT Traversal
mechanism that suits best for your voice network and program this parameter.

Web Server Management
Click Web Server Management to expand.
• HTTP Server Port: Enter the HTTP Port number. Valid range: 80, 88 or any value ranging from 1025 to
65535. Default: 80.
• HTTPS Server Port: Enter the HTTPS Port number. Valid range: 443 or any value ranging from 1025 to
65535. Default: 443.
• SPARSH Port: Enter the SPARSH port number. The system will listen for the configuration request of
Matrix Extended SPARSH clients on this port only. If you want any SPARSH Extended client to register
with the system, you must configure the SPARSH port value as the Server Port along with the Server
Address in the SPARSH client.
Valid range: 80 or any value ranging from 1025 to 65535. Default: 80.
When the system is set to default, the SPARSH Port value will not be set to default.
Layer 2 VLAN/Cos
Click Layer 2 VLAN/Cos to expand.
• Layer 2 VLAN/CoS: This parameter is to be configured if the WAN port is to be connected in VLAN
network.
This parameter enables the SARVAM UCS to add VLAN header to the packets generated by it. The VLAN
header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of
traffic108.
VLAN Tag is applied on all packets generated by system (SIP, RTP, DNS, ARP, etc.), whereas CoS bits
are applied only for SIP and RTP packets generated by system.
108. The IEEE 802.1P standard allows Layer2 switches to prioritize the traffic, thus providing Quality of Service (QoS), better handling
of data that pass over a network, thereby resulting in greater reliability and quality. Quality of Service (QoS) on Layer2 is referred to
as Class of Service (CoS) which is defined by IEEE 802.1P.

The corresponding meaning of CoS bits with respect to traffic type is as follows:
COS Traffic Type
0 Best Effort
1 Background
2 Spare
3 Excellent Effort
4 Controlled Load
5 Video
6 Voice
7 Network Control
• Enable Layer 2 VLAN/CoS: Select this check box, if you want all packets generated by the system
(SIP, RTP, DNS, ARP, etc.) to be tagged with VLAN ID as configured. The CoS bits as configured for
SIP and RTP packets will be included in the VLAN header. Default: Disabled.
• VLAN ID: Consult your network administrator and configure the VLAN ID. The valid range for this is
from 0 - 4094. Default: 1.
• SIP CoS: Define the CoS (priority) bits in all SIP packets. The range of CoS bits is from 0 to 7. Default:
3.
• RTP CoS: Define the CoS (priority) bits in all RTP packets. The range of CoS bits is from 0 to 7.
Default: 6.
UDP NAT Keep Alive

Click UDP NAT Keep Alive to expand.
• UDP NAT Keep Alive: This parameter is to be configured when the WAN port is connected behind a NAT
router109 and SIP messages are transported over UDP. UDP NAT Keep Alive messages must be sent to
refresh the UDP binding in the NAT router.
• Enable UDP NAT Keep Alive: Enable this flag to send UDP NAT Keep Alive messages periodically to
refresh the binding in the NAT router. Default: Disabled.
• Interval (sec): Select Time period after which the WAN Port should send UDP NAT Keep Alive
messages. This time period should be less than the UDP Binding Timer of the router. The valid range is
001-999 seconds. Default: 180 seconds.
109. Network Address Traversal (NAT) allows multiple hosts in the network to share the single public routable IP address. Means all the
hosts in the private network shall be identified by single public IP address in the global IP cloud.

• Type of Message: Select the type of message type to be sent when UDP NAT Keep Alive is enabled.
Select either REGISTER or NOTIFY. Default:NOTIFY.
TCP NAT Keep Alive

Click TCP NAT Keep Alive to expand.
• TCP NAT Keep Alive: This parameter is to configured when the WAN Port is connected behind a NAT
router and SIP messages are transported over TCP. TCP NAT Keep Alive messages must be sent to
refresh the TCP binding in the NAT router.
• Enable TCP NAT Keep Alive: Enable this flag to send TCP NAT Keep Alive messages periodically to
refresh the binding in the NAT router. Default: Disabled.
• Interval: Select Time period after which the WAN Port should send TCP NAT Keep Alive messages.
This time period should be less than the Binding Timer of the router. The valid range is 0001-9999
seconds. Default: 120 seconds.
Viewing Network Parameters

You can view the status of WAN Port in Jeeves. To do this,
• Click Status to open the page.

It displays the statuses of the various IPv6 and IPv4 parameters, Maintenance Port Parameters and Redundancy
Configuration parameters.
When the system receives the IP Address from the DHCP/ PPPoE Server, the system performs DAD (Duplicate
Address Detection). If DAD fails due to conflict in IP Address, the respective network parameters (WAN or LAN)
need to be re-initialized.
To re-initialize the WAN parameters, click the IPv4 Network Reinitialization or IPv6 Network Reinitialization
button under WAN. Similarly, to re-initialize the LAN parameters, click the IPv4 Network Reinitialization or IPv6
Network Reinitialization button.
Configuring WAN Port using a Telephone

To select Connection Type, dial:

• 2116-Network Connection Type
Where,
Connection Type is
1 for Static

2 for DHCP
3 for PPPoE
Default: Static
To program PPPoE User ID, dial:

• 2117 - PPPoE User ID
Where,
PPPoE User ID may consist of a maximum of 16 ASCII characters.
Default: Blank.
To program PPPoE Password, dial:

• 2123- PPPoE Password
Where,
PPPoE Password may consist of a maximum of 16 ASCII characters.
Default: Blank.
To program PPPoE Service Name, dial:

• 2124-PPPoE Service Name
Where,
PPPoE Service Name may consist of a maximum of 16 ASCII characters.
Default: Blank.
To assign IP Address to WAN Port, dial:

• 2110-IP Address
Where,
IP Address is 15 digits maximum. 000 to 255 for the first 3 Octets and 001 to 254 for the 4th Octet. Use
zeros as fillers and dial the digits in a continuous sequence. Do not dial ‘.’ in the IP Address.
For example: to change the IP Address to 192.168.50.10 dial 2110-192168050010.
To assign Network Mask to the WAN Port, dial:

• 2111-Network Mask
For example: to change the Subnet Mask to 255.255.255.0 dial 2111-255255255000.
To program the Gateway Address, dial:

• 2112-Gateway Address
Where,
Gateway Address is the Router’s LAN Interface IP Address. Gateway IP Address may be a maximum
of 15 digits max. Follow the same instructions as assigning IP Address to the WAN Port.
To select DNS Connection Type, dial:

• 2115-DNS Connection Type
Where,
DNS Address Assignment Type is
1 for Static
2 for Automatic
Default: Static
To program HTTP Port, dial:

• 2121-HTTP Port
Where,
Port range is: 80, 1025 to 65535.
Default: 80

To enable/disable Dynamic DNS, dial:
• 2125-Code
Where,
Code is
0 for Disable
1 for Enable
Default: Disable.
To program Dynamic DNS User ID, dial:

• 2126-DDNS User ID
Where,
DDNS User ID may be a maximum of 40 ASCII characters.
To program Dynamic DNS User Password, dial:

• 2127-DDNS User Password
Where,
DynDNS User Password may consist of a maximum of 24 characters.
To program Dynamic DNS Host Name, dial:

• 2128 - DDNS Host Name
Where,
Host Name may consist of a maximum of 40 ASCII characters.
To program Dynamic DNS - Retry Trial Count, dial:

• 2129 - DDNS Retry Count
Where,
Retry Trial Count is from 1 to 9.
Default: 5.
To Update Dynamic DNS IP Address binding (‘Update IP Address Now?’ flag), dial
• 2130
To program Router's Public IP Address, dial:

• 2132-IP Address
Default: Blank.
To program STUN Server Address, dial:

• 2133-STUN Server Address.
Where,
STUN Server Address may be a maximum of 40 ASCII Characters.
Default: Blank.
To program STUN Server Port, dial:

• 2134- Port
Where,
STUN Server Port range is from 1025 to 65535.
Default: 3478.
To set STUN Query Interval, dial:

• 2135 - Interval
Where,
Interval is from 0001 to 9999 minutes.
Default: 120 min.

To select IP Addressing Mode, dial:
• 2145 - IP Addressing Mode
Where,
IP Addressing Mode is
1 for IPv4 Only
2 for IPv4 and IPv6
3 for IPv6 Only
To select IPv6 Connection Type, dial:

• 2146 - IPv6 Connection Type
Where,
Connection Type is
1 for Statefull DHCP
2 for Stateless Auto-configuration
3 for PPPoE
4 for Static
Default: Static
• Exit SE mode.
Viewing WAN Status from a Telephone

• Enter SE mode from a DKP (only).
To view IP Address of the WAN Port, dial:

• 2150
To view Network Mask of the WAN Port, dial:

• 2151
To view Gateway Address of the WAN, dial:

• 2152
To view current STUN query status, dial:

• 2159
To view Router’s Public IP Address fetched by STUN, dial:

• 2160
To view Dynamic DNS Status, dial:

• 2161
To view WAN link Status, dial:

• 2162
To view the configured IPv6 Address, dial:

• 2164
• Exit SE mode.

Configuring Maintenance Port Parameters
These Parameters are applicable only for ETERNITY LENX/MENX system.
You need to configure the Maintenance Port parameters to:

• transfer the configuration files from the Active card to the Redundant (Standby) card and vise-versa.
• access jeeves of the Redundant (Standby) card.
• restrict access to the facilities — Web Jeeves, FTP, Telnet.
Configuring Maintenance Port Parameters using Jeeves

• Click Maintenance Port Parameters to open the page.
LAN Port
Click LAN Port to expand.
• IP Address: Enter the IP Address to be assigned to the LAN Port of the Active Card. The default IP
Address is 192.168.002.101. You can assign only Static IP to the LAN Port.
• Subnet Mask: Enter the Subnet Mask to be assigned to the LAN Port of the Active Card. The default
Subnet Mask is 255.255.255.0

WAN Port
Click WAN Port to expand.
• IP Address: Enter the IP Address to be assigned to the WAN Port of the Active Card. The default IP
Address is 192.168.001.101. You can assign only Static IP to the WAN Port.
• Subnet Mask: Enter the Subnet Mask to be assigned to the WAN Port of the Active Card. The default
Subnet Mask is 255.255.255.0
• Default Gateway: It displays the Gateway IP Address you configured on the Network Parameters page.
This is un-editable.
Make sure the LAN and WAN IP Addresses are in different Subnets and these do not conflict with the LAN
and WAN IP addresses configured on the Network Parameters page.
Security Settings
You can restrict unauthorized access to the system using FTP, Web, Telnet. To know more, refer to “Security
Settings”.
Click Security Settings to expand.
• In Allow Web Server Access, you can select Don’t Allow, All IP Address/es or Only Trusted IP Address/
es option. By default, All IP Address/es is selected.
If you do not want to allow access to the Web Server, select Don’t Allow option.
If you want to allow access to the Web Server from specific IP Addresses only:
• select Only Trusted IP Address/es option.
• configure the IP Address/es and their respective Subnet Mask in the Trusted IP Address table.
• enable the Allow Web Server check box in the Trusted IP Address table.

• In Allow FTP Server Access, you can select Don’t Allow, All IP Address/es or Only Trusted IP Address/es
option. By default, Don’t Allow is selected.
If you want to allow access to the FTP Server from all IP Addresses, select All IP Address/es option.
If you want to allow access to the FTP Server from specific IP Addresses only:
• enable the Allow FTP Server check box in the Trusted IP Address table.
• In Allow TELNET Server Access, you can select Don’t Allow, All IP Address/es or Only Trusted IP
Address/es option. By default, Don’t Allow is selected.
If you want to allow access to the Telnet Server from all IP Addresses, select All IP Address/es option.
If you want to allow access to the Telnet Server from specific IP Addresses only:
• enable the Allow Telnet Server check box in the Trusted IP Address table.
Click Submit.

Configuring Redundancy Parameters
These Parameters are applicable only for ETERNITY LENX/MENX system.
The Redundancy parameters need to be configured to initiate the Redundancy process, that is, to transfer all
configuration and call information from the Active Card to the Redundant (Standby) card.
Configuring Redundancy Parameters using Jeeves

• Click Redundancy Configuration to open the page.
Configuration Synchronization Interface

Click Configuration Synchronization Interface to expand.
• In Enable Configuration Sync. on, you can select LAN Port or WAN Port. The connection between the
Active and Redundant Card will be established through this port. By default, LAN Port is selected.

Redundant CPU IP Address and Port
Click Redundant CPU IP Address and Port to expand.
• In Configuration Sync. Port, enter the port number of the redundant card. The Active Card will
communicate with the Redundant Card through this port.
Make sure the same port is configured in the active card.
• In LAN Port IP Address, enter the LAN Port IP Address of the Redundant Card. If you have selected LAN
Port as the Enable Configuration Sync. on option, the Active Card will use this IP to communicate with
the Redundant Card. The default IP Address is 192.168.002.101. You can assign only Static IP to the LAN
Port.
• In WAN Port IP Address, enter the WAN IP Address of the Redundant Card. If you have selected WAN
Port as the Enable Configuration Sync. on option, the Active Card will use this IP to communicate with
the Redundant Card. The default IP Address is 192.168.001.101. You can assign only Static IP to the WAN
Port.
• The Connection Status will display the connectivity status between the Active and Redundant card. It will
display — Connected or Not Connected accordingly.
Click Submit.
Make sure you have configured these parameters for the other CPU Card also by accessing it’s respective
IP Address.

Configuring Extensions
The SARVAM UCS supports the following types of extension ports:
• SIP Extensions: Any SIP-enabled device like a Matrix VARTA UC Client, an IP-phone, a Softphone or a
Wi-Fi mobile handset can be registered with the SARVAM UCS and function as the 'SIP Extension' of the
SARVAM UCS.
SIP Extensions function like any normal DKP/SLT extension of the SARVAM UCS, allowing you to make
and receive calls to any extension user of the SARVAM UCS as well as any external numbers over PSTN,
GSM, VoIP and E&M lines, depending on the “Logical Partition” configured in the System.
SARVAM UCS supports 999 SIP Extensions. SIP Extensions are a licensed feature. To know more, refer
the topic “License Management”.
• DKP Extension Ports: The proprietary Digital Key Phone of Matrix is connected to these ports. SARVAM
UCS supports up to 96110 DKP extensions. The number of DKP ports available to you depends on the
number and configuration of the DKP Cards installed in the system.
• ISDN Terminals: These are ISDN phones connected to the BRI Ports of the SARVAM UCS. ISDN
Terminals can be connected only in a Point-to-Multipoint BRI configuration (Short or Extended Passive
Bus Configuration). Refer the topic Installing BRI Card under Installation instructions for your model of
SARVAM UCS to know more.
A maximum of 8 ISDN Terminals (phones) can be connected on a single BRI Bus line in a Point-to-
Multipoint configuration. In a Short Passive Bus Configuration, you can connect up to 8 ISDN Terminals
while in the Extended Passive Bus Configuration you can connect up to 3 ISDN Terminals.
Depending on the number of BRI ports available to you and the type of Point-to-Multipoint Configuration
(Short or Extended Passive Bus), a maximum of 64 ISDN Terminals can be connected to the SARVAM
UCS.
• SLT Extension Ports: Single Line Telephones (SLT) is connected to these ports. SARVAM UCS supports
up to 240111 SLT extensions. The number of SLT ports available to you depends on the number and
configuration of the SLT Cards installed in the system.
• Radio Extensions/Ports: Radio devices are connected to these ports and function as extensions of the
SARVAM UCS. A maximum of 16 Radio Ports are supported.
• Magneto Ports: Magneto telephones are connected to these ports and function as extensions of the
SARVAM UCS. A maximum of 16 Magneto Ports are supported.
• E&M Ports functioning as Stations: An E&M port of SARVAM UCS can be programmed to take on the
function of a Subscriber (Station), to work like an extension interface, receiving incoming calls.
Presuming that you have connected the extensions successfully, you may now configure the extensions
using Jeeves or a Telephone.
110. The maximum number of DKP ports supported in ETERNITY MENX may vary according to the type of Power Supply (whether
PSUNI or DC) being used. See Technical Specifications provided in the Appendix.
111. The maximum number of concurrent off-hook SLT ports supported may vary according to the type of Power Supply (whether
PSUNI or DC) being used. See Technical Specifications provided in the Appendix.

• Virtual Extensions. For more information see “Virtual Extension”.
Templates for Configuring Extensions

To make the task of configuring of extensions easy, the SARVAM UCS offers Templates, namely:
• SIP Hardware Template - for SIP Extensions (and SIP Trunks) only. See “SIP Hardware Template” under
Configuring Trunks.
• SLT Hardware Template - for SLT extensions.
• Station Basic Feature Template - for DKP, SLT, SIP Extensions, ISDN Terminals, and E&M Ports
functioning as Stations, Magneto ports, Radio ports and Virtual Extensions.
• Station Advanced Feature Template - for DKP, SLT, SIP Extensions, ISDN Terminals, and E&M Ports
functioning as Stations, Radio ports, Magneto ports, and Virtual Extensions.
You can use these templates to program extensions which are to be assigned the same set of features at one go,
saving you the effort for painstaking configuration of each extension.
The features in these templates are loaded with default values that fulfill the requirements of a very broad user
base. The Templates may be customized as per user requirements and applied to the extensions.
Before you start the configuration of the extensions, please read the description of the templates and how to
customize the templates according to user requirements.
SLT Hardware Template

An SLT Hardware Template is a set of features that define the behavior of the SLT hardware port, and needless to
say, it is assigned to SLT Ports only.
The SLT Hardware Template allows you to configure according to user requirements, a common set parameters
(features) like Caller ID Presentation (DTMF, FSK), Digit Pad Count, Ring Type, AC Impedance, Answer Signaling
type, etc. to be assigned to all SLT Hardware Ports.
Each of these hardware parameters are briefly described below.
SLT Hardware Template Parameters

1. CLIP Type: The SARVAM UCS provides a facility to detect the calling number and present it to the SLT.
This is known as Calling Line Identification and Presentation (CLIP). For this feature to work, the telephone
instrument connected to the SLT port must support CLIP.
The SARVAM UCS supports 3 signaling protocols for CLI on the SLT port: DTMF, FSK-V.23, and FSK-
BellCore. Select the appropriate signaling protocol.
If you want to disable CLI on the SLT port, select 'None'.
By default, DTMF protocol is set as CLIP Type.
2. Digit Pad Count: Certain SLT instruments that support CLI require a minimum number of digits in the
calling party's number to be able identify and display it. The Digit Pad Count signifies the number of zeroes

to be added with the Calling party's number before displaying it on the called party's instrument. This count
entirely depends on the instrument connected to it. By default, Digit Pad Count is set as 0.
3. Ring Type: The SLIC used with SLT port allows you to change the Ring type: Sinusoidal, Trapezoidal,
Low Sinusoidal, Low Trapezoidal. This is helpful in cases when telephone instruments, which expect
sinusoidal type of ringing current, are connected to the SLT port. By default the Ring Type for all SLT ports
is Trapezoidal.
4. SLT Gain Settings Template: You can increase or decrease the level of Incoming Speech (Receive Gain)
and Outgoing Speech (Transmit Gain) on the SLT port by changing the Rx Gain and Tx Gain to the desired
level. Different levels can be set for each port type in the SLT Gain Settings Template. By default, SLT
Gain Template 1 is assigned to all the SLT Hardware Templates. If you want to assign a different
Template, you must customize the SLT Gain Settings Template first and then assign the number of the
SLT Gain Settings Template in this Template. To customize the SLT Gain Settings, see “Gain Settings”.
5. AC Impedance: The SLIC used with each SLT port provides a facility to adjust the AC impedance of the
SLT port with the communication equipment connected to it.
Generally, most telephone instruments that are connected have nominal characteristics with AC
impedance of 600. However, the SARVAM UCS allows you to connect instruments with AC impedance
other than 600.
6. Flash Timer (msec): In Pulse Dialing, codes are dialed in pulses. A Flash key is generally used to dial this
code. Flash is breaking the loop current for 70ms to 900ms. Flash Timer defines the time period which
should be considered as Flash, if the loop current breaks.
The range of the timer is from 70 to 900 msecs. By default, the Flash Timer is set to 101-600 msec.
Program the Flash Timer as per user requirement.
If the Flash Timer range is configured as 70-100 msec, Pulse dialing for that particular SLT Phone will not
work.
7. Answer Signaling: An Answer Signal is a signal generated on the SLT port to indicate that the called
party (remote party) has answered the call and the call is now mature.
Answer Signaling on the SLT port is particularly useful when there is a PCO machine or any Billing
equipment connected to the SLT port. With Answer Signaling enabled on an SLT port, during an outgoing
call is made from that SLT port to any other port - CO/Mobile/SIP/T1E1/BRI - when the called party (remote
party) answers, the Public Network provides an Answer Signal to the trunk port to indicate call maturity.
This information can be passed on to the PCO machine billing equipment in the form of Answer Signaling.
On detecting Answer Signaling the PCO machine billing equipment can start billing.
Answer Signaling is generated in the form of Polarity Reversal or Battery Reversal, whereby the Battery
polarity of the SLT port gets reversed. For example, if the battery polarity of the SLT port is +ve for TIP and
-ve for RING in speech condition, then on call maturity, TIP becomes -ve and Ring becomes +ve.
To generate Answer Signaling on the SLT Port, select Polarity Reversal. Select None if Answer Signaling
is not be generated on the SLT port.
By default None is selected.

8. Disconnect Signaling: A Disconnect Signal is the signal generated on the SLT port to indicate that the
called party (remote party has disconnected the call.
Disconnect Signaling on the SLT port is useful when there is a PCO machine or any Billing equipment
connected to the SLT port. With Disconnect Signaling enabled on an SLT port, during an outgoing call is
made from that SLT port to any other port - CO/Mobile/SIP/T1E1/BRI - when the called party (remote
party) disconnects (goes ON Hook), the Public Network provides a Disconnect Signal to trunk port indicate
call disconnection. This signal can be generated on the on the SLT port to indicate to the PCO machine/
Billing equipment connected to this port to consider the call as disconnected and stop billing. Thus,
Disconnect Signaling on the SLT port helps prevent excessive billing.
SARVAM UCS supports two types of Disconnect Signals on the SLT Port:
• Polarity Reversal: Call Disconnection is signaled in the form of Polarity Reversal. The Battery polarity
of the SLT port will be reversed. For example, if the battery polarity of the SLT port is '+ve' for TIP and '-
ve' for RING in speech condition then on disconnection on other port, TIP will become '-ve' and Ring
'+ve'. When call is disconnected, user will get Error tone.
• Open Loop: Call Disconnection is signaled in the form of Open Loop Disconnect Pulse, whereby the
Battery voltage on the SLT port is removed for the duration of the Open Loop Disconnect Timer
programmed for that SLT port and will be restored on the expiry of this Timer. However, the Polarity of
Battery Voltage on the SLT port is not changed. When call is disconnected, the SLT extension user
gets an Error tone.
To generate Disconnect Signaling on the SLT Port, select Polarity Reversal or Open Loop as
appropriate. Select None if Disconnect Signaling is not be generated on the SLT port.
9. Open Loop Disconnect Timer (msec): This parameter is applicable only if the option Open Loop
Disconnect is selected as Disconnect Signaling type on the SLT port.
Open Loop Disconnect Timer is the time period for which the system will remove Battery Voltage on the
SLT port and restore Battery Voltage on the expiry of the Timer to signal Call Disconnection.
The range of the timer is from 68 to 952 msecs. By default, the Timer is set to 478 msec.
10. Loop Current (mA): The SLT Port provides Loop Current to the telephone instrument connected to the
SLT port to drive the telephone instrument.
The Loop Current is to be increased/decreased according to the length of the telephony wiring cable
between the wall jack (into which the SLT telephone instrument is plugged) and the MDF (into which the
cables from the SLT port are terminated).
The longer the Loop Length of the SLT port, the greater the likelihood of current dissipation, affecting
speech quality of the telephone instrument connected to the SLT port.
The system supports Loop Current of 25, 30, 35 and 40 mA. By default the Loop Current for SLT ports is
set to 25mA and the SLT port, which is sufficient to support Loop Length of 1 kilometer.
You may change the Loop Current according to the Loop Length of the SLT.

11. Loop Length: The Loop Length is the distance between the Central Office and the telephone instrument
connected to the SLT port. Select the Loop Length as Upto 5 Km or Above 5 Km according to your
12. Minimum Current for OFF-Hook Detection: The system detects OFF-Hook state of an SLT instrument
and gives dial tone on the basis of the current drawn by it from the SLT port. However, all types and brands
of SLT instruments may not uniformly draw the same minimum current; some may draw lesser and some
may draw more, making OFF-Hook detection difficult for the system. To resolve this, SARVAM UCS
provides for programmable values for threshold current for OFF-Hook detection: 10mA, 12mA, 14mA,
16mA and 18mA.
By default, the value of the Minimum Current for OFF-Hook detection is set to 12 mA. Change this value
according to the current drawn by your SLT instrument.
When an SLT instrument draws current equal to or greater than the programmed threshold value of current
for off-hook detection, the system will consider the SLT instrument as OFF-Hook and will offer dial tone to
the SLT.
13. ON-Hook Detection Current (mA): SARVAM UCS detects ON-Hook state of an SLT instrument to route
calls on the basis of the current drawn by it from the SLT port. However, as all types and brands of SLT
instruments may not uniformly draw the same current, ON-Hook detection becomes difficult for the system.
To resolve this, SARVAM UCS provides for programmable values for threshold current for ON-Hook
Detection: 10mA, 12mA, 14mA, 16mA and 18mA.
By default, the value of the ON-Hook Detection Current is set to 10 mA.
When an SLT instrument draws current equal to or lower than the programmed threshold value of current
for ON-Hook detection, SARVAM UCS will consider the SLT instrument as ON-Hook.
SLT instruments also vary in the level of current drawn during the normal 'idle' state and when Flash is
dialed112 (the simulated idle state). So, when the Flash key of an SLT instrument is pressed, and if the
instrument draws a higher current than the threshold defined for the 'idle' state, the system will not be able
to detect Flash (ON-Hook state).
Consider this when changing the value of ON-Hook Detection Current. Define the value considering the
current drawn by your SLT instrument in idle state, as well as when Flash key is pressed.
14. Low Power Mode: SARVAM UCS supports Low Power Mode option. That is, in On-Hook state the SLT
instrument will consume less power. Select the Low Power Mode check box to enable this mode.
SARVAM UCS also supports Thermal Shutdown. If the thermal temperature of the SLIC increases above
the threshold level, the SLT instrument will stop functioning. This port number appears red in SLT
Parameters. You must restart the card or the system to resume functioning of the phone.
Customizing SLT Hardware Templates

There are 50 SLT Hardware Templates that can be customized and assigned to the SLT ports. These templates
contain the default values of the above-listed parameters.
112. Dialing 'Flash' either with the 'Flash Key' or by pressing the Hook-switch causes the phone to go in ON-Hook state briefly for 600-
800 milliseconds. Thus ON-Hook state is simulated briefly. The SLT may draw a higher current when 'Flash' is dialed.

The default parameter values of the SLT Hardware Templates are country specific and are loaded in each template
according to the Country selected as the Region.
For example, when India is selected as the Region, the default value of the CLIP Type in the SLT Hardware
Templates is DTMF, whereas it is FSK-Bellcore when the Region is selected as US or Canada and FSK-V.23 when
UK is selected as Region. Similarly, the default values of AC Impedance on the SLT Hardware Templates will vary
according to the Region selected; 600 ohms for Region India, 900 for Region the Philippines, and 350 +(1000
|| 0.21µF) for Region UK.
By default SLT Hardware Template Number 01 is assigned to all the SLT ports. This template has default values
fulfilling the common requirements of a very broad user base.
If the default SLT Hardware Template 01 fulfills the user and country requirements, retain Template 01.
If you want to change the values of certain SLT Hardware Parameters, but apply the same parameter values to all
SLT ports, simply customize the desired parameters in Template 01.
However, if different hardware parameters are to be applied to different SLTs, then you can customize different the
SLT Hardware Templates using Jeeves or a Telephone.
Customizing SLT Hardware Template using Jeeves

• Under Configuration, click SLT Configuration.
• Click SLT Hardware Template to open the page.
• Select a Template number you wish to customize, for example Template 03.

• Change the values of the SLT Hardware Template parameters as desired.
• Click Submit at the bottom of the page to save your changes.
• Now, apply this SLT Hardware Template 03 on the SLT ports. To do this,
• Click SLT Parameters to open the page.
• Go to the SLT software ports to which this Template is to be assigned, for example SLT-001, 002, and 003.

• Enter the number of the Template you customized, 02, in the field SLT Hardware Template of each of
these SLT ports.
• Click Submit at the bottom of the page to save your setting.
• Repeat the same steps to customize another template.
Customizing SLT Hardware Template using a Telephone

To change the default values of a SLT Hardware Parameter in a Template, dial:
• 5702-1-Template Number-Parameter Number-Code to program the value of a parameter in a single

template.
• 5702-2-Template Number-Template Number-Parameter Number-Code to program the same value

for the parameter in a range of templates.
• 5702-*-Parameter Number-Code to program the same value for the parameter in all templates.
Where,
Template Number is the number of the SLT Hardware Template from 01 to 50.
Parameter Number is the number of the SLT Hardware Template Parameter from 01 to 13.
Code is the value for each parameter from 0 to 14.
Refer the table below for parameter numbers and meaning of codes.
For example, to change the CLIP Type in Template 02 from default the DTMF to FSK-Bell, dial 5702-1-
02-01-3
Where,
02 is the template number
01 is the parameter number for CLIP Type
3 is the code for FSK-Bell

Default values of SLT Hardware Templates

• The default values of the SLT Hardware Templates are for the default Region India. The default values
will differ according to the Region you have selected for the system.
To default SLT Hardware Template, dial:

• 5701-1-Template Number to default a single template.
• 5701-2-Template Number-Template Number to default a range of templates.
• 5701-* to default all templates.
When the SE command to set the AC impedance for the SLT is issued, the settings will be effective when
the SLT extension (for which AC Impedance is programmed) goes ON-Hook. SARVAM UCS will restart, if
the SLT is OFF-Hook.
To apply the Customized SLT Hardware Template to SLT port, dial:

• 5703-1-SLT-Template Number to assign a hardware template to a single SLT port.
• 5703-2-SLT-SLT-Template Number to assign a hardware template to a range of SLT ports.
• 5703-*-Template Number to assign a hardware template to all SLT ports.
Where,
SLT is the Software Port number of the SLT port from 001 to 512.
Template Number is the number of the customized SLT Hardware Template, from 01 to 50. Default: 01.
For example, to assign Template 02 to SLT ports 003 to 010, dial 5703-2-003-010-02
• Exit SE mode.

Station Basic Feature Template
The Station Basic Feature Template is a set of general features that completely define the basic behavior of an
extension. Instead of programming each extension individually, the Station Basic Feature makes it possible to
group together extensions that are to be assigned the same set of features, prepare a Station Basic Feature
Template with the common set of features and apply it on these extensions.
The SARVAM UCS offers 50 such Station Basic Feature Templates. A Station Basic Feature Template is assigned
to all the types of extensions, namely SLT, DKP, ISDN Terminals, E&M113, T1E1PRI and BRI114, Radio Extensions,
Virtual Extensions.
These templates have commonly used values, but can be customized as per the requirement and applied on the
extensions.
Station Basic Feature Template Parameters

The Station Basic Feature Template contains the following features:
• Time Table: A Time Table is a schedule of the three Time Zones, namely: Working Hours, Break Hours,
Non-Working hours for a week.
Certain features of the SARVAM UCS like Operator, Class of Service, Toll Control, Outgoing Trunk
Access, among others, require the extension to behave differently in each Time Zone115.
So, a Time Table is assigned to extensions defining the Time Zones for the entire week, so that the system
can execute the Time Zone-dependent features and facilities according to the Time Table.
There are 8 different Time Table templates to select from. By default, the Time Table 1 is assigned to all
Station Basic Feature Templates. All seven days of the week are 'working hours 9:00 to 18:00' with break
hours from '13:00 to14:00'.
You may also customize the default Time Table 1 OR customize and assign a different Time Table to the
Station Basic Feature Template. Please refer the topic “Time Tables” for more details.
• Operator: Define the Operator for the extensions on which the template is applied.
The system supports multiple Operators. In each Time Zone one of the 20 Operators can be programmed.
Operator can be a single extension or a group of extensions, so that call management is more efficient. For
instance certain extensions may be assigned Operator 1, certain others Operator 2 and the rest may be
assigned Operator 3.
Operator 1 is the default in the Station Basic Feature Template. If you want to assign different extensions
to different Operators, you must program a separate Station Basic Feature Template with a different
Operator for each extension group.
Refer the topic “Configuring 'Operator'” to know more.
113. Applicable only when Orientation type is 'Station'.

114. Applicable only when the T1E1PRI and BRI ports are in NT mode.
115. For example, incoming calls are to be routed to the security personnel extension, instead of the Operator when the office is closed
(non-working hours), or certain features in the Class of Service are to be allowed only during working hours, or access to outgoing
long distance calls are to be denied during non-working hours, or the extension must play a different greeting message to the call-
ers during break hours and holidays (non-working hours).

• Class of Service: Class of Service (COS) defines the set features of the system that the extension is to be
allowed access to.
Not all extensions may require the same set of features. Some extensions may require voice mail, while
another group of extensions may need the ability to forward calls to a cell phone, and still others may have
no need to make calls outside the office.
Similarly, certain features may be required during working hours, but not during break or non-working
hours.
It is possible to assign a different Class of Service to different extensions according to their feature
requirements as well as according to the Time Zones.
By default COS group 01 is assigned to the Station Basic Feature Templates for all Time Zones. If you
want to assign a different COS for each Time Zone, you must customize the COS group first and then
assign the number of the COS group in the Template.
Refer the topic “Class of Service (COS)” to know more and for instructions on how to enable or disable a
feature in a COS group.
• Call Budget: This flag is for enabling the Call Budget feature. The Call Budget feature will allot a 'budget'
limit for outgoing calls made by extensions on which the template is applied. Refer “Call Budget on
Extension” for more details.
• Toll Control: This Toll Control Level allows you to define the Call Privilege (calling permission) to be
allowed to extensions according to the time of the day, during working hours (WH), break hours (BH) and
non-working hours (NH). For each Time Zone, you may define the calling permission to be allowed to
extensions by selecting the Type of Call Privilege.
Toll Control Level 0 (WH): This Toll Control Level allows you to define the Call Privilege (calling
permission) allowed to an extension during Working Hours.
• Call Privilege: Define the type of calling permission to be allowed to the extension during the Working
Hours. The call privilege options are: No Calls, Local Calls, Regional Calls, National Calls, International
Calls, All Calls, Limited Calls.
• Allowed List: For the type of Call Privilege you select, define the list of numbers to be allowed during
Working Hours.
• Denied List: For the type of Call Privilege you select, define the list of numbers to be denied during the
Working Hours.
Toll Control Level 0 (BH): This Toll Control Level allows you to define the Call Privilege (calling
permission) allowed to an extension during Break Hours.
• Call Privilege: Define the type of calling permission to be allowed to the extension during the Break
Hours. The call privilege options are: No Calls, Local Calls, Regional Calls, National Calls, International
Calls, All Calls, Limited Calls.
Break Hours.

Break Hours.
Toll Control Level 0 (NH): This Toll Control Level allows you to define the Call Privilege (calling
permission) allowed to an extension during Non-Working Hours.
• Call Privilege: Define the type of calling permission to be allowed to the extension during the Non-
Working Hours. The call privilege options are: No Calls, Local Calls, Regional Calls, National Calls,
International Calls, All Calls, Limited Calls.
Non-Working Hours.
Non-Working Hours.
• Toll Control Level 1: This Toll Control Level allows you to define the Call Privilege (calling permission) to
be allowed to an extension, regardless of Time Zone. By default Toll Control Level 1 is set to Local Calls.
be allowed to an extension, regardless of Time Zone. By default Toll Control Level 2 is set to National
Calls.
be allowed to an extension, regardless of Time Zone. By default Toll Control Level 3 is set to No Calls.
• OG-Trunk Bundle Group (WH): This is the Outgoing Trunk Bundle Group to be allotted to the extension
for Working Hours. The extension will be allowed to make outgoing calls through the trunks in this group.
• OG-Trunk Bundle Group (BH): This is the Outgoing Trunk Bundle Group to be allotted to the extension
for Break Hours.
• OG-Trunk Bundle Group (NH): This is the Outgoing Trunk Bundle Group to be allotted to the extension
for Non-Working Hours.
Refer the topic “OG Trunk Bundle Group” for more details.
• Store Outgoing Calls: You can enable or disable the storage of call details - Station Message Detail
Records - of Outgoing Calls landing on the extensions on which the template is applied. Select the check
box to enable and clear it to disable. Refer the topic “Station Message Detail Recording-Storage” for more
details.
• Store Incoming Calls: You can enable or disable the storage of call details - Station Message Detail
Records - of Incoming Calls landing on the extensions on which the template is applied. Select the check
box to enable and clear it to disable. Refer the topic “Station Message Detail Recording-Storage” for more
details.
Customizing Station Basic Feature Template using Jeeves


• Under Configuration, click Station Basic Feature Template to open the page.
• Change the values of the Station Basic Feature Template parameters as desired.
• Now, apply this Template 10 on the SLT/DKP/ISDN Terminals/SIP Extensions.

To apply the customized template on SLT ports,
• Go to the SLT software ports to which this Template is to be assigned, for example SLT-003 and 004.
• Enter the number of the Template you customized, 10, in the field Station Basic Feature Template of
each of these SLT ports.
To apply the customized template on DKP ports,
• Under Configuration, click DKP Configuration.
• Click DKP Parameters to open the page.
• Go to the DKP software ports to which this Template is to be assigned, for example DKP-005 to 008.

each of these DKP software ports.
To apply the customized template on ISDN Terminal ports,
• Under Configuration, click the link ISDN Terminal Parameters to open the page.
• Go to the ISDN Terminal software ports to which this Template is to be assigned, for example ISDN-01.
each of the ISDN Terminal software port.

To apply the customized template on SIP Extensions,
• Under Configuration, click VoIP Configuration.
• Click SIP Extension Settings.
• Go to the SIP Extensions, for example SIP Extension 1, to which this Template is to be assigned, and
enter the Template number.
• Click Submit to save your setting.
• Repeat the same steps to customize another template and apply it on the extension ports.
Customizing Station Basic Feature Template using a Telephone

To program a feature in a Station Basic Feature Template, dial:

• 5502-1-Template Number-Feature Number-Code to program the value of a feature in a single
template.
• 5502-2-Template Number-Template Number-Feature Number-Code to program the same value for
a feature in a range of templates.
• 5502-*-Feature Number-Code to program the same value for a feature in all templates.
Where,
Template Number is the number of the Station Basic Feature Template, from 01 to 50.

Feature Number is the number of the Feature in the Template, from 01 to 39.
Code is the parameter value of the Feature.
For example, you want to customize Template number 10, by enabling Call Budget feature and select
'Local Calls' as the Toll Control when Call Budget Consumed.
Dial 5502-1-10-06-1 to enable Call Budget feature (number 06) on Template 10.
Dial 5502-1-10-19-1 to select Local Calls (1) as Toll Control when Call Budget Consumed (19) in
Template 10.
Refer the following Table for the Feature Number and Codes for the Station Basic Feature Templates.
To default a Station Basic Feature Template, dial:

• 5501-1-Template Number to default the values of a single template.
• 5501-2-Template Number-Template Number to default the values of a range of templates.
• 5501-* to default the values of all templates
To assign a Station Basic Feature Template to SLT, dial:

• 5503-1-SLT-Template Number to assign the template to a single SLT port.
• 5503-2-SLT-SLT-Template Number to assign the same template number to a range of SLT ports.
• 5503-*-Template Number to assign the same template number to all SLT ports.
Where,
SLT is the number of the Software port of the SLT, from 001 to 512.

Default: Template 01 is assigned to all SLT ports.
For example, to assign Station Basic Feature Template number 10 to the SLT software ports 004 to
010, dial 5503-2-004-010-10.
To assign a Station Basic Feature Template to a DKP, dial:

• 5504-1-DKP-Template Number to assign a template to a single DKP.
• 5504-2-DKP-DKP-Template Number to assign the same template to a range of DKPs.
• 5504-*-Template Number to assign the same template to all DKPs.
Where,
DKP is the number of the Software port of the DKP, from 001 to 128.
Default: Template 01 is assigned to all DKP ports.
For example, to assign Station Basic Feature Template 10 to DKP software ports DKP-005 to 010, dial
5504-2-005-010-10
To assign a Station Basic Feature Template to an ISDN Terminal port, dial:

• 5507-1-ISDN-Template Number to assign a template to a single ISDN Terminal port.
• 5507-2-ISDN-ISDN-Template Number to assign the same template to a range of ISDN terminal ports.
• 5507-*-Template Number to assign the same template to all ISDN terminal ports.
Where,
ISDN Terminal is the number of the ISDN Terminal Software port, from 01 to 64.
Default: Template 01 is assigned to all ISDN Terminals.
For example, to assign Station Basic Feature Template number 10 to the ISDN Terminal Software
ports 01 to 04, dial 5507-2-01-04-10
To assign a Template to an E&M (Station) port, dial:

• 5505-1-E&M-Template Number to assign a template to a single E&M port.
• 5505-2-E&M-E&M-Template Number to assign the same template to a range of E&M ports.
• 5505-*-Template Number to assign the same template to all E&M ports.
Where,
E&M is the number of the E&M Software port, from 001 to 128.
Default: Template 01 is assigned to all E&M (Station) ports
To assign a Station Basic Feature Template to a T1E1PRI port, dial:

• 5506-1-T1E1PRI-Template Number to assign a template to a single T1E1 port.
• 5506-2-T1E1PRI-T1E1PRI-Template Number to assign the same template to a range of T1E1 ports.
• 5506-*-Template Number to assign the same template to all T1E1 ports.
Where,
T1E1PRI is the number of the T1E1PRI Software port, from 01 to 08.
Default: Template 01 is assigned to all T1E1 ports.
To assign a Station Basic Feature Template to a BRI port, dial:

• 5509-1-BRI-Template Number to assign a template to a single BRI port.
• 5509-2-BRI-BRI-Template Number to assign the same template to a range of BRI ports.
• 5509-*-Template Number to assign the same template to all BRI ports
Where,

BRI is the number of the BRI Software port, from 01 to 32.
Template is the number of the Station Basic Feature Template, from 01 to 50.
Default: Template 01 is assigned to all BRI ports.
To assign a Station Basic Feature Template to a Virtual Extension, dial:

• 5513-1-Virtual Extension-Template Number to assign template to a single Virtual extension.
• 5513-2-Virtual Extension-Virtual Extension-Template Number to assign the same template to a
range of Virtual extensions.
• 5513-*-Template Number to assign the same template to all the Virtual extensions.
Where,
Virtual Extension is from 01 to 64.
Station Basic Feature Template is from 01 to 50.
By default, Template 01 is assigned.
To assign a Station Basic Feature Template to a Magneto Port, dial:

• 5511-1-Magneto Port-Template Number to assign template to a single Magneto Port.
• 5511-2-Magneto Port-Magneto Port-Template Number to assign the same template to a range of
Magneto Port.
• 5511-*-Template Number to assign the same template to all the Magneto Port.
Where,
Magneto Port is from 001 to 128.
By default, Template 01 is assigned.
• Exit SE mode.
Station Advanced Feature Template

The Station Advanced Feature Template is a set of advanced extension features, to be applied on extensions to
support features like CLIP, Call Duration Control, Storage of Internal SMDR, Call Taping, Alarm Notification, Floor
Service, Walk-In Class of Service, etc.
Instead of programming each extension individually, the Station Advanced Feature makes it possible to group
together extensions that are to be assigned the same set of features, prepare a Station Advanced Feature
Template with the common set of features and apply it on these extensions.
The SARVAM UCS offers 50 such Station Advanced Feature Templates. A Station Advanced Feature Template is
assigned to all the types of extensions viz. SLT, DKP, ISDN Terminals, E&M116, T1E1PRI and BRI117, Radio Virtual
and SIP.
These templates have commonly used values, but can be customized per the requirement and applied on the
extensions.
Station Advanced Feature Template Parameters

The Advanced Feature Template comprises the following features on extensions:
• Caller ID Presentation while Transfer: This parameter is related to the CLIP feature. It allows you to
select whether you want the CLI of the ‘Held Party’ or the CLI of the ‘Transferring Party’ to be displayed to
the transfer destination extension while the call is being transferred. Refer the feature description for
“Calling Line Identification and Presentation (CLIP)” to know more.
116. Applicable only when Orientation type is 'Station'.

117. Applicable only when the T1E1PRI and BRI ports are in NT mode.

• Call Forward No-Reply Timer: This parameter is related to the Call-Forward-No Reply/Preset Call
Forward No reply feature. The Timer is the duration for which the system will wait for an extension to
answer an incoming call, before forwarding the call it to the programmed destination phone number as Call
Forward-No Reply. By default the Timer is set to 30 seconds. Refer the feature description for “Call
Forward” and “Preset Call Forward” to know more.
• Preset Call Forward: This parameter is relevant when users do not want to set/cancel Call Forward from
their extensions. You can enable the desired type of Preset Call Forward — When Busy, When No Reply
or When Busy or No Reply and the calls will be forwarded to the selected destination— Voicemail,
Extension or Department Group. You can set Preset Call Forward for each time zone—Working Hours,
Break Hours and Non-working Hours. Refer the feature description of “Preset Call Forward” to know more.
• DDI IC Routing: This flag is relevant only if you are using DDI IC Routing. As this flag is enabled by
default, hence all the extensions configured in the IC Reference Table will behave as DDI Extensions. If
you do not want any of these extensions to function as a DDI extensions, disable this flag in the feature
template applied on that extension. By default, the flag is enabled.
Refer the topics “Direct Dialing-In (DDI)” to know more.
• Send DDI Number as CLI?: This flag is relevant only if the extension is programmed as a DDI Extensions
(the DDI IC Routing is enabled). You can choose whether to the Calling Line Identification (CLI) of the DDI
extension should be sent for outgoing calls made from that extension. By default the flag is enabled.
• Internal Calls Storage: This parameter is related to the storage of Station Message Detail Records of
internal calls, made between extensions of the SARVAM UCS. You can select the type of internal calls to
be stored by the system: i) Calls made from the extension, ii) Calls made to the extension, iii) Calls made
from as well as calls made to the extension, and iv) no internal calls. By default, Calls made from as well as
calls made to the extension are stored. Select desired type of internal call storage. Refer “Station Message
Detail Recording-Storage” to know more.
• Walk-Out Mode: This parameter is related to the feature Walk-In Class of Service. SARVAM UCS offers
two types of Walk-In: i) One-Call per Walk-In, whereby the user is automatically logged out after a call. ii)
Walk-In until Logout, whereby the user remains logged on until s/he manually walks out or a second user
walks into the same extension.
You must select the Walk-Out mode for the extension. If One-Call per Walk-In is to be supported on the
extension, select 'One Call' as Walk-Out mode.
If Walk-In until Logout is to be supported on the extension, select 'Multiple Calls' as the Walk-Out mode.
To know more about this feature, refer “Walk-In Class of Service”.
• CDC Table: This parameter is to be programmed if you have enabled the “Call Duration Control (CDC)”
feature on the extension. The system will check the Call Duration Control (CDC) Table applied to the
extension to implement this feature on the extension. So, you must first program the CDC Table and enter
the number of the CDC Table you have programmed in this field.
You can program 8 different CDC tables. By default, CDC Table No. 1 is assigned to all extensions. If CDC
is to be applied on extensions of the SARVAM UCS, simply program the default CDC Table No. 1.

To do this, click the link CDC Table to open the page. Program the CDC Table parameters and 'Submit' to
save your settings. Now return to the Station Advanced Feature Template and enter the number of the
CDC Table you programmed in the CDC Table field of the template.
Refer the feature description “Call Duration Control (CDC)” to know more and for instructions on creating
CDC Tables.
• Forced Account Code: This flag used to enable or disable the feature Forced Account Code on the SLT/
DKP/ISDN Terminal extensions. When this flag is enabled, the system will allow the extension user to dial
an external number only after entering the Account Code. Refer the feature description for “Account
Codes” to know more.
• Department Billing Group: This parameter enables you to know the total cost of the calls made by a
particular group of extensions. This parameter is used as a one of the filters for printing SMDR Reports
namely, “Print outgoing calls department group wise”. To be able to use this filter, you must assign the
extensions to a Department Bill Group. You can create as many as 99 different Department Bill Groups.
Enter the number of the Bill Group you want to assign the extensions to in this field.
• Floor Service Group: This parameter is related to the Floor Service feature. Floor Service can be floor-
wise or centralized. Floor Service requires you to program Routing Groups as landing destinations for
extension calls.
Program the Floor Service (Routing) Group first and enter this Floor Service (Routing) Group number in
this field. There are 96 different Routing Groups to be programmed as Floor Service Groups. By default,
no Routing Group is assigned to Floor Service in the Template ('00').
To know more about this feature, refer the feature description for “Floor Service”.
Calls from the extension will land on the Floor Service (Routing) Group you have assigned in this field.
• Alarm Notification Type: This parameter is related to the Alarm feature of the SARVAM UCS. You can
select any from the following options for Alarm Notification to extension users for Alarm calls:
• Voice Message: Extension users will be played a message recorded in the Voice Module, when they
answer the alarm call. By default, this option is selected.
• Music-on-Hold: Extension users will be played music-on-hold when they answer the alarm call.
• Voice Mail: The Extension users will be played the message recorded in the VMS, when they answer
the alarm call. You must have the VMS Module installed in your system if you want to select this option.
• Routing Group: Extension users will be connected to the extensions programmed in the 'Alarm
Notification Group'. For this you must have programmed a Routing Group.
If you select this option, you can also connect external Messaging Devices to play real-time updated
information like date, time, greetings, weather information, specific event announcements, etc., when
then extension users answer the alarm calls. The messaging devices must be connected to any of the
SLT ports and included in the Routing Group for alarm notification.
• Alarm Notification Routing Group: Program this parameter if you have selected 'Routing Group' as the
Alarm Notification Type. Enter the number of the Routing Group you have programmed for Alarm Calls.

By default Routing Group 31 is assigned as the Alarm Notification Routing Group. If the same Routing
Group is to be assigned to all extensions, click the link 'Alarm Notification Routing Group' to open the
Routing Groups page. Select members (extensions) in this routing group. Save your changes by clicking
'Submit' button.
You can program a different Routing Group repeating these steps. Make sure to enter the number of the
Routing Group you programmed in this field.
Refer the feature description for “Alarms” to know more.
• Help Desk: Enable this flag if you want to define the extension as a “Help Desk”. When this flag is
enabled, Auto Call Back will be automatically set whenever this extension is found busy.
• GPAX - Charge Internal Calls: This parameter is related to the GPAX application. If the extension is
programmed as a GPAX user, enable this flag for billing internal calls made by the extension. When this
flag is enabled, the system will record all calls made from the extension in the Station Message Detail
Record-Outgoing buffer for “GPAX Billing”. If the flag is disabled the calls will not be billed and will be
recorded in the Station Message Detail Record - Internal buffer as an internal call.
• Call Taping: If you want to use the “Call Taping” feature on the extension, you must program the related
parameters described below.
• Tape Calls coming without CLI: Enable this flag if you want incoming calls without CLI to be taped.
By default, the flag is disabled. The system will not tape incoming calls without CLI.
• Number List-Incoming Calls: Assign a Number List containing numbers of Incoming Calls that must
be taped. You must first program the Number List. By default, Number List 09 is assigned.
• Number List-Outgoing Calls: Assign a Number List containing numbers of Outgoing Calls that must
be taped. You must first program the Number List. By default, Number List 10 is assigned also for
outgoing calls.
If Number list 10 is already used for another application, prepare a different number list and assign it to
the template.
• Call Taping for Internal Calls: Enable this flag if you want to allow Call Taping of internal calls made
and received by the extension.
• Caller Category: Program this parameter only:
a. if the extensions are connected to T1 Trunks using RBS protocol.
b. If you want to define extensions of the systems networked over E&M lines using MFCR2 Signaling as
either 'Priority Subscriber' or 'Ordinary Subscriber'.
Select the appropriate extension category from the following options:

• Ordinary Subscriber (default)
• Priority Subscriber
• Maintenance Equipment
• Operator
• Pay Phone
• Data Transmission
• Interception Operator

• Allow External Call Forward for: This parameter defines the types of calls for which the External Call
Forwarding is to be applied. This parameter is relevant for the features “Call Forward” and “Mobility
Extension”. You may select from the following options:
• Internal Calls Only
• Trunk Calls Only
• Internal + Trunk Calls.
• Intercept Destination for DND: This parameter defines the destination extension to route incoming calls
landing on the DND set extension. You may select the desired destination for all three time zones —
Working Hours (WH), Break Hours (BH) and Non-working Hours (NH). The Destination can be a
Voicemail or any extension (SLT, DKP, SIP). If you select the Destination as an extension, enter the
desired Port Number (Port No.). For details, see “Do Not Disturb (DND)”.
• Route Global Directory Calls using: This parameter decides the OGTBG to be used to route the Global
Directory number. You may select either OGTBG configured in the Global Dir. or any other OGTBG
from 01 to 32.
• SMS for OG Call - No Reply: This parameter decides whether to send an SMS or not to the called party
for the calls that have not been answered. For more details, see “SMS on No Reply”.
Customizing Station Advanced Feature Template using Jeeves

• Under Configuration, click Station Advanced Feature Template to open the page.

• Change the values of the Station Advanced Feature Template parameters as desired.
• Now, apply this Template 02 on the SLT/DKP/ISDN Terminal ports, and Virtual Extensions.
To apply the customized template on SLT ports,
• Go to the SLT software ports to which this Template is to be assigned, for example SLT-003 and 004.

• Enter the number of the Template you customized, 02, in the field Station Advanced Feature Template
of each of these SLT ports.
To apply the customized template on DKP ports,
• Under Configuration, click the link DKP Configuration.
• Go to the DKP software ports to which this Template is to be assigned, for example DKP-005 to 008.Enter
the number of the Template you customized, 02, in the field Station Advanced Feature Template of each
of these DKP software ports.

To apply the customized template on ISDN Terminal ports,
• Under Configuration, click ISDN Terminal Parameters to open the page.
• Go to the ISDN Terminal software ports to which this Template is to be assigned, for example ISDN-01.
each of this ISDN Terminal software port.
• Repeat the same steps to customize another template and apply it on other extension ports.
Customizing Station Advanced Feature Template using a Telephone

To program a feature in a Station Advanced Feature Template, dial:

• 5602-1-Template Number-Feature Number-Code to program the value of a feature in a single
template.
• 5602-2-Template Number-Template Number-Feature Number-Code to program the same value for
a feature in a range of templates.
• 5602-*-Feature Number-Code to program the same value for a feature in all templates.
Where,
Template Number is the number of the Station Advanced Feature Template, from 01 to 50.
Feature Number is the number of the Feature in the Template, from 01 to 20.
For example, you want to customize Template number 02, by changing the default settings of the
following features:

• Call Forward No Reply Timer to set to 45 seconds, dial 5602-1-02-02-045 (Template 02, Feature
Number 02 for No Reply Timer, Code 045 for 45 seconds).
• Forced Account Code Flag to be enabled, 5602-1-02-08-1 (Template 02, Feature Number 08 for
Forced Account Code, Code 1 for enable flag).
Refer the following Table for the Feature Number and Codes for the Station Advanced Feature Templates.
To default a Station Advanced Feature Template, dial:

• 5601-1-Template Number to default the values of a single template.
• 5601-2-Template Number-Template Number to default the values of a range of templates.
• 5601-* to default the values of all templates
To assign a Station Advanced Feature Template to SLT, dial:

• 5603-1-SLT-Template Number to assign the template to a single SLT port.
• 5603-2-SLT-SLT-Template Number to assign the same template number to a range of SLT ports.
• 5603-*-Template Number to assign the same template number to all SLT ports.
Where,
SLT is the number of the Software port of the SLT, from 001 to 512.

Default: Template 01 is assigned to all SLT ports.
For example, to assign Station Advanced Feature Template number 02 to the SLT software ports 004
to 010, dial 5603-2-004-010-02
To assign a Station Advanced Feature Template to a DKP, dial:

• 5604-1-DKP-Template Number to assign a template to a single DKP.
• 5604-2-DKP-DKP-Template Number to assign the same template to a range of DKPs.
• 5604-*-Template Number to assign the same template to all DKPs.
Where,
DKP is the number of the Software port of the DKP, from 001 to 128.
Default: Template 01 is assigned to all DKP ports.
For example, to assign Station Advanced Feature Template 02 to DKP software ports DKP-005 to 010,
dial 5604-2-005-010-02
To assign a Station Advanced Feature Template to an ISDN Terminal port, dial:

• 5607-1-ISDN-Template Number to assign a template to a single ISDN Terminal port.
• 5607-2-ISDN-ISDN-Template Number to assign the same template to a range of ISDN terminal ports.
• 5607-*-Template Number to assign the same template to all ISDN terminal ports.
Where,
ISDN is the number of the ISDN Terminal Software port, from 01 to 64.
Template Number is the number of ~Template number is the number of the Station Advanced Feature
Template, from 01 to 50.
Default: Template 01 is assigned to all SIP extensions.
To assign a Station Advanced Feature Template to a Virtual Extension, dial:

• 5613 - 1 - Virtual Extension - Station Advance Feature Template to assign the template to a single
extension.
• 5613-2-Virtual Extension-Virtual Extension-Station Advance Feature Template to assign the same
template to a range of extensions.
• 5613-*- Station Advance Feature Template to assign the same template to all extensions.
Where,
Station Advance Feature Template is the number of the Station Advanced Feature Template, from 01
to 50.
Default: Template 01 is assigned to all Virtual Extensions.
To assign a Station Advance Feature Template to a Magneto Port, dial:

• 5611-1-Magneto-Station Advanced Feature Template Number to assign a template to a single
Magneto port.
• 5611-2-Magneto-Magneto- Station Advanced Feature Template Number to assign the same
template to a range of Magneto ports.
• 5611-*- Station Advanced Feature Template Number to assign the same template to all Magneto
ports.
Where,
Magneto is the Software Port number of the Magneto port from 001 to 128.
Station Advanced Feature Template is from 01 to 50.
Default: Station Advanced Feature Template Number 01.
• Exit SE mode.

Configuring SIP Extensions
SARVAM UCS supports 999 SIP extensions. You can:
• Connect SPARSH VP248, the Extended IP Phone for SARVAM UCS supplied by Matrix.
• Connect SPARSH VP330, the Touch Screen Extended IP Phones for SARVAM UCS supplied by Matrix.
• Connect SPARSH VP310, the Executive IP Phone for SARVAM UCS supplied by Matrix.
• Connect SPARSH VP510, the Premium IP Phone for SARVAM UCS supplied by Matrix.
• Register the Matrix VARTA ADR100 and VARTA AMP100 UC Clients for Mobile with SARVAM UCS.
• Register the MATRIX VARTA WIN200 Desktop UC Client with SARVAM UCS.
• Connect Matrix Open SIP Phones.
• Connect any other Open SIP phone or SIP enabled device, such as an IP phone, a Soft phone, an Analog
phone adapter.
To know more about Extended IP Phones, Mobile and Desktop UC Clients, see “Extended IP Phone/VARTA UC
Client - Operation”.
SARVAM UCS supports interoperability with the standard IP Phones. For a list of IP phones on which
various features of SARVAM UCS have been tested, see “SARVAM UCS Features tested on IP Phones of
different Brands” in the Appendix.
SIP Extensions function like any normal DKP/SLT extension of the SARVAM UCS. SIP Extensions can make and
receive calls to any extension user of the SARVAM UCS as well as to external numbers over PSTN, GSM, VoIP
and E&M lines, depending on the “Logical Partition” configured in the System.
SIP Extensions (IP Subscribers) are a licensed feature. Decide the number of IP Subscribers you will
require and buy the license. Refer the topic “License Management” to know more.
You can register a SIP Extension at three different locations as a single SIP Extension for Call Forking.
Configuring SIP Extension using Jeeves

You need to configure the following parameters for SIP Extensions:
• SIP Extension General Parameters, see “Configuring SIP Extension General Parameters”.
• SIP Extension Settings and the Extended Phone Settings as per the type of Extended Phone you have
connected, see “Configuring SIP Extension Settings as per the Extended Phone Type”.
• Voice Mail Settings, if you want to provide mailbox to the SIP extensions. See “Extension Voice Mail
Settings”.
Configuring SIP Extension General Parameters


• Click SIP Extension General Parameters link.
• Set the following SIP Extension General Parameters, as required.
• As the Source Port IP Address, select the NAT Traversal mechanism for SIP messages from the
following options:
• Use VoIP Ethernet Port IP Address: Select this option, if your system is not located behind a NAT
Router.
• Use IP Address fetched using STUN: Select this parameter, if your system is located behind a
NAT Router, and you have set 'Use IP Address fetched using STUN' as the NAT Traversal
mechanism in the “Configuring VoIP Parameters”.
• Use Router's Public IP Address: Select this option, if your system is located behind a NAT Router,
and you have set 'Router's Public IP Address' as the NAT Traversal mechanism in the “Configuring
Network Parameters”.
• You may set the Maximum Registration Timer (sec) as required. This is the Maximum Expiry Timer,
which the system will accept in the REGISTER request received. If the value of Maximum Expiry Timer
received in the REGISTER request is greater than the value you have set here, the system will send
the value you have set in the SIP message. The same timer is used for handling SUBSCRIBE
requests. The valid range of this timer is from 10 to 99999 seconds. By default it is set to 3600
seconds.
• You may set the Minimum Registration Timer (sec), as required. This is the Minimum Expiry Timer,
which the User Agent should send in its REGISTER request. If the expiry value in the REGISTER
message is less than this value, the request will be rejected. The valid range of this timer is from 10 to
99999 seconds. By default, it is set to 45 seconds.
• In the Private Key field, enter the MD5 authentication key of SARVAM UCS should use to encrypt/
decrypt the SIP messages. The Private Key may consist of a maximum of 24 characters. By default,
the field is blank.
• Click Submit to save your settings.

• You can restore the default values of any one or all the parameters on this page by clicking the Default
One and the Default button respectively.
Configuring SIP Extension Settings as per the Extended Phone Type

• If you have registered the Matrix Mobile UC Clients as SIP Extensions, for configuration instructions see
“Configuring Matrix VARTA ADR100/AMP100 UC Clients”.
• If you have registered the MATRIX VARTA WIN200 Desktop UC Client as SIP Extensions, for
configuration instructions see “Configuring Matrix VARTA WIN200 UC Client”.
• If you have connected the Matrix SPARSH VP248 as SIP Extensions, for configuration instructions see
“Configuring Matrix SPARSH VP248”.
• If you have connected Open SIP Phones or SIP enabled devices as SIP Extensions, for configuration
instructions see “Configuring Open SIP Phones”.
Viewing SIP Extension Status

You can view the Status of SIP Extension using Jeeves. To do this,

• Click SIP Extension Status.
• The SIP Extension Status page will open and display the following for each SIP Extension,
• SIP Extension number
• VoIP Ethernet Port Number to which the SIP Extension is assigned.
• Name of the SIP extension
• SIP ID assigned to the SIP Extension
• REGISTRATION status; whether the SIP Extension is registered or not.
• Contact 1
• Contact 2
• Contact 3
• You may Log out of Jeeves.
You can also view the SIP Extension Status from the Status link. To view, click the SIP Extension link
under Status.

Configuring Matrix SPARSH VP248
SPARSH VP248118, the proprietary SIP-based IP Phone for SARVAM UCS, supplied by Matrix, is a feature-rich
phone, providing voice communication over IP network. It functions like EON48, the proprietary digital key phone of
Matrix. To know the list of features supported, refer to “SARVAM UCS Features Supported in Terminals”.
For instructions on how to use SPARSH VP248, refer to the common EON48_310_SPARSH VP248_310 User
Guide.
To be able to use SPARSH VP248 - Extended IP Phone, you must configure the following:
• SIP Extension Settings, see “Configuring SIP Extension Settings using Jeeves”
• Extended IP Phone Settings, see “Configuring Matrix Extended Phone Settings using Jeeves”
• Voice Mail Settings, if you want to provide mailbox facility to the extension. See “Extension Voice Mail
Settings”.
Configuring SIP Extension Settings using Jeeves

The page of SIP Extension 001 opens.
118. SARVAM UCS supports only IPv4 Addresses for registering SPARSH VP248.

• You may select the SIP Extension number you want to configure.
The parameters of the SIP Extension number you selected will appear on this page.
For SARVAM UCS upto 999 SIP Extensions can be registered with the system. SARVAM UCS supports
IPv4 Addresses only for registering SIP Extensions.
• Select the Use SIP Extension check box to enable the SIP extension. Default:disabled.
• In the Name field, enter a name for the SIP Extension, which may be the name of the person who will use
the SIP Extension or the name of a Department. The name you enter here will be displayed as the Caller
ID of the SIP Extension on the remote user's phone, when the SIP Extension user makes calls.
The name may consist of a maximum of 18 alphanumeric characters.
• Enter the SIP ID for the extension. The SIP ID is necessary for registering the SIP Extension with the
Registrar of the system. It is the number with which you can call the SIP Extension. Any extension user of
the SARVAM UCS can call a SIP Extension by dialing the SIP ID assigned to the SIP extension. SIP ID of
each SIP Extension must be a unique number string of a maximum of 6 digits. Any combination of digits
from 0 to 9 and the characters * and # are allowed.You cannot assign the same SIP ID to more than one
extension.
To assign SIP IDs according to your preference and requirement to a range of SIP Extensions, see
“Assigning Access Codes to a Range of Extensions”.
By default, the SIP IDs are Blank.
The SIP ID will be set to default value (blank), when you restore the default settings of the system.
• In Authentication ID, enter the number which you want the system to use for user authentication of the
SIP messages received from the SIP Extension. You cannot keep this field blank. The number may be a
string of maximum 6 alphanumeric characters. All ASCII characters except < > and “ (double quote) are
allowed. Default: Blank.
• In Authentication Password, enter the password to be used by the system to authenticate the SIP
messages received from the SIP Extension. To avoid unauthorized access, we recommend you to change
the password regularly. Make sure it is strong and is kept confidential. The password must:
Default: Blank.
To provide additional security, when the Authentication fails 10 times consecutively due to wrong
Authentication ID / Authentication Password, the system will blacklist the IP Address and Port for
registration of this SIP Extension. However, you can remove the IP Address from the Blacklist IP Address
list. See “Black List IP Address - SIP Extensions” for more details. This activity will be logged in the
“System Activity Log” as well as “Simple Network Management Protocol (SNMP)”.

• In Call Appearances, define the maximum number119 of simultaneous incoming calls that the SIP
Extension user should be allowed to receive. You can set up to 10 call appearances for a SIP Extension.
Default: 2.
When Call Appearance is set to 2, the SIP Extension can receive 2 calls at a time.
• During an on-going conversation, if there is a second incoming call, the system plays beeps to indicate the
second incoming call. You can set the frequency of the Call Waiting Tone (for SPARSH VP248/VP310/
VP510) beeps as per your requirement. You can select from the following options:
• Off
• Beep Once
• Beep until Answered
Default: Beep Once
• Under Authentication, enable Authentication of any or all of the following SIP Message Options by
selecting the respective check boxes:
• INVITE Request
• SUBSCRIBE Request
By default, the SIP Message Options INVITE and SUBSCRIBE are enabled.
Make sure that the Authentication ID for the SIP Extension has been programmed.
• For secure conversations over SIP, enable SRTP Mode. The SARVAM UCS supports the following
options:
• Disable: SARVAM UCS uses normal RTP for transporting the speech packets.
• Optional: SARVAM UCS uses SRTP for transporting the speech packets. If the remote user does not
support SRTP, SARVAM UCS will use normal RTP for transporting the speech packets.
• If you select this option, you must configure the SRTP Media Type. You may select AVP or SAVP.
By default, AVP is selected as the SRTP Media Type.
• Forced: SARVAM UCS uses only SRTP (SAVP) for transporting the speech packets. If the remote
user does not support SRTP, SARVAM UCS will reject incoming calls from and drop outgoing calls
made to such users.
By default, SRTP Mode is Disabled.
• Assign a SIP Hardware Template to the SIP Extension. Default: 01. The “SIP Hardware Template”
contains voice quality related features such as Voice Codec selection, Tx and Rx Gains, Echo
Cancellation, Jitter Buffer and Fax-over-IP options and related parameters
There are 32 different templates to choose from. Each template can also be altered to suit your
requirement and preferences. By default, Template number 01 assigned to all SIP Extensions as well as to
SIP Trunks.
Check if the values in this template fulfill requirements of the SIP Extension. If Template 01 fulfills the
feature requirements, retain Template 01.
119. For the calls that are routed through the CPU, the number of Vocoder channels that will be supported would be as per the license
you purchase.

If a different set of SIP hardware features are to be allowed to this SIP Extensions, prepare another
template and assign it to this extension. To do this,
• Under VoIP Configuration, click the SIP Hardware Template link.
• Select a Template number, for example 02.
• Customize Template number 02 and click Submit.
• Return to the SIP Extension Settings page.
• Select the number of the Template you customized, Template 02 in the SIP Hardware Template
field.
Also see the topic “SIP Hardware Template” to know more about customizing the templates and
applying on the SIP Extensions.
• Assign a Station Basic Feature Template to the SIP Extension. Default: The “Station Basic Feature
Template” has a set of features like Time Table, Class of Service, Toll Control, Operator, Storage of
Incoming and Outgoing Calls, Outgoing Trunk Bundle groups. There are 50 different templates to
choose from. Each template can also be altered to suit your requirement and preferences.
If the default Station Basic Feature Template 01 fulfills the feature requirements of the SIP Extension
(“Class of Service (COS)”, “Toll Control”, “OG Trunk Bundle Group”, etc.) retain this template, you may
also customize this template. If you want to assign a different set of features to this SIP Extension,
prepare a different Station Basic Feature Template and apply it to this extension. To do this,
• Under Configuration, click the Station Basic Feature Template link.
• Select the number of the Template you customized, Template 05, in the Station Basic Feature
Template field.
• Click Submit to save changes.
Also, see the topic “Station Basic Feature Template” to know more about customizing the templates
and applying on extensions.
• Assign a Station Advanced Feature Template to the SIP Extension. Default: Template 01. The
“Station Advanced Feature Template” has a set of advanced features for extensions such as Alarm
Notification settings, Routing of Incoming Auto Attendant Calls, Call Duration Control, Floor Service,
etc. There are 50 different templates to choose from. Each template can also be altered to suit your
requirement and preferences.
Check if the default template fulfills the feature requirements of the SIP Extension by clicking the
Station Advanced Feature Template link.
You may retain this template and customize it further, or customize another template if a different set of
features are to be allowed to this SIP Extension. To customize/prepare another template,

• Under Configuration, click the Station Advanced Feature Template link.
• Select the Template number, for example 02, and customize this template.
• Click Submit and return to the 'SIP Extension Settings' page.
• In the Station Advanced Feature Template field, select the number of the template you
customized.
Also see the topic “Station Advanced Feature Template” for instructions on customizing these
templates and applying them on the extensions.
• In Voice Mail Settings, click the Voice Mail Settings link. The respective Extension Voice Mail Settings
window will open. You may edit the parameters. For details, see “Extension Voice Mail Settings”.
The Voice Mail Settings link will be visible only if you have configured the respective SIP ID.
Click Close to close the window.
Advanced Configuration Parameters

• If you want to provide other features like Personal Directory, Priority, or assign a Station Type to the SIP
Extension, click the Advanced button at the bottom of the page.

• Enter the Mobile Number of the extension user you wish to store. The Number can be a maximum of 16
digits.
• Enter the Email ID of the extension user you wish to store. The Email ID can be a maximum of 64
characters.
• You can assign the extension user to a Group. Select the desired SMS/Email Group Type from the
list.The system clubs together extension users assigned the same Group. Default: None. For details, see
“SMS/Email Group”.
• Assign the SIP Extension to a Call Pick-up Group, if required.
Call Pick Up allows the SIP Extension to 'pick up' (answer) calls ringing on any other extension, by dialing
a feature code, without physically going to the ringing extension. It also allows incoming calls for the SIP
Extension to be answered by the other extensions assigned the same Call Pick-Up group.
For this to work, both the ringing extension and the extension picking up the call must be in the same 'Call
Pick Up Group'. Refer “Call Pick Up” for instructions on how to create groups. You can create as many as
99 groups numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this SIP Extension in this field.
• You must assign the extension user to a COSEC Door Group for COSEC Integration. The users in the
same group must be assigned the same group. You can create as many as 50 groups numbered from 00
to 50. Users who are assigned COSEC Door Group ‘00’ are not a part of any group. See “COSEC
Integration” for more information.
• If this is an Operator extension and you want the system to play beeps during a conference to the
participants, to indicate the presence or absence of the Operator, select the Station Type as Assistant.
If you are using the system in the Hotel Mode, select the Station Type for the SIP Extension as
Administration/Assistant or Guest. The system will consider the options Administrator and Assistant as
same.
• You may assign a Personal Directory number to the SIP Extension. Default: 00.
A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes ("Out Going Trunk Bundle Group Index"). The Personal
Directory is necessary for using the features “Abbreviated Dialing” and “Dial By Name”.
When a Personal Directory is assigned to a SIP Extension, make sure you also configure this directory.
The Personal Directory can be programmed by the SIP Extension users and by the System Engineer.
Refer the topic “Abbreviated Dialing” for instructions on programming the Personal Directory.
If Personal Directory is not to be assigned, enter 00 in this field.
• Select a Priority Level for the SIP Extension from 1 to 9. Default; 5-Normal.
Each extension of the SARVAM UCS is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls an
extension with lower priority, a triple ring is placed on the called extension. To know more, read the feature
description “Priority”.

If this SIP extension is assigned to Operator, you may want to set a higher priority for this extension.
• Click Submit to save your SIP Extension Settings.
Configuring Matrix Extended Phone Settings using Jeeves

You can register three Matrix Extended IP Phones/UC Clients at three different locations as a single SIP
Extension. You can connect/register the same or different types of Extended Phones/UC Clients —SPARSH
VP248, SPARSH VP310, SPARSH VP510, SPARSH VP330, Matrix VARTA UC Clients or MATRIX
VARTA WIN200 Desktop UC Client — at each location. In this case we assume that SPARSH VP248 is connected
at Location 1, 2 and 3.
If you want to use more than one SPARSH VP248 Extended IP Phones as a SIP Extension, configure their settings
at Location 1, Location 2 and Location 3.
If you have connected SPARSH VP330 at any of the locations, refer to “Configuring Matrix SPARSH VP330”.
If you have registered Matrix VARTA ADR100 and VARTA AMP100 Mobile UC Clients in any of the locations, refer
to “Configuring Matrix VARTA ADR100/AMP100 UC Clients”.
If you have registered MATRIX VARTA WIN200 Desktop UC Client in any of the locations, refer to “Configuring
Matrix VARTA WIN200 UC Client”.

• Click Location 1.
• The settings of the phone at Location 1 appear.
• Select the Enable Device check box. Default: Disabled.

• Enter the Location Name for the phone to identify the phone. Location name may be the place where the
phone is located (e.g.: Head office, branch, residence). The Location Name may consist of 18 characters
(maximum). Default: Blank.
• Select MATRIX SPARSH VP248 as the Device Type at this location.
• Enter the MAC Address120 of the SPARSH VP248 connected at this location in hexadecimal format:
00:1b:09:XX:XX:XX. Default: blank.
SARVAM UCS validates the Extended Phone on the basis of the MAC Address, and provides
configuration on validation.
As SARVAM UCS allows registration of the SIP Extension from three different locations, it identifies the
SIP Extension in each location by the programmed MAC address.
• Select the appropriate Registrar Server Address to register the SPARSH VP248 with the SIP Registrar of
SARVAM UCS, according to your installation scenario:
• If the SPARSH VP248 is connected on the WAN network, select Use WAN Port IP Address as
Registrar Server IP Address.
• If the SPARSH VP248 is connected on the LAN network, select Use LAN Port IP Address as
• If the SPARSH VP248 is connected in the Global Network and SARVAM UCS is located behind a
Router, or behind a NAT Router and STUN is programmed, select Use Router/STUN's IP Address as
Make sure you configure either the Router’s Public IP Address or Simple Traversal of UDPs
through NATs (STUN) in Network Parameters. For details, see “Configuring Network Parameters”.
• If Dynamic DNS is configured in the Network Parameters, select Use Dynamic DNS Host Name as
By default, Use WAN Port IP Address is selected as the Registrar Server IP Address.
• To set the call progress tone generation standards of the country where the SPARSH VP248 is installed,
select the Call Progress Tone - Region. Default: Region 1.
See “Call Progress Tones” to know more.
• To display the Date and Time of the country where the SPARSH VP248 is installed, select the Date and
Time - Region. Default: India.
• If you want to enable Daylight Saving Time (DST) on the phone, set Apply DST? to Yes. Default: No.
The Daylight Saving Time convention followed in the country/region you selected will be automatically
applied. The SPARSH VP248 will change its date and time settings according to the DST convention of the
selected country/region.
120. MAC address is the address of the electronic hardware devices such as a computer, which is hard-coded into the device during
manufacture and cannot be modified. No two devices can have similar MAC address and thus it uniquely identifies your phone.
MAC address is assigned as per the IANA standard. The MAC Address of the phone will be used as source MAC address on all
Ethernet frames.

• Select the CO CLIP Pattern for the SPARSH VP248. This is the type of Calling Line Presentation on the
phone for incoming calls from trunks. You can select any of these options:
• Name Only (only the name of the caller will be displayed).
• Number Only (only the number of the caller will be displayed).
• Number + Name (both the name and the number of the caller will be displayed).
Default: Number + Name.
• Select the Language for the SPARSH VP248. Default: English.
SARVAM UCS provides language support for English, French, German, Spanish, Portuguese, and Italian
on the SPARSH VP248. When you select any of these languages, all the prompts and command strings
will appear in the selected language.
SIP Extension users can change the language by accessing and navigating through the phone menu.
The SA can change the Language by logging into the SA Jeeves.
• Select a Ringer Mode for the phone from the four options:
• Ring immediately (it rings immediately as a fresh calls lands on the phone).
• Ring if idle (rings only if the phone is idle).
• Ring after a delay (if the call is still not answered).
• Silent.
Default: Ring Immediate.
• If you selected Ring after a delay as Ringer Mode, set the Ring Delay Timer (sec), if required, to the
desired value.
The Ring Delay Timer is the time in seconds the system waits on receiving a call before ringing on the
phone. The range of this timer is 0 to 99 seconds. Default: 10 seconds.
• If you want to enable Ringer Auto Acknowledge mode, set the Acknowledge Timer (sec) to the desired
value.
The Ringer Auto Acknowledge mode determines when to stop the ring on the phone. There are two
options for Ringer Auto Acknowledge:
• Stop only when the call is answered.
• Stop after a delay.
To stop the ring on the phone after a delay, the Acknowledge Timer must be configured. The range of this
timer is 01 to 99 seconds. Default: 00 seconds.
To stop the ring only when the Call is answered or manually acknowledged, the Acknowledge Timer must
be set to '00'. By default, Ring Auto Acknowledge is turned OFF.
• To assign the Ring Destination for the SPARSH VP248, select the desired destination for Play Ring on.
You may choose
• Speakerphone: The ring will be played on the Speakerphone.
• Headset: The ring will be played on the Headset.
Default: Speakerphone.
When you select the Headset as the destination, make sure that you set the flag ‘Headset Connected?’ to
Yes, connect a Headset to the SPARSH VP248.

• Select the desired Ring Tune according to your/SPARSH VP248 user’s preference. Default: 1.
• Set the Ringer Volume to the desired level, from 0 to 7, according to your preference. Default: 4.
• To increase/decrease the volume of outgoing speech (Transmit Gain) on the handset of the SPARSH
VP248, set the Handset Transmit Volume Level to the desired level, from 0 to 7. Default: 4.
• To increase/decrease the volume of incoming speech (Receive Gain) on the handset of the SPARSH
VP248, set the Handset Receive Volume Level to the desired level, from 0 to 7. Default: 4.
• To increase/decrease the volume of outgoing speech (Transmit Gain) on the headset of the SPARSH
VP248, set the Headset Transmit Volume Level to the desired level, from 0 to 7. Default: 4.
• To increase/decrease the volume of outgoing speech (Receive Gain) on the headset of the SPARSH
VP248, set the Headset Receive Volume Level to the desired level, from 0 to 7. Default: 4.
• To change the Transmit Gain of the Speakerphone MIC Volume, set Speaker Transmit Volume Level to
the desired level, from 0 to 7. Default: 4.
• To change the Receive Gain of the Speakerphone MIC Volume, set Speaker Receive Volume Level to
• To use a Headset with the SPARSH VP248, select the Headset Connected? check box. Default:
Disabled.
Make sure that you connect a Headset to the SPARSH VP248, if you enable this option.
• Select the Auto Answer check box to enable this feature on the SPARSH VP248. Default: Disabled.
When you set the “Auto Answer” feature on the SPARSH VP248, the phone goes OFF-Hook automatically
after a preset period of time, without the extension user having to pick up the handset or press the speaker
or headset key. When you enable Auto Answer, you must configure the Auto Answer Timer.
• If you enabled Auto Answer on the phone, set the Auto Answer Timer (sec) to the desired value.
This timer defines the time in seconds that the SPARSH VP248 should wait before going OFF-Hook to
auto answer a call. The range of this timer is 1 to 9 seconds. Default: 1 second.
• Adjust the Backlight brightness of the phone’s LCD display, by setting the LCD Backlight Level to the
desired value, from 1 to 4. Default: 3.
• Set the Back Light Off Timer (sec) to the desired value, if required, from 000 to 999 seconds. Default: 10
seconds.
• Set the LCD Contrast Level to a level from 1 to 4 that is comfortable to you. Default: 3.
DSS Key Settings

• You can select the desired key template — Operator, Executive1, Executive2, Executive3, Hotel
Attendant, Guest or any other template you added. See “Customizing Extended IP Phone Templates using
Jeeves” for more details.
OR

• You can personalize the key map of the SPARSH VP248 for this location. To do so,
• Select Personalized as the Key Template option.
• Click Key Settings.
• The key map of the Extended Phone opens in a new window on your screen.
• Click the key you want to configure. For example, CO3.

The Functions to be Performed by the key opens in a new window.
• In the Select Function Type list, select the function to be performed by the key. For example, you want
to use the key to call the Operator.
The Operator function is a Feature, so select the option FEATURE from the Select Function Type list
box.
From the Select Offset drop down list, all the features that can be assigned to keys are listed.
• Select Operator from the list of features in the Select Offset box.

• Click OK.
The Operator feature appears on the key label.

• To take a second example, if you want to assign Remote DND to the key currently assigned CO 2 key,
click the key.
• In the Select Function Type list box, select the option SA Command, as Remote DND is a System
Administrator (SA) Command.
• In the Select Offset box, select the option Set DND for remote station.

• Click OK. The box closes. Remote DND feature will appear in abbreviated form as R-DND on the key
label.
• Follow the same instructions to assign features to other DSS keys. Selecting the appropriate Function
Type and the Offset for each feature/function.
If you want assign a feature, select FEATURE as function type, and select the desired feature as
Offset.
If you want to use the key to call a DKP or a SIP extension, select DKP or SIP Extension as Function
Type and select the number of the extension as Offset.
To assign direct access to a mobile trunk, select MOBILE as Function Type and the desired port
number 1 or 2 as Offset.
To assign direct access to a SIP Trunk, select SIP as Function Type and the desired trunk number from
1 to 4 as Offset.
Click OK, each time you select a Function Type and Offset in the dialog box.
You can reinstate default key assignment any time, by clicking the Default button at the bottom of the
window.
• When you complete assigning functions to keys, close the window.
• If you assign/re-assign functions to the following keys, the Phone will restart:
• Speaker

• Headset
• Ringer
• Acknowledge
• Local Menu
If you have upgraded your SPARSH VP248 to an Extended IP Phone with firmware V5Rx, the key labels
listed in the table below will have the following functions:
Key Label Function assumed with Firmware V5Rx
Fwd Busy Voice Mail
Fwd NR Auto Call Back
DND Forward
Reject Release
RTP Port
• Define RTP Port:
• RTP Listening Port: This is the port on which the SPARSH VP248 listens for SIP messages over TCP.
This port is also used as the source port for sending RTP packets. This port is also used as the source
port for sending RTP packets to the remote peer. The valid range for this port is 1025-65278. Default:
8000.
Quality of Service
• Set the SIP Quality of Service (QoS) for SIP signaling as:
• SIP DiffServe/ToS. Valid range is 00 to 63. Default: 26.

OR
• RTP DiffServe/ToS. Valid range is 00 to 63. Default: 46.
NAT Keep Alive

• If the SPARSH VP248 is connected behind a NAT router, configure NAT Keep Alive.
• Select the check box Enable NAT Keep Alive to send Keep Alive messages periodically to refresh the
binding in the NAT router. Default: Disabled.
• Define as Interval (sec), the time period, from 001 to 999 seconds, after which the phone should send
Keep Alive message. Default: 120 seconds.
The time period you define should be less than the binding timer of the router.
Timers
• Set the following Timers to the desired value, where required:
• SIP INVITE Timer (sec): This is the time in seconds that the phone waits for a response from the
called party after ending INVITE message. This timer starts after sending INVITE message to the
called party and stops on receipt of the provisional response or the final response or when the user

disconnects the call. On expiry of the timer, the phone terminates the call process and gives an error
tone to the user. The range of the SIP INVITE TIMER is 10-180 seconds. Default: 30 seconds.
• SIP Provisional Timer (sec): This is the time in seconds that the phone waits for final response after
receiving the provisional response from the called party. This timer starts on the receipt of the
provisional response from the called party and stops on receipt of the final response from the called
party or when the user disconnects the call. On expiry of the timer, the IP phone terminates the call
process and gives error tone to the user. The range of SIP Provisional Timer is 10-180 seconds.
Default: 60 seconds.
• General Request Timer (sec): This is the time in seconds for which the phone waits for response of a
transaction request. This timer starts on initiating a transaction. This timer stops on receipt of a
response for the request. On expiry of the timer, the phone clears the transaction. This timer is used for
Registration request, etc. The range of the General Request Timer is 10-60 seconds. Default: 20
seconds.
Debug
• To debug using Syslog Client supported by the SPARSH VP248, configure Debug parameters:
• Select the Enable Debug? check box. Default: disabled.
When the Debug flag is enabled, the phone will send the debug messages to the Syslog Server IP
address. Debug report can be viewed on the Syslog Server or any other application which can capture
the Syslog messages/debug sent by the phone.
• Enter the IP Address and port of the remote Syslog Server and as Syslog Server Address and
Server Port.
The address of the Listening Port of the Syslog Server is from 1025-65535;514. Default: 514. Syslog
uses the UDP as transport protocol and listens on the port 514 (the default listening port).
• You may select the Debug Level from the following options, by selecting the respective check box:
• SIP
• System
• Hardware
• Call
• Network
• VoPP
You may select any or all of these debug levels. The Syslog Client will send only the debug messages
for the selected level to the remote server on the IP network. For example, if the debug log of 'Call's is
required, you can select this option, and disable all others.
• Click Submit to save settings.
• If you have completed the configuration of the SPARSH VP248 Settings at Location 1, follow the same
steps as described above to configure the SPARSH VP248 at Location 2 and Location 3.
When you change any of the parameters listed below in the SIP Extension at Location 1, 2, 3, the phone
will restart automatically, if registered:
• Use SIP Extension
• SIP ID
• Authentication ID
• Authentication Password

• Registrar Server IP Address
• MAC Address
• Enable Device
• Device Type
• Key Map in the Key Template assigned to phone
• Call Progress Tone
• Date and Time
• Apply DST?
• QoS
• RTP Ports
• NAT Keep Alive
• SIP Timers
• The SE Password of SARVAM UCS is changed
• Specific parameters in VoIP Parameters are changed
• Specific parameters Network Port parameters are changed
• You restart the System
• Set the System to Default

SPARSH VP310, the Executive IP Phone is engineered to offer a contemporary design with crystal-clear audio and
feature-rich capabilities at economical price. It functions like SPARSH VP248,the proprietary SIP-based IP Phone
for SARVAM UCS, of Matrix. To know the list of features supported, refer to “SARVAM UCS Features Supported in
Terminals”.
For instructions on how to use SPARSH VP310, refer to the common EON48_310_SPARSH VP248_310 User
Guide.
• Extended IP Phone Settings, see “Configuring Matrix Extended Phone Settings using Jeeves”
Settings”.


IPv4 Addresses only for registering Extended IP Phones.
• Select the Use SIP Extension check box to enable the SIP extension. Default: disabled.
from 0 to 9 and the characters * and # are allowed. You cannot assign the same SIP ID to more than one
extension.
Default: Blank.

Default: 2.
• Off
• Beep Once
Default: Beep Once
• INVITE Request
options:
made to such users.
SIP Trunks.
you purchase.

field.

Template field.

customized.


digits.
characters.
• Assign the SIP Extension to a Call Pick-up Group, if required. Default: 01
Call Pick Up allows the SIP Extension user to 'pick up' (answer) calls ringing on any other extension, by
dialing a feature code, without physically going to the ringing extension. It also allows incoming calls for the
SIP Extension to be answered by the other extensions assigned the same Call Pick-Up group.
same.


You can register three Matrix Extended IP Phones/Soft Clients at three different locations as a single SIP
Extension. You can connect/register the same or different types of Extended Phones—SPARSH VP248, SPARSH
VP310, SPARSH VP510, SPARSH VP330, Matrix VARTA ADR100 Mobile UC Client, VARTA AMP100 Mobile UC
Client or MATRIX VARTA WIN200 Desktop UC Client — at each location. In this case we assume that SPARSH
VP310 is connected at Location 1, 2 and 3.
If you have connected SPARSH VP248 at any of the locations, refer to “Configuring Matrix Extended Phone
Settings using Jeeves”.
If you have registered Matrix VARTA Mobile UC Clients in any of the locations, refer to “Configuring Matrix VARTA
ADR100/AMP100 UC Clients”.


through NATs (STUN) in Network Parameters. For details , see “Configuring Network Parameters”.
Ethernet frames.

• Silent.
desired value.
value.
You may choose

Disabled.
seconds.
DSS Key Settings

OR


box.
• Click OK.

click the key.

label.
Offset.

To assign direct access to a Mobile Trunk, select MOBILE as Function Type and the desired port
number 01 or any other trunk number as Offset.
01 or any other trunk number as Offset.
window.
• Speaker
• Headset
• Ringer Acknowledge
• Local Menu
RTP Port
8000.
Quality of Service

OR
NAT Keep Alive

Timers

seconds.
Debug
Server Port.
• SIP
• System
• Hardware
• Call
• Network
• VoPP

• SIP ID
• MAC Address
• Enable Device
• Device Type
• Date and Time
• Apply DST?
• Transport Mode and SRTP
• QoS
• RTP Ports
• NAT Keep Alive
• SIP Timers
• Specific parameters in Network Port parameters are changed

SPARSH VP330 is the proprietary Extended IP Phone with graphical touch-screen user interface, supplied by
Matrix. The feature-rich SIP based phone support most of the features and functions of the proprietary digital key
phones of SARVAM UCS.To know the list of features supported, refer to “SARVAM UCS Features Supported in
Terminals”.
For detailed product information and operation instructions, refer to the SPARSH VP330 User Guide.
To be able to use SPARSH VP330124 as SIP Extensions, you must configure the following:
• Extended Phone Settings, see “Configuring Matrix Extended Phone Settings using Jeeves”
Settings”.


from 0 to 9 and the characters * and # are allowed.
You cannot assign the same SIP ID to more than one extension.
string of maximum 6 digits. All ASCII characters except < > and “ (double quote) are allowed. Default:
Blank.
Default: Blank.

Default: 2.
• INVITE Request
options:
made to such users.
SIP Trunks.
you purchase.

field.
Template field.

customized.

digits.
characters.

same.

You can register three Matrix Extended Phones/Soft Clients at three different locations as a single SIP Extension.

You can connect the same or different types of Extended Phones/Soft Clients —SPARSH VP248, SPARSH VP310,
SPARSH VP510, SPARSH VP330, Matrix VARTA ADR100 Mobile UC Client, Matrix VARTA AMP100 Mobile UC
Client or MATRIX VARTA WIN200 Desktop UC Client — at each location. In this case we assume that SPARSH
VP330 is connected at Location 1, 2 and 3.
If you have connected SPARSH VP248 refer to “Configuring Matrix Extended Phone Settings using Jeeves” in
Configuring Matrix SPARSH VP248 as SIP Extensions.
If you want to use more than one SPARSH VP330 Phones as a SIP Extension, configure their settings at Location
1, Location 2 and Location 3.

SARVAM UCS validates the SPARSH VP330 on the basis of the MAC Address, and provides
Ethernet frames.

applied. The IP phone will change its date and time settings according to the DST convention of the
DSS Key Settings

OR

• Click the key you want to configure. For example, SLT2.


box.

• Click OK.
Offset.
To assign direct access to a mobile trunk, select MOBILE as Function Type and the desired port
number 1 or 2 as Offset.

1 to 4 as Offset.
window.
The phone will enter the Auto Configuration mode, when you assign/re-assign certain features in the key
maps. To know more, refer to the SPARSH VP330 User Guide.
Even if you assign keys for the following feature in the Key Templates, these features will not function:
Function Type Offset
Macro
SA Command
Special Keys Digit Pause
Digit A
Digit B
Digit C
Digit D
Enter
Local Menu
Feature Enter SE Programming Mode
Enter SA Programming Mode
Personal Directory Programming
Abbreviated Dialing
Change Room Clean Status
Guest Access Code
Minibar Details
Emergency Conference
Self Ring Test
SA Command Prefix
PMS - User Defined Fields
Department Group Call Forward
Transport Mode and SRTP

• Select the protocol to be used to transport the SIP messages. You can select the Transport Mode as TCP
or TLS.
If you select TCP, make sure the SIP Over TCP check box is selected in VoIP Parameters.
If you select TLS, make sure the SIP Over TLS check box is selected in VoIP Parameters.

• For secure conversations over SIP, select the Enable SRTP? check box. The SIP messages will be
transported over SRTP only.
RTP Port
• RTP Listening Port: This is the port on which the phone listens for SIP messages over TCP. This port
is also used as the source port for sending RTP packets. This port is also used as the source port for
sending RTP packets to the remote peer. The valid range for this port is 1025-65278. Default: 8000.
Quality of Service

OR
NAT Keep Alive

Timers
seconds.

• If you have completed the configuration of the SPARSH VP330 Phone Settings at Location 1, follow the
same steps as described above to configure the SPARSH VP330 Phone at Location 2 and Location 3.
will go in Auto Configuration mode automatically, if registered:
• SIP ID
• Name
• MAC Address
• Enable Device
• Device Type
• Language
• Date and Time
• Apply DST?
• QoS
• RTP Ports
• NAT Keep Alive
• SIP Timers
• Trunk Access Code
• Emergency Numbers
The SIP Extension registered at Location 1, 2, 3, will also restart, if:


SPARSH VP510, the Premium IP Phone is engineered to offer a contemporary design with crystal-clear audio and
feature-rich capabilities at economical price. To know the list of features supported, refer to “SARVAM UCS
Features Supported in Terminals”.
For instructions on how to use SPARSH VP510, refer to the EON510_SPARSH VP510 User Guide.
Settings”.


Default: Blank.
registration of this SIP Extension. However, you can remove the IP Address from the Blacklist IP
Address list. See “Black List IP Address - SIP Extensions” for more details. This activity will be logged
in the “System Activity Log” as well as “Simple Network Management Protocol (SNMP)”.
Default: 2.
you purchase.

• Off
• Beep Once
Default: Beep Once
• INVITE Request
options:
made to such users.
Cancellation, Jitter Buffer and Fax-over-IP options and related parameters.
SIP Trunks.Check if the values in this template fulfill requirements of the SIP Extension. If Template 01
fulfills the feature requirements, retain Template 01.

field.
Template field.

customized.

digits.
characters.

• Assign the SIP Extension to a Call Pick-up Group, if required. Default: 01
same.

You can register three Matrix Extended IP Phones/Soft Clients at three different locations as a single SIP
Extension. You can connect/register the same or different types of Extended Phones/Soft Clients —SPARSH
VP248, SPARSH VP310, SPARSH VP330, SPARSH VP510, Matrix VARTA ADR100 Mobile UC Client, VARTA
AMP100 Mobile UC Client or MATRIX VARTA WIN200 Desktop UC Client— at each location. In this case we
assume that SPARSH VP510 is connected at Location 1, 2 and 3.
If you have connected SPARSH VP248 at any of the locations, refer to “Configuring Matrix Extended Phone
Settings using Jeeves”.

00:1b:09:XX:XX:XX. Default: Blank.
Ethernet frames.


• Silent.
desired value.
value.
You may choose
You can also set the Ringer Tune. For detailed instructions, refer to the EON510_SPARSH VP510 User
Guide.

Disabled.
seconds.
DSS Key Settings

OR
• Click Submit.


box.
• Click OK.

click the key.

label.
Offset.
To assign direct access to a Mobile Trunk, select MOBILE as Function Type and the desired port
number 01 or any other trunk number as Offset.
01 or any other trunk number as Offset.
window.
• Speaker
• Headset
• Ringer Acknowledge

• Local Menu
• You can also connect a DSS Console (DSS532) with SPARSH VP510. For instructions:
• to install the DSS532 with SPARSH VP510, see “Installing DSS532 with SPARSH VP510”.
• to configure the DSS keys of the Console, see “Configuring DSS Console Keys connected to
SPARSH VP510”.

or TLS.
If you select this check box, make sure you have selected SRTP Mode as Forced or Optional in the
General Parameters under SIP Extension Settings.
RTP Port
• Define the RTP Port:
8000.
Quality of Service

OR
NAT Keep Alive


Timers
seconds.
Debug
Server Port.
• SIP
• System
• Hardware
• Call
• Network
• VoPP

• SIP ID
• MAC Address
• Enable Device
• Device Type
• Date and Time
• Apply DST?
• QoS
• RTP Ports
• NAT Keep Alive
• SIP Timers

Configuring Matrix VARTA WIN200 UC Client
MATRIX VARTA WIN200, is a SIP (Session Initiation Protocol) based Unified Communication Desktop Client
running on Windows OS, delivering full-array of the System features to the user on-the-go along with an added
advantage of video calling. Through tight integration with the enterprise features of the System, UC Client provides
advance call capabilities including Conferencing, Corporate Directory Access (Global Directory), Call Logs and
Conversation Recording with one-touch access. Other than these you can take the advantage of using premium
features like Presence subscription and notification, Corporate Voicemail access to enhance your overall mobile
experience.
To use MATRIX VARTA WIN200 Desktop UC Client, make sure you have:
• Purchased and activated the VARTA Essential, VARTA Professional or VARTA Collaboration license. For
more details, see “License Management”.
• Assigned the desired license to the SIP Extension. For more details, see “VARTA License Management”.
To know the list of featured supported, refer to “SARVAM UCS Features Supported in Terminals”.
For detailed product information and operation instructions, refer to the MATRIX VARTA WIN200 User Guide.
Configuring MATRIX VARTA WIN200 using Jeeves

To be able to register and use the Desktop Client, you must configure the following:
• SIP Extension Settings, see “Configuring SIP Extension using Jeeves”.
• Extended Phone Settings, see “Configuring Matrix Extended Phone Settings using Jeeves”.
• Voice Mail Settings, if you want to provide Voice Mail to the extension. See “Extension Voice Mail
Settings”.


• In Name, enter a name for the SIP Extension, which may be the name of the person who will use the SIP
Extension or the name of a Department. The name you enter here will be displayed as the Caller ID of the
SIP Extension on the remote user's phone, when the SIP Extension user makes calls.
from 0 to 9 and the characters * and # are allowed. You cannot assign the same SIP ID to more than one
extension.

string of maximum 6 digits. All ASCII characters except < > and “ (double quote) are allowed. Default:
Blank.
Default: Blank.
Default: 2.
• INVITE Request
options:
you purchase.

made to such users.
SIP Trunks.
field.

Template field.
customized.


digits.
characters.
Call Pick Up allows the SIP Extension to 'pick up' (answer) calls ringing on any other extension, by using
the respective icon from Client GUI, without physically going to the ringing extension. It also allows
incoming calls for the SIP Extension to be answered by the other extensions assigned the same Call Pick-
Up group.

same.

You can register three Matrix Extended Phones/Soft Clients at three different locations as a single SIP Extension.
You can connect the same or different types of Extended Phones/Soft Clients — SPARSH VP248, SPARSH
VP330, SPARSH VP310, SPARSH VP510, Matrix VARTA ADR100 Mobile UC Client, Matrix VARTA AMP100
Mobile UC Client or MATRIX VARTA WIN200 Desktop UC Client — at each location. In this case we assume that
MATRIX VARTA WIN200 Desktop UC Client are registered at Location 1, 2 and 3.

If you want to use more than one UC Clients as a SIP Extension, configure their settings as Location 1, Location
2 and Location 3.
If you want to use the IM functionality in the UC Client, you must configure it at Location-1 only.

• Select MATRIX VARTA WIN200 as the Device Type at this location. Make sure you assign the desired
license to this SIP extension. For details, see “VARTA License Management”.
• In MAC Address enter the Device ID here. Default: blank.
SARVAM UCS validates the desktop/PC on which you have installed the UC Client on the basis of the
Device ID, and provides configuration on validation.
SIP Extension in each location by the programmed Device ID.
• Select the appropriate Registrar Server Address to register the UC Client with the SIP Registrar of
• If you want the UC Client to register using the WAN network, select Use WAN Port IP Address as
Registrar Server Address.
• If you want the UC Client to register using the LAN network, select Use LAN Port IP Address as
• If the UC Client is registered in the Public (Global) Network and SARVAM UCS is located behind a
the Registrar Server Address.


or TLS.
The UC Client supports RTP Relay. For detailed description, see “Configuring VoIP Parameters”.
SMS Over IP
• If you want UC Client users to send SMS to any extension user as well as receive IM from any extension
user, select the Enable SMS Over IP check box. For detailed information, see “SMS over IP”.
RTP Port
• Define the RTP Port. This is the port on which the client listens for RTP packets. This port is also used as
the source port for sending RTP packets to the remote peer. The valid range for this port is 1025-65278.
Default: 8000.
Quality of Service

OR
NAT Keep Alive

• If the UC Client is connected behind a NAT router, configure NAT Keep Alive.
Keep Alive message. Default: 120 seconds. The time period you define should be less than the binding
timer of the router.
Timers

• SIP INVITE Timer (sec): This is the time in seconds that the client waits for a response from the called
party after ending INVITE message. This timer starts after sending INVITE message to the called party
and stops on receipt of the provisional response or the final response or when the user disconnects the
call. On expiry of the timer, the client terminates the call process and gives an error tone to the user.
The range of the SIP INVITE TIMER is 10-180 seconds. Default: 30 seconds.
• SIP Provisional Timer (sec): This is the time in seconds that the client waits for final response after
party or when the user disconnects the call. On expiry of the timer, the client terminates the call
• General Request Timer (sec): This is the time in seconds for which the client waits for response of a
seconds.
• If you have completed the configuration of the UC Client at Location 1, follow the same steps as described
above to configure the UC Client at Location 2 and Location 3.
• SIP ID
• Name
• Registrar Server Address
• MAC Address/IMEI/ESN Number
• Enable Device
• Device Type
• Transport Mode
• Enable SRTP
• QoS
• SIP/RTP Ports
• RTP Listening Port
• SMS over IP
• NAT Keep Alive
• SIP Timers

Configuring Matrix VARTA ADR100/AMP100 UC Clients
Matrix offers the following Mobile UC Clients131:
• MATRIX VARTA ADR100 for Android Smartphones/Tablets

• MATRIX VARTA AMP100 for iPhones
To use MATRIX VARTA UC Clients for Mobile, make sure you have:
• Purchased and activated the VARTA Essential, VARTA Professional or VARTA Collaboration license. For
more details, see “License Management”.
MATRIX VARTA ADR100
MATRIX VARTA ADR100 is a proprietary SIP (Session Initiation Protocol) based UC Client Application
running on Android Phones/Tablets, delivering full-array of Matrix SARVAM UCS features to the user on-the-go
along with an added advantage of UC features. Through tight integration with the enterprise mobility features of the
SARVAM UCS, VARTA ADR100 provides advance call capabilities including Conferencing, Corporate Directory
Access (Global Directory), Call Logs and Conversation Recording with one-touch access. Other than these you can
take the advantage of using premium features like Video Calling, IM, IM to SMS, Presence notification and
corporate Voicemail System access to enhance your overall mobile experience.
Mobile workers can use any Wi-Fi or cellular data networks to stay connected with business communications while
working from office, home or travelling to any location. An innovative and easy to understand user interface delivers
all productivity features at fingertips that enhance speed of communication and collaboration with office users and
customers.
For detailed product information and operation instructions, refer to the MATRIX VARTA ADR100 User Guide for
Mobile/Tablet.
MATRIX VARTA AMP100
MATRIX VARTA AMP100 is a proprietary SIP (Session Initiation Protocol) based UC Client Application
running on iPhones, delivering full-array of Matrix SARVAM UCS features to the user on-the-go along with an
added advantage of UC features. Through tight integration with the enterprise mobility features of the SARVAM
UCS, VARTA AMP100 provides advanced call capabilities including Conferencing, Corporate Directory Access
(Global Directory), Call Logs and Conversation Recording with one-touch access. Other than these, you can take
the advantage of using premium features like Video Calling, IM, IM to SMS, Presence notification/subscription and
corporate Voicemail access to enhance your overall mobile experience.
all productivity features at fingertips that enhances speed of communication and collaboration with office users and
customers.
For detailed product information and operation instructions, refer to the MATRIX VARTA AMP100 User Guide.
131. SARVAM UCS supports only IPv4 Addresses for registering Mobile UC Clients.

Configuring Mobile UC Clients using Jeeves
To be able to register and use the Mobile Handsets/Tablets as Matrix VARTA Mobile UC Clients, you must
configure the following:
• Voice Mail Settings, if you want to provide Voice Mail to the extension. See “Extension Voice Mail
Settings”.

IPv4 Addresses only for registering Matrix UC Clients.

• In the Name field, enter a name of SIP Extension, which may be the name of the person who will use the
UC Client/SIP Extension or the name of a Department. The name you enter here will be displayed as the
Caller ID of the SIP Extension on the remote user's phone, when the SIP Extension makes calls.
• In the Authentication ID field, enter the number which you want the system to use for user authentication
of the SIP messages received from the SIP Extension. You cannot keep this field blank. The number may
be a string of maximum 6 digits. All ASCII characters except < > and “ (double quote) are allowed. Default:
Blank.
• In the Authentication Password field, enter the password to be used by the system to authenticate the
SIP messages received from the SIP Extension. To avoid unauthorized access, we recommend you to
change the password regularly. Make sure it is strong and is kept confidential. The password must:
Default: Blank.
Default: 2.
you purchase. Make sure the NX DBM VOCODER64 module is installed for SIP calls.

• INVITE Request
options:
made to such users.
SIP Trunks.
field.

Template field.
customized.


digits.
characters.

Call Pick Up allows the SIP Extension to 'pick up' (answer) calls ringing on any other extension, by using
the respective icon from Client GUI, without physically going to the ringing extension. It also allows
incoming calls for the SIP Extension to be answered by the other extensions assigned the same Call Pick-
Up group.
same.

You can register three Matrix Extended Phones at three different locations as a single SIP Extension. You can
connect the same or different types of Extended Phones—SPARSH VP248, SPARSH VP330, SPARSH VP310 or
Matrix VARTA Mobile UC Clients — at each location. In this case we assume that Matrix VARTA Mobile UC Clients
are registered at Location 1, 2 and 3.

If you want to use more than one Matrix UC Clients as a SIP Extension, configure their settings at Location 1,
Location 2 and Location 3.
If you want to use the IM functionality in the MATRIX VARTA ADR100/AMP100, you must configure it at
Location-1 only.

• Select MATRIX VARTA ADR100/AMP100 (for the Android/iPhone Application) as the Device Type at this
location. Make sure you assign the desired license to this SIP extension. For details, see “VARTA License
Management”.
• Enter the MAC Address/IMEI133/ESN Number of the phone/tablet on which you have installed the
application.
If you are using an iPhone, enter the Device ID here. Default: blank.
SARVAM UCS validates the phone/tablet on which you have installed the application on the basis of the
IMEI/ESN Number or Device ID, and provides configuration on validation.
SIP Extension in each location by the programmed IMEI/ESN Number/Device ID.
• Select the appropriate Internal Registrar Server Address to register the application with the SIP
Registrar of SARVAM UCS within a private network. Select the appropriate option as per your installation
scenario:
• If you want the application to register using the WAN network, select Use WAN Port IP Address as the
Internal Registrar Server Address.
133. IMEI Number is the unique identification number of the GSM engine used in the Mobile handset.

• If you want the application to register using the LAN network, select Use LAN Port IP Address as
Internal Registrar Server Address.
By default, Use WAN Port IP Address is selected as the Internal Registrar Server Address.
• Select the appropriate External Registrar Server Address to register the application with the SIP
Registrar of SARVAM UCS from a public network. Select the option according to your installation scenario:
• If you want the application to register using the WAN network, select Use WAN Port IP Address as
External Registrar Server Address.
• If the application is connected in the Public Network and SARVAM UCS is located behind a Router, or
behind a NAT Router and STUN is programmed, select Use Router/STUN's IP Address as External
External Registrar Server Address.
By default, Use WAN Port IP Address is selected as the External Registrar Server Address.

or TLS.
For secure conversations over SIP, select the Enable SRTP? check box. The SIP messages will be
The application supports RTP Relay. For detailed description, see “Configuring VoIP Parameters”.
SMS Over IP
• If you want the VARTA users to send SMS to any extension user as well as receive IM from any extension
user, select the Enable SMS Over IP check box. For detailed information, see “SMS over IP”.
RTP Port
• RTP Listening Port: This is the port on which the phone listens for RTP packets. This port is also used
as the source port for sending RTP packets to the remote peer. The valid range for this port is 1025-
65278. Default: 8000.

Quality of Service

OR
NAT Keep Alive for MATRIX VARTA ADR100

• If the MATRIX VARTA ADR100 application is connected behind a NAT router, configure NAT Keep Alive.
For MATRIX VARTA AMP100 Application the NAT Keep Alive interval is 15 seconds (not programmable).
If NAT Keep Alive is not received from the AMP100 application for 30 seconds, SARVAM UCS will
consider the applications to be in the background.
Timers
seconds.
• If you have completed the configuration at Location 1, follow the same steps as described above to
configure the application at Location 2 and Location 3.

• SIP ID
• Name
• Internal Registrar Server Address
• External Registrar Server Address
• MAC Address/IMEI/ESN Number
• Enable Matrix Extended Phone Mode
• Extended Phone Type
• Transport Mode
• Enable SRTP
• QoS
• SIP/RTP Ports
• NAT Keep Alive
• SIP Timers
Viewing SIP Extension Status

You can view the Status of SIP Extension using Jeeves. To do this,

• Click SIP Extension Status.
• The SIP Extension Status page will open and display the following for each SIP Extension,
• SIP Extension number
• VoIP Ethernet Port Number to which the SIP Extension is assigned.
• Name of the SIP extension
• SIP ID assigned to the SIP Extension
• REGISTRATION status; whether the SIP Extension is registered or not.
• Contact 1 (for MATRIX VARTA AMP100 )
When the device is Registered - It will display the SIP ID, IP Address and the Registration Expiry
Timer.
When the device is in the Background - It will display Active and the time remaining for the expiry
of the Device Inactivity Timer
When Unregistered - The existing details will be cleared and it will be blank.
To know more, refer “Apple Push Notification Service Support”.
In case of MATRIX VARTA ADR100 the Contact 1, 2, 3,

• when Registered it will display the SIP ID, IP Address and the time remaining for the expiry of the
Registration Timer.
• when Not Registered it will be blank

• Contact 2 - same as above.
• Contact 3 - same as above.
• You may Log out of Jeeves.
You can also view the SIP Extension Status from the Status link. To view, click the SIP Extension link
under Status.

VARTA License Management
To view the VARTA License Status and to assign licenses to SIP Extensions,
• Click VARTA License Management.
VARTA License Management

The following information will be displayed for the license you activated (VARTA Essential/Professional/
Collaboration).
• License Type: This displays the name of the licenses — Essential, Professional or Collaboration.
• Total Available Licenses: This displays the total number of licenses activated.
• Total Used Licenses: This displays the total number of VARTA users registered as SIP extensions.
VARTA License Assignment

After registering and configuring MATRIX VARTA WIN200/VARTA ADR100/VARTA AMP100 as a SIP Extension,
you must select the desired license in the Assigned License field below. You can also filter the SIP Extensions as
per the license assigned.
• Apply Filters: By default you can view all the SIP Extensions, as all the filters are enabled.
• Clear the VARTA Essential check box, if you do not want to view the SIP Extensions that are assigned
this license.

• Clear the VARTA Professional check box, if you do not want to view the SIP Extensions that are
assigned this license.
• Clear the VARTA Collaboration check box, if you do not want to view the SIP Extensions that are
assigned this license.
• Clear the None check box, if you do not want to view the SIP Extensions that are not assigned any
license.
• SIP Extension: This displays the SIP Extension Number with which you can register the VARTA UC
Clients.
• SIP ID: This displays the SIP ID assigned to the SIP Extension.
• Assigned License: Select the license you wish to assign to the SIP Extension and click Submit.
• Location 1, 2, 3: This displays the Device Type selected on the SIP Extension Location 1, 2 and 3.

Configuring Open SIP Phones
You can connect any of the following as SIP Extensions of the SARVAM UCS:
• Matrix SPARSH VP248

• Matrix SPARSH VP110
• Any Standard third party Open SIP Phone
• Any SIP-enabled device including PC based Soft-phone
• Analog Phone Adaptor
For detailed product information and operational instructions, refer to the product documentation supplied
with the Open SIP Phone/device.
SARVAM UCS supports two separate methods of Provisioning the Open SIP Phones. These two methods are:
• Manual Provisioning: In Manual Provisioning, the user must configure the required parameters of the
Open SIP Phone manually. So, it is not a simple plug-and-play solution for mass deployment, as it requires
intervention of authorized technical personnel for phone configuration. To configure Open SIP Phones
using this method, see “Configuring Open SIP Phones using Manual Provisioning”.
Auto Provisioning: In Auto Provisioning, the Open SIP Phone gets configured automatically by retrieving
the required configuration file from the SARVAM UCS. The configuration file contains pre-programmed
values of necessary parameters required by the Open SIP Phone. Thus, it eliminates the necessity of
manually configuring the Open SIP Phone parameters. When the Open SIP Phone starts, it then gets
configured automatically. Here, SARVAM UCS acts as the Auto Provisioning Server by providing the
configuration file to the Open SIP Phones. Auto Provisioning enables mass deployment of Open SIP
Phones and provides a plug-and-play solution for them. To configure Open SIP Phones using this method,
see “Configuring Open SIP Phones using Auto Provisioning”.
• Auto Provisioning for Matrix SPARSH Phones and Third Party SIP Phones is supported only through
the HTTP Port.
• For Auto Provisioning, you must configure the Open SIP phones at Location1 only. You will have to
manually configure the Open SIP phones at Location 2 and 3.
• If you have already configured the phones at Location 2 and 3 and you upgrade the firmware, the
phones will not function as Auto Provisioning is not supported at these locations. You will have to
configure these phones again manually.
To be able to use Open SIP Phones/Devices as SIP Extensions, you must configure the following:
• Voice Mail Settings, if you want to provide mailbox to the extensions. See “Extension Voice Mail Settings”.



IPv4 and IPv6 Addresses for registering Open SIP Phones.
If no name is assigned to the SIP Extension, the system will display the name received in the INVITE
message from the SIP Extension user when making outgoing calls.
the system can call a SIP Extension by dialing the SIP ID assigned to the SIP Extension. SIP ID of each
SIP Extension must be a unique number string of a maximum of 6 digits. Any combination of digits from 0
to 9 and the characters * and # are allowed.
string of maximum 6 digits. All ASCII characters except < > and “ (double quote) are allowed.Default:
Blank.
Default: Blank.
• In HTTP Authentication Password (Third Party IP-Phone), enter the password to be used to authenticate
the Open SIP Phone connected to the system. SARVAM UCS validates the Open SIP Phone on the basis
of the SIP ID and HTTP Authentication Password (Third Party IP-Phone) and provides configuration on
validation.

To avoid unauthorized access, we recommend you to change the password regularly. Make sure it is
strong and is kept confidential. The password must:
Default: Blank.
To provide additional security, when the HTTP Authentication fails 5 times consecutively within 10 minutes
due to wrong Authentication ID / Authentication Password, the system will block the IP Address for
configuration of this Open SIP Phone. However, you can register again after 10 minutes. This activity will
be logged in the “System Activity Log” as well as “Simple Network Management Protocol (SNMP)”.
Make sure, you have configured SIP ID as HTTP User Name and HTTP Authentication Password as HTTP
Password in the Open SIP Phone.
• In Call Appearances, define the maximum number134 of simultaneous calls that the SIP Extension should
be allowed to make/receive. You can set up to 10 call appearances for a SIP Extension. Default: 2.
When Call Appearance is set to 2, the SIP Extension can make/receive 2 calls at a time.
• INVITE Request
• If you are going to register this SIP Extension with the same SIP ID at more than one location135, you may
enable the Shared Call Appearance Subscription check box on this SIP Extension. Default: Disabled.
Shared Call Appearance provides notification on call states to all the Open SIP Phones with the same SIP
ID at different locations. To know more about this feature, see “Shared Call Appearance”.
• To provide voice mail facility to the SIP Extension, select the Voice Mail Subscription check box. Default:
disabled.
• To allow the SIP Extension to monitor the status of another extension or Trunk, select the Busy Lamp
Field136 Subscription check box. Default: disabled. See “Busy Lamp Field for Trunks” to know more.
you purchase.
135. SARVAM UCS allows you to register SIP Extension with the same SIP ID at three different locations.
136. Busy Lamp Field (BLF), a typical feature supported by the system and Key Telephone Systems, is also supported on SIP Exten-
sions.
In the system and Key Telephone Systems, this feature is typically used by the Operator to monitor the status of another exten-
sion, that is, whether it is available, ringing or busy. The status of the other extensions is indicated on the special function keys pro-
grammed on the Operator's console. This helps the Operator decide whether to place the call, or transfer the call to that extension,
or pick up the call ringing on that extension. With BLF Subscription enabled on the SIP Extension, the user can monitor the status
of another extension or trunk.

When extension's state is changed from Ringing (early state as defined in BLF) to Mature (confirm state,
as defined in BLF) state, because of implementation of SARVAM UCS, it will send 'Terminate' state while
moving from ringing to mature state. The interpretation of terminate message will vary from terminal to
terminal.
• To allow the SIP Extension to view the status of other SIP-enabled Terminals, whether they are available
or not, select the Presence Subscription check box. Default: disabled.
The SIP Extension, for which you have enabled Presence Subscription, will be able to view Presence of
only those SIP Extensions which have PUBLISH enabled.
• To allow the SIP Extension to publish presence using the PUBLISH feature supported by the SIP
Extension, select the PUBLISH Enable check box. Default: disabled.
By default, Authentication for PUBLISH message is enabled. You may disable if you do not want to use
Authentication.
You must configure the Authentication ID, if you have enabled both Publish and Authentication.
options:
made to such users.
• Key Templates are not applicable to Open SIP Phones registered with SARVAM UCS.
SIP Trunks.

field.

Template field.

customized.


digits.
characters.
Call Pick Up allows the SIP Extension to 'pick up' (answer) calls ringing on any other extension, by dialing
a feature code, without physically going to the ringing extension. It also allows incoming calls for the SIP
Extension to be answered by the other extensions assigned the same Call Pick-Up group.

same.
Configuring Open SIP Phones using Manual Provisioning

To be able to use Open SIP Phones/devices with SARVAM UCS, you must configure the following:
• Configure the SIP Extension Settings. For details, see “Configuring SIP Extension Settings using
Jeeves”.137
• Configure specific parameters (for example, SIP ID/User Name, Authentication ID, Password, Server
Address/Domain Name etc.) in the Open SIP Phone/device which are required to register it with system.
137. Some of the parameters may not be applicable depending on the Open SIP Phone you have connected to the system.

For more information, refer the product documentation supplied with the Open SIP Phone/device you want
to use.
The parameters of Location 1, 2 and 3 are not applicable to Open SIP Phones.
Configuring Open SIP Phones using Auto Provisioning

SARVAM UCS supports the following third party Open SIP Phones for Auto Provisioning-
• Panasonic UTG200B
• Grandstream GXP110x
• Grandstream GXP2200
• Grandstream GXP21xx/116x/14xx
• Grandstream GXV3140/3175
• Yealink SIP-T19P/SIP-T21P/T22P/T26P/T28P/T4X
• Yealink SIP-T20P
• Yealink SIP-T3XG
• Cisco SPA50xG/51xG SIP Phone
• Cisco SPA525G SIP Phone
• Polycom IP Phone
• Snom IP Phone
• Htek 802
Connecting the Open SIP Phone

To be able to configure and register the Open SIP Phone with SARVAM UCS using Auto Provisioning,
• Connect and reboot the Open SIP Phone.
• Make sure that DHCP is selected as the Connection Type in the Open SIP Phone.
• For Auto Provisioning, you must configure the Open SIP phones at Location1 only.
• The phone will automatically fetch the configuration file(s) from SARVAM UCS and will get registered.
Using any third party DHCP Server in your LAN

You can use any third party DHCP Server in your LAN for assigning the ‘Auto Provisioning Server Address’ and
‘Server Port’ to the Open SIP Phones.
• Make sure that the third party DHCP Server and your Open SIP Phone, both are connected in the same
subnet as that of the LAN Port of SARVAM UCS.
• Select DHCP Option 66, and configure Data type in that third party DHCP Server in the following format -
http://IP Address:SPARSH Port.
If you are using Auto Provisioning and if the Open SIP Phone has multiple SIP accounts, it is recommended
to keep the first SIP account in default settings. After Auto Provisioning, the old configuration, if present for
this account will be deleted automatically and it will get registered with the SARVAM UCS.
Configuring Panasonic Open SIP Phones

You are recommended to complete the following steps before connecting Panasonic UTG200B:

• Decide the physical location of the Open SIP Phone. It must be connected in the same subnet as that of
the LAN Port of SARVAM UCS.
• Configure the settings in the third party DHCP Server. For instructions, see “Using any third party DHCP
Server in your LAN”.
• Configure the SIP Extension Settings in SARVAM UCS. For details, see “Configuring SIP Extension
Settings using Jeeves”138
• Configure the device specific settings applicable to your Panasonic UTG200B at any one of the
Location1, 2 or 3 on the SIP Extensions page. To do so,
• Click SIP Extension Settings link.

• Click Location 1 and configure the following parameters.
Applicable Open
Parameter Description/Action to be Taken
SIP Phone Variant
Enable Panasonic
Select the Enable Device check box.
Device UTG200B
Configure the Location Name to identify the phone. The

Location Panasonic
Location Name can be a maximum of 18 characters. Default:
Name UTG200B
Blank.
Panasonic
Device Type Select Device Type as Panasonic UTG200B.
UTG200B
Enter the MAC Address of the Panasonic phone to be

connected at this location. Default: Blank.
MAC Panasonic
Address UTG200B
SARVAM UCS validates the phone on the basis of the MAC
Address and provides configuration on validation.
Select the Authenticate HTTP Provisioning request check

box to authenticate HTTP request for security reason.
Default: Disabled.
If you are using the Open SIP Phone from the Public IP
Authenticate Address, we recommend you to enable this option.
Panasonic
HTTP
UTG200B
Provisioning SARVAM UCS validates the Open SIP Phone on the basis of
request the SIP ID and HTTP Authentication Password and provides
Make sure, you have configured SIP ID as HTTP User Name

and HTTP Authentication Password as HTTP Password in
the Open SIP Phone.
138. Some of the parameters may not be applicable depending on the Open SIP Phone you have connected. Please refer the specific
Open SIP Phone manufacturer’s documentation for more details.

Select the appropriate Registrar Server Address according to
Registrar
your installation scenarioa to register the Open SIP Phone Panasonic
Server
with the SIP Registrar of SARVAM UCS. Default: Use WAN UTG200B
Address
Port IP Address.
Displays the Open SIP Phone language and it is non- Panasonic

Language
editable. UTG200B
Configure the User Passwordb of the Open SIP Phone. It can

User Panasonic
be a minimum of 6 characters to a maximum of 16
Password UTG200B
characters. Default: userpass
Configure the Admin Passwordc of the Open SIP Phone. It

Admin Panasonic
can be a minimum of 6 characters to a maximum of 16
Password UTG200B
characters. Default: adminpass
Select the Time Zone of the region/country where the Open Panasonic
Time Zone
SIP Phone is installed. Default: GMT + 05:30 UTG200B
If DST is applicable, select the Enable DST check box. A list

Enable Panasonic
of DST parameters appear. Configure them as per your
DST UTG200B
requirement. Default: Disabled.
DST Offset Configure the DST Offset in minutes. Valid Range: 0 to 720 Panasonic
(min) minutes. Default: 60 minutes. UTG200B
Daylight
Saving Time Start Day To configure the time from when DST should be applied in
Panasonic
and Time the year, select the Month, Day, Week/Ordinal and Time
UTG200B
of DST (hh:mm) from the corresponding list boxes respectively.
End Day To configure the time when DST should end, select the
Panasonic
and Time Month, Day, Week/Ordinal and Time (hh:mm) from the
UTG200B
of DST corresponding list boxes respectively.
Primary NTP Select the Primary NTP Server Address with which you want
Server to synchronize the Date and Time of the Open SIP Phone.
Panasonic
Default: None. Make sure you configure the desired NTP
UTG200B
Server Address/es in the “Third Party IP-Phone General
Parameters”.
a. If the Open SIP phone is in the same network (LAN) as SARVAM UCS, select Use LAN Port IP Address as
If the Open SIP phone is in the Global Network and SARVAM UCS is connected to Internet over WAN, select
Use WAN Port IP Address as Registrar Server Address
If the Open SIP Phone is connected in the Global Network and SARVAM UCS is located behind a NAT Router,
and STUN is programmed, select Use Router/STUN's IP Address as Registrar Server Address. Make sure the
Router’s Public IP Address is configured in the Network Parameters.
If Dynamic DNS is configured in the Network Parameters, select Use Dynamic DNS Host Name as Registrar
Server Address.
b. To avoid unauthorized access, we recommend you to change the User Password regularly. Make sure it is
strong and is kept confidential.
c. To avoid unauthorized access, we recommend you to change the Admin Password regularly. Make sure it
is strong and is kept confidential.

Configuring Grandstream Open SIP Phones
You are recommended to complete the following steps before connecting any of the Grandstream Open SIP
Phones:
• Configure the device specific settings applicable to your Grandstream at any one of the Location1, 2 or 3
on the SIP Extensions page. To do so,
Parameter Description/Action to be Taken Applicable Open SIP Phone Variant
1. Grandstream GXP110x
2. Grandstream GXP2200
Enable Device Select the Enable Device check box. 3. Grandstream GXP21xx/116x/
14xx
4. Grandstream GXV3140/3175
Configure the Location Name to identify the Open 2. Grandstream GXP2200
Location Name SIP Phone. The Location Name can be a maximum 3. Grandstream GXP21xx/116x/
of 18 characters. Default: Blank. 14xx
Select Device Type as any of the desired
Device Type 3. Grandstream GXP21xx/116x/
Grandstream Open SIP Phone you want to connect.
14xx
Enter the MAC Address of the Grandstream phone to

be connected at this location. Default: Blank.
MAC Address 3. Grandstream GXP21xx/116x/
SARVAM UCS validates the phone on the basis of
14xx
the MAC Address, and provides configuration on
validation.

Select the Authenticate HTTP Provisioning request
check box to authenticate HTTP request for security
reason. Default: Disabled.
If you are using the Open SIP Phone from the Public
Authenticate IP Address, we recommend you to enable this option.
HTTP
3. Grandstream GXP21xx/116x/
Provisioning SARVAM UCS validates the Open SIP Phone on the
14xx
request basis of the SIP ID and HTTP Authentication
Password and provides configuration on validation.
Make sure, you have configured SIP ID as HTTP

User Name and HTTP Authentication Password as
HTTP Password in the Open SIP Phone.
Select the appropriate Registrar Server Address
Registrar Server according to your installation scenarioa to register the
3. Grandstream GXP21xx/116x/
Address Open SIP Phone with the SIP Registrar of SARVAM
14xx
UCS. Default: Use WAN Port IP Address.
Select the desired Open SIP Phone language.
Language 3. Grandstream GXP21xx/116x/
Default: English
14xx
Select the Send Phone Book check box to enable 1. Grandstream GXP2200
Send Phone downloading of the Phone Book (consisting of 2. Grandstream GXP21xx/116x/
Book Extension and Global Directory Contacts) from the 14xx
SARVAM UCS. 3. Grandstream GXV3140/3175
Configure the Phone Book Download interval in 1. Grandstream GXP2200

Phone Book
minutes. Valid Range: 0 - 720 minutes. Default: 60 2. Grandstream GXP21xx/116x/
Download
minutes 14xx
interval (min)
Configure the User Passwordb of the Open SIP 2. Grandstream GXP2200
User Password Phone. It can be a minimum of 1 character to a 3. Grandstream GXP21xx/116x/
maximum of 16 characters. Default: user 14xx
c 2. Grandstream GXP2200
Configure the Admin Password of the Open SIP
Admin
Phone. It can be a minimum of 1 character to a 3. Grandstream GXP21xx/116x/
Password
maximum of 16 characters. Default: admin 14xx
Select the Time Zone of the region/country where the 2. Grandstream GXP21xx/116x/
Time Zone
Open SIP Phone is installed. Default: GMT + 05:30 14xx

Date Display Select the Date Display Format for the Open SIP 2. Grandstream GXP21xx/116x/
Format Phone. Default: yyyy-mm-dd 14xx
Time Display Select the Time Display Format for the Open SIP 2. Grandstream GXP21xx/116x/
Format Phone. Default: 24Hr 14xx
Select the Primary NTP Server Address with which 1. Grandstream GXP110x
you want to synchronize the Date and Time of the 2. Grandstream GXP2200
Primary NTP
Open SIP Phone. Default: None. Make sure you 3. Grandstream GXP21xx/116x/
Server
configure the desired NTP Server Address/es in the 14xx
“Third Party IP-Phone General Parameters”. 4. Grandstream GXV3140/3175
Server Address.
Configuring Yealink Open SIP Phones

You are recommended to complete the following steps before connecting any of the Yealink Open SIP Phones:
• Configure the device specific settings applicable to your Yealink at any one of the Location1, 2 or 3 on the
SIP Extensions page. To do so,

Applicable Open SIP Phone

Variant
1. Yealink SIP-T19P/SIP-
T21P/T22P/T26P/T28P/T4X
Enable Device Select the Enable Device check box.
2. Yealink SIP-T20P
3. Yealink SIP-T3XG
Configure the Location Name to identify the
Location T21P/T22P/T26P/T28P/T4X
Open SIP Phone. The Location Name can be a
Name 2. Yealink SIP-T20P
maximum of 18 characters. Default: Blank.
3. Yealink SIP-T3XG
Select Device Type as any of the desired T21P/T22P/T26P/T28P/T4X
Device Type
Yealink Open SIP Phone you want to connect. 2. Yealink SIP-T20P
3. Yealink SIP-T3XG
Enter the MAC Address of the Yealink phone

to be connected at this location. Default: Blank. 1. Yealink SIP-T19P/SIP-
MAC Address
SARVAM UCS validates the phone on the 2. Yealink SIP-T20P
basis of the MAC Address, and provides 3. Yealink SIP-T3XG
Select the Authenticate HTTP Provisioning

request check box to authenticate HTTP
request for security reason. Default: Disabled.
If you are using the Open SIP Phone from the

Public IP Address, we recommend you to
enable this option. 1. Yealink SIP-T19P/SIP-
Authenticate
HTTP
SARVAM UCS validates the Open SIP Phone 2. Yealink SIP-T20P
Provisioning
on the basis of the SIP ID and HTTP 3. Yealink SIP-T3XG
request
Authentication Password and provides
Make sure, you have configured SIP ID as

HTTP User Name and HTTP Authentication
Password as HTTP Password in the Open SIP
Phone.
Select the appropriate Registrar Server

Registrar Address according to your installation
Server scenarioa to register the Open SIP Phone with
2. Yealink SIP-T20P
Address the SIP Registrar of SARVAM UCS. Default:
3. Yealink SIP-T3XG
Use WAN Port IP Address.
Select the Send Phone Book check box to

enable downloading of the Phone Book
Send Phone T21P/T22P/T26P/T28P/T4X
(consisting of Extension and Global Directory
Book 2. Yealink SIP-T20P
Contacts) from the SARVAM UCS. Default:
3. Yealink SIP-T3XG
Enabled

Web User Select the desired language in which the Web
Interface User Interface of the selected Open SIP Phone
2. Yealink SIP-T20P
Language variant should be displayed. Default: English
3. Yealink SIP-T3XG
Select the desired language in which the Open
Phone User T21P/T22P/T26P/T28P/T4X
SIP Phone’s User Interface should be
Interface 2. Yealink SIP-T20P
displayed. Default: English
3. Yealink SIP-T3XG
Configure the User Passwordb of the Open SIP
User T21P/T22P/T26P/T28P/T4X
Phone. It can be maximum of 16 characters
Password 2. Yealink SIP-T20P
long. Default: user
3. Yealink SIP-T3XG
Configure the Admin Passwordc of the Open
Admin T21P/T22P/T26P/T28P/T4X
SIP Phone. It can be a maximum of up to 16
Password 2. Yealink SIP-T20P
characters. Default: admin
3. Yealink SIP-T3XG
Select the Time Zone of the region/country
Time Zone where the Open SIP Phone is installed.
2. Yealink SIP-T20P
Default: + 05:30 India (Calcutta)
3. Yealink SIP-T3XG
Daylight Select the Daylight Saving Time Mode that
Saving Time should be applied to the selected Open SIP
2. Yealink SIP-T20P
Mode Phone. Default: Automatic.
3. Yealink SIP-T3XG
Select the DST Type that should be applied to
DST Type the Open SIP Phone, either DST by Date or
2. Yealink SIP-T20P
DST by Week.
3. Yealink SIP-T3XG
Configure the time from when DST should be 1. Yealink SIP-T19P/SIP-

applied in the year by selecting the Month, Day T21P/T22P/T26P/T28P/T4X
Start Date
and Hour from the corresponding list boxes 2. Yealink SIP-T20P
respectively. 3. Yealink SIP-T3XG
Configure the time when DST should end by
If DST Type = T21P/T22P/T26P/T28P/T4X
End Date selecting the Month, Day and Hour from the
DST by Date 2. Yealink SIP-T20P
corresponding list boxes respectively.
3. Yealink SIP-T3XG
Offset Configure the DST Offset timer value in T21P/T22P/T26P/T28P/T4X
(minutes) minutes. Valid Range:-300 to +300. 2. Yealink SIP-T20P
3. Yealink SIP-T3XG

DST Start Select the Month from when DST should be T21P/T22P/T26P/T28P/T4X
Month applied. 2. Yealink SIP-T20P
3. Yealink SIP-T3XG
DST Start
Select the Day of Week from when DST should T21P/T22P/T26P/T28P/T4X
Day of
be applied. 2. Yealink SIP-T20P
Week
3. Yealink SIP-T3XG
DST Start 1. Yealink SIP-T19P/SIP-

Day of Select the DST Start Day of Week Last in T21P/T22P/T26P/T28P/T4X
Week Last Month. 2. Yealink SIP-T20P
in Month 3. Yealink SIP-T3XG
Start Hour T21P/T22P/T26P/T28P/T4X
Select the DST Start Hour of the Day.
of Day 2. Yealink SIP-T20P
If DST Type = 3. Yealink SIP-T3XG
DST by Week 1. Yealink SIP-T19P/SIP-
DST Stop T21P/T22P/T26P/T28P/T4X
Select the Month when DST should end.
Month 2. Yealink SIP-T20P
3. Yealink SIP-T3XG
DST Stop
Day of Select the Day of Week when DST should end.
2. Yealink SIP-T20P
Week
3. Yealink SIP-T3XG
DST Stop 1. Yealink SIP-T19P/SIP-

Day of Select the DST Stop Day of Week Last in T21P/T22P/T26P/T28P/T4X
Week Last Month. 2. Yealink SIP-T20P
in Month 3. Yealink SIP-T3XG
End Hour of T21P/T22P/T26P/T28P/T4X
Select the DST End Hour of the Day.
Day 2. Yealink SIP-T20P
3. Yealink SIP-T3XG
Date Display Select the Date Display Format for the Open T21P/T22P/T26P/T28P/T4X
Format SIP Phone. Default: 2. Yealink SIP-T20P
3. Yealink SIP-T3XG
Select the Time Display Format for the Open
Time Display T21P/T22P/T26P/T28P/T4X
SIP Phone. Default for T20P Phone: MM DD
Format 2. Yealink SIP-T20P
YY and for other phones WWW MMM DD
3. Yealink SIP-T3XG
Primary NTP Select the Primary NTP Server Address with

Server which you want to synchronize the Date and 1. Yealink SIP-T19P/SIP-
Time of the Open SIP Phone. Default: None. T21P/T22P/T26P/T28P/T4X
Make sure you configure the desired NTP 2. Yealink SIP-T20P
Server Address/es in the “Third Party IP-Phone 3. Yealink SIP-T3XG
General Parameters”.

Server Address.
Configuring Cisco Open SIP Phones

You are recommended to complete the following steps before connecting any of the Cisco Open SIP Phones:
• Configure the device specific settings applicable to your Cisco at any one of the Location1, 2 or 3 on the

Variant
1. Cisco SPA50xG/51xG
SIP Phone
Enable Device Select the Enable Device check box.
2. Cisco SPA525G SIP
Phone
SIP Phone
Location Name Open SIP Phone. The Location Name can be
a maximum of 18 characters. Default: Blank.
Phone

Select Device Type as any of the desired SIP Phone
Device Type
Cisco Phone you want to connect. 2. Cisco SPA525G SIP
Phone
Enter the MAC Address of the Cisco phone to

be connected at this location. Default: Blank. 1. Cisco SPA50xG/51xG
SIP Phone
MAC Address
SARVAM UCS validates the phone on the 2. Cisco SPA525G SIP
basis of the MAC Address, and provides Phone


enable this option.
Authenticate 1. Cisco SPA50xG/51xG
HTTP SIP Phone
SARVAM UCS validates the Open SIP Phone
Provisioning 2. Cisco SPA525G SIP
on the basis of the SIP ID and HTTP
request Phone

Password as HTTP Password in the Open
SIP Phone.

Address according to your installation
Registrar SIP Phone
scenarioa to register the Open SIP Phone
Server Address 2. Cisco SPA525G SIP
with the SIP Registrar of SARVAM UCS.
Phone
Default: Use WAN Port IP Address.

Send Phone SIP Phone
(consisting of Extension and Global Directory
Book 2. Cisco SPA525G SIP
Phone
Enabled
Configure the Name of the Phonebook. This 1. Cisco SPA50xG/51xG

Phonebook name will be displayed on the Phone LCD. It SIP Phone
Name can be of maximum 32 characters. Default: 2. Cisco SPA525G SIP
Corporate Directory Phone
Configure the User Passwordb of the Open 1. Cisco SPA50xG/51xG

SIP Phone. It can be a minimum of 1 SIP Phone
User Password
character to a maximum of 16 characters. 2. Cisco SPA525G SIP
Default: Blank Phone
Configure the Admin Passwordc of the Cisco
Admin SIP Phone
phone. It can be a minimum of 1 character to
Password 2. Cisco SPA525G SIP
a maximum of 16 characters. Default: Blank
Phone

Phone User Select the desired language in which the
SIP Phone
Interface Open SIP Phone’s User Interface should be
Language displayed. Default: English-US
Phone
Select the path from which the Open SIP

Phone must fetch the Language files. Default:
Language File SIP Phone
None. Make sure you configure the desired
Download Path 2. Cisco SPA525G SIP
Path (Server Address/es) in the “Third Party
Phone
IP-Phone General Parameters”.
SIP Phone
Time Zone where the Open SIP Phone is installed.
Default: GMT + 05:30
Phone
If DST is applicable, select the Enable DST 1. Cisco SPA50xG/51xG

check box. A list of DST parameters appear. SIP Phone
Enable DST
Configure them as per your requirement. 2. Cisco SPA525G SIP
Default: Disabled Phone
DST Start Select the Day of Week from when DST SIP Phone
Day of Week should be applied. 2. Cisco SPA525G SIP
Phone
DST Start 1. Cisco SPA50xG/51xG

Day of Week Select the DST Start Day of Week Last in SIP Phone
Last in Month. 2. Cisco SPA525G SIP
Month Phone
SIP Phone
Start Hour of Select the DST Start Hour of the Day.
Day
Phone
Daylight Saving DST Stop SIP Phone
Select the Month when DST should end.
Time Month 2. Cisco SPA525G SIP
Phone
Select the Day of Week when DST should SIP Phone
DST Stop
end. 2. Cisco SPA525G SIP
Day of Week
Phone
DST Stop 1. Cisco SPA50xG/51xG

Day of Week Select the DST Stop Day of Week Last in SIP Phone
Last in Month. 2. Cisco SPA525G SIP
Month Phone
End Hour of SIP Phone
Select the DST End Hour of the Day.
Day 2. Cisco SPA525G SIP
Phone

Date Display Select the Date Display Format for the Open SIP Phone
Format SIP Phone. Default: month/day 2. Cisco SPA525G SIP
Phone
Time Display Select the Time Display Format for the Open SIP Phone
Format SIP Phone. Default: 12Hr 2. Cisco SPA525G SIP
Phone
Configure the DST Offset timer value in
Offset SIP Phone
minutes. Valid Range:-300 to +300. Default:
Timer(min) 2. Cisco SPA525G SIP
000
Phone
Select the Primary NTP Server Address with

which you want to synchronize the Date and 1. Cisco SPA50xG/51xG
Primary NTP Time of the Open SIP Phone. Default: None. SIP Phone
Server Make sure you configure the desired NTP 2. Cisco SPA525G SIP
Server Address/es in the “Third Party IP- Phone
Phone General Parameters”.
If the Open SIP Phone is connected in the Global Network and SARVAM UCS is located behind a NAT
Router, and STUN is programmed, select Use Router/STUN's IP Address as Registrar Server Address. Make
sure the Router’s Public IP Address is configured in the Network Parameters.
Server Address.
Configuring Polycom Open SIP Phones

You are recommended to complete the following steps before connecting any of the Polycom Open SIP Phones:

• Configure the device specific settings applicable to your Polycom at any one of the Location1, 2 or 3 on
the SIP Extensions page. To do so,

Variant
Enable Device Select the Enable Device check box. Polycom IP Phone

Location
Open SIP Phone. The Location Name can be a Polycom IP Phone
Name
Device Type Select Device Type as Polycom IP Phone. Polycom IP Phone
Enter the MAC Address of the Polycom phone

to be connected at this location. Default: Blank.
MAC Address Polycom IP Phone

SARVAM UCS validates the phone on the
basis of the MAC Address, and provides


enable this option.
Authenticate
HTTP
SARVAM UCS validates the Open SIP Phone Polycom IP Phone
Provisioning
request

Phone.

Server scenarioa to register the Open SIP Phone with Polycom IP Phone
Use WAN Port IP Address

Send Phone
(consisting of Extension and Global Directory Polycom IP Phone
Book
Enabled

Configure the User Passwordb of the Open SIP
User
Phone. It can be maximum of 16 characters Polycom IP Phone
Password
long. Default: 123
Configure the Admin Passwordc of the Open

Admin
SIP Phone. It can be a maximum of up to 16 Polycom IP Phone
Password
characters. Default: 456
Phone User Select the desired language in which the Open

Interface SIP Phone’s User Interface should be Polycom IP Phone
Language displayed. Default: English Internal (en-in)
If DST is applicable, select the Enable DST

check box. A list of DST parameters appear.
Enable DST Polycom IP Phone
Configure them as per your requirement.
Default: Disabled

DST Type the Open SIP Phone, either DST by Date or Polycom IP Phone
DST by Week.
Select the Month from when DST should be

Start Month Polycom IP Phone
applied.
Select the Day from when DST should be

Start Day Polycom IP Phone
applied.
If DST Type = Select the Hour from when DST should be
DST by Date Start Hour Polycom IP Phone
applied.
End Month Select the Month when DST should end. Polycom IP Phone
End Day Select the Day when DST should end. Polycom IP Phone
End Hour Select the Hour when DST should end. Polycom IP Phone

DST Start Select the Month from when DST should be
Polycom IP Phone
Month applied.
DST Start
Select the Day of Week from when DST should
Day of Polycom IP Phone
be applied.
Week
DST Start
Day of Select the DST Start Day of Week Last in
Polycom IP Phone
Week Last Month.
in Month
Start Hour
Select the DST Start Hour of the Day. Polycom IP Phone
If DST Type = of Day
DST by Week DST Stop
Select the Month when DST should end. Polycom IP Phone
Month
DST Stop
Day of Select the Day of Week when DST should end. Polycom IP Phone
Week
DST Stop
Day of Select the DST Stop Day of Week Last in
Polycom IP Phone
Week Last Month.
in Month
End Hour of
Select the DST End Hour of the Day. Polycom IP Phone
Day
Date Display Select the Date Display Format for the Open
Polycom IP Phone
Format SIP Phone. Default: 1 Jan, Mon
Time Display Select the Time Display Format for the Open
Polycom IP Phone
Format SIP Phone. Default: 24Hr

Server which you want to synchronize the Date and
Time of the Open SIP Phone. Default: None.
Polycom IP Phone
Make sure you configure the desired NTP
Server Address/es in the “Third Party IP-Phone
Server Address.

Configuring Snom Open SIP Phones
You are recommended to complete the following steps before connecting any of the Snom Open SIP Phones:
• Configure the device specific settings applicable to your Snom at any one of the Location1, 2 or 3 on the
Enable Device Select the Enable Device check box. Snom IP Phone
Configure the Location Name to identify the Open

Location Name SIP Phone. The Location Name can be a maximum Snom IP Phone
of 18 characters. Default: Blank.
Device Type Select Device Type as Snom IP Phone. Snom IP Phone
Enter the MAC Address of the Snom phone to be

MAC Address Snom IP Phone

validation.

HTTP
Snom IP Phone


Snom IP Phone
Select the Send Phone Book check box to enable

Send Phone downloading of the Phone Book (consisting of
Snom IP Phone
Book Extension and Global Directory Contacts) from the
SARVAM UCS. Default: Enabled
Web User Select the desired language in which the Web User
Interface Interface of the selected Open SIP Phone should be Snom IP Phone
Language displayed. Default: English
Select the path from which the Open SIP Phone must
Web User
fetch the Language files. Default: None. Make sure
Language File Snom IP Phone
you configure the desired Path (Server Address/es) in
Download Path
the “Third Party IP-Phone General Parameters”.
Phone User Select the desired language in which the Open SIP
Interface Phone’s User Interface should be displayed. Default: Snom IP Phone
Language English
Select the path from which the Open SIP Phone must
Phone User
fetch the Language files. Default: None. Make sure
Language File Snom IP Phone
you configure the desired Path (Server Address/es) in
Download Path
the “Third Party IP-Phone General Parameters”.
Select the Time Zone of the region/country where the

Time Zone Snom IP Phone
Open SIP Phone is installed. Default: India
Date Display Select the Date Display Format for the Open SIP
Snom IP Phone
Format Phone. Default: mm/dd
Time Display Select the Time Display Format for the Open SIP
Snom IP Phone
Format Phone. Default: 12Hr
Call Progress Select the region to apply the Call Progress Tone
Snom IP Phone
Tone prevailing there. Default: India.
Configure the HTTP Server Login User Name for web

HTTP Server
access. It can be a maximum of 16 characters. Snom IP Phone
Login Username
Default: Blank
Configure the HTTP Server Login Passwordb for web

HTTP Server
access. It can be a maximum of 16 characters. Snom IP Phone
Login Password
Default: Blank
Select Admin Mode check box to allow its access to

Admin Mode Snom IP Phone
the user. Default: Enabled
Configure the Admin Passwordc of the Open SIP

Admin Mode
Phone. It can be a minimum of 1 character to a Snom IP Phone
Password
maximum of 16 characters. Default: 0000

Select the Primary NTP Server Address with which
you want to synchronize the Date and Time of the
Primary NTP
Open SIP Phone. Default: None. Make sure you Snom IP Phone
Server
configure the desired NTP Server Address/es in the
“Third Party IP-Phone General Parameters”.
a. If the phone is in the same network (LAN) as SARVAM UCS, select Use LAN Port IP Address as Registrar
Server Address.
If the phone is in the Global Network and SARVAM UCS is connected to Internet over WAN, select Use WAN
Port IP Address as Registrar Server Address
If the phone is connected in the Global Network and SARVAM UCS is located behind a NAT Router, and
STUN is programmed, select Use Router/STUN's IP Address as Registrar Server Address. Make sure the
Server Address.
b. To avoid unauthorized access, we recommend you to change the HTTP Server Login Password regularly.
Make sure it is strong and is kept confidential.
c. To avoid unauthorized access, we recommend you to change the Admin Mode Password regularly. Make
sure it is strong and is kept confidential.
Configuring Matrix SPARSH VP110 Open SIP Phone

You are recommended to complete the following steps before connecting any of the Matrix SPARSH VP110 Open
SIP Phones:
• Configure the device specific settings applicable to your Matrix SPARSH VP110 at any one of the

Variant
Select the Enable Device check box. Default:

Enable Device Matrix SPARSH VP110
Disabled.

Location
Open SIP Phone. The Location Name can be a Matrix SPARSH VP110
Name
Select Device Type as MATRIX SPARSH

Device Type Matrix SPARSH VP110
VP110.
Enter the MAC Address of the SPARSH VP110

phone to be connected at this location. Default:
Blank.
MAC Address Matrix SPARSH VP110


enable this option.
Authenticate
HTTP
SARVAM UCS validates the Open SIP Phone Matrix SPARSH VP110
Provisioning
request

Phone.

Server scenarioa to register the Open SIP Phone with Matrix SPARSH VP110

Send Key
apply the Key Settingsb to the Phone. Default: Matrix SPARSH VP110
Settings
Enabled
Select the desired Dial Plan. Default: 1
Dial Planc Matrix SPARSH VP110

The Phone will detect end of dialing as per the
rules configured in the Dial Plan selected here.
Transport Select the protocol to be used to transport the

Matrix SPARSH VP110
Mode SIP messages. Default: UDP
Select the Enable SRTP? check box for secure

Enable SRTP? Matrix SPARSH VP110
conversations over SIP. Default: Disabled
Enter the desired SIP DiffServe/ToS to set the

SIP DiffServe/
Quality of Service (QoS) for SIP packets Matrix SPARSH VP110
ToS
Default: 26

Enter the desired RTP DiffServe/ToS to set the
RTP DiffServe/
Quality of Service (QoS) for RTP packets. Matrix SPARSH VP110
ToS
Default: 46
Enter the port on which the phone will listen for

SIP messages. This port is used as source port
SIP Port in SIP messages.This port is also used to send Matrix SPARSH VP110
SIP messages to the remote peer. Default:
5060
To define a range of RTP ports, configure the
Min RTP Port Matrix SPARSH VP110
minimum local RTP port. Default: 11780

Max RTP Port Matrix SPARSH VP110
maximum local RTP port. Default: 11800
Select the Allow HTTP check box to enable

Allow HTTP Matrix SPARSH VP110
HTTP Web access. Default: Enabled
Enter the port number on which the HTTP

HTTP Port Matrix SPARSH VP110
access is to be given. Default: 80
Select the Allow HTTPS check box to enable

Allow HTTPS Matrix SPARSH VP110
HTTPS Web access. Default: Enabled
Enter the port number on which the HTTPS

HTTPS Port Matrix SPARSH VP110
Select the type of contacts (Extension or/and

Global Directory Contacts) the phone must
download from the SARVAM UCS.
Local Phone
Matrix SPARSH VP110
Bookd These will be stored in the phone’s Local
Phone Book. Default: Do not send

Remote Phone
Matrix SPARSH VP110
Book
These will be stored in the phone’s Remote
Phone Book. Default: Send first Extension
Numbers, remaining Global Directory Numbers

Interface User Interface should be displayed. Default: Matrix SPARSH VP110
Language English

Interface Phone’s User Interface should be displayed. Matrix SPARSH VP110
Language Default: English
Configure the User Passworde of the Open SIP

User
Phone. It can be maximum of 16 characters Matrix SPARSH VP110
Password
long. Default: user

Configure the Admin Passwordf of the Open
Admin
SIP Phone. It can be a maximum of up to 16 Matrix SPARSH VP110
Password
When in Headset mode, select the ring

Ringer Device
destination for the SPARSH VP110. Default: Matrix SPARSH VP110
for Headset
Use Speaker
Select the Enable Distinctive Ring check box,

to set different ringing patterns to distinguish
between different types of call events. The
following types of call events are supported:
• Internal Call
• Trunk Call
Enable • Auto Call Back
Distinctive • Auto Redial Matrix SPARSH VP110
Ring • Alarm
• Emergency
• Operator Alarm
• Message Wait
• Priority
• Emergency Conference
Default: Disabled
Call Progress Select the region to apply the Call Progress

Matrix SPARSH VP110
Tone Tone prevailing there. Default: Custom.

Time Zone where the Open SIP Phone is installed. Matrix SPARSH VP110

Saving Time should be applied to the selected Open SIP Matrix SPARSH VP110

DST Type the Open SIP Phone, either DST by Date or Matrix SPARSH VP110
DST by Week. Default: DST by Date

Start Month Matrix SPARSH VP110
applied.

Start Day Matrix SPARSH VP110
applied.
Select the Hour from when DST should be

Start Hour Matrix SPARSH VP110
applied.
If DST Type =
DST by Date End Month Select the Month when DST should end. Matrix SPARSH VP110
End Day Select the Day when DST should end. Matrix SPARSH VP110
End Hour Select the Hour when DST should end. Matrix SPARSH VP110

Offset Timer
minutes. Valid Range:-300 to +300. Default: Matrix SPARSH VP110
(min)
Blank

Matrix SPARSH VP110
Month applied.
DST Start
Day of Matrix SPARSH VP110
be applied.
Week
DST Start
Matrix SPARSH VP110
Week Last Month.
in Month
Start Hour
Select the DST Start Hour of the Day. Matrix SPARSH VP110
If DST Type = of Day
DST by Week DST Stop
Select the Month when DST should end. Matrix SPARSH VP110
Month
DST Stop
Day of Select the Day of Week when DST should end. Matrix SPARSH VP110
Week
DST Stop
Matrix SPARSH VP110
Week Last Month.
in Month
End Hour of
Select the DST End Hour of the Day. Matrix SPARSH VP110
Day
Matrix SPARSH VP110
Format SIP Phone. Default: WWW MMM DD
Matrix SPARSH VP110
Format SIP Phone. Default: 24 Hr

Matrix SPARSH VP110
Server Address.

b. To apply the desired Key Template:
i) Select the Send Key Settings check box.
ii) Click Key Template under Configuration.
iii) Click the desired key template - Operator/ Executive/ Hotel Attendant/Guest of VP110. Assign features
facilities to the keys in this template as per your requirement.
For instructions, see “Customizing Extended IP Phone Templates using Jeeves”
iv) Click SIP Extension Settings under VoIP Configuration. Select the SIP Extension Number on which
SPARSH VP110 is registered.
v) Under General Parameters, scroll to Key Template and select the template you configured as per your
requirement.
The Personalized option of Key Template is not applicable for SPARSH VP110.
c. If you want to apply the rules of the Dial Plan configured in SARVAM UCS, see “Dial Plan for SIP Extension”.
You can also configure rules for the Dial Plan from each phone. To do so, refer to the SPARSH VP110 User
Guide.
d. If an option other than "Do not send" is selected in Local Phone Book, it will overwrite all Local Phone Book's
contacts of the phone.
e. To avoid unauthorized access, we recommend you to change the User Password regularly. Make sure it is
f. To avoid unauthorized access, we recommend you to change the Admin Password regularly. Make sure it is
Configuring Htek 802 Open SIP Phones

You are recommended to complete the following steps before connecting the Htek 802 Open SIP Phones:
• Decide the physical location of the Open SIP Phone. It must be connected in the same subnet as that of the
LAN Port of SARVAM UCS.
Settings using Jeeves”
• Configure the device specific settings applicable to Htek at any one of the Location1, 2 or 3 on the SIP
Extensions page. To do so,
Enable Device Select the Enable Device check box. Htek 802 IP Phone
Configure the Location Name to identify the Open

Location Name SIP Phone. The Location Name can be a maximum Htek 802 IP Phone
of 18 characters. Default: Blank.

Device Type Select Device Type as Htek 802. Htek 802 IP Phone
Enter the MAC Address of the Htek 802 phone to be

MAC Address Htek 802 IP Phone

validation.

HTTP
Htek 802 IP Phone


Htek 802 IP Phone
Configure the Admin Passwordb of the Open SIP

Admin
Phone. It can be a maximum of up to 16 characters. Htek 802 IP Phone
Password
Default: admin
Assign Voice Enable this check box to assign the first

Htek 802 IP Phone
Mail Key programmable key of the phone as Voice Mail.
Enter the port on which the phone will listen for SIP
SIP Port messages. This port is used as source port in SIP Htek 802 IP Phone
messages.This port is also used to send SIP
messages to the remote peer. Default: 5060
Min RTP Port Htek 802 IP Phone

Max RTP Port Htek 802 IP Phone
Apply System Select this check box if you want to apply the
Htek 802 IP Phone
Time Zone System’s Time Zone to the Open SIP Phone.
Select the Time Zone of the region/country where the

Time Zone Htek 802 IP Phone
Open SIP Phone is installed. Default: GMT + 05:30
Select the Primary NTP Server Address with which

you want to synchronize the Date and Time of the
Primary NTP
Open SIP Phone. Default: None. Make sure you Htek 802 IP Phone
Server
configure the desired NTP Server Address/es in the
“Third Party IP-Phone General Parameters”.

Select the Daylight Saving Time Mode that should be
Daylight Saving
applied to the selected Open SIP Phone. Default: Htek 802 IP Phone
Time Mode
Automatic.
a. If the phone is in the same network (LAN) as SARVAM UCS, select Use LAN Port IP Address as Registrar
Server Address.
If the phone is in the Global Network and SARVAM UCS is connected to Internet over WAN, select Use WAN
Port IP Address as Registrar Server Address
If the phone is connected in the Global Network and SARVAM UCS is located behind a NAT Router, and
STUN is programmed, select Use Router/STUN's IP Address as Registrar Server Address. Make sure the
Server Address.
b. To avoid unauthorized access, we recommend you to change the Admin Password regularly. Make sure it is
Configuring Matrix SPARSH VP710 Open SIP Phone

You are recommended to complete the following steps before connecting any of the Matrix SPARSH VP710 Open
SIP Phones:
Settings using Jeeves”
• Configure the device specific settings applicable to your Matrix SPARSH VP710 at any one of the

Variant
Select the Enable Device check box. Default:

Enable Device Matrix SPARSH VP710
Disabled.

Location
Open SIP Phone. The Location Name can be a Matrix SPARSH VP710
Name
Select Device Type as MATRIX SPARSH

Device Type Matrix SPARSH VP710
VP710.

Enter the MAC Address of the SPARSH VP710
phone to be connected at this location. Default:
Blank.
MAC Address Matrix SPARSH VP710


enable this option.
Authenticate
HTTP
SARVAM UCS validates the Open SIP Phone Matrix SPARSH VP710
Provisioning
request

Phone.

Server scenarioa to register the Open SIP Phone with Matrix SPARSH VP710
Select the desired Dial Plan. Default: 1
Dial Planb Matrix SPARSH VP710

The Phone will detect end of dialing as per the
rules configured in the Dial Plan selected here.
Transport Select the protocol to be used to transport the

Matrix SPARSH VP710
Mode SIP messages. Default: UDP
Select the Enable SRTP? check box for secure

Enable SRTP? Matrix SPARSH VP710
conversations over SIP. Default: Disabled
Select the appropriate DTMF Type to

determine how the DTMF digits will be sent
DTMF Type Matrix SPARSH VP710
over the IP Network, when a DTMF digit is
pressed.
Enter the desired SIP DiffServe/ToS to set the

SIP DiffServe/
Quality of Service (QoS) for SIP packets Matrix SPARSH VP710
ToS
Default: 26
Enter the desired RTP DiffServe/ToS to set the

RTP DiffServe/
Quality of Service (QoS) for RTP packets. Matrix SPARSH VP710
ToS
Default: 46

SIP messages. This port is used as source port
SIP Port in SIP messages.This port is also used to send Matrix SPARSH VP710
SIP messages to the remote peer. Default:
5060
SIP messages transported over TLS. This port
SIP TLS Port is used as source port in SIP messages.This Matrix SPARSH VP710
port is also used to send SIP messages to the
remote peer. Default: 5061

Min RTP Port Matrix SPARSH VP710

Max RTP Port Matrix SPARSH VP710
Select the Allow HTTP check box to enable

Allow HTTP Matrix SPARSH VP710
HTTP Web access. Default: Enabled
Enter the port number on which the HTTP

HTTP Port Matrix SPARSH VP710
Select the Allow HTTPS check box to enable

Allow HTTPS Matrix SPARSH VP710
HTTPS Web access. Default: Enabled
Enter the port number on which the HTTPS

HTTPS Port Matrix SPARSH VP710

Local Phone download from the SARVAM UCS.
Matrix SPARSH VP710
Bookc
These will be stored in the phone’s Local
Phone Book. Default: Do not send
Enable this check box if you want to allow

usage of Personal Directory.
Send Personal
Matrix SPARSH VP710
Directory
These contacts will be stored in the phone’s
Local phone Book.

Remote Phone
Matrix SPARSH VP710
Book
These will be stored in the phone’s Remote
Phone Book. Default: Send first Extension
Numbers, remaining Global Directory Numbers

Interface User Interface should be displayed. Default: Matrix SPARSH VP710
Language English

Interface Phone’s User Interface should be displayed. Matrix SPARSH VP710
Language Default: English

Configure the User Passwordd of the Open SIP
User
Phone. It can be maximum of 16 characters Matrix SPARSH VP710
Password
long. Default: user
Configure the Admin Passworde of the Open

Admin
SIP Phone. It can be a maximum of up to 16 Matrix SPARSH VP710
Password
When in Headset mode, select the ring

Ringer Device
destination for the SPARSH VP710. Default: Matrix SPARSH VP710
for Headset
Use Speaker
Select the Enable Distinctive Ring check box,

to set different ringing patterns to distinguish
between different types of call events. The
following types of call events are supported:
• Internal Call
• Trunk Call
Enable • Auto Call Back
Distinctive • Auto Redial Matrix SPARSH VP710
Ring • Alarm
• Emergency
• Operator Alarm
• Message Wait
• Priority
• Emergency Conference
Default: Disabled
Call Progress Select the region to apply the Call Progress

Matrix SPARSH VP710
Tone Tone prevailing there. Default: Custom.

Time Zone where the Open SIP Phone is installed. Matrix SPARSH VP710

Saving Time should be applied to the selected Open SIP Matrix SPARSH VP710

DST Type the Open SIP Phone, either DST by Date or Matrix SPARSH VP710
DST by Week. Default: DST by Date

Start Month Matrix SPARSH VP710
applied.

Start Day Matrix SPARSH VP710
applied.
Select the Hour from when DST should be

Start Hour Matrix SPARSH VP710
applied.
If DST Type =
DST by Date End Month Select the Month when DST should end. Matrix SPARSH VP710
End Day Select the Day when DST should end. Matrix SPARSH VP710
End Hour Select the Hour when DST should end. Matrix SPARSH VP710

Offset
(minutes)
Blank

Offset
(minutes)
Blank

Matrix SPARSH VP710
Month applied.
DST Start
Day of Matrix SPARSH VP710
be applied.
Week
DST Start
Matrix SPARSH VP710
Week Last Month.
in Month
If DST Type =
Start Hour
DST by Week Select the DST Start Hour of the Day. Matrix SPARSH VP710
of Day
DST Stop
Select the Month when DST should end. Matrix SPARSH VP710
Month
DST Stop
Day of Select the Day of Week when DST should end. Matrix SPARSH VP710
Week
DST Stop
Matrix SPARSH VP710
Week Last Month.
in Month
End Hour of
Select the DST End Hour of the Day. Matrix SPARSH VP710
Day
Matrix SPARSH VP710
Format SIP Phone. Default: WWW MMM DD
Matrix SPARSH VP710
Format SIP Phone. Default: 24 Hr

Matrix SPARSH VP710
Server Address.
b. If you want to apply the rules of the Dial Plan configured in SARVAM UCS, see “Dial Plan for SIP Extension”.
You can also configure rules for the Dial Plan from each phone. To do so, refer to the SPARSH VP710 User
Guide.
c. If an option other than "Do not send" is selected in Local Phone Book, it will overwrite all Local Phone Book's
contacts of the phone.
d. To avoid unauthorized access, we recommend you to change the User Password regularly. Make sure it is
e. To avoid unauthorized access, we recommend you to change the Admin Password regularly. Make sure it is

Third Party IP-Phone General Parameters
To configure the Third Party IP-Phone General Parameters,
• Click Third Party IP-Phone General Parameters link.
The Third Party IP-Phone General Parameters page opens.
Primary NTP Server for all Third Party IP-Phones and Matrix SPARSH VP110
• Under NTP Server Address, configure the Primary NTP Server Address with which you want to
synchronize the Date and Time of the Open SIP Phones. Default: Blank.
You may configure maximum 5 different Server Addresses.
Language File Download Path for Cisco IP-Phone
• Under Path, configure the path from which you want the Open SIP Phone to fetch the Language files.
Default: Blank.
You may configure maximum 5 different Paths.
Phone User Language File Download Path for Snom IP-Phone
• Under Path, configure the path from which you want the Open SIP Phone to fetch the Language files for
the Phone user. Default: Blank.

Web User Language File Download Path for Snom IP-Phone
• Under Path, configure the path from which you want the Open SIP Phone to fetch the Language files for
the Web user. Default: Blank.
The Server Addresses/Path you configure on this page will appear in the Combo box of the respective parameter
on the SIP Extension page.

Device Management
SARVAM UCS supports Auto Detection and Auto Provisioning of third party IP Phones as well as Extended
Clients. Device Management is a touch-free, plug and play feature and is an ideal solution for a large deployment of
phones. To use this feature you must plug all the IP phones in same network (recommended) and also make sure
that these phones are registered as SIP Extensions at Location 1 only. The system will support Auto Detection and
provisioning of all these phones (third party IP Phones and Extended Clients).
Once you connect the phones the details will be displayed on the Device Management page.To use this feature,
• Click Device Management.
• The following information will be displayed for each connected phone:
• Device Name: It displays the name of the phone you connected/application you register.
• MAC Address / Device ID: It displays the MAC or Device ID of the phone.
• IP Address: It displays the IP Address of the phone.
• Last Seen: It displays the date and time, when the system detects the connected phone.
• Assign: If the phone is already configured, it displays the Name, Extension Number and Location
Number. If you wish to edit the details, click on this detail link.
If the device is not configured, it displays ‘Not Assigned’. If you wish to assign an extension to the
phone, click on the Not Assigned link.
A new Assign Extension window opens. You can configure/edit the details:
• Select SIP Extension: Select the SIP Extension Number you want to assign to the phone.
• Name: Enter a name for the SIP Extension, which may be the name of the person who will use the
SIP Extension or the name of a Department. The name you enter here will be displayed as the
Caller ID of the SIP Extension on the remote user's phone, when the SIP Extension user makes
calls. The name may consist of a maximum of 18 alphanumeric characters.
• SIP ID: Enter the SIP ID for the extension. The SIP ID is necessary for registering the SIP
Extension with the Registrar of the system. It is the number with which you can call the SIP
Extension. Any extension user of the SARVAM UCS can call a SIP Extension by dialing the SIP ID

assigned to the SIP extension. SIP ID of each SIP Extension must be a unique number string of a
maximum of 6 digits. Any combination of digits from 0 to 9 and the characters * and # are allowed.
• Authentication ID: Enter the number which you want the system to use for user authentication of
the SIP messages received from the SIP Extension. You cannot keep this field blank. The number
may be a string of maximum 6 alphanumeric characters. All ASCII characters except < > and “
(double quote) are allowed.Default: Blank.
You must configure the Authentication ID, if any of the SIP Message Authentication Options, namely
INVITE or SUBSCRIBE or PUBLISH, is enabled.
• In the Authentication Password field, enter the password to be used by the system to
authenticate the SIP messages received from the SIP Extension. To avoid unauthorized access, we
recommend you to change the password regularly. Make sure it is strong and is kept confidential.
The password must:
• all ASCII characters (except Percentage %, Hash #, Equal to =, Plus +, And &, Backslash \,
Less than <, Greater than >, Apostrophe ’, Double Quote " and Space) and digits 0 to 9 are
allowed.
Default: Blank.
registration of this SIP Extension. However, you can remove the IP Address from the Blacklist IP
Address list. See “Black List IP Address - SIP Extensions” for more details. This activity will be
logged in the “System Activity Log” as well as “Simple Network Management Protocol (SNMP)”.
• Select Location: Select the location 1, 2 or 3 on which you want the extension to register.
• Location Name: Enter the name that you want the system to display for the location you selected.
• Registrar Server Address: Select the appropriate Registrar Server Address to register the device
with the SIP Registrar of SARVAM UCS, according to your installation scenario.
• Click Submit. The window closes and the details are displays on the Device Management page.
• Reboot: Click this button to reboot the phone remotely.
The phone will reboot only if it supports remote reboot.
For Grandstream, Snom and Polycom IP Phones reboot is not supported.
• Click Clear All Assigned Entries, to clear all the assignments.

Black List IP Address - SIP Extensions
Blacklist IP Address enables you to restrict unauthorized access to SARVAM UCS.
To use this feature, you must enable Black List SIP Extension IP Address:Port on multiple Authentication Failure
Attempts check box in the “Security Settings”. The SARVAM UCS blacklists the IP Address from which an
unauthorized attempt is made for registration. When any user attempts to register as a SIP Extension using false
credentials— Authentication ID or Authentication Password and the authentication attempt fails for 10 times
consecutively within 10 minutes, SARVAM UCS blacklists the IP Address and port used for registration.
The blacklisted IP Address/es and ports are stored in the Black List IP Address - SIP Extensions table along with
the date and time. This activity will also be logged in the “System Activity Log” as well as “Simple Network
Management Protocol (SNMP)”.
If you want to allow access of SARVAM UCS to such black listed IP Address, you must remove it from the Black
List IP Address - SIP Extensions table. If this IP Address is a Trusted IP Address, you can configure it in the
Trusted IP Address/es table to avoid further black listing. For details, see “Security Settings”.
Blacklist IP Address table stores upto 500 entries. When the buffer is full, SARVAM UCS follows FIFO method to
store further entries. You cannot edit any entries in this table. However, if required, you may remove a entry from
this table.
To clear a entry from the Black List IP Address - SIP Extensions table,
• Log in as System Engineer.
• Click the Black List IP Address - SIP Extensions link.
The Black List IP Address - SIP Extensions page opens.

• Select the Clear IP Address & Port check box of the entries that you want to remove from the Black List
IP Address - SIP Extensions table.
• Click the Clear Selected button. The selected entries will be removed from the table.
• Click Clear All button to clear all the entries stored in the table.

Configuring DKP Extensions
The number of DKP extensions available to you for configuration depends on the number of DKP ports supported
by SARVAM UCS and the number of DKP ports you have specified on the “System Pre-requisites” page.
If you have enabled 'On-Site Configuration', the system will provide you only those ports that are actually present in
the system for configuration.
Configure DKP port parameters using Jeeves or by dialing commands from a Telephone.
Configuring DKP Parameters using Jeeves

The DKP Ports appear in tabs, with eight DKP Ports in each tab, 001-008, 009-016, 017-024, and so forth.
• For each DKP Port number, configure the following parameters:
• DKP H/w Slot - Port: 'Slot' is the number of the Universal Slot in which the DKP Card is inserted. 'Port'
is the number of the DKP hardware port on which the proprietary DKP EON is connected.
The SARVAM UCS can automatically detect and assign the hardware slot and port numbers
automatically to the DKP software ports.
For example: if you have inserted the DKP8 Card in Slot number 03 and DKP16 Card in Slot number
04 of SARVAM UCS, the system will assign the hardware slot 03 and port numbers 01-08 to the DKP
Software Ports from 001 to 008 respectively. The system will assign Slot 04 and port numbers 09-16 to
the DKP Software Ports 009 to 024. Refer the topic “Software Port and Hardware ID” to know more.
However, if required, you may change the Hardware Slot and Port assigned to the DKP software port.
In this case, enter the desired Hardware Slot and Port number in this field.

If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields.
• Access Code: Assign Station Access Codes to the DKP Port. Station Access Codes are commonly
referred to as Extension Numbers. These may be a maximum of 6 digits, which are dialed to call the
DKP port to which they are assigned.
All DKP ports are assigned the following Station Access Codes as default.
DKP (Software) Port No. Access Codes
001 3001
002 3002
003 3003
: :
: :
128 3128
You may either apply the default Station Access Codes to the DKP ports or assign them according to
your requirement and preferences.
To assign Station Access Codes according to your preference and requirement to a range of DKP
Ports, see “Assigning Access Codes to a Range of Extensions”.
If you decide to customize the Station Access Codes, make sure that the numbers do not clash with any
other Access Code in the 'Dial' phase. Refer the topics “Access Codes” and “Conflict Dialing” to know
more.
• Name: Assign a 'Name' to the DKP port. The name may be of the person who will use the DKP or the
name of a department. This name will be displayed on the LCD of the DKP of the user and on other
extension user's phones, provided these are also a model or EON or are equipped with Caller ID.
You can program a name of maximum 18 alphanumeric characters.
• Station Basic Feature Template: Assign a “Station Basic Feature Template” to the DKP.
A Station Basic Feature Template includes a set of features that completely define the behaviour of the
Extension, such as Time Table, Operator access, Trunk Access, Class of Service, Toll Control, Call
Budget, and Station Message Detail Records (storage of Incoming and Outgoing Call details).
By default, Station Basic Feature Template 01 is assigned to all extensions of the system that includes
DKP ports as well as DKP ports, ISDN Terminals, E&M Lines with Station as Orientation Type, and SIP
Extensions.
Check if the default template fulfills the feature requirements (like “Class of Service (COS)”, “Toll
Control”, “OG Trunk Bundle Group”, etc.) of the DKP.
If the default Template 01 fulfills the feature requirements and if the same features are to be allowed to
all DKPs, retain Template 01.
If different sets of features are to be allowed to different DKPs, then prepare separate Station Basic
Feature Templates and apply them on the ports. To do this,

• Under Configuration, click the link “Station Basic Feature Template” to open the page.
• Customize Template number 11 and click 'Submit' at the bottom of the page.
• Now go back to the DKP Parameters page.
• Enter the number of the Template you customized, Template 11, in the Station Basic Feature
Template field of the DKP Port, for instance DKP-005, on which you want to apply this template. If
you want to apply this template to other ports too, like DKP-006, 007, and 008, assign the Template
11 to all these ports.
• Repeat the same steps to customize and assign a different Template to another DKP port.
Also, refer the topic Station Basic Feature Templates to know more about customizing the
templates and applying on the ports.
• Station Advanced Feature Template: Assign a Station Advanced Feature Template to the DKP. The
Advanced Feature Template consists of a set of less commonly used features like Alarm Notification
Type, Caller ID Presentation for Transferred Calls, DDI Incoming Call Routing, Storage of Internal
Calls, Call Duration Control, Floor Service, Call Taping, etc.
By default Station Advanced Feature Template 01 is assigned to all extensions of the SARVAM UCS,
which includes DKP Ports, SLT Ports, ISDN Terminals, E&M Lines configured as Stations, and SIP
Extensions.
Check if this default template fulfills the feature requirements of the DKP Ports.
If the default Template 01 fulfills the feature requirements, and if the same features are to be allowed to
all DKP ports, retain Template 01.
If different sets of features are to be allowed to different DKP Ports, then prepare separate Station
Advanced Feature Templates and apply them on the ports.
To do this,
• Customize Template number 02.
• Click Submit at the bottom of the page.
• Now go back to the DKP Parameters page.
• Enter the number of the Template you customized, Template 02 in the Station Advanced Feature
Template field of the DKP Port, for instance DKP-001, on which you want to apply this template. If
you want to apply this template to other ports too, like DKP-002, 003, and 004, assign the Template

• Repeat the same steps to customize and assign a different Template to another DKP port.
Also refer the topic “Station Advanced Feature Template” for instructions on customizing these
templates and applying them on the ports.
• Call Capacity: Call Capacity is the number of Call Appearances (also referred to as 'call loops')
assigned to a (DKP) extension. It is the ability of an Extension to handle multiple calls simultaneously. A
Call Appearance allows an extension user to attend to more than one calling party at a time.
A minimum of two Call Appearances must be assigned to a DKP Extension - Operator extension or
Executive extension - so that the Extension user can put one party on hold while talking to another. A
third Call Appearance allows the extension user to put two calls on hold, make/attend a third call and
toggle between three calls.
The higher the call capacity (the more the number of Call Appearances assigned to an extension), the
more the number of calls the Extension user can handle.
The SARVAM UCS supports a maximum of 10 Call Appearances as Call Capacity of the DKP
extensions.
DKP extensions for Executives are usually assigned 2 Call Appearances, while the Operator Extension
is assigned 6 Call Appearances to handle 6 calls simultaneously. The default call capacity of the DKP
ports is 02.
Now, select the number of Call Appearances you wish to assign to the DKP port in the column, 'Call
Capacity'.
• Call Waiting Tone: During an on-going conversation, if there is a second incoming call, the system
plays beeps to indicate the second incoming call. You can set the frequency of the beeps as per your
requirement. You can select from the following options:
• Off
• Beep Once
Default: Beep Once
• Key Map: EON is designed to function as Operator, Executive, Hotel Attendant, and Hotel Guest
extensions, providing default key settings (key maps) for all these functions. All you need to do is
assign a Key Map Template according to the intended user of the DKP. For example, if the DKP is to be
used by the Operator, select ‘Operator's Template’. The DKP will be assigned the key template with the
special features required by Operators, such as more DSS keys for Trunk Access and Call
Appearances, a Call Release Key, etc.
Similarly, if the user of the DKP is a Hotel Attendant, select 'Hotel Attendant's Template'. The key map
with the specific Front Desk User features such as Check-In, Check-Out, Guest In/Out, Change Room
Clean Status, Room Shift, will be automatically assigned to the DKP.
The option ’Personalized’ allows you to customize the key map of the DKP and assign functions to keys
as per your requirement. Click DSS Keys to assign the features and facilities as per your requirement.

To know more about key maps, key templates and how to customize them, see “DSS Keys
Programming”.
• Call Pick-Up Group: Program this parameter if you want to assign the DKP to a particular “Call Pick
Up” group.
Call Pick Up allows the DKP extension user to 'pick up' (answer) calls ringing on any other extension,
by dialing a feature code, without physically going to the ringing extension. For this to work, both the
ringing extension and the extension picking up the call must be in the same 'Call Pick Up Group'. Refer
“Call Pick Up” for instructions on how to create groups. You can create as many as 99 groups,
numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this DKP in this field.
• Station Type: Make sure you select the option Assistant, if it is the Operator Extension. The system
will play beeps during the conference to the participants to indicate the presence or absence of the
Operator. To know more, see “Conference-3 Party”, “Conference-Multiparty” and “Conference Dial-In”.
For the Hotel Application of the SARVAM UCS, extensions are identified as — Administrator, Assistant
or Guest extensions according to the intended user of the DKP. The system will consider the
Administrator and Assistant options as same. When the Station Type is selected, the system will
automatically assign the Guest and Administrator (Hotel Staff) features to the DKP. To know more,
refer the SARVAM UCS Hospitality System Manual.
• DSS Key Settings: This parameter is to be configured if you wish to Personalize the phone Key Map
or have attached a Direct Station Selection Console with the DKP.
To Personalize the phone Key Map, refer to “Personalizing Key Maps” in “DSS Keys Programming”
Matrix offers two models of DSS — DSS64, DSS532.
If you are using EON48/310 you can attach two DSS64 to increase the number of DSS keys for Direct
Station Calling or for one touch feature access.
If you are using EON510 you can attach four DSS532 to increase the number of DSS keys for Direct
Station Calling or for one touch feature access.
Refer the topic “Installing DSS532 with EON510” for installation instructions.
For instructions on configuring the DSS Keys of the Consoles, refer to “Programming DSS Console
Keys” in “DSS Keys Programming”.
The Voice Mail Settings link will be visible only if you have configured the respective access codes.
Advanced Configuration
The above listed parameters fulfill the basic DKP extension port configuration requirements of most users.
However, it is anticipated that some users may need to configure the more advanced features like Personal
Directory, Language Selection, DKP Settings - Backlight Brightness and Contrast, Headset/Handset/Speaker

Volume levels, Select a Ringer Tune, attach Direct Station Selection Consoles, user details etc. For such users,
you may click the 'Advanced' button.
• Configure the following advanced DKP Parameters:
• Mobile Number: Enter the Mobile Number of the extension user you wish to store. The Number can be a
maximum of 16 digits.
• Email ID: Enter the Email ID of the extension user you wish to store. The Email ID can be a maximum of
64 characters.
• SMS/Email Group Type: You can assign the extension user to a Group. Select the desired SMS/Email
Group Type from the list.The system clubs together extension users assigned the same Group. Default:
None. For details, see “SMS/Email Group”.
• Personal Directory: Enter the number of the “Personal Directory” that you want to assign to the DKP. A
Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes (“OG Trunk Bundle Group”). The Personal Directory is
necessary for using the features “Abbreviated Dialing” and “Dial By Name”.
To be able to assign a Personal Directory to a DKP you must first program it. Refer the topic “Abbreviated
Dialing” for instructions on programming the Personal Directory.
• Priority: Select a Priority Level for the DKP from 1, 2, 3... to 9, with '1' being lowest Priority and '9' being
highest Priority. Whenever an extension (DKP) with higher priority calls an extension with lower priority, a
triple ring is placed on the called extension. To know more, read the feature description “Priority”.
By default, the Priority of all DKP ports is set to '5-Normal'.
• CO CLIP Pattern: This parameter allows you to select the type of Calling Line Presentation on the DKP for
incoming calls from trunks. You can select any of the below options:

• Name Only: only the name of the caller will be displayed.
• Number Only: only the number of the caller will be displayed.
• Number + Name: both the name and the number of the caller will be displayed.
By default, Number + Name is selected as the CO CLIP Pattern for all DKPs.
• Language: SARVAM UCS provides language support for English, French, German, Spanish, Portuguese,
and Italian. When you select any of these languages, all the command strings and prompts will appear in
the selected language. By default English is selected.
DKP users can change the language by accessing and navigating through the phone menu.
• Ringer Mode: You can select a Ringer mode for each DKP from the four options:
• Ring immediately (it rings immediately as a fresh calls lands on the DKP).
• Ring if idle (rings only if the DKP is idle).
• Silent.
By default the Ringer Mode is set to 'Ring Immediately'. Change the Ringer Mode on the DKP as per the
requirement and preferences of the DKP users.
• Ring Delay Timer: The Ring Delay Timer is the time in seconds the SARVAM UCS will wait to ring on
receiving a call. This Timer needs to be set only if you have selected 'Ring after a delay' as the Ringer
Mode for the DKP in the previous parameter.
By default, the Ringer Delay Timer is set to 10 seconds. You may change the Ring Delay Timer according
to the preferences and requirements of the DKP user.
• Acknowledge Timer: This Timer is to be programmed to enable the Ringer Auto Acknowledge mode.
This mode determines when to stop the ring on the DKP. There are two options for ringer auto
acknowledge:

To stop the ring on the DKP after a delay, the Acknowledge Timer must be programmed. The range of the
timer is 01 to 99 seconds.
• Play Ring ON: With this parameter, you can assign the Ring Destination for the DKP; you can choose
whether the Ring should be played on the Speakerphone or Headset of the DKP. Default: Speakerphone is
selected.
When you select the Headset as the destination, ensure that you have enabled Headset Connectivity flag
and have connected a Headset to the DKP.

The speech path of both, the Headset and the Handset is common. If the Headset is not connected and
you have selected the Headset as the ring destination, the ring will be played on the speaker of the
Handset.
• Ring Tune: You can select from different ringer tunes for each DKP according to your preferences and
requirement. By default, Ringer Tune is set to 1.
• Ring Volume: You can select from different ringer volumes for each DKP according to your preferences
and requirement. By default, Ringer Volume is set to 5.
• Handset Transmit Volume Level: This parameter is used for increasing or decreasing the volume of
outgoing speech (Transmit Gain) on the Handset of the DKP. Select the desired Handset Tx Volume Level
from 0 to 9. By default, Handset Tx Volume Level is 4.
• Handset Receive Volume Level: This parameter is used for increasing or decreasing the volume of
incoming speech (Receive Gain) on the Handset of the DKP. Select the desired Handset Rx Volume Level
from 0 to 9. By default, Handset Rx Volume Level is 4.
• Headset Transmit Volume Level: This parameter is used for increasing or decreasing the volume of
outgoing speech (Transmit Gain) on the Headset port of the DKP. Select the desired Headset Tx Volume
Level from 0 to 9. By default, Headset Tx Volume Level is 4.
• Headset Receive Volume Level: This parameter is used for increasing or decreasing the volume of
incoming speech (Receive Gain) on the Headset port of the DKP. Select the desired Headset Rx Volume
Level from 0 to 9. By default, Headset Rx Volume Level is 4.
• Hands-free Transmit Volume Level: With this parameter you may change the Volume level of Transmit
Gain of the Speaker phone MIC volume from 0 to 9, as per your preference. This parameter is to be used
for increasing or decreasing the volume levels of outgoing speech on the Speaker of the DKP. By default,
Hands-free Tx volume level is 4.
• Hands-free Receive Volume Level: With this parameter you may change the Volume level of Receive
Gain of the Speaker phone MIC volume from 0 to 9, as per your preference. This parameter is to be used
for increasing or decreasing the volume levels of incoming speech on the Speaker of the DKP. By default,
Hands-free Rx volume level is 4.
• Right Handset Transmit Volume Level (EON74145): This parameter is applicable only when EON74 is
connected to the DKP Port. This parameter is used for increasing or decreasing the volume of outgoing
speech (Transmit Gain) on the Right Handset of EON74. Select the desired Tx Volume Level from 0 to 9.
By default, the Right Handset Tx Volume Level is 4.
• Right Handset Receive Volume Level (EON74146): This parameter is applicable only when EON74 is
connected to the DKP Port. This parameter is used for increasing or decreasing the volume of incoming
speech (Receive Gain) on the Right Handset of EON74. Select the desired Headset Rx Volume Level from
0 to 9. By default, the Right Handset Rx Volume Level is 4.
• Handsfree High gain Mode (For EON510)147: This parameter is applicable only when EON510 is
connected to the DKP Port. This parameter is used for increasing the volume of incoming speech or Ring
145. Check for availability. EON74 is not supported by ETERNITY GENX.

146. Check for availability. EON74 is not supported by ETERNITY GENX.
147. This parameter is not applicable in this version.

Tone in Handsfree mode (Speaker). By default, the Handsfree High gain Mode (For EON510) check box is
disabled. Select the check box to enable.
Handsfree High gain Mode (For EON510) is supported in EON510 firmware version V2R3 onwards.
• Key Click Volume Level: You may change the Key Click Volume (Key DTMF Side tone) of the DKP. Key
Click Volume is the tone you hear as you press the dial pad keys of EON. Select the desired volume level
from 0 to 9. By default, the volume level is set to 5.
• DTMF Generation Flag: This flag is used to enable or disable DTMF dialing on the DKP. By default, the
flag is enabled. To disable the flag, click the check box.
• DTMF Transmit Level: You can select the desired Transmit Level from 0 to 9 for DTMF generation from
the DKP. By default, the DTMF Transmit Level is set to 2.
• Headset Connected?: Enable this parameter by selecting the check box if you want to use a Headset
with the DKP.
Make sure that you have also connected a Headset to the DKP.
• Auto Answer: Enable this parameter by selecting the check box if you want to set the “Auto Answer”
feature on the DKP.
When this feature is set, the DKP goes OFF-Hook automatically after a preset period of time, without the
user having to pick up the handset or press the speaker or headset key.
• Auto Answer Timer (sec): This parameter is to be programmed if you have enabled the “Auto Answer”
feature on the DKP.
When the Auto Answer feature is enabled, the Auto Answer Timer must be programmed. This timer
defines the time in seconds that the DKP should wait before going OFF-Hook. The range of this timer is 1
to 9 seconds. By default, the Auto Answer Timer is set to 1 second.
• LCD Backlight Level: You can change the LCD Backlight Brightness of EON. The intensity of the
backlight brightness increases from 0 to 4, where '0' will cause the backlight to be turned OFF. '1' signifies
minimum intensity, '4' signifies maximum intensity. Select any of the levels from 1 to 4 from the list.
• LCD Backlight OFF Timer: The backlight of the LCD display of EON can be kept switched ON
continuously, or can be set to switch OFF automatically after a predefined period of time, known as the
Backlight OFF Timer. The range of the Backlight OFF Timer is 000-999 seconds. By default it is set to
switch OFF after 010 seconds.
• LCD Contrast Level: The EON offers 4-level contrast control for its LCD display. Level 1 signifies
minimum and level 4 is the maximum. The contrast increases in steps of 1 to 4. By default the contrast is
set to level 3. You may adjust it to the level comfortable to you. Select a level from 1-4.
• Line Echo Cancellation (LEC): Enable this parameter to cancel the echo generated on the other end.
You must also configure the Line Echo Cancellation Start Timer. When DKP users go Off-Hook to make/
receive calls, the Line Echo Cancellation Timer starts. After the expiry of this timer the SARVAM UCS will
start Line Echo Cancellation. To configure the Echo Cancellation Start Timer, see “System Timers and
Counts”.

• You are recommended to enable Line Echo Cancellation only if you hear echo after you have set the
AC Impedance value in the CO Hardware Template according to the Accurate Test conducted by you.
For details see, “AC Impedance Test” and “CO Hardware Template”.
• The Call Progress Tones heard by the DKP user may be affected, if you enable LEC.
• LEC is supported only in EON48D, Version V3R15 or later.
• CPLD Version: This parameter displays the CPLD Version of the Digital Key Phone Card.
• The CPLD Version will be displayed only if supported by your DKP Card.
• Current Firmware: This parameter displays the current firmware of the Digital Key Phone connected to
the DKP Port.
• Upgrade Firmware: Enable this parameter by selecting the check box, if you want to upgrade the
firmware of the Digital Key Phone connected to the SARVAM UCS.
• You can upgrade firmware of any two DKP phones at a time. Before upgrading, make sure both these
phones are in idle state. You cannot use these phones while their upgradation is in process.
• You can view and Upgrade the current firmware of the DKP phone only from Jeeves.
Configuring DKP Extensions using a Telephone

To assign Hardware Slot and Port, dial:

• 1102-DKP-Slot-Port offset on the card
Where,
DKP is the Software Port number of the DKP port from 001 to 128.
Slot is Slot number in which the DKP Card is inserted from 01 to 16
Port Offset is the number of the DKP Port on the card from 01 to 99.
To clear the hardware Slot and Port assigned to the DKP software port, dial:
• 1102-DKP-00-00
To assign an Access Code to a DKP Port, dial:

• 3102-1-DKP-Access Code-#*
Where,
Access Code is a number string of any combination of 1, 2, 3, 4, 5 or 6 digits. Terminate the command
with #* if the number string has fewer than 6 digits.
To clear the Access codes assigned to the DKP, dial:

• 3102-1-DKP-#* to clear the Access Code of a single DKP port.
• 3102-2-DKP-DKP-#* to clear the Access Codes of a range of DKP ports.
• 3102-*-#* to clear the Access Codes of all DKP Ports.
To assign default Station Access Codes to DKP, dial:

• 3152-1-DKP to default Access Code of a single DKP Port.
• 3152-2-DKP-DKP to default Access Codes of a range of DKP ports.

• 3152-* to default Access Codes of all DKP ports.
To assign a Name to a DKP Port, dial:

• 5403-1-DKP-Name-#* to assign a Name to a single DKP port.
• 5403-2-DKP-DKP-Name-#* to assign the same Name to a range of DKP ports.
• 5403-*-Name-#* to assign the same Name to all DKP ports.
Where,
Name is a string of a maximum of 18 alphanumeric characters. Terminate the commands with #* if the
number string has fewer than 18 characters.
To clear the Name of the DKP Port, dial:

• 5403-1-DKP-#* to clear the Name of a single DKP port.
• 5403-2-DKP-DKP-#* to clear the Names of a range DKP ports.
• 5403-*-#* to clear the Names of all DKP ports.
To assign a Station Basic Feature Template to a DKP Port, dial:

• 5504-1-DKP-Template Number to assign a template to a single DKP port.
• 5504-2-DKP-DKP-Template Number to assign the same template to a range of DKP ports.
• 5504-*-Template Number to assign the same template to all DKP ports.
Where,
Template Number is the number of the Station Basic Feature Template, from 01 to 50. Default: 01
To assign a Station Advanced Feature Template to a DKP Port, dial:

• 5604-1-DKP-Template Number to assign a template to a single DKP port.
• 5604-2-DKP-DKP-Template Number to assign the same template to a range of DKP ports.
• 5604-*-Template Number to assign the same template to all DKP ports.
Where,
Template Number is the number of the Station Advanced Feature Template, from 01 to 50. Default: 01
To define the Call Capacity of a DKP Port, dial:

• 1201-1-DKP-Call Capacity to define the call capacity for a single DKP port.
• 1201-2-DKP-DKP- Call Capacity to define the same call capacity for a range of DKP ports.
• 1201-*- Call Capacity to define the same call capacity for all DKP ports.
Where,
Call Capacity is from 02 to 10. Default: 02.
To assign a Key Map for a DKP Port, dial:

• 1221-1-DKP-DKP Key Template Number to assign a Key Map to a single DKP port.
• 1221-2-DKP-DKP-DKP Key Template Number to assign the same Key Map to a range of DKP ports.
• 1221-*-DKP Key Template Number to assign the same Key Map to all DKP ports.
Where,
DKP Key Template Number is
1 for Operator's Template
2 Executive1 Template
3 Hotel Attendant's Template
4 Guest's template.

Default: 1
To assign the DKP Port to a Call Pick-Up Group, dial:

• 3902-1-DKP-Call Pick-Up Group to assign a single DKP port to a Call Pick-Up group.
• 3902-2-DKP-DKP-Call Pickup Group to assign a range of DKP ports to the same Call Pick-Up group.
• 3902-*-Call Pickup Group to assign all DKP ports to the same Call Pick-Up group.
Where,
Call Pickup Group is from 00 to 99. Default: 99.
To remove a DKP from a Call Pick Up group, dial:

• 3902-1-DKP-00 to remove a single DKP port from a Call Pick-Up group.
• 3902-2-DKP-DKP-00 to remove a range of DKP ports from a Call Pick-Up group.
• 3902-*-00 to remove all DKP ports from a Call Pick-Up group.
To select a Station Type for a DKP Port, refer the SARVAM UCS Hospitality System Manual.
• For Advanced Configuration of the DKP Ports, use the following commands:
To assign a Personal Directory to a DKP Port, dial:

• 1906-1-DKP-Personal Directory to assign directory to a single DKP port.
• 1906-2-DKP-DKP- Personal Directory to assign the same directory to a range of DKP ports.
• 1906-*-Personal Directory to assign the same directory to all DKP ports.
Where,
Personal Directory number is from 01 to 50.
To clear the Personal Directory assigned to a DKP Port, dial:

• 1906-1-DKP-00 to clear the directory of a single DKP port.
• 1906-2-DKP-DKP-00 to clear the directory of a range of DKP ports.
• 1906-*-00 to clear the directory from all DKP ports.
To define the Priority for a DKP Port, dial:

• 3912-1-DKP-Priority to define Priority for a single DKP port.
• 3912-2-DKP-DKP-Priority to define the same Priority for a range of DKP ports.
• 3912-*-Priority to define the same Priority for all DKP ports.
Where,
Priority is from 1 to 9. Default: 5-Normal
To select CO CLIP Pattern for a DKP Port, dial:

• 1243-1-DKP-CO CLIP Pattern to select the CLIP Pattern for a single DKP port.
• 1243-2-DKP-DKP-CO CLIP Pattern to select the same CLIP Pattern for a range of DKP ports.
• 1243-*-CO CLIP Pattern to select the same CLIP Pattern for all DKP ports.
Where,
CO CLIP Pattern is
1 for Name only
2 for Number only
3 for Number+Name
Default: Number+Name

To select Language for a DKP port, dial:
• 1224-1-DKP-Language to select language for a single DKP port.
• 1224-2-DKP-DKP-Language to select the same language for a range of DKP ports.
• 1224-*-Language to select the same language for all DKP ports.
Where,
Language is
1 for English
2 for Français
3 for Deutsch
4 for Español
5 for Português
6 for Italian
Default: English
To select Ringer Mode for a DKP Port, dial:

• 1204-1-DKP-Ringer Mode to select Ringer mode for a single DKP port.
• 1204-2-DKP-DKP-Ringer Mode to select Ringer mode for a range of DKP ports.
• 1204-*-Ringer Mode to select Ringer mode for all DKP ports.
Where,
Ringer Mode is
1 for Ring Immediate
2 for Ring if Idle
3 for Ring after Delay
4 for Ring OFF
Default: Ring Immediate
To select Ring Delay Timer for a DKP Port, dial:

• 1205-1-DKP-Ring Delay Timer to select Delay Timer for a single DKP port.
• 1205-2-DKP-DKP- Ringer Delay Timer to select Delay Timer for a range of DKP ports.
• 1205-*- Ringer Delay Timer to select Delay Timer for all DKP ports.
Where,
Ring Delay Timer is 01 to 99 seconds. Default: 10 seconds.
To set the Acknowledgement Mode, dial:

• 1206-1-DKP-Ringer Auto Acknowledge Mode to select mode for a single DKP port.
• 1206-2-DKP-DKP-Ringer Auto Acknowledge Mode to select the same mode for a range of DKP
ports.
• 1206-*-Ringer Auto Acknowledge Mode to select the same mode for all DKP ports.
Where,
Ringer Auto Acknowledge Mode is
0 for OFF
1 for ON
To set the Ringer Auto Acknowledge Timer, dial:

• 1207-1-DKP-Ringer Auto Acknowledge Timer to set Timer for a single DKP port.
• 1207-2-DKP-DKP-Ringer Auto Acknowledge Timer to set the same Time for a range of DKP ports.
• 1207-*-Ringer Auto Acknowledge Timer to set the same Time for all DKP ports.
Where,

Ringer Auto Acknowledge Timer is from 01 to 99 seconds. Default: 00.
To select Destination for 'Play Ring ON' for a DKP Port, dial:
• 1220-1-DKP-Ring Destination to select Play Ring ON destination for a single DKP port.
• 1220-2-DKP-DKP-Ring Destination to select the same Play Ring ON destination for a range of DKP
ports.
• 1220-*-Ring Destination to select the same Play Ring ON destination for all DKP ports.
Where,
Ring Destination is
1 for Play Ring on Speaker Phone
2 for Play Ring on Headset
To select Ring Tune for a DKP Port, dial:

• 1202-1-DKP-Ring Tune to select a Tune for a single DKP port.
• 1202-2-DKP-DKP-Ring Tune to select the same Tune for a range of DKP ports.
• 1202-*-Ring Tune to select the same Tune for all DKP ports.
Where,
Ringer Tune is from 0 to 9. Default: 1.
To set Ringer Volume for a DKP Port, dial:

• 1203-1-DKP-Ringer Volume to set Ringer Volume for a single DKP port.
• 1203-2-DKP-DKP-Ringer Volume to set the same volume level for a range of DKP ports.
• 1203-*-Ringer Volume to set the same volume level for all DKP ports.
Where,
Ringer Volume is from 0 to 9. Default: 5.
Setting Ringer Volume to '0' will cause the DKP volume to be turned OFF.
To set Handset Transmit (Tx) Volume Level for a DKP, dial:

• 1208-1-DKP-Handset MIC Volume Level to set transmit volume for a single DKP port.
• 1208-2-DKP-DKP-Handset MIC Volume Level to set the same transmit volume level for a range of
DKP ports.
• 1208-*-Handset MIC Volume Level to set the same transmit volume level for all DKP ports.
Where,
Handset MIC Volume Level is from 0 to 9. Default: 5.
Setting volume level to '0' will cause the Handset MIC volume to be turned OFF.
To set Handset Receive (Rx) Volume Level for a DKP, dial:

• 1209-1-DKP-Handset Speaker Volume Level to set receive volume for a single DKP port.
• 1209-2-DKP-DKP-Handset Speaker Volume Level to set the same receive volume level for a range
of DKP ports.
• 1209-*-Handset Speaker Volume Level to set the same receive volume level for all DKP ports.
Where,
Handset Speaker Volume Level from 0 to 9. Default: 5.
Setting volume level to '0' will cause the Handset Speaker volume to be turned OFF.

To set Right Handset Transmit (Tx) Volume Level for DKP Port, dial:
• 1225-1-DKP-Right Handset MIC Volume Level to set transmit volume for a single DKP port.
• 1225-2-DKP-DKP-Right Handset MIC Volume Level to set the same transmit volume level for a
range of DKP ports.
• 1225-*-Right Handset MIC Volume Level to set the same transmit volume level for all DKP ports.
Where,
Handset Volume Level is from 0 to 9. Default: 5.
To set Right Handset Receive (Rx) Volume Level for a DKP, dial:
• 1226-1-DKP-Handset Speaker Volume Level to set receive volume for a single DKP port.
• 1226-2-DKP-DKP-Handset Speaker Volume Level to set the same receive volume level for a range
of DKP ports.
• 1226-*-Handset Speaker Volume Level to set the same receive volume level for all DKP ports.
Where,
Handset Speaker Volume Level from 0 to 9. Default: 5.
To set Headset Transmit (Tx) Transmit Volume Level for a DKP, dial:
• 1222-1-DKP-Headset MIC Volume Level to set receive volume for a single DKP port.
• 1222-2-DKP-DKP- Headset MIC Volume Level to set the same receive volume level for a range of
DKP ports.
• 1222-*- Headset MIC Volume Level to set the same receive volume level for all DKP ports.
Where,
Headset MIC Volume Level is from 0 to 9. Default: 5.
To set Headset Receive (Rx) Volume Level for a DKP, dial:

• 1223-1-DKP-Headset Speaker Volume Level to set receive volume for a single DKP port.
• 1223-2-DKP-DKP- Headset Speaker Volume Level to set the same receive volume level for a range
of DKP ports.
• 1223-*- Headset Speaker Volume Level to set the same receive volume level for all DKP ports.
Where,
Headset Speaker Volume Level is from 0 to 9. Default: 5.
To set Hands-free Transmit (Tx) Volume Level for a DKP, dial:

• 1210-1-DKP-Speaker Phone MIC Volume Level to set volume level for a single DKP port.
• 1210-2-DKP-DKP-Speaker Phone MIC Volume Level to set the same volume level for a range of
DKP ports.
• 1210-*-Speaker Phone MIC Volume Level to set the same volume level for all DKP ports.
Where,
Speaker Phone MIC Volume Level from 0 to 9. Default: 5.
Setting volume level to '0' will cause the Speaker Phone MIC volume to be turned OFF.

To set Hands-free Receive (Rx) Volume Level for a DKP, dial:
• 1211-1-DKP-Speaker Phone Speaker Volume Level to set volume level for a single DKP port.
• 1211-2-DKP-DKP-Speaker Phone Speaker Volume Level to set the same volume level for a range of
DKP ports.
• 1211-*-Speaker Phone Speaker Volume Level to set the same volume level for all DKP ports.
Where,
Speaker Phone Speaker Volume is from 0 to 9. Default: 5.
To set Key Click Volume Level for a DKP, dial:

• 1212-1-DKP-Key Click Volume to set volume level for a single DKP port.
• 1212-2-DKP-DKP-Key Click Volume to set the same volume level for a range of DKP ports.
• 1212-*-Key Click Volume to set the same volume level for all DKP ports.
Where,
Key Click Volume Level is from 0 to 9. Default: 5.
To enable/disable DTMF Generation Flag for a DKP, dial:

• 1241-1-DKP-DTMF Generation to enable/disable the flag for a single DKP port.
• 1241-2-DKP-DKP-DTMF Generation to enable/disable the flag for a range of DKP ports.
• 1241-*-DTMF Generation to enable/disable the flag for all DKP ports.
Where,
DTMF Generation is
0 for Disable
1 for Enable
Default: Enable.
To set DTMF Transmit Level for a DKP, dial:

• 1218-1-DKP-DTMF Transmit Level to set Transmit level for a single DKP port.
• 1218-2-DKP-DKP-DTMF Transmit Level to set Transmit level for a range of DKP ports.
• 1218-*-DTMF Transmit Level to set Transmit level for all DKP ports.
Where,
DTMF Transmit Level is 0 to 9. Default: 2.
To enable/disable Headset Connectivity for a DKP, dial:

• 1213-1-DKP-Headset Connectivity Flag to enable/disable the flag for a single DKP port.
• 1213-2-DKP-DKP-Headset Connectivity Flag to enable/disable the flag for a range of DKP ports.
• 1213-*-Headset Connectivity Flag to enable/disable the flag for all DKP ports.
Where,
Headset Connectivity Flag is
0 for Disable
1 for Enable
Default: Disabled
To enable/disable Auto Answer for a DKP, dial:

• 1214-1-DKP-Auto Call Answer Mode to enable/disable Auto Answer for a single DKP port.
• 1214-2-DKP-DKP-Auto Call Answer Mode to enable/disable Auto Answer for a range of DKP ports.
• 1214-*-Auto Call Answer Mode to enable/disable Auto Answer for all DKP ports.
Where,

Auto Call Answer Mode is
1 for OFF (manual)
0 for ON (auto)
Default: OFF
To set Auto Answer Timer (sec) for a DKP, dial:

• 1215-1-DKP-Auto Call Answer Timer to set Timer for a single DKP port.
• 1215-2-DKP-DKP-Auto Call Answer Timer to set the same Timer duration for a range of DKP ports.
• 1215-*-Auto Call Answer Timer to set the same Timer duration for all DKP ports.
Where,
Auto Call Answer Timer is from 1 to 9 seconds. Default: 3 seconds.
To set LCD Back Light Level of a DKP, dial:

• 1216-1-DKP-LCD Backlight Level to set brightness of a single DKP.
• 1216-2-DKP-DKP-LCD Backlight Level to set the same brightness level for a range of DKPs.
• 1216-*-LCD Backlight Level to set the same brightness level for all DKPs.
Where,
Backlight Level is Brightness level from 1 to 4. Default: 3
To change LCD Backlight OFF Timer of a DKP, dial:

• 1219-1-DKP-LCD Backlight OFF Timer to change the timer for a single DKP.
• 1219-2-DKP-DKP-LCD Backlight OFF Timer to set the same timer duration for a range of DKPs.
• 1219-*-LCD Backlight OFF Timer to set the same timer duration for all DKPs.
Where,
Backlight OFF Timer is from 000-999 seconds. Default: 010 seconds.
To change LCD Contrast Level of a DKP, dial:

• 1217-1-DKP-LCD Contrast Level to set the Contrast level for a single DKP
• 1217-2-DKP-DKP-LCD Contrast Level to set the same Contrast level for a range of DKPs.
• 1217-*-LCD Contrast Level to set the same Contrast level for all DKPs.
Where,
LCD Contrast Level is from 1 to 4. Default: 3.
To assign Hardware Slot-Port to DSS connected to a DKP, dial:

• 1103-DKP-DSS-Slot-Port offset on the card
Where,
DSS is 1 for the first DSS Console and 2 for the second DSS Console attached to the DKP
Slot is the number of the Universal Slot in which the DKP Card is located, to which the DSS Console is
connected, from 01 to 16.
Port is the number of the hardware port on the card on which the DSS Console is connected.
To clear the hardware Slot-Port assigned to the DSS software port, dial:
• 1103-DKP-DSS-00-00
• Exit SE mode.
DKP extension users can change several phone settings according their preferences and requirement.
These are referred to as DKP Personal Settings and include:

• Ringer Volume
• Ringer Tune
• Ringer Mode
• Ringer Acknowledge Mode
• Speech Volume (Transmit/Receive)
• User Status (Present/Absent)
• Keypad Security (Lock/Open)
• Call Answer Type - Manual/Auto
• Headset/Handset Connectivity option
To be able change the DKP personal settings, the DKP users must access and navigate the phone menu.
Refer “Digital Key Phone-Operation”.

DSS Keys Programming
The DSS (Direct Station Selection) Keys when personalized, provides you quick access to Stations, Trunks,
Features/Functions of the SARVAM UCS. A few DSS Keys are provided on the phone itself. The number of DSS
Keys provided on the phone depends on the type of phone. For details, see “Digital Key Phone-Operation” and
“Extended IP Phone/VARTA UC Client - Operation”.
Matrix offers the DSS Consoles when attached to the phone, works as extensions to the phone. You can customize
the DSS Console Keys as per your requirement by assigning the desired features/functions. For details, see “Direct
Station Selection Console”.
To configure the DSS Keys on the phone, refer “Customizing Key Templates” and “Personalizing Key Maps”.
To configure the DSS Console Keys, refer “Programming DSS Console Keys”.
Key Templates
EON, the proprietary digital key phone (DKP) of Matrix and SPARSH IP Phones of Matrix, can be the extension of
the Operators and Executives in an enterprise, and in hotels, it can be the extension of the Front Desk Attendants
and Guests.
Each of these groups of users may require a different set of features on their phones. For example, when EON/
SPARSH is an Operator's extension, for efficient call management, more DSS keys may be required for Trunk
Access, Call Appearances, Call Release, Direct Station Calling, than for accessing features.
When EON/SPARSH is an Executive's extension, more DSS keys may be required for single-touch access to
features, and fewer keys for Trunk Access and Direct Station Calling.
Similarly, when EON/SPARSH is a Hotel Attendant's extension, keys are required for specific hotel functions such
as Check-In/Check-Out, Changing Room Clean Status, Room Shift, etc. But, a different set of keys with special
functions are required when EON/SPARSH is provided as a guest extension, because guests are allowed only a
limited number of features of the SARVAM UCS, such as calling the Front Desk/Floor Service, setting Do Not
Disturb, Wake-up Calls, Call Forward, and checking Voice Mail.
Given the varying requirements of these groups of extension users, SARVAM UCS provides programmable
Templates of Key Maps for the Operator, Executive1, Executive2, Executive3, Hotel Attendant, and Hotel Guest.
The default Key Maps in these Templates can be customized to match the requirement of the intended user group.
The default Key Maps of Executive1, 2 and 3 are the same.
The customized Template is assigned to the DKPs/SPARSH IP Phones. For example, you may customize the Key
Template for the Operator and assign it to the Operator DKPs. Likewise, you may customize the Key Template for
Executive and assign it to the Executive extensions.
The default Key Maps vary according to the models of the DKP/SPARSH IP Phone in use, as illustrated below.
In case of Extended IP Phones, you can customize the Templates for the Operator, Executive1, Executive2,
Executive3, Hotel Attendant, and Hotel Guest as well as add new templates. You can add upto 64 templates. The
new template you add, can be edited or remove as per your requirement. However you cannot remove the default
Templates — Operator, Executive1, Executive2, Executive3, Hotel Attendant, and Hotel Guest. These can only be
customized.

You can also Personalize the DKP/SPARSH IP Phone key maps, see “Personalizing Key Maps” for instructions.
The keys in the Key Templates are numbered only for the purpose of locating the keys when programming.
Key numbers do not appear on the key labels on the phone body.
Default Key Maps
SPARSH VP248 Key Template (default)

The default Key Template is same for the Operator, Executive, Hotel Attendant and Guest.
SPARSH VP110
All the default templates are same for SPARSH VP110.
You can customize the following keys only — History, Directory, DND, Menu, Up, Down, Right, Left, OK, Mute and
Transfer.

EON48 Key Templates (default)
Operator/Executive Hotel Attendant Guest

Voice Mail ACB Set Cancel Mute 3PConf Transfer Voice Mail ACB Set Cancel Mute 3PConf Transfer Voice Mail ACB Set Cancel Mute 3PConf Transfer
17 18 19 23 24 25 17 18 19 23 24 25 17 18 19 23 24 25
20 21 22 26 27 28 20 21 22 26 27 28 20 21 22 26 27 28
CallFwd DND Names Redial Release Hold CallFwd DND Names Redial Release Hold CallFwd DND Names Redial Release Hold
01 09 2 01 09 2 01 09 2
1 abc 3 def 1 abc 3 def 1 abc 3 def
02 10 02 10 02 10
4 ghi 5 jkl 6 mno 4 ghi 5 jkl 6 mno 4 ghi 5 jkl 6 mno
03 11 03 11 03 11
04 12 7 pqrs 8 tuv 9 wxyz 04 12 7 pqrs 8 tuv 9 wxyz 04 12 7 pqrs 8 tuv 9 wxyz

Retrv
05 CA 4 13 05 Msg. 13 05 DKP2 13
0 # 0 # 0 #
06 CA 3 14 * 06 CA 3 14 * 06 DKP1 14 *
07 CA 2 15 07 CA 2 15 07 CA 2 15
08 CA 1 16 29 08 CA 1 16 29 08 CA 1 16 29
There is no LED on the following key numbers:

• 19 (Cancel)
• 22 (Names)
• 24 (Conference)
• 25 (Transfer)
• 26 (Redial)
• 27 (Release)
• 28 (Hold)
Operator Key Map Executive Key Map
01 01
02 02
03 03
04 04
05 05
06 06
07 07
08 08
09 09
10 10
11 11
12 12

Guest Key Map
01
02 01
03 02
04 03
05 04
05
06
06
07
07
08
08
09
09
10
10
11
11
12
12

• 15 (Hold)
• 18 (Conference)
• 20 (Contacts)
• 21 (Transfer)
Operator / Executive
CO4
CO3
CO2
CO1
1 2 abc 3 def
4 ghi 5 jkl 6 mno
7 pqrs 8 tuv 9 wxyz
#
* 0 CA4
CA3
CA2
CA1

Hotel Attendant Guest
R-V Alarm Operator

Floor Ser Floor Ser
R-Forward MACRO 4
R-DND Retrv. Msg.
R-Budget Voice Help
Call Block CO3
Msg Wait CO2
1 2 abc 3 def CO1
1 2 abc 3 def CO1
BT VMS Alarm-VG
4 ghi 5 jkl 6 mno 4 ghi 5 jkl 6 mno
Print RS SLT3
7 pqrs 8 tuv 9 wxyz Print AS 7 pqrs 8 tuv 9 wxyz SLT2
Clean SLT1
#
* 0 # Retrv. Msg.
CA3
* 0 DKP2
DKP1
CA2 Ca2
CA1 CA1

• Hold
• Redial
• Transfer
• Conference
EONSOFT Key Template (default)
Operator/Executive Hotel Attendant Guest

Options Help Options Help Options Help
SLT1 SLT 2 SLT3 R-V Alarm R- Forward R-DND Operator Floor Ser Alarm-VG
01 02 03 01 02 03 01 02 03
DKP1 DKP2 DKP3 DKP4 Floor Service DKP1 DKP2 Voice Mail Voice Mail DND Voice Help Mute
06 07 08 09 06 07 08 09 06 07 08 09
CO1 CO2 CO3 CO3 Retrv Msg Print RS Print AS Clean DKP1 DKP2 SLT1 SLT2
11 12 13 14 11 12 13 14 11 12 13 14
CO1 CO1 CO1 CO1 CO1 CO2 Call Back AR-Set CO1 CO2 Call Log BGM
16 17 18 19 16 17 18 19 16 17 18 19
CA1 CA2 CA3 CA4 CA1 CA2 CA3 CA4 CA1 CA2 Recall Release
21 22 23 24 21 22 23 24 21 22 23 24
26 26 26

By using Key Templates you can prepare and assign common key maps to all or as many DKPs and SPARSH IP
Phones as you want, at one go.
SARVAM UCS also offers the flexibility to personalize the Key Maps of each DKP/SPARSH IP Phone, instead of
using the Key Templates. For example, if you have assigned a common Executive Key Template to 12 DKPs, but
you want to reassign some of the keys on two of these DKPs, SARVAM UCS allows you to selectively personalize
the key maps of these two DKPs. To personalize the key maps for DKPs, see “Personalizing Key Maps” for
instructions.
Customizing Key Templates

You can customize the Key Templates for:
• EON48 using Jeeves as well as a Telephone.
• EON310 using Jeeves only.
• EON510 using Jeeves only.
• SPARSH VP248 using Jeeves as well as a Telephone.
• SPARSH VP330 using Jeeves only.
• SPARSH VP310 using Jeeves only.
• SPASH VP110 using Jeeves only
• EONSOFT using Jeeves as well as a Telephone.
• Before you begin programming the keys,

• List the features/facilities that you want to change in each of the existing (default) Key Templates of
Operator, Executive, Hotel Attendant, and Guest.
• Create customized Key Templates on sheets of paper.
• For each template, decide the keys that will be reassigned the features you listed.
• You may use the key templates printed above to decide the position of keys.
• For each template that you customize, list down the DKPs which will be assigned the template,
along with their Software port numbers and their corresponding Hardware Slot and Port Offset.
Similarly list SPARSH IP Phones along with their Software port numbers (SIP Extension numbers).
• Similarly, list the DKPs and the SPARSH IP Phones which are to be assigned personalized Key
Maps, along with their Software port numbers and their corresponding Hardware Slot and Port
Offset.

Illustrated below is an example of a customized Operator's Template for the EON48 model:
Voice Mail ACB Set Cancel Mute 3PConf Transfer

17 18 19 23 24 25
20 21 22 26 27 28
CallFwd DND Names Redial Release Hold
01 09
1 2 abc 3 def
02 10
4 ghi 5 jkl 6 mno
03 11
04 12 7 pqrs 8 tuv 9 wxyz

05 CA 4 13
0 #
06 CA 3 14 *
07 CA 2 15
08 CA 1 16 29
This customized Template is to be applied on 2 Operator DKPs:

• DKP-001, connected on H/w Slot-Port - 17-01
• DKP-002, connected on H/w Slot-Port - 17-02
You can customize the key template and assign it to DKPs using Jeeves or a Telephone.
EON48 and SPARSH VP248 have 12 Touch-sense feature keys. While you can reassign the features on
these keys, you cannot re-label the keys. Avoid reassigning features on touch-sense keys.
EON310 has 9 feature keys, displaying feature icons as labels. While you can reassign the features on
these keys, you cannot re-label the keys. Avoid reassigning features on these keys.
EON510 has 8 feature keys, displaying feature icons as labels. While you can reassign the features on
these keys (except Headset and Mute key), you cannot re-label the keys. Avoid reassigning features on
these keys.
Customizing Digital Key Phone Templates using Jeeves

To customize key templates for EONSoft, EON42, EON310, EON510 follow the instructions given below.
• Under Configuration, click Key Template to open the page.
• There are separate links for each model of EON/IP Phone. Click the desired phone model.The respective
phone model is displayed.
• Click Key Template to select the default key template — Operator, Executive1, Executive2, Executive3,
Hotel Attendant and Guests — you want to customize.

Let us attempt to program the sample Operator template we customized earlier for EON48.
• Click EON48 to open the page. Select Operator as the Key Template.
• The Operator Key template for EON48 appears.
• Refer the key template we customized on paper. As per the customized template the keys that need to be
reassigned features are as follows:
Existing function on the key To be replaced by
DKP4 Hotline
DKP3 SA Command for “Remote DND”
DKP2 DND-Override
SLT4 Mobile Trunk 1 (MOBILE1)
SLT3 SIP Trunk 1 (SIP1)
SLT2 BRI Trunk 1 (BRI 1)
SLT1 T1E1 Channel Number 2 of T1E1 Trunk 1
• To assign Hotline, click the DKP4 key. A new window opens.

• As Hotline is a feature, in the Select Function Type list, click Feature.
All features that can be assigned to keys will appear in the Select Offset list.

• In the Select Offset list, click Hotline.
• Click on OK in the dialog box. The box will close.

• The Hotline feature will appear on the key label.
• To assign Remote DND, click DKP3 key.

• As Remote DND is an SA Command, select the option SA Command in the Select Function Type list.
• In the Select Offset list, click Set DND for remote station.

• Click OK. The window closes. The Remote DND feature appears in abbreviated form as R-DND on the
key label.
• Repeat these steps to reassign other keys, selecting the appropriate Function Type and the Offset for each
feature/function.
For example, DND-Override is a feature, so select 'Feature' as the function type.
To assign direct access to Mobile Trunk 1, select “Mobile” as function type and '01' as Offset.
To assign direct access to SIP Trunk 1, select “SIP” as function type and '01' as Offset.
To assign direct access to BRI Trunk 1, select “BRI” as function type and ‘01’ as Offset. When you select
BRI as function type, you will be asked to ‘Select Channel’. Since you do not want to assign direct access
to any particular BRI channel of this trunk, retain the option ‘All Channels” for ‘Select Channel’.

To assign direct access to Channel Number 2 of T1E1 Trunk 1, select 'T1E1' as function type, '01' as
Offset.
When you select T1E1, as function type, you will be given the ‘Select Channel’ option. Since you want to
assign direct access to Channel 2 of T1E1 Trunk 1, select the option ‘Channel 2’ in the ‘Select Channel’
list.
Always click OK when you select a Function Type and Offset.
• When you have completed assigning functions to keys, click Submit at the bottom of the page to save
your settings.
• Follow the same steps to program the other Key Templates.
• Assign the key templates you created to the respective DKPs.
To take the above example further, the Operator key template customized for EON48 is to be assigned to
the DKPs 001 and 002. To do this,
• Click the DKP Parameters link to open the page.
• Scroll with the horizontal bar to reach the parameter Key Map of this DKP 001.

• Select Operator as the Key Map.
• Similarly, for DKP-002, select Operator as the Key Map.
• Click Submit.
• Also refer the topic “Configuring DKP Parameters using Jeeves” for instructions on assigning the
Key Map in the DKP Parameters.
Customizing Extended IP Phone Templates using Jeeves

You can:
• customize the existing key templates — Operator, Executive1, Executive2, Executive3, Hotel Attendant
and Guests.
• add new key templates.
Customizing Existing Key Templates

To customize existing key templates for SPARSH VP248, SPARSH VP330, SPARSH VP310 and SPARSH VP110
follow the instructions given below.
• There are separate links for each model of IP Phone. Click the desired phone model.The respective phone
model is displayed.
• Click Key Template to select the desired key template — Operator, Executive1, Executive2, Executive3,
Hotel Attendant and Guests — you want to customize.
Let us attempt to configure the sample Operator template for VP248.

Click VP248 to open the page. Select Operator as the Key Template.
For example, to customize the key template the keys that need to be reassigned features are as
follows:
Existing function on the key To be replaced by
DKP3 Hotline
SLT3 SIP Trunk 1 (SIP1)
• To assign Hotline, click the DKP3 key. A new window opens.
• As Hotline is a feature, in the Select Function Type list, click Feature.
All features that can be assigned to keys will appear in the Select Offset list.
• In the Select Offset list, click Hotline.

• Click on OK in the dialog box. The box will close.
• The Hotline feature will appear on the key label.
• To assign direct access to SIP Trunk 1, click SLT3 key.

Select SIP Trunk as function type and 01 as Offset.
SIP 1-1 appears as the key label.
When you have completed assigning functions to keys, click submit to save the settings.
Similarly, you can customize the other templates — Executive, Hotel Attendant, Guest.
To customize the templates for other phone models, follow the instructions as given above.
Add, Edit or Remove Key Templates

If you do not wish to customize any existing template, you can add new templates. By default, the Operator
Template will be displayed as the phone key map. You can customize the key map as per your requirement or add
a new template.

You can add upto 64 new templates for each phone model.
To add a template,
• There are separate links for each model of IP Phone. Click the desired phone model.The respective phone
model is displayed.
• Click Add Template. The Add Template window opens.
• Assign a name to the template. This template appears as one of the Key Template options.
• To customize the key template to match the requirement, click John as the Key Template option.
Follow the instructions given under “Customizing Key Templates”.

• When you have completed assigning functions to keys, follow the same steps if you wish to add other Key
Templates.
• Assign the key templates you created to the respective SIP Extension location using Jeeves. See “DSS
Key Settings” in “Configuring Matrix SPARSH VP330”, “DSS Key Settings” in “Configuring Matrix SPARSH
VP248” “DSS Key Settings” in “Configuring Matrix SPARSH VP510” and “DSS Key Settings” in
“Configuring Matrix SPARSH VP310” for instructions.
• To Edit the template name, click Edit Template.
• To delete the template, click Remove Template.
If you are customizing the key maps of SPARSH VP110, you can customize the following keys only —
History, Directory, DND, Menu, Up, Down, Right, Left, OK, Mute, Transfer.
Customizing Key Templates using a Telephone
EON310 / SPARSH VP110 Key Templates can be customized using Jeeves only.
When customizing key templates using a Telephone, two attributes of a key must be programmed, namely:
a. Location - the physical location of the key on the phone body.

b. Function - the function that the key should perform, that is, as direct access to a feature or an extension, a
trunk type, an SA command, etc.
Follow the numbering of keys on the default Key Maps illustrated earlier in this topic.
To assign a feature to a key in a Template, dial:

• 1261-1-Key Template-Terminal Type-Key Number-Function Type-Function Number-Channel
To assign the same feature to the same key number in a range of templates, dial:
• 1261-2-Key Template-Key Template-Terminal Type-Key Number-Function Type-Function
Number-Channel
To assign the same feature to the same key number in all the templates, dial:
• 1261-*-Terminal Type-Key Number-Function Type-Function Number-Channel
Where,
Key Template is
1 for Operator
2 for Executive1
3 for Hotel Attendant
4 for Guest
5 for Executive2
6 for Executive3
Terminal Type is
1 for EON45/EONSOFT
2 for EON42
3 for EON48
4 for SPARSH VP248
6 for EON310
7 for SPARSH VP330
Key Number is the number of the key which is to be reassigned the feature.
Key Numbers on EONSOFT are from 01 to 25
Key Numbers on EON42 are from 01 to 25
Key Numbers on EON48 are from 01 to 29.
Key Numbers on SPARSH VP248 are from 01 to 29.
Key Numbers on SPARSH VP330 are from 01 to 15.
For numbering of the keys, refer the default Key Maps illustrated at the beginning of this topic.
Function Type is the function to be performed by the key, from 00 to 34.
Function Number defines the exact function under each Function Type selected for the key.
Channel is the number of the channel in the BRI or PRI Line to be accessed.
Channel number for BRI Line is

01 for Channel 1
02 for Channel 2
00 for both Channels

Channel number for PRI Line is
01 to 30 for Channels 1 to 30 respectively.
00 for all Channels
Channel number for all function types other than BRI and PRI is 00
Refer the tables below for the complete list of function types and function numbers, and channel
numbers.
Table: Function Types and Function Numbers
Function Function
Meaning Meaning
Type Number
00 None - -
01 SLT (Direct Station 001 to 512 Number of the SLT (software) port which is to be
Call) called when the key is pressed.
02 DKP (Direct Station 001 to 128 Number of the DKP (software) which is to be called
Call) when the key is pressed.
03 CO (Direct Trunk 001 to 128 Number of the CO (software) port which is to be

Access) called when the key is pressed.
04 BRI (Direct Trunk 001 to 032 Number of the BRI (software) port which is to be called
Access and Direct when the key is pressed.
BRI Channel
Access) Number of the Channel in the BRI Line, which is to be
accessed when the key is pressed.
Channel numbers are:
01 for BRI Channel 1

02 for BRI Channel 2
05 T1E1 (Direct Trunk 001 to 008 Number of the T1E1 (software) port which is to be
Access and Direct called when the key is pressed.
PRI Channel
Access) Number of the Channel in the PRI Line which is to be
accessed when the key is pressed.
Channel numbers are:
01 for T1E1 Channel 1

:
:
00 for all Channels
06 E&M (Direct Trunk/ 001 to 128 Number of the E&M (software) port which is to be
Station Access) called when the key is pressed.
11 Dept. Group (Direct 01 to 16 Number of the Department Group which is to be

Station Call) called when the key is pressed.

Function Function
Meaning Meaning
Type Number
12 Quick Dial 001 to 999 The Global Directory Index number on which the
(Abbreviated External Number is stored. Call will be made to that
Dialing) External Number when this key is pressed
13 LOOP (Loop Count 01 to 10 The number of Call Appearance/Call Loop which will
or Call Capacity) be accessed when the key is pressed.
14 MACRO 01 to 25 The number string of the Key Board Macro which will
be dialed when the key is pressed. Refer “Macros”
15 TAC (Direct Trunk 1 to 6 The Trunk Access Code that will be dialed when the
Access) key is pressed.
18 FEATURE (Direct 01 to 87 The feature that will be invoked when the key is
Feature Access) pressed. For a complete list of features, refer the
“Table: “Function Type 18: Features””.
21 Voice Mail 1 The Voice Mail System Port to be dialed when the
key is pressed.
22 SA Command 001 to 173 The SA Command to be invoked when the key is

(Direct Dialing of pressed. For a complete list of SA Commands, refer
SA Command) the “Table: “Function Type 22: SA Commands””.
24 Special Keys 01 to 15 The Special Digits to be dialed when the key is

pressed. For a complete list of Special Keys, refer
the “Table: “Function Type 24: Special Keys””.
25 MOBILE (Direct 01 to 32 The number of the Mobile Port to be accessed when

Trunk Access) the key is pressed.
26 SIP (Direct Trunk 01 to 32 The number of the SIP Trunk to be accessed when
Access) the key is pressed
28 ISDN Terminal 01 to 64 The number of the ISDN Terminal (software) port

(Direct Station Call) which is to be dialed when the key is pressed.
29 Magneto (Direct 001 to 128 The number of the Magneto (software) port to be
Station Call) dialed when the key is pressed.
31 ROOM (Direct 001 to 512 The number of the Hotel Room to be dialed when the
Station Call) key is pressed.
34 SIP Extension 001 to 999 The number of the SIP Extension to be dialed when
(Direct Access) the key is pressed.
36 Virtual Ext. 01-64 The number of the Virtual Extension to be dialed

when the key is pressed.
37 Radio Port 01-16 The number of the Radio Port to be dialed when the
key is pressed.
Table: “Function Type 18: Features”
Feature Feature Number DSS LED Activity?
Enter SE Programming Mode 1
Enter SA Programming Mode 2

Call Pickup - Group 3
Call Pickup - Selective 4
Auto Call Back - Set 5 Yes
Auto Call Back - Cancel 6
Redial 7
Auto Redial - Set 8 Yes
Auto Redial - Cancel 9
Personal Directory Programming 10
Abbreviated Dialing 11
Operator 12
Call Forward 13 Yes
Dynamic Lock 14
Hotline 15 Yes
Alarm 16 Yes
Do Not Disturb 17 Yes
Interrupt Request 18
Barge-In 19
Raid 20
Trunk Reservation 21
Call Toggle 22
Conference 23 Yes
- 24
Dial-In Conference 25
Call Park 26
Call Park - Retrieve 27
Room Monitor 28
Last Caller Recall 29
Voice Help 30
Walk-In Class of Service 31
Change User Password 32
Paging 33
DISA Login 34
Trunk to Trunk Call Release 35
Cancel all Feature Settings 36
Selective Port Access 37

Flashing on Trunk 38
User Absent/Present 39 Yes
Account Code by Number 40
Account Code by Name 41
Meet Me Paging 43
Hot Desk 44
Do Not Disturb Override 45
Presence 46 Yes
Live Call Screening 47 Yes
Conversation Recording 48
Forced Release 49
Transfer 50
Live Call Supervision 51
Forced Answer 52
Change Room Clean Status 53
Guest Number Prefix 54
Minibar Details 55
Mute 56 Yes
Emergency Conference 57
Self Ring Test 58
Call Chaining 59 Yes
SA Command Prefix 60
COSEC Door Open 61
Floor Service 62
Keypad Lock 63 Yes
CLI Restriction 64 Yes
Call Cost Display 65
Reminder 66 Yes
Alarm - Voice Guided 67
Reminder - Voice Guided 68
Blind Transfer to Voice Mail 69
Message Wait Set/Cancel 70 Yes
Retrieve New Message 71 Yes
PMS - User Defined Fields 72
Open a Door 75

Magneto Ring Enable 77
Invoke RCOC 78
Scheduled Call Forward 79 Yes
E&M Manual Priority Intrusion 80
E&M Forced Release Order 81
Department Call Forward 82
General Mailbox 83
Intercom 84
Terminate Conference 86
Leave Temp. / Rejoin Conf. 87
Call Forward - When Not Registered 88 Yes
DSS Call Pick-up-Station 89 Yes
DSS Call Pick-up-Trunk 90 Yes
Trunk Access Code 1(TAC1) -
Voice Mail - Yes
Table: “Function Type 22: SA Commands”
SA Command Name Function Number DSS LED Activity?
Check-In 001
Check-Out 002
Guest Name 003
Guest Group 004
Guest-In/Out 005
Guest Title 006
Change Check-In Profile of Room 007
Change Occupancy Status of Room / Extension 008
Change Clean status of Room/Extension 009
Room Shift 010
Reprint Check Out Report 011
Print Room Status Report 012

Print Alarm Status Report 013
Delete Checked Out calls 014
Set DND-Remote 015
Set Dynamic Lock settings - Remote 016
Set Alarm -Remote 017
Assign Call Budget to a station 018
Assign / un-assign Mailbox to a Station - Remote 019
Set Call Forward - Remote 020
Set Call Forward for all Stations - Remote 021
Assign Station User Greeting Message 022
Display & Acknowledge System Activity 023 Yes
Display & Acknowledge System Fault 024 Yes
Station Budget Display - Remote 025
Change User Password of a Station - Remote 026
Lock/unlock DKP's Keypad - Remote 027
User Absent / Present - Remote 028
Change SA password 029
Change SA mode timer 030
Display Registered GSM Network ID 031
Set Day/Night Mode 032 Yes
Terminate Security dialing 035
Clear System Activity Log 036
Start/Abort SAL in Offline mode 037
Start/Abort SAL in Online mode 038
- 039
Cancel Dial in Conference 040
Start/Abort SFL in Offline mode 041
Start/Abort SFL in Online mode 042
Display Port Parameters 043
Start/Abort Online OG Report 044
OG Print Filter: Print calls originated from station(s) 045
OG Print Filter: To Print calls terminated from CO 046
OG Print Filter: To Print calls terminated from BRI 047
OG Print Filter: To Print calls terminated from T1E1 048
OG Print Filter: To Print calls terminated from E&M 049

OG Print Filter: To Print calls terminated from Mobile 050
OG Print Filter: To Print calls terminated from SIP 051
OG Print Filter: To Print calls Department Bill Group wise 052
OG Print Filter: To print calls made on date(s) 053
OG Print Filter: Print calls made between time 054
OG Print Filter: To Print calls made to numbers matching with 055

the numbers programmed in the Number List
OG Print Filter: To Print calls of Duration more than this time 056
OG Print Filter: To Print calls of Units more than the units 057
programmed
OG Print Filter: To Print calls made to account code 058
Assign default OG Print filters 059
Start/Abort offline report 060
Enable/ Disable OG Schedule Reports 061
Program Time for Daily OG Scheduled Reports 062
Program Day and Time for OG Weekly Scheduled Reports 063
Program Date and Time for OG Monthly Scheduled Reports 064
Delete calls made by station(s) 065
Delete calls made on/from date 066
Clear SMDR OG buffer 067
Start/Abort Internal calls Report 068
Set filter to print Internal calls Report Station wise 069
Set filter to print internal calls with duration greater than that 070
given here
Start/Abort Offline Internal Call Report 071
Enable/ Disable Internal Scheduled Reports 072
Program Time for Internal Daily Scheduled Reports 073
Program Day and Time for Internal Weekly Scheduled Reports 074
Program Date and Time for Internal Monthly Scheduled Reports 075
Clear SMDR Internal Buffer 076
Start/Abort Online - IC Report 077
Set filter to print all Normal calls 078
Set filter to print all Built-In Auto Attendant calls 079
Set filter to print all Unanswered calls 080
Set filter to print all Built-In Auto Attendant Unanswered calls 081
Set filter to print all DISA calls 082

Set filter to print all calls with speech duration More than timer 083
Set filter to print all calls unanswered for duration More than 084
timer
Set filter to print all calls kept on hold for duration more than 085
timer
Set filter to print all IC calls received by the station 086
Set filter to print all IC calls recd. On the CO 087
Set filter to print all IC calls recd. On the BRI 088
Set filter to print all IC calls recd. On the T1E1 089
Set filter to print all IC calls recd. On the E&M 090
Set filter to print all IC calls from Mobile 091
Set filter to print calls received from SIP 092
Set filter to print all IC calls recd. on/from date 093
Set filter to print all IC calls recd. at/from-to Time 094
Set filter to print all IC calls recd. From nos. matching the 095
Number List
Default IC Print filters 096
Abort/Start IC Offline Report 097
Enable/ Disable IC Scheduled Report 098
Program Time for IC Daily Scheduled Reports 099
Program Day and Time for IC Weekly Scheduled Reports 100
Program Date and Time for IC Monthly Scheduled Reports 101
Clear SMDR-IC buffer 102
Start/Abort Printing of Online T1E1 Performance Report 103
Start/Abort Offline T1E1 Performance Report 104
Signal Strength of Mobile Port 105
Enable/Disable Call Cost Display for a Station 106
Start/Abort Hotel/Motel Activity log in Offline mode 107
Start/Abort Hotel/Motel Activity log in Online mode 108
Display and Acknowledge Hotel/Motel Activity 109 Yes
Change Guest VIP Status of Station 110
Change Phone Ringing Pattern of Room 111
Print Reminder Status Report 112
Reminder - Remote 113
Voice Guided Alarm - Remote 114
Voice Guided Reminder - Remote 115

Redirect Messages of a Station 116
To record Holiday message 117
Enable/Disable Scheduled Alarm Report 118
Program Time for Scheduled Alarm Report 119
Enable/Disable Scheduled Reminder Report 120
Program Time for Scheduled Reminder Report 121
Request Database Synchronization to PMS 122
Enable/Disable Scheduled Room Status Report 123
Program Time for Schedule Room Status Report 124
Enable/Disable Scheduled Change of Room Clean Status 125
Program Time for Schedule Change of Room Clean Status 126
- 127
Enable/Disable Internal Call Block For Guest Phones 128 Yes
Software Version Revision Display of CPU Card 129
User Definable Fields 130
OG Print Filter: To print calls originated from CO 131
OG Print Filter: To print calls originated from BRI 132
OG Print Filter: To print calls originated from T1E1 133
OG Print Filter: To print calls originated from E&M 134
OG Print Filter: To print calls originated from Mobile 135
OG Print Filter: To print calls originated from SIP 136
To change CPU from Active state to Standby (only for ULSB MK 137
III)
Set Budget Type for CO Port 138
Program Budget Amount for CO 139
Program Free Minutes on CO 140
Set Budget Type for T1E1 Port 141
Program Budget Amount for T1E1 142
Program Free Minutes on T1E1 143
Set Budget Type for BRI Port 144
Program Budget Amount for BRI 145
Program Free Minutes on BRI 146
Set Budget Type for SIP 147
Program Budget Amount for SIP 148
Program Free Minutes on SIP 149
Set Budget Type for MOBILE 150

Program Budget Amount for MOBILE 151
Program Free Minutes on MOBILE 152
Call Budget Reset Mode for CO 153
Scheduled Date to Rest Call Budget Statistics on CO 154
Reset consumed Budget Amount /minutes on CO manually 155
Call Budget Reset Mode for T1E1 156
Scheduled Date to Reset Call Budget Statistics on T1E1 157
Reset consumed Budget Amount /minutes on T1E1 manually 158
Call Budget Reset Mode for BRI 159
Scheduled Date to Rest Call Budget Statistics on BRI 160
Reset consumed Budget Amount /minutes on BRI manually 161
Call Budget Reset Mode for SIP 162
Scheduled Date to Rest Call Budget Statistics on SIP 163
Reset consumed Budget Amount /minutes on SIP manually 164
Call Budget Reset Mode for MOBILE 165
Scheduled Date to Rest Call Budget Statistics on MOBILE 166
Reset consumed Budget Amount /minutes on MOBILE manually 167
Swap Ports for CO8/SLT8 - MAG8 Card 168
Reset mobile ‘avg call dur’ & ‘ans/seizure’ 169
Set/Cancel Scheduled Call Forward 170
- 171
- 172
- 173
- 174
To Broadcast Message 175
Table: “Function Type 24: Special Keys”
Special Key Name Function Number DSS LED Activity?
Hold 001 No
Exclusive Hold 1 to 8 002 to 009 Yes
Pause 010 No
One Touch Transfer 011 No
Ringer Mute 012 Yes
Release 015 No
Acknowledge 016 No

Special Key Name Function Number DSS LED Activity?
Enter 017 No
Speed 018 No
Emergency Alarm Log 019 Yes
Speaker 020 Yes
Wakeup Call Log 021 Yes
Headset 022 Yes
Cancel 023 No
Answer 024 No
Yes (for Missed

Call Log 025
Call)
Local Menu 026 No
Examples:
Let us attempt to program the same sample Operator Template we customized earlier for EON48 using
a telephone. For this, we need to know the location of the key, that is, the key number.
Existing function on the key To be replaced by Location of the key on the Key Map
DKP4 Hotline Key No. 01
DKP3 SA Command for “Remote DND” Key No. 02
DKP2 DND-Override Key No. 03
SLT4 Mobile Trunk 1 (MOBILE1) Key No. 09
SLT3 SIP Trunk 1 (SIP1) Key No. 10
SLT2 BRI Trunk 1 (BRI 1) Key No. 11
SLT1 T1E1 Channel Number 2 of T1E1 Key No. 12

Trunk 1
To assign Hotline to the key currently assigned to DKP4 key (key 01) in the Operator Template, dial:
• 1261-1-1-3-01-18-15-00
Where,
1 is Operator Template
3 is for EON48 Terminal
01 is the Key Number
18 is Function Type for Features
15 is Function Number for Hotline feature.
00 is Channel.
To assign Mobile Trunk 1 in place of SLT4 on DSS key 09 on the Operator Template, dial:
• 1261-1-1-3-09-25-01-00
Where,

25 is Function Type for Mobile Trunks
01 is Function Number, that is, the port offset of the Mobile Port.
00 is Channel
To assign BRI Trunk 1 in place of SLT2 on DSS Key 11 on the Operator Template, dial:
• 1261-1-1-3-11-04-01-00
Where,
04 is Function Type for BRI Trunks
01 is Function Number, that is, number of the BRI port.
00 is Channel number.
To assign Channel 2 of T1E1 Trunk 1 in place of SLT1 on DSS Key 12 on Operator Template, dial:
• 1261-1-1-3-12-05-01-02
Where,
05 is Function Type for T1E1 Trunks.
01 is Function Number, that is, number of the T1E1 port.
02 is Channel number.
To assign a DKP Key template to a DKP, dial:

• 1221-1-DKP-DKP Key Template to assign the template to a single DKP.
• 1221-2-DKP-DKP-DKP Key Template to assign the same template to a range of DKPs
• 1221-*-DKP Key Template to assign the same template to all DKPs
Where,
DKP is number of the Software Port of the DKP to which the Template is to be assigned, from 001 to
128.
DKP Key Template is
0 for Personalized key map
1 for Operator Template
2 for Executive's Template
3 for Hotel Attendant's Template
4 for Guest Template.
For example, to assign the customized Operator Template to DKP 001 and DKP002, you may dial:
• 1221-1-001-1 to assign the template to DKP001
• 1221-1-002-1 to assign the template to DKP002
OR
• 1221-2-001-002-1 to assign the same template to both DKPs at one go.
• Exit SE Mode.
Personalizing Key Maps

You can personalize the Key Maps of individual DKPs (EON) and SPARSH IP Phones, instead of using the Key
Templates.
When you personalize the Key Map of a DKP, make sure 'Personalized' option is selected as the Key Map in the
DKP Parameters of the DKP.

Similarly, when you personalize the Key Map of an SPARSH IP phone, make sure 'Personalized' option as the Key
Template in the SIP Extension Settings - Location of the phone.
The Personalized option is not applicable for SPARSH VP110.
Personalizing Key Map using Jeeves

• Under Configuration, click DKP Parameters.
• Go to the DKP you want to assign a personalized key map, for example, DKP001.
• Select Personalized as the Key Map option for the DKP.
• Click Submit to save Key Map selection.

• Click the Key Settings link under DSS Key Settings.
• The current Key Map of the DKP opens.

• To assign features to keys, follow the same steps as you did for customizing the key templates.
• Click the key you want to assign the function. For example, you want Barge-In on the key assigned to
DKP4, click this key.
• The options for the Functions to be Performed by the key will open in a new window.
• Click the option Feature in the Function Type list.
• Click the option Barge-In in the Select Offset list.

• Click OK. The window will close. The new label will appear on the key.
• Repeat the same steps to program another key on this key map.
• When you have completed personalizing the key map, close the window.
• Follow the same steps to personalize the key map of another DKP.
For instructions on personalizing the Key Map of the Extended IP Phone, see “DSS Key Settings” in “Configuring
Matrix SPARSH VP330”, “DSS Key Settings” in “Configuring Matrix SPARSH VP248”, “DSS Key Settings” in
“Configuring Matrix SPARSH VP310” and “DSS Key Settings” in “Configuring Matrix SPARSH VP510”. The
Personalized option is not applicable for SPARSH VP110.
Personalizing Key Maps using a Telephone

• Enter SE mode.
To assign a Personalized Key map to a DKP, dial:

• 1221-1-DKP-0 to assign personalized key map to a single DKP.
• 1221-2-DKP-DKP-0 to assign the personalized key map to a range of DKPs.
• 1221-*-0 to assign personalized key map to all DKPs.
Where,
DKP is number of the Software Port of the DKP to which the Template is to be assigned, from 001 to
128.
0 is for Personalized key map.
For example, to assign the personalized Key map to DKP 003, dial: 1221-1-003-0

To assign a function to a DKP key in a personalized key map, dial:
• 1252-1-DKP-Key Number-Function Type-Function Number-Channel to assign a function to a DKP
Key on a single DKP.
• 1252-2-DKP-DKP-Key Number-Function Type-Function Number-Channel to assign the same
function on the same key number on a range of DKPs.
• 1252-*-Key Number-Function Type-Function Number-Channel to assign the same function on the
same key number on all DKPs
Where,
DKP is the number of the Software Port of the DKP from 001 to 128.
Key Number is the number of the key which is to be assigned the function/feature. The Key numbers
vary according to the EON Terminal Type being used.
Key Numbers on EONSOFT are from 01 to 25.
Key Numbers on EON42SR are from 01 to 25.
For numbering of the keys, refer the default Key Maps illustrated at the beginning of this topic.
Function Type is the function to be performed by the key, from 00 to 34.
Function Number defines the exact function under each Function Type selected for the key. Refer the
tables above for the complete list of function types and function numbers.
Channel is the number of the channel in the BRI or T1E1 PRI Line to be accessed.
Channel number for BRI Line is

01 for Channel 1
02 for Channel 2
Channel number for T1E1 PRI Line is

01 to 30 for Channels 1 to 30 respectively.
00 for all Channels
Channel number for all function types other than BRI and PRI is 00
For example, you want Barge-In on key number 04 of DKP-003, dial:

• 1252-1-003-04-18-19-00
Where,
003 is DKP-003
04 is the Key Number on DKP-003
18 is the Function Type for Features
19 is the Function Number for Barge-In
00 is Channel number
• Exit SE Mode.
After you have finished assigning key templates and key maps, test the functioning of the keys.

Programming DSS Console Keys
Direct Station Selection (DSS) Consoles are devices that function as extensional buttons for EON/SPARSH VP510
Phones, providing more buttons for single-touch Direct Station Calling and feature access.
You can attach two DSS64 Consoles to a single EON48/310 and four DSS532 Consoles to a single EON510/
VP510 to further increase the number of keys. Refer “Direct Station Selection Console” to know more.
For installation instructions, refer the topics Installing EON, Installing DSS Consoles, under “Installing ETERNITY
LENX”, “Installing ETERNITY MENX” and “Installing ETERNITY GENX”.
For instructions:
• to install the DSS64/DSS532 with EON, see “Installing DSS Consoles”.
• The system does not support any DSS Console for Turret.
• The system supports two soft embedded DSS Consoles for EONSOFT. These will not be displayed in
DSS Status.
Programming DSS Console Keys using Jeeves

You can configure the DSS Console either offline, that is before you have connected the DSS Console or after you
have connected the Console.
To configure DSS Console keys:

• refer to “Configuring DSS Console Keys connected to DKP Phones (EON)”, if the console is connected to
a DKP.
• refer to “Configuring DSS Console Keys connected to SPARSH VP510”, if the console is connected to
SPARSH VP510.
Configuring DSS Console Keys connected to DKP Phones (EON)

Configuring DSS Console Keys Offline,
• Log into Jeeves as System Engineer
• Go to the DKP Port number to which you wish to attach the DSS Console.
• Using the horizontal scroll bar on the page, scroll to DSS Key Settings.

The Key Settings page opens.
• In Device Type select the type of DKP — EON48, EON310, EON510 — you wish to connect to this port.
• Click ADD DSS.

• If you have selected Device Type as EON48 or EON310, the ADD DSS window opens.
In H/w Slot-Port, enter the Hardware slot and port on which you wish to connect the DSS Console.
The Hardware slot port assigned now cannot be changed from Jeeves later.
Click Submit. The ADD DSS window closes.
The default key map of DSS64 appears on your screen.

• For EON510 as the Device Type, click on ADD DSS to add a DSS532 to the default key map of the
phone.
• To add a DSS, click ADD DSS.
• By default None is assigned to all the DSS keys, you can now personalise the DSS Key map as per your
requirement.
• Click the key on which you want to assign a feature/function.

• Select the desired Function Type to be assigned to the key and the desired Offset for the Function Type.
• Click OK.
• The function you selected will appear as the key label.

• Similarly, add another DSS and configure the keys.
• If you wish to clear all the key assignments, click the respective DSS Console check box and then click
Clear Key Assignment.
• If you wish to interchange the key assignments of the DSS, click the check boxes of the respective DSS
Consoles whose keys you wish to swap and then click Interchange DSS Keys.
The option Interchange DSS Keys is applicable only for DSS532.
• If you wish to remove a DSS Console, click Remove DSS.
• To view the status of all the DSS Consoles, click “DSS Status”.
Configuring DSS Console Keys after you have connected the phone as well as the Console,
If you have connected the phone and the DSS Consoles, the system automatically detects them.
In case of DSS64 the connected DSS Console appears under Unassigned DSS64 in DSS Status. You must first
assign the DSS Console to a DKP Port and then you will be able to configure the DSS Keys. For instructions, see
“DSS Status”.
If DSS532 is connected the default key map of the DSS Console will automatically appear on the screen and you
will be able to configure the DSS Keys.

• Go to the DKP number to which DSS Console has been connected.
• Using the horizontal scroll bar on the page, scroll to DSS Key Settings.
• Click Key Settings. The Key Settings page opens.
The page displays the following:
Phone Details
• Device Type displays the model of EON connected — EON48, EON310 or EON510.
• Assigned to displays the Extension number assigned to the phone.
• Port No displays the software port assigned to the phone.
• H/w Slot-Port displays the Hardware slot and port on which the phone is connected.
DSS Console Details

• Type of DSS Console connected along with the number and Status, that is, DSS64/DSS532 - 1 Connected
or Not Connected.
• H/w Slot-Port displays the Hardware slot and port on which the Console is connected.
Hardware Slot-Port is applicable for DSS64 only.
• By default None is assigned to all the DSS keys, you can now personalize the DSS Key map as per your
requirement.

• Click OK.
• Similarly, configure other keys.
The option Interchange DSS Keys is applicable only for DSS532.
• To add another DSS Console, click ADD DSS.
Configuring DSS Console Keys connected to SPARSH VP510

You can connect DSS532 to SPARSH VP510 for quick access to Stations, Trunks, Features/Functions of the
ETERNITY or at the touch of a single key, making call operations easy.
You can configure the DSS Console keys offline, that is before connecting the console or after you have connected
the Console. If you have connected the DSS Console the system will automatically detect the same.
• Log into Jeeves as System Engineer
• Under VoIP Configuration, click SIP Extension Settings.
• Select the desired SIP Extension Number, click the desired Location 1 or 2 or 3.
• On the Location Page make sure you have selected SPARSH VP510 as the Device Type. Scroll to DSS
Key Settings.

• Click Key Settings. The Key Settings page opens.
Phone Details
• Device Type displays SPARSH VP510.
• Assigned to displays the Extension number assigned to the phone.
• Port No displays the software port number as well as the Location, for example: SIP Extension -1 -
Location - 1.
• Click ADD DSS.
If the DSS Consoles are connected the default Key maps will be displayed automatically. Click ADD DSS if
you wish to add another DSS Console.
• The default key map of DSS532 appears on your screen.
• By default None is assigned to all the DSS keys, you can now personalize the DSS Key map as per your
requirement.

• Click OK.
• Similarly, configure other keys.
The following features cannot be assigned to DSS Keys:

• Headset
• Speaker
• Ringer Acknowledge Key
• Local Menu
• If you wish to remove a DSS Console, click Remove DSS.
• Click ADD DSS again to add another Console.

DSS Status
To know the details of the DSS Consoles at a glance,
• Under Maintenance, click DSS Status.
The following details are displayed:
• Total assigned DSS in the system
• Total connected DSS to the system
• Click DSS Status to expand.
The following information is displayed:

• Serial Number
• Assigned to displays the extension number to which the DSS Console is connected.
• Device Type displays the model of EON — EON48, EON310, EON510, SPARSH VP510 — to which
the DSS Console is connected.
• DSS532 displays the status of the console, that is Assigned or Connected.
• DSS64 displays the status of the console, that is Assigned or Connected.
• DSS Key Settings provides the link to Key Settings. Click this link if you wish to configure the DSS
Keys.

• Click Unassigned DSS64 to expand.
If have connected any DSS64, you must assign it to a particular DKP Port and only then will you be able to
configure the DSS Console Keys. You can view all unassigned DSS Consoles here.
For each unassigned console the following is displayed:

• Serial Number
• H/w Slot - Port is the Hardware Slot Port assigned to the DSS Console.
• Assign DSS displays a link Assign. To assign the DSS Console to a DKP Port click the link.
• The Assign DSS window opens.
• In Select DKP Extension, select the DKP Port number to which to wish to assign the console.
• Click Submit.
• After you have assigned the DSS Console, under Configuration click DKP Configuration.
• Click DKP Parameters. Go to the DKP Port to which you assigned the DSS Console.
• Scroll to DSS Key Settings and click Key Settings.
• The DSS Console you assigned to this DKP Port appears on the screen. You can now personalize the
DSS Key map as per your requirement.

Configuring ISDN Terminals
Depending on the number of BRI ports available to you and the type of Point-to-Multipoint Configuration (Short or
Extended Passive Bus), a maximum of 64 ISDN Terminals can be connected to the SARVAM UCS.
The number of ISDN Terminal extensions available to you for configuration also depends on the number of ISDN
Terminals you have specified on the “System Pre-requisites” page.
If you have enabled On-Site Configuration, the system will provide you only those ISDN terminal ports that are
actually present in the system for configuration.
Configure ISDN Terminal Parameters using Jeeves or by dialing commands from a Telephone.
Configuring ISDN Terminals using Jeeves

• Under Configuration, click ISDN Configuration.
• Click ISDN Terminal Parameters to open the page.
• Configure the following parameters for each ISDN Terminal on this page:
• ISDN Terminal: This non-editable field is the number of the software port of the ISDN Terminal.
• BRI Port: This is the number of the BRI Software port to which the ISDN Terminal is connected on a
Short/Extended Passive Bus line.

• Access Code: Assign Station Access codes to the ISDN Terminal. Station Access codes are
commonly referred to as Extension Numbers. These may be a combination of a maximum of 6 digits,
which are dialed to call the ISDN Terminal to which they are assigned.
A maximum of 6 digits are allowed in an Access Code. By default, the Station Access Codes are blank
for all ISDN Terminal ports.
To assign Station Access Codes according to your preference and requirement to a range of ISDN
Terminals, see “Assigning Access Codes to a Range of Extensions”.
more.
• Name: Assign a 'Name' to the ISDN Terminal. The name may be of the person who will use the ISDN
Terminal or the name of a department. This name will be displayed to the called extension.
• Station Basic Feature Template: Assign a “Station Basic Feature Template” to the ISDN Terminal.
A Station Basic Feature Template includes a set of features that completely define the behaviour of the
Extension, such as Time Table, Operator access, Trunk Access, Class of Service, Toll Control, Call
Budget, and Station Message Detail Records (storage of Incoming and Outgoing Call details).
By default, Station Basic Feature Template 01 is assigned to all extensions of the system that also
includes SLT ports, ISDN ports and E&M Lines with Station as Orientation Type.
Control”, “OG Trunk Bundle Group”, etc.) of the ISDN Terminal.
all ISDN Terminals, retain Template 01.
If different sets of features are to be allowed to different ISDN Terminals, then prepare separate Station
Basic Feature Templates and apply them on the ports. To do this,
• Select a Template number, for example Template number 12.
• Customize Template number 12 and click 'Submit' at the bottom of the page.
• Now go back to the ISDN Terminal Parameters page.
• Enter the number of the Template you customized, Template 12, in the Station Basic Feature
Template field of the ISDN Terminal, for instance, ISDN-01, on which you want to apply this
template. If you want to apply this template to other terminals too, like ISDN-02, 03, and 04, assign
the Template 12 to all them.
• Repeat the same steps to customize and assign a different Template to another ISDN Terminal.

Also, refer the topic “Station Basic Feature Template” to know more about customizing the templates
and applying on extension ports.
• Station Advanced Feature Template: Assign a Station Advanced Feature Template to the ISDN
Terminal. The Advanced Feature Template consists of a set of less commonly used features like Alarm
Notification Type, Caller ID Presentation for Transferred Calls, DDI Incoming Call Routing, Storage of
Internal Calls, Call Duration Control, Floor Service, Call Taping, etc.
which includes ISDN ports, SLT ports, DKP Ports and E&M Lines configured as Stations.
Check if this default template fulfills the feature requirements of the ISDN Terminal by clicking the 'Station
Advanced Feature Template' link.
If the default Template 01 fulfills the feature requirements, and if the same features are to be allowed to all
ISDN Terminals, retain Template 01.
If different sets of features are to be allowed to different ISDN Terminals, then prepare separate Station
To do this,
• Now go back to the ISDN Terminal Parameters' page.
• Enter the number of the Template you customized, Template 04 in the Station Advanced Feature
Template field of the ISDN Terminal, for instance, ISDN-01, on which you want to apply this template.
If you want to apply this template to other terminals too, like ISDN-01, 02, and 03, assign the Template
• Repeat the same steps to customize and assign a different Template to another ISDN Terminal.
Also refer the topic “Station Advanced Feature Template” for instructions on customizing these templates
and applying them on the extension ports.
• Station Type: Make sure you select the option Assistant, if it is the Operator Extension. The system will
play beeps during the conference to the participants to indicate the presence or absence of the Operator.
To know more, see “Conference-3 Party”, “Conference-Multiparty” and “Conference Dial-In”.
For the Hotel Application of the SARVAM UCS, extensions are identified as — Administrator, Assistant or
Guest extensions according to the intended user of the ISDN Terminal. The system will consider the
automatically assign the Guest and Administrator (Hotel Staff) features to the ISDN Terminal. To know
more, refer the SARVAM UCS Hospitality System Manual.

Advanced features
The above listed parameters fulfill the ISDN Terminal configuration requirements of most users. If you need to use
other features like Personal Directory, Call Pick-Up Groups, user details, you may click the Advance button.
Configure the following parameters:
64 characters.
• Personal Directory: Enter the number of the “Personal Directory” that you want to assign to the ISDN
Terminal. A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index
number (location code), Name and Trunk Access Codes (“OG Trunk Bundle Group”). The Personal
The Personal Directory number that you assign to an ISDN Terminal must also be programmed either by
you, the System Engineer, or by the ISDN Terminal user. Refer the topic “Abbreviated Dialing” for
instructions on programming the Personal Directory.

• Priority: Select a Priority Level for the ISDN Terminal from 1, 2, 3... to 9, with '1' being lowest Priority and
'9' being highest Priority. Whenever an extension (ISDN Terminal) with higher priority calls an extension
with lower priority, a triple ring is placed on the called extension. To know more, read the feature
By default, the Priority of all ISDN Terminals is set to '5-Normal'.
• Call Pick-Up Group: Configure this parameter if you want to assign the ISDN Terminal to a particular “Call
Pick Up” group.
Call Pick Up allows the ISDN Terminal user to 'pick up' (answer) calls ringing on any other extension, by
dialing a feature code, without physically going to the ringing extension. For this to work, both the ringing
extension and the extension picking up the call must be in the same 'Call Pick Up Group'. Refer “Call Pick
Up” for instructions on how to create groups. You can create as many as 99 groups numbered from 01 to
99.
Enter the number of the Call Pick-Up Group you created for this ISDN Terminal in this field.
• Repeat the same steps to configure other ISDN Terminals.
• If you have completed configuration of all the above listed ISDN Terminal Parameters, click 'Submit' at the
bottom of the page to save your changes.
Configuring ISDN Terminals using a Telephone

• Enter SE mode using a DKP/SLT.
To assign BRI Software Port to an ISDN Terminal, dial:

• 7301-1-ISDN Terminal-BRI to assign BRI Software Port to a single ISDN Terminal.
• 7301-2-ISDN Terminal-ISDN Terminal-BRI to assign the same BRI Software Port to a range of ISDN
Terminals.
• 7301-*-BRI to assign the same BRI Software Port to all ISDN Terminals.
Where,
ISDN Terminal is the Software Port number of the Terminal from 01 to 64.
BRI is the number of the BRI Software Port to which the ISDN Terminal is connected, from 01 to 32.
5. Default: No port is assigned, 00.
To de-assign ISDN Terminal from a BRI Software Port, dial:

• 7301-1-ISDN Terminal-00 to de-assign a single Terminal from a BRI port.
• 7301-2-ISDN Terminal-ISDN Terminal-00 to de-assign a range of Terminals from a BRI port.
• 7301-*-00 to de-assign all Terminals from a BRI port.
To assign Access Code to an ISDN Terminal, dial:

• 3103-1- ISDN Terminal -Access Code-#*
Where,
Access Code is a number string of any combination of a maximum of 6 digits. Terminate the command
with #* if the number string has fewer than 6 digits. (0 to 9, #, * are allowed)
To clear the Access codes assigned to the ISDN Terminal, dial:

• 3103-1-ISDN Terminal-#* to clear the Access Code of a single terminal.
• 3103-2-ISDN Terminal-ISDN Terminal-#* to clear the Access Code of a range of terminals.
• 3103-*-#* to clear the Access Codes of all terminals.

To assign default Station Access Codes to ISDN Terminal, dial:
• 3153-1-ISDN Terminal to default Access Code of a single terminal.
• 3153-2-ISDN Terminal-ISDN Terminal to default Access Codes of a range of terminals.
• 3153-* to default Access Codes of all terminals.
To assign a Name to an ISDN Terminal, dial:

• 5409-1-ISDN Terminal-Name-#* to assign a Name to a single terminal.
• 5409-2-ISDN Terminal-ISDN Terminal-Name-#* to assign the same Name to a range of terminals.
• 5409-*-Name-#* to assign the same Name to all terminals.
Where,
Name is a string of a maximum of 18 alphanumeric characters. Terminate the commands with #* if the
To clear the Name of the ISDN Terminal, dial:

• 5409-1-ISDN Terminal-#* to clear the Name of a single terminal.
• 5409-2-ISDN Terminal-ISDN Terminal-#* to clear the Names of a range terminals.
• 5409-*-#* to clear the Names of all terminals.
To assign a Station Basic Feature Template to an ISDN Terminal, dial:

• 5507-1-ISDN Terminal-Template Number to assign a template to a single terminal.
• 5507-2-ISDN Terminal-ISDN Terminal-Template Number to assign the same template to a range of
terminals.
• 5507-*-Template Number to assign the same template to all terminals.
Where,
Template Number is the number of the Station Basic Feature Template, from 01 to 50. Default: 01
To assign a Station Advanced Feature Template to an ISDN Terminal, dial:

• 5607-1-ISDN Terminal-Template Number to assign a template to a single terminal.
• 5607-2-ISDN Terminal-ISDN Terminal-Template Number to assign the same template to a range of
terminals.
• 5607-*-Template Number to assign the same template to all terminals.
Where,
Template Number is the number of the Station Advanced Feature Template, from 01 to 50. Default: 01
To select a Station Type for an ISDN Terminal, refer the SARVAM UCS Hospitality System Manual.
For Configuration of Advanced Features, use the following commands:
To assign a Personal Directory to an ISDN Terminal, dial:

• 1907-1-ISDN Terminal-Personal Directory to assign directory to a single terminal.
• 1907-2-ISDN Terminal-ISDN Terminal-Personal Directory to assign the same directory to a range of
terminals.
• 1907-*-Personal Directory to assign the same directory to all terminals.
Where,
Personal Directory number is from 01 to 50.
To clear the Personal Directory assigned to an ISDN Terminal, dial:

• 1907-1-ISDN Terminal-00 to clear the directory of a single terminal.
• 1907-2-ISDN Terminal-ISDN Terminal-00 to clear the directory of a range of terminals.
• 1907-* to clear the directory from all terminals.

To define the Priority for an ISDN Terminal, dial:
• 3913-1-ISDN Terminal-Priority to define Priority for a single terminal.
• 3913-2-ISDN Terminal-ISDN Terminal-Priority to define the same Priority for a range of terminals.
• 3913-*-Priority to define the same Priority for all terminals.
Where,
To assign an ISDN Terminal to a Call Pick-Up Group, dial:

• 3903-1-ISDN Terminal-Call Pick-Up Group to assign a single terminal to a Call Pick-Up group.
• 3903-2-ISDN Terminal-ISDN Terminal-Call Pickup Group to assign a range of terminals to the same
Call Pick-Up group.
• 3903-*-Call Pickup Group to assign all terminals to the same Call Pick-Up group.
Where,
To remove an ISDN Terminal from a Call Pick-Up Group, dial:

• 3903-1-ISDN Terminal-00 to remove a single terminal from a Call Pick-Up group.
• 3903-2-ISDN Terminal-ISDN Terminal-00 to remove a range of terminals from a Call Pick-Up group.
• 3903-*-00 to remove all terminals from a Call Pick-Up group.
• Exit SE mode.

Configuring SLT Extensions
The number of SLT extensions available to you for configuration depends on the number of SLT ports supported by
SARVAM UCS and the number of SLT ports you have specified on the “System Pre-requisites” page.
If you have enabled On-Site Configuration, the system will provide you only those ports that are actually present
in the system for configuration.
Configure SLT port parameters using Jeeves or by dialing commands from a Telephone. If you have installed a
Voice Mail module in SARVAM UCS, you may provide voice mail facility to all or selected SLT extensions. To
provide voice mail to extensions, configure VMS Settings.
Configuring SLT Parameters using Jeeves

• Configure the following parameters for each SLT port on this page:
• SLT Port No.: This non-editable field is the number of the software port of the SLT port.
• Hardware Slot - Port: 'Slot' is the number of the Universal Slot in which the SLT Card is inserted. 'Port'
is the number of the SLT hardware port on which the telephone instrument is connected.
automatically to the SLT software ports.
For example: if you have inserted the SLT8 Card in Slot number 02 and SLT16 Card in Slot number 03
of SARVAM UCS, the system will assign the hardware slot 02 and port numbers 01-08 to the SLT
Software Ports from 001 to 008 respectively. The system will assign Slot 03 and port numbers 09-24 to
the SLT Software Ports 009 to 024. Refer the topic “Software Port and Hardware ID” to know more.
However, if required, you may change the Hardware Slot and Port assigned to the SLT software port. In
this case, enter the desired Hardware Slot and Port number in this field.

• Access Code: Assign Station Access Codes to the SLT Port148. Station Access Codes are commonly
referred to as Extension Numbers. These may be a maximum of 6 digits, which are dialed to call the
SLT port to which they are assigned.
All SLT ports are assigned the following Station Access Codes as default.
SLT Software Port Access Codes
001 2001
002 2002
003 2003
: :
: :
512 2512
You may either apply the default Station Access Codes to the SLT ports or assign them according to
your requirement and preferences.
To assign Station Access Codes according to your preference and requirement to a range of SLT
more.
• Name: Assign a 'Name' to the SLT port. The name may be of the person who will use the SLT or the
name of a department. This name will be displayed on the LCD of the remote user's phone, if it is
equipped with Caller ID.
You can program a name of a maximum of 18 alphanumeric characters.
• Station Basic Feature Template: Assign a “Station Basic Feature Template” to the SLT. A Station
Basic Feature Template includes a set of features that completely define the behaviour of the extension
port, such as Time Table, Operator access, Trunk Access, Class of Service, Toll Control, Call Budget,
and Station Message Detail Records (storage of Incoming and Outgoing Call details).
By default, Station Basic Feature Template 01 is assigned to all extensions of the system that includes
DKP ports, ISDN Terminals, SIP extensions, and E&M Lines with Station as Orientation Type.
Control”, “OG Trunk Bundle Group”, etc.) of the SLT extension user.
all SLTs, retain Template 01.
If different sets of features are to be allowed to different SLTs, then prepare separate Station Basic
Feature Templates and apply them on the ports. To do this,
148. The number of SLT ports vary according to the variant of SARVAM UCS.

• Customize Template number 10 and click Submit at the bottom of the page.
• Now go back to the SLT Parameters page.
• Enter the number of the Template you customized, Template 10 in the Station Basic Feature
Template field of the SLT Port, for instance SLT-003, on which you want to apply this template.
• Click 'Submit' at the bottom of the page to save changes.
• Repeat the same steps to customize and assign a different Template to another SLT port.
and applying on the ports.
• Station Advanced Feature Template: Assign a Station Advanced Feature Template to the SLT. The
Advanced Feature Template consists of a set of less commonly used features like Alarm Notification
Type, Caller ID Presentation for Transferred Calls, DDI Incoming Call Routing, Storage of Internal
Calls, Call Duration Control, Floor Service, Call Taping, etc.
which includes DKP Ports, SLT Ports, ISDN Terminals, SIP Extensions, and E&M Lines configured as
Stations.
Check if this default template fulfills the feature requirements of the SLT extension users by clicking the
link Station Advanced Feature Template.
all SLT ports, retain Template 01.
If different sets of features are to be allowed to different SLT Ports, then prepare separate Station
To do this,
• Return to SLT Parameters page.
• Enter the number of the Template you customized, template 02, in the Station Advanced Feature
Template field of the SLT Ports on which you want to apply this template.
• Repeat the same steps to customize and assign a different Template to another SLT port.

• SLT Hardware Template: Assign an SLT Hardware Template to the SLT port. An SLT Hardware
Template is a set of features that define the behavior of the SLT hardware port. The SLT Hardware
Template allows you to configure according to user requirements a common set of features for all SLT
Hardware Ports, like Caller ID Presentation (DTMF, FSK), Digit Pad Count, Ring Type, AC Impedance,
Answer Signaling type, Speech Transmit and Receive Gains, Open Loop Disconnect, Loop Current,
and Fax connectivity.
There are 50 SLT Hardware Templates that can be customized and assigned to the SLT ports. By
default SLT Hardware Template Number 01 is assigned to all the SLTs. This template has default
values fulfilling the common requirements of a very broad user base.Check if the values in this
template fulfill your requirements.
If the default SLT Hardware Template 01 fulfills the feature requirements and if the same features are
to be allowed to all SLTs, retain Template 01.
If different sets of hardware features are to be allowed to different SLTs, then prepare separate SLT
Hardware Templates and apply them on the ports. To do this,
• Click SLT Hardware Template to open the page.
• Now go back to the SLT Parameters page.
• Enter the number of the Template you customized, template 02, in the SLT Hardware Template
field of the SLT Ports on which you want to apply this template.
• Repeat the same steps to customize and assign a different SLT Hardware Template to another SLT
port.
Also, refer the topic “SLT Hardware Template” to know more about customizing the templates and
applying on the SLT ports.
• Call Pick-Up Group: Configure this parameter if you want to assign the SLT to a particular “Call Pick
Up” group.
Call Pick Up allows the SLT extension user to 'pick up' (answer) calls ringing on any other extension, by
dialing a feature code, without physically going to the ringing extension.
For this to work, both extensions, the ringing extension and the extension picking up the call, must be in
the same Call Pick Up Group. Refer “Call Pick Up” for instructions on how to create groups. You can
create as many as 99 groups numbered from 01 to 99.
Enter the number of the Call Pick-Up Group you created for this SLT in this field.

• COSEC Door Group: You must assign the extension user to a COSEC Door Group for COSEC
Integration. The users in the same group must be assigned the same group. You can create as many
as 50 groups numbered from 00 to 50. Users who are assigned COSEC Door Group ‘00’ are not a part
of any group. See “COSEC Integration” for more information.
or Guest extensions according to the intended user of the SLT. The system will consider the
automatically assign the Guest and Administrator (Hotel Staff) features to the SLT. To know more, refer
the SARVAM UCS Hospitality System Manual.
or Guest extensions according to the intended user of the SLT. The system will consider the
automatically assign the Guest and Administrator (Hotel Staff) features to the SLT. To know more, refer
the SARVAM UCS Hospitality System Manual.
For extension users who need to be provided features like Personal Directory or assigned a Priority, as well as to
enter the user details, you may click the Advance button at the bottom of the page and program the following
parameters:

64 characters.
• Personal Directory: Enter the number of the Personal Directory that you want to assign to this SLT. A
Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes (“OG Trunk Bundle Group”). The Personal Directory is
necessary for using the features “Abbreviated Dialing” and “Dial By Name”.
When a Personal Directory is assigned to an SLT, it must also be configured. The Personal Directory can
be configured by the SLT users and by the System Engineer. Refer the topic “Abbreviated Dialing” for
instructions on configuring the Personal Directory.
• Priority: Select a Priority Level for the SLT.
lowest Priority and '9' being highest Priority. Whenever an extension phone with higher priority calls an

By default, the Priority of all SLT ports is set to '5-Normal'. So, decide what Priority Level you will assign to
each of the SLTs and set the desired level for each port.
• If you have completed configuring SLT parameters, click Submit at the bottom of the page to save your
settings.
It is possible to default all the parameters by clicking the Default button. You can also restore default
values of the parameters of a single SLT by clicking the Default One button and specifying the SLT you
want to set to default.
Configuring SLT Parameters using a Telephone149

To assign Hardware Slot-Port to an SLT Port, dial:

• 1101-SLT-Slot-Port offset on the card
Where,
Slot is Slot number in which the SLT Card is inserted from 01 to 16
Port Offset is the number of the SLT Port on the card from 01 to 99.
To de-assign the hardware slot and the hardware port of an SLT port, dial:
• 1101-SLT-00-00
To assign Access Code to an SLT Port, dial:

• 3101-1-SLT-Access Code-#*
Where,
Access Code is a number string of any combination of 1, 2, 3 or 4 digits. Terminate the command with
#* if the number string has fewer than 4 digits.
To clear the Access codes assigned to the SLT, dial:

• 3101-1-SLT-#* to clear the Access Codes of a single SLT port.
• 3101-2-SLT-SLT-#* to clear the Access Codes of a range of SLT ports.
• 3101-*-#* to clear the Access Codes of all SLT ports.
To assign default Access Codes assigned to SLT, dial:

• 3151-1-SLT to default Access Codes of a single SLT port.
• 3151-2-SLT-SLT to default Access Codes of a range of SLT ports.
• 3151-* to default Access Codes of all SLT ports.
To assign a Name to an SLT, dial:

• 5402-1-SLT-Name-#* to assign a Name to a single SLT port.
• 5402-2-SLT-SLT-Name-#* to assign the same Name to a range of SLT ports.
• 5402-*-Name-#* to assign the same Name to all SLT ports.
Where,
Name is a string of a maximum 18 alphanumeric characters. Terminate the commands with #* if the
149. Using commands you can configure upto 999 ports in ETERNITY LENX. The remaining ports can be configured from Jeeves only.

To clear the name of the SLT, dial:
• 5402-1-SLT-#* to clear the Name of a single SLT port.
• 5402-2-SLT-SLT-#* to clear the Names of a range of SLT ports.
• 5402-*-#* to clear the Names of all SLT ports.
To assign an SLT Hardware Template to an SLT port, dial:

• 5703-1-SLT-Template Number to assign a hardware template to a single SLT port.
• 5703-2-SLT-SLT-Template Number to assign a hardware template to a range of SLT ports.
• 5703-*-Template Number to assign a hardware template to all SLT ports.
Where,
Template Number is the number of the SLT Hardware Template, from 01 to 50. Default: 01.
To assign a Station Basic Feature Template to an SLT Port, dial:

• 5503-1-SLT-Template Number to assign a Template to a single SLT port.
• 5503-2-SLT-SLT-Template Number to assign the same Template to a range of SLT ports.
• 5503-*-Template Number to assign the same Template to all SLT ports.
Where,
Template is the Station Basic Feature Template from 01 to 50. Default: 01.
To assign a Station Advanced Feature Template to an SLT Port, dial:

• 5603-1-SLT-Template Number to assign a Template to a single SLT port
• 5603-2-SLT-SLT-Template Number to assign the same Template to a range of SLT ports.
• 5603-*-Template Number to assign the same Template to all SLT ports.
Where,
Template is the Station Advanced Feature Template from 01 to 50. Default: 01.
To assign a Call Pick-Up Group to an SLT Port, dial:

• 3901-1-SLT-Call Pickup Group to assign a single SLT port to a Call Pick-Up Group.
• 3901-2-SLT-SLT-Call Pickup Group to assign a range of SLT ports to the same Call Pick-Up Group.
• 3901-*-Call Pickup Group to assign all SLT ports to the same Call Pick-Up Group.
Where,
To remove an SLT from a Call Pick Up group, dial:

• 3901-1-SLT-00 to remove a single SLT port from a group.
• 3901-2-SLT-SLT-00 to remove a range of SLT ports from a group.
• 3901-* to remove all SLT ports from a group.
To define the Station Type for an SLT Port, Refer SARVAM UCS Hospitality System Manual.
To assign a Personal Directory to an SLT Port, dial:

• 1905-1-SLT-Personal Directory to assign a Personal Directory to a single SLT port.
• 1905-2-SLT-SLT-Personal Directory to assign the same Personal Directory to a range of SLT ports.
• 1905-*-Personal Directory to assign the same Personal Director to all SLT ports.
Where,
Personal Directory Number is from 01 to 50.

To clear the Personal Directory assigned to the SLT, dial:
• 1905-1-SLT-00 to clear the Personal Directory of a single SLT port.
• 1905-2-SLT-SLT-00 to clear the Personal Directory of a range of SLT ports.
• 1905-* to clear the Personal Directory of all SLT ports.
To define the Priority for an SLT Port, dial:

• 3911-1-SLT-Priority to define Priority for a single SLT port.
• 3911-2-SLT-SLT-Priority to define the same Priority for a range of SLT ports.
• 3911-*-Priority to define the same Priority for all SLT ports.
Where,
• Exit SE mode.

Configuring Radio Interface
The Radio Card acts as an interface between the system and the Radio devices such as Radio Phones, Radio
Repeaters, to add two-way radio functionality in SARVAM UCS. In Two-way radio, the speech can be transmitted
as well as received by the radio devices. Radio devices are also called Radio Transceivers /Combat-Net Radio
devices. The Two-way radio works on High Frequency (HF), Very High Frequency (VHF) or Ultra High Frequency
(UHF).
A Radio Transceiver offers the following interfaces:

• Frequency tuner to set the frequency for receiving and sending audio messages
• Speaker to playback the audio received
• Microphone to send the audio message
• Push-to-Talk (PTT) button to activate radio transmission
Generally, the Radio Transceivers remain in the receiving (Rx) mode, so that the broadcasted audio messages can
be heard. To transmit speech, you must press the PTT button on the Radio Transceiver.
Additional interfaces are also supported on Radio Transceivers, like Dial Pad, to dial out the DTMF digits as audio
message. Such interfaces can be used where the Radio Transceivers are interfaced with the system, which allows
you to dial the DTMF digits.
The Radio Interface is supported on SARVAM UCS. A maximum of 16 Radio Ports are supported.
Matrix does not supply Radio Transceivers.
How the Radio Interface works

The Radio Port of SARVAM UCS works as a Station.
Making a Call from the Radio Port

An extension user dials the Radio Port Access Code to call a Radio device.
• The Radio Port transmits the audio signal to the Radio device and the extension user gets Ring Back Tone
for 3 seconds.
• Since the Radio Port is always in the receiving mode, the system connects the speech path between the
Radio device user (connected to the Radio Port) and the extension user.
• The extension user (caller) can now converse with the Radio device user.
• When the Radio device user wants to talk, s/he must press the PTT button of the Radio device and talk.
The extension user can now listen to the Radio device user.
• When the extension user wants talk to the Radio device user, s/he must press the ‘#’ key. Similarly to listen
to the Radio device user the extension user must press ‘#’ key again.
To connect/disconnect speech path using the ‘#’ key, make sure the option Using '#' key is selected as
PTT Activation in Radio Extension Parameter.
• The extension user can disconnect the call by going On-hook.

If the Radio port is busy, the extension user will hear the Busy Tone.
The Radio extension user (3000) dials another Radio extension user (3001).
• Audio signal is detected on the Radio port (3001) and the Radio extension user (3000) gets Ring Back
Tone for 3 seconds.
• Since the Radio Port (3001) is always in the receiving mode, the system connects the speech path
between both the Radio extension users.
• Both users must press the PTT button of their Radio device to talk. At a time only one user can speak.
• If the system detects silence for one minute, it disconnects the call.
If the Radio port is busy, the extension user will hear the Busy Tone.
You must configure the Radio parameters so as to use the Radio functionality in the SARVAM UCS. You can
configure Radio parameters only through Jeeves.
Configuring Radio Parameters using Jeeves

• Under Configuration, click Radio Extension Parameters to open the page.
• Configure the following parameters.
• Enable Port: This flag is for enabling or disabling a Radio port. When a Radio port is disabled, neither
incoming nor outgoing calls can be made from that port.
By default, the port is enabled. Clear the check box to disable the ports.
• Hardware Slot and Port: 'Slot' is the number of the Universal Slot in which the Radio Card has been
inserted. 'Port' is the number of the hardware port on the card to which the Radio device is connected.

By default the SARVAM UCS can detect and assign the hardware slot and port numbers automatically
to the Radio (software) ports. However, you may change the Hardware Slot and Port assigned to the
Radio software port. If required, enter the desired Hardware Slot and Port number in this field.
If you want to de-assign the Hardware Slot and Port, enter '00' in both fields.
• Access Code: Assign Station Access Codes to the Radio Port. Station Access Codes are commonly
referred to as Extension Numbers. These may be number strings of a maximum 6 digits, which are to
be dialed to call the Radio port to which they are assigned.
To assign Station Access Codes according to your preference and requirement to a range of Radio
By default, the Station Access Codes are blank for all Radio ports.
• If you decide to customize the Station Access Codes, make sure that the numbers do not clash with
any other Access Code in the 'Dial' phase. Refer the topics “Access Codes” and “Conflict Dialing” to
know more.
• Name: Assign a 'Name' to the Radio port. The name may be of the person who will use the Radio
device or the name of the department or location of the device. This name will be displayed on the LCD
of the Operator/extension user's phone, if it is equipped with Caller ID.
• On Detecting Voice: Select Route to Operator, to route incoming calls on the Radio Port to the
Operator.
Select Greet to Dial, if you want the callers to dial the desired extension number. The call will be routed
to the dialed number. This option must be selected only when you have a Dial Pad connected to the
Radio Device. You can customize the message played to the callers, if required. For detailed
instructions, see “Voice Message Applications”.
If you select Greet to Dial, the parameter Time Between Consecutive PTT will be un-editable and the
parameter PTT Count to Place a call will be set to 1 and will be un-editable.
• Station Basic Feature Template: As the Radio Port functions as a station, assign a “Station Basic
Feature Template” to the Radio port.
Only the following features of the Station Basic Feature Template are applied on the Radio Port:
• Time Table
• Operator
• Class of Service (Only Global Directory Part 1, 2 and 3, Basic Features and Privacy from Built-In
Auto Attendant)
• Call Budget
• Call Privilege
• Toll Control-Call Budget consumed
• Outgoing Trunk Bundle Group (WH, BH and NH)
• Store Outgoing Calls
• Store Incoming Calls
By default, Station Basic Feature Template number 01 is also applied on Radio ports.

Check if the default settings of the features applied on the Radio ports (Time Table, Operator, Class of
Service, Storage of Outgoing and Incoming Calls etc) match your requirements for the Radio ports. If
yes, retain the default Station Basic Feature Template 01.
If you want to change any of the feature settings in for the Radio Ports, you may prepare a different
Template150, for example, Template 14 and apply it on the Radio Ports.
Also, if different feature settings are to be applied on different Radio Ports prepare separate Station
• Click the Station Basic Feature Template link to open the page.
• Customize the Radio Port related features (listed above) in Template number 14 and click Submit
to save.
• Now go back to the Radio Parameters page.
• Enter the number of the Template you customized, Template 14 in the 'Station Basic Feature
Template' field of the Radio Port, for example: Port 1, on which you want to apply this template. If
you want to apply this template to other ports too, like 2, 3, and 4, assign the Template 14 to all
these ports.
• If required, repeat the same steps to customize and assign a different Template to another Radio
port.
Refer the topic “Station Basic Feature Template” to know more about customizing the templates and
applying on the ports.
• Station Advance Feature Template: Assign a Station Advanced Feature Template to the Radio Port.
Only the following features in Station Advance Feature Templates are applied on Radio Ports.
• DDI IC Routing
• Send DDI Number as CLI?
• Internal Calls Storage
• CDC Table
• Call Tapping
• Caller Category
By default Station Advanced Feature Template 01 is assigned to all the Radio ports. Check if the
default settings of the features applied on the Radio ports (Internal Call Storage flag and Call Forward
Ring Timer) fulfill the feature requirements of the Radio ports, by opening the 'Station Advanced
Feature Template' link.
If the default template fulfills your requirement, retain the default Station Basic Feature Template 01 for
the Radio ports.
150. This is recommended because changing the values of the default Template will be applied on all other extension types to which the
Template is assigned.

However, if you want to change any of the feature settings in for the Radio Ports, you may prepare a
different Template151, for example, Template 05 and apply it on the Radio Ports. To do this,
• Click the Station Advanced Feature Template link to open the page.
• Now go back to the Radio Parameters page.
• Enter the number of the Template you customized, Template 05 in the 'Station Advanced Feature
Template' field of the Radio Port, for example, 1, on which you want to apply this template. If you
want to apply this template to other terminals too, like 2, 3, and 4, assign the Template 05 to all
these ports.
• Repeat the same steps to customize and assign a different Template to another Radio Port.
templates and applying them on the station ports.
• Priority: Select a Priority Level for the Radio Port from 1, 2, 3... to 9, with '1' being lowest Priority and '9'
being highest Priority. Whenever an extension (Radio Port) with higher priority calls an extension with
lower priority, a triple ring is placed on the called extension. To know more, read the feature description
“Priority”.
By default, the Priority of all Radio ports is set to '5-Normal'.
The above listed parameters fulfill the basic Radio extension port configuration requirements of most users.
However, it is anticipated that some users may need to configure the more advanced features like Radio VAD

Threshold Level, Minimum Silence Time for valid IC call (msec), Minimum Speech Time to activate PTT (msec) etc.
For such users, you may click the 'Advance' button.
• Scroll with the horizontal bar, and configure the following advanced Radio Parameters:
• PTT Activation: Select the desired option to activate PTT. You can select:
• On Speech Detection
• Using ‘#’ Key
If you select On Speech Detection, the PTT activation/deactivation will be done automatically. It will
depend on the values of Minimum Speech Time to activate PTT (msec) and Minimum Silence Time
to deactivate PTT (msec).
If you select Using ‘#’ Key, the user must press the ‘#’ key to activate as well as deactivate PTT. It is not
depend on the values of Minimum Speech Time to activate PTT (msec) and Minimum Silence Time
to deactivate PTT (msec).

• Radio VAD Threshold Level for Valid IC Call: This parameter defines the level below which the
Radio port would not validate the audio signal as valid speech packet for incoming call. Set the Radio
VAD Threshold Level for Valid IC Call to the desired value. The range of this level is 0 to -96 dBm. By
default, it is -15 dBm.
• Radio VAD Threshold Level to activate PTT: This parameter defines the level below which the Radio
port would not validate the audio signal as valid speech packet to activate PTT. Set the Radio VAD
Threshold Level to activate PTT to the desired value. The range of this level is 0 to -96 dBm. By default,
it is -15 dBm.
• VAD Hang Timer (msec): This timer is applicable after the Minimum Silence Time for valid IC call
(msec). The system will wait for the duration of this timer to expire after the expiry Minimum Silence
Time for valid IC call (msec) timer to treat silence as valid silence before speech packets. The range is
50 - 32767. By default, it is 500 msec.
• Minimum Speech Time for valid IC call (msec): Set the duration for which the system must receive
speech packets for a valid incoming call. The range is 100 to 9999 msec. By default, it is 500 msec.
• Minimum Silence Time for valid IC call (msec): Set the duration for which the system may detect
and accept silence before actual speech packets, to treat this silence as a valid incoming call. The
range is 100 to 9999 msec. By default, it is 1200 msec.
• Minimum Sample Value for Speech Detection: This parameter defines the minimum value for
detecting speech on the Radio Port be considered by the system.The valid range is 100 to 900. By
default, it is 300.
• Time Between Consecutive PTT (msec): This parameter defines the maximum Time Between
Consecutive PTT. The valid range is 2000 to 6000 msec. By default, it is 4000 msec.
This parameter is applicable only if you have selected Route to Operator as the On Detecting Voice
option.
• PTT Count to Place call: This parameter defines the count after which the system will place the call.
The valid range is 1 to 5. By default, it is 3.
This parameter is applicable only if you have selected Route to Operator as the On Detecting Voice
option.
• Minimum Speech Time to activate PTT (msec): Set this timer to ensure continuous communication,
once the call matures. As per the set time, the PTT will be activated and speech will be transmitted.
The range is 100 to 9999 msec. By default, it is 700 msec.
• Minimum Silence Time to deactivate PTT (msec): Set the duration of silence after which the system
must deactivate PTT to disconnect the call. The range is 100 to 9999 msec. By default, it is 400 msec.
• Play Error Tone after Routing or Speech: This flag is for enabling or disabling the tone to be played
by the system. If you want the system to play error tone after routing the call or after disconnecting the
speech select the check box to enable. By default, it is disabled.

Configuring DTMF Detection on Radio
The DTMF Detection and Generation parameters are applicable only when there are Dial Pads connected to the
Radio devices.
• Under Configuration, click System Parameters.

• Click Radio to expand.
• DTMF Detection - Minimum Level (dB): This parameter signifies the minimum level (dB) of the DTMF
digit to be considered as valid. By default, Minimum levels set to -10.5dB.
• DTMF Detection - Minimum ON Time (msec): This parameter signifies the minimum time period for
which the DTMF signal should be present in order to be detected. The valid range of this time is 17 to
204 milliseconds. By default, Minimum ON Time is set to 17 milliseconds.
• DTMF Detection - Minimum OFF Time (msec): This parameter signifies the minimum time period
between successive DTMF digits. The valid range of this time is 17 to 204 milliseconds. By default,
Minimum OFF Time is set to 17 milliseconds.
• DTMF Generation - DTMF ON Time (msec): It is the width of DTMF digit to be dialed out by the Radio
port. By default the ON Time is set to 102 milliseconds.
• DTMF Generation - Inter Digit Pause Time (msec): When the Radio port dials out the DTMF digits, it
waits for the Inter Digit Pause Timer, while dialing the DTMF digits. By default the timer is set to 102
milliseconds.
• Click Submit.

Configuring Magneto Interface
A magneto telephone is a local battery telephone set, in which signaling current is provided by a hand generator,
usually a magneto. The hand generator, commonly referred to as 'crank', is located on the right hand side of the
telephone set and is turned to produce energy to ring other phones.
Tip and Ring wires of the magneto phone are dry, that is, they do not carry battery voltage when idle or in speech.
This means the wires carry only AC voice signals and not the DC voltage of the battery.
The Battery is used to power the microphone only and ringing generator is used to provide ring voltage to alert the
other phone.
Magneto Telephones are widely used by defense establishments as field phones in front lines, and by other
establishments such as railroad companies (signaling emergencies, crossings, etc.), electric utilities, pipeline
companies, who need to have their networks at places that are too remote to be serviced by public telephone
networks.
SARVAM UCS supports the Magneto Interface. A maximum of 16 ports are supported in SARVAM UCS. The
availability of ports depends on the number of Magneto Cards installed in the system.
Matrix does not supply Magneto Telephones.
How the Magneto Interface works

The Magneto port of SARVAM UCS works as a Station.
Making a Call to the Magneto Port

An extension user (SLT/DKP/ISDN Terminal) dials the Magneto Port Access Code to call a Magneto Field
Telephone.
• Ring is played on the Magneto Field Telephone connected to the Magneto Port.
• The extension user gets Ring Back Tone for the duration of the Ring Back Tone Timer.
• The extension user must press # or the Magneto Ring Enable (MRE) Key (on the DKP) before the expiry of
the Ring Back Tone Timer (programmable; default: 45 seconds) to check if the Magneto User has
answered the call152.
• The system considers pressing of the # or MRE Key as call maturity and connects the speech path
between the Magneto Field Telephone (connected to the Magneto Port) and the extension phone.
• If the Magneto User has answered the call, the extension user may start speech on hearing the Magneto
User's voice.
• If there is a period of silence, the extension user may press the # or MRE Key again to generate Ring on
the Magneto Telephone, and press the # and MRE Key once again during the Ring Back Tone Timer to
check speech with the Magneto User.
152. As there is no Answer Signaling or Call Disconnect feature on the Magneto line, there is no way for the Extension user to know
whether speech has been established with the Magneto User, but to wait to hear the Magneto User's voice.

• Either party can disconnect the call:
• The extension user either goes ON-Hook while in speech. Error Tone will be played to the Magneto
user for the duration of the Error Tone Timer and the system releases the Magneto Port.
• The Magneto User cranks the hand generator (Ring Down Signal) while in speech to disconnect the
call. The system detects the Ring Current and releases the Magneto Port.
• If the Magneto port is busy, the extension user will hear the Busy Tone.
• If the extension user goes ON-Hook while the Magneto Phone is ringing, the system will stop the ring
on the Magneto phone.
Making a Call from the Magneto Port

• The Magneto Field Telephone user cranks the hand generator to generate ringing current.
• The system detects the ringing current. If the ring current is present for more than 500msec, the system
treats it as a Call request from the Magneto Port.
• The system rings the Operator extension assigned to the Magneto Port for the duration of the Ring Back
Tone Timer and plays Ring Back Tone to the Magneto Telephone user.
• If the Operator goes OFF-Hook, before the expiry of the Ring Back Tone Timer, speech is established
between the Operator and the Magneto User.
• The Operator can now transfer the call to another extension or external number.
• Either party can disconnect the call:
• The Operator can go ON-Hook while in speech to disconnect. Error Tone will be played to the Magneto
user.
OR
• The Magneto User can cranks the hand generator again (Ring Down Signal) while in speech to
disconnect the call.
• The system releases the Magneto Port and the port with which the Magneto User is in speech.
• If neither the Operator nor any of the two Magneto station users in speech disconnects the call, the system
will automatically disconnect the call if silence (no speech) is detected for more than a specified duration of
time.
• For this, the flag 'Enable Silence Detection on Magneto' must be enabled in the System Parameters and
the 'Magneto Silence Disconnection Timer (default: 60 seconds) must be programmed. When Silence
Disconnection is enabled on Magneto port, the system will start the Magneto Silence Disconnection Timer
as soon as it detects silence. If the continuous silence is detected till the expiry of the timer the system
considers the conversation as over and releases the Magneto stations. The call is disconnected. However,
if speech is detected during this timer, the timer will be stopped.
• The call can be disconnected also using Forced Release by any extension user (Operator or any other
extension user) having the feature 'Forced Release' in its Class of Service.

• If the Operator is busy, Busy Tone is played to the Magneto user.
• The Magneto user may crank the hand generator again to send Ring Current on the expiry of the Ring
Back Tone Timer. However, if the Magneto user sends the Ring Current again during the Ring Back
Tone Timer, the system will treat it as a call disconnect event. It will stop ringing the Operator's
extension and release the Magneto Port.
• The Magneto Ring Enable (MRE) Key should be programmed on the DKP extensions that are to be
allowed access to call Magneto ports.
• On SLT extensions and ISDN Terminals, the # key will serve the function of the MRE key.
Configuring Magneto Ports using Jeeves

• Under Configuration, click Magneto Configuration.
Configure the following Magneto port parameters:
• Magneto No.: This non-editable field is the number of the software port of the Magneto port.
• HW Slot-Port: 'Slot' is the number of the Universal Slot in which the Magneto Card is inserted. 'Port' is
the number of the Magneto hardware port on which the Magneto Field telephone instrument is
connected.
automatically to the Magneto software ports.
However, if required, you may change the Hardware Slot and Port assigned to the Magneto software
port. In this case, enter the desired Hardware Slot and Port number in this field.

• Enable Port: This flag is for enabling or disabling a Magneto port. When a Magneto port is disabled,
neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by selecting the check
box.
• Access Code: Assign Station Access Codes to the Magneto Port. Station Access Codes are
commonly referred to as Extension Numbers. These may be number strings of a maximum 6 digits,
which are to be dialed to call the Magneto port to which they are assigned.
A maximum of 6 digits are allowed in an Access Code. By default, the Station Access Codes are blank
for all Magneto ports.
To assign Station Access Codes according to your preference and requirement to a range of Magneto
more.
• Name: Assign a 'Name' to the Magneto port. The name may be of the person who will use the Magneto
telephone or the name of the department or location of the telephone. This name will be displayed on
the LCD of the Operator/extension user's phone, if it is equipped with Caller ID.
• Station Basic Feature Template: As the Magneto Port functions as a station, assign a “Station Basic
Feature Template” to the Magneto port.
Only the following features of the Station Basic Feature Template are applied on the Magneto Port:
• Time Table
• Operator
• Store Outgoing Calls
• Store Incoming Calls
By default, Station Basic Feature Template number 01 is assigned to all extension types of the
SARVAM UCS (SLT, DKP, ISDN Terminals, E&M Lines with Station as Orientation Type). Template 01
is also applied on Magneto ports by default.
Check if the default settings of the features applied on the Magneto ports (Time Table, Operator, Class
of Service, Storage of Outgoing and Incoming Calls) match your requirements for the Magneto ports. If
yes, retain the default Station Basic Feature Template 01.
If you want to change any of the feature settings in for the Magneto Ports, you may prepare a different
Template153, for example, Template 14 and apply it on the Magneto Ports.
Also, if different feature settings are to be applied on different Magneto Ports prepare separate Station
• Click the Station Basic Feature Template link to open the page.

• Customize the Magneto Port related features (listed above) in Template number 14 and click
Submit to save.
• Now go back to the Magneto Parameters page.
Template' field of the Magneto Port, for example: MAG-001, on which you want to apply this
template. If you want to apply this template to other ports too, like MAG-002, 003, and 004, assign
the Template 14 to all these ports.
• If required, repeat the same steps to customize and assign a different Template to another Magneto
port.
Refer the topic “Station Basic Feature Template” to know more about customizing the templates and
applying on the ports.
• Station Advance Feature Template: Assign a Station Advanced Feature Template to the Magneto
Port.
Only the following features in Station Advance Feature Templates are applied on Magneto Ports.
• Internal Calls Storage
• Call Forward Ring Timer
which also includes DKP ports, SLT ports, ISDN Terminals and E&M Lines configured as Stations.
Check if this default template fulfills the feature requirements of the Magneto Ports by opening the link
'Station Advanced Feature Template'.
Check if the default settings of the features applied on the Magneto ports (Internal Call Storage flag
and Call Forward Ring Timer) fulfill the feature requirements of the Magneto ports, by opening the
'Station Advanced Feature Template' link.
If the default template fulfills your requirement, retain the default Station Basic Feature Template 01 for
the Magneto ports.
However, if you want to change any of the feature settings in for the Magneto Ports, you may prepare a
different Template154, for example, Template 05 and apply it on the Magneto Ports. To do this,
• Now go back to the Magneto Parameters page.

• Enter the number of the Template you customized, Template 05 in the 'Station Advanced Feature
Template' field of the Magneto Port, for example, MAG-001, on which you want to apply this
template. If you want to apply this template to other terminals too, like MAG-002, 003, and 004,
assign the Template 05 to all these ports.
• Repeat the same steps to customize and assign a different Template to another Magneto Port.
templates and applying them on the station ports.
• Priority: Assign a Priority Level from 1 to 9 to the Magneto Port, with '1' being lowest Priority and '9'
being highest Priority. Whenever a Magneto Port with higher priority calls an extension with lower
priority, a triple ring is placed on the called extension. To know more, read the feature description
“Priority”.
By default, the Priority of all Magneto Ports is set to '5-Normal'.
• If you have completed configuration of the desired Magneto Ports, click Submit at the bottom of the
page to save your settings.
• Now, configure the Magneto Ring Enable (MRE) Key for DKP extensions.
Configuring MRE Key for DKPs

The Magneto Ring Enable (MRE) Key is to be configured on DKP extensions that will allow calls to be made to the
Magneto Ports.
Any DSS key may be configured as MRE Key. There are two ways to do this:
• Assign the MRE Key function in the DKP Key Template assigned to the DKPs, by customizing a template
and assigning it to the DKPs.
OR
• Assign the MRE Key function individually in each DKP, by selecting 'Personalized' Key map for each DKP.
For detailed information and instructions, see “DSS Keys Programming”.
Configuring Silence Detection on Magneto

If you want calls from Magneto ports to be disconnected automatically, if silence is detected, you must enable the
Silence Detection on Magneto flag and program the Magneto Silence Disconnection Timer, if required.

• Click the Magneto sub-link to expand.
• Enable Silence Detection on Magneto: By default this flag is enabled. If not, select the check box to
enable this flag.
• Magneto Silence Disconnection Timer: Set the timer to the desired value. The range of this timer is
from 001 to 255 seconds. By default it is set to 30 seconds.
• Magneto VAD Threshold Level: Set the level to the desired value. The range of this level of the 0 to
-96 dBm. By default it is -25 dBm.
• Click Submit at the bottom of the page to save your settings.
• Log out of Jeeves or continue with configuration tasks, as desired.
Configuring Magneto Parameters using a Telephone

To assign Slot-Port Assignment to a Magneto Port, dial:

• 1110-Magneto Trunk-Slot-Port Offset on the Card
Where,
Slot is Slot number in which the Magneto Card is inserted from 01 to 16
Port Offset is the number of the Magneto Port on the card from 00, 01 to 08.
Default: Hardware Slot and Port Offset is automatically assigned.
To enable/disable Magneto Port, dial:

• 6801-1-Magneto-Code to enable/disable a single Magneto port.
• 6801-2-Magneto-Magneto-Code to enable/disable a range of Magneto ports.
• 6801-*-Code to enable/disable all Magneto ports.
Where,
Code is

1 for Enable
2 for Disable
Default: Enable.
To program Access Code for Magneto Port, dial:

• 3107-1-Magneto-Access Code-#*
Where,
Access Code is a number string of any combination of up to 6 characters, including the digits from to 9,
and the characters #, *, A, B, C, D, F, P, and Blank space.
Terminate the command with #* if the number string has fewer than 6 digits.
Default: Blank.
To clear the Access Code assigned to the Magneto port, dial:

• 3107-1-Magneto-#* to clear Access Code assigned to a single Magneto port.
• 3107-2-Magneto-Magneto-#* to clear Access Code assigned to a range of Magneto ports.
• 3107-*-#* to clear the Access Codes assigned to all Magneto ports.
Where,
Magneto is from 001 to 128.
To assign default Access Codes to Magneto Port, dial:

• 3156-1-Magneto to default Access Codes of a single Magneto port.
• 3156-2-Magneto-Magneto to default Access Codes of a range of Magneto ports.
• 3156-* to default Access Codes of all Magneto ports.
To assign a Name to the Magneto Port, dial:

• 5411-1-Magneto-Name-#* to assign Name to a single Magneto port.
• 5411-2-Magneto-Magneto-Name-#* to assign the same to a range of Magneto ports.
• 5411-*-Name-#* to assign the same name to all Magneto ports.
Where,
Name is an alphanumeric string of Maximum 18 characters. Terminate the command with #* if fewer
than 18 characters are to be used as name.
To clear the Name of the Magneto Port, dial:

• 5411-1-Magneto-#* to clear the name of a single Magneto port.
• 5411-2-Magneto-Magneto-#* to clear the name of a range of Magneto ports.
• 5411-*-#* to clear the names of all Magneto ports
To assign Priority level of a Magneto Port, dial:

• 3919-1-Magneto-Priority to assign Priority Level to a single Magneto port.
• 3919-2-Magneto-Magneto-Priority to assign the same Priority Level to a range of Magneto ports.
• 3919-*-Priority to assign the same Priority Level to all Magneto ports.
Where,
Priority is from 1 to 9.
Default: 5-Normal.
To assign a Station Basic Feature Template to Magneto Port, dial:

• 5511-1-Magneto-Station Basic Feature Template Number to assign a template to a single Magneto
port.
• 5511-2-Magneto-Magneto- Station Basic Feature Template Number to assign the same template to
a range of Magneto ports.

• 5511-*- Station Basic Feature Template Number to assign the same template to all Magneto ports.
Where,
Default: Station Basic Feature Template Number 01.
To assign a Station Advanced Feature Template for Magneto Port, dial:

• 5611-1-Magneto-Station Advanced Feature Template Number to assign a template to a single
Magneto port.
• 5611-2-Magneto-Magneto- Station Advanced Feature Template Number to assign the same
template to a range of Magneto ports.
• 5611-*- Station Advanced Feature Template Number to assign the same template to all Magneto
ports.
Where,
Station Advanced Feature Template is from 01 to 50.
Default: Station Advanced Feature Template Number 01.
To enable or disable the "Enable Silence Detection on Magneto?” flag, dial:

• 5357-Flag
Where,
Flag is
1 for Enable
0 for Disable
By default, the flag is enabled.
To set Magneto-Silence Detection Timer, dial:

• 5356-Silence Detection Timer
Where,
Silence detection timer value range is from 001 to 255 seconds.
By default, the Timer is set to 30 seconds.
To set the Magneto Threshold Level, dial:

• 5358 - Magneto VAD Threshold Level
Where,
The value of the Level is from 0 to -96.
By default, the value of the Level is -25 dBm
• Exit SE mode.

Configuring 'Operator'
Users understand the term 'Operator' as a person who handles multiple simultaneous calls and functions as the link
between callers and called parties.
For the system however, an 'Operator' is a Routing Group; a group of extensions to which calls made by extensions
by dialing '9' are to be landed. This also includes Auto Attendant calls on trunks during which the caller dials '9'.
Depending on the size of the Enterprise and the amount of call traffic to be managed, more than one Operator may
be employed. Also, it is not uncommon to have different Operator extensions according to the time of the day. For
instance, during working hours calls may be handled by the Receptionists or Front Desk Personnel, whereas during
non-working hours, calls may be handled by the Security Personnel.
To meet this requirement, SARVAM UCS offers configuration of up to 20 different Operators (Routing Group).
However, at a time, only one Operator can be assigned to the extensions and trunks.
Each 'Operator' is assigned a Routing Group for the Time Zones - Working Hours, Break Hours and Non-Working
Hours.
Each 'Operator' is assigned a Time Table, which defines the Working Hours, Break Hours and Non-working Hours
for a week. The system follows this Time Table to assign a Routing Group as 'Operator' according to the current
Time Zone.
Configuration of 'Operator' involves the following steps:
1. Configuring Routing Groups as 'Operator': A routing group may be made up of one or more than one
extensions, depending on user requirement. If the user requires only one extension as 'Operator', include
only one extension as member in the Routing Group for Operator. If the user requires five extensions as
'Operator', create a Routing Group of the five desired extensions to be used as 'Operator'.
If the user requires Time-Zone based 'Operator', then prepare a different routing group for each Time
Zone. If the user requires the same Operator for all Time Zones, use the same Routing Group number in
all Time Zones.
2. Configuring a Time Table for Operator: This is applicable only when Operator extensions are different
for different Time Zones.
Define the Time Table to be followed for the Operator selection. The Time Table may be the same as the
Time Tables assigned to trunks and extensions of the SARVAM UCS or may be configured to match with
the timings of the persons who work as Operators. For example if Operators in the Enterprise are working
in shifts, the Time Table can be configured to match their timings.
3. Assignment of 'Operator' to Extensions: SIP Users, UC Clients, SLT, DKP, ISDN Terminals and Virtual
Extensions of the SARVAM UCS can be assigned to an 'Operator' in their Station Basic Feature Template.
All extensions may be assigned to the same Operator, or different groups of extensions may be assigned
to different Operators, so that call management is more efficient.
Operator 1 is the default in the Station Basic Feature Templates. If you want to assign different extensions
to different Operators, you must program separate Station Basic Feature Templates with a different
Operator for each extension group.

4. Assignment of 'Operator' to Trunks: Trunks are also assigned an 'Operator', so that when a caller dials
'9' using Built-In Auto Attendant, the call is routed to the Routing Group defined as Operator for the trunk
for a particular Time Zone. For example, the during working hours, a caller on trunk 001 dials '9', the call
lands on 3001; when a caller on trunk 001 dials '9' during non-working hours the call lands on 3003 and
when the caller dials '9' during break hours the call lands on 3002.
Similarly, it is also possible to assign different Operators to different trunks.
Decide the number of Operators to be configured on the basis of the user's requirement.
Configuring Operator using Jeeves

• Under Configuration, click Routing Group to open the page.
• Select a Routing Group you want to program for Operator. By default, Routing Group 32 is assigned to
Operator. You may configure this group, or select another one.
• Select the type of extension to be included in the group as Member Type. The extension may be a DKP,
an SLT, an ISDN terminal, an OTBG, a SIP Extension, a Virtual Extension or a Voice Mail Auto Attendant.
• Enter the Port Number (software port number) to which the SLT/DKP/ISDN Terminal/SIP extension is
connected. If a Virtual Extension is selected as the Member Type, enter the Port number of the Landing
Destination here. You can program as many as 32 members in the Routing Group. If the user requires only
one extension as Operator, configure the first Member and disable all other 'members' of the routing group
by setting Member Type to None.
• Assign the desired Voice Mail Auto Attendant (VMAA) Menu, if you have selected the Voice Mail Auto
Attendant as the Port Type.
You may click the Voice Mail Auto Attendant (VMAA) Menu link to edit the parameters of the desired
VMAA Menu. For details, see “Voice Mail Auto-Attendant Menu”.

• Click Submit to save your changes.
• Repeat the same steps to configure another Routing Group.
If you have finished configuring Routing Groups for Operator, configure the Time Table for Operator.
• Under Configuration, click Time Table to open the page.
• By default Time Table 1 is assigned to all Operators. If this time table meets your requirement, retain it. If
not, select another Time Table. Customize it by defining the Working Hours, Break Hours and Non-
Working Hours for the week.
• If you want to assign different Time Tables to different Operators, repeat the above steps to prepare the
other Time Tables.

If you have completed configuring the Time Table,
• Under Configuration, click Operators to open the page.
• Select the Operator number you want to configure. By default Operator 1 is assigned to all extensions and
trunks.
• Select the number of the Time Table you prepared for the selected Operator.
• Enter the number of the Routing Group you prepared for the selected Operator for Working Hours, for
Break Hours and for Non-Working hours. If the same Routing Group is to be kept for all Time Zones, enter
the same number in fields of all three time zones.
• Repeat the above steps to configure another Operator.
Now, you may assign the 'Operator' groups you have configured to the SLT, DKP, ISDN Terminal, and SIP
extensions by configuring the number of the Operator (1-20) in the “Station Basic Feature Template”
applied on these extensions.
Similarly, you may assign the 'Operator' groups you have configured to the trunks in the “Trunk Feature
Template” applied on these Trunks.

Configuring Operator using a Telephone
Prepare the required Routing Groups and Time Table first. Refer the topics “Routing Group” and “Time Tables” for
SE commands.
To assign a Time Table to an Operator, dial:

• 1602-1-Operator-Time Table to assign a time table to a single Operator.
• 1602-2-Operator-Operator-Time Table to assign the same time table to a range of Operators.
• 1602-*-Time Table to assign the same time table to all Operators.
Where,
Operator is from 1 to 20.
Time Table is from 1 to 8. Enter the number of the Time Table you customized for Operator.
To define an Operator for Working Hours, dial:

• 1611-1-Operator-Routing Group
Where,
Routing Group is from 01 to 96. Enter the number of the Routing Group you prepared for Operator for
working hours. Default: Routing Group is 32.
To define an Operator for Break Hours:

Break Hours. Default: Routing Group is 32.
To define an Operator for Non-working Hours, dial:

Break Hours. Default: Routing Group is 32.
For SE commands to assign an operator to extensions and trunks, refer the topics “Customizing
Station Basic Feature Template using a Telephone” and “Customizing Trunk Feature Template using a
Telephone”.
To default an Operator, dial:

• 1601-1-Operator to restore default values of a single Operator.
• 1601-2-Operator-Operator to restore default values of a range of Operators.
• 1601-* to restore default values of all Operators.
Where,
On issuing this command, Timetable 1 is assigned to Operator and Routing Group 32 is assigned to
Operator for all Time Zones.
• Exit SE mode.

Extension Search
Using Extension Search, you can search for the extensions configured by you as well as find out the Software ports
that are not assigned Extension Numbers.
You can search for any extension by:

• H/w Slot Port
• Extension Number
• Extension Name
• Software Ports not assigned Extension Numbers
• The number of extensions that the system will search for depends on the total extensions programmed
in System Pre-requisites.
• If you are searching using H/W Slot-Port, the search is valid only for those extensions that are
connected to the port.
• If you are searching using Extension Name, make sure you enter the name as configured in the
system.
Configuring Extension Search using Jeeves

• Under Configuration, click Extension Search.
• Select the desired Extension Search option from the following:
• H/w Slot Port: If you select this option, enter the H/w Slot Port of the extension you want to search.
The system will perform the search for the active ports only.
• Extension Number: If you select this option, enter the Number of the extension you want to search.
• Extension Name: If you select this option, enter the Name of the extension you want to search.
• Software Ports not assigned Extension Numbers: If you select this option, select the type of
extension—SLT, DKP, Department Group, Magneto, SIP Extension, Virtual Extension, Extension Over
Q-SIG, Radio Port—for which software ports are not assigned any extension numbers.
• Click Search. The search result appears on the screen.

Configuring Trunks
The SARVAM UCS supports the following types of trunk ports:
• SIP Trunks
• ISDN T1E1 PRI Lines
• ISDN BRI Lines
• Mobile Trunks
• CO/ Two-Wire Trunks
• E&M Lines
Templates for Configuring Trunk Lines

SARVAM UCS offers the following Hardware and Software Feature Templates to make the configuration of Trunks
easy.
• SIP Hardware Template (for SIP Trunks and SIP Extensions)

• CO Hardware Template (for CO Trunk Lines).
• E&M Feature Template (for E&M Lines and T1E1PRI Lines that use E&M signaling)
• Trunk Feature Template (for All Trunk Types)
Using these templates, you can configure all Trunks that are to be assigned the same set of hardware and software
features at one go, instead of configuring each trunk individually.
SIP Hardware Template

The SIP Hardware Template contains a set voice quality related features such as Voice Codec selection, Tx and
Rx Gains, Echo Cancellation, Jitter Buffer and, Fax-over-IP options and related parameters
A SIP Hardware Template must be assigned to all SIP Trunks as well as SIP Extensions. Using the SIP Hardware
Template, you can configure SIP Trunks and Extensions the same set of features at one go.
SARVAM UCS offers 32 SIP Hardware Templates, which can be customized as per the requirement and applied
on SIP Trunks and Extensions.
SIP Hardware Template Parameters

Each of the SIP Hardware Template parameters is described below in brief.
• Vocoder Preference: Vocoders are the various voice codecs used to compress the data in RTP packets
for optimum use of bandwidth and for ensuring voice quality. You can set 7 Vocoder options in the order of
preference for a SIP trunk.
The Vocoders supported by the system of the SARVAM UCS in the order of preference, from 1st to 7th, by
default are: G.723, G.729AB, GSM FR, iLBC - 30 ms, iLBC - 20 ms, G. 711-Law, G. 711 A-Law and
G.722.
If you do not want to select any Vocoder, you can select the option 'None' in the Template. However, if all
Vocoder Preferences from 1 to 7 are set to 'None', incoming and outgoing calls will be blocked.
G.722 is not applicable for when RTP Mode is set as Transcoding in “Configuring VoIP Parameters”.

• G.723 Bit Rate (kbps): You can select the Bit Rate for G.723 codec as 5.3 kbps or 6.3 kbps. When G.723
is negotiated, the selected Bit Rate will be applied only when sending the RTP packets. When receiving
RTP packets from the remote end, both Bit Rates of G.723 will be accepted. The default G.723 Bit Rate is
6.3kpbs.
• Silence Suppression for Vocoders: This parameter suppresses the 'Silence' packets, allowing only the
Voice packets through. SARVAM UCS supports Silence Suppression for all Vocoders except GSM FR.
Default: Disabled.
Silence Suppression must be disabled if you have selected 'Pass Through' as the “Fax Type”.
• SIP Gain Settings Template: You can increase or decrease the level of Incoming Speech (Receive Gain)
and Outgoing Speech (Transmit Gain) on the SIP Trunks/Extensions by changing the Rx Gain and Tx
Gain to the desired level. Different levels can be set for each port type in the SIP Gain Settings Template.
By default, SIP Gain Template 1 is assigned to all the SIP Hardware Templates. If you want to assign a
different Template, you must customize the SIP Gain Settings Template first and then assign the number
of the SIP Gain Settings Template in this Template. To customize the SIP Gain Settings, see “Gain
Settings”.
• DTMF Type: This parameter will determine how the DTMF digits will be sent over the IP Network, when a
DTMF digit is pressed. The SARVAM UCS supports three DTMF options: i) RTP (RFC 2833), ii) SIP Info,
and iii) In-Band. You may select the appropriate option. By default RTP (RFC 2833) is selected.
• RFC2833 Payload Type: If you have selected RTP (RFC 2833) as the DTMF Type, you must configure
the value of RFC2833 Payload Type. The RTP packets will be tagged as DTMF as per the set value. The
value of RFC2833 Payload Type can be set from 96 to 127.
• Echo Cancellation: SARVAM UCS supports Echo Cancellation for SIP to CO trunk calls and SIP to
Digital Trunks (BRI, T1E1, Mobile, SIP) and Extensions (DKP, ISDN Terminals). If you want to apply Echo
Cancellation, you must enable configure the following parameters.
• Enable: This flag is to be enabled to apply Echo Cancellation on SIP to CO and SIP to Digital Trunks/
Extensions. By default Echo Cancellation is enabled.
• Tail Length (msec) for CO: Echo Cancellation Tail Length for SIP to CO trunks can be 32/64/128
milliseconds. By default, Echo Cancellation Tail Length for CO is set to 128 milliseconds.
• Tail Length (msec) for Stations and Digital Trunks: Echo Cancellation Tail Length for SIP to Digital
Trunks/Extensions can be 32/64/128 milliseconds. By default, Echo Cancellation Tail Length for Digital
Trunks/Extensions is set to 32 milliseconds.
• Jitter Buffer: The speed at which voice packets travel through a network depends on the condition of the
network. All voice packets may not come at the same speed. This variation in the delay in receiving
packets, known as Jitter, affects voice quality. Jitter Buffer helps overcome the delay in receiving voice
packets and improves voice quality. Jitter Buffer receives voice packets, stores them and sends it to the
DSP to process it at evenly spaced intervals, thus improving voice quality.
SARVAM UCS supports two types of Jitter Buffer: Static and Dynamic. Static Jitter Buffer's internal delay is
static, whereas, the Dynamic Jitter Buffer's internal delay adapts itself to the jitter in the network.
• Type: Select the type of Jitter Buffer - Static or Dynamic - you want to use. If you selected Static Jitter
Buffer, you may set the 'Minimum Delay'. The value configured in the Minimum Delay determines the
size of the Static Jitter buffer.

If you selected Dynamic Jitter Buffer, configure the 'Optimization Factor' and 'Minimum Delay'. The
minimum size of the Dynamic Jitter buffer depends on the 'Minimum Delay' you configured. The
Optimization Factor determines the rate of adaptation of the Dynamic Jitter Buffer to the jitter in the
network.
• Minimum Delay (msec): This parameter is to be configured for both Static and Dynamic Jitter Buffer.
The Minimum Delay determines the size of the Static Jitter Buffer and When Jitter Buffer type is
selected as Static, the Minimum Delay defines the size of the Static Jitter Buffer. The Static Jitter Buffer
will store each received voice packets for the time you set and then it will send it to DSP for voice
processing.
When Jitter Buffer type is Dynamic, the Minimum Delay specifies the minimum time for which the
Dynamic Jitter Buffer will store the received voice packet before sending it to the DSP for voice
processing. 'Minimum Delay' can be from 10 to 280 milliseconds. By default Minimum Delay is set to
150 milliseconds.
• Fax Type: This parameter allows you to select the protocol of Fax over IP (FoIP). You can send/receive
Fax from a Fax machine connected to the SLT port of the SARVAM UCS.
The SARVAM UCS supports the fax options: None, T.38 (UDPTL), T.38 (RTP), and Pass Through.
If you select None, the server will not detect the fax tone.
If you select any other option both, calls and fax tone will be detected.
• 'Pass Through' and 'T.38' will work only if the peer devices also support the same option.
• If you select 'Pass through' as Fax type, you must disable 'Silence Suppression'.
• If the fax sent using T.38 is rejected, SARVAM UCS will use Pass Through for sending the Fax.
• T.38 Fax Parameters: This parameter is relevant only if you have selected T.38 as the Fax Type for Fax
over IP. Receiving a good quality fax over SIP trunk depends on high 'Eye Quality Monitor' (EQM). The
higher the Eye Quality Monitor, the better the Fax quality. To improve the quality of T.38 fax reception, you
may configure the below parameters.
• Version: Configure this parameter as supported by the Remote Peer, which may be a Proxy Server or
a SIP Device. While sending a fax, the Version will be sent in the re-INVITE Message to the Remote
Peer. While receiving a fax SARVAM UCS will accept a Version equal to or less than the configured
Version.
A different Version can be configured for each SIP Trunk. This is useful when you have proxy SIP
Trunks registered with different service providers supporting different versions.
The valid range for the Version is from 0 to 3. By default Version when Fax Type is T.38 (UDPTL) is 0
and when Fax Type is T.38 (RTP) it is 1.
• Max Rate (Kbps): This parameter controls the Fax image transfer speed. As EQM is inversely
proportional to Fax Max Rate, if you receive poor quality fax, the Fax Max Rate should be reduced. The
default Max rate is 14.4 kbps.
• Image Redundancy Level: The Fax Image Level is redundancy level for output Image, which can be
None, 1 or 2.

Fax Image transfer speed is inversely proportional to this parameter. If this parameter is low then fax is
transferred faster. EQM is directly proportional to this parameter. If this parameter is high, good quality
fax can be achieved.
Increase the Fax Image Level, if the quality of fax does not improve with Fax Max Rate.
Level None means no redundancy. By default Image Redundancy level is set to 2.
• Data Redundancy Level: This is a redundancy level for T.30 control data. Fax Data Level can be set
as None, 1 or 2. Level None means no redundancy. The higher the Redundancy level, the slower
would be the fax transmission.
EQM is directly proportional to this parameter. The higher the Fax Data Redundancy Level, the better
the EQM. By default, Data Redundancy Level is set to None.
• Pass Through FAX Codec: When the Fax option is selected as Pass Through, you must configure the
Pass Through FAX Codec as supported by the Remote Peer. The Remote Peer may be a Proxy Server or
a SIP Device.
You may select the Codec as G.711 A-law or G.711 LawWhile sending a fax this Codec is sent in the
re-INVITE message to the Remote Peer, but while receiving a fax SARVAM UCS will accept the fax with
any Codec.
Customizing SIP Hardware Template

You can customize 32 SIP Hardware Templates using Jeeves or by dialing SE commands from a Telephone. By
default, SIP Hardware Template 01 is assigned to all SIP Trunks as well as SIP Extensions.
If the default SIP Hardware Template 01 fulfills the user requirements, retain Template 01. If you want to change
the values of certain SIP Hardware Parameters, but apply the same parameter values to all SIP Trunks and
Extensions, simply customize the desired parameters in Template 01.
If different hardware parameters are to be applied to different SIP Trunks and SIP Extensions, customize different
the SIP Hardware Templates using Jeeves or a Telephone and apply them to the SIP Trunks and SIP Extensions
as appropriate.
Customizing SIP Hardware Template using Jeeves


• Click SIP Hardware Template.
• Change the values of the SIP Hardware Template parameters as desired.
• Now, apply this Template 02 on the SIP Trunks and SIP Extensions.
To apply the customized template on SIP Trunks,
• Under VoIP Configuration, click SIP Trunk Parameters.
• Go to the SIP Trunks to which this Template is to be assigned, for example SIP Trunk 02, 03 and 04.
• Enter the number of the Template you customized, 02, in the field SIP Hardware Template of each of
these SIP Trunks.
• Click Submit to save your template.
To apply the customized template on SIP Extensions,
• Go to the SIP Extension software ports to which the Template is to be assigned, for example SIP
Extensions-005 to 008.
• Enter the number of the Template you customized, 02, in the field SIP Hardware Template of each of
these SIP Extensions.
• Repeat the same steps to customize another template and apply it on the SIP Trunks and Extensions.

CO Hardware Template
The CO Hardware Template contains a set of features, such as AC Termination Impedance, Pulse-Tone Dialing,
Answer Supervision, Disconnect Supervision, DTMF detection, etc. that define the behavior of the hardware of the
CO ports of the SARVAM UCS.
A CO Hardware template must be assigned to all the CO trunk ports. Using the CO templates, you can configure
CO ports which are to be assigned the same set of features at one go, instead of configuring port-by-port.
The SARVAM UCS offers 50 CO Hardware Templates. These templates have commonly used values, but can be
customized per the requirement and applied on the extensions.
SARVAM UCS supports only 'Loop Start' on CO Interface.
CO Hardware Template Parameters

The following is an overview of the parameters of these templates and a brief description of each.
• Trunk Type: Three types of Trunks can be interfaced to a CO port of the SARVAM UCS:
• Normal Dial type: This is the conventional CO trunk available from the PSTN.
• Hotline type: The CO trunk connecting two destinations immediately on grabbing the trunk.
• Delayed Hotline: A special CO trunk available from the PSTN, which works as a normal dial type for
some time and works as a hotline thereafter.
By default all the CO ports are set as Normal Dial type. You may select the CO Dial Type you want to
assign to the CO port.
• Dial Type: You can select the Dialing method as Pulse or Tone (with configurable Pulse Ratio and DTMF
ON-OFF period) according to the Dialing method supported by the CO network to which the CO port is
connected.
By default, Tone is selected as the Dial Type.
• Pulse Dial Ratio: This parameter is to be configured if you have selected Pulse as the Dial Type in the
previous parameter. The system supports the six different Pulse Dialing Ratios on CO ports. Select the
appropriate Pulse Dial Ratio from the following according to the type of Pulse Dialing Ratio supported by
your CO Network:
• 10PPS, 1:2
• 10PPS, 2:3
• 10PPS, 1:1
• 20PPS, 1:2
• 20PPS, 2:3
• 20PPS, 1:1
By default, 10PPS, 1:2 is selected as the Pulse Dial Ratio.
• Rx CLI Type: SARVAM UCS detects the CLI sent by the CO network and sends this information to the
landing extension/operator with the ringing signal. You must select the CLI Type supported by your CO
network from the following options:
• Any ETSI DTMF format

• Any FSK V.23 format
• Any FSK Bellcore format
• 1st Ring, ETSI DTMF, 2nd Ring
• Polarity Reversal, ETSI DTMF, 1st Ring
• 1st Ring, FSK, 2nd Ring
• DT-AS, FSK, 1st Ring
• RP-AS, FSK. 1st Ring
• Polarity Reversal DT-AS, FSK, 1st Ring
• Any DTMF Format (without Start/Stop Code)
By default, Any ETSI DTMF format is selected as the Rx CLI Type.
• AC Impedance: The AC Termination Impedance of the CO port must match with the AC Termination
Impedance supported by the PSTN network. The system supports the following AC Termination
Impedance:
• 600
• 900
• 270 + (750 || 150 nF)
• 220 + (820 || 115 nF)
• 370+ (620 || 310 nF)
• 320 + (1050 || 230 nF)
• 370 + (820 || 110 nF)
• 275 + (780 || 115 nF)
• 120 + (820 || 110 nF)
• 350 + (1000 || 210 nF)
• 200 + (680 || 100 nF)
• 600 + 2.16 µF
• 900 + 1 µF
• 900 + 2.16 µF
• 600 + 1 µF
• Global complex impedance
By default, the AC Termination Impedance is set as per the “Region” you have selected.
• CO Termination: This parameter allows you to increase near-end echo cancellation on the CO trunk.
Near-end echo is primarily caused by the mismatch between AC Termination Impedance (presented by
CO port of SARVAM UCS to the line) and CO Termination (Impedance presented by the Central Office to
the line), and to some extent by the transmit and receive signal path.
By correcting the line impedance mismatch, you can increase near-end echo cancellation. This is done by
selecting the AC Termination Impedance and CO Termination, and selecting a Line Type that most closely
models the line that connects the CO port of SARVAM UCS to the Central Office.
In the CO Termination list, select the appropriate line impedance match. This would depend on the region
where SARVAM UCS is deployed. For example, if AC Termination Impedance in your location is 600and
the CO Termination impedance is 900 in series with 2.16µF, select AC Impedance as 600and CO
Termination as 900 + 2.16µF. By default, None is selected.
Now, select the line model to be used from the CO Line Type list.

You are recommended to conduct the AC Impedance Test for the line connected to the CO Trunk port on
which you will apply this template. The AC Impedance Test will help you determine the most appropriate
values for the AC impedance, CO Termination and the CO Line Type. For more information see the topic
“AC Impedance Test”
• CO Line Type: This parameter allows you to select the Line model for the CO Termination you have
selected. You need to select a line type that most closely models the line connecting SARVAM UCS to the
Central Office. In the CO Line Type list, you may select a specific EIA line model from the eight options
(EIA-0 to EIA-7) or a specific wire gauge and length (2000 ft. 22/24/26awg). By default, None is selected.
• CO Gain Settings Template: You can increase or decrease the level of Incoming Speech (Receive Gain)
and Outgoing Speech (Transmit Gain) on the Trunk by changing the Rx Gain and Tx Gain to the desired
level. Different levels can be set for each port type in the CO Gain Settings Template. By default, CO Gain
Template 1 is assigned to all the CO Hardware Templates. If you want to assign a different Template, you
must customize the CO Gain Settings Template first and then assign the number of the CO Gain Settings
Template in this Template. To customize the CO Gain Settings, see “Gain Settings”.
• Answer Supervision: It is a signal from the CO network to indicate to call maturity. Whenever you make
an outgoing call from CO trunk, the CO network will give answer signaling when the called party answers
the call.
This feature is particularly useful if you want to use “Call Cost Calculation (CCC)” to enable accurate
billing. When the signal is received, the billing will start and in the absence of this signal, the call will not be
billed, ensuring that unanswered and unsuccessful call attempts are not billed.
Answer Supervision Signal has three options:
• Pseudo Answer: It is used when no signaling is available from the PSTN. If this option is selected, the
call will be considered as matured on the expiry of the 'Pseudo Answer Supervision Timer'
(configurable; default 10 seconds), irrespective of whether or not the call actually gets matured. After
this, the Call Duration Timer starts. Finally, the system starts detecting the “Disconnect Supervision”
signal configured for the CO port.
Select this option only if there is no Answer Supervision Signal supported.
• Polarity Reversal: It is used as maturity signal when the answer signaling is given in the form of
Battery Reversal. If the battery polarity of the line is -ve for TIP and +ve for RING, when the called party
has answered the call, the CO network will reverse the battery polarity, TIP becomes +ve and Ring -ve.
After this, the Call Duration Timer is started. Finally system starts detecting the Disconnect Supervision
signal configured for the CO port.
By default, Answer Supervision is set as Pseudo Answer for each CO port.
Select the same Answer Supervision signal as provided by your CO Network. If the type of Answer
Supervision signal selected in the system does not match with that of the CO network, the call will not be
stored in the Station Message Detail Record (SMDR) buffer. For example, if the CO network does not
support Answer Supervision, but you have set Polarity Reversal as Answer Supervision Type, the call will
be considered as matured and will not be stored in the Station Message Detail Record (SMDR) buffer.
• Pseudo Answer Supervision Timer: Configure this timer if you have selected 'Pseudo Answer' as
Answer Supervision Signal option.

This is the time period for which the system will wait before treating a call as matured (regardless of
whether or not it was answered). The range of this Timer is from 001 to 255 seconds. By default the
Pseudo Answer Supervision Timer is set to 10 seconds.
When Pseudo Answer is selected as Answer Supervision signal, the call duration measured by the system
will not accurately reflect the actual call duration because the Pseudo Answer Supervision Timer is not
related to the actual call maturity. For example, if the Pseudo Answer Supervision Timer is set to 015
seconds, the call will be considered as matured after 015 seconds, even if it is answered after 20 seconds.
Similarly, if this Timer is set to 080 seconds, but the call was answered after 020 seconds and
disconnected after 040 seconds, this call will never be considered as matured as it ends before 080
seconds.
• Disconnect Supervision: It is a signal from the CO network to indicate call disconnection. Whenever a
call (incoming or outgoing) made from the CO trunk is disconnected by the remote party, the CO network
will send Disconnect signal to the CO port. SARVAM UCS will detect this signal and release the CO port.
Disconnect Supervision signal is important when a PCO machine is connected to the (SLT Port) SARVAM
UCS and or when SARVAM UCS is deployed in a Gateway application.
In such application scenarios, it is desirable that calls that are disconnected by either end - calling party or
called party - is terminated by the system and the port is released. If the called (remote) party has
disconnected the call but the calling party (extension that made the outgoing call from SARVAM UCS) has
not disconnected the call, the call remains live within the system.
So, Disconnect Supervision signal is important, particularly when calls are routed from CO-to-CO ports, to
indicate to the system that it needs to disconnect the call and release the port.
Disconnect Supervision signal has three options:
• None: When there this no signaling supported. Select this option only if there is no Disconnect
Supervision signal supported.
• Polarity Reversal: Call disconnection is signaled as Polarity Reversal when the call is disconnected by
the remote user. For example, if the battery polarity of the CO port is '+ve' for TIP and '-ve' for RING in
speech condition then on disconnection by the remote user, TIP will become '-ve' and Ring '+ve'. The
user gets an Error tone and the CO port is released.
• Open Loop Disconnect: Call Disconnection is signaled in the form of Open Loop, whereby the Battery
voltage on the CO port is removed for a short duration. Voltage is restored after this short duration.
However, the Polarity of Battery Voltage on the CO port is not changed.
This option is to be selected when call disconnection is signaled in the form of Open Loop Disconnect
pulse by the CO network. System will check Open Loop Disconnect signal for the time configured for
Open Loop Disconnect Timer for each CO port. If the time of the Open Loop signal detected is less
than the Open Loop Disconnect Timer configured, it will not be considered as valid Open Loop signal
for releasing the CO port. But if the Open loop is detected continuously for at least for the time set as
the Open Loop Disconnect timer, it is considered as a valid Disconnect Supervision signal. The call will
be released and caller will get error tone.
By default, Disconnect Supervision is set to None for each CO port.
Select the same Disconnect Supervision signal as provided by your CO Network.

• Select the same Answer Supervision and Disconnect Supervision signal type supported by your CO
network for the CO ports. Consider the following case:
• The CO network supports Polarity Reversal signal as Answer and Disconnect Supervision.
• But you have configured 'Pseudo Answer' as Answer Supervision signal and 'Polarity Reversal' as
Disconnect Supervision signal for the CO ports in the system.
• In this case, when a call is made through the CO port, the call will be considered as matured after
the Pseudo Answer Supervision Timer.
• Now, when the called party answers the call, the CO generates 'Polarity Reversal' as answer
supervision signal on the CO port.
• But as 'Polarity Reversal' is also configured as the Disconnect Supervision for the port, the system
will interpret this (Answer Signaling) signal as Disconnect Supervision signal and disconnect the
call.
• Open Loop Disconnect Timer (msec): This parameter is applicable only if the option Open Loop
Disconnect is selected as Disconnect Supervision on the CO port.
The range of this timer is from 017 to 986 milliseconds. By default, the Timer is set to 204 msec.
• Disconnect Tone Detection: This parameter is to be configured if Call Disconnection is signaled by the
CO network in the form of Disconnect Tone.
When there is an incoming/outgoing call on/from the CO port is answered, the system will check whether
the flag “Disconnect Tone detection is enabled. Only if the flag is enabled, the system will detect the
Disconnect Tone.
If Disconnect Tone is detected, the system will consider the call as ended and will release the CO port.
• Disconnect Tone Cadence: To enable the system to detect the Disconnect Tone accurately, you must
set the Cadence (ON-OFF time) and Frequency of the Disconnect Tone, as supported by the CO network.
Configure the following Disconnect Tone Cadence parameters:
• Frequency 1 (Hz): Frequency 1 is from 300 to 1400 Hz. Default: 400Hz
• Operator: This parameter has 3 options: No operator, Modulation (*), Addition (+). Default: No.
If ‘No’ operator is selected frequency 2 will not be applicable.
If Modulation is selected, frequency 1and frequency 2 will be used as modulation, F1* F2
If Addition is selected, frequency 1 and frequency 2 will be used as addition, F1 + F2.
• Frequency 2 (Hz): Frequency 2 is from 20 to 1400 Hz. Select Frequency 2 if the Disconnect Tone
supported by the CO network is Dual Frequency. Default: 25Hz.
• ON Time 1 (ms), OFF Time 1 (ms): Select Cadence for the first cycle ON Time1 and OFF Time 1. It
may be 0 to 9999 milliseconds. Default: 750 ms ON Time 1, 750 ms OFF Time 1
• ON Time 2(ms), OFF Time 2 (ms): Select Cadence for the second cycle ON Time 2 and OFF Time 2.
It may be 0 to 9999 milliseconds. Default: 750 ms ON Time 2, 750 ms OFF Time 2.

• ON Time 3(ms), OFF Time 3 (ms): Select Cadence for the third cycle ON Time 3 and OFF Time 3. It
may be 0 to 9999 milliseconds. Default: 0 ms ON Time 3, 0 ms OFF Time 3.
• ON Time 4(ms), OFF Time 4 (ms): Select Cadence for the fourth cycle ON Time 4 and OFF Time 4. It
may be 0 to 9999 milliseconds. Default: 0 ms ON Time 4, 0 ms OFF Time 4.
When disconnect tone detected on the port matches the Frequency and Cadences you have set, the
call will be disconnected and the CO port will be released.
When Disconnect cadence is zero, SARVAM UCS will skip that cadence and match the next cadence.
SARVAM UCS will match the cadence for 3 cycles and then release the trunk.
• Speech Delay Timer: It is the time after which the system gives dial tone to the extension, when the
extension user grabs the CO.
To understand the significance of this timer, let us consider a situation. Extension 2001 does not have
calling permission for long distance numbers. The user of extension 2001 grabs a CO trunk, and dials a
number 1022-6305555. The system dials out this number, as it starts with '1', but since the actual dial tone
from the CO comes after some time, the CO interprets this number as 022-6305555 and establishes
speech. This way an extension user who does not have permission for long distance calling, can dial out a
long distance number. This situation can be prevented by setting the Speech Delay Timer to an
appropriate value.
The range of this timer is from 000 to 255 seconds. By default it is set to '0' second.
• Pause Timer: This Timer is required for inserting delay while digits of a number string are out dialed from
the CO trunk. The Pause Timer is applied when the features “Closed User Group (CUG)”, “Multi-Stage
Dialing”,“Emergency Dialing”, “Last Number Redial”, “Auto Redial”, “Abbreviated Dialing”, “Call Back on
Trunk Ports”, “Quick Dial”, “RCOC (Return Call to Original Caller)”, Least Cost Routing (“Configuring LCR”)
are used to dial out the numbers from the CO port. The range of this timer is from 0500 to 2500
milliseconds. By default the timer is set to 1000 milliseconds.
• Ring Cadence OFF Timer: Configure this timer to set OFF time for Ring cadence. During the incoming
call on CO port, if the CO gives ring in which the Ring OFF period is quite long, the system will consider
that the ring has been stopped, and will stop ringing the SLT port, even though the incoming call is still
present.
To get accurate indication, the system supports Ring Cadence OFF timer on CO port so that ring can
continue even for incoming calls with long Ring OFF period.
The range of the Ring Cadence OFF timer is from 1 to 9 seconds. By default the timer is set to 6 seconds.
• DTMF Out Dial: While dialing out the DTMF digits from the CO port, the following attributes of DTMF
signal are critical.
• DTMF Signal ON Time (ms): It is the width of DTMF digit to be dialed out by the CO port and is
configurable. By default the ON Time is set to 102 milliseconds.
• DTMF Inter-Digit Pause Timer (ms): When the CO port dials out the DTMF digits on the CO, it waits
for the Inter Digit Pause Timer, while dialing the DTMF digits on CO trunk. This timer is configurable. By
default the timer is set to 102 milliseconds.

The 'level' of each DTMF digit is fixed, at -6.0 dB, but you may configure these parameters to match the
CO network requirement.
These DTMF Out Dial attributes are applied when the features Redial, Auto Redial and Abbreviated
Dialing are used to dial out the numbers from the CO port. These attributes are also applicable when
you make a call from a DKP that has DTMF generation disabled.
• DTMF Detection: The default settings of DTMF Detection serve the requirements of most of the
applications. However, you may fine tune the following parameters if you face any problems in DTMF
detection.
• Minimum Level (dB): This parameter signifies the minimum level (dB) of the DTMF digit to be
considered as valid. By default, Minimum levels set to -4.5dB.
• Minimum ON Time (msec): This parameter signifies the minimum time period for which the DTMF
signal should be present in order to be detected. The valid range of this time is 17 to 204 milliseconds.
By default, Minimum ON Time is set to 34 milliseconds.
• Minimum OFF Time (msec): This parameter signifies the minimum time period between successive
DTMF digits. The valid range of this time is 17 to 204 milliseconds. By default, Minimum OFF Time is
set to 68.
• Flash Timer (msec): This parameter is relevant for dialing out Flash on the CO trunk to access some of
the features of the PSTN. Configure the desired time of Flash to be generated on the CO trunk.
The range of the timer is from 83 to 900 msecs. By default the Flash Timer is set to 600 msecs.
• ON Hook Speed (msec): This parameter allows you to set the amount of time for the line-side device to
go on-hook.
The ON-Hook speed specified is measured from the time the ON-Hook bit is cleared until loop current
equals zero. Select the desired ON-Hook Speed from the following options:
• <0.5ms
• 3 ms
• 26 ms
By default, <0.5ms is selected as ON-Hook Speed.
• OFF Hook Speed (ms): This parameter defines the time to settle the line transients after which
transmission or reception can occur. Select the desired OFF-Hook Speed from the following options:
• 512 ms
• 128 ms
• 64 ms
• 8 ms
By default, OFF-Hook Speed is set to 8 milliseconds.
• Current Limiting: With this flag you can enable Loop Current Limiting mode. When this flag is enabled,
the Loop Current will be limited to a maximum of 60mA.
By default, the flag is disabled.
• Minimum Loop Current (mA): This parameter sets the minimum loop current at which DAA module of the
CO port can operate. Select the minimum operational loop current from the following options as per your
requirement:
• 10

• 12
• 14
• 16
The minimum Operational Loop Current set by default is set to '10 mA'.
• Tip Ring Voltage (Volts): This parameter allows you TIP/Ring Voltage Adjustment on the line side.
Countries where Low voltage is required should use lower TIP/RING voltage. Adjust the values of the Tip
Ring Voltage to match your country requirements from the following options:
• 3.1
• 3.2
• 3.35
• 3.5
The default Tip/Ring voltage is 3.5.
• Ringer Impedance: Set the Ringer Impedance - High or Synthesized - for the CO port according your
country-specific requirement.
'High' signifies 20Mohm Ringer Impedance. This is the default Ringer Impedance provided on the line side
by the DAA module of the CO port. The DAA Module can provide higher impedance when 'Synthesized'
impedance is selected.
Some countries like Poland, South Africa and Slovenia require higher ring impedance which is achieved by
the DAA module, when Ringer Impedance is set to 'Synthesized' impedance.
By default 'High' (20Mohm) is selected.
• Ringer Threshold (Vrms): This parameter defines the level below which the CO port would not validate
the Ring signal and the level above which it validate the Ring signal. Set Ringer Threshold to the desired
value from the following options:
• 13.5 - 16.5
• 19.35 - 23.65
• 40.5 - 49.5
By default 13.5 - 16.5 Vrms is set as Ringer Threshold.
• PPDC: 'Pre-PSTN Digit Count' or PPDC is parameter is to be configured, only if the CO Trunk ports on
which the template is applied are in a “Behind the System Application”.
PPDC is the number of digits (dialed by an extension) to be ignored by the system before toll control check
is begun. It is the same as the number.
In Behind the System Applications, another System may be connected to the SARVAM UCS, with some of
its CO Trunks terminating into the Stations of the other System and other trunks directly connected to the
PSTN.
PPDC for CO Trunk ports directly connected to the PSTN must be set to '0'.
For Trunk ports connected to stations of another System, PPDC must be configured as per the number of
digits in the Trunk Access Code defined for that System.

If the TAC is a single digit, select '1'. If TAC is double or triple digit number, accordingly select '2' or '3' as
the PPDC.
To know more about this feature, refer “Behind the System Application”.
• Gateway Application - Answer Signaling: This parameter is to be configured if the CO Port is being
used in a gateway application as a source port (from where calls originate).
The calls originating on the source port (CO port) are routed using another Trunk port, the terminating port,
which may be any trunk port, for example: T1E1. When call made from the terminating port gets matured,
this is signaled to the source port in the form of DTMF digits.
.
• Use: Enable this flag if you want the CO port to be used in a Gateway Application.
• DTMF String (max. 4 digits): Configure the DTMF digits to be sent to signal call maturity to the source
port.
• Category (Logical Partitioning): This parameter assigns the CO Port to a trunk category for the purpose
of Logical Partitioning. By default all CO Ports are assigned to Category 1155.
If you have re-defined Category 1 or have assigned CO ports to a different category, say Category 3, enter
the same number here.
You may configure the call permission between the Category assigned to CO Ports and other Categories
(assigned to other Trunk ports). Refer the feature description “Logical Partition” to know more.
• Rx Gain at SIP Trunk (Pass Through Fax): This parameter allows you to improve the quality of Fax over
IP156. Configure this parameter if you have selected Pass Through Fax as the Type of Fax over IP on SIP
Trunks, and if Pass Through Fax is to be received on a CO Trunk.
• Data Gain (dB): select the dB Level for Data Gain. By default, Data Gain is set to -11 dB.
• Bypass Gain (dB): select the dB Level for Bypass Gain. By default, Bypass gain is set to -9 dB.
• Idle Wait Timer: This is the time taken by the Central Office to detect and release the line after the CO
trunk has been released by SARVAM UCS. This time may vary from Central Office to Central Office. Set
the Idle Wait Timer as per the time taken by your Central Office.
Set this timer accurately. If the set time is more than the actual time taken by the CO to release the line, it
will result in delay to the caller. If the set time is less than the actual time taken by the Central Office, no
dial tone will be played to the caller.
Valid range 001 to 255. By default, it is set as 002 seconds.
Customizing the CO Hardware Templates

The CO ports provide the interface to connect SARVAM UCS to the POTS network. The standards and features
supported by POTS networks across the world vary. For example, some networks support Caller ID Presentation
155. Trunk ports interfaced with PSTN /PLMN (Public Land Mobile Network) are assigned this category.
156. Normally, fax calls require less gain compared to voice calls. However, to improve fax reception, SARVAM UCS allows the config-
uring of gain settings for fax. Fax gain settings consist of Data Gain and Bypass Gain. SARVAM UCS supports Fax Receive Gain
for SIP to CO Trunks, SIP to Digital Trunk calls as well as SIP to SLT Calls.

using DTMF signaling, while some support Caller ID Presentation using FSK signaling; some networks offer 600
Ohms Impedance, while others offer complex impedance.
SARVAM UCS's versatile architecture allows it to be connected to such networks differing in their characteristics.
You can configure the CO hardware features to match the standards supported by the POTS network of the
country where the system is installed.
The 50 CO Hardware Templates offered by SARVAM UCS contain the default values of the above-listed
parameters. The default parameter values of these are country specific and are loaded in each template according
to the Country selected as the“Region”.
For example, when India is selected as the Region, the default value of the AC Impedance is 600, whereas it is
900 ohms when Philippines is selected as Region, and 320 + (1050 || 230 nF) for Region UK. Similarly, the
default Rx CLI Type for Region India is 'Any ETSI DTMF format' while the same is 'Any FSK V.23 format' for Region
UK.
By default, CO Hardware Template number 01 is assigned to all CO Trunk ports.
While the default CO Hardware Template number 01 has all with commonly used values to match your country-
specific requirements, you can still customize each of the 50 Templates to match your preference or requirement.
If the CO Hardware Template number 01 fulfills your requirements, and if the same features at to be applied on all
CO trunk ports, retain Template 01. Similarly, if you want only a few changes to be made to Template 01 and apply
it on all CO Ports, make the changes and retain the template.
However, if different sets of features are to be allowed to different CO hardware ports, then prepare separate CO
Hardware Templates and apply them on the ports as required.
You can use Jeeves or a Telephone to customize the CO Hardware Template.
Customizing CO Hardware Templates using Jeeves

• Under Configuration, click CO Configuration.

• Click CO Hardware Template to open the page.
• Select a CO Hardware Template Number you wish to customize. For example, Template 07.
• Customize the CO Hardware Template number 07 by setting the parameters to the desired values.
• Now, apply this CO Hardware Template 07 on the CO ports.
• To do this, click the link CO Parameters to open the page.
• Go to the CO software ports to which this Template you customized (07) is to be assigned, for example
CO-007 and 008.

• Enter the number of the Template you customized, 07, in the field CO Hardware Template of each of
these CO ports.
• Repeat the same steps to customize another template and apply it to the CO Port.
Customizing CO Hardware Templates using a Telephone

To change the default value of a CO Hardware Parameter in a Template, dial:

• 5902-1-CO Hardware Template Number-Feature Number-Code to change values of a parameter in
a single template
• 5902-2- CO Hardware Template Number-CO Hardware Template Number-Feature Number-Code
to set the same value for the parameter in a range of templates.
• 5902-*-Feature Number-Code to set the same value for the parameter in all templates.
Template Number is the number of the CO Hardware Template from 01 to 50.
Feature Number is the number of the CO Hardware Template Parameter from 01 to 50.
Code is the value for each parameter from 0 to 58. Refer the following table.

Default Values of CO Hardware Templates (Region Code: India)

Table 1:
For example, to change the Rx CLI Type in Template 07 from 'Any ETSI DTMF format' to 'Any FSK V.23
format', dial 5902-1-07-01-3
Where,
05 is the parameter number for Rx CLI Type
2 is the code for Any FSK V.23 format.
To default CO Hardware Template, dial:

• 5901-1- CO Hardware Template Number to default a single template.
• 5901-2- CO Hardware Template Number - CO Hardware Template Number to default a range of
templates.
To assign a CO Hardware Template to a CO port, dial:

• 5903-1-CO-CO Hardware Template Number to assign a hardware template to a single CO port.
• 5903-2-CO-CO-CO Hardware Template Number to assign a hardware template to a range of CO
ports.
• 5903-*-CO Hardware Template Number to assign a hardware template to all CO ports.
Where,
CO is the Software Port number of the CO port from 001 to 128.
Template Number is the number of the customized CO Hardware Template, from 01 to 50. Default:
Template 01.
• Exit SE mode.

E&M Feature Template
The E&M Feature Template is a set of features specific to E&M signaling, which define the behaviour of the E&M
ports, according to their 'Orientation Type', whether they are functioning as Stations, Trunks or Tie-Lines.
The E&M Feature Template is applied also on T1E1PRI trunks which use E&M signaling.The SARVAM UCS offers
50 such Templates. The E&M templates are loaded with default values that serve the requirements of a broad user
base, but can be customized as per user requirements.
E&M Feature Template Parameters

The E&M Feature Template has the following parameters:
• Seizure Type: E&M Trunk Seizure Type is the line protocol that defines how the equipment seizes the
E&M Trunk. It is also referred to as Start Dial Supervision Signaling Protocol. SARVAM UCS supports the
following Seizure Types:
• Immediate: The method of seizing the E&M Line using Immediate Start for Outgoing and Incoming
calls is illustrated below.
0V
Immediate Start
-48V
2001 3001
M M
E E
Rx Rx
System-A System-B
Tx Tx
SA SA
2002 3002
SB SB
Outgoing Call:
While making an outgoing call, when the extension user of System A seizes the E&M Port of System A,
the status of the "M" wire of its E&M port will go high, indicating that it has seized the E&M line.
There will not be any signaling over the "E" wire of System A's E&M Port during seizure.
Incoming Call:
While receiving an incoming call over its E&M port, System-B will be ready to receive digits as soon as
it detects high state on its "E" wire.
There will not be any signaling over the "M" wire of E&M Port of System B while receiving an incoming
call.
• Immediate with Ack: The method of seizing the E&M Line using Immediate with Acknowledgment for
Outgoing and Incoming calls is as follows:
Outgoing Call:
If this seizure type is selected, while an outgoing call is made by seizing the E&M Port, the 'M" wire will
go high immediately.
The remote end will acknowledge this by making its 'M' wire high, which in turn will activates (high) 'E'
wire of the E&M port of System-A.

On sensing high signal on 'E' wire, System-A will start dialing out the DTMF/Pulse digits.
Incoming Call:
On detecting high signal on "E" wire of the E&M port, the system will consider it to be an incoming call
seizure and hence it will immediately make its 'M" wire high, which will allow the remote end to dial out
the DTMF/Pulse digits.
Call Disconnection:
If the parameter 'Release Type' is configured as 'Status Change' and for this type of seizure, the "M"
wire at the remote end goes Low for some call condition, the call will be disconnected. For example,
"M" wire at the remote end will go 'Low' in the following conditions:
• Remote end user dials invalid number and does not hang up on getting Error Tone.
• Remote end user dials valid extension number and after conversation remote end hangs up first.
• Remote end user dials valid number and extension does not reply, but the remote party does not
• Remote end has made 'Orientation Type' of E&M port as 'Trunk' and for Incoming calls, all the
extensions in the trunk landing group are busy and also second call is not allowed to the extension
user, the system will disconnect the call after the expiry of the Ring Back Tone Timer, by making the
'M' wire Low.
• Immediate + Wink: The method of seizing the E&M Line using Immediate + Wink Start for Outgoing
and Incoming calls is described below.
0V 0V
Wink
-48V -48V Pulse
2001 3001
M M
E E
Rx Rx
System-A System-B
Tx Tx
SA SA
2002 3002
SB SB
Outgoing Call:
While making an outgoing call when the System A attempts a seizure (grab), the state of the "M" wire of
the E&M port of System A will go high.
To acknowledge this, the E&M port of System B will send Wink signal over its "M" wire, when System B
is ready to receive digits.
System A will wait for the duration of the 'Wait Wink Timer'. On receiving the acknowledgment in the
form of Wink signal on the "E" wire of its E&M port, before the expiry of the Wait Wink Timer, System A
will consider the trunk seizure as successful. Digits will be dialed out from E&M port.
If Wink is not received from System B within the ‘Wait Wink Timer’, System-A will drop the call.

Incoming Call:
While receiving an incoming call over the 'M' wire of its E&M port, System B will send the Wink signal to
the System A, which has initiated the seizure (grab).
The ‘Wink’ signal will be sent by the System B when it is ready to receive the digits from the System A.
You can change the 'wink' pulse width by configuring the 'Wink Pulse Timer'.
The width of the Wink Pulse ranges from 0000 to 9999 milliseconds.
• Immediate with Ack + Wink (MFC R2): The method of seizing the E&M Line using Immediate with
Ack + Wink (MFC R2) for outgoing and incoming calls is described below.
0V 0V
Wink
-48V -48V Pulse
2001 3001
M M
E E
Rx Rx
System-A System-B
Tx Tx
SA SA
2002 3002
SB SB
Outgoing Call:
When System-A attempts a seizure of (attempts to grab) the E&M Port, the 'M' Wire of the E&M Port of
System-A will go high and wait for the E&M Port of System B to turn its 'M' wire high.
System-B detects this on its 'E' wire. To acknowledge this, the E&M port of System-B will turn its 'M'
wire high and send a Wink signal over its 'M' Wire. Sending of the wink signal indicates the readiness of
System-B to receive the digits of the called party number.
System-A will wait for the duration of the 'Wait Wink Timer' to receive the acknowledgment in form of
the Wink Signal on the 'E' wire of its E&M port.
When System-A receives the acknowledgment from System-B, before the Wait Wink Timer expires,
System-A considers the trunk seizure as successful and starts dialing out DTMF digits as per the
MFCR2 Signaling protocol.
However, if System-A does not receive the Wink Signal within the ‘Wait Wink Timer', or if invalid Wink
Pulse is received (not according to Wink Pulse Timer), System-A will drop the call by turning its 'M' wire
low.

Incoming Call:
On detecting high signal on 'E' wire of the E&M port, System-B will consider it to be an incoming call
seizure and hence it will immediately make its 'M' wire high and send the Wink signal on the 'M' wire of
System-A to indicate its readiness to receive the called party number digits. The width of the Wink
Pulse (referred to as 'Proceed To Send' in MFCR2 Signaling) can be changed by setting the 'Wink
Pulse Timer'.
System-A dials out the digits as per the MFCR2 Signaling protocol.
When the called extension of System-B answers the call, System-B sends the Wink signal on the 'M'
wire of its E&M Port to indicate the call maturity.
Call Disconnection:
The call can be disconnected by the calling party, System-A, or the called party, System-B by changing
the status of the 'M' wire to Low.
Call Disconnection takes place when 'M' wire is low. So, it is recommended that the Call 'Release Type'
of the E&M Port for this Seizure Type (Immediate with Ack+Wink) be set to 'Status Change'.
If you select Immediate with Ack+Wink as the Seizure Type, you must configure the MFCR2 Signaling
parameters.
• Seizure Pulse: The method of seizing the E&M Line using Seizure Pulse for Outgoing and Incoming
Calls is described below.
0V
-48V
M M
E E
Rx Rx
System-A System-B
Tx Tx
SA SA
SB SB
Outgoing Call:
While making an out going call from the E&M port of System-A, it will send Seizure Pulse over the "M"
wire of its E&M port to seize the line.
Incoming Call:
While receiving an incoming call over the E&M Port System B will detect valid Seizure Pulse over the
"E" wire of its E&M port.
Seizure Pulse can be set for various time periods T1, T2 and T3 as required.

The Seizure Pulse for T1, T2 and T3 ranges from 000 to 999 milliseconds.
• Seizure Pulse + Wink: The method of seizing the E&M Line using Seizure Pulse + Wink for Outgoing
and Incoming Calls is as follows:
Outgoing Call:
While making an OG call, "Seizure Pulse" (as configured) will be sent on the "M" wire of E&M Port and
will start ‘Wait Wink Timer’ and expect ‘Wink’ from the remote device.
On receiving a valid Wink Pulse from the remote end within the Wait Wink Timer, digits will be dialed
out on the E&M Port.
If a valid Wink Pulse is not received from the remote end, digits will be dialed out on the expiry of ‘Wait
Wink Timer’.
Incoming Call:
On detecting valid Seizure pulse (matching with configured value of seizure pulse) on "E" wire of the
E&M Port, the E&M Port will send Wink pulse (of configured value) on "M" wire, and the call will be
considered to be present.
• Express: Express Signaling works as illustrated below.
0V
Till DSS Key is pressed

-48V
M M
E E
Rx Rx
System-A System-B
Tx Tx
SA SA
SB SB
To make a call from System A to System B, the caller from System A presses the DSS Key of the
desired E&M port.
For as long as the DSS key is pressed by caller from System A, the signal on the "M" wire of E&M port
of System A will be high.
When the destination extension on System B answers, the caller from System A releases the DSS key,
as the line seizure is successful.
When the caller from System A releases the DSS key, the signal on the 'M' wire of the E&M port of
System A, and the "E" wire of the E&M port of System B will go low.
• Radio A: This seizure type to be used for supporting Combat Net Radio (CNR) signaling on the E&M
port.
The E&M port of the SARVAM UCS will detect this pulse on the 'E' wire of its E&M port and recognize it
as a seizure signal (incoming call indication).

The length of this input signal (pulse) can be defined by setting the 'Minimum Pulse Width for Radio
Seizure' Timer. The SARVAM UCS will recognize the input signal on the 'E' wire of its E&M port as a
seizure signal only if the signal is present for the duration of this timer.
Once the incoming call is detected by the E&M port, the call is routed to the Routing Group number as
per the call routing logic configured for the E&M port.
When Routing Group member (extension) answers the call, speech is established with the CNR user.
Outgoing Call:
When any extension user of the System wants to contact the CNR user (Soldier), extension user must
grab the E&M port by dialing TAC/Selective Trunk Access / DSS key of E&M Port.
When E&M Port is grabbed by the extension user, the 'M' wire of the E&M port is made high, to indicate
the seizure signal to the radio equipment. The radio equipment then passes on the call to the CNR
user's wireless phone.
Incoming Call:
When status of "E" wire of the E&M port goes high for greater than or equal to the 'Minimum Pulse
Width for Radio Call' it is considered to be an incoming call, and the call is routed as per the current call
routing logic.
The Minimum Pulse Width for Radio Call' timer will be applicable only if the seizure type is configured
as 'Radio A' or 'Radio B'. The range of this timer is from 0000 to 9999 milliseconds. The default value of
this pulse is 150 milliseconds.
When an extension user of the System answers an incoming call, the 'M' wire of the E&M Port will be
made high.
The call between the extension user and the CNR user can be disconnected only if the extension user
disconnects the call. So, it is recommended that the call 'Release Type' of the E&M port be set to
'None'.
• Radio B: Characteristics of M and E wire for seizure type 'Radio B' are as follows:
Outgoing Call:
When any extension user of the System wants to contact the CNR user (Soldier), extension user must
grab the E&M port by dialing TAC/Selective Trunk Access / DSS key of E&M Port.
When E&M Port is grabbed by the extension user, the 'M' wire of the E&M port is made high, to indicate
the seizure signal to the radio equipment. The radio equipment then passes on the call to the CNR
user's wireless phone.
Incoming Call:
When the status of "E" wire of the E&M port goes high for greater than or equal to 'Minimum Pulse
Width for Radio Call' SARVAM UCS considers it to be an incoming call, and routes the call as per the
current call routing logic.
When an extension user answers the call, there is no signaling on the 'M' wire of the E&M port by
SARVAM UCS.

The call between the extension user and the CNR user can be disconnected only if the extension user
disconnects the call. So, it is recommended that the call 'Release Type' of the E&M port be set to
'None'.
The following table shows the status of the "M" wire of an E&M Port while making Outgoing calls and
while receiving Incoming calls.
Status of 'M' Wire after Status of 'M' Status of 'M' Wire on

E&M Seizure Type seizure and in conversation Wire when an IC answering and when in
in an OG Call call is initiated conversation in an IC Call
Immediate High Low High
Immediate with Ack High Low High
Immediate + Wink High Wink High
Immediate with Ack High Wink High

+Wink (MFC R2)
Seizure Pulse Low Low Low
Seizure Pulse + Wink Low Wink Low
Express Low Low Low
Radio A High Low High
Radio B High Low Low
• Orientation Type: Configure the Orientation Type of the E&M Port according to your installation
scenario.
Select 'Station' if the E&M port is to function as Extension. All Extension-related parameters, the
“Station Basic Feature Template” and the “Station Advanced Feature Template”, will be applicable to
this port.
Select 'Trunk' as orientation type if the port is to be assigned the feature of a trunk line. All trunk-related
parameters, “E&M Feature Template”, will be applicable to this port.
Select 'Tie Line'157 if the E&M port is to function as both a Station and a Trunk. The system will regard
the port as a Station for incoming calls and as a Trunk for outgoing calls. The Station Basic and
Advanced Feature Templates as well as the Trunk Feature Template will be applied on this port.
By default, the Orientation Type of all E&M ports is 'Station'.
• Dial Type: Digits can be dialed over E&M Tie Line by two methods:
• Tone: In this Dial Type, the DTMF signals will be sent on the "Tx" of the E&M port of the originating
side and it will be received over the "Rx" of the E&M port of the terminating side.
• Pulse: In this Dial Type, the dialed digits will be sent on the "M" wire of the E&M port of the
originating side and will be received over the "E" wire of the E&M port of the terminating side.
157. Such as in Systems used as Transit Exchanges as in a PLCC Network.

The way digits are sent varies according to the Trunk Seizure Type, as described for each Seizure Type
below.
• Express: the caller can make call by pressing the DSS key.
• Immediate: the System seizes the Tie Line and the system will start the Pause Timer. On the expiry of
the Pause Timer the System sends the digits to the remote System.
• Seizure Pulse + Wink: on receiving Wink signal from the terminating end, the originating side (which
initiates seizure) will start the Pause Timer and on expiry of Pause Timer it will start sending digits.
• Seizure Pulse: the originating System system will start the Pause Timer after sending the Seizure
Pulse. On expiry of the Seizure Pulse Timer it will send digits to the terminating end.
• Pulse Dial Ratio: This parameter is to be configured if 'Pulse' is selected as Dial Type in the previous
parameter. Select the Pulse Dial Ratio from any of the following values:
• 10PPS, 1:2
• 10PPS, 2:3
• 10PPS, 1:1
• 20PPS, 1:2
• 20PPS, 2:3
• 20PPS, 1:1
The default Pulse Dial Ratio is 10PPS, 1:2
• Wink Wait Timer (sec): This parameter is to be configured, if 'Immediate + Wink' or 'Seizure Pulse +
Wink' have been selected as Seizure Type for the E&M port.
The Wait Wink Timer is the Time period for which the system waits for the acknowledgment in the form
of Wink Signal on "E" wire of the E&M port of the SARVAM UCS to consider it as a successful seizure.
The range of this timer is from 000 to 255 seconds. By default the timer is set to '000' seconds.
• Wink Pulse Timer (msec): This parameter is to be configured, if 'Immediate + Wink' or 'Seizure Pulse
+ Wink' have been selected as Seizure Type for the E&M port.
The Wink Pulse Timer defines the width of the Wink Pulse. The range of this timer is from 0000-9999
milliseconds. By default the timer is set to '0000' milliseconds.
• Seizure Pulse Timers (msec): This parameter is to be configured, if 'Seizure Pulse' is selected as
Seizure Type for the E&M port.
• T1: This is the time period of the first ON period of the 'Seizure Pulse'.
• T2: this is the time period of the second ON period of the 'Seizure Pulse'
• T3: this is the time period of the third ON period of the 'Seizure Pulse'.
The range of T1, T2 and T3 is from 000-999 milliseconds. By default the timer is set to '000'
milliseconds.
• Minimum Pulse Width for Radio Seizure (msec): This Timer is to be configured if 'Radio A' or 'Radio
B' have been selected as the Seizure Type for the E&M ports.

The Minimum Pulse Width for Radio Seizure defines the time for which the SARVAM UCS will wait to
detect the width of the pulse sent by the Radio Interface Device158 on the 'E' wire of the E&M port and
recognize it as a seizure signal (incoming call indication).
The range of this timer is from 000 to 999 milliseconds. By default the timer is set to '150' milliseconds.
• Release Type: SARVAM UCS supports four methods to 'release' E&M calls based on, which end will
release the call and 'release pulse width'. These are:
• None: Select this option if you have selected "Express" as Trunk Seizure Type for the E&M port. It
is advisable to keep the Release Type as "None" in case the protocol does not support any
signaling for disconnecting the E&M port, for the reason that 'Trunk-to-Trunk Inactivity Timer' will be
started if the E&M port with Release type "None" is involved in a Trunk-to-Trunk call.
• Release Pulse: Select this option if the specific Pulse width of the Release Pulse is to be used to
disconnect the call. This Pulse width is configurable, as shown below:
The call can be disconnected by either party by sending Release Pulse. Consider the following
example:
0V
Release Pulse
-48V
2001 3001
M M
E E
Rx Rx
System-A System-B
Tx Tx
SA SA
2002 3002
SB SB
SLT 2001 is connected to System-A and SLT 3001 is connected to System-B.
SLT 2001 user goes ON-Hook.

• System-A will send Release Pulse over the "M" wire of its E&M port.
• System-A will release its E&M port.
• The Release Pulse sent by System-A will be received and detected on the "E" wire of the E&M port
of the System-B.
• System-B will release its E&M port.
SLT 3001 user goes ON-Hook.

• System-B will send Release Pulse over the "M" wire of its E&M port.
• System-B will release its E&M port.
• The Release pulse sent by System-B will be received and detected on the "E" wire of the E&M port
of the System-A.
• System-A will release its E&M port.
158. Connected on the 'M' wire of the E&M port. The Radio Interface Device sends pulse of approximately 100msec or higher.

• Status Change: Select this option, if Status change of 'M' wire (‘M’ wire is low) is to be considered
for release of the E&M call.
By default, 'Status Change' is selected as the Release Type for all E&M ports.
If you selected ‘Immediate with Ack + WInk’ as Seizure Type for the E&M Port, select ‘Status
Change’ as Release Type for the port.
• Release Pulse Timer (msec): This timer is to be configured if you selected the option 'Release Pulse'
as the Release Type for E&M calls in the previous parameter.
This timer defines the specific Pulse width of the Release Pulse is to be used to disconnect the call.
The range of this timer is from 0000-9999 milliseconds. By default the timer is set to '0000'
milliseconds.
• CCS - When End Point: CCS (Compander Control Signal) is a type of signal used by PLCC Networks
to improve the quality of speech transmission. The PLCC network awaits this signal from the System
when speech is established. SARVAM UCS supports CCS. The system sends CCS signal to the PLCC
panel.
This parameter is relevant if the E&M line is being used in a Power Line Communication Network
(PLCC).
This flag should be enabled if the E&M port is used as an Endpoint in a PLCC network. When the E&M
Port is used as an Endpoint, the system sends CCS to the PLCSS panel while making an outgoing call
through the E&M port and when receiving an incoming call on the E&M port.
• CCS - When Transit Exchange: This flag is to be enabled if the E&M port is used as a Transit
exchange in a PLCC network.
When the E&M Port is used as a Transit Exchange: The system sends CCS to the PLCC panel when
there is an Incoming/Outgoing Transit call through the E&M port.
• DTMF Detection: This flag is relevant when the "Dial Type' for the E&M Port is selected as 'Tone'. This
flag is of significance while receiving incoming calls.
• Max. OG Pulse Digit Count: This count defines the maximum number of digits that can be dialed out
to make a call. When dialing out the number, if the number of digits exceeds this count, the port which
is used for dialing these numbers is released automatically.
• Idle Wait Timer (sec): This timer signifies the time after which the codes could be simply (Station
Numbers or Station Numbers with Exchange ID) dialed over the E&M trunk.
The Idle Wait Timer is useful in two conditions:
• When Forced Disconnection is used. For example, two exchanges A and B are connected
through E&M trunk. Extension 2002 of System A is talking to extension 3001 of System B over the
E&M line.
Extension 2001 of System A calls extension 3001 of System B and finds it to be busy.

Extension 2001 is allowed to use forced disconnection feature. Extension 2001 issues the forced
disconnection command. The System A disconnects the Extension 2002. It then waits for the Idle
Wait Timer to expire and then dials 3001 over the E&M trunk.
• To stop any station from grabbing the E&M trunk until call is released. For example, when
Extension 2001 of System A goes On-Hook, System A sends a release signal over the E&M trunk
to System B. In turn, System B sends a release signal to System A as an acknowledgment.
The E&M Idle Wait Timer set in System A does not allow any other extension of System A to grab
the E&M trunk. Similarly, E&M Idle Wait Timer set in System B does not allow any extension of
System to grab the E&M trunk.
• Flash Timer (msec): This Timer is significant when the System acts as a Transit exchange for a call.
The flash received on one E&M Port is generated on another E&M Port involved in a Transit call.
• Pause Timer (msec): This Timer defines the time for which the system waits before dialing the outside
number after grabbing the E&M trunk.
Sometimes an extension user may not get a dial tone immediately on grabbing a trunk, in this case, the
extension user may wait for the dial tone before dialing out the number. However, when the system
dials out the number, if there is no pause time, it is possible that the system may dial out the number
before getting the dial tone. This may result in a wrong number being dialed out. The Pause Timer
helps avoid this.
The range of this timer is from 0000 to 9999 msec. By default the Timer is set to 800 msec.
• Pseudo Answer Supervision Timer (sec): This is the time period after which, the system will
consider the call as matured, irrespective of whether the call was answered or not. At the end of the
Timer, the system will start detecting Disconnect Supervision.
The range of this timer is from 000 to 255 seconds. By default the Timer is set to 030 seconds.
• Ring Timer (sec): This is the time for which the extensions connected to SARVAM UCS ring for
incoming calls.
The Ring Timer is useful in situations where the users may not be able to immediately answer on the
first few rings. The range of this timer is from 000 to 255 seconds. By default the Timer is set to 255
seconds.
• Inter-Digit Pause Timer (msec): This Timer defines the time gap to be inserted between digits of a
number string being dialed by the system.
The range of this timer is from 000 to 999 milliseconds. By default the Timer is set to 750 milliseconds.
• DTMF Out Dial: This parameter is of significance when the system dials out DTMF digits to enable the
device at the remote end (in this case a System) to detect and decode the Tones. You must configure
both the DTMF ON Time and the Level (dB) according to the DTMF digit detection capacity of the
remote System.
For example, System A and System B are connected over E&M Line. System B detects DTMF digits
only if the tone remains present (ON) for 100 milliseconds frequency and at a transmit level of 4 dB.
The DTMF Out Dial parameter for System A should be configured accordingly. The DTMF Out Dial ON
Time should be set to 100ms and Level to 4dB.

• DTMF ON Time (msec): This is the Time for which the DTMF digit tone will remain ON, while being
dialed out by the SARVAM UCS. The range of this timer is 50 to 500 milliseconds. By default the
ON Time is set to 100 milliseconds.
This Timer must be configured according to the DTMF digit detection capacity of the remote device.
• DTMF Tx Level (dB): This is the Transmit Level of the DTMF digit dialed out by the system. The
range of DTMF Out Dial 'Level' is from 0 to 7. By default DTMF Out Dial Level is set to 3.
• MFC R2 Signaling: This parameter is relevant only if you selected 'Immediate with Ack + Wink' as the
Seizure Type. Configure the following timers related to MFCR2 Signaling.
• Forward Tone Maximum ON Time (T1) (sec): the range of this timer is from 1 to 99 seconds. By
default it is set to 15 seconds.
• Forward Tone Maximum OFF Timer (T2) (sec): The range of this timer is from 1 to 99. By default
it is set to 24 seconds.
• Maximum Compelled Cycle Timer (T3) (sec): the range of this timer is from 1 to 99. By default it
is set to 15 seconds.
• Pulse Duration for Pulse Signal (msec): The range of this timer is from 001 to 999. By default, it
is set to 150 seconds.
• Pulse Signal Maximum Wait Timer (sec): The range of this timer is from 1 to 99. By default, it is
set to 15 seconds.
• First Forward Tone Wait Timer (sec): The range of this timer is from 8 to 24. By default, it is set to
15 seconds.
• Minimum MF Signal Persist Timer (msec): The range of this timer is from 1 to 255 seconds. By
default, it is set to 20 seconds.
• Inter Digit Wait Timer (sec): By default it is set to 10 seconds.
• Ask CLI: By default, it is enabled.
• Prefix String: This parameter is useful when the System’s connected using E&M lines do not send 0
when the user dials a number of another exchange. You must configure this parameter as zero along
with the exchanges’s ID and appropriately configure the strip digit count parameter in the CUG Table.
For details, see “Prefix String Feature: This feature is programmed only for few Exchanges like ET1,
Genesis, or BPL which does not send ‘0’ when user dials a number to call an extension number of the
exchange. For other Exchanges like ‘ET2’ it is not required to be programmed as it sends ‘0’ also if it is
dialed.”.
• Category (Logical Partitioning): This parameter assigns the E&M Port to a trunk category for the
purpose of Logical Partitioning. By default all E&M Ports are assigned to Category 3159.
If you have re-defined Category 3 or have assigned E&M ports to a different category, say Category 2,
enter the same number here.
159. Trunk ports used to interconnect two Systems are assigned this category.

You may configure the call permission between the Category assigned to E&M Ports and other
Categories. Refer the feature description “Logical Partition” to know more.
Customizing E&M Feature Template

By default E&M Feature Template 01 is applied on all E&M Ports. This template has 'Station' as the default
Orientation Type.
If all the E&M Ports are to be configured as 'Stations', then retain this template.
If all the E&M Ports are to be configured as 'Trunks', use the default E&M Feature Templates 09 and 10 which have
'Trunks' as Orientation Type.
If some of the E&M Ports are to be configured as Stations, some as Trunks and yet others as Tie Lines, prepare
different E&M Feature Template for each Orientation Type and apply them to the related ports.
The E&M Feature Template can be customized using Jeeves and a Telephone.
Customizing E&M Feature Template using Jeeves

• Log in to Jeeves as System Engineer.
• Under Configuration, click E&M Configuration.
• Click E&M Feature Template.
• Select an E&M Feature Template Number.
• Customize the E&M Feature Template.

• Now, apply the E&M Feature Template you customized to the E&M Ports and T1E1PRI ports.
To apply E&M Feature Template on E&M ports:
• Click the E&M Parameters to open the page.
• Apply the E&M Feature Template you customized to the E&M Port by entering the template number in the
E&M Feature Template field of this port.
To apply E&M Feature Template on T1E1PRI ports:
• Click T1E1 Configuration.
• Click the E&M Signaling.

• Click the tab of the desired T1E1 trunk port number (1 to 8), you wish to apply the E&M feature Template.
• Enter the number of the Template you customized in the field E&M Feature Template of the selected
T1E1 trunk port.
Customizing E&M Feature Template using a Telephone

To change the default values of a Parameter in an E&M Feature Template:

• 6002-1-Template Number-Parameter Number-Code to change the value of a parameter in a single
template.
• 6002-2-Template Number-Template Number-Parameter Number-Code to set the same value for
the parameter in a range of templates.
• 6002-*-Parameter Number-Code to set the same value for the parameter in all templates.
Where,
Template Number is the number of the E&M Feature Template from 01 to 50.
Parameter Number is the number of the E&M Feature Template Parameter from 01 to 35.
Code is the parameter value.
Refer the following table for the Parameter Number and default parameter values of all E&M Feature
Templates.

Default E&M Feature Templates
For example, you want to Immediate with Ack +Wink in the E&M Template number 2.
• Dial 6002-1-2-01-9
Where
2 is for E&M Template number 2
01 is parameter number for Seizure Type
9 is the value of Immediate with Ack+Wink
To restore default values to the Parameters of an E&M Feature Template:

• Dial 6001-1-Template Number to restore default values of the parameters of a single template.
• Dial 6001-2-Template Number-Template Number to restore the default values of a range of
templates.
• Dial 6001-* to restore the default values of all templates.

To assign an E&M Feature Template to an E&M port
• Dial 6003-1-E&M-Template Number to assign a template to a single E&M port.
• Dial 6003-2-E&M-E&M-Template Number to assign the same template to a range of ports.
• Dial 6003-*-Template Number to assign the same template to all ports.
Where,
E&M is the number of the E&M Software port from 001 to 128.
Default: Template 01 is assigned to all E&M ports.
To assign E&M Feature Template to a T1E1 Port:

• Dial 6004-1-T1E1-Template Number to assign a template to a single T1E1 port.
• Dial 6004-2-T1E1-T1E1-Template Number to assign the same template to a range of T1E1 ports.
• Dial 6004-*-Template Number to assign the same template to all T1E1 ports.
Where,
T1E1 is the number of the T1E1PRI Software port, 01 to 08.
• Exit SE mode.
Trunk Feature Template

The Trunk Feature Template is a set of general features that define the behavior of a Trunk port. SARVAM UCS
offers 50 such Templates.
A Trunk Feature Template is assigned to all the Trunk types: SIP, T1E1 PRI, BRI, Mobile, CO and E&M.
Trunk Feature Template Parameters

The Trunk Feature Template contains the following features:
• Time Table: Select a Time Table for the Trunk ports.
A Time Table is a schedule of the three Time Zones, namely: Working Hours, Break Hours, Non-Working
hours for a week.
Certain features of the SARVAM UCS like Operator, Auto Attendant, DISA, Trunk Landing Group, require
the trunk to behave differently in each Time Zone. For example, it can be made to land on the Operator
extension during working hours, and on the extension of the dining area during Break (lunch) hours, and
on the extension of the Security Personnel during non-working hours.
So, a Time Table is assigned to extensions defining the Time Zones for the entire week, so that the system
can execute the Time Zone-dependent features and facilities according to the Time Table.
There are 8 different Time Table templates to select from. By default, the Time Table 1 is assigned to all
Trunk Feature Templates. All seven days of the week are 'working hours 9:00-18:00' with break hours
'13:00 -14:00 hrs'.
You may also customize the default Time Table 1 OR customize and assign a different Time Table to the
Trunk Feature Template. Please refer the topic “Time Tables” for more details.
• Operator: Define the Operator for the Trunks on which the template is applied. Operator is used to route
the call when the caller dials '9' during an Auto Attendant call. This parameter is of significance only if Built-
In Auto Attendant/DISA is enabled on the trunk.

The system supports multiple Operators. In each Time Zone any one of the 20 Operators can be selected.
Trunks may be assigned to a single Operator, or different groups of Trunks may be assigned to different
Operators, so that call management is more efficient. For instance certain Trunks may be assigned to
Operator 1, while some may be assigned to Operator 2 and the rest to Operator 3.
Operator 1 is the default in the Trunk Feature Template. If you want to assign different trunks to different
Operators, you must create a separate Trunk Feature Template with a different Operator for each trunk
group.
Refer the topic “Operator” to know more.
• CLI Based Routing: Select the check box to enable CLI Based Routing on the Trunk for each Time Zone:
Working Hours (WH), Break Hours (BH) and Non-working Hours (NH). Default: disabled.
If you enable CLI Based Routing on the trunk for a Time Zone, make sure you also configure the CLI
Based Routing Table. To know more, refer the feature description “CLI Based Routing”.
• Trunk Landing Group (Routing Group): This parameter allows you to configure the group of extensions
on which incoming calls on the trunks (to which this template is assigned) are to be landed. This group of
extensions is referred to as 'Trunk Landing Group' (TLG).
To configure the TLG, you must first configure Routing Groups. Refer “Trunk Landing Group (TLG)” for
instructions on configuring trunk landing groups. Also refer the topic “Routing Group”.
There are as many as 96 Routing Groups which can be assigned as TLG. By default, Routing Group 01 is
assigned as TLG for all Time Zones. If you have prepared a different TLG for each Time Zone, for
example, Routing Group 02 for Working Hours and Break Hours, Routing Group 3 for Non-Working Hours,
then enter the number of these Routing Groups in the TLG field.
• Auto Attendant: This parameter is to be configured if you want to enable “Auto Attendant” on the trunk
ports on which you will apply the template.
Auto Attendant can be enabled or disabled for each Time Zone, namely Working Hours (WH), Break
Hours (BH) and Non-Working Hours (NH).
• For each Time Zone, you may select the desired Auto Attendant Type from the following:
• OFF: Select this option if you want to disable Auto Attendant for the Time Zone.
• Built-In Auto Attendant: Select this option if you want the calls to be answered by the built-in Auto
Attendant of the SARVAM UCS. In Built-In Auto Attendant, SARVAM UCS answers the call using
Voice Modules, if assigned, or it answers the call and plays the appropriate call progress tone - Dial
tone, Ring Back tone, Busy tone - for each call state.
If you select this option, make sure you also configure the Built-In Auto Attendant related Timers
and Flags, record and assign the Built-In Auto Attendant related Voice Message and set the Start
Time for the Greeting Messages. Refer the topics “Auto Attendant” , “Voice Message Applications”
and “Greeting Message Time” in System Parameters for instructions.
• Voice Mail Auto Attendant: Select this option if you want to the calls to be answered by the Auto
Attendant of the Voice Mail System. The Voice Mail System of SARVAM UCS answers calls and
processes them according to the Voice Mail Auto Attendant Menu assigned to the trunk.

By default, Auto Attendant is disabled (OFF) for all the Time Zones.
• Auto Attendant Delayed Timer: Set this Timer, if you want to enable “Delayed Auto Attendant” on the
trunk.
When you enable Delayed Auto Attendant, SARVAM UCS routes the incoming call on the trunk to the
Trunk Landing Group assigned to this trunk. It waits for the duration of the Auto Attendant Delayed
Timer for any of the extensions in the Trunk Landing Group to answer the call.
If none of the extensions in the Trunk Landing Group answers the call before the expiry of the Auto
Attendant Delayed Timer, SARVAM UCS processes the call according to the type of Auto Attendant -
Built-In Auto Attendant or Voice Mail Auto Attendant, set for the trunk.
To enable Delayed Auto Attendant, set the timer to the desired value from the list. By default, it is set as
Never that is it is disabled.
• Voice Mail Auto Attendant (VMAA) Menu: if you have selected the Voice Mail Auto Attendant as the
Auto Attendant Type, select the VMAA Menu to assign to the respective Trunk Feature Template.
You may click the Voice Mail Auto Attendant (VMAA) Menu link to edit the parameters of desired VMAA
Menu. For details, see “Voice Mail Auto-Attendant Menu”.
• DISA: This parameter is to be configured if you want to enable “Direct Inward System Access (DISA)” on
the trunk ports on which you will apply the template.
DISA can be enabled or disabled for each Time Zone, namely Working Hours (WH), Break Hours (BH) and
Non-Working Hours (NH).
For each Time Zone, you may select the desired DISA option from the following:
• Disabled: Select this option if you want to disable DISA.
• PIN Auth.-Multiple calls: Select this option if you want to enable DISA with PIN Authentication and
allow multiple calls during the DISA login session.
• CLI Auth.-Multiple calls: Select this option if you want to enable DISA login with CLI Authentication
and allow multiple calls to be made during the DISA login session.
Caller numbers that do not match with the CLI Table will be routed as per the logic of the Trunk Feature
Template.
• CLI Auth.-One call: Select this option if you want to enable DISA session with CLI Authentication, and
allow only a single call to be made during the DISA login session. This form of DISA is used when
SARVAM UCS is installed in a Gateway application. This form of DISA is applicable on CO Trunks
only.
By default, DISA is disabled for all Time Zones.
• Trunk Auto Answer: This parameter is relevant only if you want to enable the “Trunk Auto Answer”
feature on the Trunk ports on which this template is applied.
Trunk Auto Answer enables calls landing on a trunk to be answered automatically by greeting the caller
with a voice message before the call is actually handled.

You can set the following types of Trunk Auto Answer:
• OFF: Select this option if you do not want Trunk Auto Answer on the trunk.
• For all Calls: Select this option if you want all incoming calls landing on the trunk line to be answered.
• When Busy: Select this option if you want the system to answer incoming calls on the trunk to be
answered if the landing destination is busy.
• Delayed: Select this option if you want SARVAM UCS to answer the incoming calls on the trunk if not
answered by the landing destination within a certain time period, set as the Delayed Trunk Auto
Answer Timer.
The system first routes the incoming calls to the Trunk Landing Group. It waits for the duration of the
Delayed Trunk Auto Answer timer for any of the extensions in the Trunk Landing Group to answer the
call.
If the call is not answered by any of the extensions, before the expiry of this timer, the system answers
the call and routes it as per the Trunk Auto Answer logic.
• If you select Delayed as the Trunk Auto Answer option, you must also configure the Delayed Trunk
Auto Answer timer. Valid Range of the timer is 01 to 99 seconds. By default, it is set as 10
seconds.
Trunk Auto Answer can be enabled or disabled for each Time Zone, namely Working Hours (WH), Break
Hours (BH) and Non-Working Hours (NH).
By default, Trunk Auto Answer is Off.
If you have enabled Trunk Auto Answer for All Calls, When Busy or Delayed, you must also set the Trunk
Auto Answer Greeting Message.
• Trunk Auto Answer Greeting Message: This parameter is to be configured only if you have enabled
Trunk Auto Answer For All Calls or When Busy for a Time Zone in the previous parameter.
Assign the number of the Trunk Auto Answer Greeting with which callers will be greeted. For this you must
first record a Voice Module with the desired Greeting Message and assign it to this parameter.
You can assign up to 4 Greetings Messages for Trunk Auto Answer and assign a different Greeting
message for each Time Zone. Refer the topic “Voice Message Applications” for instructions on configuring
the greetings.
Enter the number of the Greeting Message you want to be played for each Time Zone. By default, Trunk
Auto Answer Greeting number 1 is assigned to all Time Zones.
• Trunk Auto Answer RBT Message Type: This parameter is to be configured only if you have enabled
Trunk Auto Answer 'For All Calls' or 'When Busy' for a Time Zone in the previous parameter.
When Trunk Auto Answer is enabled on a trunk, the system will answer the caller with a Greeting message
once, and play the Ring Back Tone (RBT) Message Type you have selected. You can select an RBT
Message Type from the following options:
• None: The system will play Ring Back Tone to the caller after the Trunk Auto Answer Greeting
Message.

• Internal MOH: The system will play internal music-on-hold to the caller after playing the Trunk Auto
Answer Greeting Message.
• RBT Message: The system will play a voice message continuously to the caller.
If you select this option, you must first record a Voice Module with the desired RBT Greeting Message.
You can set up to 4 RBT Messages. You can also assign a different RBT message for each Time Zone.
Refer the topic “Voice Message Applications” for instructions on configuring the RBT Message.
Assign the number of the RBT Message you want to be played to callers in each Time Zone.
By default, 'None' is selected as RBT Message Type for all Time Zones.
• Trunk Auto Answer Busy Bye Message: This parameter is to be configured only if you have enabled
Trunk Auto Answer ('For All Calls' or 'When Busy') for a Time Zone.
When Trunk Auto Answer is enabled on a trunk, the system will answer the caller with a Greeting message
once, and play the Ring Back Tone (RBT) Message Type you have selected (see previous parameter)
continuously for the duration of the Built-In Auto Attendant Inactivity Timer. If the landing destination (called
extension) is busy on the expiry of this Timer, the system will inform the caller about the busy state in two
ways, which you can select from the following options:
• None: The system will play a Busy Tone to the caller.
• Bye Message: The system will play a voice message to the caller.
If you select this option, you must first record a Voice Module with the desired Bye Message.
You can set up to 4 Busy Bye Message. You can also assign a different Busy Bye message for each
Time Zone. Refer the topic “Voice Message Applications” for instructions on configuring the Busy Bye
Message.
Assign the number of the Bye Message you want to be played to callers in each Time Zone.
By default 'None' is selected as Busy Bye Message for all Time Zones.
• Priority: Select a Priority Level for the trunks on which the template will be applied.
Each trunk of the SARVAM UCS can be assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority.
Whenever there are incoming calls on multiple trunks, the call on the trunk with higher priority will be
answered by the Operator extension first. To know more, read the feature description “Priority”.
By default, the Priority of all trunks is set to '9-Highest'. Decide what Priority Level you will assign to the
trunks and set the desired level for the trunk.
• SMDR-OG Storage: This flag is used to enable or disable the storage of details of outgoing calls from the
trunk. Please refer the topic “Station Message Detail Recording-Storage” for more details. By default,
storage of outgoing calls is enabled.

• SMDR-IC Storage: This flag is used to enable or disable storage of details of incoming calls on the trunk.
Please refer the topic “Station Message Detail Recording-Storage” to know more. By default, storage of
incoming calls is enabled.
• Hold on DSS Key Press: This flag defines the 'Hold' state of the external called party, when an extension
user presses a DSS key to dial another port.
For example, the DKP extension user (on DKP-001 port) is in the middle of speech with an external party
on a Trunk port CO-002.
If extension user of DKP-001 presses a DSS key to call another extension port DKP-003, two situations
are possible, depending on whether the Hold on DSS Key Press flag is enabled or disabled:
• When the Hold Flag is enabled: CO-002 will be played music-on-hold. DKP-001 will hear Ring Back
Tone and the call will be placed on DKP-003.
• When Hold Flag is disabled: CO-002 will be disconnected. DKP-001 will hear Ring Back Tone, and
call will be placed on DKP-003.
By default the Hold On DSS Key Press flag is enabled.
• Forced Account Code: This parameter is related to the “Account Codes” feature of the SARVAM UCS.
This flag must be enabled, if the feature Forced Account Code is to be applied on the trunks.
When this flag is enabled, the system will prompt extension users to dial the Account Code whenever they
grab a trunk to dial out a number. The system will allow extension users to dial out numbers only when
after they have dialed the Account Code or Name.
By default, the flag is disabled. Refer the feature description for “Account Codes” to know more.
Account Codes feature must also be enabled in the Class of Service of extension users who are to be
allowed this feature.
• Call Cost Calculation Pulse Rate Option: This parameter is to be configured only if you want to apply
the “Call Cost Calculation (CCC)” feature on the trunks on which the template is applied.
You have four options for Pulse Rate Types. Select from Pulse Rate Type for Pulse Rate Option 1 to 4
which you want to apply on the trunks.
• Call Cost Calculation Time Schedule: This parameter is to be configured only if you want to apply the
“Call Cost Calculation (CCC)” feature on the trunks on which the template is applied.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases, you
must first define the Time Zone (time of the day) for which a particular Pulse Rate should be applied and
the Time Schedule for each Time Zone.
You can configure up to four different Time Zones - T1, T2, T3 and T4 with different Pulse rates in the
CCC-“Configuring Pulse Rate Types”.
Now, configure the Call Cost Calculation Time Schedule, by specifying the Start Time and the End time (in
24hours: minutes format) for each Time Zone.

The default Time Schedule (starts and end time) for each Time Zone Index are as follows:
Start
Time Zone Index End Time
Time
T1 00:00 23:59
T2 00:00 23:59
T3 00:00 23:59
T4 00:00 23:59
If your service provider offers the same Pulse Rate for the entire day,
• configure only one Time Zone Index with the Pulse Rate, for instance, T1, in the CCC-Normal Pulse
Rate Table.
• Now, set the Time Schedule for Time Zone, T1, with the start and end time in Hours: Minutes format;
• set the start and end time of the other Time Zone Index, T2 to T4, to 00:00 (hours: minutes).
Similarly, if your service provider supports two different Pulse Rates in a day, set the Start and the End time
for two Time Zones and set the other two to 00:00.
• Call Duration Control: This parameter is to be configured only if you want to apply the “Call Duration
Control (CDC)” feature on the trunks on which the template is applied.
To enable CDC for incoming calls, select the Call Duration Control - For IC Calls check box.
To enable CDC for outgoing calls, select the Call Duration Control - For OG Calls check box.
• Call Taping: This parameter is to be configured only if you want to apply the “Call Taping” feature on the
trunks on which the template is applied.
Select the Call Taping check box to enable.
• Click Submit.
Customizing Trunk Feature Templates

By default, Trunk Feature Template 01 is assigned to all Trunk Types: SIP, T1E1 PRI, BRI, Mobile, CO and E&M.
The default values of the parameters of this template are sufficient to meet the common requirements of most
users.
If the default values of the Template fulfill the requirements of all Trunk types, retain this template. If you want to
change some of the feature settings and apply the template to all trunk types, you may simply customize this
template.
However, if you want to assign different feature settings for different trunk types, you are recommended to prepare
and apply separate Trunk Feature Templates for each Trunk type.
This can be done using Jeeves and from a telephone.

Customizing Trunk Feature Template using Jeeves
• Under Configuration, click Trunk Feature Template.
• Select a Trunk Feature Template Number, for instance 02.
• Customize the Template by changing the values of the desired parameters.
• Apply the Trunk Feature Template you customized to the CO Port type.
• To apply the template on CO Trunks,
• Open CO Parameters page.
• Go to the CO software ports to which this Template you customized is to be assigned, for instance CO-
001 to 003.

• Enter the number of the Trunk Feature Template you customized, 02, in the Trunk Feature Template
field of this port.
• To apply the template on Mobile Ports,
• Click Mobile Configuration.
• Click Mobile Port Parameters to open the page.

• Enter the number of the template you customized in the field Trunk Feature Template of the Mobile
Software ports to which you want to assign this template.
• Repeat the same steps to create another template and apply it on the desired Trunk Port type.
To apply the template on E&M ports, open E&M Parameters page under E&M Configuration.
To apply the template on SIP Trunks, open SIP Parameters page under VoIP Configuration.
To apply the template to BRI Trunks, open BRI Parameters page under BRI Configuration.
To apply the template on T1 lines, open T1 Port Parameters page under T1E1 Configuration.
To apply the template on E1 lines, open E1 Port Parameters page under T1E1 Configuration.
• Remember to click Submit to save your settings on each page.
Customizing Trunk Feature Template using a Telephone

To change the default value of a Trunk Feature Parameter in a Template, dial:
• 5802-1-Trunk Feature Template Number-Feature Number-Code to change the value of a parameter

in a single template
• 5802-2-Trunk Feature Template Number-Trunk Feature Template Number-Feature Number-Code
to set the same value for the parameter in a range of templates.
• 5802-*-Feature Number-Code to set the same value for the parameter in all templates.
Template Number is the number of the Trunk Feature Template from 01 to 50.
Parameter Number is the number of the Trunk Feature Template Parameter from 01 to 40.
Code is the value for each parameter from 0 to 9.
Refer the following table for the parameter numbers and the values for the codes.

Trunk Feature Templates - Parameter Numbers and Default Values

For example: Assign Trunk Landing Group 03 to Working Hours and Break Hours and Trunk Landing
Group 04 to Non-Working Hours in Template 02.
To assign TLG 03 to Working Hours, dial:

• 5802-1-02-03-03
Where,
03 is the parameter number for TLG-Working Hours
03 is the code for the TLG to be assigned to Working Hours.
To assign TLG 03 to Break Hours, dial:

• 5802-1-02-04-03
Where,
04 is the parameter number for TLG-Break Hours
03 is the code for the TLG to be assigned to Break Hours.
To assign TLG 04 to Non-Working Hours, dial:

• 5802-1-02-05-03
Where,
05 is the parameter number for TLG-Non-Working Hours
03 is the code for the TLG to be assigned to Non-Working Hours.
To default Trunk Feature Templates, dial:

• 5801-1-Trunk Feature Template Number to default a single template.
• 5801-2-Trunk Feature Template Number-Trunk Feature Template Number to default a range of
templates.
Where,
Trunk Feature Template is the template number from 01 to 50.

• To assign a Trunk Feature Template to a Trunk Port type:
For CO trunks, dial:

• 5803-1-CO- Trunk Feature Template Number to assign a feature template to a single CO port.
• 5803-2-CO-CO- Trunk Feature Template Number to assign a feature template to a range of CO
ports.
• 5803-*- Trunk Feature Template Number to assign a feature template to all CO ports.
Where,
Template Number is the number of the customized Trunk Feature Template, from 01 to 50. Default:
Template 01.
For BRI trunks, dial:

• 5804-1-BRI- Trunk Feature Template Number to assign a feature template to a single BRI port.
• 5804-2-BRI-BRI- Trunk Feature Template Number to assign a feature template to a range of BRI
ports.
• 5804-*- Trunk Feature Template Number to assign a feature template to all BRI ports.
Where,
BRI is the Software Port number of the CO port from 01 to 32.
Template 01.
For E&M Trunks, dial:

• 5805-1-E&M- Trunk Feature Template Number to assign a template to a single E&M port.
• 5805-2-E&M-E&M- Trunk Feature Template Number to assign the same template to a range of E&M
ports.
• 5805-*- Trunk Feature Template Number to assign the same template to all E&M ports.
Where,
E&M is the Software Port number of the port from 001 to 128.
Trunk Feature Template 01.
For T1E1 Trunks, dial:

• 5806-1-T1E1- Trunk Feature Template Number to assign a template to a single T1E1 port.
• 5806-2-T1E1- Trunk Feature Template Number to assign the same template to a range of T1E1
ports.
• 5806-*- Trunk Feature Template Number to assign the same template to all T1E1 ports.
Where,
T1E1is the Software Port number of the port from 01 to 08.
Trunk Feature Template 01.
For Mobile ports, dial:

• 5807-1-Mobile- Trunk Feature Template Number to assign a template to a single Mobile port.
• 5807-2-Mobile-Mobile- Trunk Feature Template Number to assign the same template to a range of
Mobile Ports.
• 5807-*- Trunk Feature Template Number to assign the same template to all Mobile ports.
Where,
Mobile Port Number is the number of the software port from 001 to 064.
Template Number is the number of the Trunk Feature Template, from 01 to 50.
Default: Trunk Feature Template 01.
• Exit SE mode.

Configuring VoIP Parameters
Configuring VoIP Parameters using Jeeves

• Click VoIP Parameters to open the page.
VoIP Server Domain

Click VoIP Server Domain to expand.
• VoIP Server Domain: This parameter is of relevance only if you are configuring SIP Extensions.
The system is capable of maintaining a domain for registering SIP clients (any SIP-enabled device) as SIP
Extensions.
Configure the Server Domain if you want SIP clients to register with the Registrar Server of the system
using the domain handled by the system160. The Domain Name can be a maximum of 40 characters.
Default: Blank.
If you configure Server Domain for registration of SIP clients, you must also map the Domain Name and
the IP Address of the WAN Port to the DNS Server in the network.
160. SIP clients can be registered with the system either using the domain handled by the CPU or using the WAN or LAN Port IP
Address.
If domain is programmed, the system will listen for the SIP message which is redirected to the programmed domain only. It will also
listen for SIP messages on the WAN IP address and LAN IP address.
But if domain is not programmed, the system will listen for SIP messages only on the WAN IP Address and LAN IP address.

RTP Mode
Click RTP Mode to expand.
• RTP Mode: Select the desired RTP mode using which you want SARVAM UCS to route speech in SIP to
SIP calls.
You can select from the following:
• Transcoding: When this option is selected, RTP packets will be routed using Vocoder channels161 for
SIP to SIP calls. SIP users (Extended SIP Clients and Open SIP Clients) will be able to access all the
features of SARVAM UCS. This option uses two Vocoder channels for SIP to SIP calls. Thus the
maximum number of SIP to SIP calls per system is equal to the number of Vocoder channels divide by
2.
• RTP Relay: When this option is selected, RTP packets will be routed without using Vocoder channels
for SIP to SIP calls. The system will use Vocoder channels to route SIP to TDM calls and vice versa.
System will use Vocoder channels for some features and Open SIP Clients will be able to use limited
features of SARVAM UCS. Refer to “SARVAM UCS Features supported with RTP/Direct RTP” for more
details.
• Direct RTP: When this option is selected, no Vocoder channel will be used for SIP to SIP calls and
RTP packets will be sent to and fro directly between SIP end points. If transfer of RTP packets is not
possible between SIP end points, system will use RTP Relay as the fallback option. The system will
use Vocoder channels to route SIP to TDM calls and vice versa. System will use Vocoder channels for
some features and Open SIP Clients will be able to use limited features of SARVAM UCS. Refer to
“SARVAM UCS Features supported with RTP/Direct RTP” for more details.
• Direct RTP is not supported on SIP Trunks and MATRIX VARTA ADR100/AMP100/WIN200, therefore
the system will use RTP Relay.
• If RTP Relay or Direct RTP is selected,

• Maximum 550 Audio calls or 55 Video calls are supported in SARVAM UCS.
• Certain voice messages will not be played. See “Voice Message Applications” for more details.
• If you change the RTP Mode, the system will release all ongoing calls.
• MoH Vocoder Preference: If RTP Mode is set as RTP Relay or Direct RTP, the system will play the MoH
stored in the Vocoder module when the system holds any SIP end point. The MoH is played as per the
configured MoH Vocoder Preference.
Select the Vocoders in the order of their preference, for MoH Vocoder Preference 1, MoH Vocoder
Preference 2 and MoH Vocoder Preference 3.
161. The number of Vocoder channels that will be supported would be as per the license you purchase.

If required, you can customize the MoH to be played as per your requirement. For detailed instructions,
see “Uploading Custom MoH”.
Quality of Service (QoS)

Click Quality of Service (QoS) to expand.
• Quality of Service (QoS): QoS refers to priority of IP packets on network layer. It can be programmed for
both signaling (SIP) and media (Voice, Video and Fax). Configure the following types QoS:
• SIP DiffServe/ToS: The system sends all the SIP signaling messages with this QoS setting. This field
defines the priority bits for SIP messages. The Valid DiffServe range is from 00-63, default: 26
• Voice DiffServe/ToS: The system sends all the Voice packets with this QoS setting. This field defines
the priority bits for Voice packet. The Valid DiffServe range is from 00-63, default: 46
• Video DiffServe/ToS: The system sends all the Video packets with this QoS setting. This field defines
the priority bits for Video packet. The Valid DiffServe range is from 00-63, default: 46
• Fax DiffServe/ToS: The system sends all the Fax packets with this QoS setting. This field defines the
priority bits for Fax packet. The Valid DiffServe range is from 00-63, default: 46
QoS parameters are applicable for all packets (SIP/ Media) leaving both LAN and WAN port as well as
TCP connection.
Channel Reservation for SIP Trunks

Click Channel Reservation for SIP Trunks to expand.
• Channels Reserved for SIP Trunks: The Vocoder module supports up to 128 voice channels, which can
be used by SIP Extensions and SIP Trunks.
It may happen that SIP Extension users use most of the channels of the Vocoder module, leaving too few
or none for making/receiving SIP Trunk calls.
This can be avoided by reserving some voice channels exclusively for SIP Trunk calls.
Specify the minimum number of voice channels you want to reserve for SIP Trunk calls. By default, no
channel is reserved.

SIP 100rel
Click SIP 100rel to expand.
• SIP 100rel: This parameter is to be configured if you want to support reliable transmission of (SIP)
provisional responses. Enable 100rel by selecting the check box, if you want the Vocoder module to use
100rel for reliable transmission of SIP provisional responses and to use PRACK (Provisional
Acknowledgment). Default: Disabled.
SIP Over TCP

Click SIP Over TCP to expand.
• SIP Over TCP: SARVAM UCS supports transporting of SIP messages over User Datagram Protocol
(UDP), Transfer Control Protocol (TCP) as well as Transport Layer Security connection (TLS). Despite the
advantages that SIP over TCP and SIP over TLS offer, it is more common to use UDP to transport SIP
messages.
By default, SIP Over TCP is enabled. To be able to send SIP messages over TCP, you must configure
‘TCP’ as the ‘Default Transport for Outgoing Messages’ on the ‘SIP Trunk Parameters’ page.
If you do not want to transport SIP messages using TCP, clear the SIP Over TCP check box.
SIP Over TLS

Click SIP Over TLS to expand.
• SIP Over TLS: SARVAM UCS supports transporting of SIP messages over Transport Layer Security. TLS
offers secure SIP signaling.
By default, SIP Over TLS is enabled. To be able to send SIP messages over TLS, you must configure
‘TLS’ as the ‘Default Transport for Outgoing Messages’ on the ‘SIP Trunk Parameters’ page.
If you do not want SIP messages to be transported using TLS, clear the SIP Over TLS check box.

SIP/RTP Ports
Click SIP/RTP Ports to expand.
• SIP UDP Port: This port defines the port on which SARVAM UCS listens for SIP messages transported
over UDP. This port is also used as the source port for sending SIP messages to the remote peer. The
valid range for this port is 1025-65535. Default: 05060.
• SIP TCP Port: This port defines the port on which SARVAM UCS listens for SIP messages transported
over TCP. This port is also used as the source port for sending SIP messages to the remote peer. The valid
range for this port is 1025-65535. The default SIP TCP Port is 05060.
• SIP TLS Port: This port defines the port on which SARVAM UCS listens for SIP messages transported
over TLS. This port is also used as the source port for sending SIP messages to the remote peer. The valid
range for this port is 1025-65535. The default SIP TLS Port is 05061.
• RTP Listening Port: This port defines the port on which SARVAM UCS listens for RTP Packets. This port
is also used as the source port for sending RTP packets to the remote peer. The valid range for this port is
1025-65278. The default RTP Listening Port is 08000.
Timers
Click Timers to expand.
• SIP Invite Timer (sec): This is the time in seconds for which the system waits for a response from the
called party after ending INVITE message. This timer starts after sending INVITE message to the called
party and stops on receipt of the provisional response or the final response or when the user disconnects
the call. On expiry of the timer, the call process is terminated by the SARVAM UCS and an error tone is
played to the user. The range of the SIP Invite Timer is 010-180 seconds. Default: 30 seconds.
• SIP Provisional Timer (sec): This is the time in seconds for which the system waits for the final response
after receiving the provisional response from the called party. This timer starts on the receipt of the
provisional response from the called party and stops on receipt of the final response from the called party
or when the user disconnects the call. On expiry of the timer, the SARVAM UCS terminates the call
process and plays error tone to the user. The range of SIP provisional Timer is 010-180. Default: 60
seconds.
• General Request Timer (sec): The time in seconds for which the system waits for response for a
transaction request. This timer starts on the initiation of a transaction. This timer stops on receipt of a
response for the request. On expiry of the timer, the system clears the transaction. This timer is used for
Registration request, etc. The range of the General Request Timer is 10-60 seconds. Default: 20 seconds.

Advance Option
Click Advance Option to expand.
• Check User-Agent for routing call: If you enable this check box, SARVAM UCS will check if the call is
originated from a trunk or an extension.
By default, Check User-Agent for routing call is disabled.
Make sure the PBX connected on both the ends is of Matrix. Only then the Check User-Agent for routing
call parameter will be applicable.

Configuring SIP Trunks
SIP trunks may be Proxy or Non-Proxy. All SIP trunks are considered as Proxy trunks by default.
Regardless of whether a SIP Trunk is Proxy or Non-Proxy, it must be assigned to the LAN/WAN Port, which is to be
used for that SIP trunk and the SIP trunk must be enabled. VoIP Calls can be initiated after suitable configuration of
the SIP trunk number in the Outgoing Trunk Bundle Group.
For Proxy SIP Trunks, you must program the following parameters required for registration with the Proxy Server.
• Enable the SIP trunk.
• Program the SIP ID, registrar Server Address, Registrar Server Port, Authentication ID, Authentication
Password as provided by your ITSP.
• If your ITSP uses Outbound Proxy, Enable the Outbound Proxy for the SIP trunk and also program the
Outbound Proxy Server Address and Port as provided by your ITSP.
The maximum number of SIP trunks supported by SARVAM UCS is 99.
If you have installed a single VoIP module, you can configure all SIP trunks on the same module.
Configuring SIP Trunks using Jeeves

• Click SIP Trunk Parameters to open the page.
Configure the following SIP Trunk parameters for the SIP trunks:
• SIP Trunk No.: This non-editable field is the number of the SIP trunk. The SIP trunks are numbered
from 01 to 99.

• Enable SIP Trunk: This flag is for enabling or disabling the SIP trunk. To be able to make incoming and
outgoing calls from the SIP trunk, click to enable the SIP trunk. By default, the SIP trunk is disabled,
disallowing incoming and outgoing calls. You may disable SIP trunks that are not functioning.
• Name: You may assign a 'Name' to each SIP trunk to facilitate identification. Whenever there is an
incoming call without CLI on this port, the Name you have programmed will be displayed on the landing
extension, provided it is a DKP.
The Name assigned to the SIP trunk may consist of a maximum of 18 characters. The Name of the port
may be the name of the ITSP the SIP trunk is subscribed with (recommended).
• SIP ID: Enter the SIP User ID provided by the ITSP. SIP User ID is the ID that callers will use to call this
SIP trunk.
The SIP User ID may be a number or text for remote parties to call on the SIP trunk. For example, if
SIP URI provided by the ITSP is 12345@abc.com, enter 12345 in this field. SIP User ID may consist of
a maximum of 40 characters. All ASCII characters are allowed.
• SIP Trunk Mode: You may select SIP Trunk Mode as Proxy or Peer-to-Peer, according to your
requirement.
If you are using the services of an Internet Telephony Service Provider (ITSP), select Proxy to register
this SIP trunk with the ITSP.
If you are not using this service, select Peer-to-Peer.
• During Incoming Call Routing to select a SIP trunk, SARVAM UCS compares the SIP ID received in the
Request URI of the INVITE message with the SIP ID configured on the SIP Trunk.
If you do not want SARVAM UCS to check the SIP ID received in the Request URI of the INVITE
message, clear the Check SIP ID During Incoming Call check box. Default: Enabled.
• Treat Incoming calls as: If you select Peer-to-Peer as the SIP Trunk mode, you may configure the
trunk to Treat Incoming call as: Trunk or Station.
If you select Trunk, the incoming calls will be routed as per the Trunk Feature Template assigned to
the SIP Trunk.
If you select Station, the system will route the incoming call as follows:
• When only a number is received in the “To:” field of the INVITE message, SARVAM UCS will check
the number in the Closed User Group Table. If a match is found in the CUG table the call will be
routed as per the corresponding Outgoing Trunk Bundle Group.
• If the CUG Table is not configured or if no match is found for the number received in the “To:” field of
the INVITE message, the system will check if there is an extension number that matches with the
number received in the “To:” field of the INVITE message. If a match is found the call is routed to the
desired extension number.
• When a Trunk Access Code and a number is received in the “To:” field of the INVITE message, the
system will route the call as per the Outgoing Trunk Bundle Group assigned in the Station Basic
Feature Template of the SIP Trunk.

By default, Trunk is selected.
• If Station is selected as the option for Treat Incoming call as, the user will only be able to:
• Dial Flexible Numbers
• Dial Operator Code
• Dial Trunk Access Code for making outgoing calls
• Access the Global Directory
• Make calls within the Closed User Group
• Registrar Server Address: Enter the Proxy/Registrar Server Address. Both IPv4 and IPv6 addresses
are supported. The Server Address may be an IP Address or a Domain name, of maximum 40
characters
• Registrar Server Port: Enter the Registrar Server Listening Port. The valid rang is from 1025 to
65535. By default, 5060 is set as the Listening Port of the Registrar Server.
• Re-registration Time (sec): The Registrar Server deletes an entry of its client from its database on
expiry of a fixed timer, which is set by the Registrar Server. SARVAM UCS sends a registration request
before this Timer expires to remain registered on the server.
Enter the value of the Timer after which the system should send registration request to maintain
registration binding with the server. The valid range of this timer is from 00001- 65535. By default the
Timer is set to 3600 seconds.
• Registration Retry Time (sec): This Timer stands for the period between retries for registration. If the
registration attempt fails, SARVAM UCS sends the registration request on the expiry of this Timer
again. The system continues to send the registration request till it gets registered. The valid range of
this timer is from 00001- 65535. By default the Timer is set to 00010 seconds.
• Authentication User ID: Enter the Authentication ID provided by the ITSP for this SIP trunk. The
Authentication User ID may be a string of 40 characters (maximum), including ASCII characters.
• Authentication Password: Enter the Authentication Password provided by the ITSP for this SIP trunk.
The Authentication Password may be a string of 24 characters (maximum), including ASCII characters.
• Outbound Proxy: This parameter is relevant only if the ITSP has a SIP outbound server to handle
voice calls. If yes, program the following parameters:
• Enable: Select this check box to enable Outbound Proxy. By default the flag is disabled.
• Server Address: Enter the Outbound Proxy Server's address. Both IPv4 and IPv6 addresses are
supported. It may be an IP Address or Domain name. A maximum of 48 characters, including ASCII
characters are allowed.
• Server Port: Enter the Outbound Proxy Server's Listening Port. The valid range for this is 1025-
65535. By default the Server Port is 5060.
• Trusted IP Address/es: You must configure IP Address table to allow incoming calls from specific IP
addresses on this SIP Trunk. Both IPv4 and IPv6 addresses are supported.
If you select Peer to Peer as the SIP Trunk Mode and you do not configure this table, incoming calls on
this SIP Trunk will be rejected.

If you select Proxy as the SIP Trunk Mode and you do not configure this table, incoming calls will be
allowed only from the Registrar Server Address or Outbound Proxy Address. All other calls on this SIP
Trunk will be rejected.
To configure the IP Address table,
• Click IP Address Table.
If you have selected SIP Trunk mode as Peer to Peer, configure the following.

• Allow from all IP Addresses: Enable this check box, if you want to allow incoming calls on this SIP
Trunk from all IP Addresses. Default: Disabled.
• Apply Digest Authentication: Enable this check box, if you want to allow incoming calls from
callers only after the callers have authenticated themselves (with their User ID and Password). If the
caller does not enter valid credentials in two attempts, the system will reject the call.
If you have enabled Allow from all IP Addresses check box, Apply Digest Authentication check box
will be enabled automatically. You must configure the Digest Authentication table.
To configure the table, click the Apply Digest Authentication link. For detailed instructions, see
“Digest Authentication”.
• Consider Peer to Peer Table for Trusted IP Address: Enable this check box, if you also want to
allow incoming calls from the domain names or IP Addresses configured in the Peer to Peer table.
Default: Enabled. To know more about Peer to Peer table, see “Peer-to-Peer Calling”.
If you have selected SIP Trunk mode as Proxy or Peer to Peer, configure the Trusted IP Address table.
The first entry in the table will display the Proxy/Registrar Server Address:Port or Outbound Proxy
Address: Port as configured for this SIP Trunk (applicable only for Proxy Mode). For the Index numbers
1 to 10,
• Enter the IP Address and the corresponding Port from which you want to allow incoming calls.
Do not configure the port, if you want to allow incoming calls from all the ports for a particular IP
Address.
• Click Submit to save your entries and close the window.
• SIP Hardware Template: Assign a “SIP Hardware Template” to the SIP Trunk. The SIP Hardware
Template contains voice quality related features such as Voice Codec selection, Tx and Rx Gains,
Echo Cancellation, Jitter Buffer and, Fax-over-IP options and related parameters.
requirement and preferences. By default, Template number 01 assigned to all SIP Trunks. Template
number 01 is also assigned to all SIP Extensions.
First, check if the values in Template 01 fulfill the feature requirements of the SIP Trunks. Retain this
template, if it fulfills the feature requirements of all SIP Trunks and if the same features are to be
allowed to all SIP Trunks.
If different sets of SIP hardware features are to be allowed to different SIP Trunks, then prepare
separate SIP Hardware Templates and apply them on the SIP Trunks. To do this,
• Under ‘VoIP Configuration’, click the 'SIP Hardware Template' link.

• Customize Template number 03 and click 'Submit'.
• Now go back to the 'SIP Trunks' page.
• Enter the number of the Template you customized, Template 03 in the 'SIP Hardware Template' field
of the SIP Trunk on which you want to apply this template.

• Repeat the same steps to customize and assign a different SIP Hardware Template to another SIP
Trunk.
• Also, refer the topic “SIP Hardware Template” to know more about customizing the templates and
applying on the SIP Trunks.
• Trunk Feature Template: A Trunk Feature Template is a set of features like Time Table, Operator,
Auto Attendant, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR Storage, etc., that defines the
behavior of a Trunk. Apply a Trunk Feature Template to the SIP trunk. By default, Trunk Feature
Template 01 is applied on all SIP trunks as well as all other trunk types. Refer the topic “Trunk Feature
Template” to know more.
Click the 'Trunk Feature Template’ link to open the page. Check if the default Template 01 fulfills your
requirement for the SIP trunk.
If the default Template 01 does not fulfill your requirement, prepare another Trunk Feature
Template162, and enter the newly prepared Template number for the SIP trunk.
• Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the SIP
trunk.
Cost Factor is a number assigned to each trunk for identification. This number also serves as a
preference number for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same
preference must be assigned the same Cost Factor. Different trunk types can also be assigned the
same Cost Factor. These trunks are used for routing calls.
Assign a Cost Factor to the SIP trunk, for instance, 02 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '9' through the SIP trunk 01
only,
• You must first assign a Cost Factor (01-99) to SIP trunk 01, for example, 02.
• Click the 'Least Cost Routing - Number Based' link to open the page.
• Enter '9' in the 'Number' column, Cost Factor '02' as Preference 1, 2, 3 and 4.
• Click 'Submit' to save your settings.
All outgoing calls assigned Cost Factor trunk 02 will be made from SIP trunk 01.
• Simultaneous Calls: This parameter is for defining the number of simultaneous calls to be allowed on
the SIP trunk. SARVAM UCS supports up to 32 simultaneous calls. You can configure as many
simultaneous calls as supported on the SIP trunk by the ITSP.
If the ITSP supports less than 8 simultaneous calls on SIP Trunks, you must program this parameter
accordingly. However, if you ITSP supports more than 8 simultaneous calls, you need not change this
parameter.
• Click Submit to save your configuration settings.
162. The default template is applied on the ports of all trunk types supported by SARVAM UCS. Changes to the default template will be
applied on all trunk types also. So, you are advised to prepare a new template and apply it to the desired trunk types.

The above listed parameters fulfill the basic SIP trunk configuration requirements of most users. However, it is
anticipated that some users may desire other less commonly used parameters such as RCOC, Accept Anonymous
Calls on SIP trunks, Call Budget on SIP trunks, etc.
For such users, you may click the Advance button and configure the following parameters:
• Return Call to Original Caller (RCOC): Enable this flag if you want to apply the 'Return Call to Original
Caller' on this SIP trunk.
If this feature is enabled on the SIP trunk, the system will route calls returned by remote parties back to the
extensions that originally made the call from this Trunk (the original callers' extensions). To know more,
refer the feature description for “RCOC (Return Call to Original Caller)”.
• Station Basic Feature Template: Assign a “Station Basic Feature Template” to the SIP trunk. There are
50 different templates to choose from. Each template can also be altered to suit your requirement and
preferences. Default: Template number 01 assigned to all SIP trunks.
• Station Advanced Feature Template: Assign a “Station Advanced Feature Template” to the SIP trunk.
requirement and preferences. Default: Template number 01 assigned to all SIP trunks.
• Add 'rinstance' in Register: 'rinstance' is any random value which can be used by the WAN Port to fetch
its own contact binding, that is, to know the Registration Expiry Timer assigned by the server. By default,
this flag is enabled.
• Send REGISTER Message: With this parameter you can select whether or not the system should send
REGISTER message from the SIP trunk. By default, this flag is enabled allowing REGISTER message to
be sent from the SIP trunk.
• Allow OG Calls without Registration: This parameter is to be enabled to allow outgoing calls to be made
from the SIP trunk, even when the SIP trunk is not registered. If this flag is disabled the system will not

allow outgoing calls to be made if the status of the SIP trunk is 'not registered'. By default, this flag is
disabled.
• Send OPTIONS as Heartbeat: With this parameter you can select whether or not the system should send
the OPTIONS message periodically to the Proxy Server to monitor its availability. Calls can be made and
received only if the Proxy Server is alive.
If the Proxy Server is unavailable, like no response is received, the status of the SIP Trunk will display
“Heartbeat Failed” along with the Reason for Failure.
To view status of the Proxy Server, go to SIP Trunk Status.
If you enable Send OPTIONS as Heartbeat, you must configure the Heartbeat Interval.
• Heartbeat Interval: Define the Heartbeat Interval (Seconds), the time period, from 10 to 999
seconds, after which SARVAM UCS should send the OPTIONS message to the Proxy Server. Default:
30 seconds.
• DNS Record Type: Set this parameter as provided to you by your Service Provider. You may select any of
the following options:
• A/AAAA Record: Select this option, if you want SARVAM UCS to send the query to the DNS Server to
fetch the IP Address of the Target Server on which further SIP messages are to be sent.
A Record maps to an IPv4 Address of the Target Server, while AAAA (quad-A) Record maps to an IPv6
Address.
• SRV Record: Select this option, if you want SARVAM UCS to send the SRV query to the DNS Server
to fetch the Destination Port. The system will also make a DNS A query to fetch the IP Address of the
Target Server on which further SIP messages are to be sent.
• NAPTR/SRV Record: Select this option, if you want SARVAM UCS to send queries to the DNS Server
to fetch the Transport, Port and IP Address of the Target Server on which further SIP messages are to
be sent.
By default, A Record is set as the DNS Record Type.
This parameter will be checked when:

• SIP Trunk Mode is configured as Proxy.
• Registrar Server Address is configured as Domain and Outbound Proxy Address is blank OR Registrar
Server Address is configured as Domain/IP Address and Outbound Proxy Address is configured as
Domain.
• Send CLI in FROM field: This parameter allows you to configure the CLI of the SIP Trunk to be sent to the
remote party on outgoing calls made using the SIP trunk. You may select any of the following options as
desired:
• CLIR: Select this option if you do not want the CLI to be sent.
• SIP ID: You may select this option if you want the SIP ID programmed on the SIP Trunk to be sent as
CLI.
• Calling Party Wise: Select this option if you want to send the Calling Extension Number (the number
of the extension making the outgoing call through the SIP trunk) as CLI.

When reverse DDI is programmed on the SIP Trunk, the DDI number of the calling extension will be
sent, instead of its extension number.
If the calling extension has disabled the parameter ‘Send DDI as CLI’ in its Station Advanced Feature
Template, then its Pilot number configured in the Outgoing Reference Table will be sent as CLI.
If calling extension has enabled CLIR, no CLI will be sent on the SIP Trunk.
• Fixed Number: Select this option if you want a specific number to be sent as CLI. When you select this
option, you must also define the number to be sent as CLI.
You may select this option if you wish to send any of your trunk line numbers as CLI on the SIP Trunk
so as to enable the called party to call back the calling party using this CLI.
Since it is not possible to call back a SIP ID, Fixed Number offers you a solution, using which you can
send a trunk line number as CLI on the SIP Trunk. Using this CLI, the called party can call back the
calling party.
If you select this option, you must configure the Fixed Number. The Fixed Number may consist of a
maximum of 40 characters. Valid characters are from 0-9 and plus (+).
By default, SIP ID is set as the Send CLI in FROM field for all SIP Trunks.
When extension number of the calling extension is blank, and the ‘Send CLI in FROM field’ programmed
for the SIP Trunk is other than "SIP ID", then also SIP ID will be sent as CLI.
When Emergency numbers are dialed using the SIP Trunk and even if CLIR is set as the ‘Send CLI in
FROM field’ option, the system will send the number of the caller as CLI.
• Use “tel” URI type in: If your ITSP requires “tel” in the URI so that it can handle/ route calls to/from global
numbers, select the desired option from any of the following:
• TO, Request URI
• FROM
• TO, Request URI and FROM
The system will send 'tel' URI in the selection headers and Request URI according to the selection you
make. This will be used while making an Outgoing call.
• Send "user=phone": Select this check box, if you want SARVAM UCS to add user=phone in the Request
URI/FROM/TO header of the INVITE message. Default: Disabled.
SARVAM UCS will send user=phone, only if the SIP ID is numeric.
• P-Asserted-Identity/P-Preferred-Identity for Outgoing calls: Configure the following P-Asserted-

Identity/P-Preferred-Identity for Outgoing calls parameters as provided to you by your Service
Provider.
• Select Add P-Asserted-Identity/P-Preferred-Identity check box, if supported by your Service

Provider.
• Select P-Preferred-Identity or P-Asserted-Identity as Identity.

• Select the desired Caller ID option:
• SIP ID: If you want the SIP ID configured on the SIP Trunk to be displayed as the Caller ID.
• Calling Party Wise: If you want the Calling Extension Number (the number of the extension
making the outgoing call through the SIP trunk) to be displayed as the Caller ID.
• Fixed Number: If you want a specific number to be sent as the Caller ID. When you select this
option, you must also define the number to be sent as the Caller ID.
The Fixed Number may consist of a maximum of 40 characters. Valid characters are from 0-9 and
plus (+) .
By default P-Asserted-Identity/P-Preferred-Identity for Outgoing calls is disabled.
• P-Asserted-Identity/P-Preferred-Identity for Incoming calls: Configure the following P-Asserted-

Identity/P-Preferred-Identity for Incoming calls parameters as provided to you by your Service
Provider.
• Select Get CLI from P-Asserted-Identity/P-Preferred-Identity check box, if supported by your

Service Provider.
• Select P-Preferred-Identity or P-Asserted-Identity as the Preference.
• By default Handle Privacy Header is enabled. If the system receives Privacy Header as:
• Privacy: ID, anonymous will be displayed as the CLI to the called extension.
• Privacy: None, the CLI received from the Service Provider will be displayed to the called extension.
If No Privacy header is received the CLI received from the Service Provider will be displayed to the
called extension.
• If you do not want the system to handle the Privacy Header, clear the check box.
By default P-Asserted-Identity/P-Preferred-Identity for Incoming calls is disabled.
• Redirection: When an incoming call is diverted/forwarded to an external number using the same SIP
Trunk, configure the following parameters as required by your ITSP:
• Select the desired Call Redirection Type. You may select Generate new Call or Send 302. Default:
Generate new Call.
• If you want the system to send 181 response to the source upon redirection, enable the Sending of
181 responses upon Redirection check box. Default: Disabled.
• If you have selected Generate new Call as the Call Redirection Type, you may select the Include
"Diversion" while diverting the call check box. The system includes incoming call request
information in Diversion header, along with the reason of redirection. Default: Disabled.
• If you have selected Generate new Call as the Call Redirection Type, you may select the Include
"History-Info" while diverting the call check box. The system includes the incoming call request

information as well as the new call information in the INVITE/REINVITE requests generated on the
target. Default: Disabled.
• If you have selected Generate new Call as the Call Redirection Type, select the desired option in
CLI while Diversion/Redirection. You may select Self or Received. Default: Self.
If you select Self, the system will send SIP Trunks' Identity and if you select Received the system will
send the received CLI to the target.
• Call Transfer Type: Select the Call Transfer Type as supported by your ITSP. System will send the Call
Transfer request to the transferee only if calls are routed through the same SIP Trunk. You may select:
• System: In this option, the SARVAM UCS will handle the Call Transfer locally.
• Network: In this option, the Call Transfer is handled by the Network.
Default: System.
• Accept Anonymous Calls?: The option is for accepting calls without CLI that land on the SIP trunk. By
default the flag is enabled. You may disable this flag to disallow calls without CLI on this SIP trunk port.
• Source Port IP Address: Select the Source Port IP Address for the SIP trunk. You may select from any of
the following options, as applicable to your installation scenario.
• Use VoIP Ethernet Port IP Address: Select this option, if the WAN Port is connected directly to the
public Internet.
• Use IP Address fetched using STUN: Select this option, if the WAN Port is located behind a NAT
router other than Symmetric.
• Use Router's Public IP Address: Select this option if the WAN Port is located behind a NAT router
(any type).
• Handle rport: Select the desired option from:
• Force NAT: By default, this option is selected. The system will not check Contact/Via header etc. while
sending SIP messages and will follow Symmetric Signaling.
• RFC 3581: Select this option, if you want the system to follow Standard RFC’s while sending SIP
messages.
Default: Force NAT
• Use Symmetric RTP?: The use of Symmetric RTP makes it possible for a SIP device to send the RTP on
the same connection on which it is listening for RTP. This is done only on peer to peer SIP trunks.
Enable this flag, if the WAN Port is located on a public IP and you want outgoing calls to the SIP Client
located behind the NAT Router. OR if you need to receive incoming calls from the SIP Client located
behind the NAT router. By default, Use of Symmetric RTP is disabled.
• Default Transport for Outgoing Message: Configure this parameter for Proxy SIP Trunks only. The
SARVAM UCS supports three options for transporting outgoing SIP messages:

• UDP: Outgoing messages are transported using UDP. By default this option is selected.
• TCP: Outgoing messages are transported using TCP. If you select this option, you must enable ‘SIP
Over TCP’ on the ‘VoIP Parameters’ page.
• TLS: Outgoing messages are transported using TLS. If you select this option, you must enable ‘SIP
Over TLS’ on the ‘VoIP Parameters’ page.
By default, UDP is selected as the Default Transport for Outgoing Message.
The Default Transport for Outgoing Message options are checked only if you have enabled SIP over TCP
or SIP over TLS.
If the SIP over TCP and SIP over TLS are disabled, all outgoing SIP messages will be transported over
UDP only.
• SRTP Mode: SARVAM UCS supports SRTP (Secure Real Time Protocol) for secure conversations over
SIP. The SARVAM UCS supports the following options:
made to such users.
• Call Budget: If you want to enable Call Budget on Trunk feature on this SIP trunk, configure the
following parameters:
• Type: Select the type of Call Budget on Trunk—Amount or Minutes or Calls—to be applied on this SIP
trunk. By default, no Call Budget type is selected.
• Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field. By
default the Amount is set to 99999.
• Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this field. By
default the number of minutes is set to 99999.
• Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By default
the number of calls is set to 9999.
• Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to be
reset on a particular date of every month.
• Scheduled (Date): Select the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan suggests
so.

The consumed Call Budget Amount/Minutes/Number of Calls can be reset also using SA and SE
commands, referred to as Manual Reset. Refer the feature description “Call Budget on Trunk”.
• Call Back: This parameter is related to the “Call Back on Trunk Port’ feature. If you want to enable the
'Call Back on Trunk Port' feature on this SIP trunk, configure the following parameters:
• Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this flag
is disabled on all trunk port types. By default, the flag is disabled.
• Call Back Timer: This is the duration for which the system waits for the caller to disconnect before
applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to 10
seconds for all SIP Trunks.
• Call Back Mode: Select from the following options how a ‘Call Back’ call answered by the remote party
should be routed:
• Built-In Auto Attendant
• PIN Authentication - Multiple Calls
• CLI Authentication - Multiple Calls
• CLI Authentication - Single Call - Answer Signaling
• Operator
By default, Operator is selected as the Call Back Mode.
• Call Back on: This parameter allows you to select if the call back should be made to the same number
that was received or to a different number. If you want the call back to be made to the same number
select the ‘CLI number”. If you want the call back to be made to a different number, select ‘Alternate
Number’.
By default, CLI number is selected for Call Back.
• Incoming Number List: Program the number strings that are eligible for Call Back in this List. By
default, Number List 15 is assigned to Call Back Incoming Number List.
Number List 15 is also assigned to all SIP trunks as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all SIP trunks and Trunk Port types, retain this
list.
If you want a different set of number strings to be programmed for this SIP Trunk, select a different
Number List, and assign it to the SIP trunk port.
You may program the Incoming Number List either from the ‘Number List’ page or by clicking the
‘Incoming Number List’ link to reach the Number List page.
Refer the topic “Number List” to know more, and for configuration instructions.
• Outgoing Number List: Program the number strings that are to be called back in this List. For each
number string you programmed in the ‘Incoming Number List’, you must program in the corresponding
index in the Outgoing Number List a number to which the call back is to be made. For example, for the
number string programmed at Index 1 in the Incoming Number List, a corresponding number string at
the same Index, Index 1, should be programmed in the ‘Outgoing Number List’.
By default, Number List 16 is assigned to Outgoing Number List.The same Number List 16 is also
assigned to all SIP trunks as well as all other Trunk port types.

You may program the default number list, or a different number list and assign it to this SIP Trunk port.
You may program the Outgoing Number List either from the ‘Number List’ page or by clicking the
‘Outgoing Number List’ link to reach the Number List page.
• Call Back from: This parameter determines the trunk port to be used to make the call back. The call
back can be made using the same port or an Outgoing Trunk Bundle Group (OGTBG).
Select ‘Same port’ if you want the call back to be made using the same port on which the missed call is
received. If you select OGTBG, the call back will be made using the OGTBG, which you have defined.
By default, Same port is selected.
• OGTB Group: If you selected OGTBG for making the call back in the previous parameter, you must
define the OGTBG that must be used in this parameter.
By default, OGTBG 01 is assigned to all SIP trunks.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost
Routing on the OGTBG that you define here for Call Back.
• Incoming (IC) Reference ID: Assign an Incoming Reference ID for the SIP trunk for Working Hours,
Break Hours, Non-Working Hours. By default, 00 is assigned as Incoming Reference ID for all three time
zones.
• Outgoing (OG) Reference ID: Assign an Outgoing Reference ID for the SIP Trunk, from 00 to 99. By
default, 00 is assigned as Outgoing Reference ID.
• Pause Timer (sec): This Timer is required for inserting delay while digits of a number string are out dialed
from the SIP trunk. The Pause Timer will be applicable when the letter 'P' is configured in the DTMF
number string which is to be out dialed as DTMF digits on the SIP trunk. The range of this timer is from 1 to
9 seconds. By default the Timer is set to 3 seconds.
For example, if 'PPP3' is to be out dialed and Pause timer is programmed as 3 seconds, the SARVAM
UCS will out dial the digit 3 after 9 seconds, after a delay of individual P (3+3+3 =9). The range of this
Timer is from 1 to 9.
This parameter is used for the “Multi-Stage Dialing” feature.
• DTMF ON Timer: This is the time for which the DTMF digit will remain ON, while being out dialed by the
SARVAM UCS. This parameter finds its application in the feature “Multi-Stage Dialing”. The range of this
timer is from 051 to 255 milliseconds. By default, ON Time is defined as 102 msec.
• DTMF Inter Digit Pause (msec): This is the time for which the SARVAM UCS will wait before dialing the
successive digits.
• Gateway Application - Answer Signaling: This parameter is to be programmed if the SIP trunk is being
used in a gateway application as a source port (from where calls originate). The calls originated on the
source port (SIP trunk) are routed using another Trunk port, the terminating port, which may be any trunk
port, like T1E1. When call made from the terminating port gets matured, this is signaled to the source port
in the form of DTMF digits.

• Use?: Enable this flag if you want the SIP trunk to be used in a Gateway Application.
• DTMF String: Program the DTMF digits to be sent to signal call maturity to the source port.
• Send Re-INVITE over SIP Trunk on Hold: With this parameter you can select whether or not the system
should send Re-INVITE message from the SIP Trunk to the Remote Peer, when an external call over the
SIP Trunk is put on hold by the extension user. Set this parameter as per the requirement of the Remote
Peer. The Remote Peer can be a Proxy Server or a SIP Device.
By default, Send Re-Invite over SIP Trunk on Hold is disabled. For more information see “Call Hold”.
• Delayed Offer: Enable this flag, if you want the SIP Trunk to generate INVITE without SDP. Default:
Disabled.
• Fetch Called Party Number From: SARVAM UCS extensions may be assigned DDI numbers provided
by the ITSP. When INVITE is received from ITSP, the ITSP may send the DDI number either in the
“Request URI” of the INVITE message or in the “To:” field of the INVITE message.
By default, Request URI is selected. Ask your ITSP if you need to change this parameter.
• Called Party Number as CLI: If you want SARVAM UCS to display the called party number received in
the INVITE message as the CLI, select the Display Called Party Number as CLI check box. By default,
Display Called Party Number as CLI option is disabled.
This parameter is useful when a single SIP Trunk having DDI Numbers and Operator are shared by more
than one organization. If you enable this option, make sure:
• you configure the names and corresponding numbers of the organizations sharing the SIP Trunk in the
Global Directory of SARVAM UCS.
• the Operator has a DKP or an Extended IP Phone or a Mobile UC Client.
With this option enabled the Operator will be able to handle calls more efficiently. When there is an
incoming call, SARVAM UCS matches the number with the numbers in the Global Directory. If a match is
found SARVAM UCS displays the company name configured for that entry to the Operator, that is, the CLI
will display the called party number and name.
After the Operator answers the call, the CLI will change and display the calling party number and name (if
configured in the Global Directory).
If you keep this option disabled, the calling party number and name will be displayed as the CLI, both
during an incoming call and after the call is answered by the Operator.
You can configure the Display Called Party Number as CLI option only from Jeeves.
• On Connecting Media Send: If Built-In Auto Attendant or DISA is enabled on the SIP Trunk, select the
response you want the system to send on Connecting Media, that is 200 OK or 183 Session Progress.
• Answer Source Trunk on Receiving: When a call received on any trunk of SARVAM UCS is routed
through the SIP Trunk, select the response after which the call on the source trunk must be answered. You
can select Early Media or 200 OK.
• If you have configured the above parameters, click Submit at the bottom of the page to save your
configuration settings.

Viewing SIP Trunk Status
You can also view the settings of the SIP Trunk Parameters on Jeeves.
To do this,
• Under VoIP Configuration, click the SIP Trunk Status link.
• For each SIP trunk (number), the following settings will be displayed:
• SIP Trunk Number
• Status
• Registration Time
• Registration Retry Count
• Reason for Failure
• Call Budget Type
• Allotted Amount/Minutes/Calls
• Consumed Amount/Minutes/Calls
• Scheduled Reset
• Budget Reset Scheduled (Date)
• Reset Consumed Budget (this is not a status indicator. It is for resetting the Consumed Call Budget
manually)
You can also view the SIP Trunk Status from the Status link. To view, click the SIP Trunk link under Status.

Configuring PRI Trunks
What’s this?
• This topic explains the connection of PRI line, and application of PRI.
• For MSN Numbering System, please refer “Configuring BRI Trunks”.
• PRI port of the SARVAM UCS configured for NT mode can be connected to the PRI Port of another
System configured for TE mode. In such case, the SARVAM UCS will behave as Transit Exchange.
• Dialing method on the PRI port is not programmable. The PRI port (configured for TE mode) will send the
called party number in Enbloc mode.
• All the switch variants are applicable to the PRI port whether programmed in TE or NT mode.
Applications:
Applications for PRI-NT Port are as described below:
• Video conferencing system connected to the PRI-NT port with IC/OG calls.
• Data Calls support.
• Networking of Systems.
Video conferencing system connected to the PRI-NT port:
SARVAM UCS
ISDN PRI TE
Video
Conferencing
PRI NT System
• Connect the Video Conference system (H.320) to the PRI-NT port of the SARVAM UCS.
• This feature works only if it is supported by the Service Provider. Video Conference is established mainly
by the Video Conferencing (VC) equipment.
• Video Conferencing requires 6 B-channels. But Video Conferencing can also be done at lower bit rates
also using the “aggregation” of 6 B channels which must be supported by the VC equipment.
• The VC equipment uses two methods:H.221 and H.242 BONDING, for aggregation.
• More than one Video conferencing system can be connected to the SARVAM UCS to each PRI-NT port.
• It also supports internal calling like, one Video Conferencing system to another.

OG calls from VC
• After suitable connections to the VC equipment, user follows following sequence:
• Program the OGTBG (OG Trunk Bundle Group) such that the user gets at least 6 B-channels of the same
PRI port.
• Go OFF-Hook.
• Dial the trunk access code to grab the PRI trunk.
• Dial the destination number.
• If VC at the called party responds, the call is established which occupies 6-B Channels.
• If 6 B-channels on the same PRI port are not available, user at VC will get busy tone.
• The user at calling party’s VC gets dial tone of the ISDN exchange.
• The user at VC starts dialing the destination number. The destination number is sent in keypad IE by the
VC user to the PRI-NT port whereas it is dialed on the PRI-TE port in the method programmed for the port
that is, Enbloc.
• Rest of the signaling is done between the VC equipment and the called party’s VC equipment.
• If VC equipment supports Phone Book feature, user can make call using the feature.
• At the end of VC, all the 6 B-channels are freed and available for other users.
• It is preferable that the SE will assign an OGTBG to the Video Conferencing equipment in such a manner
that the same group is not assigned to any other station. This is to allow Video Conferencing call at any
time.
IC Video Call:
• Following steps are followed for IC call:
• Program the system to place IC video calls to the PRI-NT port using DDI Routing Table.
• For this, program the DDI Routing Table. Refer related chapter.
• You can also prepare the Trunk Landing Group (Routing Group) to land the call to PRI-NT port. For this
purpose, Routing Group is to be modified.
Data Calls from PRI-NT:

• SARVAM UCS PRI-NT port supports data communication also.
• For data-communication, connect the Router supporting ISDN-PRI or a Computer with SARVAM UCS
Card T1E1PRI to the PRI-NT port as shown below:

SARVAM UCS
ISDN PRI TE
PRI NT Hub
Router
Using this feature following applications are supported:
• The computers connected in the LAN can browse the net through the PRI.
• Remote LAN Access the Computers in the LAN can access the computer/computers in LAN at the remote
end (Branch office/Home office).
• Files can be transferred from one LAN to another.
OG Data calls from the Router

• The data call can be made by the router requesting desired number of channels. This establishes a live
connection between the Router and the ISP through the System. The users on the LAN can browse the
net as normal using Internet Explorer or Netscape Navigator.
• For this, the SARVAM UCS will allocate data channels only on the PRI-TE port so as to leave other
channels for speech calls when the system detects the call to be a data call.
• Similarly, a Remote Computer can be accessed (Remote LAN Access) by dialing the Remote users’
number (The remote end System should be so programmed that the call made to a number lands directly
on the Router.) This establishes a permanent connection between the two Routers (and hence two LAN
networks).
• Now the user at System-A can access the computer in LAN at the remote end in the same way as
accessing another computer on the same network.
IC Data calls to the Router:

• Program the Trunk Landing Group such that all data calls will land on the PRI-NT port to which the Router
is connected using CLI based Routing logic or DDI based routing logic.
• However, while routing call on the PRI-NT port, the System will check that the data channels reserved for
data communication on the PRI-NT port are enough to establish the call. Otherwise the call will be
rejected.
• The call will be rejected if the number of channels, reserved for data calls are already busy with one data-
call.

Networking of Systems:
One of the applications is also for connecting multiple Systems using PRI-NT and PRI-TE ports. A simple
connection of only two Systems is explained below. Refer following figure for connecting another System to the
PRI-NT port.
2001
System-A System-B 3001
ISDN PRI TE ISDN PRI TE
PRI NT PRI TE
Various cases of Calls from System-A to System-B can be:
Case-1: (2001 calls 3001)
• A station 2001 of System-A can call a station 3001 of System-B by dialing 3001.
• Such a call can be routed using CUG table of System-A. Refer chapter “Closed User Group (CUG)”.
Case-2: (2001 calls a subscriber of ISDN network connected to System-B)
Method 1
• Program (for System-A) an OGTBG containing the PRI-NT port and assign Trunk access code to it.
• Grab the PRI-NT by dialing access code. User of 2001 will get dummy dial tone of the System-A.
• Dial a station number 3001 of System-B.
Method 2
• User of 2001 can dial trunk access code of System-A, and dial a trunk access code of System-B. He will
be connected to ISDN network through System-B.
• Then he can make OG call on PRI-NT port as per method programmed for the PRI-NT port. (For example,
PRI-NT port can be programmed as a Trunk port and when station user of System-B grabs the PRI-NT
port, his call will be placed on the destination programmed for PRI-NT port in the Trunk Landing Group).
Case-3: (3001 to 2001)
Station 3001 of System-B will go Off-Hook.
• Using CUG feature of System-B, 3001 can call 2001 through PRI-NT port. Refer chapter “Closed User
Group (CUG)” for more details.
• 3001 can also dial Trunk Access Code to grab its PRI-TE port. He will get dial tone of PRI-NT Port of
System-A. Now 3001 can dial 2001 or trunk access code to make a call to the ISDN network connected to
System-A.

How to configure
For programming of the PRI, please refer topic “Configuring E1 Trunks” and “Configuring T1 Trunks” for more
details.
• Trunk software ports are automatically assigned to the PRI port by the system depending on the slot in
which they are inserted.
• Please take care to terminate the PRI line on the HDSL interface only of the ISDN modem.
• The DDI Routing and Routing Tables shall be programmed as explained in chapters on DDI.

Configuring T1 Trunks
What’s this?
Digital Signal Level 1 (T1E1) trunks use Bit-Oriented Signaling (BOS) and multiplexes 24 channels (T1 service) or
32 channels (E1 service) into a single data stream. T1E1 can be used for voice or voice-grade data and for data-
transmission protocols. T1 trunk service multiplexes 24 channels into a single 1.544-Mbps data stream. E1 trunk
service multiplexes 32 channels into a single 2.048-Mbps stream. Both T1 and E1 provide a digital interface for
trunk groups.
Signaling Modes
Common Channel Signaling (CCS) is an industry-standard technique where any one of a group of channels carries
the signals for the other channels. Matrix uses the 24th channel of a group for signaling. This signaling technique
differs from 24-channel signaling. When the system is configured for Facility-Associated Signaling, 24-channel
signaling uses the 24th channel in a T1E1 facility to carry signals. This technique also is called clear channel, out-
of-band or alternate voice data (AVD) signaling.
ROBBED-BIT signaling is a per-channel in-band signaling technique for transmitting signaling bits on each channel
in a T1E1 facility. The least-significant bit in every 6th transmitted information frame is removed and replaced by a
signaling bit. This technique is also called in-band signaling. The maximum transmission rate for each bearer
channel with ROBBED-BIT signaling is 56 Kbps. T1 RBS Protocol supports 24 channels (from 01 to 24) and the
protocol doesn’t consume any channel for signaling so that there are total 24 channels available for the users.
ISDN-PRI signaling is carried on the 24th channel for a 1.544 Mbps (T1) connection and on the 16th channel for a
2.048 Mbps (E1) connection. In case of T1 PRI Protocol supports 24 channels (from 01 to 24), in which channel no.
24 is used for the signaling, so effectively there are 23 Voice channels are available.
QSIG is an ISDN based protocol for signaling between nodes of a Private Integrated Services network.
Any of the common trunks, except for PCOL (Personal Central Office Line) trunks, can be analog or digital. (PCOL
trunks can only be analog.) Administering a digital trunk group is very similar to administering its analog
counterpart, but digital trunks must connect to a T1E1 port and this port must be administered separately.
Configuring T1E1 Port Parameters using Jeeves

• Under Configuration, click T1E1 Configuration.
• Click Port Parameters to open the page.

• Select the T1E1 Port number you want to configure by clicking the respective tab, and program the
following port parameters:
• SARVAM UCS will assign the Hardware Slot-Port automatically, when any card is inserted in the
system.
Hardware slot is the number of the Universal slot of SARVAM UCS in which the T1E1 Card is inserted.
Range of slot number is 01-16. Port is the number of T1E1 hardware port on the card to which the
T1E1 line is connected. Range of Port is from 00-99.
If you want to de-assign the Hardware Slot and Port, Enter '00' in both fields. By default, Hardware slot-
Port is 00–00.
• Keep the Enable T1E1 Port flag enabled.
Clear the flag, only if you do not want to use this port. By default, it is enabled.
• You may assign a Name to the T1E1 Port for identification of the port. The Name may consist of a
maximum of 12 characters. By default, it is blank.
• Select the Carrier type as T1. The Carrier type will automatically be assigned to the port when you
select the region. You may it if required.
• Select Signal Type. Signal Type signifies the type of signaling to be used on E1 line. The E1 Signaling
supported are:
• PRI
• RBS
• QSIG
• E&M
Default: PRI

If you select PRI or QSIG as the Signal Type, configure the PRI/QSIG Parameters.
If you select RBS as the Signal Type, configure the RBS Parameters.
If you select E&M as the Signal Type, configure the E&M Parameters.
• Set the Orientation Type. Select the orientation type from the following options:
• Terminal
• Network
• Tie-Line
Default: Terminal
• If you have set Terminal as the Orientation Type, you must select the type of network with which it is
to be Interfaced With. The network may be:
• Public ISDN
• Private ISDN
Default: Public ISDN
• You may configure the port to Treat Incoming call as Trunk or Station.
If you select Trunk, the system will treat all incoming calls as external calls landing on the trunk. The
calls will be routed as per the Trunk Feature Template assigned to the T1E1 Port.
If you select Station, you must also assign a Station Basic Feature Template and Station Advanced
Feature Template to the T1E1 Port.
When you select Station, the system will treat the calling party as an extension user. The user will have
access to all the features and facilities of the system, as per the Station Basic Feature Template and
Station Advanced Feature Template assigned to the T1E1 Port.
• If Point-to-Point is selected as the Interface Type, you can select the option Trunk or Station for the
parameter Treat Incoming call as.
• If Point-to-Multipoint is selected as the Interface Type, only Station can be set as the option for the
• Line coding is a pattern that data assumes as it is propagated over a communication channel. Select
the Line Coding Mechanism from the following:
• AMI-Basic
• B8ZS
• NRZ (Fiber Optic)
Default: AMI-Basic.

• Framing means to form a set of 24 or 32, 8 bits time slot that is to be treated as single transmission
unit. The Framing Modes supported by SARVAM UCS are:
• SF (D4)
• ESF
Default: SF (D4)
• Line Buildout Parameter field reduces the outgoing signal strength by a fixed amount. The
appropriate level of loss depends on the distance between your switch (measured by cable length from
the smart jack) and the nearest repeater. Where another switch is at the end of the circuit, as in campus
environments, use the cable length between the 2 switches to select the appropriate setting from the
list. This field is relevant if the Near-end CSU type field is integrated. By default, 0-133ft is selected as
the Line Buildout Parameter.
• Configure Outgoing (OG) Reference ID. By default, OG Reference ID is 00.
• Configure Incoming (IC) Reference ID for working hours, non-working hours and break hours. By
default, IC Reference ID is 00.
• Assign Trunk Feature Template to the T1E1 Port. Trunk Feature Template is a set of general features
that define the behavior of a Trunk Port. By default, Template 01 is assigned to all T1E1 Ports.
For more details, see “Trunk Feature Template”.
• Station Basic Feature Template assigned to the T1E1 Port is displayed in this field. Station Basic
Feature Template is a set of general features that define the basic behavior of a station. By default,
Template 01 is assigned to all T1E1 Ports.
For more details, see “Station Basic Feature Template”.
• Station Advanced Feature Template assigned to the T1E1 Port is displayed in this field. Station
Advanced Feature Template is a set of advanced features, to be applied on extensions such as CLIP,
Floor Service, Walk-in Class of Service. By default, Template 01 is assigned to all T1E1 Ports.
For more details, see “Station Advanced Feature Template”.
• Select Priority for the T1E1 Port. Priority is the precedence given to certain trunks and extensions over
others in being answered by the destination extension. You can select from 1 to 9. By default, Priority 5-
Normal is set for all T1E1Ports.
For know more about Priority feature, see “Priority”.
• Assign a Cost Factor to the T1E1 Port. By default, all the T1E1 Ports are assigned Cost Factor 01.
For more details, see “Cost Factor”.
• ISDN glare occurs if the system initiates an outgoing call on a B-Channel at the same time the network
initiates an incoming call on that same B-channel. You may configure the Glare Option as Proceed or
Held Back. While processing a glare condition, the configured Glare Option on T1E1 port will be
considered.
• Select Category (Logical Partition) for the T1E1 Port. You may select from the following options:
• 1
• 2
• 3

By default, 1 is selected for all T1E1 Ports.
• Idle Code is the 8-bit sequence that occupies the time slot on a E1/T1 trunk channel when it is not
being used. By default, 127 is configured as the Idle Code.
• If you want SARVAM UCS to display the called party number as the CLI for incoming calls, select the
Display Called Party Number as CLI check box. By default, Display Called Party Number as CLI
option is disabled.
This option is useful when a single T1E1 line connection and Operator are shared by more than one
organization. If you enable this option, make sure:
• you configure the names and corresponding numbers of the organizations sharing the line in the
incoming call, SARVAM UCS matches the number with the numbers in the Global Directory. If a match
is found SARVAM UCS displays the company name configured for that entry to the Operator, that is,
the CLI will display the called party number and name.
After the Operator answers the call, the CLI will change and display the calling party number and name
(if configured in the Global Directory).
• Select Return Call to Original Caller (RCOC) flag to enable this feature on the T1E1 Port. By default,
RCOC flag is disabled.
For know more about RCOC feature, see “RCOC (Return Call to Original Caller)”.
• In Channel Reserved for Data Call, configure the channels you want to reserve for Data Calls. By
default, 00 channels are reserved.
• In Channel Reserved for Outgoing Call, configure the channels you want to reserve for making
outgoing calls. By default, 30 channels are reserved.
• In Channel Reserved for Incoming Call, configure the channels you want to reserve for receiving
incoming calls. By default, 30 channels are reserved.
• When a caller dials the trunk access code or selective trunk access code for dialing the number directly
on the trunk port, the caller waits for the dial tone before dialing the number. But some exchanges do
not give Dial Tone for the T1E1 Port. For Example, when T1E1 port as E1CAS type is used in Delhi, it
is observed that the exchange does not give dial tone when direct dialing on the trunk is used.
Enable the Feed Dial Tone flag. SARVAM UCS will provides the dial tone to the caller when the T1E1
Port is accessed.
It is applicable only when Online dialing is used as, Store and Forward dialing, the dial tone is given to
the caller. The dial tone is played as per the Dial Tone Timer.

By default, Feed Dial Tone flag is disabled.
• When dial tone flag is disabled, user will hear the dial tone of the exchange if provided, otherwise, user
will hear the silence.
• If the user is making the call from the SLT port and dial tone is not provided by the exchange, user will
not know when to start dialing the number. In this case, it is possible that some digits are not out dialed
on the port and wrong number is dialed out because system will out dial the number only if Outgoing
call Acknowledge is received but the user is not aware of this condition. Hence, it is required to enable
this flag, if exchange is not providing the dial tone.
• When Online dialing or Store and Forward dialing are used, some exchanges do not provide any tone
while routing/processing the call. Thus, the caller does not know whether the call is being processed or
not as there is silence. In such case, You must enable the Feed Routing Tone flag, SARVAM UCS will
play the routing tone to the caller. SARVAM UCS stops playing the routing tone will be stopped when
an alert message or connect message or disconnect message is received from the T1E1 Card.
By default, the Feed Routing Tone flag is disabled.
• To customize the pulse width option and set the pulse shapes configure the Custom Pulse
parameters. SARVAM UCS generates pulse shapes which match the country standard, where it is
installed. However, if the standard pulse shape does not match, SARVAM UCS enables you to
customize the pulse width to match your exact requirements.
To use customize pulse width option and set the pulse shape in 1 to 4 phases, keep the T1/E1 Custom
Pulse Width (CPW) flag enabled.
• Word 1: Set the pulse width for setting pulse shape in the 1st phase. The range of Custom Pulse
Width-Word 1 is 001 to 127. Default: 109.
• Word 2: Set the pulse width for setting pulse shape in the 2nd phase. The range of Custom Pulse
• Word 3: Set the pulse width for setting pulse shape in the 3rd phase. The range of Custom Pulse
• Word 4: Set the pulse width for setting pulse shape in the 4th phase. The range of Custom Pulse
• Select the Bearer Service supported by your service provider. You can select from:
• Speech
• 3.1 KHz Audio
By default, Speech is selected.
• The Overlap Receiving Timer is relevant while receiving the called party number information in
overlap receiving mode. It is not relevant for the port in overlap sending mode.
Range of Overlap Receiving Timer is from 01 to 99 seconds. Default: 15 seconds.
• Configure Pause Timer for theT1E1 Port. Range of Pause Timer is from 1 to 9 seconds. By default, it
is set to 3 seconds for all T1E1 Ports.

This Timer is required to insert delay between the digits while dialing out DTMF digits on the T1E1 port.
One of the applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage
Dialing”.
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the SARVAM
UCS will out dial the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
• When the SETUP Message is sent by SARVAM UCS to the network (ISDN exchange), the exchange
responds by sending SETUP ACK (Acknowledgment), and dial tone is played to the caller. The time
taken by the exchange to respond to the SETUP message may vary from exchange to exchange. Set
the SETUP Response Timer (sec) as per the time taken by the network to respond to the SETUP
message and play dial tone to the caller.
Valid Range of the timer is 01 to 20 seconds. By default it is set to 4 seconds.
Change the default settings only if required. If the time you set is less than the time taken by the exchange
to respond, no dial tone will be played to the caller.
You can configure the SETUP Message Timer only from Jeeves.
• Configure DTMF On Time for theT1E1 Port. Range of DTMF On Time is from 051 to 255 ms. By
default, it is set to 102 ms for all T1E1 Ports.
The DTMF On Time is the time for which the DTMF digit which is to be outdialed by the SARVAM UCS
remain On. One of the applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-
Stage Dialing”.
• Configure DTMF Inter Digit Pause Timer for the T1E1 Port. Range of Inter Digit Pause Timer is from
051 to 255 ms. By default, it is set to 102 ms for all T1E1 Ports.
Inter Digit Pause Timer is the time for which the system will wait while receiving the dialing digits to
consider it as end-of-dialing.
• Configure the Minimum ON Time (msec) for which the DTMF signal should be present in order to be
detected. The valid range of this time is 10 to 200 milliseconds. By default, Minimum ON Time is set to
20 milliseconds.
• Configure the Minimum OFF Time (msec). This parameter signifies the minimum time period between
successive DTMF digits. The valid range of this time is 10 to 200 milliseconds. By default, Minimum
OFF Time is set to 20 milliseconds.
Configure the Minimum Level (dB) for the DTMF digit to be considered as valid. The valid range of
this time is 0 to -36.5 dB. By default, Minimum levels is set to -36.5dB.
• If SARVAM UCS is to be used as a Gateway, enable Gateway Application-Answer Signaling on the

T1E1 Port and configure DTMF String. By default, Gateway Application-Answer Signaling is disabled
and CCC is configured as DTMF String.
For more details, see “Gateway Application-Answer Signaling”.
• Configure Call Budget parameters for the T1E1 Ports. Call Budget is an expense control feature of
SARVAM UCS that allows you to keep track of the cost of phone calls made from the T1E1 Port.

• Type: Select the type of Call Budget, that is, Amount or Minutes or Calls to be applied on the T1E1
Port. By default, no Call Budget type is selected.
• Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field.
By default the Amount is set to 999999.
• Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this
field. By default the number of minutes is set to 999999.
• Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By
default the number of calls is set to 9999.
• Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to
be reset on a particular date of every month.
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan
suggests so.
• Call Back: This parameter is related to the ‘Call Back on Trunk Port’ feature. If you want to enable the
'Call Back on Trunk Port' feature on this T1E1 Port, configure the following parameters:
• Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this
flag is disabled on all trunk port types. By default, the flag is disabled.
• Call Back Timer (sec): This is the duration for which the system waits for the caller to disconnect
before applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set
to 10 seconds.
• Call Back Mode: Select from the following options how a ‘Call Back’ call answered by the remote
party should be routed:
• Operator
• Call Back on: This parameter allows you to select if the call back should be made to the same
number that was received or to a different number. If you want the call back to be made to the same
number select the ‘CLI number”. If you want the call back to be made to a different number, select
‘Alternate Number’.
Number List 15 is also assigned to all T1E1 Ports as well as all other Trunk port types. If you want
the same numbers strings to be programmed commonly for all T1E1 Ports and Trunk Port types,
retain this list.

If you want a different set of number strings to be programmed for this T1E1 Port, select a different
Number List, and assign it to the T1E1 Port.
Refer the topic “Number Lists” to know more, and for configuration instructions.
• Outgoing Number List: Program the number strings that are to be called back in this List.
For each number string you programmed in the ‘Incoming Number List’, you must program in the
corresponding index in the Outgoing Number List a number to which the call back is to be made.
For example, for the number string programmed at Index 1 in the Incoming Number List, a
corresponding number string at the same Index, Index 1, should be programmed in the ‘Outgoing
Number List’.
assigned to all T1E1 Ports as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this T1E1 Port.
• Call Back from: This parameter determines the trunk port to be used to make the call back. The
call back can be made using the Same Port or an “OG Trunk Bundle Group”.
Select ‘Same Port’ if you want the call back to be made using the same port on which the missed
call is received. If you select OGTB Group, the call back will be made using the OGTB Group, which
you have defined.
By default, Same Port is selected.
• OGTB Group: If you selected OGTB Group for making the call back in the previous parameter, you
must define the OGTB Group that must be used in this parameter.
By default, OGTB Group 01 is assigned.
Routing on the OGTB Group that you define here for Call Back.
• FDL is used for communicating general maintenance information or for transmitting user defined
information within the T1 link. General maintenance information is in the form of Performance Message
Report which is generated by the SARVAM UCS Card T1E1PRI and depending upon the FDL Protocol,
the Performance Message Report is sent every second or on request.
Select the FDL flag to enable. This parameter is applicable only if Framing = ESF. If the Network
(Public or Private) to which the SARVAM UCS is connected does not support FDL then FDL will be
disabled. By default, the T1 FDL is disabled.

If you have enabled FDL, configure the FDL Protocol. SARVAM UCS supports ANSI T1.403 and AT&T
54016 protocols of reporting the performance monitoring. By default, the T1 FDL Protocol is ANSI
T1.403.
For more information, see “E1/T1 Maintenance”
• Enter appropriate Debug Code (Level 1 to 4), to obtain debug information of various parts of T1E1
Card on the COM Port. By default, debug is off for all T1E1 ports for all levels.
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from
000 to 255. Code value '000' for each level will turn off that level's debug.
Level and Code for T1E1 Port are as specified below:
Level 1:
Unused Unused Unused Unused Layer 4 CAS DSP MFC R2 CAS
001 CAS
002 MFC R2
004 CAS DSP
008 Layer 4
000 Debug Off
Level 2:
Unused Unused Unused HDLC (D-Channel) FDL ABCD Bits Counters Alarms
001 Alarms
002 Counters
004 ABCD Bits
008 FDL
HDLC (D
016
Channel)
000 Debug Off
Level 3:
Variable
Unused Flow Debug NLS Debug LAP Debug SVC Primitives State Primitives
s
001 Primitives
002 State
004 Variables

008 SVC Primitives
016 LAP Debug
032 NLS Debug
064 Flow Debug
000 Debug Off
Level 4:
Unused Unused Unused Unused Unused Unused NI Debug OS Task
001 OS Task
002 NI Debug
000 Debug Off
PRI/QSIG Parameters
If you have selected PRI as Signal Type on the Port Parameters page,
• Under T1E1 Configuration, click PRI/QSIG Signaling and configure the PRI/QSIG parameters.
• ISDN Switch Variant: ISDN supports a variety of service provider switches. Different countries use
specific type of ISDN switch. This switch is designed using ISDN standard protocol. The type of switch
determines various factors such as how many ISDN devices would be handled, which B-channel will
support voice, video, data etc. Select the ISDN Switch Variant from the list. Default: ETSI NET5.
• Offer continuous Bearer Channel Mapping(01-30): Select this check box for continuous Bearer
Channel Mapping for E1-QSIG/E1-PRI.
• D-Channel: Enter the Channel number that is used for Data Signaling. By default, D-Channel is 16.
Valid Range is 1 to 31.

• Send Called Party Number Using: Select the appropriate option from the following for Send Called
Party Number Using:
• Called Party Number IE (Information Element)
• Keypad Facility IE (Information Element)
Default: Called Party Number IE is selected.
• Dialing Type for Called Party Number: Select the option from the following as supported by your
exchange:
• Enbloc
• Digit-by-Digit
• Any
Default: Any.
• Caller - Type of Numbering Plan (TON): Select the appropriate option from the following for sending
the type of numbering plan of the calling party:
• Unknown
• International
• National
• Network Specific
• Subscriber
• Abbreviated
• Reserved
Default: Unknown.
• Caller- Numbering Plan Identification (NPI): Select the appropriate option from the following for
sending the numbering plan identification of the calling party:
• Unknown
• ISDN Numbering
• Data Numbering
• Telex Numbering
• National Numbering
• Private
• Reserved
Default: ISDN Numbering.
• Called - Type of Numbering Plan (TON): Select the appropriate option from the following for sending
the type of numbering plan of the called party:
• Unknown
• International
• National
• Subscriber
• Abbreviated
• Reserved
Default: Unknown.
• Called - Numbering Plan Identification (NPI): Select the appropriate option from the following for
sending the numbering plan identification of the called party:
• Unknown
• ISDN Numbering
• Data Numbering
• Telex Numbering

• Private
• Reserved
• Receive Equalization Mode: You can set the Receive Equalization Mode as Auto or Manual. By default,
Auto is selected as the Receive Equalization Mode.
• Receive Equalization Parameters: This field increases the strength of incoming signals by a fixed
amount to compensate for line losses. Select the required option from the list. By default, the receive
equalization parameters of T1E1 is 8dB.
• Feed Inband Tones on T1E1-NT, before sending DISCONNECT: Select this flag, if you want to feed
inband tones on T1E1-NT before sending DISCONNECT message. This flag is applicable only when
T1E1 port is configured as 'Network'. When this flag is enabled, inband tones shall be feed for 15 seconds
(fixed, non programmable) before sending DISCONNECT message.
When this flag is disabled, inband tones (Busy/Error as applicable for the state of the call) shall not be feed
before sending the DISCONNECT message. However when DISCONNECT message is sent from T1E1-
NT port, inband tones will always be sent with 'progress indicator 8'. By default, this flag is disabled.
• Send SETUP ACK with PI: Select this flag if you want the system to send PI (Progress indicator) element
in Setup Ack message to other end. This option is relevant only when call landing on the T1E1 port are to
be routed to SLT/CO/Mobile/DKP port. If T1E1 port is configured as 'Network’, by default it is enabled and
if T1E1 port is configured as 'Terminal’, by default it is disabled.
• Send PROCEED with PI: Select this flag if you want the system to send PI (Progress indicator) in
Proceed message to other end. This option is relevant only when call landing on the T1E1 port are to be
routed to SLT/CO/Mobile/DKP port. If T1E1 port is configured as 'Network’, by default it is enabled and if
T1E1 port is configured as 'Terminal’, by default it is disabled.
• Send PROGRESS with PI: Select this flag if you want the system to send PI (Progress indicator) in
Progress message to other end. This option is relevant only when call landing on the T1E1 port are to be
routed to SLT/CO/Mobile/DKP port. By default it is enabled.
• Send ALERT with PI: Select this flag if you want the system to send PI (Progress indicator) in Alert
message to other end. This option is relevant only when call landing on the T1E1 port are to be routed to
SLT/CO/Mobile/DKP port. If T1E1 port is configured as 'Network’, by default it is enabled and if T1E1 port
is configured as 'Terminal’, by default it is disabled.
• Transparently pass PI in case of ISDN to ISDN / SIP call: Select this flag, if you want the system to just
pass the PI as received to the other end.This option is relevant only when call landing on the T1E1 port are
to be routed to ISDN or SIP ports. By default it is enabled.

E&M Signaling
Under T1E1 Configuration, click E&M Signaling and configure the E&M parameters.
• E&M Feature Template: Assign an E&M Feature Template to the T1E1 Port. The E&M Feature Template
is a set of features specific to E&M signaling, which define the behavior of the E&M ports, according to
their 'Orientation Type', whether they are functioning as Stations, Trunks or Tie-Lines. By default, Template
01 is assigned to all T1E1 Ports.
For more details, see “E&M Feature Template”.
• B Bit Pattern: Select the Bit Pattern from Same as Bit A or Fixed Value. By default, the Code is 1 (Same
as A bit).
• B Bit Value: Configure the B bit value, the value can be 0 or 1. By default, B bit value is 0.
• CD Bit Value: Configure the CD bit value, the valid range of the value is 1 to 3. By default, B bit value is 0.
• Invert Bit A: This parameter signifies whether A-bit is to be inverted before transmitting and on receiving.
Select the check box to Invert Bit A.
Default: Disabled (Do Not Invert Bit A).
• Invert Bit B: This parameter signifies whether B-bit is to be inverted before transmitting and on receiving.
Select the check box to Invert Bit B.
Default: Disabled (Do Not Invert Bit B1)

• Invert Bit C: This parameter signifies whether C-bit is to be inverted before transmitting and on receiving.
Select the check box to Invert Bit C.
Default: Disabled (Do Not Invert Bit C).

• Invert Bit D: This parameter signifies whether D-bit is to be inverted before transmitting and on receiving.
Select the check box to enable, that is to Invert Bit D.
Default: Disabled (Do Not Invert Bit D).
RBS Parameters
Under T1E1 Configuration, click T1 RBS Signaling and configure the RBS parameters.
• Line Signaling Variant: Select the T1 Line Signaling Variant from following options:
• SLT Loop Start
• CO Loop Start
• SLT Ground Start
• CO Ground Start
• E&M Immediate Dial/Start
• E&M Wink Start
• E&M Wink Start FGD
Default: E&M Wink Start FGD.
• Wink Timer (milliseconds): Wink timer refers to the momentary Off-Hook condition to acknowledge
end of making an outgoing call. The Wink Timer ranges from 0001 ms to 9999 ms. Default: 160 msec.
• Wink Wait Timer (milliseconds): Wink Wait Timer signifies the maximum time the system should wait
before sending a wink start signal after an incoming seizure is detected. Wink Wait Timer ranges from
0001 to 9999 msec. Default: 30msec.
Ensure that this timer is greater than the Wink Wait Timer of the other end.

• Wait Wink Timer (milliseconds): Wait Wink Timer signifies the time for which SARVAM UCS will wait
for receiving the DNIS after sending the outgoing seizure signal. Wait Wink Timer ranges from 001 to
999 msec. Default: 5000 msec.
Make sure that this timer is greater than the Wait Wink Timer of the other end.
• Delay Duration (milliseconds): This duration signifies the time after which the DNIS information is to
be sent while making an outgoing call. Range of the Delay Duration is from 0001 to 9999 msec.
Default: 100 msec.
• Start Delay Timer: Start Delay Timer signifies the time for which SARVAM UCS waits for receiving
DNIS from the network. This timer is loaded on receiving the Off-hook (I/C Seizure) on the receive
channel (while receiving an incoming call). The Start Delay Timer ranges from 0001 to 9999 ms.
Default: 20 msec.
• Register Signaling Variant: The Register Signaling Variant for T1/E1 Ports is set as DTMF.
• Inbound ANI/DNIS Format: Select the Inbound ANI/DNIS Format for T1/E1 Ports from the following
options:
• ANI
• DNIS
• ?ANI?
• ?DNIS?
• ?ANI?DNIS?
• ?DNIS?ANI?
Default: ?ANI?DNIS?.
• Inbound Delimiter (?) Character: Define the Inbound Delimiter Character in this field. Characters
supported in this field are 0-9, #, *, A, B, C and D. Default: *
• Outbound ANI/DNIS Format: Select the Outbound ANI/DNIS Format for T1/E1 Port from the following
options:
• ANI
• DNIS
• ?ANI?
• ?DNIS?
• ?ANI?DNIS?
• ?DNIS?ANI?
• Outbound Delimiter (?) Character: Define the Outbound Delimiter Character in this field. Characters
The commands explained below should be referred as:
• To program a single port: XXXX-1
• To program a range of ports: XXXX-2
• To program all the ports: XXXX-*

Port Parameters
T1E1-1
Hardware Slot-Port
Use following command to assign hardware ID to a T1E1 software port.

1107-T1E1-Slot-Port offset
Where,
T1E1 is from 01 to 08.
Slot is the number of the universal slot, where the T1E1 Card is installed, from 01 to 16.
Port is the number of theT1E1 port on the card, from 01 to 32.
Use following command to de-assign the hardware slot and the hardware port assigned to the T1E1 software port.
1106-T1E1-00-00
Port Status
Use the following command to enable/disable the port:

6101-1-T1E1-Port Status
6101-2-T1E1-T1E1-Port Status
6101-*-Port Status
Where,
Port Status Meaning
0 Disable
1 Enable
By default, the T1E1 Port is enabled.
Name
Use the following command to assign a name to the port:

5407-1-T1E1-Name
5407-2-T1E1-T1E1-Name
5407-*-Name
Where,
Name can be of upto 18 characters.
Carrier
Use the following command to select the carrier:

6108-1-T1E1-Carrier Type
6108-2-T1E1-T1E1-Carrier Type
6108-*-Carrier Type
Where,
Carrier
Meaning
Type
1 E1

Carrier
Meaning
Type
2 T1
Line type
Use following command to program signaling type/ Line type of a T1E1:

6105-1-T1E1-Line Type
6105-2-T1E1-T1E1-Line Type
6105-*-Line Type
Where,
Line Type Meaning
1 ISDN_E1_PRI
2 ISDN_T1_PRI
3 ISDN_E1_CAS
4 ISDN_T1_RBS
5 ISDN_E1_QSIG
6 ISDN_T1_QSIG
7 ISDN_E1_E&M
8 ISDN_T1_E&M
By default, signaling type/ of a T1E1 is 1.
DDI Routing is not supported on T1/E1 trunk line if you have selected E&M as the Signal Type.
Orientation Type
Use following command to program 'Orientation Type' for the T1E1 port:
6106-1-T1E1-Orientation Type
6106-2.T1E1-T1E1-Orientation Type
6106-*-Orientation Type
Where,
T1E1=01 to 08.
Orientation Meaning
1 Terminal
2 Network
3 Tie Line
By default Type = 1.
When Orientation = Terminal, the port will be regarded as trunk. All the trunk related parameters will be applicable.

When Orientation = Network, the port will be regarded as station. All the station related parameters will be
applicable.
When Orientation = Tie-line, the port will be regarded as station for all IC calls to it and as trunk for all OG calls to be
made through it.
Line Coding Mechanism
Use following command to program the Line Coding Mechanism for the T1E1 port:
6103-1-T1E1-Line Coding
6103-2-T1E1-T1E1-Line Coding
6103-*-Line Coding
Where,
Line Coding Meaning
1 AMI-Basic
2 B8ZS
3 CMI
By default, Line Coding is AMI-Basic.
Set Line Coding = AMI or B8ZS for T1 line. However, B8ZS is recommended.
Set Line Coding = AMI or HDB3 for E1 line. However, HDB3 is recommended.
CMI is used in Japan and since SARVAM UCS does not support Japan, this option will never be used.
Framing Mode
Use following command to program the Framing Mode for the T1E1 port:
6104-1-T1E1-Framing
6104-2-T1E1-T1E1-Framing
6104-*-Framing
Where,
Framing Meaning
1 SF (D4) for T1
2 ESF for T1
By default, Framing is SF (D4).
Set Framing = SF or ESF for T1 line. However, ESF is recommended since it supports advanced features
like CRC and FDL, which provide the performance reports.
Line Buildout Parameters
Use the following command to program the line build out parameters of a T1E1:
6162-1-T1E1-Code
6162-2-T1E1-T1E1-Code
6162-*-Code
Where,

Code Meaning
1 0-133ft
2 133-266ft
3 266-399ft
4 399-533ft
5 533-665ft
6 -7.5dB or equidistance
7 -16dB or equidistance
8 -22.5dB or equidistance
By default, the Line Build Out Parameters is 0-133ft.
DDI Routing
OG Reference ID
Use the following command to assign OG Reference ID to T1E1 port:

6131-1-T1E1-OG Reference ID
6131-2-T1E1-T1E1-OG Reference ID
6131-*-OG Reference ID
Where,
OG Reference ID is from 00 to 99.
By default, OG Reference ID is 00.
IC Reference ID for Working Hour
Use the following command to assign IC Reference ID for Working Hour:

6132-1-T1E1-IC Reference ID
6132-2-T1E1-T1E1-IC Reference ID
6132-*-IC Reference ID
Where,
IC Reference ID is from 00 to 99.
By default, IC Reference ID is 00.
IC Reference ID for Break Hour
Use the following command to assign IC Reference ID for Break Hour:

Where,
By default, IC Route Reference ID is 00.

IC Route Reference ID for Non-working Hour
Use the following command to assign IC Route Reference ID for Non-working Hour:
Where,
By default, IC Route ID is 00.
Templates
Use the following command to assign a Trunk Feature Template to the T1E1 Trunks, dial:
5806-1-T1E1- Trunk Feature Template Number to assign a template to a single T1E1 port.
5806-2-T1E1- Trunk Feature Template Number to assign the same template to a range of T1E1 ports.
5806-*- Trunk Feature Template Number to assign the same template to all T1E1 ports.
Where,
Template Number is the number of the customized Trunk Feature Template, from 01 to 50. Default: Trunk Feature
Template 01.

5506-1-T1E1PRI-Template Number to assign a template to a single T1E1 port.
5506-2-T1E1PRI-T1E1PRI-Template Number to assign the same template to a range of T1E1 ports.
5506-*-Template Number to assign the same template to all T1E1 ports.
Where,
Station Advance Features Template
To assign a Station Advanced Feature Template to a T1E1PRI port

Where,
Others
Priority
To assign priority to T1E1

3914-1-T1E1-Priority to assign a template to a single T1E1 port.
3914-2-T1E1PRI-T1E1PRI-Priority to assign the same priority to a range of T1E1 ports.
3914-*-Template Number to assign the same Priority to all T1E1 ports.
Where,

Priority is from 1 to 9. Default: 5-Normal.
Cost Factor
Use following command to assign a name to the T1E1 port:

6102-1-T1E1-SP
6102-2-T1E1-T1E1-SP
6102-*-SP
Where,
SP is from 01 to 99.
By default, Service Provider is 01.
Glare Option
Use following command to program Glare Option for the T1E1 port:
6112-1-T1E1-Glare Option
6112-2-T1E1-T1E1-Glare Option
6112-*-Glare Option
Where,
Glare Option Meaning
1 Proceed
2 Held Back
By default, Glare Option is 2.
Set Glare=Proceed, if SARVAM UCS is to be given priority in event of Glare.

Set Glare=Held Back, if the other end of the link is to be given priority in event of Glare.
The Glare settings should be complimentary on either side of the link.
Category (Logical Partitioning)
Use following command to set the Category (Logical Partitioning) for the T1E1 port:
6121-1-T1E1-Category
6121-2-T1E1-T1E1-Category
6121-*-Category
Where,
Category is from 1 to 3.
By default, Category is set as 1.
Idle Code
Use the following command to program the Idle Code of a T1E1:

6113-1-T1E1-Idle Code
6113-2-T1E1-T1E1-Idle Code
6113-*-Idle Code
Where,

Idle Code is from 000 to 255 (corresponding to 8 bits).
By default, the idle code is 127 (7F).
The binary equivalent of the programmed value (000 to 255) is sent on the channel to signify that the channel is
idle. (or Unused) This setting depends on the network. Most commonly applicable values are 7F and FF (Binary
equivalent is 0111 1111 and 1111 1111, decimal equivalent is 127 and 255).
Use message mode of the Digital Switch IC to send the idle channel code.
RCOC
To enable RCOC on T1E1 Trunk

Dial 6145-1-T1E1-Code to enable the feature on a single trunk.
Dial 6145-2-T1E1-T1E1-Code to enable the feature on a range of trunks.
Dial 6145-*-Code to enable the feature on all trunks.
Where,
T1E1 is the software port number of the trunk from 01 to 08.
Code is
0 for Disable
1 for Enable
Default: Disable
Channels
Return Call to Original Caller (RCOC)
Use the following command to enable RCOC on T1E1 Trunk:

6145-1-T1E1-Code to enable the feature on a single trunk.
6145-2-T1E1-T1E1-Code to enable the feature on a range of trunks.
6145-*-Code to enable the feature on all trunks.
Where,
Code is
0 for Disable
1 for Enable
Default: Disable
Channel Reserved for Data Call
Use the following command to reserve channels for data transmission on T1E1:
6135-1-T1E1-Channel Count (Data)
6135-2-T1E1-T1E1-Channel Count (Data)
6135-*-Channel Count (Data)
Where,
Channel Count (Data) is from 00 to 30.
By default, Channel Count for data transmission is 00.
Channel Reserved for Outgoing Call
Use the following command to program number of channels reserved for OG channel count:
6136-1-T1E1-Channel Count (OG)
6136-2-T1E1-T1E1-Channel Count (OG)
6136-*-Channel Count (OG)

Where,
Channel Count (OG) is from 00 to 30. "It specifies the number of channels to be reserved for making an OG calls.
For example, If OG channel count is programmed as 15, simultaneous 15 (maximum) OG calls can be made from
the T1E1 port".
By default, OG Channel Count is 30.
Channel Reserved for Incoming Call
Use the following command to program number of channels reserved for IC channel count:
6137-1-T1E1-Channel Count (IC)
6137-2-T1E1-T1E1-Channel Count (IC)
6137-*-Channel Count (IC)
Where,
Channel Count (IC) is from 00 to 30. "It specifies the number of channels to be reserved for making an IC calls. For
example, If IC channel count is programmed as 10, simultaneous 10 (max.) IC calls can be received on the T1E1
port".
By default, Channel Count (IC) is 30.
Tone
Feed Dial Tone
Use following command to program the dial tone flag for T1E1 port:
6115-1-T1E1-Flag
6115-2-T1E1-T1E1-Flag
6115-*-Flag
Where,
Flag Meaning
0 Disable
1 Enable
By default, Dial Tone Flag is '0' for all the T1E1 ports.
Feed Routing Tone
Use following command to program the routing tone flag for T1E1 port:
6116-1-T1E1-Flag
6116-2-T1E1-T1E1-Flag
6116-*-Flag
Where,
Flag Meaning
0 Disable
1 Enable
By default, Routing Tone Flag is '0' for all the T1E1 ports.

Custom Pulse
T1/E1 Custom Pulse Width (CPW)
Use following command to enable/disable Custom Pulse Width (CPW) Flag for the T1E1 port for T1 signaling:
6171-1-T1E1-Flag
6171-2-T1E1-T1E1-Flag
6171-*-Flag
Where,
Flag Meaning
0 Disable
1 Enable
By default, Custom Pulse Width Flag is '0'.
T1/E1Custom Pulse Width Word 1
This command is used for Pulse shaping in first phase.
Use following command to program Custom Pulse Width Word 1 for the T1E1 port for T1 signaling:
6172-1-T1E1-Custom Pulse Width Word 1
6172-2-T1E1-T1E1-Custom Pulse Width Word 1
6172-*-Custom Pulse Width Word 1
Where,
Custom Pulse Width Word 1 is from 000 to 127 volt.
By default, Custom Pulse Width Word 1 is 63 volt.
This command is used for Pulse shaping in second phase.
Where,
This command is used for Pulse shaping in third phase.
Where,

This command is used for Pulse shaping in fourth phase.
Where,
Timer
Pause Timer
Use following command to program Pause Timer:

6109-1-T1E1- Pause Timer
6109-2-T1E1-T1E1-Pause Timer
6109-*-Pause Timer
Where,
T1E1 is from 01 to 08
Pause Timer is from 1 to 9 seconds
By default, Pause Timer is 3 seconds.
DTMF ON Time
Use following command to program DTMF ON Time:

6117-1-T1E1-DTMF ON Time
6117-2-T1E1-T1E1-DTMF ON Time
6117-*-DTMF ON Time
Where,
DTMF ON Time is from 051 to 255 msec.
By default, DTMF ON Time is 102 msec.
DTMF Inter Digit Pause Timer
Use following command to program DTMF Inter digit Pause Timer:

6118-1-T1E1-DTMF Inter digit Pause Time
6118-2-T1E1-T1E1-DTMF Inter digit Pause Time
6118-*-DTMF Inter digit Pause Time
Where,
DTMF Inter digit Pause Time is from 051 to 255 msec.
By default, DTMF Inter digit Pause Time is 102 msec.
Gateway
Use Gateway Application - Answer Signaling?
Use following command to set flag for 'Gateway Application-Answer Signaling' on T1E1 trunk:
6119-1-T1E1-Gateway Application-Answer Signaling flag

6119-2-T1E1-T1E1-Gateway Application-Answer Signaling flag
6119-*- Gateway Application-Answer Signaling flag
Where,
Flag Meaning
0 Disable
1 Enable
By default, Gateway Application-Answer Signaling flag is 'disable'.
Gateway Application - Answer Signaling DTMF Digits

Use following command to program DTMF digits string to be dialed as Gateway Application-Answer Signaling:
6120-1-T1E1-Gateway Application-Answer Signaling DTMF String
6120-2-T1E1-T1E1-Gateway Application-Answer Signaling DTMF String
Where,
DTMF Digits allowed for DTMF string are from (0 - 9), *, #, A, B, C, D.
Maximum 4 DTMF digits can be programmed. If you need less than 4 digits for DTMF string, terminate the
command using #*.
To program #, *, A, B, C, D use following codes:
Digit Code for programming through command
A #4
B #5
C #6
D #7
* **
# ##
By default Gateway Application-Answer Signaling DTMF String is 'CCC'.
Call Budget
Call Budget Type
To program Call Budget Type on T1E1 Port, dial:

6122-1-T1E1-Budget Type to program call budget type for a single trunk port.
6122-2-T1E1-T1E1-Budget Type to program the same call budget type for a range of trunk ports.
6122-*-Budget Type to program the same call budget type for all trunk ports.
Where,
T1E1is the number of the T1E1 software port from 01 to 08.
Budget Type is
0 for None
1 for Amount
2 for Minutes

3 for Number of Calls
By default, Budget Type is None.
Call Budget Amount
To program Call Budget Amount on T1E1 Port, dial:

6123-1-T1E1-Budget Amount to program call budget amount for a single trunk port.
6123-2-T1E1-T1E1-Budget Amount to program the same amount for a range of trunk ports.
6123-*-Budget Amount to program the same amount for all trunk ports.
Where,
Budget Amount is of 6 digits max. Use leading zeros if amount to be programmed has fewer than 6 digits.
By default Budget Amount is 999999.
Call Budget Minutes
To program Call Budget Minutes on T1E1 Port:

6124-1-T1E1-Minutes to program minutes for a single trunk port.
6124-2-T1E1-T1E1-Minutes to program the same minutes for a range of trunk ports.
6124-*-Minutes to program the same minutes for all trunk ports.
Where,
Minutes is of 6 digits max. Use leading zeros if Minutes to be programmed has less than 6 digits.
By default, Minutes is 999999.
Call Budget Number of Calls
To program Call Budget Number of Calls on T1E1 Port, dial:

6125-1-T1E1-Number of calls to program number of calls for a single trunk port.
6125-2-T1E1-T1E1-Number of calls to program the same number of calls for a range of trunk ports.
6125-*-Number of calls to program the same number of calls for all trunk ports.
Where,
Number of Calls is of 4 digits from 0001 to 9999. Use leading zeros if number of calls to be programmed has fewer
than 4 digits.
By default, Number of calls is 9999.
Call Budget Reset Mode
To program Call Budget Reset Mode for T1E1, dial:

6138-1-T1E1-Call Budget Reset Mode to program reset mode for a single trunk port.
6138-2-T1E1-T1E1-Call Budget Reset Mode to program the same reset mode for a range of trunk ports.
6138-*-Call Budget Reset Mode to program the same reset mode for all trunk ports.
Where,
1 for Scheduled reset
2 for Manual reset
By default, Call Budget Reset Mode is Scheduled.
Call Budget Date for Scheduled Reset mode
To program the Date for Scheduled Reset mode for T1E1, dial:
6139-1-T1E1-Date to program date for a single trunk port.
6139-2-T1E1-T1E1-Date to program the date for a range of trunk ports.
6139-*-Date to program the date for all trunk ports.
Where,

Date is
01 to 31 for Scheduled date to reset every month.
00 for Scheduled reset Daily.
By default, Reset date is 1st. of every month.
Call Back
Use the following commands to program Call Back on T1E1 Trunk ports. To know more about this feature, refer the
topic Call Back on Trunk Ports.
To enable/disable Call Back on T1E1 port:

6176-1-T1E1- Call Back Flag
6176-2-T1E1-T1E1-Call Back Flag
6176- *-Call Back Flag
Where,
T1EI is from 01 to 08
Call Back Flag Meaning
0 Disable
1 Enable
By default, Call Back flag is disabled.
Call Back Timer
To program Call Back Timer for T1E1 port:

6177-1-T1E1-Call Back Timer
6177-2-T1E1-T1E1-Call Back Timer
6177-*-Call Back Timer
Where,
Pause Timer Range is from 01 to 99 Sec.
By default, Pause Timer is 10 Seconds
Call Back Mode
To program Call Back Mode on T1E1 port:

6178-1-T1E1-Call Back Mode
6178-2-T1E1-T1E1-Call Back Mode
6178-*-Call Back Mode
Where,
Call Back Mode is 1 to 5.
Call Back Mode Meaning
1 Built-In Auto Attendant
2 PIN Auth. - Multiple Calls
3 CLI Auth. - Multiple Calls
4 CLI Auth. - Single Call - Ans. Sig.

5 Operator
By default, Call Back mode is Operator
Call Back On
To program Call Back On method for T1E1 port:

6179-1-T1E1-Call Back on selection
6179-2-T1E1-T1E1-Call Back on selection
6179-*-Call Back on selection
Where,
Call back on selection is
Call Back on Meaning
1 CLI Number
2 Alternate Number
By default, Call Back on selection is CLI Number.
Incoming Number List
To assign Call Back - Incoming Number List to a T1E1 port:

6180-1-T1E1-Incoming Number List
6180-2-T1E1-T1E1-Incoming Number List
6180-*-Incoming Number List
Where,
Incoming Number List is from 01 to 16.
By default, Incoming Number List is 15.
Outgoing Number List
To assign a Call Back - Outgoing Number List to a T1E1 port:

6147-1-T1E1-Outgoing Number List
6147-2-T1E1-T1E1-Outgoing Number List
6147-*-Outgoing Number List
Where,
Outgoing Number List is from 01 to 16.
By default, Outgoing Number List is 16.
Call Back From
To define Call Back From for a T1E1 port:

6148-1-T1E1-Call Back From
6148-2-T1E1-T1E1-Call Back From
6148-*-Call Back From
Where,
Call Back From is
1 for Same Port

2 for OGTB Group
By default, Same Port is selected as Call Back From.
OGTB Group
To assign a Call Back - OGTB Group for a T1E1 port:

6149-1-T1E1-OGTB Group
6149-2-T1E1-T1E1-OGTB Group
6149-*-OGTB Group
Where,
OGTB Group is from 01 to 32
By default, OGTBG is 01.
FDL
FDL Flag - copied from T1 Maintenance
Use following command to enable/disable T1 FDL on a T1E1PRI port:

6164-1-T1E1PRI-T1 FDL
6164-2-T1E1PRI-T1E1PRI-T1 FDL
6164-*-T1 FDL
Where,
T1E1PRI is from 01 to 08.
T1 FDL Meaning
0 Disable
1 Enable
By default, the T1 FDL is disabled.
FDL Protocol
Use following command to program the T1 FDL Protocol for a T1E1PRI port:
6165-1-T1E1PRI-T1 FDL Protocol
6165-2-T1E1PRI-T1E1PRI-T1 FDL Protocol
6165-*-T1 FDL Protocol
Where,
T1 FDL Protocol Meaning
0 Disable
1 AT&T 54016
2 ANSI T1.403
By default, the T1 FDL Protocol is ANSI T1.403.
Refer “E1/T1 Maintenance”

Debug
SARVAM UCS supports debug of parameters (debug codes) depending on the Level of debug. On issuing this
command the SARVAM UCS Card T1E1 will send the debug details to the COM port of the T1E1 port.
Debug Level Command
Debug Level-1 6191-1-T1E1-1-XXX
Option 1
Use following command to start/stop debug the parameters for the T1E1 port:
6191-1-T1E1-Level-Debug Code
6191-2-T1E1-T1E1-Level-Debug Code
6191-*-Level-Debug Code
Where,
T1E1 Port is from 01 to 08.
Level is from 1 to 4 (As shown below).
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from 000 to 255.
Code value ‘000’ for each level will turn off that level’s debug.
Level 1:
001 CAS
002 MFC R2
004 CAS DSP
008 Layer 4
000 Debug Off
Level 2:
001 Alarms
002 Counters
004 ABCD Bits
008 FDL
HDLC (D
016
Channel)
000 Debug Off

Level 3:
Unused Flow Debug NLS Debug LAP Debug SVC Primitives Variables State Primitives
001 Primitives
002 State
004 Variables
008 SVC Primitives
016 LAP Debug
032 NLS Debug
064 Flow Debug
000 Debug Off
Level 4:
001 OS Task
002 NI Debug
000 Debug Off
Default: Debug Code = ‘Debug OFF’ for all T1E1 ports for all levels.
T1E1:1 - PRI/QSIG Signaling

ISDN Switch Variant
Use following command to program the ISDN PRI Switch Variant:

6107-1-T1E1-ISDN Switch Variant
6107-2-T1E1-T1E1-ISDN Switch Variant
6107-*-ISDN Switch Variant
Where,
ISDN PRI Variant Meaning
1 ATT 4ESS
2 ATT 5ESS
3 Australia
4 DMS
5 ETSI NET5
6 NTT INS64
7 SWV Hongkong
8 US NI12

9 QSIG E1
10 QSIG T1
By default, ISDN PRI Switch Variant of a T1E1 is ETSI NET5.
Send Called Party Number Using
Use the following command to program overlap receiving timer:

6146-1-T1E1-Type
6146-2-T1E1-T1E1-Type
6146-*-Type
Where,
Type is
1 for Keypad Facility IE
0 for Called Party IE
Default: Called Party Number IE
Overlap Receiving Timer

6114-1-T1E1-Timer
6114-2-T1E1-T1E1-Timer
6114-*-Timer
Where,
By default, Overlap Receiving Timer is 15 seconds.
Caller-Type of Number (TON)
Use following command to program the Caller-Type of Number (TON) for a T1E1:
6126-1-T1E1-Source TON
6126-2-T1E1-T1E1-Source TON
6126-*-Source TON
Where,
Source TON Meaning
Unknown: This is used when the user or network has no a prior information about
1 the numbering plan. In this case, the Address Value field is organized according to
the network dialing plan. For example, prefix or escape digits might be present.
2 International Number.
3 National Number: Prefix or escape digits shall not be included.
Network Specific Number: This is used to indicate administration/Service number

4
specific to the serving network. For example, used to access an operator.
Subscriber Number: This is used when a specific short number representation is

5
stored in one or more SCs as part of a higher layer application.
6 Abbreviated Number.

Source TON Meaning
7 Reserved Number.
By default, Caller-Type of Number (TON) of T1E1 is 1.
Caller-Numbering Plan Identification (NPI)
Use following command to program Caller-Numbering Plan Identification (NPI) for T1E1:
6127-1-T1E1-Source NPI
6127-2-T1E1-T1E1-Source NPI
6127-*-Source NPI
Where,
Source NPI Meaning
1 Unknown
2 ISDN Numbering Plan
3 Data Numbering Plan
4 Telex Numbering
5 National Numbering Plan
6 Private Numbering Plan
7 Reserved for Extension
By default, Caller-Numbering Plan Identification (NPI) for T1E1 is 2.
Called-Type of Number (TON)
Use following command to program Called-Type of Number (TON) for a T1E1:

6128-1-T1E1-Destination TON
6128-2-T1E1-T1E1-Destination TON
6128-*-Destination TON
Where,
Destination TON Meaning
1 Unknown: This is used when the user or network has no a prior information about
the numbering plan. In this case, the Address Value field is organized according to
4 Network Specific Number: This is used to indicate administration/Service number

5 Subscriber Number: This is used when a specific short number representation is


7 Reserved Number.
By default, Called-Type of Number (TON) of T1E1 is 1.
Called-Numbering Plan Identification (NPI)
Use following command to program Called-Numbering Plan Identification (NPI) for T1E1:
6129-1-T1E1-Destination NPI
6129-2-T1E1-T1E1-Destination NPI
6129-*-Destination NPI
Where,
Destination NPI Meaning
1 Unknown
4 Telex Numbering
By default, Called-Numbering Plan Identification (NPI) for T1E1 is 2.
Receive Equalization Mode
Use the following command to program auto receive equalization mode:

6110-1-T1E1-Mode
6110-2-T1E1-T1E1-Mode
6110-*-Mode
Where,
Mode Meaning
0 Manual
1 Auto
By default, Receive Equalization Mode is 1.
Receive Equalization Parameters
Use the following command to program the receive equalization parameters of a T1E1:
6111-1-T1E1-Receive Equalization Parameters
6111-2-T1E1-T1E1-Receive Equalization Parameters
6111-*-Receive Equalization Parameters
Where,

Receive Equalization Parameters Meaning
1 None
2 8 dB
3 16 dB
4 24 dB
5 32 dB
6 40 dB
7 48 dB
By default, the receive equalization parameters of T1E1 is 1.
Feed Inband Tones on T1E1-NT, before sending DISCONNECT
Use following command to program to select whether the inband tones should be feed on T1E1-NT before sending
DISCONNECT message?
6130-1-T1E1-Flag
6130-2-T1E1-T1E1-Flag
6130-*-Flag
Where,
Flag Meaning
0 No
1 Yes
Default = NO.
E&M Signaling
Assign E&M Feature Template to T1E1 using following command:

Use Following command to assign E&M Feature Template to T1E1 Port
6004-1-T1E1-Template Number
6004-2-T1E1-T1E1- Template Number
6004-*- Template Number
Where,
Template Number is 01 to 50.
By default, Template 01 is assigned to T1E1.
Now proceed to program other parameters for E&M on T1E1 using following commands:
B Bit Pattern
Use following command to select B Bit Pattern

7191-1-T1E1-Code
7191-2-T1E1-T1E1-Code

7191-*-Code
Where,
Code Meaning
1 Same as A bit
2 Fixed Value
By default, the Code is 1 (Same as A bit)
B Bit Value
Use following command to program B Bit Value

7192-1-T1E1-B Bit Value
7192-2-T1E1-T1E1- B Bit Value
7192-*- B Bit Value
Where,
B bit value can be 0 or 1.
By default, B bit value is 0.
CD Bit Value
Use following command to program CD Bit Value

7193-1-T1E1-CD Bit Value
7193-2-T1E1-T1E1-CD Bit Value
7193-*- CD Bit Value
Where,
CD bit value can be 1 or 3.
By default, CD bit value is 1.
Invert Bit A
Use following command to program to invert/don't invert Bit A for the T1E1 port:
7162-1-T1E1-Invert Bit A
7162-2-T1E1-T1E1-Invert Bit A
7162-*-Invert Bit A
Where,
Invert Bit A Meaning
0 Disable
1 Enable
By default, Invert Bit A is 0.
Invert Bit B
Use following command to program to invert/don't invert Bit B for the T1E1 port:
7163-1-T1E1-Invert Bit B
7163-2-T1E1-T1E1-Invert Bit B
7163-*-Invert Bit B

Where,
Invert Bit B Meaning
0 Disable
1 Enable
By default, Invert Bit B is 0.
Invert Bit C
Use following command to program to invert/don't invert Bit C for the T1E1 port:
7164-1-T1E1-Invert Bit C
7164-2-T1E1-T1E1-Invert Bit C
7164-*-Invert Bit C
Where,
Invert Bit C Meaning
0 Disable
1 Enable
By default, Invert Bit C is 0.
Invert Bit D
Use following command to program to invert/don't invert Bit D for the T1E1 port:
7165-1-T1E1-Invert Bit D
7165-2-T1E1-T1E1-Invert Bit D
7165-*-Invert Bit D
Where,
Invert Bit D Meaning
0 Disable
1 Enable
By default, Invert Bit D is 0.
RBS Signaling
To assign a Line Signaling Variants to the T1E1 port, dial:
6181-1-T1E1-T1 Line Signaling Variants
6181-2-T1E1-T1E1-T1 Line Signaling Variants
6181-*-T1 Line Signaling Variants
Where,

T1 Line Signaling Variants Meaning
1 SLT Loop Start
2 CO Loop Start
3 SLT Ground Start
4 CO Ground Start
5 E&M Immediate Start/Dial
6 E&M Wink Start
7 E&M Wink Start FGD
Default: E&M Wink Start.
To set the Wink Timer, dial:

6182-1-T1E1-Wink Timer
6182-2-T1E1-T1E1-Wink Timer
6182-*-Wink Timer
Where,
Wink Timer is from 0001 to 9999 ms.
By default, T1 Wink Timer is 0200 ms.
Use the following command to program the T1 wink wait timer for T1E1:
6183-1-T1E1-Wink Wait Timer
6183-2-T1E1-T1E1-Wink Wait Timer
6183-*-Wink Wait Timer
Where,
Wink Wait Timer is from 0001 to 9999 ms.
By default, T1 Wink Wait Timer is 0200 ms.
Use the following command to program the T1 wait wink timer for T1E1:
6184-1-T1E1-Timer
6184-*-Timer
Where,
Timer is from 0001 to 9999 ms.
By default, T1 Wait Wink Timer is 0200 ms.
Use the following command to program the T1 delay duration for T1E1:
6185-1-T1E1-Delay Duration
6185-2-T1E1-T1E1-Delay Duration
6185-*-Delay Duration
Where,
Delay Duration is from 0001 to 9999 ms.
By default, T1 Delay Duration is 0140 ms.

Use the following command to program the T1 Start Delay Timer for T1E1:
6186-1-T1E1-Start Delay Timer
6186-2-T1E1-T1E1-Start Delay Timer
6186-*-Start Delay Timer
Where,
Start Delay Duration is from 001 to 255 seconds.
By default, T1 Start Delay Timer is 020 seconds.
To assign a Register Signaling Variant to the T1E1 port, dial:

6161-1-T1E1-T1 Register Signaling Variant
6161-2-T1E1-T1E1-T1 Register Signaling Variant
6161-*-T1 Register Signaling Variant
Where,
T1 Register Signaling Variant Meaning
1 T1 RBS DTMF
T1 RBS DTMF: DNIS is transmitted in the corresponding speech channel using the DTMF signals as per the ITU-T
Q.23.
To select the Inbound ANI/DNIS Format, dial:

6166-1-T1E1-Code
6166-2-T1E1-T1E1-Code
6166-*-Code
Where,
Code is
T1 Register Signaling Variant Code
ANI 1
DNIS 2
?ANI? 3
?DNIS? 4
?ANI?DNIS? 5
?DNIS?ANI? 6
By default,?ANI?DNIS? is selected.
To assign the Inbound Delimiter (?) character, dial:

6167-1-T1E1-Character
6167-2-T1E1-T1E1-Character
6167-*-Character
Where,
Character can be #, *, A,B,C,D and 0-9.
By default, * is set.

To assign the Outbound ANI/DNIS Format, dial:
6168-1-T1E1-Code
6168-2-T1E1-T1E1-Code
6168-*-Code
Where,
Code is
ANI 1
DNIS 2
?ANI? 3
?DNIS? 4
?ANI?DNIS? 5
?DNIS?ANI? 6
By default,?ANI?DNIS? is selected.
To assign the Outbound Delimiter (?) character

6169-*-Character
Where,

Viewing T1E1 Trunk Status
You can view the status of T1E1 Trunks on Jeeves only. To do this,
• Under T1E1 Configuration, click Status.
• For each T1E1 Trunk, the following settings will be displayed:

• T1E1 Port No.
• Name
• Layer 1
• Layer 2
• Alarms
• Performance Monitoring Counter
• Call Budget Reset Mode
• Call Budget Reset Scheduled (Date)
• Reset Consumed (this is not a status indicator. It is for resetting the Consumed Call Budget manually)
You can also view the T1E1 Trunk Status from the Status link. To view, click the T1E1 link under Status.

T1 RBS Parameters
What’s this?
Some countries like North America support the standard of 1.544Mbps of PCM trunk. This is known as T1 Trunks.
The T1 type of PCM Trunks use Robbed Bit Signaling. ROBBED-BIT signaling is a per-channel signaling technique
for transmitting signaling bits on each channel in a T1E1 facility. The least-significant bit in every 6th transmitted
information frame is removed and replaced by a signaling bit. This technique is also called in-band signaling. The
maximum transmission rate for each bearer channel with ROBBED-BIT signaling is 56 Kbps.
ISDN-PRI signaling is carried on the 24th channel for a 1.544 Mbps connection and on the 16th channel for a 2.048
Mbps connection. There are two types of parameters:
• Line Signaling (ABCD Bits)
• Register Signaling
Line signaling is described by following types:

• E&M Wink Start
• SLT Ground start
• SLT Loop Start
• CO Loop Start
• CO Ground Start
T1 Line signaling type is applicable when the Line Type is programmed as T1 RBS for the T1E1 Port.
The Register signaling supported by SARVAM UCS is DTMF.
T1 Line signaling is applicable when the Line Type is programmed as ISDN_T1_RBS for the T1E1 Port.
Refer chapter “Configuring T1 Trunks” to program Line Type as T1 RBS.
The various Line Signaling Variant are explained below:
E&M Wink Start

Bits A and B are set to 1 to indicate Off-hook. Bits A and B are set to 0 to indicate On-Hook. Bits C and D follow bits
A and B. The application for making calls and receiving calls is explained below:
Making an OG Call
• To transmit Off-Hook, bits A, B, C and D on the transmit channel are set to 1.
• The far end (network) sends a wink (a momentary OFF-Hook for 200ms), that is, bits A, B, C and D on the
receive channel receive a pulse (Active High) of 200ms.
• On receipt of the wink signal, DNIS (Dialed Number Identification Signal) is sent on the speech channels
using the Register Signaling type (DTMF or Decadic (A-Bit) or R1 MFC or R2 MFC).
DNIS is Dialed Number In Service. It is the ISDN number that is being dialed. This is provided by the telco in
the call setup messages. DNIS can be used to provide differentiated service to dialing users.
The call goes through when the called party answers the call.

Receiving an IC Call
• Bit A, B, C and D =1 are received on the receive channel.
• The System sends a wink.
• The far end sends the DNIS in the speech channels using Register Signaling.
Disconnect
• By the Network-Bits A and B on the receive channel are 0. Bits C and D are also 0 in ESF. Following this,
the bits on the transmit channel are set to 0 by the System.
• By the System-Bit A and B on the transmit channel are set to 0. Bit C and D are also set to 0 in case of
ESF. Following this, the bits A and B (C and D in ESF) are received as 0 on the receive channel.
E&M Wink Start FGD

• Bits A and B are set to 1 to indicate OFF-Hook. Bits A and B are set to 0 to indicate ON-Hook. Bits C and
D follow bits A and B (Incase of ESF).
Making an OG Call
• To transmit OFF-Hook, bits A, B, C and D on the transmit channel are set to 1.
• The far end (network) sends a wink (a momentary OFF-Hook for 200ms.), that is, bits A, B, C and D on the
receive channel receive a pulse (Active High) of 200ms.
• On receipt of the wink signal, DNIS (Dialed Number Identification Signal). It is the ISDN number that is
being dialed. This is provided by the telco in the call setup messages. DNIS can be used to provide
differentiated service to dialing users.) is sent on the speech channels using the Register Signaling type.
• The call goes through when the called party answers the call.
• Bit A, B, C and D =1 are received on the receive channel.
• The System sends a wink.
• The far end sends the DNIS in the speech channels using DTMF.
• On expiry of DTMF Inter-digit timer, the System sends a wink again.
• The System goes off hook when the call is answered by the called party.
Disconnect
• By the Network-Bits A and B on the receive channel are 0. Bits C and D are also 0 in ESF. Following this,
the bits on the transmit channel are set to 0 by the System.
• By the System-Bit A and B on the transmit channel are set to 0. Bit C and D are also set to 0 in case of
ESF. Following this, the bits A and (C and D in ESF) are received as 0 on the receive channel.
Please note that while using T1 RBS, only DID is being sent/received and not the CLI/ANI.

E&M Immediate Start/Dial
Same as E&M Wink Start. But here the O/G side sends the DNIS immediately after sending Off-hook on its transmit
channel.
SLT Loop Start

The T1E1 port acts as a SLT port whereas the network acts as CO port. When configured for this type of signaling,
the SLT side (the System) uses the A-bit whereas the CO side (the Network) uses the B-bit. Bits C and D follow A
and B bits.
Direction State A B C D
Transmit ON-Hook 0 1 0 1
Transmit OFF-Hook/Loop Closed 1 1 1 1
Receive ON-Hook 0 1 0 1
Receive OFF-Hook 0 1 0 1
Receive Ringing 1 1 1 1
Making an OG Call
• To transmit OFF-Hook, bits A (and Bit C if ESF) on the transmit channel is set to 1.
• The far end provides dial tone. There is no signaling change.
• The SARVAM UCS sends the DNIS (Dialed Number Identification Signal) on the speech channels using
the DTMF signals.
• Bit B (and bit D if ESF) toggles as per the ringing pattern.
• The Network sends the DNIS on the speech channel using DTMF signals.
• The System detects toggling of bit B. When the called station of the System answers, the System transmits
Off-hook state by changing bit-A from 0 to 1.
Disconnect
By the Network:
• No indication from the Network. The System will detect error tone to detect a disconnect from the network.
On detecting on-hook from the network, the System transmits on-hook by setting Bit A from 1 to 0.
By the System:
• The System transmits on-hook by setting Bit A from 1 to 0.

SLT Ground Start
The T1E1 port acts as a SLT port whereas the network acts as CO port. When configured for this type of signaling,
the SLT side (the System) uses the A-bit (however, the System can use B-bit but generally it is ignored by the
Network) whereas the CO side (the Network) uses the A and B-bits both. Bits C and D follow A and B bits.
Direction State A B C D
Transmit ON-Hook/Loop Open 0 1 0 1
Transmit Ground on Ring 0 0 0 0
Transmit OFF-Hook/Loop Closed 1 1 1 1
Receive ON-Hook/No TIP Ground 1 1 1 1
Receive OFF-Hook/TIP Ground 0 1 0 1
Receive Ringing 0 0 0 0
Making an OG Call
• To transmit Off-hook, bits A (and Bit C if ESF) and B (and Bit C if ESF) on the transmit channel is set to 0.
• The network detects this change and goes off-hook. The A-bit on the receive channel goes from 1 to 0.
The B-bit is set to 1.
• Bit A and Bit B are set to 1.
• Network sends dial tone.
• System detects dial tone and sends DNIS (DTMF digits) on the corresponding speech channel.
• Bit A (and bit C if ESF) goes from 1 to 0.
• Bit B (and bit D if ESF) toggles as per the ringing.
• The Network sends the DNIS on the speech channel using DTMF signals.
• The System detects toggling of bit B and the seizure on Bit A.
• When the called station of the System answers, the PBS transmits Off-hook state by changing bit-A from 0
to 1.
• The Network stops toggling of bit B. Bit B is set to 1.
Disconnect
• By the Network-A bit on the receive channel of the System goes from 0 to 1. On detecting on-hook from
the network, the System transmits on-hook by setting Bit A from 1 to 0.
• By the System-The System transmits on-hook by setting Bit A from 1 to 0. On detecting on-hook from the
system, the network transmits on-hook by setting Bit A from 0 to 1.

• CO Loop Start or CO Ground Start is used when the SARVAM UCS is connected to the Network, that is,
when the T1E1 port is configured for Terminal mode (Connection mode).
• Whereas SLT Loop Start or SLT Ground Start is used when the SARVAM UCS is connected to another
SARVAM UCS, that is, when the T1E1 port is configured for Network mode.
CO Loop Start-This is complementary to SLT Loop Start explained above.

CO Ground Start-This is complementary to SLT Ground Start explained above.

• Click T1 RBS Parameters to open the page and configure the following:
• Line Signaling Variant: Select the T1 Line Signaling Variant from following options:
• SLT Loop Start
• CO Loop Start
• SLT Ground Start
• CO Ground Start
• E&M Wink Start
Default: E&M Wink Start FGD.
• Wink Timer (milliseconds): Wink timer refers to the momentary Off-Hook condition to acknowledge
end of making an outgoing call. The Wink Timer ranges from 0001 ms to 9999 ms. Default: 160 msec.
• Wink Wait Timer (milliseconds): Wink Wait Timer signifies the maximum time the system should wait
before sending a wink start signal after an incoming seizure is detected. Wink Wait Timer ranges from
0001 to 9999 msec. Default: 30msec.
Ensure that this timer is greater than the Wink Wait Timer of the other end.
• Wait Wink Timer (milliseconds): Wait Wink Timer signifies the time for which SARVAM UCS will wait
for receiving the DNIS after sending the outgoing seizure signal. Wait Wink Timer ranges from 001 to
999 msec. Default: 5000 msec.
Make sure that this timer is greater than the Wait Wink Timer of the other end.
• Delay Duration (milliseconds): This duration signifies the time after which the DNIS information is to
be sent while making an outgoing call. Range of the Delay Duration is from 0001 to 9999 msec.
Default: 100 msec.
• Start Delay Timer: Start Delay Timer signifies the time for which SARVAM UCS waits for receiving
DNIS from the network. This timer is loaded on receiving the Off-hook (I/C Seizure) on the receive
channel (while receiving an incoming call). The Start Delay Timer ranges from 0001 to 9999 ms.
Default: 20 msec.
• Register Signaling Variant: The Register Signaling Variant for T1/E1 Ports is set as DTMF.

• Inbound ANI/DNIS Format: Select the Inbound ANI/DNIS Format for T1/E1 Ports from the following
options:
• ANI
• DNIS
• ?ANI?
• ?DNIS?
• ?ANI?DNIS?
• ?DNIS?ANI?
• Inbound Delimiter (?) Character: Define the Inbound Delimiter Character in this field. Characters
• Outbound ANI/DNIS Format: Select the Outbound ANI/DNIS Format for T1/E1 Port from the following
options:
• ANI
• DNIS
• ?ANI?
• ?DNIS?
• ?ANI?DNIS?
• ?DNIS?ANI?
• Outbound Delimiter (?) Character: Define the Outbound Delimiter Character in this field. Characters
Configuring T1E1 Port Parameters using Telephone

• To assign a Line Signaling Variants to the T1E1 port, dial:

6181-1-T1E1-T1 Line Signaling Variants
6181-2-T1E1-T1E1-T1 Line Signaling Variants
6181-*-T1 Line Signaling Variants
Where,
T1 Line Signaling Variants Meaning
1 SLT Loop Start
2 CO Loop Start
3 SLT Ground Start
4 CO Ground Start
5 E&M Immediate Start/Dial
6 E&M Wink Start
7 E&M Wink Start FGD
Default: E&M Wink Start.

• To set the Wink Timer, dial:
6182-1-T1E1-Wink Timer
6182-2-T1E1-T1E1-Wink Timer
6182-*-Wink Timer
Where,
Wink Timer is from 0001 to 9999 ms.
By default, T1 Wink Timer is 0200 ms.
• To configure the T1 wink wait timer for T1E1, dial:

6183-1-T1E1-Wink Wait Timer
6183-2-T1E1-T1E1-Wink Wait Timer
6183-*-Wink Wait Timer
Where,
Wink Wait Timer is from 0001 to 9999 ms.
By default, T1 Wink Wait Timer is 0200 ms.
• To program the T1 wait wink timer for T1E1, dial:

6184-1-T1E1-Timer
6184-*-Timer
Where,
Timer is from 0001 to 9999 ms.
By default, T1 Wait Wink Timer is 0200 ms.
To program the T1 delay duration for T1E1, dial:

6185-1-T1E1-Delay Duration
6185-2-T1E1-T1E1-Delay Duration
6185-*-Delay Duration
Where,
Delay Duration is from 0001 to 9999 ms.
By default, T1 Delay Duration is 0140 ms.
• To program the T1 Start Delay Timer for T1E1, dial:

6186-1-T1E1-Start Delay Timer
6186-2-T1E1-T1E1-Start Delay Timer
6186-*-Start Delay Timer
Where,
Start Delay Duration is from 001 to 255 seconds.
By default, T1 Start Delay Timer is 020 seconds.
• To assign a Register Signaling Variant to the T1E1 port, dial:

6161-1-T1E1-T1 Register Signaling Variant
6161-2-T1E1-T1E1-T1 Register Signaling Variant
6161-*-T1 Register Signaling Variant
Where,

T1 Register Signaling Variant Meaning
1 T1 RBS DTMF
T1 RBS DTMF: DNIS is transmitted in the corresponding speech channel using the DTMF signals as
per the ITU-T Q.23.
• To select the Inbound ANI/DNIS Format, dial:

6166-1-T1E1-Code
6166-2-T1E1-T1E1-Code
6166-*-Code
Where,
Code is
ANI 1
DNIS 2
?ANI? 3
?DNIS? 4
?ANI?DNIS? 5
?DNIS?ANI? 6
By default, ?ANI?DNIS? is selected.
• To assign the Inbound Delimiter (?) character, dial:

6167-*-Character
Where,
• To assign the Outbound ANI/DNIS Format, dial:

6168-1-T1E1-Code
6168-2-T1E1-T1E1-Code
6168-*-Code
Where,
Code is
ANI 1
DNIS 2
?ANI? 3

?DNIS? 4
?ANI?DNIS? 5
?DNIS?ANI? 6
By default, ?ANI?DNIS? is selected.

• To assign the Outbound Delimiter (?) character, dial:
6169-*-Character
Where,
• Exit SE mode.

Configuring E1 Trunks
What’s this?
Digital Signal Level 1 (T1E1) trunks use Bit-Oriented Signaling (BOS) and multiplexes 24 channels (T1 service) or
32 channels (E1 service) into a single data stream. T1E1 can be used for voice or voice-grade data and for data-
transmission protocols. T1 trunk service multiplexes 24 channels into a single 1.544-Mbps data stream. E1 trunk
service multiplexes 32 channels into a single 2.048-Mbps stream. Both T1 and E1 provide a digital interface for
trunk groups.
Signaling Modes
Common Channel Signaling (CCS) is an industry-standard technique where any one of a group of channels carries
the signals for the other channels. Matrix uses the 24th channel of a group for signaling. This signaling technique
differs from 24-channel signaling. When the system is configured for Facility-Associated Signaling, 24-channel
signaling uses the 24th channel in a T1E1 facility to carry signals. This technique also is called clear channel, out-
of-band or alternate voice data (AVD) signaling.
Channel Associated Signaling (CAS) is similar to common-channel signaling and is used only when the Bit Rate is
2.048 Mbps (the trunk is used with an E1 interface). Signaling is carried on the 16th channel.
Common-channel signaling and channel associated signaling provide a maximum transmission rate of 64 Kbps for
bearer channels.
ROBBED-BIT signaling is a per-channel in-band signaling technique for transmitting signaling bits on each channel
in a T1E1 facility. The least-significant bit in every 6th transmitted information frame is removed and replaced by a
signaling bit. This technique is also called in-band signaling. The maximum transmission rate for each bearer
channel with ROBBED-BIT signaling is 56 Kbps.
ISDN-PRI signaling is carried on the 24th channel for a 1.544 Mbps (T1) connection and on the 16th channel for a
2.048 Mbps (E1) connection.
QSIG is an ISDN based protocol for signaling between nodes of a Private Integrated Services network. Any of the
common trunks, except for PCOL (Personal Central Office Line) trunks, can be analog or digital. (PCOL trunks can
only be analog.) Administering a digital trunk group is very similar to administering its analog counterpart, but digital
trunks must connect to a T1E1 port and this port must be administered separately.
User interface for E1_PRI and E1_CAS channels:
• In case of ISDN_E1_PRI and ISDN_E1_CAS the protocol supports 32 Channels ranging from 00 to 31, out
of which 2 channels (channel no. 00 and 16) are used for framing/signaling. So effectively user has 30
channels for OG/IC calls.

• For better understanding of the user the channel IDs are mapped as shown below. Thus for the E1_PRI
and E1_CAS the T1E1 port supports total 30 channels ranging from 01 to 30, which you can use for
making and receiving calls.
Channel ID as
00 01 02 03 04 14 15 16 17 18 31
per Protocol
Channel ID for
01 02 03 04 14 15 16 17 30
User Interface
Mapping of Actual Channel ID with User Interface (E1_PRI and E1_CAS)
• The system Debug is for trouble shooting and so the channel ID in debug will be as the actual channel ID
as supported by the E1_PRI and E1_CAS protocols.
• Similarly, in case of T1 PRI, Protocol supports 24 channels (from 01 to 24), in which channel no. 24 is used
for the signaling, so effectively there are 23 Voice channels are available.
• But in case of T1 RBS, Protocol supports 24 channels (from 01 to 24) and the protocol doesn’t consume
any channel for signaling so that there are total 24 channels available for the users.

• Click Port Parameters to open the page.
• Select the T1E1 Port number you want to configure by clicking the respective tab, and program the
following port parameters:
• SARVAM UCS will assign the Hardware Slot-Port automatically, when any card is inserted in the
system.

Hardware slot is the number of the Universal slot of SARVAM UCS in which the T1E1 Card is inserted.
Range of slot number is 01-16. Port is the number of T1E1 hardware port on the card to which the
T1E1 line is connected. Range of Port is from 00-99.
Port is 00–00.
• Keep the Enable T1E1 Port flag enabled.
Clear the flag, only if you do not want to use this port. By default, it is enabled.
• You may assign a Name to the T1E1 Port for identification of the port. The Name may consist of a
maximum of 12 characters. By default, it is blank.
• Select the Carrier type as E1. The Carrier type will automatically be assigned to the port when you
select the region. You may it if required.
• Select Signal Type. Signal Type signifies the type of signaling to be used on E1 line. The E1 Signaling
supported are:
• PRI
• CAS
• QSIG
• E&M
Default: PRI
If you select PRI or QSIG as the Signal Type, configure the PRI/QSIG Parameters.
If you select CAS as the Signal Type, configure the CAS Parameters.
If you select E&M as the Signal Type, configure the E&M Parameters.
• Set the Orientation Type. Select the orientation type from the following options:
• Terminal
• Network
• Tie-Line
Default: Terminal
• If you have set Terminal as the Orientation Type, you must select the type of network with which the
port is to be Interfaced With. Select the type of network from the following:
• Public ISDN
• Private ISDN
If you select Trunk, the system will treat all incoming calls as external calls landing on the trunk. The
calls will be routed as per the Trunk Feature Template assigned to the T1E1 Port.
Feature Template to the T1E1 Port.

Station Advanced Feature Template assigned to the T1E1 Port.
• Line coding is a pattern that data assumes as it is propagated over a communication channel. Select
the Line Coding Mechanism from the following:
• AMI-Basic
• HDB3
• NRZ (Fiber Optic)
Default: HDB3.
• Framing means to form a set of 24 or 32, 8 bits time slot that is to be treated as single transmission
unit. The Framing Modes supported by SARVAM UCS are:
• CEPT1 MF (No CRC)
• CEPT1 MF (Forced CRC)
• CEPT1 MF (Auto CRC)
Default: CEPT1 MF (Auto CRC).
• Configure Outgoing (OG) Reference ID. By default, OG Reference ID is 00.
• Configure Incoming (IC) Reference ID for working hours, non-working hours and break hours. By
default, IC Reference ID is 00.
• Assign Trunk Feature Template to the T1E1 Port. Trunk Feature Template is a set of general features
that define the behavior of a Trunk Port. By default, Template 01 is assigned to all T1E1 Ports.
• Station Basic Feature Template assigned to the T1E1 Port is displayed in this field. Station Basic
Feature Template is a set of general features that define the basic behavior of a station. By default,
Template 01 is assigned to all T1E1 Ports.
• Station Advanced Feature Template assigned to the T1E1 Port is displayed in this field. Station
Advanced Feature Template is a set of advanced features, to be applied on extensions such as CLIP,
Floor Service, Walk-in Class of Service. By default, Template 01 is assigned to all T1E1 Ports.

• Select Priority for the T1E1 Port. Priority is the precedence given to certain trunks and extensions over
others in being answered by the destination extension. You can select from 1 to 9. By default, Priority
5-Normal is set for all T1E1Ports.
• Assign a Cost Factor to the T1E1 Port. By default, all the T1E1 Ports are assigned Cost Factor 01.
• ISDN glare occurs if the system initiates an outgoing call on a B-Channel at the same time the network
initiates an incoming call on that same B-channel. You may configure the Glare Option as Proceed or
Held Back. While processing a glare condition, the configured Glare Option on T1E1 port will be
considered.
• Select Category (Logical Partition) for the T1E1 Port. You may select from the following options:
• 1
• 2
• 3
By default, 1 is selected for all T1E1 Ports.
• Idle Code is the 8-bit sequence that occupies the time slot on a E1/T1 trunk channel when it is not
being used. By default, 127 is configured as the Idle Code.
Display Called Party Number as CLI check box. By default, Display Called Party Number as CLI
option is disabled.
This option is useful when a single T1E1 line connection and Operator are shared by more than one
• you configure the names and corresponding numbers of the organizations sharing the line in the
incoming call, SARVAM UCS matches the number with the numbers in the Global Directory. If a match
is found SARVAM UCS displays the company name configured for that entry to the Operator, that is,
the CLI will display the called party number and name.
After the Operator answers the call, the CLI will change and display the calling party number and name
(if configured in the Global Directory).
• Select Return Call to Original Caller (RCOC) flag to enable this feature on the T1E1 Port. By default,
RCOC flag is disabled.

• In Channel Reserved for Data Call, configure the channels you want to reserve for Data Calls. By
default, 00 channels are reserved.
• In Channel Reserved for Outgoing Call, configure the channels you want to reserve for making
outgoing calls. By default, 30 channels are reserved.
• In Channel Reserved for Incoming Call, configure the channels you want to reserve for receiving
incoming calls. By default, 30 channels are reserved.
When a caller dials the trunk access code or selective trunk access code for dialing the number directly
on the trunk port, the caller waits for the dial tone before dialing the number. But some exchanges do
not give Dial Tone for the T1E1 Port. For Example, when T1E1 port as E1CAS type is used in Delhi, it
is observed that the exchange does not give dial tone when direct dialing on the trunk is used.
• Enable the Feed Dial Tone flag. SARVAM UCS will provides the dial tone to the caller when the T1E1
Port is accessed.
It is applicable only when Online dialing is used as, Store and Forward dialing, the dial tone is given to
the caller. The dial tone is played as per the Dial Tone Timer.
By default, Feed Dial Tone flag is disabled.
• When dial tone flag is disabled, user will hear the dial tone of the exchange if provided, otherwise, user
will hear the silence.
• If the user is making the call from the SLT port and dial tone is not provided by the exchange, user will
not know when to start dialing the number. In this case, it is possible that some digits are not out dialed
on the port and wrong number is dialed out because system will out dial the number only if Outgoing
call Acknowledge is received but the user is not aware of this condition. Hence, it is required to enable
this flag, if exchange is not providing the dial tone.
• When Online dialing or Store and Forward dialing are used, some exchanges do not provide any tone
while routing/processing the call. Thus, the caller does not know whether the call is being processed or
not as there is silence. In such case, You must enable the Feed Routing Tone flag, SARVAM UCS will
play the routing tone to the caller. SARVAM UCS stops playing the routing tone when an alert message
or connect message or disconnect message is received from the T1E1 Card.
By default, the Feed Routing Tone flag is disabled.
• To customize the pulse width option and set the pulse shapes configure the Custom Pulse
parameters. SARVAM UCS generates pulse shapes which match the country standard, where it is
installed. However, if the standard pulse shape does not match, SARVAM UCS enables you to
customize the pulse width to match your exact requirements.
To use customize pulse width option and set the pulse shape in 1 to 4 phases, keep the T1/E1 Custom
Pulse Width (CPW) flag enabled.
• Word 1: Set the pulse width for setting pulse shape in the 1st phase. The range of Custom Pulse
• Word 2: Set the pulse width for setting pulse shape in the 2nd phase. The range of Custom Pulse

• Word 3: Set the pulse width for setting pulse shape in the 3rd phase. The range of Custom Pulse
• Word 4: Set the pulse width for setting pulse shape in the 4th phase. The range of Custom Pulse
• Speech
• 3.1 KHz Audio
• The Overlap Receiving Timer is relevant while receiving the called party number information in
overlap receiving mode. It is not relevant for the port in overlap sending mode.
Range of Overlap Receiving Timer is from 01 to 99 seconds. Default: 15 seconds.
• Configure Pause Timer for theT1E1 Port. Range of Pause Timer is from 1 to 9 seconds. By default, it
is set to 3 seconds for all T1E1 Ports.
This Timer is required to insert delay between the digits while dialing out DTMF digits on the T1E1 port.
One of the applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage
Dialing”.
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the SARVAM
UCS will out dial the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
responds by sending SETUP ACK (Acknowledgment), and dial tone is played to the caller. The time
taken by the exchange to respond to the SETUP message may vary from exchange to exchange. Set
the SETUP Response Timer (sec) as per the time taken by the network to respond to the SETUP
message and play dial tone to the caller.
• Configure DTMF On Time for theT1E1 Port. Range of DTMF On Time is from 051 to 255 ms. By
default, it is set to 102 ms for all T1E1 Ports.
Stage Dialing”.
• Configure DTMF Inter Digit Pause Timer for the T1E1 Port. Range of Inter Digit Pause Timer is from
051 to 255 ms. By default, it is set to 102 ms for all T1E1 Ports.

• Configure the Minimum ON Time (msec) for which the DTMF signal should be present in order to be
detected. The valid range of this time is 10 to 200 milliseconds. By default, Minimum ON Time is set to
20 milliseconds.
• Configure the Minimum OFF Time (msec). This parameter signifies the minimum time period between
successive DTMF digits. The valid range of this time is 10 to 200 milliseconds. By default, Minimum
OFF Time is set to 20 milliseconds.
Configure the Minimum Level (dB) for the DTMF digit to be considered as valid. The valid range of
this time is 0 to -36.5 dB. By default, Minimum levels is set to -36.5dB.
• If SARVAM UCS is to be used as a Gateway, enable Gateway Application-Answer Signaling on the

T1E1 Port and configure DTMF String. By default, Gateway Application-Answer Signaling is disabled
and CCC is configured as DTMF String.
• Configure Call Budget parameters for the T1E1 Ports. Call Budget is an expense control feature of
SARVAM UCS that allows you to keep track of the cost of phone calls made from the T1E1 Port.
• Type: Select the type of Call Budget, that is, Amount or Minutes or Calls to be applied on the T1E1
Port. By default, no Call Budget type is selected.
• Amount: If you selected 'Amount' as the Call Budget Type, enter the Budget Amount in this field.
By default the Amount is set to 999999.
• Minutes: If you selected 'Minutes' as the Call Budget Type, enter the number of Minutes in this
field. By default the number of minutes is set to 999999.
• Calls: If you selected 'Calls' as the Call Budget Type, enter the number of Calls in this field. By
default the number of calls is set to 9999.
• Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes/Number of Calls to
be reset on a particular date of every month.
Amount/Minutes/Number of Calls to be reset every month. You may select 'Daily' if your plan
suggests so.
• Call Back: This parameter is related to the ‘Call Back on Trunk Port’ feature. If you want to enable the
'Call Back on Trunk Port' feature on this T1E1 Port, configure the following parameters:
• Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, this
flag is disabled on all trunk port types. By default, the flag is disabled.
before applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set
to 10 seconds.
• Call Back Mode: Select from the following options how a ‘Call Back’ call answered by the remote
party should be routed:
• Built-in Auto Attendant

• Operator
• Call Back on: This parameter allows you to select if the call back should be made to the same
number that was received or to a different number. If you want the call back to be made to the same
number select the ‘CLI number”. If you want the call back to be made to a different number, select
‘Alternate Number’.
Number List 15 is also assigned to all T1E1 Ports as well as all other Trunk port types. If you want
the same numbers strings to be programmed commonly for all T1E1 Ports and Trunk Port types,
retain this list.
If you want a different set of number strings to be programmed for this T1E1 Port, select a different
Number List, and assign it to the T1E1 Port.
corresponding index in the Outgoing Number List a number to which the call back is to be made.
For example, for the number string programmed at Index 1 in the Incoming Number List, a
corresponding number string at the same Index, Index 1, should be programmed in the ‘Outgoing
Number List’.
assigned to all T1E1 Ports as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this T1E1 Port.
• Call Back from: This parameter determines the trunk port to be used to make the call back. The
call back can be made using the Same Port or an “OG Trunk Bundle Group”.
Select ‘Same Port’ if you want the call back to be made using the same port on which the missed
call is received. If you select OGTB Group, the call back will be made using the OGTB Group, which
you have defined.

• FDL is used for communicating general maintenance information or for transmitting user defined
information within the T1 link. General maintenance information is in the form of Performance Message
Report which is generated by the SARVAM UCS Card T1E1PRI and depending upon the FDL Protocol,
the Performance Message Report is sent every second or on request.
Select the FDL flag to enable. This parameter is applicable only if Framing = ESF. If the Network
(Public or Private) to which the SARVAM UCS is connected does not support FDL then FDL will be
disabled. By default, the T1 FDL is disabled.
If you have enabled FDL, configure the FDL Protocol. SARVAM UCS supports ANSI T1.403 and AT&T
54016 protocols of reporting the performance monitoring. By default, the T1 FDL Protocol is ANSI
T1.403.
For more information, see “E1/T1 Maintenance”
• Enter appropriate Debug Code (Level 1 to 4), to obtain debug information of various parts of T1E1
Card on the COM Port. By default, debug is off for all T1E1 ports for all levels.
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from
000 to 255. Code value '000' for each level will turn off that level's debug.
Level and Code for T1E1 Port are as specified below:
Level 1:
001 CAS
002 MFC R2
004 CAS DSP
008 Layer 4
000 Debug Off
Level 2:
001 Alarms
002 Counters

004 ABCD Bits
008 FDL
HDLC (D
016
Channel)
000 Debug Off
Level 3:
001 Primitives
002 State
004 Variables
008 SVC Primitives
016 LAP Debug
032 NLS Debug
064 Flow Debug
000 Debug Off
Level 4:
001 OS Task
002 NI Debug
000 Debug Off

PRI/QSIG Parameters
• Under T1E1 Configuration, click PRI/QSIG Signaling and configure the PRI/QSIG parameters.
• ISDN Switch Variant: ISDN supports a variety of service provider switches. Different countries use
specific type of ISDN switch. This switch is designed using ISDN standard protocol. The type of switch
determines various factors such as how many ISDN devices would be handled, which B-channel will
support voice, video, data etc. Select the ISDN Switch Variant from the list. By default, ETSI NET5 is
selected as the ISDN Switch Variant.
• Offer continuous Bearer Channel Mapping(01-30): Select this check box for continuous Bearer
Channel Mapping for E1-QSIG/E1-PRI.
• D-Channel: Enter the Channel number that is used for Data signaling. By default, D-Channel is 16.
Valid Range is 1 to 31.
• Send Called Party Number Using: Select the appropriate option from the following for Send Called
Party Number Using:
• Called Party Number IE (Information Element)
• Keypad Facility IE (Information Element)
By default, Called Party Number IE is selected.
• Dialing Type for Called Party Number: Select the type of dialing supported by your exchange. You
can select:
• Enbloc
• Digit -by - Digit
• Any
By default, Any is selected as the Dialing Type for Called Party Number.
• Caller - Type of Numbering Plan (TON): Select the appropriate option from the following for sending
the type of numbering plan of the calling party:
• Unknown
• International

• National
• Subscriber
• Abbreviated
• Reserved
Default: Unknown.
• Caller- Numbering Plan Identification (NPI): Select the appropriate option from the following for
sending the numbering plan identification of the calling party:
• Unknown
• ISDN Numbering
• Data Numbering
• Telex Numbering
• Private
• Reserved
• Called - Type of Numbering Plan (TON): Select the appropriate option from the following for sending
the type of numbering plan of the called party:
• Unknown
• International
• National
• Subscriber
• Abbreviated
• Reserved
Default: Unknown.
• Called - Numbering Plan Identification (NPI): Select the appropriate option from the following for
sending the numbering plan identification of the called party:
• Unknown
• ISDN Numbering
• Data Numbering
• Telex Numbering
• Private
• Reserved
• Screening Indicator: Select the option as provided to you by your exchange. You can select from:
• User provided, not screened
• User Provided, verified and passed
• User Provided, verified and failed
• Network Provided
By default, User provided, not screened.
• Receive Equalization Mode: You can set the Receive Equalization Mode as Auto or Manual. By default,
Auto is selected as the Receive Equalization Mode.
• Receive Equalization Parameters: This field increases the strength of incoming signals by a fixed
amount to compensate for line losses. Select the required option from the list. By default, the receive
equalization parameters of T1E1 is 8dB.

• Feed Inband Tones on T1E1-NT, before sending DISCONNECT: Select this flag, if you want to feed
inband tones on T1E1-NT before sending DISCONNECT message. This flag is applicable only when
T1E1 port is configured as 'Network'. When this flag is enabled, inband tones shall be feed for 15 seconds
(fixed, non programmable) before sending DISCONNECT message.
When this flag is disabled, inband tones (Busy/Error as applicable for the state of the call) shall not be feed
before sending the DISCONNECT message. However when DISCONNECT message is sent from T1E1-
NT port, inband tones will always be sent with 'progress indicator 8'. By default, this flag is disabled.
• Send SETUP ACK with PI: Select this flag if you want the system to send PI (Progress indicator) element
in Setup Ack message to other end. This option is relevant only when call landing on the T1E1 port are to
be routed to SLT/CO/Mobile/DKP port. If T1E1 port is configured as 'Network’, by default it is enabled and
if T1E1 port is configured as 'Terminal’, by default it is disabled.
• Send PROCEED with PI: Select this flag if you want the system to send PI (Progress indicator) in
Proceed message to other end. This option is relevant only when call landing on the T1E1 port are to be
routed to SLT/CO/Mobile/DKP port. If T1E1 port is configured as 'Network’, by default it is enabled and if
T1E1 port is configured as 'Terminal’, by default it is disabled.
• Send PROGRESS with PI: Select this flag if you want the system to send PI (Progress indicator) in
Progress message to other end. This option is relevant only when call landing on the T1E1 port are to be
routed to SLT/CO/Mobile/DKP port. By default it is enabled.
• Send ALERT with PI: Select this flag if you want the system to send PI (Progress indicator) in Alert
message to other end. This option is relevant only when call landing on the T1E1 port are to be routed to
SLT/CO/Mobile/DKP port. If T1E1 port is configured as 'Network’, by default it is enabled and if T1E1 port
is configured as 'Terminal’, by default it is disabled.
• Transparently pass PI in case of ISDN to ISDN / SIP call: Select this flag, if you want the system to just
pass the PI as received to the other end.This option is relevant only when call landing on the T1E1 port are
to be routed to ISDN or SIP ports. By default it is enabled.

E1 CAS Parameters
Under T1E1 Configuration, click E1 CAS Signaling and configure the CAS parameters.
• E1 Line Signaling Variant: The E1 Line Signaling Variant supported by SARVAM UCS is ITU T Q.400-
Q.490.
• E1 Register Signaling Variant: Select the appropriate option from the following for the E1 Register
Signaling Variant:
• DTMF - DNIS/ANI is transmitted in the corresponding speech channel using the DTMF signals as per
ITU-T Q.23.
• MFC R2 - DNIS/ANI is transmitted in the corresponding speech channel using the MFC R2 signals as
per ITU-T Q.400-Q490.
By default, E1 Register Signaling Variants is MFC R2.
Register Signaling Parameters
• Forward Tone Maximum On Timer (T1) (Seconds): This timer signifies the maximum time for which the
forward signal remains ON, from the outbound end.
The range of Forward Time Maximum ON Timer is from 01 to 99 seconds. Default: 15 seconds.
The DSP (Digital Signaling Processor) device will send the forward tone for this timer and will expect
backward signal within this timer. If no backward signal is received during this time, a timeout condition will
occur in this case, an alert signal will be sent to the CPU Card, error tone be issued to the calling party and
a clear forward signal will be sent on the line.
• Forward Tone Maximum Off Timer (T2) (Seconds): This timer signifies the maximum time between two
out going forward signals. During this time the forward tone will remain OFF. If the outbound end does not
send a forward signal for this time, the inbound end will interpret it as per its condition and shall take action
accordingly.

The range of Forward Time Maximum OFF Timer is from 01 to 99 seconds. Default: 24 seconds.
• Maximum Compelled Cycle Timer (T3) (Seconds): This timer signifies the maximum time within which
one compelled signaling cycle shall end.
The range of Maximum Compelled Cycle Timer is from 01 to 99 seconds. Default: 15 seconds.
• Pulse Duration for Pulse Signal (Milliseconds): Backward signals A-3, A-4, A-6 and A-15 are pulsed to
the outbound end. Pulse duration of these signals vary from country to country.
The range of Pulse Duration for Pulsed Signals is from 001 to 999 ms. Default: 150 ms.
It is recommended that tolerance be fixed at +/- 25 ms.
• Pulsed Signal Maximum Wait Timer (Seconds): This timer signifies the time for which the outbound end
waits for the pulsed signal. If the pulsed signal is not received during this time, the compelling signaling is
said to be complete.
The range of Pulsed Signal Maximum Wait Timer is from 01 to 99 seconds. Default: 15 seconds.
• First Forward Tone Wait Timer (Seconds): This timer signifies the time between receipt of line seizure
signal and the first forward signal.
The range of the First Forward Tone Wait Timer is from 08 to 24 seconds. Default:15 seconds.
• Minimum MF Signal Persist Timer (Milliseconds): This timer signifies the minimum time for which the
forward/backward signal shall be sustained on the line by the receiving end.
The range of Minimum MF Signal Persist Timer is from 001 to 255 ms. Default: 20 ms.
Outbound Parameters
• Dialed Number Identification Signal (DNIS) End Type: This parameter is applicable only when DNIS
length is set to 99 (that is, variable). The outbound end indicates end of DNIS using a group I tone or using
time out.
Range of DNIS End Type is from 00, 11 to 15; where 00 indicates End of DNIS as time out. 11 to 15
indicates group I tone to declare End of DNIS. Default: 15.
• Address Number Information (ANI) Send Position: This parameter signifies the number of DNIS digits
after which address number information is to be sent. Address number information is usually sent on
receiving the backward tone Send next digit or Send next ANI digit.
If send next address number information tone is received then this parameter is not applicable. But if same
tone is used by the inbound end to request the next ANI digit and the next DNIS digit, ANI is sent after the
number of digits as set in this field.
The range of ANI Send Position is from 00 to 99. Default: 00.
• Is Address Number Information (ANI) Available: This parameter indicates Group A tone (received from
the inbound tone) that is to be interpreted as a question by the inbound end asking the outbound end
whether the outbound end has ANI digits to be sent.

The range of Is ANI Available, Group A tone is from 01 to 15. Default: 05.
• Positive Response to Is ANI Available: This parameter signifies the Group 1 tone that the outbound end
will send to the inbound end as a response to Is ANI Available tone from the inbound end. The tone
defined in this parameter indicates the Group 1 tone with which the Outbound end will respond to the
inbound end to indicate that it has ANI digits to be sent.
The range of Positive Response to Is ANI Available, Group 1 tone is from 01 to 15. Default: 01.
• Negative Response to Is ANI Available: This parameter signifies the Group 1 tone that the outbound end
will send to the inbound end as a response to Is ANI Available tone from the inbound end. The tone
defined in this parameter indicates the Group 1 tone with which the Outbound end will respond to the
inbound end to indicate that it does not have ANI digits to be sent.
The range of Negative Response to Is ANI Available, Group 1 tone is from 01 to 15. Default: 10.
• ANI End Tone Presentation Allowed: This parameter signifies the Group 1 tone used to signify end of
ANI digits with Presentation Allowed.
The range of End of ANI with Presentation Allowed, Group 1 tone is from 00, 11 to 15. Default:15.
• ANI End Tone Presentation Restricted: This parameter signifies the Group 1 tone used to signify end of
ANI digits with Presentation Restricted.
The range of End of ANI with Presentation Restricted, Group 1 tone is from 00, 11 to 15. Default: 00.
Inbound Parameters
• Dialed Number Identification Signal (DNIS) End Type: This parameter is applicable only when the DNIS
length is set to 99 (that is, variable). The outbound end indicates end of DNIS using a group I tone or using
time out.
The range of DNIS End Type is from 00, 11 to 15; where 00 indicates End of DNIS as time out, 11 to 15
indicates group I tone to declare End of DNIS. Default:15.
• Dialed Number Identification Signal (DNIS) Digit Length: This parameter signifies the number of DNIS
digits required by inbound end to indicate the Called party number during MFC R2 signaling.
The range of DNIS Length is from 01 to 99. Default: 99.
DNIS Digit length (01 to 98) will be expected by the inbound end. (Practical value would be 01 to 10)
DNIS Digit length 99 indicates DNIS length is variable. Further action is taken after timeout or on receipt of
I-15. Refer parameter 'DNIS End Type (Inbound)'.
• Address Number Information (ANI) Request Position: The inbound end may or may not request ANI
digits. It may request ANI digits after receiving the first DNIS or after receiving second DNIS or even after
receiving all the DNIS digits.
The range of ANI Request Position is as follows:
ANI Request Meaning
00 Never request ANI digits

ANI Request Meaning
01-98 Request ANI digits on receipt of these

many DNIS digits
99 Request after receiving all the DNIS

digits (complete DNIS)
Default: 99.
• Address Number Information (ANI) Length: This parameter signifies the number of ANI digits that would
be expected by the inbound side as Calling Party Number during MFC R2 signaling. This parameter at the
inbound side guides the inbound register to switch from requesting ANI digits back to requesting DID
digits.
The range of ANI Length is from 00 to 99. Default: 99.
• ANI Length = 00, ANI is not sent by the Outbound end.
• ANI Length = 99, ANI Length is variable. If ANI length is variable, the logic waits for End of ANI from the
outbound side. The inbound end will sense for I-12 and I-15. I-12 is used to signify that no ANI digits
are available whereas I-15 is used to signify end of ANI digits. Some countries like China use I-15 to
signify both the events viz. End of ANI and no ANI digits available.
• Ask Address Number Information (ANI): This parameter specifies the backward group A tone used to
ask the outbound end whether it has ANI digits to be sent. This parameter is also known as Request ANI
Category.
The range of Ask ANI is from 00, 01 to 15. If no tone is sent by the inbound end, set this parameter to 00.
For India, this parameter is set to 04. Default: 05.
• Positive Response to Ask ANI: This parameter specifies that the Group 1 forward tone is to be received
by the inbound end from the outbound which in turn indicates that outbound end has ANI digits to be sent.
This parameter is also known as ANI category.
The range of Positive Response to Ask ANI is from 01 to 15. Default: 01.
For example, In India I-1 or I-10 is sent by the outbound end. In Kuwait, I-6 is sent. This parameter cannot
be zero because; Is ANI Available request will be made by the inbound end only if the country supports this
protocol.
• Negative Response to Ask ANI: This parameter specifies the Group 1 forward tone to be received by the
inbound end from the outbound which would indicate that outbound end has ANI digits to be sent. This
parameter is also known as ANI category.
The range of Negative Response to Ask ANI is from 01 to 15. Default: 10.
For example, in India I-1 or I-10 is sent by the outbound end. In Kuwait, I-6 is sent. This parameter cannot
be zero because; Is ANI Available request will be made by the inbound end only if the country supports this
protocol.
• ANI End Tone Presentation Allowed: This parameter specifies the Group I tone that the inbound end
should expect from the outbound end to consider End of ANI digits with information that the Presentation
of ANI by the outbound end is allowed.

The range of ANI End Tone Presentation Allowed is 00 or from 11 to 15. Default: 15.
If no tone is sent, set this parameter to 00. For India use A-4, for China use A-1.
• ANI End Tone Presentation Restricted: This parameter specifies the group I tone that the inbound end
shall expect from the outbound end to consider End of ANI digits with an information that the Presentation
of ANI by the outbound end is Restricted.
The range of ANI End Tone Presentation Restricted is 00 or from 11 to 15. Default: 00.
If no tone is sent, set this parameter to 00. For India use A-4, for China use A-1.
• Ask Calling Party Sub Category: This parameter specifies the group 1 tone that the inbound end shall
expect from the outbound end to consider End of ANI digits with an information that the Presentation of
ANI by the outbound end is Restricted. Select this flag to enable. Default: Disabled.
Forward Group II
• Ordinary Subscriber: This parameter specifies the forward group II tone used to inform the inbound end
that the calling party is an Ordinary Subscriber. This signal is sent in response to Calling Party Category
signal Request from the inbound end.
Ordinary Subscriber is 00, 01 to 15. Default: 01. If this parameter is not applicable, assign 00.
• Priority Subscriber: This parameter specifies the forward group II tone used to inform the inbound end
that the calling party is a Priority Subscriber. This signal is sent in response to Calling Party Category
signal Request from the inbound end.
Priority Subscriber is 00, 01 to 15. Default: 02. If this parameter is not applicable, assign 00.
• Maintenance Equipment: This parameter specifies the forward group II tone used to inform the inbound
end that the calling party is Maintenance equipment.
Maintenance Equipment is 00, 01 to 15. Default: 03. If this parameter is not applicable, assign 00.
• Operator: This parameter specifies the forward group II tone used to inform the inbound end that the
calling party is Operator.
Operator is from 00, 01 to 15. Default: 05. If this parameter is not applicable, assign 00.
• Pay Phone: This parameter specifies the forward group II tone used to inform the inbound end that the
calling party is Pay Phone (Coin box).
Pay Phone is from 00, 01 to 15. Default: 00. If this parameter is not applicable, assign 00.
• Data Transmission: This parameter specifies the forward group II tone used to inform the inbound end
that the call is a Data Call.
Data Transmission is from 00, 01 to 15. Default: 06. If this parameter is not applicable, assign 00.
• Interception Operator: This parameter specifies the forward group II tone used to inform the inbound end
that the call is from Interception Operator.
Interception Operator is from 00, 01 to 15. Default: 00. If this parameter is not applicable, assign 00.

Backward Group A
• Send Next Digit (N+1) (DNIS): This parameter specifies the backward group A tone used to request next
digit. Be it ANI digit or DNIS digit.
Send next Digit range is 00, 01 to 15. Default: 01. If you do not want to use any tone, assign 00.
For India, use A-1 to signify Send DNIS Digit event.
• Send Last But One Digit (N-1) (DNIS): This parameter specifies the backward group A tone used to
request last but one digit that is, N-1 digit. Be it ANI digit or DNIS digit.
Send last but one digit range is 00, 01 to 15. Default: 02. If you do not want to use any tone, assign 00.
For India, use A-9 to signify Send last but one digit event.
• Send Last But Two Digits (N-2) (DNIS): This parameter specifies the backward group A tone used to
request last but two digits that is, N-2 digit. Be it ANI digit or DNIS digit.
Send last but two digits range is 00, 01 to 15. Default: 07. If you do not want to use any tone, assign 00.
For India, use A-7 to signify Send last but two digits event.
• Send Last But Three Digits (N-3) (DNIS): This parameter specifies the backward group A tone used to
request last but three digits that is, N-3 digit. Be it ANI digit or DNIS digit.
Send last but three digits range is 00, 01 to 15. Default: 08. If you do not want to use any tone, assign 00.
For India use A-8 to signify Send last but three digits event.
• Send Caller Party Category and ANI Digit: It is to send calling party’s category requests transmission of
a group II signal. Range is 00 to 15. By Default, it is 05.
• Address Completed, Change over of Group B: This parameter specifies the backward group A tone
used to inform the inbound end that the incoming register at the inbound end needs no additional address
digit and is about to go over to transmission of a group B signal conveying the status of equipment at the
subscriber at the inbound end.
Address-Complete, Changeover to reception of Group B signal range is 00, 01 to 15. Default: 03. If you do
not want to use any tone, assign 00.
• Send Calling Party Category and Change to Group C: This parameter specifies the backward group A
tone used by the inbound end to request Calling Party Category from the outbound end. This tone also
informs the outbound end to change to reception of Group C signal.
Send Calling Party Category and Change to Group C range is from 00, 01 to 15. Default: 00. If you do not
want to use any tone, assign 00.
• Congestion in the National Network: This parameter specifies the backward group A tone used to
inform the congestion at the inbound end.
Congestion in National Network range is 00, 01 to 15. Default: 04. If you do not want to use any tone,
assign 00.

• Send Calling Party Category: This parameter specifies the backward group A tone used to request
calling party category.
Send calling party's category range is 00, 01 to 15. Default: 05. If you do not want to use any tone, assign
00.
For India, use A-7 to signify 'Send calling party's category' event.
• Address Completed, Charge, Set Speech Condition: This parameter specifies the backward group A
tone used to inform the inbound end that the incoming register at the inbound end needs no additional
address digit, but will not send Group B signals. Also charge the call on answer.
The range of Address-Complete, Charge, Set-up Speech conditions is 00, 01 to 15. Default: 06. If you do
not want to use any tone, assign 00.
• Repeat DNIS Digits from Beginning: This parameter specifies the backward group A tone used to inform
the outbound end to send all the DNIS digits from the beginning.
The range of Repeat DNIS digits from beginning is 00, 01 to 15. Default: 00. If you do not want to use any
tone, assign 00.
• Send Next ANI Digits: This parameter specifies the backward group A tone used to request next (first)
ANI digit.
The range of Send Next ANI Digit is 00 or 01 to 15.Default: 00. If no such tone is sent, set this parameter to
00.
A few countries use different tone to request next ANI digit and next DNIS digits. For example, India uses
A-4, China uses A-1.
Backward Group B
• Send Special Information Tone: This parameter specifies the backward group B tone used to inform the
outbound end that the call cannot be made through because of reasons beyond those which are
considered by the Protocol. Hence Special Information tone will be sent to the calling party. SARVAM UCS
will send only the Group B signal and then disconnect the call.
The range of Send Special Information Tone is from 00, 01 to 15. Default: 02. If you do not want to use any
tone, assign 00.
• Send Special Information Tone and Setup Speech Conditions: This parameter specifies the backward
group B tone used to inform the outbound end that the call cannot be made through because of reasons
beyond those which are considered by the Protocol. Hence, Special information tone will be sent to the
calling party and request the outbound end to setup speech conditions.
In this case, SARVAM UCS shall connect the calling party to the voice message of the system informing
the caller that the call cannot be connected.
The range of Send Special Information Tone and setup speech conditions is from 00, 01 to 15. Default: 02.
If you do not want to use any tone, assign 00.
• Subscriber Line Busy: This parameter specifies the backward group B tone used to inform the outbound
end that the called subscriber is busy.

Subscriber Line busy range is from 00, 01 to 15. Default: 03. If you do not want to use any tone, assign 00.
• Subscriber Line Free, Charge: This parameter specifies the backward group B tone used to inform the
outbound end that the called subscriber is free and the call is to be charged on answer.
The range of Subscriber Line free, Charge is from 00, 01 to 15. Default: 06. If you do not want to use any
tone, assign 00.
• Subscriber Line Free, No charge: This parameter specifies the backward group B tone used to inform
the outbound end that the called subscriber is free, but the call is not to be charged on answer. This signal
permits non-chargeable calls without the need for transferring no charge information by line signals.
The range of Subscriber Line free, NO Charge is 00, 01 to 15. Default: 07. If you do not want to use any
tone, assign 00.
• Congestion: This parameter specifies the backward group A tone used to inform that congestion is
encountered after changeover from Group-A to Group-B signals.
Congestion range is from 00, 01 to 15. Default: 04. If you do not want to use any tone, assign 00.
• Unallocated Number: This parameter specifies the backward group B tone used to inform the outbound
end that the number received is not in use.
The range of Unallocated Number is from 00, 01 to 15. Default: 05. If you do not want to use any tone,
assign 00.
• Subscriber Line Out of Order: This parameter specifies the backward group B tone used to inform the
outbound end that the called subscriber's line is out of order.
Range of Subscriber's Line out of order is from 00, 01 to 15. Default: 08. If you do not want to use any
tone, assign 00.
• Call Rejected, No Indication: This parameter specifies the Group B backward tone used to inform the
outbound end that the call is rejected but there is no indication of cause.
Call rejected, No indication range is from 00, 01 to 15. Default: 00. If this parameter is not applicable,
assign 00.
• Alternative Answer Tone: This parameter specifies the Group B backward tone used to inform the
outbound end that the call is accepted and the speech path is made through.
Alternative Answer Tone range is from 00, 01 to 15. Default: 00. If this parameter is not applicable, assign
00.
• Changed Number: This parameter specifies the Group B backward tone used to inform the outbound end
that the number dialed by the calling party is changed. However, this parameter is rarely used.
The range of Changed Number (announcement on line) is from 00, 01 to 15. Default: 00. If this parameter
is not applicable, assign 00.

Backward Group C
• Send Next ANI Digit: This parameter specifies the backward group C tone to request next (even first) ANI
digit from the outbound end.
The range of Send next ANI digit (Group C) is from 00, 01 to 15. Default: 00. If this parameter is not
applicable, assign 00.
• Request Transition Back to Group A and Restart from First DNIS (Group C): This parameter specifies
the backward group C tone to restart from the first DNIS and request transition to Group A.
The range of Request transition to Group A and restart from first DNIS is from 00, 01 to 15. Default: 00. If
this parameter is not applicable, assign 00.
• Address Completed, Change to Reception of Group B: This parameter specifies the backward group C
tone used to signify Address completed, change to reception of Group B signal.
The range of Address completed, change to reception of Group B signal is from 00, 01 to 15. Default: 00. If
• Congestion: This parameter specifies the backward group C tone used to signify Congestion.
The range of Congestion is from 00, 01 to 15. Default: 00. If this parameter is not applicable, assign 00.
• Request Transition Back to Group A and Sent Next DNIS: This parameter specifies the backward
group C tone used to signify request transition back to group A, and send next DNIS.
The range of Request transition back to group A, and send next DNIS is from 00, 01 to 15. Default: 00. If
• Request Transition Back to Group A and Repeat the Last DNIS: This parameter specifies the
backward group C tone used to signify request transition back to group A, and repeat the last DNIS.
The range of Request transition back to group A, and repeat the last DNIS is from 00, 01 to 15. Default: 00.
If this parameter is not applicable, assign 00.
Line Signal Parameters
• Line Signaling: By default this flag is enabled. If you are unable to make outgoing calls, check with your
Service Provider and disable this option. This option will be relevant only when you Tie-up SARVAM UCS
with another PBX.
• C & D Bits: This parameter indicates the default values of C and D bits when the T1/E1 Port transmits line
signals.
CD Bits Meaning (Binary Value)
0 00 (C=0, D=0)
1 01
2 10
3 11
Default: 01 that is, C and D Bit is 1

The C and D bits received during an IC call should be ignored by the system.
• Invert Bit A: This parameter signifies whether A-bit is to be inverted before transmitting and on receiving.
• E1 Metering Bit: This parameter signifies the bit used by the network to signal metering pulses.
You can select from the following options:
• None
• Bit-A
• Bit-B
• Bit-C
• Bit-D
Default: Bit-A.
• E1 Metering Pulse Minimum Timer (Milliseconds): This timer signifies the minimum time for which the
metering bit is changed, to be recognized as a genuine metering pulse subject to E1 Metering Pulse
Minimum timer.
All Changes occurred for time less than this timer is ignored. The range of E1 Metering Pulse Minimum
timer is from 20ms to 1000ms. Default: 150ms.
• Clear Back Signal: This parameter signifies the signal used to signify that the called party has
disconnected the line first. This is indicated in two ways: Release Guard (Ab =1) or Forced Release (Bb =
0). This parameter is country specific.
Default: Release Guard.
• Release Timer (Milliseconds): This timer signifies the time for which the clear back signal should persist
on the line to be recognized as a genuine clear back signal. This is also known as Clear Back timer.
The range of Release Timer is from 20ms to 1000ms. Default: 400 ms.
• Line Seizure Acknowledge Wait Timer (Milliseconds): This timer signifies the time for which the
outbound end waits for seizure acknowledgement from the inbound end after sending the line seizure

signal. On expiry of this timer, clear forward signal is sent by the outbound end. Alarm is to be generated.
This timer is applicable only when acting as outbound end.
The range of Line Seizure acknowledge Wait Timer is from 0001ms to 9999 ms. Default: 200ms.
• Release Guard Timer (Milliseconds): This timer signifies the time for which inbound register waits before
declaring the channel idle (sending idle signal) when clear forward line signal is received from the
outbound end. This timer is applicable for Forced Release signal. This timer is applicable only when acting
as inbound end. This timer depends on the speed of switching and processing.
The range of Release Guard Timer is from 0000 ms to 9999 ms. Default: 200ms.
E&M Signaling
Under T1E1 Configuration, click E&M Signaling and configure the E&M parameters.
• E&M Feature Template: Assign an E&M Feature Template to the T1E1 Port. The E&M Feature Template
is a set of features specific to E&M signaling, which define the behavior of the E&M ports, according to
their 'Orientation Type', whether they are functioning as Stations, Trunks or Tie-Lines. By default, Template
01 is assigned to all T1E1 Ports.
For more details, see “E&M Feature Template”.
• B Bit Pattern: Select the Bit Pattern from Same as Bit A or Fixed Value. By default, the Code is 1 (Same
as A bit).
• B Bit Value: Configure the B bit value, the value can be 0 or 1. By default, B bit value is 0.
• CD Bit Value: Configure the CD bit value, the valid range of the value is 1 to 3. By default, B bit value is 0.
• Invert Bit: This parameter signifies whether A-bit is to be inverted before transmitting and on receiving.

Port Parameters
T1E1-1
Hardware Slot-Port
Use following command to assign hardware ID to a T1E1 software port.

1107-T1E1-Slot-Port offset
Where,
Slot is the number of the universal slot, where the T1E1 Card is installed, from 01 to 16.
Port is the number of theT1E1 port on the card, from 01 to 32.
Use following command to de-assign the hardware slot and the hardware port assigned to the T1E1 software port.
1106-T1E1-00-00
Port Status
Used to enable/disable the port. When the Port is disabled, it will not be allotted to the user on grabbing the port.
Instead the user will get error tone.
Use the following command to enable/disable the port:

6101-1-T1E1-Port Status
6101-2-T1E1-T1E1-Port Status
6101-*-Port Status
Where,

Port Status Meaning
0 Disable
1 Enable
By default, the T1E1 Port is enabled.
Name
Use the following command to assign a name to the port:

5407-1-T1E1-Name
5407-2-T1E1-T1E1-Name
5407-*-Name
Where,
Name can be of upto 18 characters.
Carrier
Use the following command to select the carrier:

6108-1-T1E1-Carrier Type
6108-2-T1E1-T1E1-Carrier Type
6108-*-Carrier Type
Where,
Carrier
Meaning
Type
1 E1
2 T1
Line type
Use following command to program signaling type/ Line type of a T1E1:

6105-1-T1E1-Line Type
6105-2-T1E1-T1E1-Line Type
6105-*-Line Type
Where,
Line Type Meaning
1 PRI
2 QSIG
3 CAS
4 E&M
By default, signaling type of a T1E1 is 1.

Orientation Type
Use following command to program 'Orientation Type' for the T1E1 port:
6106-1-T1E1-Orientation Type
6106-2-T1E1-T1E1-Orientation Type
Where,
T1E1=01 to 08.
Orientation Meaning
1 Terminal
2 Network
3 Tie Line
When Orientation = Terminal, the port will be regarded as trunk. All the trunk related parameters will be applicable.
When Orientation = Network, the port will be regarded as station. All the station related parameters will be
applicable.
When Orientation = Tie-line, the port will be regarded as station for all IC calls to it and as trunk for all OG calls to be
made through it.
Line Coding Mechanism
Use following command to program the Line Coding Mechanism for the T1E1 port:
6103-1-T1E1-Line Coding
6103-2-T1E1-T1E1-Line Coding
6103-*-Line Coding
Where,
Line Coding Meaning
1 AMI-Basic
2 HDB3
3 NZR (Fiber Optic)
By default, Line Coding is HDB3.
Framing Mode
Use following command to program the Framing Mode for the T1E1 port:
6104-1-T1E1-Framing
6104-2-T1E1-T1E1-Framing
6104-*-Framing

Where,
Framing Meaning
1 CEPT1 MF (No CRC)
2 CEPT1 MF (Forced CRC)
3 CEPT1 MF (Auto CRC)
By default, Framing is CEPT1 MF (Auto CRC).
DDI Routing
OG Reference ID
Use the following command to assign OG Reference ID to T1E1 port:

6131-1-T1E1-OG Reference ID
6131-2-T1E1-T1E1-OG Reference ID
Where,
OG Reference ID is from 00 to 99.
By default, OG Reference ID is 00.
IC Reference ID for Working Hour
Use the following command to assign IC Reference ID for Working Hour:

Where,
By default, IC Reference ID is 00.
IC Reference ID for Break Hour
Use the following command to assign IC Reference ID for Break Hour:

Where,
By default, IC Route Reference ID is 00.
IC Route Reference ID for Non-working Hour
Use the following command to assign IC Route Reference ID for Non-working Hour:
Where,

By default, IC Route ID is 00.
Templates
Use the following command to assign a Trunk Feature Template to the T1E1 Trunks, dial:
5806-1-T1E1- Trunk Feature Template Number to assign a template to a single T1E1 port.
5806-2-T1E1- Trunk Feature Template Number to assign the same template to a range of T1E1 ports.
5806-*- Trunk Feature Template Number to assign the same template to all T1E1 ports.
Where,
Template Number is the number of the customized Trunk Feature Template, from 01 to 50. Default: Trunk Feature
Template 01.

Where,
Station Advance Features Template
To assign a Station Advanced Feature Template to a T1E1PRI port

Where,
Others
Priority
To assign priority to T1E1

3914-1-T1E1-Priority to assign a template to a single T1E1 port.
3914-2-T1E1PRI-T1E1PRI-Priority to assign the same priority to a range of T1E1 ports.
3914-*-Template Number to assign the same Priority to all T1E1 ports.
Where,
Priority is from 1 to 9. Default: 5-Normal.
Cost Factor
Assign a Cost Factor to the T1E1 port. This is useful in LCR.
Use following command to assign a name to the T1E1 port:

6102-1-T1E1-SP

6102-2-T1E1-T1E1-SP
6102-*-SP
Where,
SP is from 01 to 99.
By default, Service Provider is 01.
Glare Option
Use following command to program Glare Option for the T1E1 port:
6112-1-T1E1-Glare Option
6112-2-T1E1-T1E1-Glare Option
6112-*-Glare Option
Where,
Glare Option Meaning
1 Proceed
2 Held Back
By default, Glare Option is 2.
Set Glare=Proceed, if SARVAM UCS is to be given priority in event of Glare.

Set Glare=Held Back, if the other end of the link is to be given priority in event of Glare.
The Glare settings should be complimentary on either side of the link.
Category (Logical Partitioning)
Use following command to set the Category (Logical Partitioning) for the T1E1 port:
6121-1-T1E1-Category
6121-2-T1E1-T1E1-Category
6121-*-Category
Where,
By default, Category is set as 1.
Idle Code
Use the following command to program the Idle Code of a T1E1:

6113-1-T1E1-Idle Code
6113-2-T1E1-T1E1-Idle Code
6113-*-Idle Code
Where,
Idle Code is from 000 to 255 (corresponding to 8 bits).
By default, the idle code is 127 (7F).
Use message mode of the Digital Switch IC to send the idle channel code.

RCOC
To enable RCOC on T1E1 Trunk, dial:

Where,
Code is
0 for Disable
1 for Enable
Default: Disable
Channels
Use the following command to enable RCOC on T1E1 Trunk:

Where,
Code is
0 for Disable
1 for Enable
Default: Disable
Channel Reserved for Data Call
Use the following command to reserve channels for data transmission on T1E1:
6135-1-T1E1-Channel Count (Data)
6135-2-T1E1-T1E1-Channel Count (Data)
6135-*-Channel Count (Data)
Where,
Channel Count (Data) is from 00 to 30.
By default, Channel Count for data transmission is 00.
Channel Reserved for Outgoing Call
Use the following command to program number of channels reserved for OG channel count:
6136-1-T1E1-Channel Count (OG)
6136-2-T1E1-T1E1-Channel Count (OG)
6136-*-Channel Count (OG)
Where,
Channel Count (OG) is from 00 to 30. "It specifies the number of channels to be reserved for making an OG calls.
For example, If OG channel count is programmed as 15, simultaneous 15 (maximum) OG calls can be made from
the T1E1 port".
By default, OG Channel Count is 30.
Channel Reserved for Incoming Call
Use the following command to program number of channels reserved for IC channel count:
6137-1-T1E1-Channel Count (IC)

6137-2-T1E1-T1E1-Channel Count (IC)
6137-*-Channel Count (IC)
Where,
Channel Count (IC) is from 00 to 30. "It specifies the number of channels to be reserved for making an IC calls. For
example, If IC channel count is programmed as 10, simultaneous 10 (max.) IC calls can be received on the T1E1
port".
By default, Channel Count (IC) is 30.
Tone
Feed Dial Tone
Use following command to program the dial tone flag for T1E1 port:
6115-1-T1E1-Flag
6115-2-T1E1-T1E1-Flag
6115-*-Flag
Where,
Flag Meaning
0 Disable
1 Enable
By default, Dial Tone Flag is '0' for all the T1E1 ports.
Feed Routing Tone
Use following command to program the routing tone flag for T1E1 port:
6116-1-T1E1-Flag
6116-2-T1E1-T1E1-Flag
6116-*-Flag
Where,
Flag Meaning
0 Disable
1 Enable
By default, Routing Tone Flag is '0' for all the T1E1 ports.
Custom Pulse
T1/E1 Custom Pulse Width (CPW)
Use following command to enable/disable Custom Pulse Width (CPW) Flag for the T1E1 port for T1 signaling:
6171-1-T1E1-Flag
6171-2-T1E1-T1E1-Flag
6171-*-Flag
Where,

Flag Meaning
0 Disable
1 Enable
By default, Custom Pulse Width Flag is '0'.
This command is used for Pulse shaping in first phase.
Where,
This command is used for Pulse shaping in second phase.
Where,
This command is used for Pulse shaping in third phase.
Where,
This command is used for Pulse shaping in fourth phase.

Where,
Timer
Pause Timer

6109-1-T1E1- Pause Timer
6109-2-T1E1-T1E1-Pause Timer
6109-*-Pause Timer
Where,
DTMF ON Time

6117-1-T1E1-DTMF ON Time
6117-2-T1E1-T1E1-DTMF ON Time
6117-*-DTMF ON Time
Where,
DTMF Inter Digit Pause Timer

6118-1-T1E1-DTMF Inter digit Pause Time
6118-2-T1E1-T1E1-DTMF Inter digit Pause Time
Where,
Gateway
Use Gateway Application - Answer Signaling?
Use following command to set flag for 'Gateway Application-Answer Signaling' on T1E1 trunk:
6119-1-T1E1-Gateway Application-Answer Signaling flag
6119-2-T1E1-T1E1-Gateway Application-Answer Signaling flag
6119-*- Gateway Application-Answer Signaling flag
Where,
Flag Meaning
0 Disable

Flag Meaning
1 Enable
By default, Gateway Application-Answer Signaling flag is 'disable'.
Gateway Application - Answer Signaling DTMF Digits

6120-2-T1E1-T1E1-Gateway Application-Answer Signaling DTMF String
Where,
command using #*.
A #4
B #5
C #6
D #7
* **
# ##
By default Gateway Application-Answer Signaling DTMF String is 'CCC'.
Call Budget
Call Budget Type

6122-1-T1E1-Budget Type to program call budget type for a single trunk port.
6122-2-T1E1-T1E1-Budget Type to program the same call budget type for a range of trunk ports.
6122-*-Budget Type to program the same call budget type for all trunk ports.
Where,
Budget Type is
0 for None
1 for Amount
2 for Minutes
Call Budget Amount

6123-1-T1E1-Budget Amount to program call budget amount for a single trunk port.

6123-2-T1E1-T1E1-Budget Amount to program the same amount for a range of trunk ports.
6123-*-Budget Amount to program the same amount for all trunk ports.
Where,
Budget Amount is of 6 digits max. Use leading zeros if amount to be programmed has fewer than 6 digits.
Call Budget Minutes

6124-1-T1E1-Minutes to program minutes for a single trunk port.
6124-2-T1E1-T1E1-Minutes to program the same minutes for a range of trunk ports.
6124-*-Minutes to program the same minutes for all trunk ports.
Where,
Call Budget Number of Calls

6125-1-T1E1-Number of calls to program number of calls for a single trunk port.
6125-2-T1E1-T1E1-Number of calls to program the same number of calls for a range of trunk ports.
6125-*-Number of calls to program the same number of calls for all trunk ports.
Where,
T1E1 is the number of the T1E1 software port from 01 to 08.
Number of Calls is of 4 digits from 0001 to 9999. Use leading zeros if number of calls to be programmed has fewer
than 4 digits.
Call Budget Reset Mode

6138-1-T1E1-Call Budget Reset Mode to program reset mode for a single trunk port.
6138-2-T1E1-T1E1-Call Budget Reset Mode to program the same reset mode for a range of trunk ports.
6138-*-Call Budget Reset Mode to program the same reset mode for all trunk ports.
Where,
2 for Manual reset
Call Budget Date for Scheduled Reset mode
6139-1-T1E1-Date to program date for a single trunk port.
6139-2-T1E1-T1E1-Date to program the date for a range of trunk ports.
6139-*-Date to program the date for all trunk ports.
Where,
Date is

Call Back
Use the following commands to program Call Back on T1E1 Trunk ports. To know more about this feature, refer the
To enable/disable Call Back on T1E1 port:

6176-1-T1E1- Call Back Flag
6176-2-T1E1-T1E1-Call Back Flag
6176- *-Call Back Flag
Where,
0 Disable
1 Enable
Call Back Timer
To program Call Back Timer for T1E1 port:

6177-1-T1E1-Call Back Timer
6177-2-T1E1-T1E1-Call Back Timer
Where,
Call Back Mode
To program Call Back Mode on T1E1 port:

6178-1-T1E1-Call Back Mode
6178-2-T1E1-T1E1-Call Back Mode
Where,
Call Back Mode is 1 to 5.
1 Built-in Auto Attendant
5 Operator
By default, Call Back mode is Operator
Call Back On
To program Call Back On method for T1E1 port:

6179-1-T1E1-Call Back on selection
6179-2-T1E1-T1E1-Call Back on selection
Where,
1 CLI Number
2 Alternate Number
By default, Call Back on selection is CLI Number.
Incoming Number List
To assign Call Back - Incoming Number List to a T1E1 port:

6180-1-T1E1-Incoming Number List
6180-2-T1E1-T1E1-Incoming Number List
Where,
By default, Incoming Number List is 15
Outgoing Number List
To assign a Call Back - Outgoing Number List to a T1E1 port:

6147-1-T1E1-Outgoing Number List
6147-2-T1E1-T1E1-Outgoing Number List
Where,
Outgoing Number List is from 01 to 16.
Call Back From
To define Call Back From for a T1E1 port:

6148-1-T1E1-Call Back From
6148-2-T1E1-T1E1-Call Back From
Where,
Call Back From is
1 for Same Port
2 for OGTB Group
OGTB Group
To assign a Call Back - OGTB Group for a T1E1 port:

6149-1-T1E1-OGTB Group
6149-2-T1E1-T1E1-OGTB Group

6149-*-OGTB Group
Where,
FDL
FDL Flag - copied from T1 Maintenance
T1 FDL can be enabled/disabled. This parameter is applicable only if Framing = ESF. If the Network (Public or
Private) to which the SARVAM UCS is connected does not support FDL then T1 FDL will be disabled.

6164-*-T1 FDL
Where,
T1 FDL Meaning
0 Disable
1 Enable
FDL Protocol
SARVAM UCS will support both the protocols of reporting the performance monitoring. This parameter is applicable
only if T1 FDL is enabled and Framing = ESF. This parameter will match the protocol expected by the other end of
the link.
Where,
0 Disable
1 AT&T 54016
2 ANSI T1.403
Refer “E1/T1 Maintenance”

Debug
SARVAM UCS supports debug of parameters (debug codes) depending on the Level of debug. On issuing this
command the T1E1 Card will send the debug details to the COM port of the T1E1 port.
Debug Level Command
Use following command to start/stop debug the parameters for the T1E1 port:
6191-1-T1E1-Level-Debug Code
6191-2-T1E1-T1E1-Level-Debug Code
6191-*-Level-Debug Code
Where,
T1E1 Port is from 01 to 08.
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from 000 to 255.
Code value ‘000’ for each level will turn off that level’s debug.
Level 1:
001 CAS
002 MFC R2
004 CAS DSP
008 Layer 4
000 Debug Off
Level 2:
001 Alarms
002 Counters
004 ABCD Bits
008 FDL
HDLC (D
016
Channel)
000 Debug Off

Level 3:
001 Primitives
002 State
004 Variables
008 SVC Primitives
016 LAP Debug
032 NLS Debug
064 Flow Debug
000 Debug Off
Level 4:
001 OS Task
002 NI Debug
000 Debug Off
Default: Debug Code = ‘Debug OFF’ for all T1E1 ports for all levels.
Refer “T1 RBS Parameters”
T1E1:1 - PRI/QSIG Signaling

ISDN Switch Variant
This field selects the variant for ISDN PRI when ISDN PRI is selected as signaling mode. This must match with the
other end of the link.
Use following command to program the ISDN PRI Switch Variant:

6107-1-T1E1-ISDN Switch Variant
6107-2-T1E1-T1E1-ISDN Switch Variant
6107-*-ISDN Switch Variant
Where,
1 ATT 4ESS
2 ATT 5ESS
3 Australia
4 DMS

5 ETSI NET5
6 NTT INS64
7 SWV Hongkong
8 US NI12
9 QSIG E1
10 QSIG T1
By default, ISDN PRI Switch Variant of a T1E1 is ETSI NET5.
Send Called Party Number Using

6146-1-T1E1-Type
6146-2-T1E1-T1E1-Type
6146-*-Type
Where,
Type is
1 for Keypad Facility IE
0 for Called Party IE
Default: Called Party Number IE

6114-1-T1E1-Timer
6114-*-Timer
Where,
By default, Overlap Receiving Timer is 15 seconds.
Caller-Type of Number (TON)
Use following command to program the Caller-Type of Number (TON) for a T1E1:
6126-1-T1E1-Source TON
6126-2-T1E1-T1E1-Source TON
6126-*-Source TON
Where,
Source TON Meaning
Unknown: This is used when the user or network has no a prior information about
1 the numbering plan. In this case, the Address Value field is organized according to

Source TON Meaning
Network Specific Number: This is used to indicate administration/Service number

4
Subscriber Number: This is used when a specific short number representation is

5
7 Reserved Number.
By default, Caller-Type of Number (TON) of T1E1 is 1.
Caller-Numbering Plan Identification (NPI)
Use following command to program Caller-Numbering Plan Identification (NPI) for T1E1:
6127-1-T1E1-Source NPI
6127-2-T1E1-T1E1-Source NPI
6127-*-Source NPI
Where,
Source NPI Meaning
1 Unknown
4 Telex Numbering
By default, Caller-Numbering Plan Identification (NPI) for T1E1 is 2.
Called-Type of Number (TON)
Use following command to program Called-Type of Number (TON) for a T1E1:

6128-1-T1E1-Destination TON
6128-2-T1E1-T1E1-Destination TON
6128-*-Destination TON
Where,
1 Unknown: This is used when the user or network has no a prior information about
the numbering plan. In this case, the Address Value field is organized according to

4 Network Specific Number: This is used to indicate administration/Service number

5 Subscriber Number: This is used when a specific short number representation is

7 Reserved Number.
By default, Called-Type of Number (TON) of T1E1 is 1.
Called-Numbering Plan Identification (NPI)
Use following command to program Called-Numbering Plan Identification (NPI) for T1E1:
6129-1-T1E1-Destination NPI
6129-2-T1E1-T1E1-Destination NPI
6129-*-Destination NPI
Where,
Destination NPI Meaning
1 Unknown
4 Telex Numbering
By default, Called-Numbering Plan Identification (NPI) for T1E1 is 2.
Receive Equalization Mode
Use the following command to program auto receive equalization mode:

6110-1-T1E1-Mode
6110-2-T1E1-T1E1-Mode
6110-*-Mode
Where,
Mode Meaning
0 Manual
1 Auto
By default, Receive Equalization Mode is 1.

Receive Equalization Parameters
This field increases the strength of incoming signals by a fixed amount to compensate for line losses.
Use the following command to program the receive equalization parameters of a T1E1:
6111-1-T1E1-Receive Equalization Parameters
6111-2-T1E1-T1E1-Receive Equalization Parameters
6111-*-Receive Equalization Parameters
Where,
Receive Equalization Parameters Meaning
1 None
2 8 dB
3 16 dB
4 24 dB
5 32 dB
6 40 dB
7 48 dB
By default, the receive equalization parameters of T1E1 is 1.
Feed Inband Tones on T1E1-NT, before sending DISCONNECT
Use following command to program to select whether the inband tones should be feed on T1E1-NT before sending
DISCONNECT message?
6130-1-T1E1-Flag
6130-2-T1E1-T1E1-Flag
6130-*-Flag
Where,
Flag Meaning
0 No
1 Yes
Default = NO.
E1 CAS Signaling
T1E1-1
E1 Line Signaling Variant
The following command is applicable when the Bit Rate=E1.
Use following command to program E1 Line Signaling Variant for the T1E1 port:
6152-1-T1E1-E1 Line Signaling Variant
6152-2-T1E1-T1E1-E1 Line Signaling Variant
6152-*-E1 Line Signaling Variant

Where,
E1 Line Signaling Variant Meaning
01 ITU T Q.400-Q.490
By default, E1 Line Signaling Variant is 01.
E1 Register Signaling Variant
Use following command to program E1 Register Signaling Variant for the T1E1 port:
6153-1-T1E1-E1 Register Signaling Variant
6153-2-T1E1-T1E1-E1 Register Signaling Variant
6153-*-E1 Register Signaling Variant
Where,
E1 Register Signaling Variant Meaning
1 DTMF
2 MFC R2
By default, E1 Register Signaling Variants is 2.
• DTMF: DNIS/ANI is transmitted in the corresponding speech channel using the DTMF signals as per ITU-
T Q.23.
• MFC R2: DNIS/ANI is transmitted in the corresponding speech channel using the MFC R2 signals as per
ITU-T Q.400-Q490.
Register Signaling Parameters

Forward tone Maximum ON timer (T1)
This timer signifies the maximum time for which the forward signal can be ON, from the outbound end.
Use the following command to program Forward Tone Maximum ON Timer:
7101-1-T1E1-Forward Tone Maximum ON Timer
7101-2-T1E1-T1E1-Forward Tone Maximum ON Timer
7101-*-Forward Tone Maximum ON Timer
Where,
Forward Tone Maximum ON Timer is from 01 to 99 seconds.
By default, Forward Tone Maximum ON Timer is 15 secs.
Forward Tone Maximum OFF Timer (T2)
Use the following command to program Forward Tone Maximum Off Timer:
7102-1-T1E1-Forward Tone Maximum OFF Timer
7102-2-T1E1-T1E1-Forward Tone Maximum OFF Timer
7102-*-Forward Tone Maximum OFF Timer
Where,
Forward Tone Maximum OFF Timer is from 01 to 99 seconds.

By default, Forward Tone Maximum Off Timer is 24 seconds. This timer will be less than the Incoming R2 register
timeout timer programmed by the inbound end.
Maximum Compelled Cycle Time (T3)
Use the following command to program the Maximum compelled cycle time:
7103-1-T1E1-Maximum Compelled Cycle Time
7103-2-T1E1-T1E1-Maximum Compelled Cycle Time
7103-*-Maximum Compelled Cycle Time
Where,
Total Call Set Up Timer is from 01 to 99 seconds.
By default, Maximum Compelled Cycle Time is 15 secs.
Pulse Duration for Pulse Signal
Use the following command to program Pulse duration for pulsed signals:
7104-1-T1E1-Pulse Duration for Pulse Signal
7104-2-T1E1-T1E1-Pulse Duration for Pulse Signal
7104-*-Pulse Duration for Pulse Signal
Where,
Tolerance is fixed at +/-25ms.
By default, the Pulse Duration for Pulsed Signals is 150ms.
Pulse Signal Maximum Wait Timer
Use the following command to program the Pulsed Signal Maximum Wait Timer:
7105-1-T1E1-Pulse Signal Maximum Wait Timer
7105-2-T1E1-T1E1-Pulse Signal Maximum Wait Timer
7105-*-Pulse Signal Maximum Wait Timer
Where,
Pulsed Signal Maximum Wait Timer is from 01 to 99 seconds.
By default, Pulsed Signal Maximum Wait Timer is 15 secs.
First Forward Tone Wait Timer
Use the following command to program first forward tone wait timer:
7106-1-T1E1-First Forward Tone Wait Timer
7106-2-T1E1-T1E1-First Forward Tone Wait Timer
7106-*-First Forward Tone Wait Timer
Where,
First forward tone wait timer is from 08 to 24 seconds.
By default, the First Forward Tone Wait Timer is 15 seconds.
Minimum MF Signal Persist Timer
Use following command to program Minimum MF Signal Persist timer:

7107-1-T1E1-Minimum MF Signal Persist Timer
7107-2-T1E1-T1E1-Minimum MF Signal Persist Timer
7107-*-Minimum MF Signal Persist Timer
Where,
Minimum MF Signal Persist timer is from 001 to 255 ms.

By default, Minimum MF Signal Persist timer is 20 ms.
Outbound Parameters
Dialed Number Identification Signal (DNIS) END Type
Use following command to program to set DNIS END Type (outbound) for T1E1 port:
7108-1-T1E1-End of DNIS
7108-2-T1E1-T1E1-End of DNIS
7108-*-End of DNIS
Where,
End of DNIS is from 00, 11 to 15.
00 indicates End of DNIS as time out.
01 to 15 indicates group 1 tone to declare End of DNIS.
By default, DNIS End Type (Outbound) is 15.
Address Number Information (ANI) Send Position
Use following command to program the ANI Send Position:

7110-1-T1E1-ANI Send Position
7110-2-T1E1-T1E1-ANI Send Position
7110-*-ANI Send Position
Where,
ANI Send Position is from 00 to 99.
By default, ANI Send Position is 00.
Is Address Number Information (ANI) Available
Use following command to program Is ANI Available (Outbound):

7111-1-T1E1-Is ANI Available (Outbound)
7111-2-T1E1-T1E1-Is ANI Available (Outbound)
7111-*-Is ANI Available (Outbound)
Where,
Is ANI Available (Outbound) is Group A tone from 00 to 15. If no question by the inbound end, set this parameter to
00.
By default, Is ANI Available is '05'.
Positive Response to Is ANI Available
Use the following command to program the Positive Response to Is ANI Available (Outbound):
7112-1-T1E1-Positive Response to Is ANI Available
7112-2-T1E1-T1E1-Positive Response to Is ANI Available
7112-*-Positive Response to Is ANI Available
Where,
Positive Response to Is ANI Available is a Group 1 tone from 01 to 15.
By default, Positive Response to Is ANI Available is 01.
Negative Response to Is ANI Available
Use following command to program the Negative Response to Is ANI Available (Outbound):
7113-1-T1E1-Negative Response to Is ANI Available
7113-2-T1E1-T1E1-Negative Response to Is ANI Available

7113-*-Negative Response to Is ANI Available
Where,
Negative Response to Is ANI Available is a Group 1 tone from 01 to 15.
By default, Negative Response to Is ANI Available is 10.
ANI End Tone Presentation Allowed
Use the following command to program the End of ANI with Presentation Allowed (Outbound):
7114-1-T1E1-ANI End Tone with Presentation Allowed (Outbound)
7114-2-T1E1-T1E1-ANI End Tone with Presentation Allowed (Outbound)
7114-*-ANI End Tone with Presentation Allowed (Outbound)
Where,
ANI End Tone with Presentation Allowed (Outbound) is a Group 1 tone from 00, 11 to 15. If no such tone is sent, set
this parameter to 00.
By default, ANI End Tone with Presentation Allowed (Outbound) is 15.
ANI End Tone Presentation Restrict
Use the following command to program the End of ANI with Presentation Restricted (Outbound):
7115-1-T1E1-ANI End Tone with Presentation Restrict (Outbound)
7115-2-T1E1-T1E1-ANI End Tone with Presentation Restrict (Outbound)
7115-*-ANI End Tone with Presentation Restrict (Outbound)
Where,
ANI End Tone with Presentation Restrict (Outbound) is a Group 1 tone from 00, 11 to 15. If no such tone is sent, set
this parameter to 00.
By default, ANI End Tone with Presentation Restrict (Outbound) is 00.
Inbound Parameters
Dialed Number Identification Signal (DNIS) End Type
Use following command to program DNIS End Type:

7109-1-T1E1-DNIS End Type
7109-2-T1E1-T1E1-DNIS End Type
7109-*-DNIS End Type
Where,
DNIS End Type is from 00, 11 to 15.
00 indicates End of DNIS as time out.
01 to 15 indicates group 1 tone to declare End of DNIS.
By Default, DNIS End Type (Inbound) is 15.
Dialed Number Identification Signal (DNIS) Digit Length
Use following command to program the DNIS Digit Length:

7116-1-T1E1-DNIS Digit Length
7116-2-T1E1-T1E1-DNIS Digit Length
7116-*-DNIS Digit Length
Where,
DNIS Digit Length is from 01 to 99.
By default, DNIS Digit Length is 99.

• DNIS Digit length (01 to 98) will be expected by the inbound end. (Practical value would be 01 to 10)
• DNIS Digit length 99 indicates DNIS length is variable. Further action is taken after timeout or on receipt of
I-15. Refer parameter 'DNIS End Type (Inbound)'.
Address Number Information (ANI) Request Position
Use following command to program the number of DNIS digits after which ANI digits should be requested by the
inbound end:
7117-1-T1E1-ANI Request Position
7117-2-T1E1-T1E1-ANI Request Position
7117-*-ANI Request Position
Where,
ANI Request Position is from 00, 01 to 99.
ANI Request Position=00 indicates Never request ANI digits.
ANI Request Position=01 to 98 indicates Request ANI digits on receipt of these many DNIS digits.
ANI Request Position = 99 indicates Request after receiving all the DNIS digits (complete DNIS).
By default, ANI Request Position is 99.
Address Number Information (ANI) Length
Use following command to program ANI Length (Inbound):

7118-1-T1E1-ANI Length
7118-2-T1E1-T1E1-ANI Length
7118-*-ANI Length
Where,
ANI Length = 00 to 99.
By default, ANI Length is 99.
• ANI Length=00 indicates ANI is not sent by the Outbound end.

• ANI Length = 99 indicates ANI Length is variable. If ANI length is variable, the logic waits for End of ANI
from the outbound side. The inbound end will sense for I-12 and I-15. I-12 is used to signify that no ANI
digits are available whereas I-15 is used to signify end of ANI digits. Some countries like China use I-15 to
signify both the events viz. End of ANI and no ANI digits available.
Ask Address Number Information (ANI)
Use following command to program the ASK ANI available:

7119-1-T1E1-ASK ANI
7119-2-T1E1-T1E1-ASK ANI
7119-*-ASK ANI
Where,
ASK ANI Available is from 00 or 01 to 15.
If no such tone is sent by the inbound end, set this parameter to 00. For India, this should be set to 04
By default, Ask ANI (Inbound) is '05'.
Positive Response to Ask ANI
Use following command to program Positive Response to Ask ANI:

7120-1-T1E1-Positive Response to Ask ANI
7120-2-T1E1-T1E1-Positive Response to Ask ANI
7120-*-Positive Response to Ask ANI
Where,

Positive Response to Ask ANI is from 01 to 15.
By default, Positive Response to Ask ANI is 01.
• This cannot be zero. This is because; Is ANI Available request would be made by the inbound end only if
the country supports this protocol. In such event, Is ANI Available request will be responded to.
For example, In India I-1 or I-10 is sent by the Outbound end. In Kuwait, I-6 is sent.
Negative Response to Ask ANI
Use following command to program Negative Response to Ask ANI:

7121-1-T1E1-Negative Response to Ask ANI
7121-2-T1E1-T1E1-Negative Response to Ask ANI
7121-*-Negative Response to Ask ANI
Where,
Negative Response to Ask ANI is from 01 to 15.
By default, Negative Response to Ask ANI is 10.
• This cannot be zero. This is because; Is ANI Available request would be made by the inbound end only if
the country supports this protocol. In such event, Is ANI Available request will be responded to.
For example, In India I-1 or I-10 is sent by the Outbound end. In Kuwait, I-6 is sent.
ANI End Tone Presentation Allowed
Use following command to program the ANI End Tone Presentation Allowed:
7122-1-T1E1-ANI End Tone Presentation Allowed
7122-2-T1E1-T1E1-ANI End Tone Presentation Allowed
7122-*-ANI End Tone Presentation Allowed
Where,
By default, ANI End Tone Presentation Allowed is 15.
• ANI End Tone Presentation Allowed is 00 or 11 to 15. If no such tone is sent, set this parameter to 00. For
For example, India uses A-4, China uses A-1, etc.
ANI End Tone Presentation Restricted
Use following command to program the ANI End Tone Presentation Restricted:
7123-1-T1E1-ANI End Tone Presentation Restricted
7123-2-T1E1-T1E1-ANI End Tone Presentation Restricted
7123-*-ANI End Tone Presentation Restricted
Where,
By default, ANI End Tone Presentation Restricted (Inbound) is 00.
ANI End Tone Presentation Restricted is 00 or 11 to 15. If no such tone is sent, set this parameter to 00. For
example: India uses A-4, China uses A-1, etc.
Ask Calling Party Sub Category
Use following command to program the Ask Calling Party Sub Category:
7160-1-T1E1-Code
7160-2-T1E1-T1E1-Code
7160-*-Code

Where,
Code is
1 to enable
0 to disable
Forward Group B
Ordinary Subscriber
Use following command to program the Ordinary Subscriber:

7124-1-T1E1-Ordinary Subscriber
7124-2-T1E1-T1E1-Ordinary Subscriber
7124-*-Ordinary Subscriber
Where,
Ordinary Subscriber is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Ordinary Subscriber is 01.
Priority Subscriber
Use following command to program the Priority Subscriber:

7125-1-T1E1-Priority Subscriber
7125-2-T1E1-T1E1-Priority Subscriber
7125-*-Priority Subscriber
Where,
Priority Subscriber is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Priority Subscriber is 02.
Maintenance Equipment
Use following command to program the Maintenance Equipment:

7126-1-T1E1-Maintenance Equipment
7126-2-T1E1-T1E1-Maintenance Equipment
7126-*-Maintenance Equipment
Where,
Maintenance Equipment is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Maintenance Equipment is 03.
Operator
Use following command to program the Operator:

7127-1-T1E1-Operator
7127-2-T1E1-T1E1-Operator
7127-*-Operator
Where,
Operator is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Operator is 05.
Pay Phone
Use following command to program the Pay Phone:

7128-1-T1E1-Pay Phone

7128-2-T1E1-T1E1-Pay Phone
7128-*-Pay Phone
Where,
Operator is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Pay Phone is 00.
Data Transmission
Use following command to program the Data Transmission:

7129-1-T1E1-Data Transmission
7129-2-T1E1-T1E1-Data Transmission
7129-*-Data Transmission
Where,
Data Transmission is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Data Transmission is 06.
Interception Operator
Use following command to program the Interception Operator:

7130-1-T1E1-Interception Operator
7130-2-T1E1-T1E1-Interception Operator
7130-*-Interception Operator
Where,
Interception Operator is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Interception Operator is 00.
Backward Group A
Send next Digit (N+1) (DNIS)
Use following command to program the Send next Digit:

7131-1-T1E1-Send Next Digit
7131-2-T1E1-T1E1-Send Next Digit
7131-*-Send Next Digit
Where,
Send next Digit is 00, 01 to 15. 00 is used for No tone.
By default, Send Next Digit is 01.
Send last but one Digit (N-1) (DNIS)
Use following command to program the Send last but one Digit:
7132-1-T1E1-Send Last But One Digit
7132-2-T1E1-T1E1-Send Last But One Digit
7132-*-Send Last But One Digit
Where,
Send Last But One Digit is 00, 01 to 15. 00 is used for No tone.
By default, Send Last But One Digit (N-1) is 02.

Send last but two digit (N-2) (DNIS)
Use following command to program the Send last but two digit:
7134-1-T1E1-Send Last But Two Digit
7134-2-T1E1-T1E1-Send Last But Two Digit
7134-*-Send Last But Two Digit
Where,
Send Last But Two Digit is 00, 01 to 15. 00 is used for No tone.
By default, Send Last But Two Digit (N-2) is 07.
Send last but three Digit (N-3) (DNIS)
This parameter specifies the backward group A tone used to request last but three digit, that is, N-3 digit. Be it ANI
digit or DNIS digit.
Use following command to program the Send Last But Three Digit:
7135-1-T1E1-Send Last But Three Digit
7135-2-T1E1-T1E1-Send Last But Three Digit
7135-*-Send Last But Three Digit
Where,
Send last but three digit is 00, 01 to 15. 00 is used for No tone.
By default, Send Last But Three Digit is 08.
Send Caller Party Category and ANI Digit
Use following command to program the Send Caller Party Category and ANI Digit:
7135-1-T1E1-Send Caller Party Category and ANI Digit
7135-2-T1E1-T1E1-Send Caller Party Category and ANI Digit
7135-*-Send Caller Party Category and ANI Digit
Where,
Send Caller Party Category and ANI Digit is 00 to 15.
By default, Send Last But Three Digit is 08.
Address-Complete, Change over to Group B
Use following command to program the Address-Complete, Change over to reception of Group B signals:
7136-1-T1E1-Address-Complete, Change Over to Reception of Group B Signals
7136-2-T1E1-T1E1-Address-Complete, Change Over to Reception of Group B Signals
7136-*-Address-Complete, Change Over to Reception of Group B Signals
Where,
Address-Complete, Change Over to Reception of Group B Signals is 00, 01 to 15. 00 is used for No tone.
By default, Address-Complete, Change Over to Reception of Group B Signals is 03.
Send Calling Party Category and Change to Group C
Use following command to program the Send Calling Party Category and Change to Group C:
7137-1-T1E1-Send Calling Party Category and Change to Group C
7137-2-T1E1-T1E1-Send Calling Party Category and Change to Group C
7137-*-Send Calling Party Category and Change to Group C
Where,

Send Calling Party Category and change to Group C is from 00, 01 to 15. 00 is used for No tone.
By default, Send Calling Party Category and Change to Group C is 00.
Congestion in National Network
Use following command to program congestion in the National Network

7138-1-T1E1-Congestion in National Network
7138-2-T1E1-T1E1-Congestion in National Network
7138-*-Congestion in National Network
Where,
Congestion in National Network is 00, 01 to 15. 00 is used for No tone.
By default, Congestion in National Network is 04.
Send Calling Party Category
Use following command to program the Send caller party's category:

7139-1-T1E1-Send Calling Party Category
7139-2-T1E1-T1E1-Send Calling Party Category
7139-*-Send Calling Party Category
Where,
Send Caller Party Category is 00, 01 to 15. 00 is used for No tone.
By default, Send Calling Party's Category is 05.
Address-Completed, Charge, Set Speech Conditions
Use following command to program the Address-Complete, Charge, Set-up Speech Conditions of Group B signals:
7140-1-T1E1-Address-Complete, Charge, Set-up Speech Conditions
7140-2-T1E1-T1E1-Address-Complete, Charge, Set-up Speech Conditions
7140-*-Address-Complete, Charge, Set-up Speech Conditions
Where,
Address-Complete, Charge, Set-up Speech Conditions is 00, 01 to 15. 00 is used for No tone.
By default, Address-Complete, Charge Set-up Speech Conditions is 06.
Repeat DNIS digits from Beginning
Use following command to program the repeat DNIS digits from beginning of Group B signals:
7141-1-T1E1-Repeat DNIS Digits from Beginning
7141-2-T1E1-T1E1-Repeat DNIS Digits from Beginning
7141-*-Repeat DNIS Digits from Beginning
Where,
Repeat DNIS Digits from Beginning is from 00, 01 to 15. 00 is used for No tone.
By default, Repeat DNIS Digits from Beginning is 00.
Send Next ANI Digit
Use following command to program the Send Next ANI Digit

7142-1-T1E1-Send Next ANI Digit
7142-2-T1E1-T1E1-Send Next ANI Digit
7142-*-Send Next ANI Digit
Where,
Send Next ANI Digit is 00 or 01 to 15. If no such tone is sent, set this parameter to 00.

By default, Send Next ANI Digit is 05.
• Few countries use different tone to request next ANI digit and next DNIS digits.
• For example: India uses A-4, China uses A-1, etc.
Backward Group B
Send Special Information Tone
Use following command to program the Send Special Information Tone:

7143-1-T1E1-Send Special Information Tone
7143-2-T1E1-T1E1-Send Special Information Tone
7143-*-Send Special Information Tone
Where,
Send Special Information Tone is from 00, 01 to 15. 00 is used for No tone.
By default, Send Special Information Tone is 02.
Send Special Information Tone and Setup Speech Conditions
Use following command to program the Send Special Information Tone and Setup Speech Condition:
7144-1-T1E1-Send Special Information Tone and Setup Speech Conditions
7144-2-T1E1-T1E1-Send Special Information Tone and Setup Speech Conditions
7144-*-Send Special Information Tone and Setup Speech Conditions
Where,
Send Special Information Tone, and Setup Speech Conditions is from 00, 01 to 15. 00 is used for No tone.
By default, Send Special Information Tone and Setup Speech Conditions is 02.
Subscriber Line Busy
Use following command to program the subscriber line busy:

7145-1-T1E1-Subscriber Line Busy
7145-2-T1E1-T1E1-Subscriber Line Busy
7145-*-Subscriber Line Busy
Where,
Subscriber Line Busy is from 00, 01 to 15. 00 is used for No tone.
By default, Subscriber Line Busy is 03.
Subscriber Line Free, Charge
Use following command to program the subscriber line free, charge:

7146-1-T1E1-Subscriber Line Free, Charge
7146-2-T1E1-T1E1-Subscriber Line Free, Charge
7146-*-Subscriber Line Free, Charge
Where,
Subscriber Line Free, Charge is from 00, 01 to 15. 00 is used for No tone.
By default, Subscriber Line Free, Charge is 06.
Subscriber Line Free, No charge
Use following command to program the subscriber line free, no charge:

7147-1-T1E1-Subscriber Line Free, NO Charge
7147-2-T1E1-T1E1-Subscriber Line Free, NO Charge

7147-*-Subscriber Line Free, NO Charge
Where,
Subscriber Line Free, NO Charge is from 00, 01 to 15. 00 is used for No tone.
By default, Subscriber Line Free, NO Charge is 07.
Congestion
Use following command to program Congestion:

7148-1-T1E1-Congestion
7148-2-T1E1-T1E1-Congestion
7148-*-Congestion
Where,
Congestion is from 00, 01 to 15. 00 is used for No tone.
By default, Congestion is 04.
Unallocated Number
Use following command to program the Unallocated Number:

7149-1-T1E1-Unallocated Number
7149-2-T1E1-T1E1-Unallocated Number
7149-*-Unallocated Number
Where,
Unallocated Number is from 00, 01 to 15. 00 is used for No tone.
By default, Unallocated Number is 05.
Subscriber's Line Out of Order
Use following command to program the subscriber's line out of order:

7150-1-T1E1-Subscriber's Line out of Order
7150-2-T1E1-T1E1-Subscriber's Line out of Order
7150-*-Subscriber's Line out of Order
Where,
Subscriber's Line out of order is from 00, 01 to 15. 00 is used for No tone.
By default, Subscribers Line Out of Order is 08.
Call Rejected, No Indication
Use following command to program the Call rejected, No indication of cause:

7151-1-T1E1-Call rejected, No indication
7151-2-T1E1-T1E1-Call rejected, No indication
7151-*-Call rejected, No indication
Where,
Call rejected, No indication is from 00 to 15.
By default, Call rejected, No indication is 00.
Alternative Answer Tone
Use following command to program the Alternative Answer Tone:

7152-1-T1E1-Alternative Answer Tone
7152-2-T1E1-T1E1-Alternative Answer Tone
7152-*-Alternative Answer Tone

Where,
Alternative Answer Tone is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Alternative Answer Tone is 00.
Changed Number
Use following command to program the Changed Number (announcement on line):

7153-1-T1E1-Changed Number
7153-2-T1E1-T1E1-Changed Number
7153-*-Changed Number
Where,
Changed Number (announcement on line) is from 00, 01-15. Use '00' when this parameter is not applicable.
By default, Changed Number (announcement on line) is 00.
Backward Group C
Send Next ANI digit
Use following command to program the Send next ANI digit:

7154-1-T1E1-Send Next ANI Digit
7154-2-T1E1-T1E1-Send Next ANI Digit
7154-*-Send Next ANI Digit
Where,
Send next ANI digit (Group C) is from 00, 01 to 15. Use '00' when this parameter is not applicable.
This parameter is applicable in Mexico only.
By default, Send Next ANI Digit (Group C) is 00.
Request Transition to Group A and Restart From First DNIS
Use following command to program the Request transition to Group A and restart from first DNIS.
7155-1-T1E1-Request Transition to Group A and Restart from First DNIS
7155-2-T1E1-T1E1-Request Transition to Group A and Restart from First DNIS
7155-*-Request Transition to Group A and Restart from First DNIS
Where,
Request Transition to Group A and Restart from First DNIS is from 00, 01 to 15. Use '00' when this parameter is not
applicable.
By default, Request Transition to Group A and Restart from First DNIS is 00.
Address Completed, Change to Reception of Group B
Use following command to program the Address completed, change to reception of Group B signal:
7156-1-T1E1-Address Completed, Change to Reception of Group B Signal
7156-2-T1E1-T1E1-Address Completed, Change to Reception of Group B Signal
7156-*-Address Completed, Change to Reception of Group B Signal
Where,
Address Completed, Change to Reception of Group B Signal is from 00, 01 to 15. Use '00' when this parameter is
not applicable.
By default, Address Completed, Change to Reception of Group B Signal is 00.

Congestion
This parameter specifies the backward group C tone used to signify Congestion
Use following command to program the tone for Congestion:

7157-1-T1E1-Congestion
7157-2-T1E1-T1E1-Congestion
7157-*-Congestion
Where,
Congestion is from 00, 01 to 15. Use '00' when this parameter is not applicable.
By default, Congestion is 00.
Request Transition Back to Group A, and Sent Next DNIS
Use following command to program the tone for request transition back to group A, and send next DNIS signal:
7158-1-T1E1-Request Transition Back to Group A, and Send Next DNIS
7158-2-T1E1-T1E1-Request Transition Back to Group A, and Send Next DNIS
7158-*-Request Transition Back to Group A, and Send Next DNIS
Where,
Request transition back to group A, and send next DNIS is from 00, 01 to 15. Use '00' when this parameter is not
applicable.
By default, Request Transition Back to Group A, and Send Next DNIS is 00.
Request Transition Back to Group A, and Restart the Last DNIS
Use following command to program the tone for request transition back to group A, and repeat the last DNIS:
7159-1-T1E1-Request Transition Back to Group A, and Restart the Last DNIS
7159-2-T1E1-T1E1-Request Transition Back to Group A, and Restart the Last DNIS
7159-*-Request Transition Back to Group A, and Restart the Last DNIS
Where,
Request Transition Back to Group A, and Restart the Last DNIS is from 00, 01 to 15. Use '00' when this parameter
is not applicable.
By default, Request Transition Back to Group A and Restart the Last DNIS is 00.
Line Signal Parameters

C and D Bits
This parameter indicates the default values of C and D bits when the SARVAM UCS transmits line signals.
Use following command to program the C and D Bits for T1E1 port:
7161-1-T1E1-C and D Bits
7161-2-T1E1-T1E1-C and D Bits
7161-*-C and D Bits
Where,
0 00 (C=0, D=0)
1 01

2 10
3 11
By default, CD Bits is 1.
The C and D bits received during an IC call are ignored by the SARVAM UCS.
Invert Bit A
7162-*-Invert Bit A
Where,
0 Disable
1 Enable
Invert Bit B
7163-*-Invert Bit B
Where,
0 Disable
1 Enable
Invert Bit C
7164-*-Invert Bit C
Where,

0 Disable
1 Enable
Invert Bit D
7165-*-Invert Bit D
Where,
0 Disable
1 Enable
E1 Metering Bit
Use following command to program the E1 Metering Bit for the T1E1 port:
7166-1-T1E1-E1 Metering Bit
7166-2-T1E1-T1E1-E1 Metering Bit
7166-*-E1 Metering Bit
Where,
E1 Metering Bit Meaning
0 None
1 Bit-A
2 Bit-B
3 Bit-C
4 Bit-D
By default, E1 Metering Bit is 1.
E1 Metering Pulse Minimum Timer
Use following command to program the Metering Pulse Minimum timer for the T1E1 port:
7167-1-T1E1-E1 Metering Pulse Minimum Timer
7167-2-T1E1-T1E1-E1 Metering Pulse Minimum Timer
7167-*-E1 Metering Pulse Minimum Timer

Where,
E1 Metering Pulse Minimum timer is from 20ms to 1000ms.
By default, E1 Metering Pulse Minimum Timer is 150ms.
Clear Back Signal
Use following command to program the Clear Back Signal for the T1E1 port:
7168-1-T1E1-Clear Back Signal
7168-2-T1E1-T1E1-Clear Back Signal
7168-*-Clear Back Signal
Where,
Clear Back Signal Meaning
1 Release Guard (Ab=1)
2 Forced Release (Bb=0)
By default, Clear Back Signal is 1.
Release Timer
Use following command to program the Release Timer for the T1E1 port:
7169-1-T1E1-Release Timer
7169-2-T1E1-T1E1-Release Timer
7169-*-Release Timer
Where,
Release Timer is from 20ms to 1000ms.
By default, Release Timer is 400 ms.
Line Seizure Acknowledge Wait Timer
Use following command to program Line seizure acknowledge wait timer:

7170-1-T1E1-Line Seizure Acknowledge Wait Timer
7170-2-T1E1-T1E1-Line Seizure Acknowledge Wait Timer
7170-*-Line Seizure Acknowledge Wait Timer
Where,
Line Seizure Acknowledge Wait Timer is from 0001 to 9999 milli seconds.
By default, Line Seizure Acknowledge Wait Timer is 200ms.
Release Guard Timer
Use following command to program Release Guard Timer:

7171-1-T1E1-Release Guard Timer
7171-2-T1E1-T1E1-Release Guard Timer
7171-*-Release Guard Timer
Where,
T1E1 Release Guard Timer is from 0000 to 9999 milliseconds.
By default, Release Guard Timer is 200ms. This timer depends on the speed of switching and processing.

E&M Signaling
Assign E&M Feature Template to T1E1 using following command:

Use Following command to assign E&M Feature Template to T1E1 Port
6004-1-T1E1-Template Number
6004-2-T1E1-T1E1- Template Number
6004-*- Template Number
Where,
Template Number is 01 to 50.
By default, Template 01 is assigned to T1E1.
B Bit Pattern
Use following command to select B Bit Pattern

7191-1-T1E1-Code
7191-2-T1E1-T1E1-Code
7191-*-Code
Where,
Code Meaning
1 Same as A bit
2 Fixed Value
By default, the Code is 1 (Same as A bit)
B Bit Value
Use following command to program B Bit Value

7192-1-T1E1-B Bit Value
7192-2-T1E1-T1E1- B Bit Value
7192-*- B Bit Value
Where,
B bit value can be 0 or 1.
By default, B bit value is 0.
CD Bit Value
Use following command to program CD Bit Value

7193-1-T1E1-CD Bit Value
7193-2-T1E1-T1E1-CD Bit Value
7193-*- CD Bit Value
Where,
CD bit value can be 1 or 3.
By default, CD bit value is 1.
Invert Bit A

7162-*-Invert Bit A
Where,
0 Do not invert
1 Invert
Invert Bit B
7163-*-Invert Bit B
Where,
0 Do not invert
1 Invert
Invert Bit C
7164-*-Invert Bit C
Where,
Invert Bit C Meaning
0 Do not invert
1 Invert
Invert Bit D
7165-*-Invert Bit D
Where,

0 Do not invert
1 Invert
Viewing T1E1 Trunk Status

You can view the status of T1E1 Trunks on Jeeves only. To do this,
• Under T1E1 Configuration, click Status.
• For each T1E1 Trunk, the following settings will be displayed:

• T1E1 Port No.
• Name
• Layer 1
• Layer 2
• Alarms
• Performance Monitoring Counter
You can also view the T1E1 Trunk Status from the Status link. To view, click the T1E1 link under Status.

Configuring Data Parameters
The SARVAM UCS supports 4 Data ports. For more information about this card, see the topic “The Data Card”.
You can configure the Data Port Parameters only when you have set E1 as the Carrier type in the Port
Parameters under T1E1 Configuration. See “Configuring E1 Trunks” for more details. To configure the Data Port
Parameters, see “E1 Data Settings”.
Configuring Data Parameters using Jeeves

• Under Configuration, click E1-Data Settings to open the page.
E1 Data Settings
• Click E1-Data Settings to expand.
• Select the T1E1 Port tab, for which you want to configure the data parameters.
• By default, both Framing and Signaling are enabled. You can divide the E1 channels to route Voice
calls as well as for Data communication.
• Select the number of E1 Channels to be used for routing data from E1 to the desired Data port in Start
Channel and End Channel. Default: Start and End Channel are blank.
• In Data Port Number, select the desired Data Port to which the E1 Port is to be mapped. Default: 1.

• The E1 Channels not configured above to route Data, shall be used to route Voice Calls.
• By default, Framing is enabled. You can use this E1 Port for both Voice Calling and Data
Communication.
Clear this check box, if you want to use this E1 Port for Data Communication only. When you clear this
check box, Framing is disabled and you can allot all the E1 channels to a single data port only.
In Assign all channels of this E1 Port to Data Port Number, select the desired Data Port to which
you want to assign all the E1 channels.
• By default, Signaling is enabled. You can use this E1 Port for both Voice Calling and Data
Communication.

Clear this check box, if you want to use this E1 Port for Data Communication only. When you clear this
check box, Signaling is disabled and you can allot different E1 channels to different data ports.
• Select the number of E1 Channels to be used for routing data from E1 to the desired Data port in Start
Channel and End Channel. Default: Start and End Channel are blank.
• In Data Port Number, select the desired Data Port to which the E1 Port is to be mapped. Default: 1.

• Repeat the same steps to map another E1 port to a Data port and select the channels to be used for
routing data between the two ports.
DATA Card Firmware Upgrade

• Click DATA Card Firmware Update to expand.
• Select the Use ETHERNET-4 port for FTP access of DATA Card check box to access the FTP.

• Click OK.
• Enter the IP Address through which the FTP of the Data Card can be accessed. Default: Blank
• Enter the Subnet Mask. Default: Blank
• Enter the Default Gateway. Default: Blank
• Click Submit.

Configuring BRI Trunks
BRI port of the SARVAM UCS configured for NT mode can be connected to the BRI Port of another System
configured for TE mode. In such case, the SARVAM UCS will behave as Transit Exchange.
Dialing method (en bloc/overlap) on the BRI port is programmable. This is because many PSTN exchanges support
only Overlap receiving. Hence, in such cases the BRI port (configured for TE mode) will send the called party
number in overlap mode.
SARVAM UCS supports a software port entity called "ISDN Terminal" which can be connected to the BRI-NT port
and will be treated as stations of the System.
How it works
• The ISDN numbering plan with SARVAM UCS revolves around Multiple Subscriber Number (MSN) and
Direct Dialing In (DDI).
• With MSN feature, practically all the station users or other specific terminals can be given a unique
telephone number. One station can be assigned a number 2765400, other station can be assigned
2765401, etc. Hence, for one subscriber the Service Provider can assign multiple numbers. Hence this
feature is called Multiple Subscriber Number.
• The SARVAM UCS can be programmed to land all the calls coming through various channels of the BRI
line on the Operator just like in case of normal trunk.
• Alternatively, using DDI feature of ISDN the calls can be made to land directly on the desired stations. To
accomplish this requirement, each station should be given a unique directory number. On assigning
directory number, a table is formed internally called DDI table as shown below:
Station (S/w Port No.) Directory Number
000 03
005 04
006 05
008 06
009 07
• When a caller calls a MSN number, the call lands on the System. The System compares the incoming
number with the DDI table. If the incoming number matches with any number in the DDI table, it routes the
call to the specific station. If the incoming number does not match with any number in the DDI table then it
is matched with CLI number. If it matches with any number in the CLI table, the call is routed according to
the CLI table. If the number does not match with either of these tables then the call is routed to the landing
destination.

How to connect the BRI line?
ISDN
Network NT BRI Port
Power
SARVAM UCS
U-Interface S/T Interface
(2-wire)
Customer Premises
• Most of the Service Providers provide the NT1 along with the BRI line.
• At the Customer's Premises, the BRI line is terminated on the NT1. The S/T interface of the NT1 is
connected to BRI port of the SARVAM UCS.
The configuration details of U interface (RJ-45) at NT1 are given below:
Pin Number Pin Detail
4 Tx
5 Rx
The configuration details of S/T interface (RJ-45) on NT1 are given below:
Pin Number Pin Detail
3 Rx1
4 Tx1
5 Tx2
6 Rx2
Association of BRI port and ISDN Terminal:

• The 'ISDN terminal' is defined as a real port type in the system.
• Since 32 BRI ports are supported at present and as per protocol a maximum of 8 terminals can be
connected to the BRI port; at present 64 software ports are supported for ISDN terminals, namely 01 to 64
(Maximum 64 Terminals).
• ISDN terminals (software ports) do not have any hardware slot and port Id of their own.
• ISDN terminals (software ports) are associated to the BRI software port and the BRI software port has a
hardware slot and port ID of its own.
• Each ISDN terminal is assigned an access code (flexible number), station basic template, station
advanced template and a CPU group.

The ISDN terminals do not have a direct software port number as other port types have. Rather the ISDN
terminals have a derived software port number. Such derived software port numbers are used while
programming extension name or assigning Station Basic Feature or Station Advanced Feature templates
or programming Routing Group.
For Call routing from the BRI NT:
• When call is made from the ISDN terminal the BRI port will check the calling party number sent by the
ISDN Terminal. If the calling party number = Programmed access code of the ISDN Terminal which are
assigned to the BRI port, the BRI Terminal will process the call further as per the Station Basic Feature
Template and Station Advanced Feature Template assigned to the ISDN Terminal.
• If the ISDN terminal doesn't send the calling party number while making the OG call, the Station Basic/
Advanced Feature template assigned to the BRI port will be used, for call processing.
• When the ISDN terminal doesn't send the calling party number = Access code programmed for the BRI
port on which it is connected, while making the OG call, the Station Basic/Advanced Feature Template
assigned to the BRI port will be applied.
• When ISDN terminal sends calling party number which is not programmed for any ISDN terminals
assigned to that particular BRI Port, neither its access code of that particular BRI port, the call will be
dropped, by the BRI port.
BRI port Orientation Type

• BRI port can be connected to a public ISDN exchange, private ISDN exchange or to an ISDN terminal (BRI
Access).
• When the BRI port is connected to a Public ISDN exchange, it behaves as a terminal. SARVAM UCS
supports this function of BRI Access by assigning it a parameter viz. Orientation = Terminal. The signaling
used for this Orientation is Q.931-protocol.
• When the BRI port is connected to an ISDN Phone or ISDN Video phone, it behaves as a 'network'.
SARVAM UCS supports this function of BRI Access by assigning it a parameter viz. Orientation = Network.
In this case, the BRI port behaves as network. The signaling used for this Orientation is Q.931- protocol.
• When the BRI port is connected to a private ISDN (the main application of this configuration is CUG,
feature transparency, etc.), the BRI port is used as a pipe of 128Kbps. It is of no significance which end
acts as network and which acts as terminal (User) since the role of the BRI port in such case will be to
route the call depending on the signaling protocol applied on the BRI port (128Kbps link). SARVAM UCS
supports this function of BRI Access by assigning it a parameter viz. Orientation = Tie-line. In this case, the
BRI port behaves as 128Kbps link. The signaling used for this Orientation can be any of the Inter-
exchange signaling protocol for BRI Access. The most commonly used is QSIG.
TEI negotiation on BRI port

• BRI port supports both automatic and fixed TEI negotiation, so that it can be integrated with a system of
another vendor. If 'Fixed' mode is selected, its value is required to be configured using SE commands or
Jeeves. TEI is 'Terminal Endpoint Identifier' protocol for negotiation used while connecting to the BRI port
with remote BRI port.

Applications of BRI-NT Port:
Two applications are described below:
• Video Phone
• Making Data Call
Video Phone
• An important application of BRI-NT is establishing a Video Call using Video phone. It is used just like other
ISDN Phones.
• Video Phone can be used with SARVAM UCS supporting a BRI connection. SARVAM UCS does not
support call transfer from one Video Terminal to another Terminal. But the call routing is implemented by
preparing suitable OG Trunk Bundle Group (OGTBG).
• The OGTBG for the Video Phone should be so formed by the SE that a BRI channel will be allotted by the
Call Processing logic (preferably).
• If the OGTBG formed by the SE contains a non-ISDN trunk and if by the OGTBG logic this non-ISDN trunk
is to be allotted to the ISDN phone which has requested 2 B-channels then the call will be dropped. The
non-ISDN trunk is allowed only if the ISDN user makes an audio call. The system will allot first channel
even if other channels are available.
OG Call using Video Phone:

• The SARVAM UCS will support supplementary feature viz. Hold only. If the terminal equipment uses Hold
feature to hold a video call, it will be applicable.
• The SARVAM UCS does not support logic for converting audio call to video call.
• Instead, the Video Phone user will have to use one Trunk Access Code (and hence OGTBG) to make and
receive Video call and another TAC to make an audio call.
IC call to a Video Phone:

• User can program normal DDI routing logic to route the IC call to the Video Phone using the first or
additional channels.
• Depending upon terminal equipment call, the call will be answered by the Operator and transferred to the
Video Phone.

Video Conference
As shown in the figure SARVAM UCS supports Video conference from BRI-NT port.
SARVAM UCS
ISDN BRI TE
Video
Conferencing
BRI NT System
• This feature works only if it is supported by the Service Provider. Video Conference is established mainly
by the Video Conferencing (VC) equipment.
• A Video Conference system (H.320) can be connected to any of the BRI-NT port.
• For Video conferencing, three BRI-NT ports of SARVAM UCS are connected to the three ports of the VC
equipment (The remaining one port of the Video Conference system remains free).
• Thus user will assign at least 6 B-channels to the OGTBG that is to be assigned to the VC equipment.
(User can assign BRI-NT software port nos. 01 to 03 to the Video Conferencing equipment).
• The user will program the routing of IC calls to the Video Conferencing equipment using DDI routing table
for placing IC video conferencing calls.
• More than one VC-equipment can be connected to a BRI-NT Port.
• Video Conferencing generally requires 6 B-channels. But Video Conferencing can also be done at
lower bit rates also using the “aggregation” of 6 B channels which must be supported by the VC
equipment.
• Please note that a Video Conference call cannot be transferred or kept on Hold.
Making Data Calls

• SARVAM UCS BRI-NT port supports data communication also.

• For data-communication, connect the Router supporting ISDN-BRI or a Computer with the BRI Card ( BRI-
NT port) as shown below:
SARVAM UCS
ISDN BRI TE
BRI NT
Using this feature following applications are supported:
• The computers connected in the LAN can browse the net through the BRI.
• Remote LAN Access, the Computers in the LAN can access the computer/computers in LAN at the
remote end (Branch office/Home office).
• Files can be transferred from one LAN to another.
OG Data calls from the Router:

• The data call can be made by the router requesting desired number of channels. This establishes a live
connection between the Router and the ISP through the system. The users on the LAN can browse the net
as normal using Internet Explorer or Netscape Navigator.
• For this, the SARVAM UCS will allocate data channels only on the BRI-TE port so as to leave other
channels for speech calls when the system detects the call to be a data call.
• Similarly, a Remote Computer can be accessed (Remote LAN Access) by dialing the Remote users'
number (The remote end system should be so programmed that the call made to a number lands directly
on the Router.) This establishes a permanent connection between the two Routers (and hence two LAN
networks).
• Now the user of the system can access the computer in LAN at the remote end in the same way as
accessing another computer on the same network.
IC Data calls to the Router:

• Program the Trunk Landing Group such that all data calls will land on the BRI-NT port to which the Router
is connected using CLI based Routing logic or DDI based routing logic.
• However, while routing call on the BRI-NT port, the system will check that the data channels reserved for
data communication on the BRI-NT port are enough to establish the call. Otherwise the call will be
rejected.
• The call will be rejected if the number of channels reserved for data calls, are already busy with one data-
call.

Configuring BRI Parameters using Jeeves
• Under Configuration, click BRI Configuration.
• Click BRI Parameters.
• SARVAM UCS will assign the Hardware Slot-Port automatically, when any card is inserted in the system.
Hardware slot is the number of the Universal slot of SARVAM UCS in which the BRI Card is inserted.
Range of slot number is 01-16. Port is the number of BRI hardware port on the card to which the BRI line
is connected. Range of Port is from 00-99.
Port is 00–00.
• Keep the BRI Port enabled.
Clear the Enable Port check box, only if you do not want to use this port. By default, it is enabled.
• You may assign a Name to the BRI Port for identification of the port. The Name may consist of a maximum
of 12 characters. By default, it is blank.
• Set Orientation Type for BRI Port. You may select from the following options:
• Terminal
• Network
• Tie Line
By default, Orientation Type is Terminal.

Orientation Type - Terminal
If you select Terminal as Orientation Type, configure the following:
• Enable Power Feed flag, if you want SARVAM UCS to feed power to the terminal equipment. By default, it
is disabled.
• Select Interface Type as Point-to-Point or Point-to-Multipoint according to your installation. By default, it is

Point-to-Multipoint.
• The TEI Negotiation mode will be set automatically as per the Interface Type selected for the BRI Port.
• If the Point-to-Point is selected as the Interface Type, the TEI Negotiation will be set as Fixed.
• You must also configure TEI Negotiation Value (for Fixed Mode). TEI Value range is from 00 to
63. By default, it is 00. TEI Value of BRI (TE) port of SARVAM UCS should match with the TEI Value
expected by the NT equipment at other end.
• If Point-to-Multipoint is selected as the Interface Type, the TEI Negotiation will be set as Auto.
When you change the TEI mode on any port, the BRI Card will restart.
If you select Trunk, the system will treat all incoming calls as external calls landing on the trunk. The calls
will be routed as per the Trunk Feature Template assigned to the BRI Port.
Feature Template to the BRI Port.
Station Advanced Feature Template assigned to the BRI Port.
• Different countries use specific type of ISDN switch. The type of switch determines various factors such as
how many ISDN devices would be handled, which B-channel will support voice, video, data etc. Select
ISDN Switch Variant supported by your country. You may select from the following options:
• ATT_4ESS
• ATT_5ESS

• AUSTRALIA
• AUTO CONFIG
• DMS_100
• ETSI_NET3
• NTT_INS64
• SWV_HONG_KONG
• US_NI1
• US_NI2
• VN_X
By default, it is ETSI_NET3.
• Configure Outgoing (OG) Reference ID for working hours, non-working hours and break hours. By
default, OG Reference ID is 00.
• Configure Incoming (IC) Reference ID for working hours, non-working hours and break hours. By default,
IC Reference ID is 00.
• Assign Trunk Feature Template to the BRI Port. Trunk Feature Template is a set of general features that
define the behavior of a Trunk Port. By default, Template 01 is assigned to all BRI Ports.
• Assign a Cost Factor to the BRI Port. By default, all the BRI Ports are assigned Cost Factor 01.
• Station Basic Feature Template assigned to the BRI Port is displayed in this field. Station Basic Feature
Template is a set of general features that define the basic behavior of a station. By default, Template 01 is
assigned to all BRI Ports.
• Station Advanced Feature Template assigned to the BRI Port is displayed in this field. Station Advanced
Feature Template is a set of advanced features, to be applied on extensions such as CLIP, Floor Service,
Walk-in Class of Service. By default, Template 01 is assigned to all BRI Ports.
• If you have set Terminal as the Orientation Type, you must select the type of network with which the port
is to be Interfaced With. Select the type of network from the following:
• Public ISDN
• Private ISDN
Display Called Party Number as CLI check box. By default, Display Called Party Number as CLI option
is disabled.
This option is useful when a single BRI line connection and Operator are shared by more than one
• you configure the names and corresponding numbers of the organizations sharing the line in the Global
Directory of SARVAM UCS.

• Select Layer 1 Mode depending upon the type of line terminated on BRI port of SARVAM UCS.
The Public ISDN provides different types of lines in different countries as mentioned below:
• On Demand
• Always ON
In 'On Demand' type of line, layer 1 (physical layer) remains 'down' when the line is idle. When the network
places incoming call, it activates the layer 1 and places the call. When the terminal makes out going call,
the layer 1 gets activated automatically.
In 'Always ON' type of line, layer 1 is always 'up', in normal condition. By default, Layer 1 Mode is 'Always
ON'.
When the system detects 'layer 1 down', it removes the unhealthy line from the OGTBG. If you have
assigned a DSS Key to the BRI Port/channel, the faulty condition is indicated by LED indication.
DSS key LED behavior

When Layer 1 of BRI line is down or not connected, DSS key LED will glow 1 sec On and 3 sec Off in violet
color.
If Layer 1 of BRI line is up, the DSS key LED will be turned off.
• SARVAM UCS uses the BRI Port to place call only if the layer 1 is 'up'. When layer 1 goes 'down',
SARVAM UCS considers the line as un-healthy and will not use this BRI as destination port. SARVAM
UCS will place the call using the alternate port programmed in the same routing group.
• The default Layer 1 Mode is 'Always ON'. Hence, if the interfaced line is of the type 'On Demand', the
calls will not get routed through BRI port, unless the 'Layer 1 Mode' is changed to 'On Demand'.
• When the port is un-healthy, the SARVAM UCS routes the call using other healthy port. However, this
depends upon the member selection method and other ports programmed in the OG Trunk Bundle
Group (OGTBG). Refer chapter “OG Trunk Bundle Group” for more details.
• Select Priority for the BRI Port. Priority is the precedence given to certain trunks and extensions over
Normal is set for all BRI Ports.

• Select Return Call to Original Caller (RCOC) check box to enable this feature on the BRI Port. By
default, RCOC flag is disabled.
• Set Overlap Receiving Timer for BRI Port. This timer is relevant while receiving the called party number
information in overlap receiving mode. By default, it is set to 15 seconds.
responds by sending SETUP ACK (Acknowledgment), and dial tone is played to the caller. The time taken
by the exchange to respond to the SETUP message may vary from exchange to exchange. Set the
SETUP Response Timer (sec) as per the time taken by the network to respond to the SETUP message
and play dial tone to the caller.
• Configure Idle Code for the BRI Port. Range of Idle Code is from 000 to 255. By default, it is 127 (7F).
The binary equivalent of the configured value (000 to 255) is sent on the channel to signify that the channel
is idle (or Used). This setting depends on the network. Most commonly applicable values are 7F and FF
(Binary equivalent is 0111 1111 and 1111 1111, decimal equivalent is 127 and 255).
• Select the number of Channels to be reserved for Data Communication. Channel count is from 0 to 2.
By default, number of channels reserved for Data Communication is 02.
• Select the number of Channels to be reserved for Outgoing Calls. Channel count is from 0 to 2. By
default, number of channels reserved for Outgoing Calls is 02.
• Select the number of Channels to be reserved for Incoming Calls. Channel count is from 0 to 2. By
default, number of channels reserved for Incoming Calls is 02.
• Select the required option for sending the Caller- Type of Numbering Plan (TON) from the following:
• Unknown
• International
• National
• Subscriber
• Abbreviated
• Reserved
By default, Unknown is selected.
• Select the required option for sending the Caller- Numbering Plan Identification (NPI) from the following:
• Unknown
• ISDN Numbering
• Data Numbering

• Telex Numbering
• Private
• Reserved
By default, ISDN Numbering is selected.
• Select the required option for sending the Called-Type of Numbering Plan (TON) from the following:
• Unknown
• International
• National
• Subscriber
• Abbreviated
• Reserved
• Select the required option for sending the Called-Numbering Plan Identification (NPI) from the following:
• Unknown
• ISDN Numbering
• Data Numbering
• Telex Numbering
• Private
• Reserved
• Select the Screening Indicator as provided to you by your exchange. You can select from:
• User provided, not screened
• User Provided, verified and passed
• User Provided, verified and failed
• Network Provided
By default, User provided, not screened.
• Speech
• 3.1 KHz Audio
• Configure Call Budget parameters for the BRI Ports. Call Budget is an expense control feature of
SARVAM UCS that allows you to keep track of the cost of phone calls made from the BRI Port.
• Type: Select the type of Call Budget, that is, Amount or Minutes or Calls to be applied on the BRI Port.
By default, no Call Budget type is selected.

so.
• Call Back: This parameter is related to the ‘Call Back on Trunk Port’ feature. If you want to enable the 'Call
Back on Trunk Port' feature on this BRI Port, configure the following parameters:
before applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to
10 seconds.
should be routed:
• Operator
Number’.
Number List 15 is also assigned to all BRI Ports as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all BRI Ports and Trunk Port types, retain this
list.
If you want a different set of number strings to be programmed for this BRI Port, select a different
Number List, and assign it to the BRI Port.

• Outgoing Number List: Program the number strings that are to be called back in this List. For each
number string you programmed in the ‘Incoming Number List’, you must program in the corresponding
index in the Outgoing Number List a number to which the call back is to be made. For example, for the
number string programmed at Index 1 in the Incoming Number List, a corresponding number string at
the same Index, Index 1, should be programmed in the ‘Outgoing Number List’.
assigned to all BRI Ports as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this BRI Port.
back can be made using the same port or an “OG Trunk Bundle Group”.
Select ‘Same Port’ if you want the call back to be made using the same port on which the missed call is
received. If you select OGTB Group, the call back will be made using the OGTB Group, which you have
defined.
• Configure Pause Timer for the BRI Port. Range of Pause Timer is from 1 to 9 seconds. By default, it is set
to 3 seconds for all BRI Ports.
This Timer is required to insert delay between the digits while dialing out DTMF digits on the BRI port. One
of the applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage Dialing”.
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the SARVAM UCS
will out dial the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
• Configure DTMF On Time for the BRI Port. Range of DTMF On Time is from 051 to 255 ms. By default, it
is set to 102 ms for all BRI Ports.
Stage Dialing”.
• Configure DTMF Inter Digit Pause Timer for the BRI Port. Range of Inter Digit Pause Timer is from 051 to
255 ms. By default, it is set to 102 ms for all BRI Ports.

• Select Category (Logical Partition) for the BRI Port. You may select from the following options:
• 1
• 2
• 3
By default, 1 is selected for all BRI Ports.
• If SARVAM UCS is to be used as a Gateway, enable Gateway Application-Answer Signaling on the BRI
Port and configure DTMF String. By default, Gateway Application-Answer Signaling is disabled and CCC
is configured as DTMF String.
• Enter appropriate Debug Code (Level 1 to 4) to obtain debug information of various parts of BRI Card on
the COM Port. By default, debug is off for all BRI Ports for all levels.
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from 000
to 255. Code value '000' for each level will turn off that level's debug.
Level and Code for BRI Port are as specified below:
Level 1
Unused Flow NLS LAP SVC Primitives Variables State Primitives
Code Meaning
001 Primitives
002 State
004 Variables
008 SVC Primitives
016 LAP
032 NLS
064 Flow Debug
000 Debug Off
Level 2
Unused Unused Unused DTMF Digit Unused Layer 4 Unused Unused
Code Meaning
004 Layer 4
016 DTMF Digit
000 Debug Off

Level 3
Unused Unused Unused HDLC D-Channel Unused Unused Unused Unused
Code Meaning
016 HDLC D-Channel
000 Debug Off
Level 4
Unused Unused Unused Unused Unused Unused OS Task NI
Code Meaning
001 Network Interface (NI)
002 OS Task
000 Debug Off
Example:
• If '004' decimal value is entered as Debug Code for Level 1, then its binary equivalent '100' (0000100)
indicates that debug for "Variables" will get enabled.
If "007" decimal value is entered as a Debug Code for the Level 1, then its binary equivalent "111"
(00000111) indicates that debug for "Primitives", "States" and "Variables" will get enabled.
Orientation Type - Network

If you select Network as Orientation Type, configure the following:
• Select BRI-NT Interface Type as Point-to-Point or Point-to-Multipoint according to your installation. By

default, it is Point-to-Multipoint.
• The TEI Negotiation mode will be set automatically as per the Interface Type selected for the BRI Port.
• If the Point-to-Point is selected as the Interface Type, the TEI Negotiation will be set as Fixed.
• You must also configure TEI Negotiation Value (for Fixed Mode). TEI Value range is from 00 to
63. By default, it is 00. TEI Value configured in the BRI (NT) should match with the TEI Value
configured in the Terminal Equipment connected with it.
• If the Point-to-Multipoint is selected as the Interface Type, the TEI Negotiation will be set as Auto.
• When you change the TEI mode on any port, the BRI Card will reset.
If you select Trunk, the system will treat all incoming calls as external calls landing on the trunk. The calls
will be routed as per the Trunk Feature Template assigned to the BRI Port.

Feature Template to the BRI Port.
Station Advanced Feature Template assigned to the BRI Port.
• Different countries use specific type of ISDN switch. The type of switch determines various factors such as
how many ISDN devices would be handled, which B-channel will support voice, video, data etc. Select
ISDN Switch Variant supported by your country. You may select from the following options:
• ATT_4ESS
• ATT_5ESS
• AUSTRALIA
• AUTO CONFIG
• DMS_100
• ETSI_NET3
• NTT_INS64
• SWV_HONG_KONG
• US_NI1
• US_NI2
• VN_X
By default, it is ETSI_NET3.
• Configure Outgoing (OG) Reference ID for working hours, non-working hours and break hours. By
default, OG Reference ID is 00.
• Configure Incoming (IC) Reference ID for working hours, non-working hours and break hours. By default,
IC Reference ID is 00.
• Assign Trunk Feature Template to the BRI Port. Trunk Feature Template is a set of general features that
define the behavior of a Trunk Port. By default, Template 01 is assigned to all BRI Ports.
• Assign a Cost Factor to the BRI Port. By default, all the BRI Ports are assigned Cost Factor 01.

• Station Basic Feature Template assigned to the BRI Port is displayed in this field. Station Basic Feature
Template is a set of general features that define the basic behavior of a station. By default, Template 01 is
assigned to all BRI Ports.
• Station Advanced Feature Template assigned to the BRI Port is displayed in this field. Station Advanced
Feature Template is a set of advanced features, to be applied on extensions such as CLIP, Floor Service,
Walk-in Class of Service. By default, Template 01 is assigned to all BRI Ports.
Display Called Party Number as CLI check box. By default, Display Called Party Number as CLI option
is disabled.
This option is useful when a single BRI line connection and Operator are shared by more than one
• you configure the names and corresponding numbers of the organizations sharing the line in the Global
Directory of SARVAM UCS.
• Select Layer 1 Mode depending upon the type of line terminated on BRI port of SARVAM UCS.
The Public ISDN provides different types of lines in different countries as mentioned below:
• On Demand
• Always ON
In 'On Demand' type of line, layer 1 (physical layer) remains 'down' when the line is idle. When the network
places incoming call, it activates the layer 1 and places the call. When the terminal makes out going call,
the layer 1 gets activated automatically.
In 'Always ON' type of line, layer 1 is always 'up', in normal condition. In this type of line, 'layer 1 down'
indicates fault condition and calls get failed.
By default, Layer 1 Mode is 'Always ON'.

• When Layer 1 mode is set as 'Always ON', the SARVAM UCS uses this BRI to place call only if the
layer 1 is 'up'. When layer 1 goes 'down', SARVAM UCS considers the line as un-healthy and will not
use this BRI as destination port. SARVAM UCS will place the call using the alternate port programmed
in the same routing group.
• When Layer 1 mode is set as 'On Demand', SARVAM UCS will not check the Layer 1 condition while
placing call through this BRI.
• The default Layer 1 Mode is 'Always ON'. Hence, if the interfaced line is of the type 'On Demand', the
calls will not get routed through BRI port, unless the 'Layer 1 Mode' is changed to 'On Demand'.
• When the port is un-healthy, the SARVAM UCS routes the call using other healthy port. However, this
depends upon the member selection method and other ports programmed in the OG Trunk Bundle
Group (OGTBG). Refer chapter “OG Trunk Bundle Group” for more details.
• Select Priority for the BRI Port. Priority is the precedence given to certain trunks and extensions over
Normal is set for all BRI Ports.
• Select Return Call to Original Caller (RCOC) check box to enable this feature on the BRI Port. By
default, RCOC flag is disabled.
• Set Overlap Receiving Timer for BRI Port. This timer is relevant while receiving the called party number
information in overlap receiving mode. By default, it is set to 15 seconds.
responds by sending SETUP ACK (Acknowledgment), and dial tone is played to the caller. The time taken
by the exchange to respond to the SETUP message may vary from exchange to exchange. Set the
SETUP Response Timer (sec) as per the time taken by the network to respond to the SETUP message
and play dial tone to the caller.
You can configure the SETUP Message Timer only from Jeeves
• Configure Idle Code for the BRI Port. Range of Idle Code is from 000 to 255. By default, it is 127 (7F).
The binary equivalent of the configured value (000 to 255) is sent on the channel to signify that the channel
is idle (or Used). This setting depends on the network. Most commonly applicable values are 7F and FF
(Binary equivalent is 0111 1111 and 1111 1111, decimal equivalent is 127 and 255).
• Select the number of Channels to be reserved for Data Communication. Channel count is from 0 to 2.
By default, number of channels reserved for Data Communication is 02.

• Select the number of Channels to be reserved for Outgoing Calls. Channel count is from 0 to 2. By
default, number of channels reserved for Outgoing Calls is 02.
• Select the number of Channels to be reserved for Incoming Calls. Channel count is from 0 to 2. By
default, number of channels reserved for Incoming Calls is 02.
• Select the required option for sending the Caller- Type of Numbering Plan (TON) from the following:
• Unknown
• International
• National
• Subscriber
• Abbreviated
• Reserved
• Select the required option for sending the Caller- Numbering Plan Identification (NPI) from the following:
• Unknown
• ISDN Numbering
• Data Numbering
• Telex Numbering
• Private
• Reserved
• Select the required option for sending the Called-Type of Numbering Plan (TON) from the following:
• Unknown
• International
• National
• Subscriber
• Abbreviated
• Reserved
• Select the required option for sending the Called-Numbering Plan Identification (NPI) from the following:
• Unknown
• ISDN Numbering
• Data Numbering
• Telex Numbering
• Private
• Reserved
• Speech
• 3.1 KHz Audio

• Configure Call Budget parameters for the BRI Ports. Call Budget is an expense control feature of
SARVAM UCS that allows you to keep track of the cost of phone calls made from the BRI Port.
• Type: Select the type of Call Budget, that is, Amount or Minutes or Calls to be applied on the BRI Port.
By default, no Call Budget type is selected.
so.
• Configure Call Back parameters for the BRI Ports. Call Back is used to respond to missed calls from
particular numbers on the BRI Port.
• Enable Call Back: Enable this flag to activate the Call Back on Trunk Port feature. By default, the flag
is disabled.
before applying the Call Back. The range of this timer is from 01 to 99 seconds. By default, it is set to
10 seconds.
• Call Back Mode: From the following options select how a ‘Call Back’ call answered by the remote party
should be routed:
• Operator
Number’.

Number List 15 is also assigned to all the BRI Ports as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all the BRI Ports and Trunk Port types, retain
this list.
If you want a different set of number strings to be programmed for this CO Trunk, select a different
Number List, and assign it to the CO trunk port.
corresponding index in the Outgoing Number List a number to which the call back is to be made. For
example, for the number string programmed at Index 1 in the Incoming Number List, a corresponding
number string at the same Index, Index 1, should be programmed in the ‘Outgoing Number List’.
assigned to all BRI Ports as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this BRI port.
By default, OGTBG 01 is assigned.
Routing on the OGTBG that you define here for Call Back.
• Configure Pause Timer for the BRI Port. Range of Pause Timer is from 1 to 9 seconds. By default, it is set
to 3 seconds for all BRI Ports.
This Timer is required to insert delay between the digits while dialing out DTMF digits on the BRI port. One
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the SARVAM UCS
will out dial the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.

• Configure DTMF On Time for the BRI Port. Range of DTMF On Time is from 051 to 255 ms. By default, it
is set to 102 ms for all BRI Ports.
Stage Dialing”.
• Configure DTMF Inter Digit Pause Timer for the BRI Port. Range of Inter Digit Pause Timer is from 051 to
255 ms. By default, it is set to 102 ms for all BRI Ports.
• Select Category (Logical Partition) for the BRI Port. You may select from the following options:
• 1
• 2
• 3
By default, 1 is selected for all BRI Ports.
• If SARVAM UCS is to be used as a Gateway, enable Gateway Application-Answer Signaling on the BRI
Port and configure DTMF String. By default, Gateway Application-Answer Signaling is disabled and CCC
is configured as DTMF String.
• Enter appropriate Debug Code (Level 1 to 4), to obtain debug information of various parts of BRI Card on
the COM Port. By default, debug is off for all BRI ports for all levels.
Code is the value for the specified level to turn ON the debug for the parameters. Code range is from 000
to 255. Code value '000' for each level will turn off that level's debug.
Level and Code for BRI Port are as specified below:
Level 1
Code Meaning
001 Primitives
002 State
004 Variables
008 SVC Primitives
016 LAP
032 NLS
064 Flow Debug
000 Debug Off

Level 2
Code Meaning
004 Layer 4
016 DTMF Digit
000 Debug Off
Level 3
Code Meaning
016 HDLC D-Channel
000 Debug Off
Level 4
Code Meaning
002 OS Task
000 Debug Off
Configuring BRI Parameters using Commands

To program a single port: XXXX-1
To program a range of ports: XXXX-2
To program all the ports: XXXX-*
Hardware Slot-Port
Use following command to assign hardware ID to a BRI software port:
1106-BRI-Slot-Port offset
Where,
BRI is from 01 to 32.
Slot is the number of the universal slot, where the BRI Card is installed, from 01 to 16.
Port is the number of the BRI port on the card, from 00 to 99.
Use following command to de-assign the hardware slot and the hardware port assigned to the BRI software port.
1106-BRI-00-00

Name
Use following command to assign a name to the BRI port:

5405-1-BRI-Name
5405-2-BRI-BRI-Name
5405-*-Name
Where,
Name is an alpha-numeric string of 12 characters. Terminate the command with #*.
By default, Name field is Blank.
Enable BRI Port
This parameter is used to enable/disable the port. When the Port configured in TE mode is disabled, it will not be
allotted to the user on grabbing the port. Instead the user will get error tone. Also no IC call will be allowed.
Likewise, when the port is configured in NT mode is disabled, the IC call will not be allowed to land on this port. The
port will be treated as absent and accordingly other activities will be performed like routing the call to other stations
in the group, etc.
Use following command to enable/disable BRI port:

6201-1-BRI-Flag
6201-2-BRI-BRI-Flag
6201-*-Flag
Where,
Flag is 0 for disable, 1 for enable.
By default, the BRI Port is enabled.
Orientation Type
Use following command to program Orientation Type for the BRI port:
6204-1-BRI-Orientation Type
6204-2-BRI-BRI-Orientation Type
Where,
Orientation Type Meaning
1 Terminal
2 Network
3 Tie Line
When Orientation=Terminal, the port will be regarded as trunk. All the trunk related parameters will be applied.
When Orientation=Network, the port will be regarded as station. All the station related parameters will be applied.
When Orientation = Tie-line, the port will be regarded as station for all incoming calls on it and as a trunk for all
outgoing calls made through it.

Power Feed
Use following command to Feed Power to the BRI port:

6227-1- BRI Port- Feed Power to Port
6227-2- BRI Port-BRI Port- Feed Power to Port
6227-*- Feed Power to Port
Where,
BRI Port is from 01 to 32
Feed Power to port is 0 for disable, 1 for enable.
Default it is disabled.
BRI NT Interface Type
This parameter is relevant only when BRI port is used as Network Interface, i.e BRI NT mode.
Use the following command to select the Interface type as Point-to-Point or Point-to-Multipoint:
6226-1-BRI - Interface Type
6226-2-BRI-BRI-Interface Type
6226-*-Interface Type
Where,
Interface Type is
1 for Point-to-Point
2 for Point-to-Multipoint
Default =Point to Point
When ISDN Phones are to be connected with BRI NT of the SARVAM UCS, this parameter should be set as 'Point
to Multipoint'.
When an ISDN System is to be interfaced with BRI-NT of the SARVAM UCS, this parameter should be set as 'Point
to Point'.
BRI ISDN Switch Variant
Use following command to program the ISDN BRI Switch Variant of the BRI port:
6203-1-BRI-ISDN BRI Switch Variant
6203-2-BRI-BRI-ISDN BRI Switch Variant
6203-*-ISDN BRI Switch Variant
Where,
ISDN BRI Variant Meaning
01 ATT_4ESS
02 ATT_5ESS
03 AUSTRALIA
04 AUTO CONFIG
05 DMS_100
06 ETSI_NET3
07 NTT_INS64
08 SWV_HONG_KONG
09 US_NI1

ISDN BRI Variant Meaning
10 US_NI2
11 VN_X
By default, ISDN BRI Switch Variant is 06.
OG Reference ID-WH
Use following command to program an OG Reference ID to a BRI port:

6231-1-BRI-OG Reference ID
6231-2-BRI-BRI-OG Reference ID
Where,
OG Reference ID is from 00 or 01 to 99.
By default, OG Reference ID for BRI is 00.
This command is significant only if the BRI port is configured in TE mode.
OG Reference ID-BH

Where,
OG Reference ID-NH

Where,
IC Reference ID-WH
Use following command to program an IC Reference ID-WH on the BRI port:

6232-1-BRI-IC Reference ID
6232-2-BRI-BRI-IC Reference ID
Where,
By default, IC Reference ID-WH for BRI is 00.

IC Reference ID-BH
Use following command to program an IC Reference ID-BH on the BRI port:

Where,
IC Reference ID is from 00 or 01 to 99.
By default, IC Reference ID-BH for BRI is 00.
IC Reference ID-NH
Use following command to program an IC Reference ID-NH on the BRI port:

Where,
IC Reference ID is from 00 or 01 to 99.
By default, IC Reference ID-NH for BRI is 00.
Use the following command to assign a Trunk feature Template to the BRI Port:
5804-1-BRI- Trunk Feature Template Number

5804-2-BRI-BRI- Trunk Feature Template Number.
5804-*- Trunk Feature Template Number.
Where,
Template Number is the number of the customized Trunk Feature Template, from 01 to 50. Default: Template 01.
Cost Factor
Use following command to assign a Cost Factor to the BRI port:

6202-1-BRI-SP
6202-2-BRI-BRI-SP
6202-*-SP
Where,
Cost Factory is from 01 to 99.
By default, all the BRI ports are assigned Cost Factor = 01.
This parameter is insignificant when BRI port is configured in NT mode.
Use the following command to assign a Station Basic Feature Template to a BRI port:
5509-1-BRI-Template Number
5509-2-BRI-BRI-Template Number.

5509-*-Template Number
Where,
Template is the number of the Station Basic Feature Template, from 01 to 50.
Use the following command to assign a Station Advanced Feature Template to a BRI port:
5609-1-BRI-Template Number
5609-2-BRI-BRI-Template Number
5609-*-Template Number
Where,
Template is the number of the Station Advanced Feature Template, from 01 to 50.
Layer 1 Mode
• This parameter is applicable for BRI port, when orientation type is 'Terminal'.
• The Public ISDN provides different types of lines in different countries as mentioned below:
• On Demand
• Always ON
• In 'On Demand' type of line, layer 1 (physical layer) remains 'down' when the line is idle.
• When the network places incoming call, it activates the layer 1 and places the call.
• When the terminal makes out going call, the layer 1 gets activated automatically.
• In 'Always ON' type of line, layer 1 is always 'up', in normal condition.
• In this type of line, 'layer 1 down' indicates fault condition and calls get failed.
Use following command to program 'Layer 1 Mode' for BRI port:

6225-1-BRI-Layer 1 Mode
Where,
Layer 1 Mode Meaning
1 Always ON
2 On Demand
Select the 'Layer 1 Mode' parameter, depending upon the type of line, terminated on BRI port of SARVAM UCS.
By default, Layer 1 Mode is 'Always ON'.
TEI Negotiation
SE can select the Automatic or Fixed TEI negotiation for each BRI port as required. If Fixed TEI negotiation is
selected, the value of fixed TEI negotiation is required to be programmed.
Use following command to select TEI Negotiation on a BRI port:

6238-1-BRI-TEI Negotiation Mode
6238-2-BRI-BRI-TEI Negotiation Mode
6238-*-TEI Negotiation Mode
Where,
TEI Negotiation mode Meaning
0 Automatic (Non-fixed)
1 Fixed
By default, TEI Negotiation Mode is Automatic.
TEI Value (for Fixed Mode)
Use following command to program TEI Negotiation value when programmed as Fixed:
6239-1-BRI-TEI Value
6239-2-BRI -BRI-TEI Value
6239-*-TEI Value
Where,
TEI Value is from 00 to 63
By default, TEI Value is 00.
The System Engineer should take care of the following points when using the above commands for TEI negotiation:
• When you change the TEI mode on any port, the BRI Card will get reset.
• If you have selected 'Fixed' mode, program the value as per the value of port connected at remote end as
explained below:
• TEI Value programmed in the BRI (NT) should match with the TEI value programmed in the Terminal
equipment connected with it.
• TEI value of BRI (TE) port of SARVAM UCS should match with the TEI value expected by the NT
equipment at other end.
Priority
This command is applicable when BRI port is configured to act as a station.

Use following command to assign priority to BRI:
3916-1-BRI-Priority
3916-2-BRI-BRI-Priority
3916-*-Priority
Where,
BRI is from 01 to 32
Priority is from 1 to 9
By default, Priority for BRI is 5-Normal.
Use the following command to enable RCOC on BRI trunk:

6220-1-BRI-Code
6220-2-BRI-BRI-Code
6220-*-Code

Where,
Code is
0 for Disable
1 for Enable
Default: Disabled
Use following command to set overlap receiving timer for BRI:

6208-1-BRI-Timer
6208-2-BRI-BRI-Timer
6208-*-Timer
Where,
Timer is from 00 to 99 seconds.
This timer is relevant while receiving the called party number information in overlap receiving mode. It is not
relevant for overlap sending mode.
Idle Code
Use following command to program the Idle Code for the BRI port:
6207-1-BRI-Idle Code
6207-2-BRI-BRI-Idle Code
6207-*-Idle Code
Where,
Idle Code is from 000 to 255.
The binary equivalent of the programmed value (000 to 255) is sent on the channel to signify that the channel is
idle. (or Unused) This setting depends on the network. Most commonly applicable values are 7F and FF (Binary
equivalent is 0111 1111 and 1111 1111, decimal equivalent is 127 and 255).
Default: Idle Code for the BRI port is 127 (7F).
Channel Reserved for Data Communication
Use the following command to program number of channels reserved for data transmission:
6235-1-BRI-Channel Count
6235-2-BRI-BRI-Channel Count
6235-*-Channel Count
Where,
Channel Count is from 00 to 02.
By default, Number of Channels is 02.
Channels Reserved for OG Calls
Use following command to reserve the number of channels for OG calls on a BRI port.
Where,

By default, Channel is reserved for OG Calls for BRI is 2. Both the channels are used for receiving OG calls.
Channels reserved for IC Calls
Use following command to reserve the number of channels for IC calls on a BRI port.
Where,
By default, Channel is reserved for IC Calls for BRI is 2. Both the channels are used for receiving IC calls.
Caller - Type of Number (TON)
Use following command to program a Caller TON for the BRI port:
6221-1-BRI-Caller TON
6221-2-BRI-BRI-Caller TON
6221-*-Caller TON
Where,
Caller TON Meaning
1 Unknown
2 International Number
3 National Number
4 Network Specific Number
5 Subscriber Number
6 Abbreviated Number
7 Reserved Number
By default, Caller TON for BRI is Unknown.
Caller Numbering Plan Identification (NPI)
Use following command to program a Caller NPI for the BRI port:
6222-1-BRI-Caller NPI
6222-2-BRI-BRI-Caller NPI
6222-*-Caller NPI
Where,
Caller NPI Meaning
1 Unknown
2 ISDN Numbering
3 Date Numbering
4 Telex Numbering

Caller NPI Meaning
5 National Numbering
6 Private
7 Reserved
By default, Caller NPI for BRI is ISDN Numbering.
Called - Type of Number (TON)
Use following command to program a Called Party destination TON for the BRI port:
6223-1-BRI-Called Party TON
6223-2-BRI-BRI-Called Party TON
6223-*-Called Party TON
Where,
Called Party TON Meaning
1 Unknown
2 International Number
3 National Number
4 Network Specific Number
5 Subscriber Number
6 Abbreviated Number
7 Reserved Number
By default, Called Party TON for BRI is Unknown.
Called Numbering Plan Identification (NPI)
Use following command to program a Called Party NPI for the BRI port:
6224-1-BRI-Called Party NPI
6224-2-BRI-BRI-Called Party NPI
6224-*-Called Party NPI
Where,
Called Party NPI Meaning
1 Unknown
2 ISDN Numbering
3 Date Numbering
4 Telex Numbering
5 National Numbering
6 Private
7 Reserved

By default, Called Party NPI for BRI is ISDN Numbering.
Call Budget
Refer the topic “Call Budget on Trunk” for command strings.
Call Back
Use the following commands to program Call Back on BRI Trunk ports. To know more about this feature, refer the
To enable/disable Call Back on BRI port:

6242-1-BRI- Call Back Flag
6242-2-BRI-BRI-Call Back Flag
6242- * - Call Back Flag
Where,
0 Disable
1 Enable
To program Call Back Timer for BRI port:

6243-1-BRI-Call Back Timer
6243-2-BRI-BRI-Call Back Timer
Where,
To program Call Back Mode on BRI port:

6244-1-BRI-Call Back Mode
6244-2-BRI-BRI-Call Back Mode
Where,
Call Back Mode is 1 to 5
1 Built-in Auto Attendant
5 Operator
By default, Call Back mode is Operator.

To program Call Back On method for BRI port:
6245-1-BRI-Call Back on selection
6245-2-BRI-BRI-Call Back on selection
Where,
1 CLI Number
2 Alternate Number
By default, Call Back on selection is CLI Number
To assign Call Back - Incoming Number List to a BRI port:

6246-1-BRI-Incoming Number List
6246-2-BRI-BRI-Incoming Number List
Where,
Incoming Number List is from 01 - 16.
By default, Incoming Number List is 15.
To assign a Call Back - Outgoing Number List to a BRI port:

6247-1-BRI-Outgoing Number List
6247-2-BRI-BRI-Outgoing Number List
Where,
Outgoing Number List is from 01 - 16.
To define Call Back From for a BRI port:

6248-1-BRI-Call Back From
6248-2-BRI-BRI-Call Back From
Where,
Call Back From is
1 for Same Port
2 for OGTB Group
To assign a Call Back - OGTB Group for a BRI port:

6249-1-BRI-OGTB Group
6249-2-BRI-BRIOGTB Group
6249-*-OGTB Group
Where,

Pause Timer

6209-1-BRI- Pause Timer
6209-2-BRI-BRI-Pause Timer
6209- * - Pause Timer
Where,
This Timer is used to provide delay for the number dialing by the BRI port. Pause timer will be applicable when any
'P' digit is configured in the DTMF number string which is to be outdialed as DTMF digits on BRI port. One of the
applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage Dialing”.
For example, if PPP2 is to be outdialed and Pause timer is programmed as 3 seconds, the SARVAM UCS will out
dial the digit 2 after 9 seconds i.e delay of individual P i.e 3+3+3 =9.
DTMF ON Time (msec)

6210-1-BRI-DTMF ON Time
6210-2-BRI-BRI-DTMF ON Time
6210-*-DTMF ON Time
Where,
This parameter decides for how much time the DTMF digit will be ON, while out dialed by the SARVAM UCS. One
DTMF Inter digit Pause Timer

6211-1-BRI- DTMF Inter digit Pause Time
6211-2-BRI-BRI- DTMF Inter digit Pause Time
Where,
This parameter decides how much time the pause (gap) should be present between two digits while dialed by the
SARVAM UCS. One of the applications for using this parameter is Multi-stage dialing. Refer chapter “Multi-Stage
Dialing”.
Category (Logical Partition)
Use the following command to program 'Category (Logical Partition)' for BRI port:
6213-1-BRI- Category
6213-2-BRI-BRI- Category
6213-*-Category
Where,

1 for CO
2 for Leased
3 for Private.
By default, Category is CO.
Gateway Application-Answer Signaling
This command is used to enable Gateway Application-Answer Signaling on BRI trunk. Refer chapter “Gateway
Application-Answer Signaling” for more details.
Use following command to set flag for Gateway Application-Answer Signaling on BRI trunk:
6206-1-BRI-Gateway Application-Answer Signaling Flag
6206-2-BRI-BRI-Gateway Application-Answer Signaling Flag
6206-*-Gateway Application-Answer Signaling Flag
Where,
Flag Meaning
0 Disable
1 Enable
By default, Gateway Application-Answer Signaling Flag is 'disable'.
6212-1-BRI-Gateway Application-Answer Signaling DTMF String
6212-2-BRI-BRI-Gateway Application-Answer Signaling DTMF String
6212-*-Gateway Application-Answer Signaling DTMF String
Where,
command using #*.
A #4
B #5
C #6
D #7
* **
# ##
By default, Gateway Application-Answer Signaling DTMF String is 'CCC'.
Debug Code
Debug information of various parts of the card can be obtained on the COM port of the card.

Use following command to get appropriate debug information for the BRI port:
6291-1-BRI-Level-Code
6291-2-BRI-BRI-Level-Code
6291-*-Level-Code
Where,
Code is the value for the specified level to turn ON the debug for the parameters Code range = 000 to 255. Code
value '000' for each level will turn off that level's debug.
Level and Code for BRI port are as specified below:
Level 1
Code Meaning
001 Primitives
002 State
004 Variables
008 SVC Primitives
016 LAP
032 NLS
064 Flow Debug
000 Debug Off
Level 2
Code Meaning
004 Layer 4
016 DTMF Digit
000 Debug Off
Level 3
Code Meaning
016 HDLC D-Channel
000 Debug Off

Level 4
Code Meaning
002 OS Task
000 Debug Off
By Default Debug = off for all BRI ports for all levels.
Viewing BRI Port Status

You can view the status of BRI ports on Jeeves only. To do this,
• Under BRI Configuration, click Status.
• For each BRI Port, the following settings will be displayed:

• BRI Port Number
• Port Name
• Layer 1
• Layer 2
If you have assigned a DSS Key to the BRI Port/channel and when Layer 1 of BRI line is down or not
connected, DSS key LED will glow 1 sec On and 3 sec Off in violet color to indicate the faulty condition.
You can also view the BRI Port Status from the Status link. To view, click the BRI link under Status.

Configuring Mobile Trunks
The SARVAM UCS supports a maximum of 40 Mobile ports. Before you begin configuring the Mobile ports, ensure
that the GSM/3G/CDMA Card has been installed correctly.
Configuration of Mobile Trunks involves:
1. Customizing the Mobile Port Parameters

2. Network Selection
3. Viewing Mobile Port Status
You can customize mobile port parameters and select the network using Jeeves and a Telephone. However, mobile
port status can be viewed using Jeeves only.
Mobile Port Parameters
Configuring Mobile Port Parameters using Jeeves

• Under Configuration, click Mobile Configuration.
• Click Mobile Port Parameters to open the page.
Configure the following port parameters:
• Hardware Slot and Port: 'Slot' is the number of the Universal Slot in which the Mobile Card has been
inserted. 'Port' is the number of the Mobile port in which you have installed the SIM card.

The number of the Mobile Port depends on the configuration of the Mobile Card. For instance, if you
have installed GSM8 Card, the number of the ports will be from 1 to 8.
By default the SARVAM UCS can detect and assign the hardware slot and port numbers automatically
to the mobile (software) ports. However, if required, you may change the Hardware Slot and Port
assigned to the mobile software port. In this case, enter the desired Hardware Slot and Port number in
this field.
If you want to de-assign the Hardware Slot and Port, enter '00' in both fields.
• Enable Port: This flag is for enabling or disabling a Mobile Trunk port. When a Mobile Trunk port is
disabled, neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by clicking the check box.
• Name: You may assign a 'Name' to each Mobile Trunk to facilitate identification. Whenever there is an
extension.
The Name of the port may be the name of the Service Provider of the SIM card you have installed on
this port (recommended) or the phone number assigned to the SIM card on this port.
The Name may comprise a maximum of 18 characters.
• Band Selection163 (Freq in MHz) (GSM): The Frequency Band supported by the mobile networks
varies from country to country. SARVAM UCS's mobile card supports frequency bands of most
countries. Select the Frequency Band used by your GSM/ UMTS Provider for the mobile port.
By default, Mobile Frequency Band 'All Bands’ is selected.
This is applicable only for GSM Mode.
When you change the Frequency Band, the change will be effected after the next system restart or the
next Mobile Port restart.
• Preferred Network Mode164: Select the Preferred Network Mode for the Mobile Port as Dual, GSM or
UMTS. By default, Dual Mode is selected.
• SIM PIN165: If you have enabled PIN protection and changed the SIM PIN of the card to the default
value '1234' using a mobile handset166, you can assign a new PIN to the SIM card from the SARVAM
UCS.
Make sure that the PIN stored on the SIM card and that of the system are the same.
163. Not applicable if CDMA Mobile Card is installed in your system.

166. Refer the topic Installing the Mobile Card for instructions. If you have not enabled PIN protection before installing the GSM Card,
you will not be able to change the SIM PIN.

• You must click ‘Submit’ after you enter the new PIN. Wait for 5 seconds, and then refresh this page to
view the new SIM PIN.
• If you have enabled PIN protection, and the SIM PIN on the Card and the SIM PIN programmed in the
SARVAM UCS are not the same, the SIM card may get blocked and would require the Personal
Unblocking Number (PUK) from the Service Provider to reactivate it again.
The SIM PIN will not be set to default value, when you set the system to default.
• Incoming Calls: Select the Incoming Call Mode on the Mobile Port from the options:
• Accept': incoming calls will be allowed and incoming call logic is applied.
• ‘Ignore': incoming calls will not be processed further and call logic will not be applied.
• ‘Reject': incoming calls will be rejected immediately and mobile port will be freed (released).
By default, incoming calls are accepted. This feature is particularly useful in outbound call centers for
blocking incoming calls on a SIM number (mobile port) of the SARVAM UCS.
• Trunk Feature Template: A Trunk Feature Template is a set of features like Time Table, Operator,
Auto Attendant, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR Storage, etc., that defines the
behavior of a Trunk. Apply a Trunk Feature Template to the Mobile Trunk. By default, Trunk Feature
Template 01 is applied on all Mobile Trunks as well as all other trunk types. Refer the topic “Trunk
Feature Template” to know more.
Click the 'Trunk Feature Template’ link to open the page. Check if the default Template 01 fulfills your
requirement for the mobile port.
If the default Template 01 does not fulfill your requirement, prepare another Trunk Feature
Template167, and enter the newly prepared Template number for the Mobile port.
• CLIR168: Enable this flag if you want to activate CLIR for all outgoing calls made through the Mobile
Port.
When CLIR is enabled, the called party will not be able to see the subscriber number of the Mobile
Port.
By default CLIR is disabled for all Mobile ports.
This feature will work only if subscribed/supported by your mobile service provider.
• Return Call to Original Caller (RCOC)169: Enable this flag if you want to apply the RCOC feature.
If this feature is enabled on the Mobile trunk port, the system routes calls returned by remote parties
back to the extensions that originally made the call from this port (the original callers' extensions). To
know more, refer the feature description for Return Call to Original Caller (RCOC).
167. The default template is applied on the ports of all trunk types supported by SARVAM UCS. Changes to the default template will be
applied on all trunk types also. So, you are advised to prepare a new template and apply it to the desired trunk types.

• Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the
mobile port.
Assign a Cost Factor to the Mobile Trunk port, for example, 03 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '9' through the SIM installed in
Mobile Port Number 01 only,
• You must first assign a Cost Factor (01-99) to Mobile Port 01, for example, 03.
• Click 'Least Cost Routing - Number Based' to open the page.
All outgoing calls assigned Cost Factor trunk 03 will be made from Mobile Port 01.
The above listed parameters fulfill the basic mobile trunk port configuration requirements of most users. However, it
is anticipated that some users may need to configure other less commonly used features on the mobile ports, such
as Call Budget, Call Back on Mobile Port, or they may want to use the Mobile port as a Gateway application.
For such users, you may click the 'Advance' button and program the following parameters:

• N/w Registration Retry Count: The mobile port is programmed to automatically locate and register with
the Network that supports the SIM card installed on it. Also, at each power ON, the mobile port (SIM) will
automatically register with the Network that supports the SIM on it.
However, if the Mobile port fails to register, it will restart the process of network registration on the expiry of
the Network Registration Retry Timer170. On the expiry of this timer, the system will retry registration for
the programmed Count (number of times) and with each re-try attempt, the count will be decremented by
one.
By default the Retry Count is set to 2. If required you may change the Count to the desired value.
• Mobile Gain Settings Template: You can increase or decrease the level of Incoming Speech (Receive
Gain) and Outgoing Speech (Transmit Gain) on the Mobile port by changing the Rx Gain and Tx Gain to
the desired level. Different levels can be set for each port type in the Mobile Gain Settings Template. By
default, Mobile Gain Template 1 is assigned. If you want to assign a different Template, you must
customize the Mobile Gain Settings Template first and then assign the number of the Mobile Gain Settings
Template here. To customize the Mobile Gain Settings, see “Gain Settings”.
If you change the Tx or Rx Gain during an active call, the change you made will not apply on the current
call. It will be applied on the next call.
Back on Trunk Port' feature on this Mobile trunk, configure the following parameters:
seconds.
should be routed:
• Operator
Number’.
170. The Network Registration Retry Timer defines the time for which the Mobile port, which has failed to register with the network,
should wait before attempting to re-register with the network. Network registration retry timer is 2 minutes and is non-programma-
ble.

Number List 15 is also assigned to all Mobile trunks as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all Mobile trunks and Trunk Port types, retain
this list.
If you want a different set of number strings to be programmed for this Mobile Trunk, select a different
Number List, and assign it to the Mobile trunk port.
assigned to all Mobile trunks as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this Mobile Trunk
port.
By default, OGTBG 01 is assigned.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost Routing
on the OGTBG that you define here for Call Back.
• Call Budget: If you want to enable 'Call Budget on Trunk' feature, configure the following parameters for
this mobile trunk port:
• Type: Select the type of Call Budget on Trunk—Amount, Minutes or Number of Calls—to be applied on
this mobile trunk port. By default, no Call Budget type is selected.

• Calls: If you selected ‘Calls’ as the Call Budget Type, enter the number of calls in this field. By default,
on a particular date of every month.
• Scheduled (Date): Enter the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be every month. You may select ‘Daily’ if your plan suggests so.
The consumed Call Budget Amount/Minutes/Number of Calls can be reset from SE and SA Mode, referred
to as Manual Reset. Refer the feature description “Call Budget on Trunk”.
• Accept Anonymous Calls: The flag is for accepting calls without CLI that land on the mobile port. By
default the flag is enabled. You may disable this flag to disallow calls without CLI on this Mobile port.
• Pause Timer: This Timer is used for providing delay in number dialing from the Mobile port. The Pause
Timer will be applicable when the digit 'P' is configured in the DTMF number string which is to be out dialed
as DTMF digits on the Mobile port.
For example, if PPP2 is to be out dialed and Pause timer is programmed as 3 seconds, the SARVAM UCS
will out dial the digit 2 after 9 seconds, that is, after a delay of individual P (3+3+3 =9). The range of this
time is from 1 to 9. By default the Timer is set to 1 seconds.
This parameter is used for the “Multi-Stage Dialing” feature.
• DTMF Outdial Option: You can select whether to send the DTMF digits from the Mobile Ports Inband or
through signaling, that is, AT Command. By default, DTMF Outdial Option is Inband.
When you select DTMF Outdial using AT Command, the length of the DTMF digits will be determined by
the DTMF ON Time you set.
If CDMA Mobile Card is installed in your system, select DTMF Outdial option as AT Command for
efficient functionality.
• DTMF ON Timer: This parameter determines the time for which the DTMF digit will remain ON, while
being out dialed by the system. This parameter finds its application in the feature “Multi-Stage Dialing”' and
in DTMF Outdialing using AT Command. By default, DTMF ON Timer is 100 ms.
• DTMF Detection Mode: You can select whether to detect the DTMF digits using the GSM Modules or
through DSP. Default: Using DSP.
When you select DTMF Detection Using Module, the length of the DTMF digits will be determined by the
DTMF Detection Timer you set.
If CDMA Mobile Card is installed in your system, select DTMF Detection Mode as Using Module (GSM)
for efficient functionality.

• DTMF Detection Duration: This is the minimum ON to consider the DTMF digit as a valid digit. The valid
range of this timer is 20 to 100ms. Default: 30ms.
If CDMA Mobile Card is installed in your system, set the DTMF Detection Duration as 20ms.
• Min. Level (dB): This parameter signifies the minimum level (dB) of the DTMF digit to be considered as
valid. By default, Minimum level is set to -30dB.
If the Min. Level set is very low, the DTMF digits might be detected in Voice and if it is very high, the DTMF
digits may be lost.
If received DTMF digit level is higher than or equal to the set value of Min. Level (dB), the system will
accept the DTMF.
If received DTMF digit level is lower than the set value of Min. Level (dB), the system will ignore the DTMF.
• Category (Logical Partition): This parameter assigns the Mobile Port to a trunk category for the purpose
of Logical Partitioning. By default all Mobile Ports are assigned to Category 1. Do not change the default
setting.
If you want to change the call permission between the mobile port and other trunks, click the 'Category' link
to open the Logical Partitioning page. You may program the call permission between Category 1 (assigned
to Mobile Trunk Ports) and other Categories. Refer the feature description “Logical Partition” to know
more.
• Gateway Application-Answer Signaling: This parameter is to be programmed if the Mobile Trunk Port is
being used in a gateway application as a source port (from where calls originate). The calls originated on
the source port (mobile port) are routed using another Trunk port, the terminating port, which may be any
trunk port, for example: T1E1. When call made from the terminating port gets matured, this is signaled to
the source port in the form of DTMF digits.
• Enable: Enable this flag if you want the Mobile port to be used in a Gateway Application.
• DTMF String (max. 4 digits): Program the DTMF digits to be sent to signal call maturity to the source
port.
• SMS Parameters171: If you want to use this Mobile Trunk Port for the SMS Server application to send/
receive SMS, configure the following parameters:
• SMS Center Number: The system displays the default SMS Center Number of the network. If
required, you can change the SMS Center Number. The SMS Center Number can be a maximum of 16
digits. Valid Range: 0 to 9 and +
• Send SMS: Enable this check box if you want to use this Mobile Trunk Port to send SMS.
• Number of SMS to be sent (Total): Enter the maximum number of SMS that can be sent using this
Mobile Trunk Port. Default: 0999999999. Valid Range: 1 to 4294967296.
• Number of SMS to be sent (Daily): Enter the maximum number of SMS that can be sent using this
Mobile Trunk Port daily. Default: 20000. Valid Range: 1 to 20000.

• Receive SMS: Enable this check box if you want to use this Mobile Trunk Port to receive SMS.
• Debug: Enable this flag by selecting the check box if you want to initiate debugging for the Mobile Port. By
default, debugging is disabled.
• Repeat the same steps to configure other mobile ports.
• If you have completed configuration of all the above listed Mobile Port Parameters, click Submit at the
Configuring Mobile Port Parameters using a Telephone

To assign Hardware Slot and Port, dial:

• 1108-Mobile-Slot-Port Offset on the card
Where,
Mobile Port Number is the number of the software port from 001 to 064
Slot is the number of the universal slot from 01 to 16
Port Offset on the card is 00, 01 to 99.
To de-assign the Hardware Slot and Port, dial:

• 1108-Mobile-00-00
To enable/disable the Mobile Port, dial:

• 8000-1-Mobile-Flag to enable/disable a single mobile port.
• 8000-2-Mobile-Mobile-Flag to enable/disable a range of mobile ports.
• 8000-*-Flag to enable/disable all mobile port.
Where,
Flag is
1 for Enable
0 for Disable
To assign a Name to the Mobile Port, dial:

• 5408-1-Mobile-Name-#* to assign a name to a single mobile port.
• 5408-2-Mobile-Mobile-Name-#* to assign the same name to a range of mobile ports.
• 5408-*-Name-#* to assign the same name to all mobile ports.
Where,
Name is a string of a maximum of 18 alphanumeric characters. Terminate the command with #* if the
number string is fewer than 18 characters.
To clear the Name of the Mobile Port, dial:

• 5408-1-Mobile-#* to clear the name of a single mobile port.
• 5408-2-Mobile-Mobile-#* to clear the names of a range of mobile ports.
• 5408-*-#* to clear the names of all mobile ports.
To select the frequency Band172 for the Mobile Port, dial:

• 8009-1-Mobile Port Number-Mobile Frequency Band Code to select the band for a single mobile
port.

• 8009-2-Mobile Port Number -Mobile Port Number -Mobile Frequency Band Code to select the
same band for a range of mobile ports.
• 8009-*-Mobile Frequency Band Code to select the same band for all mobile ports.
Where,
Mobile Port is the software port number of the mobile port, from 001 to 064.
Mobile Frequency Band Code is
1 for 900
2 for 1800
3 for1900
4 for 850+1900
5 for 900+1800
Default: 900 + 1800
To select the Preferred Network Mode173 for the Mobile Port, dial:
• 8038-1-Mobile Port Number-Preferred Network Mode to select the band for a single mobile port.
• 8038-2-Mobile Port Number -Mobile Port Number -Preferred Network Mode to select the same
band for a range of mobile ports.
• 8038-*-Preferred Network Mode to select the same band for all mobile ports.
Where,
Mobile Port is the software port number of the mobile port, from 001 to 064.
Preferred Network Mode is:
1 for Dual
2 for GSM
3 for UMTS
To assign SIM PIN174 to the Mobile Port, dial:

• 8006-1-Mobile Port Number-SIM PIN-#* to assign SIM PIN to a single mobile port.
Where,
Mobile Port Number is the software port number of the mobile port, from 001 to 064.
SIM PIN is the new SIM PIN, any combination of 4 to 8 digits. Terminate the command with '#*' if SIM
PIN is fewer than 8 digits.
To select Incoming Call mode on the Mobile port, dial:

• 8005-1-Mobile-Mode to select the mode for a single mobile port.
• 8005-2-Mobile-Mobile-Mode to select the same mode for a range of mobile ports.
• 8005-*-Mode to select the same mode for all mobile ports.
Where,
Mode is
1 for Accept Incoming Calls.
2 for Ignore Incoming Calls
3 for Reject Incoming Calls
Default: Accept
To assign a Trunk Feature Template to the Mobile Port, dial:

• 5807-1-Mobile Port Number-Template Number to assign a template to a single mobile port.
• 5807-2-Mobile Port Number-Mobile Port Number-Template Number to assign the same template to
a range of mobile ports.
• 5807-*-Template Number to assign the same template to all mobile ports.
Where,


Template Number is the number of the Trunk Feature Template from 01 to 50.
To enable/disable CLIR175 on the Mobile Port, dial:

• 8031-1-Mobile Port Number-CLIR to enable/disable CLIR on a single mobile port.
• 8031-2-Mobile Port Number-Mobile Port Number-CLIR to enable/disable CLIR on a range of mobile
ports.
• 8031-*- CLIR to enable/disable CLIR on all mobile ports.
Where,
CLIR is
0 for Disable
1 for Enable
Default: Disable
To enable RCOC176 on Mobile Port, dial:

• 8030-1-Mobile -Code to enable the feature on a single mobile port.
• 8030-2-Mobile-Mobile-Code to enable the feature on a range of mobile ports.
• 8030-*-Code to enable the feature on all mobile ports.
Where,
Code is
0 for Disable
1 for Enable
Default: Disabled
To assign Cost Factor to a Mobile Port, dial:

• 8001-1-Mobile-Cost Factor to assign cost factor to a single mobile port.
• 8001-2-Mobile-Mobile-Cost Factor to assign the same cost factor to a range of mobile ports.
• 8001-*-Cost Factor to assign the same cost factor to all mobile ports.
Where,
Cost Factor is from 01 to 99.
• For Advanced Configuration of the Mobile Ports, use the following commands:
To set Network Registration Retry Count, dial:

• 8004-1-Mobile-N/w Registration Retry Count to set the count for a single mobile port.
• 8004-2-Mobile-Mobile-N/w Registration Retry Count to set the same count for a range of mobile
ports.
• 8004-*-N/w Registration Retry Count to set the same count for all mobile ports.
Where,
N/w Registration Retry Count is from 001 - 255: Default: 002
• For commands to program Call Budget on Mobile Ports, refer the topic “Call Budget on Trunk”.
To enable/disable Call Back on Mobile Port, dial:

• 8010-1-Mobile-Code to enable/disable Call Back on a single mobile port.
• 8010-2-Mobile-Mobile-Code to enable/disable Call Back on a range of mobile ports.
• 8010-*-Code to enable/disable Call Back on all mobile ports.


Where,
Code is
0 for Disable
1 for Enable
Default: Disabled
To change Call Back Timer for the Mobile Port, dial:

• 8011-1-Mobile-Call Back Timer to change Call Back Timer for a single mobile port.
• 8011-2-Mobile-Mobile-Call Back Timer to set the same Call Back Timer duration for a range of mobile
ports.
• 8011-*-Call Back Timer to set the same Call Back Timer duration for all mobile ports.
Where,
Call Back Timer is from 01 to 99 seconds.
To select the Call Back Mode, dial:

• 8012-1-Mobile-Call Back Mode to select Call Back Mode for a single mobile port.
• 8012-2-Mobile-Mobile Call Back Mode to select the same Call Back Mode for a range of mobile ports.
• 8012-*-Call Back Mode to select the same Call Back Mode for all mobile ports.
Where,
Call Back Mode is
1 for Built-In Auto Attendant
2 for PIN Authentication - Multiple Calls
3 for CLI Authentication - Multiple Calls
4 for CLI Authentication - Single Call - Answer Signaling
5 for Operator
Default: Operator
To program Call Back on for the Mobile Port, dial:

• 8013-1-Mobile-Call Back on to program call back on for a single mobile port.
• 8013-2-Mobile-Mobile-Call Back on to program the same call back on for a range of mobile ports.
• 8013-*-Number List to program the same call back on for all mobile ports.
Where,
Call Back on is
1 for CLI Number
2 for Alternate Number.
To assign a Call Back - Incoming Number List to a Mobile Port, dial:

• 8034-1-Mobile-Incoming Number List to assign a list to a single mobile port.
• 8034-2-Mobile-Mobile-Incoming Number List to assign the same list to a range of mobile port.
• 8034-*-Incoming Number List to assign the same list to all mobile ports.
Where,
Incoming Number List is from 01 to 16
Default: 15
To assign a Call Back - Outgoing Number List to a Mobile Port, dial:

• 8035-1-Mobile-Outgoing Number List to assign a list to a single mobile port.
• 8035-2-Mobile-Mobile-Outgoing Number List to assign the same list to a range of mobile port.

• 8035-*-Outgoing Number List to assign the same list to all mobile ports.
Where,
Outgoing Number List is from 01 to 16
Default: 16
To select Call Back From for a Mobile Port, dial:

• 8036-1-Mobile-Call Back From to select call back from for a single mobile port.
• 8036-2-Mobile-Mobile-Call Back From to select the same call back from option for a range of mobile
ports.
• 8036-* -Call Back From to select the same call back from option for all mobile ports.
Where,
Call Back From is
1 for Same Port
2 for OGTB Group
Default: Same Port
To assign a Call Back - OGTB Group for a Mobile Port, dial:

• 8037-1-Mobile-OGTB Group to assign an OGTBG to a single mobile port.
• 8037-2-Mobile-Mobile-OGTB Group to assign the same OGTBG to a range of mobile ports.
• 8037-*-OGTB Group to assign the same OGTBG to all mobile ports.
Where,
Default: 01
To enable/disable Accept Anonymous Calls on Mobile Port, dial:

• 8029-1-Mobile-Code to enable/disable Anonymous Calls on Mobile Port on a single mobile port.
• 8029-2-Mobile-Mobile-Code to enable/disable Anonymous Calls on Mobile Port on a range of mobile
ports.
• 8029-*-Code to enable/disable Anonymous Calls on Mobile Port on all mobile ports.
Where,
Scheduled Reset Date is from 01 to 31.
To program the Pause Timer on Mobile Port, dial:

• 8014-1-Mobile-Pause Timer to set the timer on a single mobile port.
• 8014-2-Mobile-Mobile-Pause Timer to set the same timer on a range of mobile ports.
• 8014-*-Pause Timer to set the same timer on all mobile ports.
Where,
Pause Timer is from 1 to 9 seconds. Default: 1 second.
To program DTMF ON Time on Mobile Port, dial:

• 8015-1-Mobile-DTMF ON Time to set the time on a single mobile port.
• 8015-2-Mobile-Mobile-DTMF ON Time to set the same time on a range of mobile ports.
• 8015-*-DTMF ON Time to set the same time on all mobile ports.
Where,
DTMF ON Time is
1 for 100 msec
2 for 200 msec

Default: 100 msec
To set DTMF Detection Mode on Mobile Port, dial:

• 8051-1-Mobile-Mode to set the DTMF Detection Mode on a single mobile port.
• 8051-2-Mobile-Mode to set DTMF Detection Mode on a range of mobile ports.
• 8051-*-Mobile-Mode to set the same DTMF Detection Mode on all mobile ports.
Where,
DTMF Detection Mode is
1 for Using DSP
2 for Using Module
Default: Using DSP
To program DTMF Detection Duration on Mobile Port, dial:

• 8052-1-Mobile-Duration to set the DTMF Detection Duration on a single mobile port.
• 8052-2-Mobile-Duration to set the DTMF Detection Duration on a range of mobile ports.
• 8052-*-Mobile-Duration to set the DTMF Detection Duration on all mobile ports.
Where,
DTMF Detection Duration range is from 20 to 100 msec.
Default: 30 msec
To assign the Mobile Port to a 'Category' for Logical Partition, dial:

• 8018-1-Mobile-Category to assign a single mobile port to a category.
• 8018-2-Mobile-Mobile-Category to assign a range of mobile ports to the same category.
• 8018-*-Category to assign all mobile ports to the same category.
Where,
1 for Trunk ports interfaced with PSTN /PLMN (Public Land Mobile Network)
2 for Leased lines terminated in the trunk ports.
3 Trunk ports used to interconnect two Systems.
Default: 1
To enable/disable Gateway Application on the Mobile Port, dial:

• 8016-1-Mobile-Gateway Application flag to enable/disable on a single mobile port.
• 8016-2-Mobile-Mobile-Gateway Application flag to enable/disable on a range of mobile ports.
• 8016-*-Gateway Application flag to enable/disable on all mobile ports.
Where,
Gateway Application flag is.
0 for Disable
1 for Enable
Default: Disable
To program the DTMF String for the Gateway Application on the Mobile Port, dial:
• 8017-1-Mobile-DTMF String to program the string on a single mobile port.
• 8017-2-Mobile-Mobile-DTMF String to program the same string on a range of mobile ports.
• 8017-*-DTMF String to program the same string on all mobile ports.
Where,
DTMF String is a maximum of 4 digits. Default: CCC

To enable/disable Debug on a Mobile Port, dial:
• 8028-1-Mobile port number-Debug Code to enable/disable debug on a single mobile port.
• 8028-2-Mobile port number-Mobile port number-Debug Code to enable/disable debug on a range
of mobile ports.
• 8028-*-Debug Code to enable/disable debug all mobile ports.
Where,
Debug Code is
1 for Enable
0 for Disable
Default: Disable
To program SIP Rx Gain on the Mobile Port, dial:

• 8041-1-Mobile-SIP Rx Gain to program the SIP Rx Gain on a single mobile port.
• 8041-2-Mobile-Mobile-SIP Rx Gain to program the same SIP Rx Gain on a range of mobile ports.
• 8041-*-SIP Rx Gain to program the same SIP Rx Gain on all mobile ports.
Where,
Rx and Tx Gain is 01 to 63 dB: Default: 4.
For Rx and Tx Gain at SIP codes, see the tables below.
To program SIP Tx Gain on the Mobile Port, dial:

• 8042-1-Mobile-SIP Tx Gain to program the SIP Tx Gain on a single mobile port.
• 8042-2-Mobile-Mobile-SIP Tx Gain to program the same SIP Tx Gain on a range of mobile ports.
• 8042-*-SIP Tx Gain to program the same SIP Tx Gain on all mobile ports.
Where,
Rx and Tx Gain is 01 to 63 dB: Default: 4.
For Rx and Tx Gain at SIP codes, see the tables below.
• Exit SE mode.
Network Selection
After the Mobile Card is successfully installed and powered on, the mobile port is programmed to automatically
locate and register with the Network that supports the SIM card installed in. Also, at each power ON, the mobile

port (SIM) will automatically register with the Network that supports the SIM on it. However, if the Mobile port fails to
register, it will restart the process of network selection on the expiry of the Network Registration Retry Timer177.
If the SARVAM UCS is located in a border area where more than one Network Operator is available, it is possible
that the SIM card may register with another available network and result in 'Roaming' charges. To avoid this, you
must disable automatic network selection and program manual network selection.
When you enable manual network selection, you must program the Network Operator Priority Table. This table
requires you to program the Network Operator Codes (MCC-MNC)178 in order of priority for a Mobile Port. So,
whenever you register with the network manually, select the Network Operator that matches in order of priority. If
the Mobile port fails to register, it will restart the process of network selection on the expiry of the Network
Registration Retry Timer.
If no match is found, the Mobile port (SIM) will not get registered with any of the available network operators and no
calls can be made or received on this port.
Configuring Network Selection using Jeeves

While still logged in as System Engineer,
• Under Mobile Configuration, click Network Selection.
• Set the Network Selection Mode to Manual.
If CDMA Mobile Card is installed in your system, keep the Network Selection Mode as Automatic. Manual
mode is not applicable in this case.
177. The Network Registration Retry Timer defines the time for which the Mobile port, which has failed to register with the network,
should wait before attempting to re-register with the network. Network registration retry timer is 2 minutes and is non-programma-
ble.
178. The Network Operator Code comprises of the Mobile Country Code (MCC) appended by the Mobile Network Code (MNC). The
MCC is usually a 3-digit code that identifies a country. A single country may be assigned more than one MCC. For example the
MCC assigned to India is 404, but same code applies to all network operators in the country.
The MNC is usually a 2/3-digit code. The MCC-MNC combination uniquely identifies the home network of the mobile terminal or
the mobile user. For example, AirTel, a GSM network operator in India, has different MNC assigned to its networks in various
states. The MNC for AirTel in the state of Maharashtra is 90, while the same for the state of Gujarat is 98.

• Enter the Network Operator Codes (MCC-MNC) in order of priority. The codes must not exceed 8 digits.
You can store up to 9 Network Operator Codes in the order of priority.
• Repeat the same steps to set network selection mode for other mobile ports.
• When you change the Network Selection Mode to ‘Manual’ and the Network Operator Code manually,
the change you made will not come into effect until you have restarted the Mobile Port.
• To restart the Mobile Port,

• go back to Mobile Port Parameters.
• Click the check box to disable the Enable Port flag.
• Wait for the page to refresh.
• Click the check box to enable the Enable Port flag.
• The Mobile Port will be restarted.
Configuring Network Selection using a Telephone

To select Network Selection mode, dial:

• 8007-1-Mobile Port Number-Code to select network selection mode for a single mobile port.
• 8007-2-Mobile Port Number-Mobile Port Number-Code to select the same network selection mode
for a range of mobile ports.
• 8007-*-Code to select the same network selection mode for all mobile ports.
Where,
Code is
1 for Automatic Selection
2 for Manual Selection
Default: Automatic Selection
For example, to program Manual Network Selection on Mobile Port number 1, dial 8007-1-01-2
To program the network operator codes in order of priority for Mobile port, dial:
• 8008-1-Mobile-Priority-Network Operator Code-#* to program network operator codes for a single
mobile port.
• 8008-2-Mobile-Mobile-Priority-Network Operator Code-#* to program the same network operator
codes for a range of mobile ports.
• 8008-*-Priority-Network Operator Code-#* to program the same network operator codes of all mobile
ports.
Where,
Network Operator Code is MCC-NCC maximum 8 digits. Terminate the command with #* if the number
of digits is fewer than 8.
For example, following Network Operator Codes need to be programmed in the order of priority on
Mobile Port number 01:
Priority 1=40498
Priority 2 = 40425

Priority 3 = 40421
Dial:
8008-1-01-1-40498-#*
8008-1-01-2-40425-#*
8008-1-01-3-40421-#*
• Exit SE mode.
Viewing Mobile Port Status

You can view the status of Mobile ports on Jeeves only. To do this,
• Under Mobile Configuration, click Status.
• Wait for 4-5 seconds after the Port Status page is opened.
• Click Refresh at the bottom of the page for current view.
• The Port status page will display the following parameters for all Mobile ports that have been enabled:
• Port Name: This is the name by which the Mobile Port is programmed.
• Port Status: This is the status of the connection - showing Initialization with the Network, Registering
with the Network, Idle or Busy state of the network. It also shows errors and alerts when SIM is absent,
the wrong SIM PIN has been entered, SIM PUK is required.
• IMEI: This is the unique identification number of (the GSM engine) each Mobile port.

If CDMA Mobile Card is installed in your system, ESN No.(Electronic Serial Number) will be displayed as
the parameter name instead of IMEI.
In absence of RUIM Card, “00000000” will be displayed as the ESN No.
• Network Operator Code: This is the MCC-MNC code of the network with which the mobile port is
registered.
• Network Operator Name: This is the name of the service provider/network operator with which the
Mobile Port is registered.
• SMSC Number: This is the number of the SMS Center of the network operator. If CDMA Mobile Card
is installed in your system, SMSC Number status will be displayed as blank.
• Signal Strength (dBm): This is the signal strength in '-dBm' as received from the network with which
the Mobile port is registered.
• Bit Error Rate (BER): BER is Bit Error Rate which defines the quality of the channel.
• Ec/Io (dB)179: This is the ratio of the received energy per chip (= code bit) and the interference level,
given in dB. In case no true interference is present, the interference level is equal to the noise level.
This parameter is significant only when the GSM engine is registered with the 3G network.
The range of this parameter can be from 0 to 63dB. A ratio of 10dB to 14dB is normal, numbers going
higher than that is progressively worse.
• Call Duration: This is the total call duration of matured outgoing calls180 on the Mobile port. This data
is used for calculating Answer Seizure Ratio (ASR) for the port. It is displayed in MMMMMM:SS format.
• Dialed Calls: This is the total number of outgoing calls181 made from the Mobile port. This data is used
for calculating Answer Seizure Ratio (ASR) for the port.
• Successful Calls: This is the total number of matured outgoing calls made from the Mobile port. This
data is used for calculating Answer Seizure Ratio (ASR) and Average Call Duration for the port.
• ASR % : This is the Answer Seizure Ratio (ASR) calculated by the system for the Mobile port, in terms
of percentage. ASR is the sum of all outgoing matured calls from the Mobile port, divided by the total
number of outgoing calls made from the Mobile port, multiplied by 100. The system calculates ASR
after the completion of the outgoing call.
• ACD: This is the Average Call Duration (ACD) of outgoing calls made from the Mobile port. It is an
indicator for monitoring the network condition. Decreasing ACD is indicative of trouble in the network
condition.
The system calculates ACD after the completion of the outgoing calls, by dividing the total call duration
by the number of outgoing matured calls.

180. Matured calls are outgoing calls for which 'CONNECT' message was received from the network.
181. The total number of outgoing calls made includes the number of times the ATD has been sent from the Mobile port to the network.

• Reset ASR and ACD: This field allows the System Engineer to manually the ASR and the ACD of the
Mobile port.
The parameters Total Call duration, Number of matured calls, Total Number of OG Calls, ASR and
ACD are saved in the configuration, and are not on Power OFF condition. The system maintains the
statistics for the last 999 calls. When the total number of outgoing calls exceeds 999, the system will
stop calculating ACD and ASR and will display ASR and ACD calculated on the basis of the last 999
calls only.
Therefore, the System Engineer must manually ASR and ACD when the total number of calls reaches
999. When you ASR and ACD the number of call matured and the number of calls dialled is to 0.
ASR and ACD can be anytime, even when the total number of calls is less than 999.
When ACD is, only the 'Total Call Duration' maintained for the ACD calculation will be. The 'Total Call
Duration' of the Call Budget, the consumed minutes maintained for the Call Budget on the mobile port will
remain unaffected.
• SIM ID182: This is the Integrated Circuit Card ID (ICC-ID) of the SIM card inserted in the Mobile port.
Each SIM is internationally identified by its ICC-ID. ICC-IDs are stored in the SIM card and are also
printed on the SIM card body.
• IMSI183: International Mobile Subscriber Identity (IMSI) is a unique number stored in the SIM card.
• Cell ID: This is the 16-bit identifier that identifies the cell. The cell is the radio coverage area given by
one BTS (Base Transceiver Station).
If CDMA Mobile Card is installed in your system, NID (Network Identification) will be displayed as the
parameter name instead of Cell ID.
• Location Area Code (LAC): The LA (Location Area) is a group of cells defined by the Operator. The
LAC (Location Area Code) uniquely identifies a LA within a PLMN (Public Land Mobile Network).
If CDMA Mobile Card is installed in your system, SID (System Identification) will be displayed as the
parameter name instead of LAC.
• Call Budget Type: This shows the Call Budget Type, whether Amount, Minutes or Number of Calls,
are set on the Mobile port.
• Allotted Amount/ Minutes/Calls: This shows the sum/number of minutes/number of calls allotted as
Call Budget on the Mobile port.
• Consumed Amount/ Minutes/Calls: This shows the sum/number of minutes/number of calls of the
allotted Call Budget that has been used up on the Mobile port.
• Scheduled Reset: This shows if you have enabled Scheduled Reset.
• Call Budget Reset Schedule (Date): This shows whether the consumed Call Budget on the Mobile
port is to be Daily or on a particular date of a month.


• Reset Consumed Amount/Minutes/Calls: This editable field allows the you to reset the consumed
Call Budget Amount/Minutes/Calls at any time, manually.
• SMS Budget Scheduled Reset: This shows if you have enabled Scheduled Reset for the SMS budget.
If CDMA Mobile Card is installed in your system, any parameters related to SMS will not be supported.
• SMS Budget Reset Schedule (Date): This shows the date on which the SMS Budget will be reset.
• SMS Budget Consumed SMS (Total): This shows the number of SMS of the allotted SMS Budget that
has been used up on the Mobile port.
• Consumed SMS (Daily): This shows the number of SMS of the allotted Daily SMS Budget that has
been used up on the Mobile port.
• Reset Consumed SMS (Total) Budget: This editable field allows the you to reset the consumed SMS
Budget, manually. It will reset both Consumed SMS (Total) and Consumed SMS (Daily).
The Consumed SMS Budget can be reset from the System Engineer mode as well as the System
Administrator mode manually at any time or on a scheduled date from the System Administrator mode
only. Refer Resetting SMS Budget.
• Registered with Network: This shows the type of network with which the Mobile port is registered,
whether GSM, GSM Compact, 3G, CDMA or UMTS.
• Firmware Version of Engine: This shows the firmware version of the mobile Engine.
You can also view the Mobile Port Status from the Status link. To view, click the Mobile link under Status.
Resetting SMS Budget184

You manually reset the Consumed SMS Budget from the SE mode from the Status page (as explained above) or
from the SA mode.
To reset the SMS Budget from the SA mode, follow the steps given below:
• Log in as System Administrator.
• Under Trunks Properties, click the Mobile link.
• Consumed SMS (Total): This shows the number of SMS of the allotted SMS Budget that has been
used up on the Mobile port.
• Consumed SMS (Daily): This shows the number of SMS of the allotted Daily SMS Budget that has
been used up on the Mobile port.
• Scheduled reset consumed SMS (Total) budget: Enable this check box if you want only the
Consumed SMS (Total) Budget to be reset on a particular date of every month.
• Budget Reset Schedule (Date): Select the date of the month (Daily or 1-31) on which you want the
SMS Budget to be reset every month.

• Reset Consumed SMS (Total) Budget: Enable this check box to reset the consumed SMS Budget,
manually. It will reset both Consumed SMS (Total) and Consumed SMS (Daily).
Resetting ASR and ACD using a Telephone

To ASR and ACD calculation for Mobile Port, dial:

• 8032-1-Mobile-1 to ASR-ACD calculation for a single mobile port.
• 8032-2-Mobile-Mobile-1 to ASR-ACD calculation for a range of mobile ports.
• 8032-*-1 to ASR-ACD calculation for all mobile ports.
• Exit SE mode.
The System Administrator can also
• view the ID of the Mobile Network Operator with which the Mobile Port is currently registered.
• check Signal Strength of a mobile port, whenever there is trouble placing calls over a Mobile port to rule
out weak signal as the cause.
• Reset ASR and ACD calculation.
To do this,
• Enter SA mode from a DKP.
• Dial 1072-017-Mobile Port Number to view ID of Network Operator.

You will get a confirmation tone.
If the port is successfully registered, the name of the network Operator will be displayed on the LCD of
the DKP.
If the port is not registered, the message 'Error' will be displayed on the LCD.
• Dial 1072-032-Mobile Port Number to check signal strength.
• Wait for the confirmation tone.

When the network responds, you the Mobile Network Signal Strength will be displayed on the LCD of
your DKP and programming beeps will be played.
The Signal Strength values are in -dBm. '-113' indicates weak signal, whereas '-51' indicates maximum
signal strength.
• Dial 1072-022-Mobile Port Number
You will get the confirmation tone and the confirmation message “ASR-ACD Reset” will appear on your
phone’s display.

Configuring CO Trunks
The SARVAM UCS supports a maximum of 64 Analog Two-Wire Trunk Lines. Before you begin configuring the CO
trunk ports, ensure that the CO trunk Card has been installed correctly.
You may configure the CO ports from Jeeves and using a Telephone.
Configuring CO Trunks using Jeeves

• Click CO Parameters.
Configure the following port parameters:
• CO No.: This non-editable field is the number of the software port of the CO Trunk.
• Hardware Slot and Port: 'Slot' is the number of the Universal Slot in which the CO Card has been
inserted. 'Port' is the number of the CO trunk port on that card.
By default the SARVAM UCS can detect and assign the hardware slot and port numbers automatically to
the CO (software) ports. However, if required, you may change the Hardware Slot and Port assigned to the
CO software port. In this case, enter the desired Hardware Slot and Port number in this field.
• Enable Port: This flag is for enabling or disabling a CO Trunk port. When a CO Trunk port is disabled,
neither incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning by clearing this check box.
You may disable CO port in case of trouble with the CO line.

• Name: You may assign a 'Name' to each CO Trunk to facilitate identification. Whenever there is an
extension.
The Name of the port may be the name of the Service Provider of this Trunk Line (recommended).
• CO Hardware Template: A CO Hardware Template is a set of features that completely define the behavior
of the hardware port of the CO, such as Type of CO Trunk, AC Termination Impedance, Pulse-Tone
Dialing, Answer Supervision, Disconnect Supervision, DTMF detection, etc.
Apply a CO Hardware template to the CO trunk port. The SARVAM UCS offers 50 CO Hardware
Templates. By default, CO Hardware Template number 01 is assigned to all CO Trunks. Refer the topic
“CO Hardware Template” to know more.
Check if this default template fulfills the feature requirements of the CO Hardware Ports by clicking the 'CO
Hardware Template' link.
If CO Hardware Template 02 fulfills your requirements, and if the same features at to be applied on all CO
trunk ports, retain Template 02. Similarly, if you want only a few changes to be made to Template 02 and
apply it on all CO Ports, make the changes and retain the template.
However, if different sets of features are to be allowed to different CO hardware ports, then prepare
separate CO Hardware Templates and apply them on the ports as required. To do this,
• Click the CO Hardware Template link to open the page.
• Select a CO Hardware Template Number.
• Customize the CO Hardware Template.
• Go back to the CO Parameters page.
• Apply the CO Hardware Template you customized to the CO Port by entering the template number in
the CO Hardware Template field of this port.
• Repeat the same steps to customize another template and apply it to the CO Port.
To know more about the hardware port features and customizing templates, refer the topic “CO Hardware
Template”.
• Trunk Feature Template: A Trunk Feature Template is a set of features like Time Table, Operator, Auto
Attendant, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR Storage, etc., that defines the behavior
of a Trunk. Apply a Trunk Feature Template to the CO Trunk port. By default, Trunk Feature Template 01 is
applied on all CO Trunks as well as all other trunk types like ISDN BRI, ISDN T1/E1/PRI, GSM, and VoIP.
Refer the “Trunk Feature Template” topic to know more.

Click the Trunk Feature Template link to open the page. Check if the default Template 01 fulfills your
requirement for the CO Trunk port.
If the default Template 01 does not fulfill your requirement, you may prepare a different Trunk Feature
Template and apply on all CO Ports. For this,
• Click the Trunk Feature Template link.
• The Trunk Feature Template page will open.
• Prepare a new template, for example 02 to match your requirements.
• Go back to the CO Parameters page.
• Go to the CO Software Port Number you want to assign the Template you prepared.
• Enter the number of the Template you prepared (02) in the Trunk Feature Template field.
You may also prepare different Templates for different CO Ports, for example Template 02 for certain ports,
Template 03 for others. In this case, follow the steps described above. For each CO Port, enter the number
of the template you have prepared for that port.
To know more about customizing templates, refer the topic “Trunk Feature Template”.
• Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the CO Trunk
port.
Cost Factor is a number assigned to each trunk for identification. This number also serves as a preference
number for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same preference must be
assigned the same Cost Factor. Different trunk types can also be assigned the same Cost Factor. These
trunks are used for routing calls.
Assign a Cost Factor to the CO Trunk port, for instance 02, and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '6' through the CO Trunk Port 001
only,
• You must first assign a Cost Factor (01-99) to CO Port 001, for example, 02.
• Click the link Least Cost Routing.
• Click the Least Cost Routing - Number Based link to open the page.

• Enter '6' in the Number column, Cost Factor 02 as Preference 1, 2, 3 and 4.
All outgoing calls assigned Cost Factor trunk 02 will be made from CO Trunk Port 001.

The above listed parameters fulfill the basic CO trunk port configuration requirements of most users. However, for
users who need advanced features like Call Budget on the CO trunk ports, you may click the Advance button and
program the following parameters:
• Call Budget: If you want to enable 'Call Budget on Trunk' feature, configure the following parameters for
this CO trunk port:
• Type: Select the type of Call Budget on Trunk—Amount, Minutes or Number of Calls—to be applied on
this CO trunk port. By default, no Call Budget type is selected.
• Calls: If you selected ‘Calls’ as the Call Budget Type, enter the number of calls in this field. By default,
• Scheduled Reset: Enable this flag if you want the Call Budget Amount/Minutes to be reset on a
particular date of every month.

• Scheduled (Date): Enter the date of the month (Daily or 1-31) on which you want the Call Budget
Amount/Minutes/Number of Calls to be reset every month. You may select ‘Daily’ if your plan suggests
so.
The consumed Call Budget Amount/Minutes/Number of Calls can be reset from SE and SA Mode, referred
to as Manual Reset. Refer the feature description “Call Budget on Trunk”.
Back on Trunk Port' feature on this CO trunk, configure the following parameters:
seconds.
should be routed:
• Operator
Number’.
Number List 15 is also assigned to all CO trunks as well as all other Trunk port types. If you want the
same numbers strings to be programmed commonly for all CO trunks and Trunk Port types, retain this
list.
If you want a different set of number strings to be programmed for this CO Trunk, select a different
Number List, and assign it to the CO trunk port.

assigned to all CO trunks as well as all other Trunk port types.
You may program the default number list, or a different number list and assign it to this CO Trunk port.
back can be made using the Same Port or an Outgoing Trunk Bundle Group (OTGTBG).
Select ‘Same Port’ if you want the call back to be made using the same port on which the missed call is
received. If you select OGTB Group, the call back will be made using the OGTB Group, which you have
defined.
• If you have completed configuration of all the above listed CO Parameters, click 'Submit' at the bottom of
the page to save your changes.
Viewing CO Trunk Status

You can view the status of CO Trunks on Jeeves only. To do this,

• Under CO Configuration, click Status.
• For each CO Trunk, the following settings will be displayed:

• CO Port Number
• Port Name
• Line Status
You can also view the CO Trunk Status from the Status link. To view, click the CO link under Status.
Configuring CO Trunks using a Telephone

To assign Hardware Slot and Port to the CO Port, dial:

• 1104-CO-Slot-Port offset on the card
Where,
CO is the Software port number of the CO port from 001 to 128.
Slot is Slot number in which the CO is inserted from 01 to 16
Port Offset is the number of the Port on the card from 01 to 99.
To clear the Hardware ID assigned to a CO Port, dial:

• 1104-CO-00-00
To program a Name for a CO Port, dial:

• 5404-1-CO-Name-#* to program a name for a single CO port.
• 5404-2-CO-CO-Name-#* to program the same name for a range of CO ports.
• 5404-*-Name-#* to program the same name for all CO ports.
Where,
Name is a string of alphanumeric characters (maximum) 18 characters.
Terminate the command with #* if the name string has fewer than 18 characters.

To clear the Name of a CO trunk, dial:
• 5404-1-CO-#* to clear the name of a single CO port.
• 5404-2-CO-CO-#* to clear the names for a range of CO ports.
• 5404-*-#* to clear the names of all CO ports.
To enable/disable the CO Trunk, dial:

• 3307-1-CO-Flag to enable/disable a single CO port.
• 3307-2-CO-CO-Flag to enable/disable a range of CO ports.
• 3307-* -Flag to enable/disable all CO ports.
Where,
Flag is
0 for Disable
1 for Enable
Default: Enable
To assign a Trunk Feature Template to a CO Port, dial:

• 5803-1-CO-Trunk Feature Template Number to assign a feature template to a single port.
• 5803-2-CO-CO- Trunk Feature Template Number to assign the same template to a range of ports.
• 5803-*- Trunk Feature Template Number to assign the same template to all ports.
Where,
Trunk Feature Template number is from 01 to 50.
Default: Template 01.
To assign a CO Hardware Template to a CO Port, dial:

• 5903-1-CO-Hardware Template Number to clear the name of a single port.
• 5903-2-CO-CO-Hardware Template Number to clear the names for a range of ports.
• 5903-*-Hardware Template Number to clear the names of all ports.
Where,
Hardware Template Number is CO Hardware Template from 01 to 50.
Default: Template 01.
To assign a Cost Factor to a CO Port, dial:

• 3308-1-CO-Cost Factor to assign cost factor for a single CO port.
• 3308-2-CO- CO-Cost Factor to assign cost factor for a range of CO ports.
• 3308-*-Cost Factor to assign cost factor for all CO ports.
Where,
Cost Factor is from 01 to 99. Default: 01
• For commands to program Call Budget on CO trunks, refer the topic “Call Budget on Trunk”.
To enable / disable Call Back on CO port, dial:

• 3310-1-CO-Code to enable/disable Call Back on a single CO trunk.
• 3310-2-CO-CO-Code to enable/disable Call Back on a range of CO trunks.
• 3310-*-Code to enable/disable Call Back on all CO trunks.
Where,
CO is the number of the software port of the CO trunk from 001 to 128.
Code is
0 for Disable

1 for Enable
Default: Disabled
To program Call Back Timer on CO port, dial:

• 3311-1-CO-Call Back Timer to set timer for a single CO trunk.
• 3311-2 -CO-CO-Call Back Timer to set same timer duration for a range of CO trunks.
• 3311-*-Call Back Timer to set the same timer duration for all CO trunks.
Where,
Timer is from 01 to 99 Sec.
Default: 10 Seconds
To select Call Back Mode for CO port, dial:

• 3312 -1-CO-Call Back Mode to set call back mode for a single CO trunk.
• 3312 -2-CO-CO-Call Back Mode to set same call back mode for a range of CO trunks.
• 3312-*-Call Back Mode to set the same call back mode for all CO trunks.
Where,
Call Back Mode is from 1 to 5
1 for Built-in Auto Attendant
2 for PIN Auth. - Multiple Calls
3 for CLI Auth. - Multiple Calls
4 for CLI Auth. - Single Call - Ans. Sig.
5 for Operator
Default: Operator
To program Call Back On for CO port, dial:

• 3313-1-CO-Call Back on to program call back on for a single CO trunk.
• 3313-2-CO-CO-Call Back on to program the same call back on option for a range of CO trunks.
• 3313-*Call Back on to program the same call back on option for all CO trunks.
Where,
Call back on is
1 for CLI Number
2 for Alternate Number
Default: CLI Number
To assign Call Back - Incoming Number List to a CO port, dial:

• 3314-1 -CO-Incoming Number List to assign a list to a single CO trunk.
• 3314-2-CO-CO-Incoming Number List to assign the same list to a range of CO trunks.
• 3314-*-Incoming Number List to assign the same list to all CO trunks.
Where,
Default: 15
To assign Call Back - Outgoing Number List to a CO port, dial:

• 3315-1 -CO-Outgoing Number List to assign a list to a single CO trunk.
• 3315-2-CO-CO-Outgoing Number List to assign the same list to a range of CO trunks.
• 3315-*-Outgoing Number List to assign the same list to all CO trunks.
Where,

Default: 16
To select Call Back From port for a CO Trunk port, dial:

• 3316-1-CO-Call Back From to select Call Back From for a single CO trunk.
• 3316-2-CO-CO-Call Back From to select the same Call Back From for a range of CO trunks.
• 3316-*-Call Back From to select the same Call Back From for all CO trunks.
Where,
Call Back From is
1 for Same Port
2 for OGTB Group
Default: Same Port
To assign Call Back - OGTB Group to a CO port, dial:

• 3317-1-CO-OGTB Group to assign an OGTBG to a single CO trunk.
• 3317-2-CO-CO-OGTB Group to assign the same OGTBG to a range of CO trunks.
• 3317-*-OGTB Group to assign the same OGTBG to all CO trunks.
Where,
OGTB Group is from 01 to 32.
Default: 01
• Exit SE mode.

Configuring E&M Lines
The SARVAM UCS supports E&M interface. A maximum of 32 ports are supported in SARVAM UCS.
If you have correctly installed the E&M Cards and observed the Reset Cycle, you may now program the E&M Ports
using Jeeves or a Telephone, depending on your installation scenario.
An E&M port of SARVAM UCS can be programmed to take on the function of:
• a Subscriber (Station) - works like an extension interface, receiving incoming calls.
OR
• a Trunk - works like a trunk interface when any of the extensions of the System makes an outgoing call
through it.
OR
• a Tie Line - takes on a dual personality: functioning as both a station and a trunk. The E&M port works like
an extension interface for incoming calls. It works like a trunk interface when any extension makes an
outgoing call through it.
This dual function is used in Systems that are used as Transit Exchanges as in a PLCC Network. Read
The E&M Card of SARVAM UCS supports the following features:
• E&M Trunk Seizure Type185: Immediate, Immediate with Ack, Immediate + Wink, Immediate with
Ack+Wink (MFCR2)186, Seizure Pulse, Seizure Pulse + Wink, Express, and Radio.
The E&M Interface (Type IV and Type V connection) and the Speech Interface (2-wire speech or 4-wire)
are selected at the time of installation by changing the Jumper settings.
Configuring the E&M using Jeeves

• Under Configuration, click E&M Configuration.
Protocol.
186. Currently supported only on ETERNITY GENX E&M4 Card.

• Click the E&M Parameters.
Configure the following parameters for each E&M port on this page:
• E&M Port No.: This non-editable field is the software port number of the E&M Port. Refer the topic
“Software Port and Hardware ID” to know more.
• H/w (Hardware) Slot - Port: 'Slot' is the number of the number of the Universal Slot in which the E&M
Card is inserted. 'Port' is the number of the E&M hardware port on which the Tie Line equipment
(System, Router, Leased Line, etc.) is connected.
automatically to the E&M software ports.
For example: if you have inserted the E&M8 Card in Slot 05 and E&M4 Card in Slot 06 of SARVAM
UCS, the system will assign the Hardware Slot 05 and port numbers 01-08 to the E&M Software Ports
from 001 to 008 respectively. The system will assign hardware Slot 6 and port numbers 01-04 to the
E&M Software Ports 009 to 012. Refer the topic “Software Port and Hardware ID” to know more.
However, if required, you may change the Hardware Slot and Port assigned to the E&M software port.
In this case, enter the desired Hardware Slot and Port number in this field.
• Enable Port: This flag is for enabling or disabling an E&M port. When an E&M port is disabled, neither
incoming nor outgoing calls can be made from that port.
By default, the port is enabled. You may disable ports that are not functioning.

• Name: You may assign a Name to the E&M Port. Whenever there is an incoming call on this Port, this
name will be displayed on the destination extension, while receiving the call.
• E&M Feature Template: This parameter is applicable to all E&M Ports. The E&M Feature Template is
a complete set of E&M features to be applied on E&M Ports according to their 'Orientation Type',
whether they are Stations, Trunks or Tie-Lines.
By default E&M Feature Template 01 is applied on all E&M Ports. This template has 'Station' as the
default Orientation Type.
If all the E&M Ports are to be programmed as 'Stations' retain this template.
If all the E&M Ports are to be programmed as 'Trunks' use the default E&M Feature Templates 09 and
10 which have 'Trunks' as Orientation Type.
If some of the E&M Ports are to be programmed as Stations, some as Trunks and yet others as Tie
Lines, prepare different E&M Feature Template for each Orientation Type and apply them to the related
ports.
• Under Configuration, click E&M Feature Template to open the page.
• Select an E&M Feature Template Number.
• Customize the E&M Feature Template.
• Go to the E&M Parameters page.
• Apply the E&M Feature Template you customized to the E&M Port by entering the template number
in the 'E&M Feature Template' field of this port.
• Repeat the same steps to customize another template and apply it to another E&M Port.
Refer the topic “E&M Feature Template” for more details on customizing the templates and applying
them on E&M Ports.
• Trunk Feature Template: This parameter is relevant only if the E&M Port is to be programmed to
function as a Trunk or a Tie-Line187. To know more, refer “Trunk Feature Template”.
Assign an E&M Feature Template to the E&M Port. A Trunk Feature Template is a set of features like
Time Table, Operator, Auto Attendant, DISA, Trunk Auto Answer, Trunk Landing Group, SMDR
Storage, etc., that defines the behavior of a Trunk. Apply a Trunk Feature Template to the E&M Port.
By default, Trunk Feature Template 01 is applied on all E&M (trunk) Ports. This Template is also the
default template commonly applied on all other trunk types (CO, ISDN BRI, ISDN T1E1PRI, GSM, and
VoIP).
187. To program the E&M Port as a Trunk or a Tie-Line, you must set the 'Orientation Type' of the E&M Port to 'Trunk' or 'Tie-Line' in the
“E&M Feature Template” applied on the port.

Click the link Trunk Feature Template to open the page. Check if the default Template 01 fulfills your
requirement for the E&M Trunk port.
If not, you may prepare a different Trunk Feature Template and apply on all E&M Ports. For this,
If the default Template 01 does not fulfill your requirement,
You may prepare a different Trunk Feature Template and apply on all E&M Ports. For this,
• Click the Trunk Feature Template link.
• The Trunk Feature Template page will open.
• Prepare a new template, for example 04 to match your requirements.
• Go to the E&M Parameters.
• Go to the E&M Software Port Number you want to which you want to assign the Template you
prepared.
• Enter the number of the Template you prepared (04) in the Trunk Feature Template field.
You may also prepare different Templates for different E&M Ports, for example Template 04 for certain
ports, Template 05 for others. In this case, for each E&M Port, enter the number of the template you
have prepared for that port.
To know more about customizing templates, refer the topic “Trunk Feature Template”.
• Station Basic Feature Template: This parameter is applicable on when the E&M Port is to be
programmed to function as a Station (Orientation type = Station or Tie-Line).
Assign a “Station Basic Feature Template” to the E&M Port functioning as a Station. By default, Station
Basic Feature Template 01 is assigned to all extensions, that includes SLT and DKP ports.
Control”, “OG Trunk Bundle Group”, etc.) of the E&M Ports functioning as Stations.
all E&M (station) ports, retain Template 01.
If different sets of features are to allowed to different E&M (station) Ports, then prepare separate Station
• Click the link Station Basic Feature Template to open the page.
• Now go back to the E&M Parameters page.

Template' field of the E&M Port (for example: E&M No. 003) on which you want to apply this
template. If you want to apply this template to other ports too, like E&M No. 004, 005, and 006,
• Repeat the same steps to customize and assign a different Template to another E&M port.
and applying on the ports.
• Station Advanced Feature Template: This parameter is applicable only when the E&M Port is to be
programmed to function as a Station.
By default Station Advanced Feature Template 01 is assigned to all extensions, that includes SLT and
DKP ports as well as E&M ports with the orientation type 'Station'.
Check if this default template fulfills the feature requirements of the E&M Ports (with 'station' as
orientation type) by selecting the 'Station Advanced Feature Template' link.
all E&M (station) ports, retain Template 01.
If different sets of features are to be allowed to different E&M (station) Ports, then prepare separate
Station Advanced Feature Templates and apply them on the ports.
To do this,
• Go to the E&M Parameters page.
Template' field of the E&M Port (for example, E&M No. 003) on which you want to apply this
template. If you want to apply this template to other ports too, like E&M No. 004, 005, and 006,
• Repeat the same steps to customize and assign a different Template to another E&M (Station) port.

• Priority: This parameter is applicable only when the E&M Port is to be programmed to function as a
Station188. To know more, refer “E&M Feature Template”.
Each station of the SARVAM UCS is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls
an extension with lower priority, a triple ring is placed on the called extension. To know more, read the
feature description “Priority”.
By default, the Priority of all E&M Ports functioning as Stations is set to '9-Highest'. So, decide what
Priority Level you will assign to each of the E&M Ports functioning as Stations and set the desired level
for each port.
• Cost Factor: This parameter is of relevance only if 'Least Cost Routing' feature is applied on the E&M
Trunk port.
Assign a Cost Factor to the E&M Trunk port, for example, 03 and program Least Cost Routing Table
accordingly.
For example, if you want to route all outgoing calls starting with number '6' through the E&M Trunk Port
001 only,
• You must first assign a Cost Factor (01-99) to E&M Trunk Port 001, for example 03.
• Click the Least Cost Routing - Number Based link to open the page.
All outgoing calls assigned Cost Factor trunk 03 will be made from E&M Trunk Port 001.
• Repeat the same steps to configure other E&M Trunk/Station ports.
• If you have completed configuration of all the above listed E&M Parameters, click 'Submit' at the
Configuring the E&M using a Telephone

To assign Hardware ID to an E&M Software Port, dial:

• 1105-E&M-Slot-Port offset on the card
Where,
E&M is the Software Port number of the E&M port from 001 to 128.
Slot is Slot number in which the E&M Card is inserted from 01 to 16
188. Recall that E&M ports can function as trunks, as stations and have both functions. When an E&M Port is programmed as a Station
Interface, it can only receive incoming calls. To program the E&M Port as a Station, you must set the 'Orientation Type' of the E&M
Port to 'Station' in the E&M Feature Template applied on the port.

Port Offset is the number of the E&M Port on the card from 01 to 99.
To clear the Hardware ID assigned to an E&M Software Port, dial:

• 1105-E&M-00-00
To enable/disable an E&M Port, dial:

• 3321-1-E&M-Code to enable/disable a single E&M port.
• 3321-2-E&M-E&M-Code-#* to enable/disable a range of E&M ports.
• 3321-*-Code to enable/disable all E&M ports.
Where,
Code is
0 for Disable
1 for Enable
Default: Enable
To program a Name for an E&M Port, dial:

• 5406-1-E&M-Name-#* to program a name for a single E&M port.
• 5406-2-E&M-E&M-Name-#* to program the same name for a range of E&M ports.
• 5406-*-Name-#* to program the same name for all E&M ports.
Where,
Name is a string of alphanumeric characters 18 characters (maximum).
Terminate the command with #* if the name string has fewer than 18 characters.
To clear a name assigned to an E&M port, dial:

• 5406-1-E&M-#* to clear the name of a single E&M port.
• 5406-2-E&M-E&M-#* to clear the names for a range of E&M ports.
• 5406-*-#* to clear the names of all E&M ports.
To assign an E&M Feature Template to an E&M Port, dial:

• 6003-1-E&M-Template Number to assign a template to a single port.
• 6003-2-E&M-E&M-Template Number to assign the same template to a range of ports.
• 6003-*-Template Number to assign the same template to all ports.
Where,
Template Number is from 01 to 50.
Default: E&M Feature Template 01.
To assign a Trunk Feature Template to an E&M Port, dial:

Where,
Default: Trunk Feature Template 01.
To assign a Station Basic Feature Template to an E&M Port, dial:


Where,
Default: Station Basic Feature Template 01.
To assign a Station Advanced Feature Template to an E&M Port, dial:

Where,
Default: Station Advanced Feature Template 01
To set a Priority Level for an E&M Port, dial:
This command is applicable only when the E&M Port is configured to function as a station!
• 3915-1-E&M-Priority to set Priority for a single E&M Port.

• 3915-2-E&M-E&M-Priority to set the same Priority for a range of E&M Ports.
• 3915-*-Priority to set the same Priority for all E&M Ports.
Where,
Default: 9-Highest
To assign a Cost Factor to an E&M port, dial:

• 3322-1-E&M-Cost Factor to assign Cost Factor to a single E&M port.
• 3322-2-E&M-E&M-Cost Factor to assign Cost Factor to a range of E&M ports.
• 3322-*-Cost Factor to assign Cost Factor to all E&M ports.
Where,
Cost Factor is from 01 to 99. Default: 01
• Exit SE mode.

Configuring LCR
Least Cost Routing (also referred to as Automatic Route Selection) is an expense control feature of SARVAM UCS.
Least Cost Routing (LCR) is useful when there are different trunk lines for making outgoing calls, and the service
providers of these trunks offer different tariffs for calls made to certain locations or numbers or during a particular
time of the day.
When a call is made from an extension of the SARVAM UCS, LCR recognizes where the call is going. Depending
upon how the LCR is programmed, the system routes the call through the assigned trunks.
The system can be programmed to select the most cost effective trunk for the time of the day when the call is made
from the extension or to select the most cost effective trunk for the destination number dialed from the extension or
to select the most cost effective trunk considering both time of the day and destination number.
Accordingly, SARVAM UCS supports four types of LCR which can be programmed, namely:
1. Time-based LCR: This type of LCR may be used when you have trunk lines of more than one service
provider, and each offers a different tariff according to the time of the day.
For example, Service Provider 1 offers a lower tariff for calls made between 9am to 8pm, while Service
Provider 2 offers a lower tariff for calls made between 8pm to 9am.
When Time-based LCR is programmed, the system uses the Online-dialing logic, whereby digits dialed by
the user are directly passed on to the trunk.
2. Number based LCR: This type of LCR may be used when you have trunk lines of more than one service
provider, and each offers different tariffs according to the area or distance, or phone numbers dialed. For
instance, Service Provider 1 provides cheaper calling rates for calls made from City A to City B, than
Service Provider 2 and Service Provider 3.
3. Time and Number based LCR: This type of LCR is a combination of number and time based LCR, that is,
the service providers offer different tariffs according to the time of the day as well as area/distance.
For example, Service Provider 1 offers lower rates for calls made from City A to City B during peak hours
9am to 8pm, as compared to Service Provider 2, whereas Service Provider 2 offers cheaper rates for calls
made from City A to City B during off peak hours (8pm to 9 am).
When Time+Number-based LCR is programmed, the system uses Store and Forward dialing logic,
whereby digits dialed by the user are first stored at a memory location in the system, and then dialed out
on the assigned trunk.
4. Service Provider-based LCR: This type of LCR may be used when the same Service Providers offer
different rates for calls made to numbers within their own network and for calls made to numbers of
another Service Provider's network. For example, Service Provider 1 offers lower rates to call a Service
Provider1 number in City A and in City B, than for calling numbers of Service Provider 2 in the same cities.
This type of LCR may also be used when the same Service Providers apply different charges for different
subscriber services provided by them. For example, Service Provider 1 offers both Fixed Line as well as
GSM services and applies different charges for fixed line and GSM services.

When Service Provider-based LCR is programmed, whenever a number is dialed out, the system ignores
the area code, checks the number in the 'Service Provider-based LCR table', and routes the call according
to the trunk programmed for that number.
SARVAM UCS also supports LCR based on Carrier Pre-Selection. This type of LCR is useful where there
exist different service providers for local and long distance calls. Refer the topic “Least Cost Routing-
Carrier Pre-Selection” to know more.
Cost Factor
For LCR to work, all trunks that are allotted to extensions for making outgoing calls, must first be assigned a Cost
Factor.
Cost Factor is a number assigned to each trunk for identification. This number also serves as a preference number
for the trunk. The Cost Factor can be from 1 to 99. Trunks having the same preference must be assigned the same
Cost Factor. Different trunk types can also be assigned the same Cost Factor. These trunks are used for routing
calls.
By default all trunks are assigned Cost Factor number 01.
After assigning Cost Factor to Trunks, you must configure the Type of LCR to be used on Trunks in the Outgoing
Trunk Bundle Group (OGTBG) allotted to the extensions for making calls.
Assigning Cost Factor to Trunks

• On a sheet of paper, make a list of all the trunks assigned to extensions for making outgoing calls.
• Make a table of the trunk types and assign a cost factor to each trunk type, as shown below.
Trunk Type and

Service Provider Cost Factor
Number
CO-001 BSNL 01
CO-002 BSNL 02
MOB-001 Reliance 03
MOB-002 BSNL 04
MOB-003 Airtel 05
MOB-004 Vodafone 06
BRI-01 BSNL 07
BRI-02 Reliance 08
• Program the Cost Factor number you assigned to the Trunk types in their respective trunk parameters. For
instance, assign Cost Factor 01 to CO-001 and Cost Factor 002 to CO-002 in the “Configuring CO
Trunks”. Similarly, assign Cost Factor 03 to Mobile Trunk 001, Cost factor 04 to Mobile Trunk 002, Cost
Factor 05 to Mobile Trunk port 003, and Cost Factor 06 to Mobile Trunk port 004 in the “Mobile Port
Parameters”. Assign Cost Factor 07 to BRI Trunk port 01 and Cost Factor 08 to BRI Trunk port 02 in
“Configuring BRI Trunks”.

Configuring Time-based LCR
• You can configure Time-based LCR for as many as 8 different Time Zones.
• On a sheet of paper, make a table for Time-based LCR.
• Define the Time Zone, that is, the start and end time, when the LCR should be applied for the outgoing
calls. The Time Zone you define is stored at an Index number from 1 to 8.
• For each Time Zone that you define, select the Trunk as your first preference, that is, Preference 1. Select
the trunk of your second, third and fourth preference. When the trunk you selected as first preference is
busy, the system will route the call through the next trunk you have set that is free.
• Refer to the table you prepared for assigning Cost Factor to trunks.
• For example, you want calls made during 9am to 8pm to be routed through BSNL CO trunks (CO-001 and
CO-002). If these trunks are busy, you want the system to route calls through the BRI line of BSNL trunk.
When this line is busy, you want the system to attempt to route calls through the BRI line of Reliance.
• You want calls made between 8pm to 9am to be routed through BSNL CO trunk 001 only.
• At Time Zone Index 1, define the Time Zone start and end time in 24 Hours:Minutes format, enter the Cost
Factor you assigned to CO-001 (01) and CO-002 (02) as Preference 1 and Preference 2 respectively.
Enter the cost factor you assigned to BRI-01 (07) and BRI02 (08) as Preference 3 and Preference 4
respectively.
Time Zone Cost Factor

Time Zone
Index Start Time End Time Preference Preference Preference Preference
(HH:MM) (HH:MM) 1 2 3 4
1 09:00 20:00 01 02 07 08
2 20:01 08:59 01 01 01 01
• Similarly, at Time Zone Index 2, define the Time Zone in 24 Hours: Minutes format. Enter the Cost factor
you assigned to CO-001, that is, 01 as Preference 1, 2, 3, and 4. When calls are made during this time
period, they will be routed through CO-001 only.
• If you have finished defining Time Zones and the preferred trunks for the time zones, configure the Time-
based LCR using Jeeves or a Telephone.

Configuring Time-based LCR using Jeeves
• Under Configuration, click Least Cost Routing (LCR).
The links to the LCR options appear.
• Click Time-based to open the page.
• Enter the values of the Time-based LCR you prepared on the sheet of paper in the appropriate fields.
• Log out of Jeeves or continue programming, as required.
Configuring Time-based LCR using a Telephone

• Enter SE mode.
To program Time Zone at a Time Zone index, dial:

• 3402-Time Zone Index-Start Time-End Time
Where,
Time Zone Index is from 01 to 08.
Start Time is the time in HH:MM format when the Time zone starts.
End Time is the time in HH:MM format when the Time zone ends.
For example to program 09:00 to 20:00 hours at Time Zone Index 1, dial 3402-01-0900-2000
By default, Time Zone is 00.00 to 23.59.
To program the Cost Factor (Service Provider preference) for the Time Zone, dial:
• 3403-Time Zone Index-CF1-CF2-CF3-CF4
Where,
CF1is the first preferred (the cheapest) service provider.
CF2 is the second preferred (second cheapest) service provider.
CF3 is the third preferred service provider.

CF4 is the fourth preferred service provider.
For example, to program the preferred trunks for the Time Zone 09:00 to 20:00 hours, dial 3403-01-01-
02-07-08
It is mandatory to complete this command with CF1 to CF4. If you have only one service provider, program
the same as CF1, CF2, CF3, CF4.
To default the Time Zone based LCR table, dial:

• 3401
• Exit SE mode.
Configuring Number-based LCR

• You can configure Number-based LCR for as many as 99 different Numbers, which are stored against
Index numbers from 01 to 99.
• On a sheet of paper, make a table for Number-based LCR.
• Enter each of the number strings at an Index number from 01 to 99. A Number string may be a complete
telephone number, a truncated phone number or an area code. The number can be a maximum of 64
characters (Digits + Wildcards). Valid characters: 0 to 9, *, #, X, T, Comma [,], Hyphen [-], Caret [^ ]. See
“Wildcard Characters” to know the various number patterns you can use.
• For each number string you enter, select a Trunk as your first preference, that is, Preference 1. Select the
trunk of your second, third and fourth preference. When the trunk you selected as first preference is busy,
the system will route the call through the next trunk you have set that is free.
• Refer to the table you prepared for assigning Cost Factor to trunks.
For example, you want all mobile numbers to be routed through the Mobile Trunk ports, all local numbers
to be routed through the CO ports.
All mobile numbers start with the number '9', which is prefixed with a '0' when making long distance mobile
calls, so enter '9' and '09' as the number strings. For '9' as well as '09', select the Mobile trunks through
which the calls should be made in order of preference.
Similarly, all local numbers start with 2, so enter this number in the number string column, and select the
CO trunk in the order of preference. As in this example, you have only two CO trunks, so you may keep the
same two trunks as your preference.
Cost Factor
Index Number
Preference 1 Preference 2 Preference 3 Preference 4
1 9 04 03 05 06
2 09 04 06 05 03
3 2 01 02 01 02

Cost Factor
Index Number
Preference 1 Preference 2 Preference 3 Preference 4
99
• If you have finished entering the number strings, and selecting the preferred trunks for the numbers,
configure the Number-based LCR using Jeeves or a Telephone.
Configuring Number-based LCR using Jeeves

• Click Number-based to open the page.
• Enter the values of the Number-based LCR you prepared on the sheet of paper in the appropriate fields.
• In Number you can enter may be a complete telephone number, a truncated phone number or an area
code. You may enter upto 64 characters (Digits + “Wildcard Characters”) in this field. Valid characters: 0 to
9, *, #, X, T, Comma [,], Hyphen [-], Caret [^ ]. Default: Blank.
Wildcard Characters
SARVAM UCS supports following characters.
Character Description
X (letter X) X represents any single digit from 0 to 9.

# When # is configured in a number string, it will not be considered as End of Dialing.
* When * is configured in a number string, it will not be considered as End of Dialing.
+ (plus) can be configured as a first character of the Destination Number string in the
+
SIP Trunk only.
[-] Hyphen within the bracket, defines a range. Only digits 0-9 are allowed within a bracket.
[,] Comma within a bracket is used as a separator between the groups of numbers.
Caret within a bracket is used to deny or restrict the number or range defined after the
[^]
symbol. Only digits 0-9 are allowed after the caret.
Character T can be configured only as a last character in a number string. When

T (letter T)
configured in a number string, the system waits for End of Dialing.
Refer the following table to understand how you can configure the Numbers.
Numbers Description
1XX Allows you to dial any number in a range from 100 to 199.
[2-5]XX Allows you to dial any 3 digit number in a range from 200-599.
Allows you to dial any 3 digit number in the range from 200-299, 300-399, 800-
[2,3,8]XX
899.
[2-9]XXXXXX Allows you to dial any 7 digit number in the range from 2000000-9999999.
Allows you to dial a 4 digit number: 2301, 2311, 2331, 2341, 2351, 2361, 2371,
23[^2]1
2381, 2391.
2630[500-550] Allows you to dial a 7 digit number in the range from 2630500-2630550.
Allows you to dial a 2 digit number in the range from 00 to 99 except the numbers
[^6-7]X
from 60 to 79.
1234 Allows you to dial 1234 number only.
Allows you to dial any number starting with 011. The number must be of minimum
011T
3 digits and maximum digits must be as configured for the port.
Configuring Number-based LCR using a Telephone

• Enter SE mode.
To program a number at number Index in the Time-based LCR table, dial:

• 3411-Number Index-Number String-#*
Where,
Number Index is from 02 to 99.
Number String can be a complete telephone number, a truncated telephone number or an area code.
Number string may be a maximum of 16 digits. Terminate the command with #* if the number string has
fewer than 16 digits. Terminate the command with #* if the number string has fewer than 16 digits.
By default, Number String is 'Blank'.
For example to program '9' at Index 02, dial 3411-02-9-#*

To clear a number string programmed at a number index, dial:
• 3411-Number Index-#*
Where,
To program Cost Factor (Service Provider preference) for the each Number, dial:
• 3412-Number Index-CF1-CF2-CF3-CF4
Where,
For example, to program Cost Factor for number '9' at Index 01, dial 3412-01-04-03-05-06
To default the Number-based LCR table, dial:

• 3410
Configuring Time and Number-based LCR

• This is a combination of the Time Zone-based and Number-based LCR. You may use this feature if your
service providers offer lower call rates for calls made to certain numbers during a certain time of the day.
• On a sheet of paper, make a table for Time and Number-based LCR.
• Define the Time Zones when the service providers offer lower tariff. You can define up to 8 time zones.
• For each Time Zone you define, specify the Number strings on which lower tariff is applied during that
Time Zone. The number can be a maximum of 64 characters (Digits + Wildcards). Valid characters: 0 to 9,
*, #, X, T, Comma [,], Hyphen [-], Caret [^ ]. See “Wildcard Characters” to know the various number
patterns you can use.
• For each Number string you enter for a particular time zone, assign Cost Factor. Select a trunk as your first
preference. Select trunks of your second, third and fourth preference. Refer to the table you prepared for
assigning Cost Factor.
• You can enter up to 99 different number strings, which are stored at Index numbers from 01 to 99. The
Number strings may be complete telephone numbers, truncated phone numbers or area codes.
When the trunk you selected as first preference is busy, the system will route the call through the next
trunk you set as preference if it is free.
For example, service provider of CO-001 and CO-002 (assigned Cost Factor 01 and 02) offers the lowest
rate for calls made to Area Code 022 between 8am to 12pm, followed by service providers of Mobile
Trunk-02 (assigned cost factor 04) and Mobile Trunk-01 (assigned cost factor 03).
• Define Time Zone 1 Start and End time as 08 to 12:00 hours
• Enter area code 022 as Number string at Number Index 1

• Assign Cost Factor preference for the number string in this sequence: 01, 02, 04, 03
Time
Time Zone1 Time Zone2 Time Zone3 : Zone
8
HH MM HH MM HH MM
Start Time
08 00 12 00 09 00
End Time 12 00 18 00 20 00
Cost Factor Cost Factor Cost Factor
Prefe Prefe Prefe Prefere Prefe Prefe Prefe Prefere Prefe Prefe Prefe Prefere
Number rence rence rence nce 4 rence rence rence nce 4 rence rence rence nce 4
Index
1 2 3 1 2 3 1 2 3
1 022 01 02 04 03 04 03 02 01 03 05 06 04
2 011 01 02 04 05
3 080 01 02 05 06 03 05 06 04
99
• If you have finished defining the time zones, entering the number strings, and selecting the preferred
trunks for the number strings, configure the Number and Time-based LCR using Jeeves or a Telephone.
Configuring Time+Number-based LCR using Jeeves

• Click Time + Number based to open the page.

• Enter the values of the Time+Number-based LCR you prepared on the sheet of paper in the appropriate
fields.
Configuring Time+Number-based LCR using a Telephone

• Enter SE mode.
To define Time Zone for Time+Number-based LCR, dial:

• 3421-Time Zone Index-Start Time-End Time
Where,
Start Time is the time in HH:MM format when the Time zone starts.
End Time is the time in HH:MM format when the Time zone ends.
By default, Time Zone is 00.00 to 23.59.
For example, to define 08:00 to 12:00 as start and end time of Time Zone 1, dial 3421-1-0800-1200
To program Cost Factor (Service Provider preference) for the each Number and Time Zone, dial:
• 3423-Number Index-Time Zone Index-CF1-CF2-CF3-CF4
Where,
For example, to program Cost factor for Area code 022 during time zone 08 to 12:00 hours as entered
in the sample table, dial 3423-01-1-01-02-04-03
To default the Time and Number-based LCR table, dial:

• 3420
Configuring Service Provider-based LCR

• In Service Provider-based LCR, whenever a number is dialed out, the system ignores the area code, and
starts checking the numbers in the 'Service Provider-based LCR table' and routes the call according to the
trunk programmed for that number. For this, you must program the two parameters Area Code and Ignore
Digit Count in the Area Code Table.
• On a sheet of paper make a table for Service Provider-based LCR.
Index Area Ignore Cost Factor

Number
No. Code Digit Count Preference 1 Preference 2 Preference 3 Preference 4
01 3 080 3 08 07 01 02
02 6 022 3 07 01 02 08
: : :
99 2 03852 5 01 02 01 02

• As you can see, the Service Provider-based LCR Table is similar to the Number-based LCR table.
• You can program as many as 99 different numbers which are stored against Index numbers from 01 to 99.
• The number strings may be the complete telephone number, a truncated phone number or the first digit of
the phone number. The number can be a maximum of 64 characters (Digits + Wildcards). Valid
characters: 0 to 9, *, #, X, T, Comma [,], Hyphen [-], Caret [^ ]. See “Wildcard Characters” to know the
various number patterns you can use.
• For each number string that you enter against an Index number, you must also specify the Area Code and
the Ignore Digit Count.
• The Ignore Digit Count is the number of digits in the area code that the system should ignore before
checking the Service Provider-based LCR table. For each area code that you enter, the corresponding
Ignore Digit Count will be the number of digits in the area code. For example, the area code for the number
starting with '3' is 080, which consists of 3 digits. So, the Ignore Digit Count for the number/area code 080
will be 3.
• For each number string and area code that you enter, assign the Trunk of the service provider that you
prefer as your first, second, third and forth preference for dialing that number/area code. Refer the table
you prepared for assigning Cost Factor to trunks.
• If you have finished entering the number strings, their corresponding area codes and the Ignore Digit
Count, and the preferred trunks, configure Service Provider-based LCR using Jeeves or a Telephone.
Configuring Service Provider-based LCR using Jeeves

• To program Area Code and Ignore Digit Count, Under Configuration, click Call Cost Calculation.
• Click Area Code Table to open the page.
Each area code is stored at an index number.

• Enter the Area Codes and the corresponding Ignore Digit Counts from the sheet you prepared for Service
Provider based-LCR. You may also enter the respective name for each area code, if desired.
• Now, click the Least Cost Routing (LCR). The links for the LCR options appear.
• Click Service Provider based to open the page.
• Enter the values of the Service Provider-based LCR you prepared on the sheet of paper in the appropriate
fields.
Configuring Service Provider-based LCR using a Telephone

• Enter SE mode.
To program Area Code in the Area Code Table, dial:

• 2620-Area Code Index-Area Code-#*
Where,
Area Code Index is from 001 to 999.
Area Code is a string of maximum 7 digits. Terminate the command with #* if the number string has
fewer than 7 digits.
By default, Area Code is 'Blank'.
For example, to program Number string '080' at Number Index 001, dial 2620-001-080-#*
To clear an Area Code in the Area Code Table, dial:

• 2620-Area Code Index-#*
To program Ignore Digit Count for an Area Code, dial:

• 2623-Area Code Index-Ignore Digit Count
Where,

Ignore Digit Count is from 0 to 9.
By default, Ignore Digit Count is '0'.
Refer the topic "Area Code Table" under Call Cost Calculation to know more.
To program a number at number Index in the Service Provider-based LCR table, dial:
• 3441-Number Index-Number String-#*
Where,
Number String can be a complete telephone number, a truncated telephone number or an area code.
Number string may be a maximum of 16 digits. Terminate the command with #* if the number string has
fewer than 16 digits.
By default, Number String is 'Blank'.
For example to program '3' at Index 02, dial 3441-02-3-#*
To clear a number string programmed at a number index, dial:

• 3441-Number Index-#*
Where,
To program Cost Factor (Service Provider preference) for the each Number, dial:
• 3442-Number Index-CF1-CF2-CF3-CF4
Where,
For example, to program Cost Factor for number '3' at Index 01, dial 3412-01-08-07-01-02
To default the Number-based LCR table, dial:

• 3440
• Exit SE mode.
Configuring LCR Type on Trunks

After assigning Cost Factor to Trunks and programming the LCR Type - Time-based, Number-based, Time and
Number-based, Service Provider-based - you must now apply the desired LCR Type on the Outgoing Trunk Bundle
Group (OGTBG) allotted to the extensions. This can be done using Jeeves or a Telephone.
Configuring LCR Type in OGTB using Jeeves


• Under Configuration, click Outgoing Trunk Bundle Group.
• For each OGTBG number assigned to extensions, select the desired LCR Type: Time-based, Number-
based, Time+Number based, Service Provider-based (Cost Factor).
You can find the OGTBG number assigned to each extension from the Station Basic Feature Template
assigned to the extension.
Configuring LCR Type in OGTB using a Telephone

• Enter SE mode.
To select LCR type in OG Trunk Bundle Group, dial:

• 1404-1-OGTBG-LCR Type to select an LCR Type for a single group.
• 1404-2-OGTBG-OGTBG-LCR Type to select the same LCR Type for a range of OGTB groups.
• 1404-*-LCR Type to select the same LCR Type for all OGTB groups.
Where,
OGTBG is from 01 to 32.
LCR Type is
0 for No LCR
1 for Time zone-based LCR
2 for Number-based LCR
3 for Time+Number-based LCR
4 for Service Provider-based (Cost Factor) LCR
By default, 'No LCR' is selected as LCR Type.
• Exit SE mode.

Configuring Emergency Number Dialing
SARVAM UCS supports dialing of Emergency Numbers immediately without any blocking.
The system will disregard features such as Toll Control, Call Budget, Automatic Number Translation, Call Duration
Control on the extensions when dialing out Emergency Numbers. To know more, refer the topic “Emergency
Dialing”.
For Emergency Number Dialing, you need to configure the Emergency Number Table. In this table, each
Emergency Number is to be assigned an Outgoing Trunk Bundle Groups (OGTB) through which the Emergency
Number is to be routed.
The system loads the default Emergency Numbers and Outgoing Trunk Bundle Group (OGTBG) in the Emergency
Number Table as per the “Region” you selected for the system. The Emergency Numbers loaded by default in the
Emergency Number Table are non-editable, but you can re-assign the default OGTB, as per your requirement.
To use the feature “Emergency Calls (911) - Reporting to PSAP”, make sure you assign only T1 PRI lines in
the outgoing trunk bundle group for dialing the number 911.
The default Emergency table as per the Region you select is given below:
Emergency Emergency Emergency Emergency Emergency Emergency Emergency Emergency Emergency Emergency
Feature
Number 1 Number 2 Number 3 Number 4 Number 5 Number 6 Number 7 Number 8 Number 9 Number 10
Australia 000 106 112
OGTB 32 30 31 1 1 1 1 1 1 1
Group
Number
Banglade 999
sh
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Belgium 101 100 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Bhutan 110 112 113
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Canada 911
OGTB 32 1 1 1 1 1 1 1 1 1
Group
Number
China 110 120 119

Feature
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Germany 110 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
India 100 101 108 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Indonesia 110 118 119 113 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Italy 113 118 115 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Jordan 191 199
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Kazakhsta 03
n
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Kenya 999
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Kuwait 777
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Malaysia 999 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Maldives 102 108

Feature
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Mauritius 999 115 144
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Mexico 911
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Namibia 911
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Nepal 100
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
New 111 112 911 08

Zealand
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Oman 9999
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Pakistan 15 115 16 911 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Philippine 117 911 112

s
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Poland 997 999 998 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Russia 02 03 01 112

Feature
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Singapore 999 995 112 911
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
South 10111 10177 1022

Africa
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Spain 091 061 080 085 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Sri Lanka 119 110 111
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Sudan 110 112 113
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Sweden 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Taiwan 110 119
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Thailand 191 1669 199
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
Turkey 155 112 110
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
UAE 999 998 997 112

Feature
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
UK 999 112
OGTB 1 1 1 1 1 1 1 1 1 1
Group
Number
USA 911 112
OGTB 32 1 1 1 1 1 1 1 1 1
Group
Number
For example, if you selected USA as Region, the default Emergency Number Table would look like this:
Default Emergency Number Table for USA
Index Emergency Number OG Trunk Bundle Group
01 911 32
02 112 01
03 01
04 01
05 01
06 01
07 01
08 01
09 01
10 01
If you selected Australia as Region, the default Emergency Number Table would look like this:
Default Emergency Number Table for Australia
Index Emergency Numbers OG Trunk Bundle Group
01 000 32
02 106 30
03 112 31
04 01
05 01
06 01
07 01

Index Emergency Numbers OG Trunk Bundle Group
08 01
09 01
10 01
Each Number is stored at an index from 01 to 10. The Emergency Number fields from index 01 to 5 are non-
editable, but you can select a different OGTBG for each of these default Emergency Numbers.
You can add Emergency Numbers at index 06 to 10 in the table, and select the OGTBG as required.
Configuring Emergency Number Table using Jeeves

• After you have selected the Region.
• Under Configuration, click Emergency.
• Click Emergency Number to open the page.
Click the Default button, the system will upload the default Emergency Numbers as per the Region you
have selected.
The first five entries, at Index 01 to 05 on this table, are uneditable. These fields will be populated with the
default Emergency Numbers of your country (which you selected as Region).
If the Emergency Numbers loaded by default are applicable for your region/country, all you need to do is
re-assign, if required, the Outgoing Trunk Bundle Group assigned by default to each number.

If the Emergency Numbers loaded by default are not applicable for your region/country, you may add the
Emergency Numbers and their OGTBG at index 06 to 10 in this table.
Make sure that the trunks configured in the OGTBG for each Emergency Number belong to the correct
network and the ports through which the calls are to be routed are not disabled. For example, '112' is the
default Emergency Number for the mobile network. So, make sure that the Mobile Trunk to be used for
dialing this number is included in the OGTBG you assign to this number in the Table.
Configuring Emergency Number Table using a Telephone

• Enter SE mode from DKP/SLT.
To add Emergency Numbers in the table, dial:

• 3116-Index-Emergency Number-#*
Where,
Index is the location at which the Emergency Number is stored in the table, from 06 to 10 (01 to 05 are
reserved for default numbers).
Emergency Number is a string of 8 digits max. Use digits 0-9 only.
To assign an OG Trunk Bundle Group (OTBG) to an emergency number, dial:

• 3117-Index-OG Trunk Bundle Group
Where,
Index is the location at which the Emergency Number is stored in the table, from 01 to 10.
• Exit SE mode.

CHAPTER 10 Configuring Voice Mail
System
Before you begin configuration of the VMS related parameters, consider the following points:
• Make sure you have installed the VMS module correctly and observed the Reset Cycle. Refer the topic
“Installing the VMS Module” for instructions on installing the VMS module on the CPU Card. Also refer the
topic “VMS Channels” for license related information.
• The number of VMS channels will be considered as the maximum channels available for reservation for
Voice Mail Auto-Attendant parameter.
• The channel reserved for Voice Mail Auto-Attendant configuration will not be changed if the number of
channels configured in Channel Reserved for Voice Mail Auto Attendant parameter is increased.
• The channels reserved for Voice Mail Auto-Attendant configuration will be changed to the number of
VMS channels if the number of VMS channels available are less than the number of channels
configured in Channel Reserved for Voice Mail Auto Attendant parameter. To know more, refer
“Configuring VMS General Parameters”.
• Decide which extension users are to be provided voice mail. Make a list of these extensions by their port
type and (software) port number, access codes. Configure the VMS parameters for these extensions. See
“Extension Voice Mail Settings”.
• If you intend to use the VMS Auto Attendant on trunks,

• make a list of the trunks by their port type (SIP, T1E1 PRI, BRI, Mobile, CO and E&M) and (software)
port number.
• configure welcome and greeting messages. You may record custom welcome messages that meet
your requirements. Refer to “Voice Mail Auto-Attendant Menu”.
• configure Voice Mail Auto Attendant (VMAA) Menu for the time zones in the Trunk Feature Template(s)
of the trunks on which you want to use the VMS Auto Attendant. Refer to “Trunk Feature Template”.
• assign the desired Voice Mail Auto Attendant (VMAA) Menu to the trunks in their Trunk Feature
Template.
• The prompts used to route the call using the Voice Mail Auto Attendant can be customized as per your
requirement. Refer to “Prompts Management”.

Configuring VMS General Parameters
VMS General Parameters allows you to customize VMS settings as per your requirement. You can:
• configure the VMS General Settings
• set the default language
• configure the Play Settings
• define the Filename format
• customize Voice Mail Notification Messages
Configuring VMS General Parameters using Jeeves

• Under Configuration, click VMS Configuration.
• Click VMS General Parameters.
• Check the default values of the following parameters, and change them, if required, to the desired values.
General Settings
• Enable Extension Number Validation: The VMS Auto Attendant allows callers to directly reach the
desired party in an organization, by giving them the option of dialing the extension number.
When the Extension Number Validation flag is enabled, the VMS compares the extension number dialed
by the external caller with the extension numbers configured in the system. If no match is found, the VMS
responds with a message “Invalid Number”.
By default, Extension Number Validation is enabled when SARVAM UCS is operating in the Enterprise
mode. This flag is disabled when SARVAM UCS is operating in the Hospitality mode.

• Use SMTP Account: Select the SMTP Account you wish to use for VMS Notifications.
The SMTP Account you configure will be used for all VMS Email Notifications — Message Notifications
and Memory Usage Notifications.
You may add a new SMTP Account. To do so,

• Select Add New option for Use SMTP Account.
• Click Settings to configure the parameters of the New SMTP Account you created. For more
information, see “SMTP Settings”.
If you select None as the option, the system will not send any VMS Notifications — Memory Usage
Notification to SE,VMS E-Mail Notification, Message Wait Notification via E-Mail — even if you have
enabled and configured the respective notification.
• Memory Usage Notification to SE189: The VMS allows all the memory related notifications — VMS
memory usage and mailbox memory usage — to be sent to the System Engineer via email. Enable this
flag, if you want memory related notifications to be sent via email to the System Engineer.
You may customize the Notification Messages as per your requirement. For details, see “VMS E-Mail
Notification”.
You must also specify the email address to which the notifications are to be sent in SE Email ID.
• SE Email ID: Enter the email address on which the notifications should be sent to the System Engineer.
The System Engineer must have access to this email ID. The email ID may consist of a maximum of 64
characters.
• Save Call Taping Files in: When Call Taping feature is enabled on your extension, you can save the Call
Taping files either in the Common Mailbox or in your Personal Mailbox. By default, Call Taping files are
saved in the Common Mailbox.
• Common Mailbox for Call Taping (Enter Extension Number): If you choose to save Call Taping files in
Common Mailbox, you must specify the Access Code of any SLT, DKP, SIP Extension, Department Group
or General Mailbox, whose mailbox you want to assign for Call Taping.
If Call Taping files are saved in Common Mailbox, only the extension users who have access to the
Common Mailbox will be able to retrieve and listen to the recorded conversations.
• Make Message Notification calls using TAC: Select the Trunk Access Code to be used by the system to
make outgoing notification calls to external numbers.
• Channel Reserved for Voice Mail Auto Attendant: Select the number of channels of the VMS that you
wish to reserve for the Voice Mail Auto Attendant. These channels will be used to answer incoming calls
landing on Voice Mail Auto Attendant enabled Trunks only. These channels even if free will not be
available to extension users to access their Mailbox.
• Date Format: Select the Date Format — DD-MM-YYYY or MM-DD-YYYY. The Date format selected here
will be applicable for all the VMS features, VMS E-Mail Notifications, etc.
189. This parameter will not function if you have selected None as the Use SMTP Account option. You may configure this parameter
for future use.

This Date format is not applicable for Message Profile. The “Date Playback Format” will be applicable
there.
• Time Format: Select the Time Format — 24 Hour or 12 Hour. The Time format selected here will be
applicable for all the VMS features, VMS E-Mail Notifications, etc.
Supported VMS Language

You may add a new language, edit or delete the existing languages.
To add a new language,
• Click the Add Language button.
A window will pop up, asking you the language name.
• Enter the language name in the field. The allowed characters are A to Z, a to z, 0 to 9, - and _.
• Click OK.
To Edit a Language Name,
• Click .
• A window will pop up. You can edit the language name here.
• Click OK.
To delete a language, click .
If the language you wish to delete is already configured for any other parameter, the system will set English
as the default language for that parameter.

Play Settings
• Play ‘#’ as: Select the desired option — Hash or Pound. The system will play the selected option when ‘#’
is pressed.
• Play ‘*’ as: Select the desired option — Star or Asterisk. The system will play the selected option when ‘*’
is pressed.
• For all Options Menu, play digit: Select the desired option — after each Option, before each Option. The
system will play the digit as per the selected option for all Menu options.
• For all Options Menu, for digit dialing play: Select the desired option — Press or Dial. The system will
play the selected option for digit dialing.
Filename Format (.wav)
You can define the Filename format for Mailbox (Personal) and General Mailbox by setting the sequence of the
various parameters — None, Date, Time, Message Type, Calling Number, Called Number and Mailbox Extension
Number190— as per requirement.
While selecting the Filename format, make sure:

• None is not selected as the first two options.
• Date and Time are included in the filename.
• The same option (other than None) is not selected twice.
The table given below defines the parameters. You may sequence it as per your requirement.
Option Meaning
None The system will skip this option and move to the next option configured.
Date Select this option if you want the system to add the Datea in the filename.
Time Select this option if you want the system to add the Timeb in the filename.
Select this option if you want the system to add the Message Typec for the
Message Type
message stored.
190. The Mailbox Extension Number parameter is only applicable for General Mailbox Filename Format.

Option Meaning
Select this option if you want the system to add the calling number. If calling
Calling Number
number is Blank, the system will add ‘No Number’.
Select this option if you want the system to add the called number. If Called
number is not available, system will ignore this option and select the next
Called Number
option. This option is generally required when you wish to use the feature Call
Tapping.
This option is applicable only for General Mailbox. Select this option if you want
Mailbox Extension Number the system to add the mailbox extension number from which the message is
transferred to the General Mailbox.
a. The Date Format here will be as per the Date Format you have selected in General Settings.
b. The Time Format here will be as per the Time Format you have selected in General Settings.
c. The Message Types supported by the system are: Call forward(CF), Call Taping(CT), Conversation Recording(CR), Broadcast
Message(BM), Transfer to Mailbox(TM), Leave Message(LM), Send Message(SM), Call forward with LCS – No reply, all(LS),
Redirect Message(RM), Message forward(MF).
The default sequential Filename Format is as given below:

• For Mailbox(Personal): Date-Time-Message Type-Calling Number-Called Number.wav
• For General Mailbox: Mailbox Extension Number-Date-Time-Message Type-Calling Number-Called
Number.wav
VMS E-Mail Notification
Make sure you have configured the user Email ID for sending the notifications to user. For details, see “Message
Wait Notification via E-Mail” in “Extension Voice Mail Settings”.
To know about the Message Wait Notification feature, see “Email Based Notification”.
The table below displays the default events for which notifications will be sent to the SE. You may customize it as
per your requirement.
Event Message Description
Notification for Memory Usage
VMS Memory consumption alert to SE
[VMS] Warning! VMS USB memory This email will be sent to the SE, when 80%
USB 80% is consumed
usage is 80% consumed of the USB memory has been consumed.

[VMS] Alert! VMS USB memory usage is This email will be sent to the SE, when 100%
USB 100% is consumed
completely consumed of the USB memory has been consumed.
USB consumption is [VMS] VMS USB memory usage is in This email will be sent to SE, when USB
below 75% limit memory consumption is below 70%.
Mailbox consumption alert to SE
This email will be sent to the SE, when 80%

Mailbox of user is 80% [VMS] Warning! Mailbox <ext> is 80% of the mailbox memory has been consumed,
consumed consumed along with the Extension Number (if
configured).
This email will be sent to the SE, when 100%

Mailbox of user is 100% [VMS] Alert! Mailbox <ext> is completely of the mailbox memory has been consumed,
consumed consumed along with the Extension Number (if
configured).
Mailbox consumption alert to User
This email will be sent to the User, when 80%

Mailbox is 80% [VMS] Warning! Mailbox <ext> is 80% of the personal mailbox memory has been
consumed consumed consumed, along with the Extension Number
(if configured).
This email will be sent to the User, when

[VMS] Alert! Mailbox <ext> is completely
Mailbox is 100% 100% of the personal mailbox memory has
consumed, please delete old messages
consumed been consumed, along with the Extension
to allow storing of new messages
Number (if configured).
This email will be sent to the User, when 80%

General Mailbox is 80% [VMS] Warning! General Mailbox <ext> of the general mailbox memory has been
consumed is 80% consumed consumed, along with the Extension Number
(if configured).
[VMS] Alert! General Mailbox <ext> is This email will be sent to the User, when
General Mailbox is 100% completely consumed, please delete old 100% of the general mailbox memory has
consumed messages to allow storing of new been consumed, along with the Extension
messages Number (if configured).
Notification for New Message in Mailbox
This email will be sent to user’s personal

[VMS] <msg_type> Message received mailbox, when a message is received along
Normal Message
from <cli> with the Message Type and the Caller
Numbera.
This email will be sent to user’s personal

Conversation Recording [VMS] Message received for
mailbox, when the conversation between the
Message conversation with <num1>
extension user and the caller was recorded.
[VMS] Message received for call This email will be sent to user’s personal
Call Tapping Message recording between <num1> and mailbox, when the conversation between two
<num2> Numbers was recorded.
Notification for New Message in General Mailbox

This mail will be sent to the SE Email ID

configured in the General Settings, when a
[VMS] <msg_type> Message received
New Message message with the message typeb is received
for <ext> from <cli>
for an Extension Number (if configured) from
the Caller Number.
This mail will be sent to the SE Email ID

configured in the General Settings, when a
Conversation Recording [VMS] Message received for <ext> and
message is received for an Extension
Message conversation was with <num1>
Number (if configured) and conversation
was with the Number.
a. If Caller Number is unavailable, Caller Name will be displayed. If both are unavailable, ‘Unknown’ will be displayed.
The CLI Number can be an External Caller Number or an Internal Caller Number.
b. Message Type may be Normal or Urgent.

General Mailbox Settings
A General Mailbox is a common mailbox in the VMS, with which more than one extension users are associated.
When the personal mailbox of any extension user is full, all new messages are diverted to the General Mailbox.
To access the General Mailbox, the extension users must have the feature General Mailbox enabled in their CoS.
The extension users can listen to the messages in the General Mailbox, by dialing the General Mailbox access
code (programmable; default: 1176).
You can change the default settings of General Mailbox parameters using Jeeves.
Configuring General Mailbox Parameters using Jeeves

• Click General Mailbox Settings.
• Check the default values of the following parameters, and change them, if required, to the desired values:
• Access Code: By default, 1176 is the access code for the General Mailbox. Extension users must dial
this number, if they want to access the General Mailbox.
If required, you may assign a different access code to the General Mailbox. The Access Code you
assign may consist of a maximum of 16 digits. Digits 0-9, # and * are allowed.
• Name: You can assign a Name to the General Mailbox. The name you assign can be a maximum of 18
characters. The Name must not have space as it’s first character.
<, >, :, ", /, \, |, ? and * characters are not allowed.
• Mailbox Password: If you have selected the Ask Password to access Mailbox check box in
Extension Voice Mail Settings, extensions users can access the General Mailbox by dialing the default
Password-1111.
To avoid unauthorized access, we recommend you to change the default password. The password you
assign may consist of a maximum of 4 digits. Valid Range: 0000 to 9999. Make sure the new password is
strong and is provided to the extension users who need to access the General Mailbox only.

• Voicemail Settings: Click on this link to configure the Voicemail Settings for the General Mailbox. The
Extension Voicemail Settings window opens.
By default, the General Mailbox with access code-1176 will open. For information regarding the
configuration, see “Extension Voice Mail Settings”.
• Click Submit to save the settings.
For more information and instructions on how to access the General Mailbox, see “Accessing the General
Mailbox”.

Extension Voice Mail Settings
Extension Voice Mail Settings allows you to configure the various VMS parameters assigned to — an Extension, a
Department Group, an Operator or a General Mailbox.
Configuring Extension Voice Mail Settings

• Click Extension Voice Mail Settings.
• Select Number: You may select — Extension, Department Group, Operator or General Mailbox.
• For Extension, enter the Extension Number/Name and then select the same from the drop
down list.
• For Department Group, select the desired Department Group Number from the drop down list.
• For Operator, select the desired Operator from the drop down list.
• Click OK.
You may now configure the respective VMS parameters.

• Access Code: The Access Code of the respective Extension, Department Group or General Mailbox is
displayed as a status.
• Abbreviated Name: You may configure the abbreviated name for the respective Extension or
Department Group. It must be a minimum of 3 alphabetic characters and a maximum of 8 alphabetic
characters.
Abbreviated Name is applicable when you use Dial by Name feature to transfer the call to any
extension using VMAA Menu. For details, refer “Dial By Name”.
• Language: Select the Language which you want the system to use when accessing the Personal
Mailbox. The languages displayed as options in the drop down list is as per the Supported VMS
Language you configure in VMS General Settings.
All the prompts related to Personal Mailbox will be played in the language you select here.
• Department Group Mailbox: Select the Department Group Mailbox Number which you want to assign
to the respective Extension. You may select None or the Department Group Mailbox Number. The
system will allow the access to the department group mailbox you select here.
If you select None, the department group mailbox will not be accessible by the Extension even if the
Extension is a member of the Department Group.
• VMAA Menu: Select the VMAA Menu Number which you want to assign to the respective Extension or
Department Group. For more information, see “Voice Mail Auto-Attendant Menu”.
• Personal Mailbox: Keep the check box enabled if you want to assign the Personal Mailbox to the
respective Extension, Department Group or General Mailbox. It allows you to access your Personal
Mailbox.

For Extension and Department Group:
If you disable the check box, the Personal Mailbox will not be assigned and the system will not allow
you to access your Personal Mailbox.
For General Mailbox:

If you disable the check box, the General Mailbox will not be assigned and the system will not allow you
to access the General Mailbox.
• Mailbox Number: The Mailbox Number is displayed as a status when the Personal Mailbox is
assigned to the respective Extension, Department Group or General Mailbox, if you have enabled the
Personal Mailbox check box. If Personal Mailbox is not assigned, the Mailbox Number will be displayed
as Blank.
Mailbox191
• In Mailbox Size (min), configure the maximum allowed size for message storage. You may configure from
1 to 60000 minutes.
• In Maximum Message Length (sec), configure the maximum time for which a message can be recorded
by the caller. You may configure from 1 to 3600 seconds.
In case the Maximum Message Length is more than the Mailbox Size, the Mailbox size will be considered
for recording.
• In New Message Delivery Option in Mailbox Full Condition, select an option for delivering the New
Message when your Personal Mailbox is full.
You can select from any of the options described below:
• Do not offer to leave message, if you do not want the system to allow any new message to be
delivered when the Personal Mailbox is full.
• Deliver to General Mailbox, if you want the system to deliver the new message to the General
Mailbox when Personal Mailbox is full.
• Overwrite Old Messages, if you want the system to delete the old messages and allow the new
message to be stored when the Personal Mailbox is full.
191. Not applicable when you select Operator as the “Select Number” option.

In case, the recorded message size is greater than the old message that is to be overwritten then the
recorded message wll not be delivered. A prompt will be played for the same. Note: The old message
will be deleted.
The prompt will not be played in case of multiple recipients.
If the mailbox has no old messages, the recorded message will not be stored.
• Overwrite New Messages, if you want the system to delete the new messages to allow the new
message to be stored when the Personal Mailbox is full.
The number of messages that will bedeleted would be as per the Maximum Message Length allowed.
deliver the new message to the General Mailbox when the Personal Mailbox is full.
If the mailbox has no new messages, the recorded message will not be stored.
• Overwrite All (Old + New), if you want the system to delete the old and new messages and allow the
new message to be stored when the Personal Mailbox is full.
• In Auto Delete Message, select an option to delete the messages automatically by the system.You may
select None, Old or All.
• Select None if you do not want the system to delete the message automatically from the Personal
Mailbox.
• Select Old if you want the system to delete the old read messages automatically after the number of
days you configure in Days for Auto Delete Messages.
• Select All if you want the system to delete all messages — read or unread — automatically after the
number of days you configure in Days for Auto Delete Messages.
• In Days for Auto Delete Messages, configure the number of days after which you want the system
to automatically delete the messages.
• In Ask Password to Access Mailbox,by default the access to the mailbox is password protected. The
“User Password” is required to access the mailbox. Whenever the mailbox owner accesses the mailbox,
the VMS will ask for the (user) password.
If you want to remove password protection, clear this check box.
Since a Mailbox can be accessed using the default User Password, 1111, extension users who are
assigned a mailbox are recommended to change their User Password. To avoid unauthorized access, we
recommend extension users to change the password regularly. Make sure it is strong and is kept
confidential.
• In Message Profile, select the Message Profile you want to assign. For more information, see “Message
Profile”.
• In Mailbox Menu, select the Mailbox Menu you want to assign. For more information, see “Mailbox Menu”.

Call Transfer Settings192
• Under Working Hour - WH, select the Call Transfer Profile you want system to use for transferring the
calls during Working Hours. The drop down list will include the profiles you configure in the “Call Transfer
Profile”.
• Under Break Hour - BH, select the Call Transfer Profile you want system to use for transferring the calls
during Break Hours. The drop down list will include the profiles you configure in the “Call Transfer Profile”.
• Under Non-Working Hour - NH, select the Call Transfer Profile you want system to use for transferring
the calls during Non-Working Hours. The drop down list will include the profiles you configure in the “Call
Transfer Profile”.
Message Wait Settings193
• In Message Wait Indication, select the type of indication to be given to the extension user for new
messages in the mailbox and message wait set by another extension user. This is only applicable when
you select Extension as the “Select Number” option.
You can select from any of the four types of indicators described below for new messages:
• Stuttered Dial Tone/Voice Message: When the extension user goes OFF-Hook, s/he will hear a voice
message, if a pre-recorded Voice Module has been assigned for Message Wait Notification. If no voice
module is recorded and assigned, the extension user will hear a stuttered dial tone instead.
192. Not applicable when you select General Mailbox as the “Select Number” option.

If you want voice message to be played as message wait notification, record and assign a Voice
Module. Refer “Voice Message Applications” for instructions.
SARVAM UCS can play only 9 Voice Modules simultaneously. The Voice Module for Message Wait
Notification will not be played if there are already 9 being played simultaneously. In this case, Stuttered Dial
Tone will be played as Message Wait Indication, when the extension user goes OFF-Hook.
• Ring: The extension will ring for the duration of the Message Wait Ring Timer (configurable; default: 30
seconds), for as many times as the Message Wait Ring Count (configurable; default: 10 times), at the
interval set as the Message Wait Ring Timer Interval (configurable; default: 30 minutes).
When the extension user answers the call, the VMS informs the user of the new message and allows
the extension user to access it.
• Stuttered Dial Tone + LED Lamp (High Voltage): When the extension user goes OFF-Hook, the
extension user will hear a stuttered dial tone and if the SLT has a 'Message Wait' lamp, the lamp will
blink continuously using High Voltage. When the extension user retrieves all the waiting messages, the
LED will be turned off and the stuttered dial tone will stop.
• Stuttered Dial Tone + LED Lamp (Polarity Reversal): When the extension user goes OFF-Hook, the
extension user will hear a stuttered dial tone and if the SLT has a 'Message Wait' lamp, the lamp will
blink continuously using Polarity Reversal. When the extension user retrieves all the waiting messages,
the LED will be turned off and the stuttered dial tone will stop.
• LED Lamp (High Voltage): If the SLT has a 'Message Wait' lamp, it will blink continuously using High
Voltage. When the extension user retrieves all the waiting messages, the LED will be turned off.
• LED Lamp (Polarity Reversal): If the SLT has a 'Message Wait' lamp, it will blink continuously using
Polarity Reversal. When the extension user retrieves all the waiting messages, the LED will be turned
off.
Default: Stuttered Dial Tone + LED Lamp (High Voltage).
Refer the feature description “Message Wait” to know more.
Message Wait Notification via Call194

The message wait notification will be sent to a number(destination number). This number can be an internal or an
external number.
• In Type, you may select — Immediate, Scheduled or None.
• Select Immediate, if you want the notifications to be sent as soon as a new message arrives in the
mailbox of the extension user.
• Select Scheduled, if you want the notification to be sent at fixed time schedules.
• Select None, if you do not want to set message wait notification via call.
Default: None.
194. Applicable only when you select Extension as the “Select Number” option.

• In Schedule Profile, select the Notification via Call - Profile number according to which you want the
system to send the notifications. The Notification via Call Profile determines how notification calls are to be
made to the destination numbers. To know more, see “Message Wait Notification via Call”.
• In Destination Number, configure the number that you want the system to use for sending the notification
via calls.
The destination number can be an internal or an external number. The destination number can be a
maximum of 16 digits. Valid digits are 0 to 9, # and *.
When the notification call is answered, the VMS informs the callee about the new message and allows the
callee to access it.
Refer the feature description “Message Wait Notification via Call” to know more.
Message Wait Notification via E-Mail195

The message wait notification will be sent to the e-mail address of the extension user.
• In Notification, you may select — Do not send, Without Attachment, With Attachment or With Attachment
and mark voicemail as read. Default: Do not send.
• Select Do not send, if you do not want the system to send email notification to the user for new
message even if the email ID is configured. In this case, the system will send the Mailbox Memory
Usage notification.
• Select Without Attachment, if you want the system to send email notification to the user for new
message if the email ID is configured.
• Select With Attachment, if you want the system to send email notification to the user for new message
along with the message as attachment. Make sure the email ID is configured. The attachment will be
sent only if the message size is less than or equal to 20 MB, else the email notification will be sent
without the attachment.
• Select With Attachment and mark voicemail as read, if you want the system to send email
notification to the user for new message along with the message as attachment and also mark the
voicemail as read. Make sure the email ID is configured. The attachment will be sent only if the
message size is less than or equal to 20 MB, else the email notification will be sent without the
attachment.
• E-mail Address: Enter the email ID of the extension user to which the notification is to be sent. Maximum
allowable length of the email ID is 64 characters. Default: blank.
Extension users will receive notifications only for the mailbox memory utilization, if you configure the E-mail
Address and select Do not sent as the Notification option.
Refer the feature description “Email Based Notification” to know more.
• Click Submit to save Extension Voice Mail settings.
• Click Copy if you wish to copy the respective Extension’s Voice Mail settings to other extensions.

Copy Voice Mail Settings to window will open.
You can copy the Voice Mail Settings to a single Extension, multiple extensions or all extensions.
For single Extension, select Extension Number and enter the Extension Number.
For multiple extensions, select Extension Numbers from and enter the Extension Number range.
For all extensions, select All Extensions.
The Voice Mail Settings — Abbreviated Name, Message Wait Notification via Call parameters and
Message Wait Notification via Email parameters will not be copied.

Voice Mail Auto-Attendant Menu
Voice Mail Auto-Attendant Menu is applicable when trunk call is directly routed to VMS.
Each trunk can be assigned a VMAA Menu. For details, see “Trunk Feature Template”.
If VMAA Menu assigned to any action/trunk is deleted, the system will consider the first VMAA Menu for that action/
trunk by default.
VMS supports a maximum of 128 VMAA Menu.
By default, three VMAA Menus — Working Hour, Break Hour and Non-Working Hour— are provided to you. These
three VMAA menus cannot be deleted but you may edit their settings as per your requirement.
How to Configure
• Click Voice Mail Auto-Attendant Menu.
You may add a new menu, edit the default menus or delete a VMAA menu.
To add a new VMAA Menu,
• Click Add New Menu. A new VMAA Menu xxx will be created. You may now configure this as per your
requirement.

To delete a menu, click Delete.
The default VMAA Menus cannot be deleted.
To edit a VMAA Menu,
• Click on the VMAA Menu tab you wish to edit.
• In Menu Name, configure the name of the VMAA Menu. By default, it is VMAA Menu xxx where xxx is the
VMAA Menu Number from 001 to 128.
• In Access Code, configure the Access Code you wish to assign to the respective VMAA Menu. The caller
can transfer the call to a VMAA Menu by dialing the respective access code. Access Code can be a
Make sure, the Access Codes assigned to the VMAA Menus do not conflict with the existing access codes.
System will not save the configured VMAA access code if the same code is already assigned.
Greetings
Greetings allows you to select the prompts to greet the caller.
• In Morning Prompt, select the prompt which you wish to play to the caller as the Morning Greeting.
Select Do Not Play, if you do not wish to play any prompt.
• Click Settings . The Prompts Management page opens.
You may add a new prompt from here. For further details, see “Prompts Management”.
Once the files are uploaded these appear as options for Morning Prompt.
• In Afternoon Prompt, select the prompt which you wish to play to the caller as the Afternoon Greeting.
Once the files are uploaded these appear as options for Afternoon Prompt.
• In Evening Prompt, select the prompt which you wish to play to the caller as the Evening Greeting.

Once the files are uploaded these appear as options for Evening Prompt.
Language Settings
Language Settings allows you to set a menu language for the respective VMAA Menu and also gives the caller a
choice to select the language.
• In Menu Language, select the language which you wish to set as the default language for the respective
VMAA Menu. For all the calls routed to the VMAA Menu, the prompts and greetings will be played as per
the Menu Language set.
• Select the Language Selection check box if you wish to allow the caller to choose a language.
• In Language Selection Profile, select the language profile you want to assign to the respective VMAA
Menu.
• Click Settings to configure the parameters of the selected profile. For more information, see
“Language Selection Profile”.
From here, the caller can select a different language other than the Menu Language configured. All the
VMS prompts henceforth will be played in the language selected here.

Auto-Attendant Settings
Auto-Attendant Settings allows you to customize the Auto-Attendant Actions.
• In Auto Attendant Prompt, select the prompt which you wish to play to the caller according to the Auto-
Attendant Actions.
Once the files are uploaded these appear as options for Auto Attendant Prompt.
To add more Auto-Attendant Prompts, click . You can add a maximum of 4 Auto Attendant Prompts in
each VMAA Menu.
After playing all Auto-Attendant prompts, the system will wait for the input from the caller as per the timers
configured. For details, see “Timers”.
• Select the Extension Number Dialing check box to allow the caller to dial the Extension Number while
the Auto-Attendant prompts are being played.
• Click Settings to configure the No Match Found Settings.
• In No Match Found Prompt, select the prompt which you wish to play when caller has dialed an
invalid extension number.
Select None, if you do not wish to play any prompt.
You may add a new Prompt. To do so,

Once the files are uploaded these will appear as the options for No Match Found Prompt.
• Select the No Match Found Retry check box to prompt the caller to dial the Extension Number
again.
• In No Match Found Retry Count, select the number of times you wish to prompt the caller for
dialing the extension number. The prompt will be played repeatedly till the Retry Count expires.
• In Retry Count Expiry Prompt, select the prompt you wish to play when the Retry Count
expires.
Once the files are uploaded these appear as options for Retry Count Expiry Prompt.
• In No Match Found Action, you can select — Disconnect, Transfer to Operator, Transfer to
Extension, Transfer to Department Group, Go to VMAA Menu.
• If you select Transfer to Operator or Transfer to Department Group or Transfer to

Extension, select the respective Operator number/Department Group number or enter the
• In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement.
By default, As configured for Transfer Number is selected.
Click Settings to configure the parameters of the selected profile. For more
information, see “Call Transfer Profile”.
• If you select Go to VMAA Menu, select the desired VMAA Menu.
• If you select Disconnect, the system will disconnect the call.
• In Call Transfer Profile, select the desired Call Transfer Profile you want the system to use. Make
sure you have configured the parameters for this profile. By default, As configured for Dialed
Extension is selected.
• Select As configured for Dialed Extension if you want the VMS Auto Attendant to use the
Call Transfer Profile which is assigned to the dialed extension.
• Select Wait for Ring if you want the VMS Auto Attendant to wait for the extension to start
ringing and then transfer the call.
• Select Blind if you want the VMS Auto Attendant to transfer the call on the extension without
checking whether it is busy or free.

• Select Attended if you want the VMS Auto Attendant to transfer the call when the extension
answers (goes OFF-Hook).
Click Settings to configure the parameters of the selected profile. For more information,
see “Call Transfer Profile”.
When the Call Transfer Profiles are added, or any existing profile is edited, these changes will be reflected
in the options displayed.
Auto-Attendant Actions
The default Auto-Attendant actions taken by the system, when there is an incoming trunk call on the VMS are
displayed here.
You may add a new action, edit or delete the default actions.
To delete an Auto-Attendant action, mouse over on the respective action and click .
To add/edit an Auto-Attendant action,
• Click on the Add Action button to add a new action or click to edit an action.
• In Dial Digit, select the digit to be dialed for the respective action.
• In Action to Digit Dialed, you can select any one of the following:
• Transfer to Operator
• Transfer to Extension
• Transfer to Department Group
• Go to VMAA Menu
• Play Information
• Dial Extension Number by Name
• Leave Voice Mail
• Personal Mailbox Access
• Change Language
• Repeat Prompt
• Go to Previous Menu
• Disconnect

• If you select Transfer to Operator or Transfer to Extension or Transfer to Department Group,
select the respective Operator number or desired extension number or Department Group number.
• In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement. By
default, As configured for Transfer Number is selected.
• If you select Play Information, click Settings to configure the parameters. For details, see “Play
Information”.
• If you select Dial Extension Number by Name, the system will allow caller to dial any extension
number using name. Click Settings to configure the parameters. For details, see “Dial by Name”.
• If you select Leave Voice Mail, the system will allow caller to leave the message directly to the
extension number or department group. Click Settings to configure the parameters. For details,
see “Leave Voice Mail”.
• If you select Personal Mailbox Access, the system will provide the mailbox access to any user from
remote location. Click Settings to configure the parameters. For details, see “Mailbox Access”.
• If you select Change Language, select the Language which you want the system to use for the VMAA
Menu instead of the previously selected Language.
Click Settings to configure the parameters of the selected language profile. For more information,
see “Language Selection Profile”.
All the VMS prompts henceforth will be played in the language selected here.
• If you select Repeat prompt, the system will repeat all the Auto-Attendant prompts configured in
sequence.
• If you select Go to Previous Menu, the system will provide an option to the caller to go back to the
previous VMAA Menu.
• Select the Ignore Digit Dialed during Prompt check box, if you want the system to ignore the digits
dialed by the caller while the prompts are being played.
Play Information

• In Information Prompt, select the prompt which has the information to be played to the caller. The
information may be about the company or about the different products etc.
Select None, if you do not wish to play any information prompt to the caller.
Once the files are uploaded these appear as options for Information Prompt.
• Select the Information Prompt Repeat check box to allow the Information prompt to be repeated to the
caller.
• In Information Prompt Repeat Count, select the number of times you wish to play information prompt to
the caller. The prompt will be played repeatedly till the Repeat Count expires.
• In Repeat Count Expiry Prompt, select the prompt you wish to play when the Information Prompt Repeat
Count expires.
Once the files are uploaded these appear as options for Repeat Count Expiry Prompt.
• In Information Prompt Playover Action, select — Transfer to Operator, Transfer to Department Group,
Transfer to Extension, go to VMAA Menu or Disconnect.
• If you select Transfer to Operator or Transfer to Department Group or Transfer to Extension,

select the respective Operator number/Department Group number or enter the desired extension
number.
Click Settings to configure the parameters of the selected profile. For more information, see “Call
• Click Close to close the window.

Dial by Name
Basic Settings
• In Dial by Name Prompt, select the prompt which you wish to play to the caller allowing him/her to dial the
extension number using name.
Once the files are uploaded these appear as options for Dial by Name Prompt.
• In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement. By default, As
configured for Transfer Number is selected.
• Select the Play Dialed/Selected Name check box, if you want the system to play the name dialed by the
caller and check the option Dialed/Selected Name Confirmation.
If you disable the check box, the system will transfer the call directly to the extension number dialed by the
caller as per the Call Transfer Profile configured.

• Select the Dialed/Selected Name Confirmation check box, if you want the system to confirm with the
caller before transferring the call to the extension number dialed.
• In Confirm, assign the Digit (0-9, *, # or None) that you want the system to play to the caller for
confirming the Name dialed before transferring the call. If you assign None, the option will not be
played.
• In Re-enter, assign the Digit (0-9, *, # or None) that you want the system to play to prompt the caller
to dial the extension number again.
• In No Digit Dialed Action, select the option — Confirm or Re-enter. This is applicable when the
caller has not dialed any digit for Selected Name Confirmation. The system will function as per the
option you select.
No Digit Dialed Settings
No Digit Dialed Settings are applicable when caller has not dialed any digit and the first digit wait timer has expired.
• In No Digit Dialed Prompt, select the prompt which you wish to play when caller has not dialed any digit.
Select None, if you do not wish to play any prompt when caller has not dialed any digit.
Once the files are uploaded these appear as options for No Digit Dialed Prompt.
• Select the No Digit Dialed Retry check box to prompt the caller to dial the digit again.
• In No Digit Dialed Retry Count, select the number of times you wish to prompt the caller for dialing the
digit. The prompt will be played repeatedly till the Retry Count expires.
• In Retry Count Expiry Prompt, select the prompt you wish to play when the Retry Count expires.

• In No Digit Dialed Action, you can select — Disconnect, Transfer to Operator, Transfer to Extension,
Transfer to Department Group, Go to VMAA Menu.

number.
In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement. By default,
As Configured for Transfer Number is selected.
No Match Found Settings
• In No Match Found Prompt, select the prompt which you wish to play when caller has dialed an invalid
extension number.
Once the files are uploaded these appear as options for No Match Found Prompt.
• Select the No Match Found Retry check box to prompt the caller to dial the Extension Number again.
• In No Match Found Retry Count, select the number of times you wish to prompt the caller for dialing the
extension number. The prompt will be played repeatedly till the Retry Count expires.

• In No Match Found Action, you can select — Disconnect, Transfer to Operator, Transfer to Extension,

number.
As Configured for Transfer Number is selected
Multiple Match Found Settings
Multiple Match Found Options
You can assign the Digits (0-9, *, # or None) to each of the options — Select Name, Next Name, Repeat Name,
Repeat from First Name, Go to Dial by Name, Go to Previous Menu or Disconnect.

Make sure you do not assign the same digit to multiple options.
The digit you configure will be played as the digit to be dialed by the caller to select the respective option. The
options will be played in the sequence — 1-9, 0, *, #. The option for which the Digit selected is None will not be
played.
After the digits have been assigned, the system will proceed further as per the options selected by you. These
options will be played for each of the Names included in the Multiple Match Found list.
Brief description of each option is explained below:
• Select Name: The system will play a Name from the multiple match found list. The caller may select the
this name for if he wishes to transfer the call to this Extension.
• Next Name: The system will play the next name from the multiple match found list.
• Repeat Name: The system will re-play the last name played from the multiple match found list.
• Repeat from First Name: The system will play the multiple match found list of names from the first
matched name.
• Go to Dial by Name: The system will clear the multiple match found list and prompt caller to dial the
Extension by Name again.
• Go to Previous Menu: The system will provide an option to the caller to go back to the Previous Menu.
• Disconnect: The system will disconnect the call after playing the Disconnect prompt.
In Select Name - Digit Wait Timer (sec), enter the time for which you want the system to wait to play the next
name while playing the multiple match found list. Default: 03 seconds
No Name Selected Settings
• In No Name Selected Prompt, select the prompt you wish to play when no name is selected by the caller.
Once the files are uploaded these appear as options for No Name Selected Prompt.
• In No Name Selected Action, you can select — Transfer to Operator, Transfer to Extension, Transfer to
Department Group, Go to VMAA Menu, Go to Previous Menu or Disconnect.

number.

• If you select Go to Previous Menu, the system will go back to the previous menu.
Invalid Digit Dialed Settings
• Clear the Ignore Invalid Digit Dialed check box, if you do not want the system to ignore the invalid digit
dialed and prompt the caller for the same. By default, it is enabled.
• In Invalid Digit Dialed Prompt, select the prompt you wish to play when the caller has dialed an
invalid digit for option selection.
Select None, if you do not wish to play any prompt when an invalid digit is dialed.
Once the files are uploaded these appear as options for Invalid Digit Dialed Prompt.
• In Invalid Digit Dialed Action, you can select — Transfer to Operator, Transfer to Extension, Transfer to
Department Group, Go to VMAA Menu, Go to Previous menu or Disconnect.

number.
• If you select Go to Previous Menu, the system will go back to the previous menu.

Leave Voice Mail
• In Leave Voice Mail to, select an option where you want to leave the voice mail message.
• If you select Extension Number or Department Group, the system will prompt the caller to leave the
message for the Extension Number or Department Group configured.
• If you select Extension number dialed by caller, the system will prompt the caller to enter the number
to leave the message for the Extension Number or Department Group.
Click Settings to configure the Leave Voice Mail - Dialed Extension Number parameters.
• In Leave Voice Mail Prompt, select the prompt you wish to play to prompt the caller to leave the
message.
Once the files are uploaded these appear as options for Leave Voice Mail Prompt.
• For No Match Found Settings, refer “No Match Found Settings”.
• For No Digit Dialed Settings, refer “No Digit Dialed Settings”.
• Select the Override Message Leave Settings of Extension check box, if you want the system to apply
the Leave Voice Mail Settings configured on this page. Configure the “Message Leave Settings” and “No
Mailbox Action”.
Clear the check box, if you want the system to apply the Leave Voice Mail settings configured for the
extension.
• In Action after Leaving Voice Mail, you can select — Disconnect, Transfer to Operator, Transfer to

number.

Click Settings to configure the parameters of the selected profile. For more information, see
“Call Transfer Profile”.
Message Leave Settings
Message Leave Settings is applicable if you have enabled Overide Message Leave Settings of Extension
parameter. It allows you to configure the settings for the messages you receive from the caller i.e the message left
for you by the caller.
• Select the Play Personal Greeting check box to allow the personal greetings to be played to the caller.
If the personal greeting is not recorded or is unavailable, no prompt will be played.
Personal Greetings will not be applicable if your mailbox is defined as the destination for Call Tapping or
Conversation Recording messages.
• Select the Play Conditional Greeting check box to allow the conditional greetings to be played to the
caller.
If the conditional greeting is not recorded or is unavailable, no prompt will be played.
Conditional Greeting will be applicable only if your mailbox is defined as the destination for Call Transfer
Unsuccessful.

• In Stop Record Message Code, enter the digits (0-9,* or #) that you want the caller to dial to stop the
recording of the message. The system will play this to the caller along with the prompt before he starts
recording the message. You may configure a maximum of upto 3 digits.
If you do not assign any digit and keep it blank, the system will play the prompt without the stop code.
• Select the Message Verification check box to allow the recorded message to be verified by the caller
before storing it in the Personal Mailbox or re-record it. You must configure the “Message Leave Options”.
If you disable the check box, the system will directly store the recorded message in the Personal Mailbox.
• In Message Type, select the option according to which you want the system to store the recorded
messages. You may select — Set as Normal, Set as Urgent or Ask caller.
• Select Set as Normal, if you want the system to store the recorded message in the mailbox as Normal.
• Select Set as Urgent, if you want the system to store the recorded message in the mailbox as Urgent.
• Select Ask Caller, if you want the system to ask the caller to select the message type before storing
the recorded message in the mailbox.
• In Message Sensitivity, select the option according to which you want the system to prioritize the
messages as per their importance. You may select — Set as Normal, Set as Private or Ask caller.
• Select Set as Private, if you want the system to store the recorded message in the mailbox as Private.
The message received is considered as confidential and forwarding the message is restricted.
• Select Ask Caller, if you want the system to ask the caller to select the message sensitivity before
storing the recorded message in the mailbox.
• Select the Message Security check box, if you want the system to restrict the forwarding and
downloading of the messages.
Message Security has higher priority than Message Type and Message Sensitivity set for the message.
If you disable the check box, the system will not set any security for the message.
• Select the Message Leave Confirmation Prompt check box if you want the system to play the Message
Leave Confirmation prompt to the caller after leaving the message. The Message Leave Confirmation
prompt plays the Message Type, Message Sensitivity and Message Security that you configured.
If you disable the check box, the system will not play the Message Leave Confirmation prompt to the caller
after leaving the message.
Message Leave Options
You can assign the Digits (0-9, *, # or None) to each of the options — Re-record, Confirm, Urgent,
Private(Confidential), Listen Recorded Message and Append to Recorded Message.

played.
After the digits have been assigned, the system will proceed further as per the options selected by you. Given
below is the brief description of each option:
• Re-record: The system will clear the recorded message and let the caller record a new message again.
This is applicable only if you have enabled the Message Verification check box.
• Confirm: The system will save the recorded message as Normal message. This is applicable only if you
have enabled the Message Verification check box or selected the Ask Caller as the Message Type and/or
Message Sensitivity option.
• Urgent: The system will save the recorded message along with the Message Type as Urgent. This is
applicable only if you have selected the Ask Caller as the Message Type option.
• Private(Confidential): The system will save the recorded message along with the Message Sensitivity as
Private. This is applicable only if you have selected the Ask Caller as the Message Sensitivity option.
• Listen Recorded Message: The system will play the recorded message to the caller and then play the
message leave options again. This is applicable only if you have enabled the Message Verification check
box.
• Append to Recorded Message: The system will let the caller record a new message again and add it to
the existing recorded message. This is applicable only if you have enabled the Message Verification check
box.
System will check — Maximum Message Length, New Message Delivery Option in Mailbox Full Condition
etc — before allowing the caller to append the recorded message
The Message, that is, the recorded message along with the appended message cannot exceed the
Maximum Message Length. Hence, the system will allow recording of the appending message only till it
reaches the Maximum Message Length.
• In No Digit Dialed Action, if no digit is dialed by the caller, the system will take action as per the Message
Type and Message Sensitivity configured.
If you have configured Ask Caller option in Message Type and/or Message Sensitivity, you can select the
desired action — Normal+Normal, Urgent+Normal, Normal+Private or Urgent+Private.
No Mailbox Action
No Mailbox Action is applicable if you have enabled Overide Message Leave Settings of Extension parameter.
• In No Mailbox Action, you can select — Disconnect, Transfer to Operator, Transfer to Extension, Transfer
to Department Group, Go to VMAA Menu.

number.

Click Settings to configure the parameters of the selected profile. For more information, see
“Call Transfer Profile”.
Mailbox Access
Settings
• In Mailbox Access Prompt, select the prompt you wish to play when the caller has selected Personal
Mailbox Access as the Action on Digit Dialed option.
Once the files are uploaded these appear as options for Mailbox Access Prompt.
• For details, refer “No Match Found Settings”.
• For details, refer “No Digit Dialed Settings”.

• For details, refer “No Digit Dialed Settings”.

• For details, refer “Invalid Digit Dialed Settings”.

Timers
• In First Digit Wait Timer, enter the time for which you want the system to wait for the caller to dial the first
digit for menu option selection or for entering the Extension or Department Group number. On the expiry of
the First Digit Wait Timer, the system will apply “No Digit Dialed Settings”. Default: 07 seconds
• In Inter Digit Wait Timer, enter the time for which you want the system to wait for the caller to dial the
subsequent digit after the first digit is dialed. On the expiry of the Inter Digit Wait Timer, the system will
apply “No Match Found Settings”. Default: 05 seconds
• In Conflict Wait Timer, enter the time for which you want the system to wait for the extension user to dial
the next digit to resolve conflicting access codes dialed by the extension user. Default: 03 seconds
Disconnect
• In Disconnect Prompt, select the prompt you wish to play when the caller has selected Disconnect as the
Action on Digit Dialed option.
Once the files are uploaded these appear as options for Disconnect Prompt.

Language Selection Profile
Language Selection Profile is a menu which allows you to select the language. The languages you configure in the
“Supported VMS Language” will be displayed under Language selection options. You may add a maximum of 8
profiles.
If Language Selection Profile assigned to any parameter is deleted, the system will consider the first Language
Selection Profile.
How to Configure
• Click the Language Selection Profile.
You may add a new profile, edit the default profile or delete a Language Profile.
To add a new Language Profile,
• Click Add New Profile. A new Language Profile x will be created. You may now configure this as per your
requirement.
To delete a profile, click Delete.
The default Language Profile cannot be deleted.
To edit a Language Profile,
• In Profile Name, you may assign a name to the Language profile. By default, it is Language Selection x,
where x is the Language Profile Number from 1 to 8.

Language Selection Options
The languages that are added in the system are displayed here. For details, see “Configuring VMS General
Parameters”.
You can assign the Digits(0-9, *, # or None) to each of the Languages.
The digit you configure will be played as the digit to be dialed by the caller to select the respective language. The
options will be played in the sequence — 1-9, 0, *, # — as per the Digits configured. The language for which the
Digit selected is None will not be played.
The option you configure for a language will be played in the respective language.
For Example, you selected the Digit 5 to be played for Italian language. The option will be played along with the
digit to dial (as configured) in Italian Language.

No Digit Dialed Settings are applicable when caller has not dialed any digit to select the desired Language and the
first digit wait timer has expired.
• In No Digit Dialed Prompt, select the prompt you wish to play when caller has not dialed any digit for
option selection.
• Select the No Digit Dialed Retry check box to allow the Language selection options to be played again if
no digit has been dialed.
• In No Digit Dialed Retry Count, select the number of times you wish to prompt the caller for language
selection. The language selection options will be played repeatedly till the Retry Count expires.

• In No Digit Dialed Language, select — Use Language configured in Menu, Disconnect, or the language
configured as Supported VMS Language.
• Select Use Language configured in Menu, if you want the system to use the Language you
configured as Menu Language in VMAA Menu.
• Select Disconnect, if you want the system to disconnect the call.
If you select a language from the Supported VMS Languages you configured, the system will use the
Language you select. See “Supported VMS Language”.

Invalid Digit Dialed Settings are applicable when caller has dialed an invalid digit — a digit which is not configured
as the Language Selection option.
• In Invalid Digit Dialed Prompt, select the prompt you wish to play when the caller has dialed an
invalid digit for option selection.
Select None, if you do not wish to play any prompt when an invalid digit is dialed.
• Select the Invalid Digit Dialed Retry check box to allow the Language selection options to be played
again if an invalid digit has been dialed.
• In Invalid Digit Dialed Retry Count, select the number of times you wish to prompt the caller for
language selection. The language selection options will be played repeatedly till the Retry Count
expires.

• In Invalid Digit Dialed Language, select — Use Language configured in Menu, Disconnect, or the
language configured as Supported VMS Language.
• Select Use Language configured in Menu, if you want the system to use the Language you
configured as Menu Language in VMAA Menu.
• Select a language from the Supported VMS Languages you configured, if you want the system to
use the Language you select. See “Supported VMS Language”.

Message Profile
Message Profile allows you to customize the Message Leave Settings, Message Playback Settings and Message
Sent/Forward Settings.
VMS supports a maximum of 24 Message Profiles.
By default, two Message Profiles — User and Executive — are provided to you. These two profiles cannot be
deleted but you may edit their settings as per your requirement.
How to Configure
• Click Message Profile.
You may add a new profile, edit the default profile or delete a message profile.
To add a new Message Profile,
• Click Add New Profile. A new Message Profile xx will be created. You may now configure this as per your
requirement.
The default Message Profiles cannot be deleted.

To edit a Message Profile,
• In Profile Name, you may configure the name of the Message profile you want. By default, it is Message
Profile xx where xx is the Message Profile Number from 01 to 24.

Message Leave Settings are applicable only when the external caller reaches to your mailbox or internal caller
replies to your message.
The Personal Greetings will be played as per the timezone - Working Hours, Break hours or Non-Working
hours.
caller. The conditional greetings will be played when the call forward is set to VMS for Busy, No Reply or
Unconditional.
If you do not assign any digit and keep it blank, the system will play the prompt without the stop code. In this
case, you may dial any digit to stop recording.
before storing it in the Personal Mailbox or to re-record it. You must configure the “Message Leave
Options”.
• In Message Type, select the option according to which you want the system to store the received
messages as per the priority. You may select — Set as Normal, Set as Urgent or Ask caller.
• Select Set as Normal, if you want the system to store the messages received in the mailbox as
Normal. The messages left by the external callers or replied by the internal callers will be considered as
normal messages.
• Select Set as Urgent, if you want the system to store the messages received in the mailbox as Urgent.
This is useful when you want the messages left by the external callers or replied by the internal callers
to be considered as urgent messages with higher priority.
• Select Ask Caller, if you want the system to ask the caller to select the message type before leaving
the recorded message on your extension. If the call gets disconnected before the caller selects the
message type, the message will be set as normal by default. You must configure the “Message Leave
Options”.
While retrieving, the Urgent and Normal messages will be played separately.

• In Message Sensitivity, select the option according to which the system will decide whether the
messages left by the external callers or replied by the internal callers are allowed to be forwarded or not.
• Select Set as Normal, if you want the system to store the messages received in the mailbox as
Normal. The messages left by the external callers or replied by the internal callers will be considered as
normal messages.
• Select Set as Private, if you want the system to store the messages received in the mailbox as Private.
The message left by the external callers or replied by the internal callers will be considered as
confidential and forwarding the message will be restricted.
storing the recorded message in the mailbox. If the call gets disconnected before the caller selects the
message sensitivity, the message will be set as normal by default. You must configure the “Message
Leave Options”.
• The Message Security parameter is reserved for future use. Keep this check box disabled.
• Select the Message Leave Confirmation Prompt check box if you want the system to play the prompt to
the caller after leaving the message. The Message Type and Message Sensitivity that you configured will
be played.

Message Leave Options will be played based on the configuration of the parameters — Message Verification,
Message Type and Message Sensitivity.

options will be played in the sequence — 1-9, 0, *, #. The option for which the digit selected is None will not be
played.
• Urgent: The system will save the recorded message as Urgent message. This is applicable only if you
have selected the Ask Caller as the Message Type option.
• Private(Confidential): The system will save the recorded message as Private message. This is applicable
only if you have selected the Ask Caller as the Message Sensitivity option.
box.
box.
etc — before allowing the caller to append the recorded message.
This is applicable only if you have enabled the Message Verification check box or selected the Ask Caller
as the Message Type and/or Message Sensitivity option.
Based on the configuration, you may select the desired action — Normal+Normal, Urgent+Normal,
Normal+Private or Urgent+Private.

Message Playback Settings
Message Playback Settings allows you to configure the settings applicable when extension user accesses his/her
Personal Mailbox and plays the messages.
• In Message Playback Direction, select the option according to which you want the system to play the
messages to the extension user. You may select — Play oldest recorded message first or Play new
recorded message first.
• Select Play oldest recorded message first, if you want the system to play the old messages first and
then play the new messages.
• Select Play new recorded message first, if you want the system to play the new messages first and
then play the old messages.
• In New Message Playback, select the option defining the priority according to which you want the system
to play the new messages to the extension user. You may select — Play Normal Messages and thereafter
Urgent Messages or Play Urgent Messages first and thereafter Normal Messages.
• Select Play Normal Messages and thereafter Urgent Messages, if you want the system to first play
the new messages with message type as Normal and then play the new messages with the message
type as Urgent.
• Select Play Urgent Messages first and thereafter Normal Messages, if you want the system to first
play the new messages with message type as Urgent and then play the new messages with the
message type as Normal.
• In Old Message Playback, select the option defining the priority according to which you want the system
to play the old messages to the extension user. You may select — Play Normal Messages and thereafter
Urgent Messages or Play Urgent Messages first and thereafter Normal Messages.
• Select Play Normal Messages and thereafter Urgent Messages, if you want the system to first play
the old messages with message type as Normal and then play the old messages with the message
type as Urgent.
• Select Play Urgent Messages first and thereafter Normal Messages, if you want the system to first
play the old messages with message type as Urgent and then play the old messages with the message
type as Normal.
• Date Playback Format: Select the Date Playback Format — DD-MM-YYYY or MM-DD-YYYY— you want
the system to play while playing the message details to the extension user.

This Date Playback format is applicable only for Message Profile.
• In Message Type and Sensitivity, select the option according to which you want the system to play the
Message Type and Sensitivity of the stored messages to the extension user. You may select — Do Not
Play, Play before Message or Play after Message.
• Select Do Not Play, if you do not want the system to play the Message Type and Sensitivity.
• Select Play before Message, if you want the system to play the Message Type and Sensitivity before
playing the message.
• Select Play after Message, if you want the system to play the Message Type and Sensitivity after
playing the message.
• In Message Details, select the option according to which you want the system to play the Message Details
of the stored messages to the extension user. You may select — Play before Message, Play after
Message or Play on Demand.
• Select Play before Message, if you want the system to play the Message Details before playing the
message.
• Select Play after Message, if you want the system to play the Message Details after playing the
message.
• Select Play on Demand, if you want the system to play the Message Details only when the caller
wants. You must select the desired option to listen to the message details.
The System will first play the Message Type and Sensitivity and then play the Message Details.
• Select the Message Count check box if you want the system to play the Message sequence number to
the extension user before playing the message. The message count is just to differentiate between various
messages played to the caller.
The message count has no interaction with the number of old/new/total messages present in the mailbox.
If you disable the check box, the system will not play the Message sequence number before playing the
message to the extension user.

Message Send/Forward Settings
Message Send/Forward Settings allows you to configure the settings for the messages you send or forward.
• In Send/Forward Number Collection Prompt, select the prompt which you wish to play when you have
selected the Send Message or Forward Message option. This is to prompt the extension user for entering
the destination number/s for sending/forwarding the message.
Once the files are uploaded these appear as options for Send/Forward Number Collection Prompt.
• Select the Confirm Number Collected check box to allow the system to check the Destination Number/s
collected by the extension user. Configure the “Destination Number Confirmation” settings.
• In Stop Record Message Code, enter the digits (0-9,* or #) that you want the extension user to dial to
stop the recording of the message. The system will play this to the extension user along with the prompt
before s/he starts recording the message. You may configure a maximum of upto 3 digits.
If you do not assign any digit and keep it blank, the system will play the prompt without the stop code. In this
case, you may dial any digit to stop recording.
• Select the Message Verification check box to allow the recorded message to be verified by the extension
user before storing it in the Personal Mailbox or re-record it.

• In Message Type, select the option according to which you want the system to send/forward the recorded
• Select Set as Normal, if you want the system to send/forward the recorded message in the mailbox as
Normal.
• Select Set as Urgent, if you want the system to send/forward the recorded message in the mailbox as
Urgent with higher priority.
• Select Ask Caller, if you want the system to ask the extension user to select the message type before
storing the recorded message in the mailbox. If the call gets disconnected before the message type is
selected, the message will be set as normal by default. You must configure the “Message Send
Options”.
While retrieving, the Urgent and Normal messages will be played separately.
• Select Set as Normal, if you want the system to send/forward the recorded message in the mailbox as
Normal.
• Select Set as Private, if you want the system to send/forward the recorded message in the mailbox as
Private. The message received is considered as confidential and forwarding the message is restricted.
• Select Ask Caller, if you want the system to ask the extension user to select the message sensitivity
before storing the recorded message in the mailbox. If the call gets disconnected before the message
sensitivity is selected, the message will be set as normal by default. You must configure the “Message
Send Options”.
• The Message Security parameter is reserved for future use. Keep this check box disabled.
• Select the Message Send Confirmation Prompt check box if you want the system to play the prompt to
the extension user after sending the message on the destination number/s i.e. to the recipients. The
Message Type and Message Sensitivity that you configured will be played.
If you disable the check box, the system will not play the Message Send Confirmation prompt to the
extension user after sending the message.
Destination Number Confirmation

This option is applicable only if you have selected the Confirm Number Collected check box.
You can assign the Digits (0-9, *, # or None) to each of the options — Confirm, Re-enter.
Make sure you do not assign the same digit to both the options.

The digit you configure will be played as the digit to be dialed by the extension user to select the respective option.
The options will be played in the sequence — 1-9, 0, *, #. The option for which the Digit selected is None will not be
played.
below is a brief description of each option:
• Confirm: The system will save the destination number/s collected considering they are valid.
• Re-enter: The system will clear all the destination numbers collected and ask the extension user to record
the destination number/s again.
• In No Digit Dialed Action, select the option — Confirm or Re-enter. This is applicable when the extension
user has not dialed any digit for Destination Number Confirmation. The system will function as per the
option you select.
Forward Message Option
You can assign the Digits (0-9, *, # or None) to each of the options — With Comment at Start, With Comment at
End, Without Comment or Go to Previous Menu.
played.
below is a brief description of each option:
• With Comment at Start: The system will provide an option to the extension user for recording a message
in addition to the existing recorded message before forwarding the message to the destination number/s.
The additional message will be added as the initial part of the message i.e. will be added at the beginning
of the message.
• With Comment at End: The system will provide an option to the extension user for recording a message
in addition to the existing recorded message before forwarding the message to the destination number/s.
The additional message will be added as the final part of the message i.e. will be added at the end of the
message.
• Without Comment: The system will provide an option to the extension user for sending the recorded
message to the destination number/s without adding any additional comment to it.

• Go to Previous Menu: The system will provide an option to the extension user to go back to the Mailbox
Access Main Menu.
• In No Digit Dialed Action, select the option — With Comment at Start, With Comment at End or Without
Comment. This is applicable when the extension user has not dialed any digit for Forward Message
Option. The system will carry out the function when no digit is dialed as per the option you select.
Message Send Options

Message Send Options will be played based on the configuration of the parameters — Message Verification,
Message Type and Message Sensitivity.
The options will be played in the sequence — 1-9, 0, *, #. The option for which the digit selected is None will not be
played.
• Re-record: The system will clear the recorded message and let the extension user record a new message
again.
• Urgent: The system will save the recorded message as Urgent message. This is applicable only if you
have selected the Ask Caller as the Message Type option.
• Private(Confidential): The system will save the recorded message as Private message. This is applicable
only if you have selected the Ask Caller as the Message Sensitivity option.

• Listen Recorded Message: The system will play the recorded message to the extension user and then
play the message leave options again. This is applicable only if you have enabled the Message
Verification check box.
• Append to Recorded Message: The system will let the extension user record a new message again and
add it to the existing recorded message. This is applicable only if you have enabled the Message
Verification check box.
etc — before allowing the extension user to append the recorded message.
• In No Digit Dialed Action, if no digit is dialed by the extension user, the system will take action as per the
Message Type and Message Sensitivity configured.
This is applicable only if you have enabled the Message Verification check box or selected the Ask Caller
as the Message Type and/or Message Sensitivity option.
Based on the configuration, you may select the desired action — Normal+Normal, Urgent+Normal,
Normal+Private or Urgent+Private.
Message Delivery Options
You can assign the Digits (0-9, *, # or None) to each of the options — Request Read Receipt or Ignore Read
Receipt.
played.
After the digits have been assigned, the system will proceed further as per the options selected by you. Brief
description of each option is explained below:
• Request Read Receipt: The system will provide an option to the extension user for requesting the read
receipt to be played whenever the message is read by the destination number. In case of multiple
destination numbers, it will be played individually for each number after the message has been read.
The read receipt includes the name, destination number and the first five seconds of the message sent or
forwarded.

• Ignore Read Receipt: The system will provide an option to the extension user for ignoring the read receipt
i.e. no read receipt will be played back to the extension user.
• In No Digit Dialed Action, select the option — Request Read Receipt, Ignore Read Receipt. This is
applicable when the extension user has not dialed any digit for Message Delivery Option. The system will
carry out the function when no digit is dialed as per the option you select.
No Match Found Settings (while collecting destination number)

No Match Found Settings are applicable when no valid match is found as the Destination Number.
• In No Match Found Prompt, select the prompt which you wish to play when the extension user has dialed
an invalid destination number/s for sending/forwarding the message.
Select None, if you do not wish to play any prompt when extension user has dialed an invalid destination
number.
No Digit Dialed Settings (while collecting destination number)

No Digit Dialed Settings are applicable when extension user has not dialed the destination number and the first digit
wait timer has expired.
• In No Digit Dialed Prompt, select the prompt which you wish to play when extension user has not dialed
any digit for destination number.
Select None, if you do not wish to play any prompt when extension user has not dialed any digit.

• Select the No Digit Dialed Retry check box to prompt the extension user to dial the Destination Number
again.
• In No Digit Dialed Retry Count, select the number of times you wish to prompt the extension user for
dialing the destination number. The prompt will be played repeatedly till the Retry Count expires.
• In No Digit Dialed Action, select — Go to Previous Menu or Disconnect.
• Select Go to Previous Menu, if you want the system to provide an option to the extension user to go
back to the Mailbox Access Main Menu.

Call Transfer Profile
Call Transfer Profile allows you can select the Call Transfer Type — Blind, Wait for Ring, Wait for Answer,
Screened, None — and customize parameters related to it as per your requirement.
VMS supports a maximum of 64 Call Transfer Profiles.
By default, three Call Transfer Profiles — Wait for Ring, Blind and Attended — are provided to you. These three
profiles cannot be deleted but you may edit their settings as per your requirement.
How to Configure
• Log into Jeeves.
• Click Call Transfer Profile.
You may add a new profile, edit the default profile or delete a Call Transfer Profile.
To add a new Call Transfer Profile,
• Click Add New Profile. A new Transfer Profile xx will be created. You may now configure this as per your
requirement.
The default Call Transfer Profiles cannot be deleted.

To edit a Call Transfer Profile,
• In Profile Name, you may configure the name of the Call Transfer profile you want. By default, it is
Transfer Profile xx where xx is the Call Transfer Profile Number from 01 to 64.
• In Call Transfer Type, you can select — None, Blind, Wait for Ring, Wait for Answer or Screened.
• If you select None, the caller will be transfered to the transferor’s mailbox directly. You must configure
the Leave Voice Mail parameters.
• If you select Blind, the caller will be transfered to the transfer number directly.
• If you select Wait for Ring, the caller will be transfered to the transfer number after the number starts
ringing.
• If you select Wait for Answer, the caller will be transfered to the transfer number only after the
transferor answers the call.
• If you select Screened, the caller will be transfered only after the transferor confirms to speak to the
caller. You must configure the Call Transfer Type - Screened parameters.
• In Attended Transfer Prompt, select the prompt you want the system to play while the call is being
transfered. This is applicable only if you have selected Wait for Ring or Wait for Answer as the Call
Transfer Type option.
Once the prompts are uploaded these appear as options for Attended Transfer Prompts.
• In Blind Transfer Prompt, select the prompt you want the system to play when the call is being
transfered. This is applicable only if you have selected Blind as the Call Transfer Type option.
Select None, if you do not wish to play any prompt. You may also add a new Prompt. To do so, follow the
same steps as given in Attended Transfer Prompt.
• In Screened Transfer Prompt, select the prompt you want the system to play when the call is being
transfered. This is applicable only if you have selected Screened as the Call Transfer Type option.
Select None, if you do not wish to play any prompt. You may also add a new Prompt. To do so, follow the
same steps as given in Attended Transfer Prompt.
• In Wait for Answer Timer (sec), configure the time for which you want the system to wait before
transferring the call. This is applicable only if you have selected Wait for Answer or Screened as the Call
Transfer Type option.
If the Wait for Answer Timer value is greater than Ring Back Tone timer, then Ring Back Tone Timer will
expire first and the next action will be taken as per the Call Transfer Unsuccessful - Unconditional
configuration.

• In Extension Name, you can select — Do not play or Play Always.
Select Play Always, if you want the system to play the recorded Extension Name as well as the respective
Transfer prompt during the transfer.
Select Do not Play, if you want the system to only play the respective Transfer prompt and not the
Extension Name during the transfer.
• In Call Transfer - Music On Hold (MOH), you may select System MoH or add a new MoH. By default,
System MoH will be played. This parameter is applicable only if you have selected Wait for Ring or Wait for
Answer as the Call Transfer Type.
Once the files are uploaded, these appear as options for Call Transfer - MoH.
Extension, Transfer to Department Group or Go to VMAA Menu. This parameter is applicable only if you
have selected None as the Call Transfer Type.
• Select the respective Operator number/Department Group number or enter the desired extension
number.
• In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement. By default,
Call Transfer Profile for Transfer Number is selected.

Leave Voice Mail
This option is applicable only if you select None as the Call Transfer Type.
the Leave Voice Mail Settings configured on this page.
Clear the check box, if you want the system to apply the Leave Voice Mail settings configured for the
extension.

Message Leave Settings allows you to configure the settings for the messages you receive from the caller i.e the
message left for you by the caller.
Personal Greetings will not be applicable if your mailbox is defined as the destination for Call Tapping or
Conversation Recording messages.
caller.
Conditional Greeting will be applicable only if your mailbox is defined as the destination for Call Transfer
Unsuccessful.

If you do not assign any digit and keep it blank, the system will play the prompt without the stop code.
before storing it in the Personal Mailbox or re-record it. You must configure the “Message Leave Options”.
• In Message Type, select the option according to which you want the system to store the recorded
• Select Set as Urgent, if you want the system to store the recorded message in the mailbox as Urgent.
• Select Ask Caller, if you want the system to ask the caller to select the message type before storing
the recorded message in the mailbox.
• Select Set as Private, if you want the system to store the recorded message in the mailbox as Private.
The message received is considered as confidential and forwarding the message is restricted.
storing the recorded message in the mailbox.
• Select the Message Security check box, if you want the system to restrict the forwarding and
downloading of the messages.
Message Security has higher priority than Message Type and Message Sensitivity set for the message.
If you disable the check box, the system will not set any security for the message.
• Select the Message Leave Confirmation Prompt check box if you want the system to play the Message
Leave Confirmation prompt to the caller after leaving the message. The Message Leave Confirmation
prompt plays the Message Type, Message Sensitivity and Message Security that you configured.


played.
• Urgent: The system will save the recorded message along with the Message Type as Urgent. This is
applicable only if you have selected the Ask Caller as the Message Type option.
• Private(Confidential): The system will save the recorded message along with the Message Sensitivity as
Private. This is applicable only if you have selected the Ask Caller as the Message Sensitivity option.
box.
box.
etc — before allowing the caller to append the recorded message.
If you have configured Ask Caller option in Message Type and/or Message Sensitivity, you can select the
desired action — Normal+Normal, Urgent+Normal, Normal+Private or Urgent+Private.
Call Transfer Unsuccessful - Busy

If the called party is busy and has set Call Forward-Busy to VMS, the Call Transfer Unsuccessful - Busy
parameters will be applicable. You can customize these parameters as per your requirement.
The VMS allows you to:

• process the call immediately for the busy condition as per the options you select.
or
• process the call after being held for a certain duration as per the options you select.

Configure the following parameters, if you want the VMS to process the call immediately when the desired
extension is busy,
• Busy Prompt
• Busy Action
• Action after Leaving Voice Mail
Configure the following parameters, if you want the VMS to put the call on hold when the desired extension is busy,
• Busy Prompt
• Busy Hold Prompt
• Provide Advanced Options during Hold
• Busy Extension Status check time interval (sec) and Time Interval Expiry Prompt
• Busy Extension Status check retry count and Retry Count Expiry Prompt
• Busy Hold - Music on Hold
• Busy Action
• Action after Leaving Voice Mail
• Select Hold a Call check box, if you want the call to be held. This option is applicable only for internal calls.
• In Busy Prompt, select the prompt you wish to play to the caller when the dialed extension is busy.
Once the files are uploaded these appear as options for Busy Prompt.
• In Busy Hold Prompt, select the prompt you wish to play to the caller when put on hold as the dialed
extension is busy.
Once the files are uploaded these appear as options for Busy Hold Prompt.
• Select Provide Advanced Options during Hold check box, if you want the system to play
Advanced options to the caller.

You can assign the Digits (0-9, *, # or None) to each of the options — Leave Voice Mail, Transfer to
Operator, Transfer to Assistant, Transfer to Alternate/Mobile Number, Dial Extension Number, Go to
Main Menu, Go to Previous Menu and Disconnect.
The digit you configure will be played as the digit to be dialed by the caller to select the respective
option. The options will be played in the sequence — 1-9, 0, *, #. The option for which the Digit
selected is None will not be played.
After the digits have been assigned, the system will proceed further as per the options selected by
you. Brief description of each option is explained below:
• Leave Voice Mail: After you assign the digit, click Settings to configure the parameters.
• Select the Override Message Leave Settings of Extension check box, if you want the
system to apply Message Leave Settings customized on this page and not that of the
extension or department group number.
Clear the check box if you want the system to apply Message Leave Settings configured for
the extension or department group number.
• Transfer to Operator: After you assign the digit, click Settings to configure the parameters.
• In Operator Group, select the desired Operator Group Number. Make sure you have
configured the parameters for this group. For detailed instructions, see “Extension Voice Mail
Settings”. By default, As Configured in Extension Settings is selected, that is, the system will
use the Operator Group configured for the transfer number.
• In Call Transfer Profile, select the desired Call Transfer Profile you want the system to use.
By default, As configured for Transfer Number is selected.
• Select As configured for Dialed Extension if you want the VMS Auto Attendant to use
the Call Transfer Profile which is assigned to the dialed extension.
• Select Wait for Ring if you want the VMS Auto Attendant to wait for the extension to start
ringing and then transfer the call.
• Select Blind if you want the VMS Auto Attendant to transfer the call on the extension
without checking whether it is busy or free.
• Select Attended if you want the VMS Auto Attendant to transfer the call when the
extension answers (goes OFF-Hook).
Click Settings to configure the parameters of the selected profile.
When the Call Transfer Profiles are added, or any existing profile is edited, these changes will be reflected
in the options displayed.
• Transfer to Assistant: If you assign a digit for this option, make sure you have programmed the
Assistant Number. For detailed instructions, refer “Number Programming (Assistant/Personal)”
in “Mailbox Menu”.

Click Settings to configure the parameter.
default, Wait for Ring is selected.
• Transfer to Alternate/Mobile Number: If you assign a digit for this option, make sure you have
programmed the Alternate/Mobile Number. For detailed instructions, refer “Number
Programming (Assistant/Personal)” in “Mailbox Menu”.
• Dial Extension Number: After you assign the digit, click Settings to configure the
parameters.
• In Dial Extension Number Prompt, select the prompt you wish to play to the caller to dial
the desired extension number.
Once the files are uploaded these appear as options for Dial Extension Number Prompt.
• In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement.
By default, As configured for Dialed Extension is selected.
No Match Found Settings: No Match Found Settings are applicable when a Extension Number
is found invalid, the system will play the No Match Found prompt.
• In No Match Found Prompt, select the prompt which you wish to play when caller has
dialed an invalid extension number.
• Select the No Match Found Retry check box to prompt the caller to dial the Extension
Number again.
• In No Match Found Retry Count, select the number of times you wish to prompt the caller
for dialing the extension number. The prompt will be played repeatedly till the Retry Count
expires.

expires.

Extension, select the respective Operator number/Department Group number or enter
In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement.
By default, As Configured for Transfer Number is selected
No Digit Dialed Settings: No Digit Dialed Settings are applicable when caller has not dialed the
extension number and the first digit wait timer has expired.
• In No Digit Dialed Prompt, select the prompt which you wish to play when caller has not
dialed any digit.
• Select the No Digit Dialed Retry check box to prompts the caller to dial the Extension
Number again.
• In No Digit Dialed Retry Count, select the number of times you wish to prompt the caller for
dialing the extension number. The prompt will be played repeatedly till the Retry Count
expires.
expires.

• In No Digit Dialed Action, you can select — Disconnect, Transfer to Operator, Transfer to

Extension, select the respective Operator number/Department Group number or enter
In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement.
By default, As Configured for Transfer Number is selected.
• Go to Main Menu: Assign a digit for this option if you want the system to re-direct the caller to
the Main Menu.
• Go to Previous Menu: Assign a digit for this option if you want the system to re-direct the caller
to the Previous Menu.
• Disconnect: Assign a digit for this option if you want the system to provide the option to
• In Busy Extension Status check time interval (sec), enter the interval of time after which you want the
system to inform the caller that the extension number is busy. The system will play the Time Interval Expiry
Prompt.
• In Time Interval Expiry Prompt, select the prompt you wish to play when the time interval expires.
Once the files are uploaded these appear as options for Time Interval Expiry Prompt.
• In Busy Extension Status check retry count, select the number of times you wish the system to check if
the transfer number is free. The prompt will be played repeatedly till the Retry Count expires.

• In Busy Hold - Music on Hold, select the prompt you wish to play. This option is applicable only is you
have enabled the Hold a Call check box.
Once the files are uploaded these appear as options for Busy Hold - Music on Hold.
• In Busy Action, you can select — Disconnect, Transfer to Operator, Transfer to Extension, Transfer to
Department Group, Go to VMAA Menu, Go to Voice Mail, Give Advanced Options.

number.
• If you select Give Advanced Option, see “Advanced Options”
Extension, Transfer to Department Group, Go to VMAA Menu. This option is applicable only is you select
Go to Voice Mail or Give Advanced options as the Busy Action option.

number.

Call Transfer Unsuccessful - No Reply
If the called party has set Call Forward-No Reply to VMS, the Call Transfer Unsuccessful - No Reply parameters
will be applicable. You can customize these parameters as per your requirement.
• In No Reply Prompt, select the prompt you wish to play to the caller if the dialed extension does not reply.
• In No Reply Action, you can select — Go to Voice Mail, Give Advanced Options, Transfer to Operator,
Transfer to Extension, Transfer to Department Group, Go to VMAA Menu, Disconnect.

number.
• In Action after Leaving Voice Mail, you can select — Transfer to Operator, Transfer to Extension,
Transfer to Department Group, Go to VMAA Menu, Disconnect. This option is applicable only is you select
Go to Voice Mail or Give Advanced options as the No Reply Action option.

number.

Call Transfer Unsuccessful - Unconditional
If the called party has set Call Forward-Unconditional to VMS, the Call Transfer Unsuccessful - Unconditional
parameters will be applicable. You can customize these parameters as per your requirement.
• In Unconditional Prompt, select the prompt you wish to play to the caller.
• In Action on Unconditional, you can select — Go to Voice Mail, Give Advanced Options, Transfer to
Operator, Transfer to Extension, Transfer to Department Group, Go to VMAA Menu, Disconnect.

number.
• In Action after Leaving Voice Mail, you can select — Transfer to Operator, Transfer to Extension,
Transfer to Department Group, Go to VMAA Menu, Disconnect. This option is applicable only is you select
Go to Voice Mail or Give Advanced options as the No Reply Action option.

number.

Call Transfer Type -Screened
Screened Transfer is when the VMS connects the caller to the transfer number after confirmation from the transfer
number. If you have selected Call Transfer Type as Screened, you may customize the Call Transfer Type -
Screened parameters as per your requirement.
Screened Options
The system will play Screened Options — Accept Call, Reject Call, Reject Call - Busy, Reject Call - NoReply — to
the caller. You can assign the Digits (0-9, *, # or None) to each of the options.
played.
• Accept Call - When this option is selected, the system will transfer the call to the desired number.
• Reject Call - When this option is selected, system will consider it as Call Transfer Unsuccessful -
Unconditional and process the call further as per the settings of “Call Transfer Unsuccessful -
Unconditional”.
• Reject Call - Busy - When this option is selected, system will consider it as Call Transfer Unsuccessful -
Busy and process the call further as per the settings of “Call Transfer Unsuccessful - Busy”.
• Reject Call - No Reply - When this option is selected, system will consider it as Call Transfer Unsuccessful
- No Reply and process the call further as per the settings of “Call Transfer Unsuccessful - No Reply”.
No Option Selected Settings

While the Screened Options are being played, if the caller does not select any options, the following parameters will
be applicable.
• In Prompt if No Option Selected, select the prompt you wish to play to the caller, if no Screened Option is
selected by the caller.

Once the files are uploaded these appear as options for Prompt if No Option Selected.
• In Action if No Option Selected, you can select — Accept Call, Reject Call, Reject Call with Busy, Reject
Call with No Reply, Disconnect.
Invalid Option Selected Settings

While the Screened Options are being played, if the caller dials a digit which is not programmed, the following
parameters will be applicable.
• Clear the Ignore if Invalid Option Dialed check box, if you do not want the system to ignore the invalid
digit dialed and prompt the caller for the same. By default, it is enabled.
• In Prompt if Invalid Option Selected, select the prompt which you wish to play when the caller has
dialed an invalid digit.
Once the files are uploaded these appear as options for Prompt if Invalid Option Selected.
• In Action if No Option Selected, you can select — Accept Call, Reject Call, Reject Call with Busy,
Reject Call with No Reply, Disconnect.
No Mailbox Settings
If for any option Leave Voice Mail is selected and a mailbox is not assigned to the number, the system will play a
prompt to the caller informing the caller that a mailbox is not assigned. The system will then proceed further as per
the action you select here.
• In No Mailbox Action, you can select — Transfer to Operator, Transfer to Extension, Transfer to
Department Group, Go to VMAA Menu, Disconnect.

number.
Wait for Ring is selected.

Advanced Options
The system will play Advanced options— Leave Voice Mail, Transfer to Operator, Transfer to Assistant, Transfer to
Alternate/Mobile Number, Dial Extension Number, Go to Main Menu, Go to Previous Menu, Stay on Hold,
Disconnect — to the caller. You can assign the Digits (0-9, *, # or None) to each of the options.
played.
• Leave Voice Mail
• Transfer to Operator
• Transfer to Assistant
• Transfer to Alternate Number/Mobile Number
• Dial Extension Number
• Go to Main Menu
• Go to Previous Menu
• Stay on Hold
• Disconnect

Leave Voice Mail
After you assign the digit, click Settings to configure the parameters.
Message Leave Settings customized on this page and not that of the extension or department group
number.
Clear the check box if you want the system to apply Message Leave Settings configured for the extension
or department group number.
Transfer to Operator

• In Operator Group, select the desired Operator Group Number. Make sure you have configured the
parameters for this group. For detailed instructions, see “Extension Voice Mail Settings”. By default, As
Configured in Extension Settings is selected, that is, the system will use the Operator Group configured for
the Transfer Number.
Configured for Transfer Number is selected.
Transfer to Assistant
If you assign a digit for this option, make sure you have programmed the Assistant Number. For detailed
instructions, refer “Number Programming (Assistant/Personal)” in “Mailbox Menu”.
• In Call Transfer Profile, select the desired Call Transfer Profile you want the system to use. Make sure
you have configured the parameters for this profile. By default, Wait for Ring is selected, that is, the system
will use the Call Transfer profile configured for transfer number.
• In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement. By default,
Wait for Ring is selected.
Transfer to Alternate/Mobile Number

If you assign a digit for this option, make sure you have programmed the Alternate/Mobile Number. For detailed
instructions, refer “Number Programming (Assistant/Personal)” in “Mailbox Menu”.
Dial Extension Number

• In Dial Extension Number Prompt,select the prompt you wish to play to the caller to dial the desired
extension number.
configured for Dialed Extension is selected.
No Match Found Settings are applicable when the caller has dialed a digit or a number which does not match the
Extension Number list present in the system, the system will play the No Match Found prompt.
• In No Match Found Prompt, select the prompt which you wish to play when caller has dialed a digit that
does not match the Extension Number List.
• Select the No Match Found Retry check box to prompt the caller to dial the Extension Number again.
• In No Match Found Retry Count, select the number of times you wish to prompt the caller for dialing the
• In No Match Found Action, you can select — Disconnect, Transfer to Operator, Transfer to Extension,

number.
No Digit Dialed Settings are applicable when caller has not dialed the extension number and the first digit wait timer
has expired.
• Select the No Digit Dialed Retry check box to prompts the caller to dial the Extension Number again.
• In No Digit Dialed Retry Count, select the number of times you wish to prompt the caller for dialing the
• In No Digit Dialed Action, you can select — Disconnect, Transfer to Operator, Transfer to Extension,

number.

Go to Main Menu
Assign a digit for this option if you want the system to re-direct the caller to the Main Menu.
Go to Previous Menu
Assign a digit for this option if you want the system to re-direct the caller to the Previous Menu.
Stay on Hold
This option is applicable for Call Transfer Unsuccessful - Busy only.
• In Busy Hold Prompt, select the prompt to be played to the caller when put on hold.
Once the files are uploaded these appear as options for Busy Hold Prompt.
• Select Provide Advanced Options during Hold check box, if you want the system to play Advanced
options to the caller.

You can assign the Digits (0-9, *, # or None) to each of the options — Leave Voice Mail, Transfer to
Operator, Transfer to Assistant, Transfer to Alternate/Mobile Number, Dial Extension Number, Go to
Main Menu, Go to Previous Menu and Disconnect.
The digit you configure will be played as the digit to be dialed by the caller to select the respective
option. The options will be played in the sequence — 1-9, 0, *, #. The option for which the Digit selected
is None will not be played.
After the digits have been assigned, the system will proceed further as per the options selected by you.
Brief description of each option is explained below:
• Leave Voice Mail: After you assign the digit, click Settings to configure the parameters.
• Select the Override Message Leave Settings of Extension check box, if you want the system
to apply Message Leave Settings customized on this page and not that of the extension or
department group number.
Clear the check box if you want the system to apply Message Leave Settings configured for the
extension or department group number.
• Transfer to Operator: After you assign the digit, click Settings to configure the parameters.
• In Operator Group, select the desired Operator Group Number. Make sure you have configured
the parameters for this group. For detailed instructions, see “Extension Voice Mail Settings”. By
default, As Configured in Extension Settings is selected, that is, the system will use the Operator
Group configured for the Transfer Number.
• Transfer to Assistant: If you assign a digit for this option, make sure you have programmed the
Assistant Number. For detailed instructions, refer “Number Programming (Assistant/Personal)” in
“Mailbox Menu”.
• Transfer to Alternate/Mobile Number: If you assign a digit for this option, make sure you have
programmed the Alternate/Mobile Number. For detailed instructions, refer “Number Programming
(Assistant/Personal)” in “Mailbox Menu”.
• Dial Extension Number: After you assign the digit, click Settings to configure the parameters.


default, As configured for Dialed Extension is selected.
No Match Found Settings: No Match Found Settings are applicable when a Extension Number is
found invalid, the system will play the No Match Found prompt.
• In No Match Found Prompt, select the prompt which you wish to play when caller has dialed
an invalid extension number.
• Select the No Match Found Retry check box to prompt the caller to dial the Extension Number
again.
• In No Match Found Retry Count, select the number of times you wish to prompt the caller for
expires.

In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement. By

No Digit Dialed Settings: No Digit Dialed Settings are applicable when caller has not dialed the
extension number and the first digit wait timer has expired.
• In No Digit Dialed Prompt, select the prompt which you wish to play when caller has not dialed
any digit.
• Select the No Digit Dialed Retry check box to prompts the caller to dial the Extension Number
again.
• In No Digit Dialed Retry Count, select the number of times you wish to prompt the caller for
expires.
• In No Digit Dialed Action, you can select — Disconnect, Transfer to Operator, Transfer to

In Call Transfer Profile, select the desired Call Transfer Profile as per your requirement. By

• Go to Main Menu: Assign a digit for this option if you want the system to re-direct the caller to the
Main Menu.
• Go to Previous Menu: Assign a digit for this option if you want the system to re-direct the caller to
the Previous Menu.
• Disconnect: Assign a digit for this option if you want the system to provide the option to disconnect
the call.
• In Busy Extension Status check time interval (sec), enter the interval of time after which you want the
system to inform the caller that the extension number is busy. The system will play the Time Interval Expiry
Prompt.
• In Time Interval Expiry Prompt, select the prompt you wish to play when the time interval expires.
Once the files are uploaded these appear as options for Time Interval Expiry Prompt.
• In Busy Extension Status check retry count, select the number of times you wish the system to check if
the transfer number is free. The prompt will be played repeatedly till the Retry Count expires.
• In Busy Hold - Music on Hold, select the prompt you wish to play. This option is applicable only is you
have enabled the Hold a Call check box.
You may add a new Prompt. To do so,
Once the files are uploaded these appear as options for Busy Hold - Music on Hold.
• In Busy Action, you can select — Disconnect, Transfer to Operator, Transfer to Extension, Transfer to
Department Group, Go to VMAA Menu, Go to Voice Mail, Give Advanced Options.

number.
As configured for Transfer Number is selected.
• If you select Give Advanced Option, see “Advanced Options”.
Disconnect
Assign a digit for this option if you want the system to provide the option to disconnect the call.
No Digit Dialed Settings (during Advanced Options)

No Digit Dialed Settings are applicable when caller has not dialed the extension number and the first digit wait timer
has expired.
• In No Digit Dialed Action, you can select — Disconnect, Go to Voice Mail, Transfer to Operator, Transfer
to Assistant, Transfer to Alternate Number, Go to Main Menu, Go to Previous Menu, Stay on Hold.
Refer to the details given above under “Advanced Options”.
Invalid Digit Dialed Settings (during Advanced Options)

Invalid Digit Dialed Settings are applicable when caller has dialed an invalid digit — a digit which is not configured
as the Advanced Options.

• In Invalid Digit Dialed Prompt, select the prompt which you wish to play when the caller has dialed an
invalid digit.
• In Invalid Digit Dialed Action, you can select — Disconnect, Go to Voice Mail, Transfer to Operator,
Transfer to Assistant, Transfer to Alternate Number, Go to Main Menu, Go to Previous Menu, Stay on
Hold.
Refer to the details given above under “Advanced Options”.

Extensions Over Q-SIG
SARVAM UCS supports Voice Mail for extension users connected over QSIG. These are extensions of the remote
System (may be SARVAM UCS or any other system) connected with SARVAM UCS over QSIG.
To provide Voice Mail to the remote system extension users you must configure the Voice Mail Settings. To do so,
• Click Extensions over Q-SIG.
• For the extensions of the system you wish to assign voice mail, configure the following:
• In Extension Number, enter the number assigned to the extensions of the remote System. The
number must be entered along with the Exchange ID (if applicable).
The Extension Number can be a maximum of 6 digits max. Valid Digits are 0-9, * ,# (*, # can be used
as fist digit only).
• In Name, enter a name which you wish to assign to the extension. The name may be of the person
using the extension number of the remote System or it can be the name of a department.
It can be a maximum of 18 alphanumeric characters. Default: Blank.

• Click Default to default the page.
• Click Default One to default only one entry.
A new window will open. Select the Index Number which you wish to default.
Click OK.

Notification via Call-Profile
What’s this?
The VMS supports Notification via Call to inform the extension users about the arrival of new messages in their
mailbox.
Extension users can receive new message notification calls on a phone number of their choice. This number may
be another extension number or an external number. You can set the Type of notification calls as:
• Immediate: Users will receive notifications as soon as a new message arrives in their mailbox.
Or
• Scheduled: Extension users will receive notification at specified time intervals.
You can set the preferred time slots in a day during which notification calls should be made to extension users. In
addition to the time slot preference, you can also choose to receive notification calls on a Holiday.
Message Notification via Call for Department Group will not work if the destination number is an external
number.
How it works
For this feature to work, you must do the following configuration for the extension:
• Select the type of Notification call.
• Define the preferred time slots by configuring Time Zones. You can configure four different Time Zones,
defining the Start Time and End Time for each Time Zone.
• Configure the phone number to which the notification call is to be made. If the number is an external
number, configure the Trunk Access Code to be used for making the calls.
When Immediate is selected as the Type of notification,
• A new message arrives in the mailbox of the extension user.
• The system checks the preferred start and end time of the time zones configured for the extension. If the
message has arrived within the preferred time slot (Start and End Time) it immediately makes the
notification call on the number configured for the user.
If the number is an external number, the system dials out the number using the Trunk Access Code (TAC)
assigned for making notification calls.
• When the call is answered, the extension user gets connected to the VMS and can listen to the message.
• If the notification call is not answered, by default, the system makes three attempts (Message Notification
Retry Count; programmable) at an interval of 5 minutes (Message Notification Interval; programmable)
between each attempt.
• If the notification call remains unanswered after the third attempt, the system will not make any more
attempts to place this notification call. The next notification call will be made only when another new
message arrives in the mailbox of the user between the start and end time of the configured time zone.

When Schedule is selected as the Type of notification,
• The system checks the start time of the time zone(s) configured and the notification call will be made on
the number at the subsequent start time.
If the number is an external number the system dials out the number through the Trunk Access Code
(TAC) assigned for making notification calls.
• When the call is answered the extension user gets connected to the VMS and can listen to the message.
• If the notification call is not answered, the system makes three attempts (Message Notification Retry
Count; programmable) at an interval of 5 minutes (Message Notification Interval; programmable) for each
time zone. The system will continue to make attempts to place the notification call till the call is answered.
Thus, when Notification type is Immediate, notification call is made for each message that is received within the
start and end time configured in the time zone.
When Notification type is Scheduled, notification call is made for all messages received before the start time
configured in the time zone. Where multiple time zones are configured, notification call will be made at the start
time of the next time zone.
How to configure
For Message Wait Notification via Call, you need to configure:
• the parameters for Message Wait Notification via Call under Message Wait Settings in Extension Voice
Mail Settings.
• select the desired profile in Schedule Profile. For instructions to configure the profile parameters, see
“Configuring Notification via Call - Profile”.
• Make Message Notification call using TAC for calls to be made to external numbers. See “Configuring
VMS General Parameters”.
• if required, the Message Notification Retry Count, Message Notification Interval and Message Notification
Ring. See “System Timers and Counts”.
Configuring Notification via Call - Profile

• Click Notification via Call - Profile to open the page.
• Select the Profile Number which you want to assign to Message Wait Notification via Call.
The Message Wait Notification Profile determines how notification calls are to be made to the desired
numbers. You can configure upto 16 different profiles. In each profile, you can set different time zones
according to the user preferences.
Configure the following parameters against the Profile Number you select:
• For Each Time Zone, Time Zone 1 to 4, configure the Start Time and End Time. The valid range is
00:00 to 23:59.
• If you want to receive notifications on a holiday, select the Notify on Holiday check box.
• Now, assign the profile numbers to the desired extensions. Make sure the Message Wait Notification
via Call parameters have been configures in the Extension Voice Mail Settings of these extensions.

Mailbox Menu
Mailbox Menu offers you a group of parameters that will be used when the caller accesses his/her personal
mailbox. VMS supports a maximum of 12 Mailbox Menus.
By default, three mailbox Menus — User, Executive and Guest— are provided to you. These three mailbox menus
cannot be deleted but you can edit their settings as per your requirement.
How to Configure
• Click Mailbox Menu.
You may add a new menu, edit the default menus or delete a Mailbox menu.
To add a new Mailbox Menu,
• Click Add New Menu. A new Mailbox Menu xxx will be created. You may now configure this as per your
requirement.
To delete a menu, click Delete.
The default Mailbox Menus cannot be deleted.

To edit a Mailbox Menu,
• In Menu Name, configure the name of the Mailbox Menu you want. By default, it is Mailbox Menu xx where
xx is the Mailbox Menu Number from 01 to 12.
Mailbox Menu Features

You can assign the Digits (0-9, *, # or None) to each of the options given under the features — Mailbox Access,
Listening a Message, Mailbox Management, Message Redirection, Mailbox Greetings, Personal Greetings
Timezone Selection, Conditional Greeting, Record Greetings/Name and Number Programming (Assistant/
Personal).
Make sure you do not assign the same digit to multiple options under the respective feature. For example,
the digits assigned to various options under Listening a Message feature must be unique.
Select the Play check box to play the respective option.
If you assign the digit to an option but do not select the Play check box for the same, then the option will not be
played to the caller.
The digits you configure will be played as the digit to be dialed by the caller to select the respective option. The
options will be played in the sequence — 1-9, 0, *, #. The option for which the digit selected is None will not be
played.
Brief description of each option in the respective features are given below:
Mailbox Access
Mailbox Access is the main menu which will be played to the caller whenever s/he accesses his/her Personal
Mailbox.
• Listen New Messages: The system will play the new(unread) messages present in the caller’s personal
mailbox. The new messages will be played as per the Message Playback Settings configured in the
Message Profile. For details, refer “Message Playback Settings”.
• Listen Old Messages: The system will play the old(read) messages present in the caller’s personal
mailbox. The old messages will be played as per the Message Playback Settings configured in the
• Send Message: The system will play a prompt asking the caller to enter the Destination Number of the
recipient for sending the message. The Destination Number may include a Distribution List. The Message
will be send as per the Message Send/Forward Settings configured in the Message Profile. For details,
refer “Message Send/Forward Settings”.
• Mailbox Management: The system allows user to manage his/her personal mailbox. For further
information, see “Mailbox Management”.

Listening a Message
Listening a Message is a menu which will be played to the caller after every message (old/new).
• Replay Message: The system will play the message again.
• Play Message Details: The system will play the message details — Message Date/Time and the Name of
sender. The message details will be played as per the Message Playback Settings configured in the
• Reply Message: The system will allow the caller to reply the sender with a message. The Message will be
sent as per the Message Leave Settings configured in the Message Profile. For details, refer “Message
Leave Settings”.
After sending the reply message, the Mailbox Access menu will be played again.
Reply Message will not be played if:

• the message does not include the calling party number.
• the message includes the calling party number but no Mailbox is assigned.
• the message includes the calling party number as an external number.
• it is a Broadcasted or a Call Tapping message.
• Delete Message: The system will delete the message and play the next message(old/new).
• Listen Next Message: The system will play the next message(old/new) as per the last message played.
After playing the next message, the Mailbox Access menu will be played again.
• Forward Message: The system will play a prompt asking caller to enter the Destination Number of the
recipient for forwarding the message.
The Destination Number may include a Distribution List. A maximum of 10 Destination Numbers can be
added. The Message will be sent as per the Message Send/Forward Settings configured in the Message
Profile. For details, refer “Message Send/Forward Settings”.
After forwarding the message, the Mailbox Access menu will be played again.
The message will not be forwarded if the Message Sensitivity is set as Private. For details, refer “Message
Profile”.

• Save Message as New: The system will save the message as a New Message keeping all the message
properties — Date and Time, Caller, Sensitivity and Security — unchanged.
• Go to Previous Menu: The system will provide an option to the caller to go back to the “Mailbox Access”
menu.
Mailbox Management
Mailbox Management is a menu used to manage the Personal Mailbox.
• Record Mailbox Name: The system will allow the caller to record, play or erase the Mailbox Name.
The Record Greetings/Name options will be played where the caller can choose whether s/he want to
Record, Play or Erase the Name. For details, refer “Record Greetings/Name”.
• Message Redirection: The system will allow the caller to set or cancel Message Redirection. Message
Redirection is used when the caller wants to forward all the messages to the other user’s personal
mailbox.
The Message Redirection options will be played where caller can choose if s/he wants to set or cancel this
feature. For details, refer “Message Redirection”.
• Delete all Old Messages: The system will delete all the old messages from the caller’s Personal Mailbox.
• Delete all Messages: The system will delete all the messages (old/new) from the caller’s Personal
Mailbox.
• Record Mailbox Greetings: The system will allow the caller to record, play or erase the Mailbox Personal
and Conditional Greetings.
The options will be played in the following sequence:
• The Mailbox Greetings options will be played asking the caller to select the type of greetings —
Personal or Conditional — which he wishes to Record, Play or Erase. For details, refer “Mailbox
Greetings”.
• If the caller selects Personal, the Personal Greeting Time zone Selection options will be played.
The caller must select the timezone — Working Hour, Break Hour or Non-Working Hour for which s/
he wishes to Record, Play or Erase the personal greeting. For details, refer “Personal Greeting
Timezone Selection”.

• If the caller selects Conditional, the Conditional Greetings options will be played. The caller must
select the type of Conditional Greeting — Busy, No-Reply or Unconditional for which s/he wishes to
Record, Play or Erase the greetings. For details, refer “Conditional Greeting”.
• The Record Greetings/Name options will be played where the caller can choose whether s/he wants to
Record, Play or Erase the Mailbox Greetings. For details, refer “Record Greetings/Name”.
• Assistant Number: The system will allow the caller to enter, play or clear the Assistant Number.
The caller can call this number when the user is not available to answer the call.
The Number Programming (Assistant/Personal) options will be played from where the caller can choose
whether s/he wants to Enter, Play or clear the Assistant Number. For details, refer “Number Programming
(Assistant/Personal)”
• Personal Number: The system will allow the caller to enter, play or clear the Personal Number.
The Personal Number is used as an alternate number. The caller can call this number when the user is not
available to answer the call.
The Number Programming (Assistant/Personal) options will be played from where the caller can choose
whether s/he want to Enter, Play or clear the Personal Number. For details, refer “Number Programming
(Assistant/Personal)”.
• Go to Previous Menu: The system will provide an option to the caller to go back to the “Mailbox Access”
menu.
Message Redirection
Message Redirection is a menu which is applicable if the caller selects the Message Redirection option in “Mailbox
Management”.
• Set: The system will set the Message Redirection feature and play a prompt to the caller for entering the
Message Redirect Number.
The Message Redirect Number must be a valid Extension Number, Department Group Number or
Extension over QSIG Number. A Mailbox must be assigned to this Number.
• Cancel: The system will clear the Message Redirect Number programmed by the caller.
• Go to Previous Menu: The system will provide an option to the caller to go back to the “Mailbox
Management” menu.
Mailbox Greetings
Mailbox Greetings is a menu which is applicable if the caller selects the Record Mailbox Greetings option in the
Mailbox Management.

• Personal: The system will play the Personal GreetingTimezone Selection options. The caller must select
the timezone — Working Hour, Break Hour or Non-Working Hour for which s/he wishes to Record, Play or
Erase the Personal greetings. For details, refer “Personal Greeting Timezone Selection”.
• Conditional: The system will play the Conditional Greetings option. The caller must select the type of
Conditional Greeting — Busy, No-Reply or Unconditional for which s/he wishes to Record, Play or Erase.
For details, refer “Conditional Greeting”.
Management” menu.
Personal Greeting Timezone Selection

Personal Greeting Timezone Selection is a menu which is applicable if the caller selects the Personal Greetings
option in the Record Mailbox Greetings. You can record the personal greetings like “Good Morning”, “Good
Afternoon” or “ Good Evening” to greet the caller.
The various Timezones are:
• Working Hour: The system will play the Record Greetings/Name options and allow the caller to Record,
Play or Erase a personal greeting for the Working Hour Timezone.
• Break Hour: The system will play the Record Greetings/Name options and allow the caller to Record, Play
or Erase a personal greeting for the Break Hour Timezone.
• Non-Working Hour: The system will play the Record Greetings/Name options and allow the caller to
Record, Play or Erase a personal greeting for the Non-Working Hour Timezone.
• Go to Previous Menu: The system will provide an option to the caller to go back to the Record Mailbox
Greetings Menu.
For details, refer “Record Greetings/Name”.

Conditional Greeting
Conditional Greeting is a menu which is applicable if the caller selects the Conditional option in the Record Mailbox
Greetings. Conditional Greetings are applicable if any call forward — Busy, No Reply or Unconditional — is set on
the calling extension.
The various Conditional Greeting types are:
• Busy: The system will play the Record Greetings/Name options and allow the caller to Record, Play or
Erase the Conditional Greeting of type — Busy.
• No-Reply: The system will play the Record Greetings/Name options and allow the caller to Record, Play
or Erase the Conditional Greeting of type — No-Reply.
• Unconditional: The system will play the Record Greetings/Name options and allow the caller to Record,
Play or Erase the Conditional Greeting of type — Unconditional.
• Go to Previous Menu: The system will provide an option to the caller to go back to the Record Mailbox
Greetings Menu.
For details, refer “Record Greetings/Name”.
Record Greetings/Name
Record Greetings/Name is a menu which allows the caller to Record, Play or Erase the Mailbox Name or the
Personal/Conditional Greetings.
• Record: The system will allow the caller to Record the Mailbox Name or the Personal/Conditional
Greeting. Maximum allowed length for recording a name/greeting is 120 seconds.
• Play: The system will Play the Mailbox Name or the Personal/Conditional Greeting to the caller.
• Erase: The system will clear the recorded Mailbox Name or the Personal/Conditional Greeting.
• Go to Previous Menu: The system will provide an option to the caller to go back to the Previous Menu —
Record Mailbox Name, Personal Greeting Timezone Selection or Conditional Greeting.

Number Programming (Assistant/Personal)
Number Programming (Assistant/Personal) is a menu which allows the caller to program the Assistant Number or
Personal Number.
• Enter Number: The system will allow the caller to program the Assistant Number or Personal Number. For
Assistant Number, only the Assistant Number List present in the system will be allowed.
Make sure the Assistance Number is an extension number and the Personal Number is an external
(Mobile) number.
Make sure you do not enter the Department Group Mailbox Number as the Assistant Number or the
Personal Number.
• Play Number: The system will play the Assistant Number or Personal Number to the caller.
• Clear Number: The system will clear the Assistant Number or Personal Number programmed by the
caller.
Management”.

Distribution List
A Distribution List enables extension users to send the same message to a group of extensions at the same time.
Any extension with a mailbox can be included in a Distribution List. You can create upto 30 Distribution Lists of 50
members each.
How to configure
To configure Distribution List using Jeeves,
• Click Distribution List.
You may add a new list, edit the default list or delete a Distribution List.
To add a new List,
• Click Add New List. A new List xx will be created. You may now configure this as per your requirement.
To delete a list, click Delete.
The default Distribution List cannot be deleted.

To edit a Distribution List,
• In Name, configure the name of the Distribution List. By default, it is List xx where xx is the Distribution List
Number from 01 to 30.
• In Access Code, configure the Access Code you wish to assign to the respective Distribution List. The
caller can send or forward the message to a Distribution List by dialing the respective access code. It can
be a maximum of 6 digits.
Make sure, the Access Codes assigned to the Distribution Lists do not conflict with the existing access
codes. System will not save the configured Distribution List access code if the same code is already
assigned.
Members Selection
• All the Extension Numbers and Department Group Numbers appear in the left side box arranged
sequentially in the increasing order.
• To select the members for the Distribution List,
• Place your cursor on the desired Extension and click Select>> button. The selected extension will
appear on the right side box. It is also possible to select a range of extensions at a time by pressing
'SHIFT’ and 'Down' arrow keys.
A maximum of 50 extensions can be selected.
• To delete the members from the Distribution List,
• Place your cursor on the desired Extension and click delete key from your keyboard. The selected
extension will be deleted. It is also possible to select a range of extensions at a time by pressing
'SHIFT’ and 'Down' arrow keys.

Recording Voice Messages
The VMS of SARVAM UCS supports voice messages for different functions, which are broadly classified as:
• System Greetings: These are messages are played to the caller when a new call lands on the VMS.
Callers are greeted according to the time of the day - morning, afternoon, evening (Time Zone). You can
customize the Time Zones as per your requirement. For detailed instructions, see “Greeting Message
Time”. A different System Greeting can also be played to callers on holidays.
System Greetings are played to callers when the VMS Auto Attendant feature is enabled on trunks.
• Personal Greetings: These messages are played to callers when they are diverted to the extension
user’s mailbox to leave a message. Extension users can record personal mailbox greeting messages of
their choice.
• Conditional Greetings: These messages are played to callers when they are diverted to the extension
user’s mailbox for certain conditions—busy, no reply or unconditional/unregistered Call Forward.
Extension users can record a different message for each condition.
• Welcome Messages: These are messages are played to callers who call the VMS. Welcome messages
help the callers navigate through the VMS. Welcome messages are played according to the time of the
day, that is, the Time Zone programmed in the system.
Welcome Messages are played to callers when the VMS Auto Attendant feature is enabled on trunks.
• Holiday Messages: These messages are played to callers who call the VMS on a Holiday. These
messages are played in place of the Welcome messages and help the callers navigate through the VMS.
A different message can be played for each holiday.
Holiday Messages are played to callers when the VMS Auto Attendant feature is enabled on trunks and
the Holiday Table is configured.
• Prompts/Responses: These are voice guidance messages that are played to the caller in response to the
action taken (that is, when the caller dials a digit).
For all of these message types, audio files containing the appropriate recorded voice guidance messages are
loaded in the configuration of the VMS Module. The VMS plays the messages related to the function it is
performing.
For example, if Voice Mail Auto Attendant (the VMS Auto Attendant feature) is enabled on a trunk, the VMS plays
messages relevant to the Voice Mail Auto Attendant Menu programmed for the current Time Zone. This helps the
caller navigate through various options as the VMS plays the related message, as explained below:
• The VMS plays the default System Greeting and Welcome message to the caller according to the time of
the day, e.g.: “Good Morning”. ““Welcome! Please dial the extension number or To dial by name press
‘6’,To leave a message press ‘7’, To access your Personal Mailbox press ‘8’, For further assistance press
‘9’, To disconnect the call press ‘#’ (hash)”.
• As the caller navigates, the VMS plays the pre-recorded voice messages related to the particular option
selected by the caller. If the caller dials 6 to dial by name, the VMS plays relevant voice message, e.g.:
"Please enter first three letters of the name" and then plays “More than one match found. Matching Names

will be played one by one. To Select the name press ‘1’, to Skip the name press ‘2’, To Repeat the last
name press ‘3’.”
• If the caller dials 1 to select the name, the VMS plays the prompt: “To confirm press ‘1’, to Re-enter press
‘2’.”
• If the caller dials 1 the VMS transfers the call as per the transfer type assigned to the selected station.
In the same way, when you set a voice guided alarm, the VMS plays the alarm-related voice prompts, like: “Enter
the time, HH MM in twenty four hour format”. Thus, for every voice mail related function or feature, the VMS plays
the appropriate voice message.
The VMS gives you the option of either using the default voice guidance messages loaded in the VMS Module
configuration, or recording custom messages that better suit your purpose.
All VMS default voice messages are in English only. If you want, you may record voice messages in your local
language.
No special programming is required for using the default voice messages. However, if you want to use custom
messages, you must first:
• record the message (Greetings, Welcome Messages, Holiday Messages, Voice Guidance prompt).
• upload the new recorded message file in the VMS configuration.

When you record messages of your choice, consider these important points:
• The custom messages must be in WAV format.
• Make sure the message files have the following attributes:

• Audio Format: CCIT u-law
• Channel: 1 (mono)
• Sampling Frequency: 8 KHz
• Audio Sample Size: 8 bit
• You must record the custom messages from an external source.
• You must verify the message and then these messages must be uploaded on to the VMS configuration
files.
• Extension users can record their personal mailbox greetings on their own. Refer the topic “Recording
Personal Greetings” to know more.
• Extension users can record the conditional greetings on their own. Refer the topic “Recording Conditional
Greetings” to know more.
Uploading Custom Voice Messages
As mentioned earlier, when you record voice messages from an external source, make sure that:
• The audio file is recorded in the prescribed format (.wav) and attributes.

• The SARVAM UCS is connected to a computer (standalone or LAN).
• You can upload a single prompt or the entire folder. For detailed instructions, see “Prompts Management”

Prompts Management
Prompts Management allows you to view all the prompt folders present in the system for each of the configured
languages. You may upload or download a prompt file to/from the VMS Prompt folders for any language. The
default prompts recorded in English language are provided to you. See “VMS Prompts” for details.
You may overwrite the default prompts or upload new prompts. While uploading the prompts for other languages,
you must record them with reference to the default prompts listed under the topic VMS Prompts.
There are 21 Prompt folders present in the system. The first 11 folders are flexible while the others are fixed.
The Flexible Folders are:
• Greeting
• Auto-Attendant
• Number Dialing
• No Digit Dialed
• Invalid Digit Dialed
• Expiry of Count
• Call Transfer Type
• Call Transfer Unsuccessful
• Information
• MoH
• Disconnect
The Fixed Folders are:
• Language
• Dial by Name
• Call Transfer
• Message Record
• Message Send Forward
• Mailbox Access
• Mailbox Access Menu
• Number
• Alarm
• Miscellaneous
The Flexible Folders include the prompt files which the system uses according to the configuration done by you.
You may upload new prompt files, download and delete the existing files from the system.
The Fixed Folders include the prompt files which the system uses internally and are non-configurable. You may
overwrite or download the existing files but cannot delete these from the system.
The following table defines the maximum number of prompt files supported in each folder.
Maximum length allowed for each folder

Folder Name Maximum Prompt files supported
(seconds)
Greeting 24 180
Auto-Attendant 99 180

Maximum length allowed for each folder
Folder Name Maximum Prompt files supported
(seconds)
Number Dialing 12 120
No Digit Dialed 12 120
Invalid Digit Dialed 12 120
Expiry of Count 12 120
Call Transfer Type 12 120
Call Transfer Unsuccessful 12 120
Information 24 180
MoH 12 180
Disconnect 12 120
Language 01 120
Dial by Name 13 120
Call Transfer 16 120
Message Record 19 120
Message Send Forward 30 120
Mailbox Access 29 120
Mailbox Access Menu 50 120
Number 46 120
Alarm 53 120
Miscellaneous 24 120
How to Configure
• Click Prompts Management.
• In Select Prompt type to manage, select the desired folder.
• Click OK.

All the prompt files present in the respective language folders of the selected Prompt Type will be
displayed.
You can upload or download the entire folder for the respective Prompt Type for each of the languages.
To add a new prompt in a Flexible Folder,
• Click the Add Prompt button at the bottom of the page.
The Add/Edit Prompt window opens.
• Click the Browse button to reach the location on the local disk where the respective prompts are stored
in your PC.
• Click to upload.
Make sure the folder you upload is a zip file.
Make sure the prompt files within the folder are in .wav format and have the following attributes:
• Audio Format: u-law
• Channel : 1 (mono)
• Sampling Frequency: 8 KHz

To Upload or Download the entire Prompt Type folder,
• Click on the Prompt Type link.
The Upload/Download Prompt Type Window opens.
• The language-wise folders will be displayed.
• Click the Browse button of the desired language to reach the location on the local disk where the
respective prompt folder is stored in your PC.
Make sure, the file to be uploaded is a zip file.

The filename can be a maximum of 64 characters.
The allowed characters for the filename are A to Z, a to z, 0 to 9, - and _.
To download all the prompts of the respective prompt type and language selected, click .

The Opening Prompt Type_Language.zip window will open.
• You can either open the zip file or save the file to a location.
The above window display depends upon the browser you are using. Check the Download Settings of
your browser and set the Download path accordingly.
OR
If your browser does not ask you for the location you want to save your file, it saves it in the default location
according to the download path specified for that browser.
If you are using Mozilla Firefox (version 3.5 recommended), before you save the configuration files, set the
Downloads option of your browser as Always ask me where to save files.

• Save the file on the local disk.
• To view the files,
• Open the Prompt Type_Language folder.

• Now, open the Prompt Type folder to view all the subfolders containing the prompts.
• Open the subfolder containing the prompt.
You may now upload, download or delete a prompt file to/from a folder. The files present in the Fixed
Folders cannot be deleted or added, but the existing files can be overwritten by a new one.

To Upload or Download a Prompt,
• Click on the respective Prompt Name link.
The Add/Edit Prompt window will open.
• To upload or overwrite a prompt,
• Click the Browse button to reach the location on the local disk where the respective prompts are
stored in your PC.
Make sure, the file to be uploaded is a wav file.

The filename can be a maximum of 64 characters.
The allowed characters for the filename are A to Z, a to z, 0 to 9, - and _.
• To download a prompt, click .

The Opening Prompt Name_xx.wav window will open; where xx signifies the prompt number.
• You can either open the wav file or save the file to a location.
OR

• To delete a prompt, click .

Mailbox Status
The Mailbox Status displays the mailbox details of the respective Extension, Department Group or General
Mailbox.
Viewing Mailbox Status

• Click Mailbox Status.
OR
• Under Voice mail, click Mailbox Status.

• Select Number: You may select — Extension, Department Group or General Mailbox — whose
mailbox status you wish to view.
• For Extension, enter the Extension Number/Name and then select the same from the drop
down list.
• For Department Group, select the desired Department Group Number from the drop down list.
• Click OK.
• Click Mailbox Status to expand.
The respective mailbox parameters along with their status will be displayed.
• Mailbox Size Consumed / Allowed (minutes): It displays the minutes consumed by the Mailbox
messages out of the maximum minutes allowed for the Mailbox.
• Mailbox Number: It displays the mailbox number assigned to the respective Extension or Department
Group or the General Mailbox.
• Redirect Set: It displays ’Yes’ if the Redirect is set and ‘No’ if the Redirect is not set for the respective
mailbox user.
• Redirect Number: It displays the Redirect number, if configured, for the mailbox user.
• Assistant Number: It displays the Assistant number, if configured, for the mailbox user.
• Mobile/Alternate Number: It displays the Mobile/Alternate number, if configured, for the mailbox user.
• New Messages / Total Messages: It displays the number of New Messages out of the Total
Messages available in the respective mailbox.

• Urgent New / Urgent Old Messages: It displays the number of New Urgent Messages as well as the
Old Urgent Messages available in the respective mailbox.
• Last Cleared On: It displays the date and time when all the Mailbox Messages (old and new) were last
cleared by the user or SE.
• Last Accessed On: It displays the date and time when the Mailbox was last accessed.

Voicemail Backup
The Voicemail Backup allows you to store the Backup of the desired Extension/s’ voicemail messages along with
the Personal Greetings and recorded Station Name on the Network Drive. For details regarding the Network Drive
configuration, see “Network Drive Settings”.
The voicemail backup is not applicable for Department Group, Operator and General Mailbox. The backup
can be taken only for Extensions.
If you want to store the backup files in a PC having Windows as the Operating System, make sure it has
IPV4 address.
Backup is not possible in Apple PCs.
How to Configure
• Click Voice Mail to expand.
• Click Voicemail Backup.
You can take the voicemail backup of a single extension, multiple extensions or all extensions.
• To backup single extension’s voicemail, select Extension. Enter the Extension Number/Name and
then select the same from the drop down list.
• To backup multiple extensions’ voicemail, select Extension Range. Enter the starting and ending
Extension Number/Name in range and then select the same from the drop down list.
• To backup all extensions’ voicemail, select All Extensions.

• Select the Delete messages after backup check box if you want the system to delete the respective
extension/s voicemail messages after their backup has been stored on the network drive.
• Network Drive Status displays the IP Address and the Folder Name configured in the Network Drive
Settings. If the parameters have not been configuration for the Network Drive, it will display the error
message here.
The Voicemail Backup of the respective Extension/s will be stored in the Shared Folder you configure in Network
Drive Settings.
The folder structure will be as shown below:
The Voicemail_Backup folder consists of the folders with the folder names defined as per the Date and Time of
the backup.
The Folder Names will be as per the Date and Time Format you configure in VMS General Parameters.
Supported formats for folder names are DDMonthname_YYYY_HHMMSS and DDMonthname_YYYY_HHMMSS_

PM/AM.
Where,
DDMonthname_YYYY represents the Date, HHMMSS represents the Time in 24 Hour format and HHMMSS_PM/
AM represents the Time in 12 Hour format.
The sub-folders with the Extension Names contain the respective extension’s voicemail files.

VMS Debug
The VMS supports debug for the VMS Application and SMTP. You can view debug messages on the Syslog
Server196.
To be able to use Syslog for debug, you will need to:

• connect a PC to the LAN/WAN Port of the ETERNITY GENX.
• select the Ethernet Port as the Destination Port.
• configure the Syslog Server Address and the Server Port on which Syslog will listen for debug messages.
Configuring VMS Debug using Jeeves

• Click the Debug.
• Select the Enable Debug check box if you wish to view the VMS Debug
• In Syslog Server IP Address, configure the IP Address of the Syslog Server.
• In Syslog Server Port, configure the address of the Listening Port of the Syslog Server. Valid port
range is: 1025 to 65535; 514. By default, the remote server port address is '514'.
If you disable the check box, the system will not send the VMS Debug to the Syslog Server.
196. The SARVAM UCS supports Syslog Client, which enables the VMS to send debug messages in syslog format to the remote 'Sys-
log Server' on the IP network. You can view the system debug messages on the remote Syslog server or any other application
which can capture the Syslog debug messages.

The system supports following VMS Debug Levels. You may select the respective checkbox of the debug levels
you wish to include in the VMS Debug.
• Select the Call check box to view the debug for all types of call flow.
• Select the Mailbox check box to view the debug for all operations related to Mailbox.
• Select the SMTP check box to view the debug related to the SMTP Client handled by VMS.
• Select the Error check box to view the debug for all types of errors.
• Select the Configuration check box to view the debug for configuration update.
• Select the Status check box to view the different status of all the VMS functions — total running calls, USB
status, pending mail notification count, total messages, etc.
Protocol
• Select the Basic check box to view the debug for data received with its IE.
• Select the Extended check box to view the debug for data received in raw hex format.
Play
• Select the Host Application check box to view the debug related to play from VMS application side.
• Select the Slave Application check box to view the debug related to play from VMS Layer side.
Record
• Select the Host Application check box to view the debug related to the records from VMS application
side.
• Select the Slave Application check box to view the debug related to the records from VMS Layer side.
Click Submit.

CHAPTER 11 SMS Server
This feature is not applicable if CDMA Mobile Card is installed in your system.
The SMS Server Application enables you to:
• Send/ receive SMS to/from individuals or groups using the Mobile Port of SARVAM UCS.
• Forward SMS received on Mobile Port as Emails to users through the Email Client.
• Forward Email of the users as SMS to the Mobile users through the Mobile Port.
• Configure Personal Directory via Email. For details, see“Configuring Personal Directory via Email” under
“Abbreviated Dialing”.
The SMS Server application works as an intermediary between the GSM Short Message Service and the SARVAM
UCS. The Server supports multipart, 7 bit text messages as well as UNICODE messages.
The Server functions as an SMTP Client to send emails and as a POP3 Client to receive emails. SMS Server
supports three types of Emails—Plain Text, HTML and MIME— from its mail clients.
To use this feature, you must purchase the SMS Server License. Refer to the topic, “License
Management” to know more.
Make sure your Email Server uses SMTP and POP3 to send/receive emails.
If you have both the SMS Gateway license and the SMS Server license, and have activated both, the SMS
Gateway will be given priority.
To forward SMS as IM on SIP Extensions and IM from SIP Extensions as SMS, see “SMS over IP”.
How it works
For this feature to work,
• you must have the SMS Server license. See “License Management”.
• make sure your Email Server uses SMTP to send messages and POP3 to receive messages.
• you must configure the SMTP Client and POP3 Client parameters in SMS Server of SARVAM UCS. See
“SMS Server - Mail Settings”
• the users must have valid Email ID’s.
• you must configure the desired Groups and assign them to the extension users and/or Global/Personal
Directory contacts. See “SMS/Email Group”.
• configure the SMS parameters for the desired extension users and/or Global/Personal Directory contacts,
that is, Mobile Number, Email ID and SMS/Email Group. See “Configuring SLT Extensions”, “Configuring
DKP Extensions”, “Configuring ISDN Terminals”, “Configuring Matrix SPARSH VP330”, “Configuring

Matrix VARTA ADR100/AMP100 UC Clients”, “Configuring Matrix SPARSH VP248”,“Configuring Matrix
SPARSH VP310”, “Configuring Matrix SPARSH VP510”,“Configuring Personal Directory using Jeeves”
and “Configuring Global Directory using Jeeves” under “Abbreviated Dialing”.
• you must define the Mobile port through which the messages are to be sent/received (Fixed/LCR). See
“SMS Routing”.
• configure the SMS parameters and the SMS Budget parameters (if required) on the respective Mobile
ports, see “Configuring Mobile Trunks”.
• you must configure the SMS Server parameters as well as the multipart SMS parameters for sending/
receiving SMS (if required). See “How to Configure”.
Forwarding an incoming SMS as Email

• To forward an SMS to an Email ID, the SMS received on the Mobile Port must be in a specific format.
Illustrated below is as example of the format.
Here,
• The sender of the SMS is John (9898214345).
• The sender must send the message to the SIM Number of the Mobile Port of SARVAM UCS, that is
9898012345.
• The body of the message must contain the following:

• To: This is the destination where the Email is to be sent. In this case, James@matrixrd.org, Smith,
2001.
• Sub: This is the message that will be displayed to the recipients. In this case, ‘Meeting at 10.00 a.m.
tomorrow’.
• When the SMS is received on Mobile Port (9898012345) in the above format, the system checks the
senders number (9898214345 - John) in the Denied numbers list of the SMS Server.

If the SMS is received in multiple parts the SMS Server combines it into a single SMS.
• If a match is found in the Denied list, the SMS will be rejected. If no match is found, then it checks the
destination where the SMS is to be forwarded as an email.
The destination can be the recipients Extension number, Name/Group Name or Email ID. The To field in
the message body specifies the destination. To send the same message to multiple destinations, enter the
destinations separating them with a comma or semicolon.
When the destination is an Email ID, the system will forward the SMS to the recipients Email ID. In this
case, James@matrixrd.org
When the destination is a Name or Extension Number, the system will search for the Name/Number in its
database. When a match is found, the message will be sent to their corresponding Email IDs. In this case
the Email ID of Smith is Smith@matrixrd.org and the Email ID of the Extension user 2001 is
Bella@matrixrd.org.
If a match is not found for the Email ID, Name/Group Name or Extension Number, by default the system
rejects the message. The system also provides you the option to send the message to a specific recipient
(Send to Default recipient), if you do not want to reject the message.
• The SMS Server will convert this SMS to an Email. The email will be sent by the SMS Server to the Email
Server and the Email Server will finally deliver it to the recipients.
• When the recipients download their emails, it will be as per the format displayed below.

In this Email, the From field contains the Email address of the SMS Server. The To field contains the Email IDs of
the recipients, the Subject contains the Name and/or Number of the sender and the Body of the message contains
the message for the recipients.
The Name will be displayed only if it has been configured and it is found in the system database.
Forwarding an incoming Email as SMS

• To forward an Email as an SMS, the sender must send the Email to the SMS Server in a specific format.
Illustrated below is as example of the format.
Here,
1. The sender of the Email is John, John@matrixrd.org.
2. The To field must be the Email ID of the SMS Server, smsserver@domain.com.
3. The Subject must contain the destination where the SMS to be delivered.
The destination can be a Name/Numbers of the users to whom the SMS is to be sent. To send the SMS to
multiple destinations, enter the destinations separating them with a comma or semicolon. Here the
destination, that is the recipients are James, Smith, Bella, 9898981212, 9845544544.

4. The body must contain the message to be sent to the recipients, that is ‘Due to an urgent task tomorrow’s
meeting is postponed.’
5. The senders Email ID or Name will be displayed to the recipients, if you have configured the parameter
Send Footer/Signature in SMS.
It is recommended that you configure the parameter Send Footer/Signature in SMS, so that the recipient
knows the sender of the message. If this parameter is not configured, only the mobile port SIM number will
be displayed to the recipient.
• This mail will be sent to the Email Server and then the Email Server forwards it to the SMS Server.
• The SMS Server can receive emails from extensions users (system users) or from external users. You can
allow or deny emails from users as per your requirement.
If you want to receive Emails from extension users only, select the Enable Email to SMS forwarding
check box.
If you want to receive emails from extension users as well as external users, select the Enable Email to
SMS forwarding for External Users check box.
• Then, the Server checks the Email ID of the sender in the Denied Email list.
• If a match is found in the Denied list, the Email will be rejected. If no match is found, then it checks the
destination where the SMS is to be forwarded.
The destination can be the recipients Number or Name. The Subject field of the message specifies the
destination. To send the same SMS to multiple destinations, enter the destinations separated by comma or
semicolon.
If the destination is a Number, the server will check the number in the Denied list. If a match is found the
email will be rejected. If a match is not found, the Server will send the SMS directly to the number using the
Mobile Port of SARVAM UCS.
If the destination is a Name, the system will search for the Name in its database. When a match is found,
the SMS will be sent to the corresponding Number. In this case, the Numbers of James, Smith, Bella.
The Server sends a return email as well as a delivery status report to the SA/Sender or to Both, informing
that the SMS has been delivered/not delivered.
If a match is not found for the Name, a reply mail is sent to the Sender, informing that the Name is not
found.

The SMS Server converts the Email to a SMS. In this case the SMS delivered to each recipient, will appear
as given below.
From:9898012345
Due to an urgent task tomorrows
meeting is postponed.
From:John@matrixrd.org
Options Cancel
In the SMS sent to the recipients, the From field contains the SIM Number through which the SMS Server sent the
SMS. The Body of the message contains the message for the recipients and the Footer/Signature that is, the Email
ID/Name of the sender (if the Signature has been added).
Bulk SMS
The SMS Server supports Bulk SMS, that is a single message can be sent to multiple numbers. The message can
be sent using the email client by specific users.
How it works
• make sure you have enabled Email to SMS Forwarding. See “Email to SMS Forwarding”.
• configure the Minimum time delay required between sending two consecutive SMS. See “SMS
Configuration”.
• configure the Bulk SMS parameters and the Email IDs of the users allowed to send Bulk SMS. See “Bulk
SMS”
• you must define the Mobile port through which the messages are to be sent (Fixed or LCR). See “SMS
Routing”
• make sure your Email Server uses SMTP to send messages and POP3 to receive messages.
• you must configure the SMTP Client and POP3 Client parameters in the SMS Server of SARVAM UCS.
See “SMS Server - Mail Settings”
• users must have valid Email IDs.

To send the Bulk SMS the email must be received in the following format:
• the email must be sent with the Subject “Bulk SMS”.
• the content received in the email will be considered as the SMS text.The text in the email must not exceed
160 characters.
• the mail must contain an attachment of the numbers only or numbers and names in csv format. The csv file
can have a maximum of 1000 numbers. The csv file can have two columns Name and Number or have
only one column with numbers. Make sure the columns do not have any header.
When the system receives the email from the user who has requested for Bulk SMS, the system will check this
Email ID in the Allowed Email IDs List for sending Bulk SMS. If a match is found, the system will serve the request.
If a match is not found the Bulk SMS request will be rejected.
The system will check the Allowed Email IDs for sending Bulk SMS list, only if you have enabled the
Allowed Email IDs to send Bulk SMS check box. If the Allowed Email IDs for sending Bulk SMS
option is disabled, all the system users (extension users) will be able to send Bulk SMS requests.
The system will serve only one Bulk SMS request at a time. If one Bulk SMS is in progress and the system receives
another request, the system will reject it. Bulk SMS supports a single SMS of 160 characters only. It does not
support multi-part SMS.
While the Bulk SMS request is in progress, the system also provides you the option to stop the process at any point
of time if required. The sender must contact the System Administrator to do so.
After the Bulk SMS request has been served, the system updates the csv file and adds another column, Status. For
each entry the Status of the SMS is updated. The Status column may contain any one of the following:
• Sent - When any SMS is sent successfully by the system i.e OK is received from GSM engine.
• Failed - When GSM engine gives any error response
• Invalid - For any entry where data is found other than number in the .csv file
• Denied - When SMS number is found it the Denied list
• Limit Exceed - When Daily or Monthly limit is exceeded for all the Mobile ports used for sending the SMS
for any entry.
• No Port Available - When all Mobile Ports are configured as "Not allowed to send SMS" which are to be
used for sending the SMS for any entry.
This report is then emailed to the user who had requested for Bulk SMS. The Subject of the report email is the
same as received for Bulk SMS request.
In certain cases the system may not be able to serve the Bulk SMS request. In such cases the system sends a
reply email with the Error message to the user who requested for Bulk SMS. The possible Error conditions and
messages are mentioned below:
Condition Error Message
If Mail Subject is written as “bulk SMS” (case insensitive) No attachment found to process bulk SMS.
but No file is attached in Mail Please attach valid “.csv” file for Bulk SMS.
If multiple attached file is received Only one attachment is allowed.

Attached file format should be ".csv".
Attachment file size should be less than 1 MB.
If multiple attached file is received and from that multiple Only one attachment is allowed.
attached file one file format is .csv Attached file format should be ".csv".
Attachment file size should be less than 1 MB.

If attached file format is found to be other than .csv No attachment found to process bulk SMS.
Please attach valid “.csv” file for Bulk SMS.
If attached file is .csv but file size is more than we There is No such Error cause for this condition.
support. (i.e 2MB) In this case, the incoming email will get ignored.
If attached file format is found to be .csv but in subject it Attachment is not allowed for normal Mail to SMS.
is not received as "Bulk SMS" (string required for this
Bulk SMS feature) and not applicable for Master also.
If attached file format is found to be .csv, subject is Bulk SMS feature is disabled
matched for Bulk SMS but feature is disable
If attached file format is found to be .csv, subject is User not allowed to send Bulk SMS
matched for Bulk SMS but user is not allowed to send
Bulk SMS (user means from which email is received)
If attached file format is found to be .csv, subject is More than 160 character text is not supported for Bulk
matched for Bulk SMS, user is allowed Bulk SMS feature SMS
(user means from which email is received) but Text data
contains more than 160 characters (i.e consider that text
data is such that sms is to be send in multi-part)
If attached file format is found to be .csv, subject is Invalid ".csv" file.It should contain data only in first 2
matched for Bulk SMS, user is allowed Bulk SMS feature columns (i.e.Name, Phone Number or Phone Number,
(user means from which email is received), Text data is Name) without any header.
also for single part sms but attached file data is not as
required.
If already one .csv file for Bulk sms is in process and Already one file is in process. Please send the file again
other attached valid file is received. after x minute(s) approximately.
Here x is the time duration for which current csv file will
be processed.
If Bulk SMS file is processed, send report Please find the attached report of Bulk SMS file "xxx.csv"
received on "Date-Time" .
where xxx.csv should be replaced with filename

Date-Time should be replaced with the date and time
when email was received by system.
How to Configure
For the SMS Server to function, you need to configure the following parameters,
General Parameters
• Under Configuration, click SMS Server.
• Click General Parameters. The SMS Server-General Parameters page opens.

• By default, the SMS Server is disabled. To use the SMS Server feature, select Enable.
SMS Configuration
Click SMS Configuration to expand.
Configure the SMS related parameters:
• You can Send SMS through certain fixed Mobile Ports or through different Mobile Ports according to
specific numbers and time.
To send messages through fixed Mobile Ports, select Using Fixed Mobile Port and configure the SMS
Routing-Fixed Port table. For detailed information, see “Fixed Port Routing (SMS Server)”.
To send messages to specific numbers through certain preferred Mobile Port/s during a defined time
interval, select Based on specific Time-Number and configure the SMS Routing-LCR table. For detailed
information, see “Least Cost Routing”.
Default: Using Fixed Mobile Port.
• Configure the Allowed-Denied Numbers for sending SMS list, if you want to allow or restrict sending of
messages on specific numbers. To do this,
• Select the Allowed-Denied Numbers for sending SMS check box.
• Click on Allowed-Denied Numbers for sending SMS and the Allowed-Denied Numbers for
sending SMS table opens. You can configure upto 250 numbers.

• In the Allowed column, enter the numbers on which messages can be sent and in the Denied column,
enter the numbers on which messages cannot be sent.
• You can also configure this list by clicking the Allowed-Denied Numbers for sending SMS link under
SMS Routing.
Default: Disabled
• Configure the Allowed-Denied Numbers for receiving SMS list, if you want to allow or restrict receiving
messages from specific numbers. To do this,
• Select the Allowed-Denied Numbers for receiving SMS check box.
• Click on Allowed-Denied Numbers for receiving SMS and the Allowed-Denied Numbers for
receiving SMS table opens. You can configure upto 250 numbers.
• In the Allowed column, enter the numbers from which messages can be received and in the Denied
column, enter the numbers from which messages cannot be received.
• You can also configure this list by clicking the Allowed-Denied Numbers for receiving SMS link
under SMS Routing.
Default: Disabled
If you want to allow multipart messages configure the following parameters:
• Select the Number of parts allowed for sending SMS from the list. Default:1
• If the SMS length is more than number of parts allowed, you can select either Ignore remaining
part of the SMS or Do not send SMS and send error report to sender. Default: Do not send SMS
and send error report to sender.
• If you select Ignore remaining part of the SMS, you can select the Send error report to sender
when SMS length is more than allowed parts check box, if you want to send an error report to the
sender.
• If you enable Send error report to sender when SMS length is more than allowed parts, enter the
message you want to send to the sender in the email in Reply error report as an Email to sender
containing text. Default text: Some texts of message are ignored as length of mail is more than
allowed characters.
• Configure the Minimum time delay between sending two consecutive SMS (sec) as supported by the
network. Default: 05.
Click Submit to save the changes you made.

SMS to Email Forwarding
Click SMS to Email Forwarding to expand.
• By default the Enable SMS to Email forwarding is selected (enabled). If you do not want the SMS Server
to forward the SMS as Emails to the users, clear the check box.
• Configure the Default recipient of Email when recipient is not specified in SMS. When an SMS is
received without any recipient, the system delivers the SMS as an Email to the default recipient/s
configured here. You can configure upto 5 Email IDs. Default: Blank.
• Select the desired option for, When recipient is specified in SMS then forward Email to. You can select
Recipient specified in SMS or Default Recipient. Default: Recipient specified in SMS.
If you want all incoming SMS to be delivered as Email to a specific recipient only, select Default
Recipient.
If you want the incoming SMS to be delivered as Email to the recipients specified in the SMS, select
Recipient specified in SMS.
• Select the desired option for, If recipient (Name/Number/Email ID) not found in database. You can
select Reject SMS or Send Email to default recipient. Default: Reject SMS.
• Select the desired option for, If conflict occurs for recipient Name. You can select Reject SMS or Send
Email to default recipient. Default: Reject SMS.
• Enable the Send copy of each SMS to System Administrator as an Email check box, if you want to
send a copy of SMS as an Email to the System Administrator. Default: Disabled.
If you enable this check box, you must configure atleast one Email address of the System Administrator
under “System Administrator Email ID”.
Click Submit to save the changes you made.

Email to SMS Forwarding
Click Email to SMS Forwarding to expand.
• By default the Enable Email to SMS forwarding check box is selected (enabled). Emails received from
extension users only will be forwarded as SMS. If you do not want the SMS Server to forward Emails as
SMS to the users, clear the check box.
• By default the Enable Email to SMS forwarding for External Users check box is clear (disabled). Select
this check box, if you want the SMS Server to receive Emails from extension users as well as external
users and then forward them as SMS.
• Configure the Allowed-Denied Email IDs to send SMS list, if you want to allow or restrict certain Email
IDs to send SMS. To do this,
• Select the Allowed-Denied Email IDs to send SMS check box.

• Click on Allowed-Denied Email IDs to send SMS and the Allowed-Denied Email ID list table opens.
You can configure upto 500 Email IDs.
• Select the desired option Allow all except programmed in Denied List or Deny all except
programmed in Allowed List.
• If you select Allow all except programmed in Denied List, in Denied Email ID column, enter the
Email IDs from which SMS cannot be sent.
• If you select Deny all except programmed in Allowed List, in Allowed Email ID column, enter the
Email IDs from which SMS can be sent.
• You can also configure this list by clicking the Allowed-Denied Email ID List link under SMS Server.
Default: Disabled.
• Select the Send an Email if SMS is delivered to phone check box, if you want a confirmation email to be
sent when the delivery report is received by the Server from the network. Default: Disabled.
• If you have enabled Send an Email if SMS is delivered to phone, in Send delivery report status to,
select the recipient to whom the delivery status report must be sent. You can select Sender, System
Administrator or Both. Default: Sender.
• If you select System Administrator, the email will be sent to the email IDs configured in “System
Administrator Email ID”.

• In Email Reply Text, enter the message you want to send in the Email. The message can a maximum
of 100 characters. Default text: Your Message is delivered to Receiver.
• Select the Send an Email if SMS is not delivered to phone check box, if you want a confirmation email
to be sent when the delivery report is not received by the Server from the network. Default: Disabled.
• If you have enabled Send an Email if SMS is not delivered to phone, in Send delivery report
status to, select the recipient to whom the delivery status report must be sent. You can select Sender,
System Administrator or Both. Default: Sender.
• If you select System Administrator, the email will be sent to the email IDs configured in “System
Administrator Email ID”.
• In Email Reply Text, enter the message you want to send in the Email. The message can a maximum
of 100 characters. Default text: Your Message is not delivered to Receiver.
• Enable the Send Footer/Signature in SMS check box and select the desired option to be sent:
• Specific Text: Enter the text/message to be sent to the recipient as signature in the SMS.
• Send Email ID of Sender: Enter the Email ID of the Sender to be sent to the recipient as signature in
the SMS.
• Send Name of Sender: Enter the Name of the Sender to be sent to the recipient as signature in the
SMS.
Click Submit to save the changes.
Bulk SMS
Click Bulk SMS to expand.
• By default Bulk SMS is disabled. If you want the SMS Server to send Bulk SMS, select the check box.
• Select the Allowed Email IDs to send Bulk SMS check box and configure the Allowed Email IDs to
send Bulk SMS list. Default: Disabled. Only these users will be able to send Bulk SMS. To configure the
Email IDs,
• Click on Allowed Email IDs to send Bulk SMS and the Allowed Email IDs for sending Bulk SMS
table opens. You can configure upto 64 Email IDs.
• Enter the Email IDs of the users who can send Bulk SMS.
• You can also configure this list by clicking the Allowed Email IDs for sending Bulk SMS link under
SMS Server.

• Click Submit to save the entries.
Error Cause List

There are different activities/events/error conditions handled by the SMS Server. The Server sends an SMS/Email
as reply for each of these to the sender.
Refer to the table below, to known the possible Condition/Activity/Event:
Error Cause Condition/ Activity/Event Email reply text
Error Cause 1 When Email user sends Email to SMS Server You are not allowed to access SMS
for sending SMS and Email ID is Server features. Please consult your
programmed in Denied List, then reject administrator.
Email.
Error Cause 2 When Number is programmed in Denied List You are not allowed to send SMS on this
for Sending SMS/Receiving SMS, then reject Number <X>. Please consult your
Email (for email to SMS query) or Reject administrator.
SMS (for SMS to email query). Where <X> is the number.
Error Cause 3 When Email user sends Email to SMS Server "Name <X>" is not found in Directory
for sending SMS and Name is written in mail Where <X> is name written as recipient in
but name is not found in any Directory; then subject line.
reject Email.
Error Cause 4 If conflict occurs for the Recipient Name in Conflict occurred for "Name <X>" in
Directory (SMS to Email or Email to SMS), Directory. Please check the Name.
then reject it. Where <X> is the Name written as
recipient by sender.
Error Cause 5 If Number is not programmed for the Number is not found for Name <X> in the
Recipient Name in Directory request (SMS to Directory.
Email or Email to SMS), then reject it.
Error Cause 6 When there is an Email for SMS forwarding No port is programmed to send SMS.
but there is no port is programmed for Consult your System Administrator.
sending SMS, request for sending SMS is
rejected.
Error Cause 7 When Email User sends Mail to SMS Server No contact is added. Please add at least
having subject line blank, reject this mail. one contact.
Error Cause 8 When Email user sends Email to SMS Server Your mail is large to send the SMS,
for sending SMS and length of message please shorten your mail or contact your
body is more than allowed characters. administrator.
The email will be rejected or the remaining
part will be ignored, depending on the
configuration.
Error Cause 9 When SMS credit exceeds, send email to SMS Limit Exceeds. Please consult your
sender informing about the status. System Administrator.
Error Cause 10 When Daily SMS credit exceeds, send email Daily SMS Limit Exceeds. Please consult
to sender informing about the status. your System Administrator.

If required you can modify the Email reply text as per your requirement. To do this,
• Click the Error Cause List link to expand.
• The Error Cause table displays the Error Cause number with the corresponding Email reply text.
• Select the Error Cause Number for which you need to edit/change the reply text.
• In Email reply text, enter the desired text.
• Click Submit.
You can generate a report of all the errors, see “SMS Server Reports” for more information. These Error Causes
are also logged into the Fault Log. See “System Fault Log” for more information.
System Administrator Email ID

Click the System Administrator Email ID link to expand.

• You must Configure the System Administrator Email ID to which a copy of SMS must be sent by the
SMS Server, if you have enabled Send copy of each SMS to System Administrator as an Email check
box under “SMS to Email Forwarding”.
You can configure upto 5 Email ID’s. The Email IDs can be of maximum 64 characters.

SMS Routing
The SMS Server can send SMS using any of the following methods:
• Fixed Port Routing - through a single/fixed group of Mobile Ports.
• Least Cost Routing - through selective preferred Mobile Ports grouped together in order to utilise the
benefits offered by the service providers, such as 1000 free SMS in a month, reduced rates to send
messages etc.
The system allows you to configure different Fixed Port Routing Tables for the SMS Server application and SMS
over IP application. But, the Least Cost Routing table is common for both these applications.
Fixed Port Routing (SMS Server)

You can route the SMS through a single or fixed group of ports. To do this, you must determine the ports through
which you want to send the SMS and then configure the SMS Routing-Fixed Port table.
Configuring Fixed Port Routing

• Click General Parameters.
• Click SMS Configuration to expand.
• As the Send SMS option select Using Fixed Port.
• Click on the Send SMS link, the SMS Routing-Fixed Port table opens.

The number of Mobile Ports displayed, depend on the variant of SARVAM UCS.
• Rotation: By default, all the messages will be sent using the first Mobile Port enabled by you.
Select the Rotation check box, when you want the first SMS to be sent through the first Mobile Port,
the subsequent SMS through the next Mobile Port and so on. For example, if three Ports MOB Port 1,
MOB Port 2, MOB Port 3 are selected and there is a request to send 5 SMS. Then, the first SMS will be
sent through MOB Port 1, second SMS through MOB Port 2, third SMS through MOB Port 3, fourth
SMS through MOB Port 1 and so on.
If the Rotation check box is cleared, all the SMS will be sent using the first Mobile Port enabled by you.
For example, if three Ports MOB Port 1, MOB Port 2, MOB Port 3 are selected and there is a request to
send 5 SMS. Then, each SMS will be sent through MOB Port 1 only.
• Port No.: This is the Mobile Port number.
• Name: This is the name assigned to the Mobile Port. This will be displayed only if you have assigned a
name to the Mobile Port on the Mobile Port Parameters page. See “Configuring Mobile Trunks”.

• Enable: Select the Enable check box corresponding to the Mobile Port you want to use for sending the
SMS.
• Click Submit.
Make sure that you have selected the Send SMS check box for this Mobile Port. See “Configuring Mobile
Trunks”.
Fixed Port Routing (SMS over IP)

You can route the SMS through a single or fixed group of ports. To do this, you must determine the ports through
which you want to send the SMS and then configure the SMS over IP-Fixed Port table.
Configuring Fixed Port Routing

• Click SMS over IP Settings.
• As the Send SMS option select Using Fixed Mobile Port.
• Click on the Send SMS link.
• The SMS over IP-Fixed Port Routing table opens

• Rotation: By default, all the messages will be sent using the first Mobile Port enabled by you.
Select the Rotation check box, when you want the first SMS to be sent through the first Mobile Port,
the subsequent SMS through the next Mobile Port and so on. For example, if three Ports MOB Port 1,
MOB Port 2, MOB Port 3 are selected and there is a request to send 5 SMS. Then, the first SMS will be
sent through MOB Port 1, second SMS through MOB Port 2, third SMS through MOB Port 3, fourth
SMS through MOB Port 1 and so on.
If the Rotation check box is cleared, all the SMS will be sent using the first Mobile Port enabled by you.
For example, if three Ports MOB Port 1, MOB Port 2, MOB Port 3 are selected and there is a request to
send 5 SMS. Then, each SMS will be sent through MOB Port 1 only.
• Port No.: This is the Mobile Port number.
• Name: This is the name assigned to the Mobile Port. This will be displayed only if you have assigned a
name to the Mobile Port on the Mobile Port Parameters page. See “Configuring Mobile Trunks”.
• Enable: Select the Enable check box corresponding to the Mobile Port you want to use for sending the
SMS.
• Click Submit.
Make sure that you have selected the Send SMS check box for this Mobile Port. See “Configuring Mobile
Trunks”.
Least Cost Routing

Least Cost Routing (also referred to as Automatic Route Selection) is an expense control feature of SARVAM UCS.

Least Cost Routing (LCR) is useful when there are different Mobile Ports for sending messages, and the service
providers offer different schemes for SMS. These schemes may be for certain numbers or during a particular time
of the day.
When an SMS is sent from the system, LCR recognizes where the SMS is going to be delivered. It selects the
lowest cost port from among all the ports allotted for sending SMS, depending upon how the LCR is configured.
The system can be configured to select the most cost effective trunk for the time of the day when the SMS is sent,
or to select the most cost effective trunk for the destination number to which the SMS is sent, or to select the most
cost effective trunk considering both time of the day and destination number.
You can configure Time or Number or both as per your requirement.
• If you have configured only the Time, the system will check the time while sending the SMS and then route
it according to the selected preference. For example you want to send SMS from one group of trunks
during 9:00 a.m. to 2:00 p.m. and from 3:00 p.m. to 8:00 a.m. through another group.
• If you have configured only the Numbers, the system will check the destination number while sending the
SMS and then route it according to the selected preference. For example SMS to numbers that begin with
99 and 97 can be routed through different trunk groups.
• If you have configured both, the time and number, the system will check the time as well as the destination
number while sending the SMS and then route it according to the selected preference.
Configuring LCR
• Under Configuration, click SMS Routing.
• Click Least Cost Routing. The SMS Routing - LCR table opens.

• Time Zone 1 to 4: Configure the Start Time and End Time for each time zone. You can configure four
different time zones—Time Zone1, 2, 3 and 4 as per your requirement.
• Number: Configure the destination numbers to which the messages are to be sent.
• Preference1 to 4: Select the trunks in the order of preference through which you want to send the SMS.
You can select upto 4 preferences.

SMS Server - Mail Settings
The Server functions as an SMTP Client to send emails and as a POP3 Client to receive emails. The SMS Server
will be able to send/receive emails only after you have configured the relevant SMTP and POP3 parameters in the
SMS Server.
How to configure
• Click Mail Settings.
SMTP Configuration
Contact your Network Administrator for the following information and configure the parameters as per the
configurations done in the Email Server to register the SMS Server as an SMTP Client.
Click the SMTP Configuration link to expand.
• In Use SMTP Account select the account you want the SMS Server application to use.
You may add a new SMTP Account. To do so,
• Select Add New option for Use SMTP Account.
• Click Settings to configure the parameters of the New SMTP Account you created. For more
information, see “SMTP Settings”.
Click Submit to save settings.
POP3 Configuration
Contact you Network Administrator for the following information and configure the parameters as per the
configurations done in the Email Server to register the SMS Server as a POP3 Client.

Click the POP3 Configuration link to expand.
• If the Email Server uses authentication, select Requires Authentication as Yes. Default: No. If your Email
Server uses authentication, you must also configure the User ID and the Password.
• To transport all data in a secure manner, select Enable Secure Socket Layer (SSL) as Yes. All the data
to the Email Server will be transported over secure layer. Default: No.
• If you have enabled authentication, configure the User ID and the Authentication Password as provided to
you by your network administrator. The User ID may consist of a maximum of 40 characters and the
Password can be a maximum of 24 characters. Default: Blank.
• Configure the POP3 Server Address and POP3 Server Port. This is the Server’s IP Address and Port
number that is used to download incoming mails. Both IPv4 and IPv6 addresses are supported. For
example, Email Server address is 192.168.1.1 and port is 1400, then configure the Server Address as
192.168.1.1 and port as 1400. If port is not programmed, use the default port value equal to 110. Valid Port
range: 110;995;1025 to 65535. The Server Address can be a maximum of 46 characters.
Click Submit to save settings.
Timer
• Download interval Timer (min.) is the time interval after which the POP3 Client of the SMS Server retries
to fetch new mail from the Email Server. Valid Range is 01 to 99 minutes. Default: 01 minute.
Test POP3 Settings

• After you have configured the POP3 parameters and submitted them, the Click to Test POP3 Settings
button appears.
• Click the 'Click to Test POP3 Settings’ button to check if the POP3 parameters have been configured
correctly.
When you click this button, the alert message appears: "Testing POP3 can take up to 99 seconds. Would
you like to continue?" Click the OK button.
The message "Please refresh the web browser after few seconds to check the test mail status" appears.
Click OK button.
Refresh the web browser after a few seconds. The Test Result will be displayed in the 'Test Status' field.

• Test Status: Any one of the results listed below may appear in this field:
Test Status Message Description

"Login to POP3 Mail Server is Successful" When connection to POP3 server is established
successfully.
"Login to POP3 Mail Server is Failed" When connection to POP3 server is not
established successfully

SMS Server Reports
The SARVAM UCS maintains a database with details of all the Email and SMS transactions as well as the Errors
that occurred during these transactions.
The system has a buffer for 999 successful SMS/Email transactions and 100 Error transactions. When the buffer is
full, the system overwrites these transactions on First In First Out (FIFO) basis. The buffer can be cleared at any
time from the System Administrator mode.
You can generate different type of reports by setting the desired filters. SARVAM UCS can generate reports
whenever you want or you can obtain it Online, immediately after the Email or SMS has been sent/received.
You can generate Reports, either
• Manually: The report is generated whenever you want.

Or
• as per Schedule: The report is generated on a preset Day, Date and Time.
SARVAM UCS allows you to set a variety of filters for printing the SMS Server Reports. SARVAM UCS supports
Syslog Client for SMS Server Reports. The Syslog Client enables the system to send records in syslog format to
the remote ‘Syslog Server’. You can view the records on the remote server and print.
The Report contains the following information for each transaction:

• Index
• Email
• Direction
• Email Address
• Status
• Date
• Time
• SMS
• Direction
• Number
• Status
• Date
• Time
• Part of SMS
• Mobile Port
• Text
To generate the reports you must configure the following parameters:

• General Report Settings
• Scheduled Report Settings
• Error Report Settings
• Report Backup Parameters
• Report Filters

Configuring Report Parameters
• Click Report Settings.
• Under General Report Settings,
• Select Ethernet as the Destination Port.
• In Destination IP Address, enter the IP Address of the remote Syslog Server. Both IPv4 and IPv6
addresses are supported.
• In Port, enter the port of the remote Syslog Server. Valid port range is: 514; 1025 to 65535. Default:
514.
• Under Scheduled Report Settings,
• If you have opted for Scheduled Reports, in When scheduled backup is done, send an email to,
enter the desired email ID. The report will be generated and sent to this email ID.
To select the Scheduled Report options log into the SA mode.
• Under Error Report Settings,
• Select Ethernet as the Destination Port.

• In Port, enter the port of the remote Syslog Server. Valid port range is: 514; 1025 to 65535. Default:
514.
How to use
You can print Reports whenever you want or schedule printing of the report from the System Administrator
mode.
You must set the filters as per your requirement before you print the Report. To do this,
• Open Jeeves.
• Click SMS Server to expand.
Settings Report Filters
• Click Filter Report to set the various filters as per your requirement:
• Select the Direction. You can select SMS to Email or Email to SMS or Both.
• Select the Mobile Port/s using which the SMS are sent/received. You can select the desired range in
the From and To fields.
• Select the Dates during which the SMS/Emails are sent/received.You can select the desired range in
the From and To fields.

• Select the Time duration during which the SMS/Emails are sent/received.You can set the desired
range in the From and To fields.
• Select the type of SMS Status. You can select from the following:
• All
• Pending
• Sent
• Delivered
• Not Delivered
• Received
• Failed
• Select the type of Mail Status. You can select from the following:
• All
• Sent
• Received
• Pending
• Failed
• The minimum and maximum number of parts in which an SMS can be sent is from 1 to 8. Select the
Part of SMS for which you want to generate the report. After you have selected the Part of SMS value,
select the desired filter—All, Equal to, Less Than or More than—to be applied to that value. For
example is you select 5 as the Part of the SMS and More than as filter, the report will be generated for
all the SMS sent in more than 5 parts.
• If you want reports to be generated for certain numbers, select the Filter Numbers check box and
configure the desired numbers in the table.
• If you want reports to be generated for certain Email IDs, select the Filter Email Ids check box and
configure the desired Email IDs in the table.
• Click Submit.
• To view the report, click the Generate Report button.
• The report is displayed in a new window.
Online Report Generation

• Click the Report under SMS Server.

• To Generate online report, click the Start button.
• To stop printing, click the Stop button.
Scheduled Report Generation

• To Generate schedule backup, select the desired option a particular day, day of the week, or day of the
month.

Manual Report Generation

• To generate the report manually, click the Report under SMS Server.
• To Generate report now, click the Generate button.
• The report is generated and appears in a new window.
• Click Export, if you wish to save the report at the desired location.
The report will be saved in CVS format.

Online Error Report Generation
• To Generate Online error report, click the Start button.
• To stop printing, click the Stop button.
Bulk SMS Status

• In Current status of Bulk SMS process, the status of the Bulk SMS process is displayed.
• If the Current status of Bulk SMS process displays Running, click the Abort button to stop the ongoing
process midway.
To clear all the database, click the Clear Database button.

SMS over IP
The Unified Messaging Functionality of SARVAM UCS includes SMS over IP.
SMS over IP allows you to:
• Forward SMS received on Mobile Port from any extension user as IM to Matrix VARTA UC Users.
• Forward IM from Matrix VARTA UC Users as SMS to any extension user through the desired Mobile Port.
How it works
• you must define the Mobile port through which the messages are to be sent/received (Fixed/LCR). See
“SMS Routing”.
• configure the SMS parameters and the SMS Budget parameters (if required) on the respective Mobile
ports, see “Configuring Mobile Trunks”.
• enable SMS over IP for Matrix VARTA UC Users. See “Configuring Matrix VARTA ADR100/AMP100 UC
Clients”.
• configure the SMS over IP Settings. See “How to Configure”.
SMS over IP Functionality requires SMS Server License. Make sure you purchase the license for using
this feature. For more information, see “License Management”.
Forwarding an incoming SMS as an IM

• To forward SMS as an IM, the SMS received on the Mobile Port must be in a specific format. Illustrated
below is as example of the format.

Here,
• The sender of the SMS is John (9898214345).
• The sender must send the message to the SIM Number of the Mobile Port of SARVAM UCS, that is
9898012345.
• The body of the message must contain the following:

• To: This is the destination where the IM is to be sent. In this case 2001.
• Sub: This is the message that will be displayed to the recipients. In this case, ‘Meeting at 10.00 a.m.
tomorrow’.
• When the SMS is received on Mobile Port (9898012345) in the above format, the system checks the
senders number (9898214345 - John) in the Denied numbers list of the SMS Server.
If the SMS is received in multiple parts, the first 160 characters only will be sent as an IM.
destination where the SMS is to be forwarded as an IM.
The destination can be the recipients Extension number or Name. The To field in the message body
specifies the destination.
You cannot send the same message to multiple destinations at the same time. You must send the same
message to individual recipients separately.
• When the destination is an Email ID, the further handling of the SMS will be done by the SMS Server
Application. See “SMS Server” for more details.
• When the destination is an Extension Number, the system will search for the Extension Number in its
database. When a match is found and it is a SIP extension, the system will check if the option SMS
over IP is enabled for this SIP extension. If it is enabled the SMS will be forwarded as an IM. If this
option is disabled, the further handling of the SMS will be done by the SMS Server Application. See
“SMS Server” for more details.
When the destination is an Extension Number other than SIP extension, the further handling of the
SMS will be done by the SMS Server Application. See “SMS Server” for more details.
• When the destination is a Name, the system will search for the Name in its database. When a match is
found, the system will check the corresponding Extension number. If it is a SIP extension the SMS will
be forwarded as an IM. If the Extension Number is any other extension or if a match is not found the
further handling of the SMS will be done by the SMS Server Application. See “SMS Server” for more
details.

Forwarding an IM as an SMS
You can send a message as an IM or SMS using Matrix VARTA Mobile UC Clients only.
To forward an IM as an SMS, refer to the illustration below:
Here,
• The SIP extension user is typing a message to Jessica James. This message can either be sent as an IM
or an SMS.
• To send an SMS, user taps Message .
• The system will check if the SMS is sent by a SIP extension. If it is a SIP extension, then it checks if the
SMS over IP option is enabled for the SIP extension. If it is enabled then the SMS will be sent using the
desired Mobile Port. The Mobile Port using which the SMS will be sent to the destination will depend on the
configurations— Fixed/LCR— in SMS over IP Settings.
• When the SMS is received on Mobile Port, the system checks the Send SMS and the Call Budget
parameters configured for the Mobile Port.
If Send SMS is disabled or the Call Budget exceeds the SMS will be rejected.
If Send SMS is enabled and the Call Budget has not exceeded, the system checks the senders number in
the Denied numbers list as configured in the SMS over IP Settings.

destination where the SMS is to be delivered. The destination can be an Extension Number/ External
Number or Name.
When the destination is an Extension Number/Name the system will search for the Extension Number in
its database and then send an SMS to its corresponding Mobile Number.
When the destination is a Mobile number the system will send the SMS to this Mobile number.
• The senders Name and/or Number will be displayed to the recipients, if you have enabled Add Footer in
SMS and configured the parameter Footer text.
It is recommended that you configure the parameters Add Footer in SMS and Footer text, so that the
recipient knows the sender of the message. If this parameter is not configured, only the mobile port SIM
number will be displayed to the recipient.
How to Configure
You need to configure the following SMS over IP parameters,
• Click SIP over IP Settings.
• You can Send SMS through certain fixed Mobile Ports or through different Mobile Ports according to
specific numbers and time.
To send messages through fixed Mobile Ports, select Using Fixed Mobile Port and configure the SMS
over IP - Fixed Port Routing table. For detailed information, see “Fixed Port Routing (SMS over IP)”.
To send messages to specific numbers through certain preferred Mobile Port/s during a defined time
interval, select Based on LCR Table and configure the SMS Routing-LCR table. For detailed
information, see “Least Cost Routing”.
Default: Using Fixed Mobile Port.

• Configure the Allowed-Denied Numbers for sending SMS list, if you want to allow or restrict sending
of messages on specific numbers. To do this,
• Select the Allowed-Denied Numbers for sending SMS check box.
• Click on Allowed-Denied Numbers for sending SMS and the Allowed-Denied Numbers for
sending SMS table opens. You can configure upto 999 numbers.
• In the Allowed column, enter the numbers on which messages can be sent and in the Denied
column, enter the numbers on which messages cannot be sent.
• You can also configure this list by clicking the Allowed-Denied Numbers for sending SMS link
under SMS Routing.
Default: Disabled
• Configure the Allowed-Denied Numbers for receiving SMS list, if you want to allow or restrict
receiving messages from specific numbers. To do this,
• Select the Allowed-Denied Numbers for receiving SMS check box.
• Click on Allowed-Denied Numbers for receiving SMS and the Allowed-Denied Numbers for
receiving SMS table opens. You can configure upto 999 numbers.
• In the Allowed column, enter the numbers from which messages can be received and in the Denied
column, enter the numbers from which messages cannot be received.
• You can also configure this list by clicking the Allowed-Denied Numbers for receiving SMS link
under SMS Settings.
Default: Disabled.
• Enable the Add Footer in SMS check box and in Footer text enter the desired text/message to be
sent to the recipient as signature in the SMS. By default, the senders Name and Number will be sent.

SMS/Email Group
What's this?
In an organization you can group together extension users as per your requirements. Groups can be made
according to the departments, project-wise, product-wise, according to a particular task. The system clubs together
the extension users assigned the same Group.
You can reach all the members simultaneously in a Group, by sending a SMS or an Email.
How to configure
To use SMS/Email Group feature,
• Decide the type of groups you want to create.
• Make a list of extensions to be assigned to a particular group.
• Configure the type of Groups at each index in the SMS/Email Group table. For instructions, see
“Configuring SMS/Email Group table”.
• Assign the desired extensions to the SMS/Email Group you created as per your requirement. For
instructions, see “Configuring SLT Extensions”, “Configuring DKP Extensions”, “Configuring ISDN
Terminals”, “Configuring Matrix SPARSH VP330”, “Configuring Matrix SPARSH VP248”, “Configuring
Matrix VARTA ADR100/AMP100 UC Clients”, “Configuring Matrix SPARSH VP310”, “Configuring Matrix
SPARSH VP510”.
• Assign a Global/Personal Directory contact to the SMS/Email Group you created as per your requirement.
For instructions, see “Configuring Personal Directory using Jeeves” and “Configuring Global Directory
using Jeeves” under “Abbreviated Dialing”.
Configuring SMS/Email Group table


• Click SMS/Email Group.
You can configure maximum 99 Groups,
• At the desired index, in Type of Group configure the Group Name. The Group Name can be a
• Click Submit to save your entries.

CHAPTER 12 Computer Telephony
Integration (CTI)
CTI stands for Computer Telephony Integration. CTI is a technology that integrates a telephone and a computer.
Using CTI, you can control your telephone with your computer and vice versa. This technology contributes to the
success of a modern business.
The Matrix TAPI Service Provider (TSP) acts as a link to integrate the interactions between your telephones and
your computers. The computer in which you install this application functions as a Client and SARVAM UCS
functions as a Server. In an organization, it may happen that the data is stored in different computers. In this
scenario, the Matrix TAPI Service Provider can be installed in three (maximum) different computers, if required.
Connecting your telephone with your computer has advantages such as—initiate a call at a click of a mouse directly
from a CRM system (Customer Relationship Management) or from a database, view missed calls directly on your
screen and can call back with just one click, transfer and call recording can be initiated via a mouse click.
• The Matrix TAPI Service Provider (TSP) is developed using TAPI Standard 2.2.
• To use this feature, you must purchase the CTI License. Refer “License Management” for more details.
• To develop your own TAPI Application, refer to the Matrix TAPI Developer’s Guide. The
documentation can be found at https://www.matrixtelesol.com/product-manuals.html
How it works
Using CTI, only following features and facilities of SARVAM UCS can be accessed:
DKP/Extended IP
Features SLT Open IP Phone
Phone
Make a call   x
Answer/receive a call x  x
Put a call on hold x  x
Retrieve a held call x  x
Toggle between two calls   x
Transfer a call   x
Disconnect/Release a call   
Conference - Add/Remove a party   x

DKP/Extended IP
Features SLT Open IP Phone
Phone
Dial digits for Multi-Stage dialing   x
For CTI to work:
• You must install the Matrix TAPI Service Provider (TSP) in your computer (Client). For instructions refer to
the TAPI User Guide. The documentation can be found at https://www.matrixtelesol.com/product-
manuals.html
• Configure the Matrix TSP parameters. For instructions refer to the TAPI User Guide.
• Configure the CTI parameters in SARVAM UCS. See “Configuring CTI Parameters”.
• Make sure the necessary third party TAPI applications are installed.
This is how CTI works,
• After successful installation of the Matrix TSP in your Computer (Client) and configuring the necessary
parameters, a binding (CTI link) is established between the Client and SARVAM UCS. The SARVAM UCS
communicates with the Client via Matrix TSP.
• All the information is passed by SARVAM UCS to the Matrix TSP and vise versa.
• The third party softwares in the Client can also communicate with the Matrix TSP to get the required
information.
• You can handle calls from your computer (desktop application) as well as your extension phone.
Configuring CTI Parameters

You must configure the following CTI parameters in SARVAM UCS. To do so,
• Under Configuration, click CTI.
• Click CTI Parameters.
• Enable CTI Server.

• Enter Server Listening Port. When the CTI Link is to be established between Matrix TSP and
SARVAM UCS, then the binding will be done on this port.
Make sure the same port is configured as the Listening Port of System— SARVAM UCS in the Client.
For more details, see Matrix TSP User Guide.
• Enable Make Call Ring on SLT, if you want the SLT to ring when an outgoing call is initiated from the
CTI Client.
If Extended IP Phone or DKP is connected as an extension, the speaker of the phone is automatically
turned on when an outgoing call is initiated from the CTI Client.
• Select the Debug check box to log the CTI Server events.
• Click Resources, to view the details of the CTI events and activities. These details are used by the
Technical Team only.
• Click Status. It displays the following details of each Client:
• IP Address: It displays the IP Address of the Client in which you have installed the Matrix TSP.
• Port: It displays the Port of the Client on which the CTI binding with the Server is established.
• TSP Version: It displays the Software Version of the Matrix TSP installed in the Client.
• Status: It displays the status of the Client and Server binding, that is whether the link is up or down.

CHAPTER 13 Incoming and Outgoing Call
Routing
Incoming Call Routing
What's this?
SARVAM UCS offers various options for routing incoming calls on trunks. You can configure incoming call routing
in the following ways:
• Land the call on a group of extensions, referred to as “Trunk Landing Group (TLG)”.
• Route the call to the Built-In Auto Attendant.
• Route call to the Voice Mail Auto Attendant.
• Greet the caller with Trunk Auto Answer, and then route the call to the Trunk Landing Group.
• Route the call to a specific extension on the basis of the CLI received.
• Route the call according to the DDI numbers197.
• Route the call to a group of trunks to use the Gateway functionality of SARVAM UCS.
It is also possible to configure a different routing option for each time zone—Working Hours, Break Hours, Non-
working hours—according to your call routing requirement.
Besides the type of routing you configure, there are certain features supported by SARVAM UCS on trunks that
also determine how an incoming call will be routed, when these features are enabled on the trunk.These features
are:
• “Call Back on Trunk Ports”: Missed calls received on a trunk are returned to the same or to an alternative
number.
• “Direct Inward System Access (DISA)”: From a remote location, users can make a call to an extension of
SARVAM UCS. The call is routed to an extension after authentication and the users can use the system
resources (make calls, access features, configure the system).
• ““RCOC (Return Call to Original Caller)”198: When an unanswered outgoing call is made by an extension is
returned by the called party, this call is routed to the very extension that originally made the outgoing call.
197. Supported on ISDN and SIP trunks only.

198. Supported on digital trunks—BRI, T1E1PRI, Mobile and SIP.

How it works
When a call lands on a trunk, the system checks the trunk features and the routing option configured for the current
time zone on the trunk in the following sequence and accordingly routes the call.
1. Call Back on Trunks

2. DISA - CLI Authentication
3. RCOC (Return Call on Original Caller)
4. DDI Routing and DISA - CLI Authentication
5. DDI Routing and DISA - PIN Authentication
6. DDI Routing
7. DISA - PIN Authentication
8. Auto Attendant - Built-In or Voice Mail
9. Trunk Auto Answer
10. Trunk Landing Group
In the default Incoming Call Routing configuration, incoming calls on a trunk of SARVAM UCS are routed to a
“Trunk Landing Group (TLG)”.
Routing to a Trunk Landing Group

A Trunk Landing Group (TLG) is a “Routing Group”, consisting of a group of extension types. These extensions
may be Digital Key Phones (DKP), Single Line Telephone (SLT), SIP, ISDN Terminals or Virtual Extensions. A
Routing Group can have a single extension type or as many as 32 extension types as members.
A Routing Group can also have a Voice Mail Auto Attendant and an Outgoing Trunk Bundle Group as members,
the significance of which is described later in this topic under Routing to the Voice Mail Auto Attendant and Routing
to an Outgoing Trunk Bundle Group.
• The call is landed on the The Trunk Landing Group (TLG) assigned to the trunk for the current time zone.
By default, Trunk Landing Group number 01 assigned to all trunks for all time zones and has DKP software
port 001, SLT software ports 001 and 002 as members.
• The member extensions in the TLG start ringing in the sequence in which they are arranged in the TLG.
• Each member extension rings for the duration of the Ring Timer (programmable; default: 15 seconds).
• If Continuous Ring is enabled, each extension will ring continuously till the call is answered. The extension
continues to ring even as other extensions in the group are hunted.
If the call remains unanswered even after the last extension in the group has been hunted, the system will
loop back and start hunting from the first extension, all over again.
• If Rotation is enabled on a TLG, for each new call on a trunk, the system will land the call on the extension
next to the one that received the last call. Thus, ensuring an equal distribution of incoming calls on all
member extensions of the TLG.
When Rotation is disabled, for each new call on a trunk, the system will land the call on the first free
extension of the TLG.
To know more about how TLG works, see “Trunk Landing Group (TLG)”.

You can also route calls that remain unanswered by extensions in the Trunk Landing Group to the Voice
Routing Calls to the Auto Attendant

• When a call lands on the trunk, the system checks the trunk features and the routing option configured for
the current time zone on the trunk in the sequence mentioned earlier.
• When Auto Attendant is configured for the current time zone, the system checks for the type of Auto
Attendant configured for the current time zone.
Built-In Auto Attendant

• If Built-In Auto Attendant is configured,
• The system greets the caller with a pre-recorded voice message and prompts the caller to dial the
desired number.
• The call is placed on the extension number dialed by the caller.
• If the caller fails to dial any extension number the call is routed to the TLG configured for the current
time zone.
• If the caller dials ‘9’ for Operator, the call is routed to the Operator group assigned to the trunk. To know
more, see “Configuring 'Operator'”.
For a detailed description of how the Built-In Auto Attendant works, see “Auto Attendant”.
Voice Mail Auto Attendant

• If Voice Mail Auto Attendant is configured,
• The call is routed to the Voice Mail System (VMS).
• The VMS greets the caller with the Welcome message and the Greeting message selected for the
current time zone.
• The VMS processes the call as per the Voice Mail Auto Attendant Menu assigned to the trunk.
For a detailed description of how the Voice Mail Auto Attendant works, see “Auto Attendant”.
You can also route calls that remain unanswered by extensions in the Trunk Landing Group to the Voice
Delayed Auto Attendant

• If Delayed Auto Attendant is configured,
• The system first places the call on the TLG.
• It waits for the configured Auto Attendant Delayed Timer to expire.
• If the call remains unanswered by the TLG, on the expiry of the Auto Attendant Delayed Timer, the
system checks for the type of Auto Attendant selected for the current time zone.
• If Built-In Auto Attendant is selected, the call is routed accordingly. See description for“Built-In Auto
Attendant” in this topic.
• If Voice Mail Auto Attendant is selected the call is processed as per the Voice Mail Auto Attendant
Menu assigned to the trunk.
To know more about Delayed Auto Attendant, see “Auto Attendant”.

Greeting with Trunk Auto Answer
• When a call lands on the trunk, the system checks the trunk features and the routing option configured for
the current time zone on the trunk in a sequence mentioned earlier.
• When Trunk Auto Answer is enabled on a trunk, the system checks for the type of Trunk Auto Answer set
on the trunk.
• If Trunk Auto Answer - For All Calls is selected,

• the call immediately answered with a Greeting message, known as the Trunk Auto Answer Greeting.
• The call is then routed to the landing destination selected for the current time zone.
• The system waits for the duration of the Built-In Auto Attendant Inactivity Timer (default: 60 seconds)
as it waits for the call to be answered.
• If the call is answered by the landing destination, within this timer, the caller is connected to the
answering extension.
• If the call remains unanswered until the expiry of this timer, the system plays the Trunk Auto Answer
Busy Bye message or a Busy Tone and releases the trunk port.
• If Trunk Auto Answer - When Busy is selected,

• the system checks will first check the state of the landing destination for the current time zone.
• If the landing destination is busy, the system answers the call with the Trunk Auto Answer Greeting
message.
• The system waits for the duration of the Built-In Auto Attendant Inactivity Timer (default: 60 seconds)
for the landing destination to become free.
• If the landing destination is free before the timer expires, the call is placed.
• If the landing destinations is busy till the end of this timer, the system plays the Trunk Auto Answer
Busy Bye message or a Busy tone to the caller and releases the trunk port.
• If Trunk Auto Answer - Delayed is selected,

• the system first routes the incoming calls to the TLG selected for the current time zone.
• It waits for the duration of the Delayed Trunk Auto Answer Timer (programmable; default:10 seconds)
for any of the extensions in the TLG to answer the call.
• If the call is not answered by any of the extensions within this timer, the system answers the call with
the Trunk Auto Answer Greeting message, and rings the landing destination.
• The system starts the Built-In Auto Attendant Inactivity Timer (default: 60 seconds) as it waits for the
call to be answered and repeatedly plays the Ring Back Tone message or tone to the caller.
• If the call is answered before the Inactivity Timer by the landing destination, the system stops the timer
and the Ring Back tone message/tone and connects the caller to the destination.
• If the call remains unanswered till the expiry of the Inactivity Timer, the system plays the Trunk Auto
Answer Busy Bye message or a Busy tone to the caller and releases the trunk port.
Feature Interactions
As mentioned earlier, certain trunk features also determine incoming call routing. Consider the following feature
interactions between Trunk Auto Answer and other trunk features when configuring Trunk Auto Answer as a routing
option on a trunk.
• Call Back on Trunks: This feature should be disabled when using Trunk Auto Answer.
• DISA - CLI Authentication: This feature should be disabled when using Trunk Auto Answer.
• Return Call to Original Caller (RCOC): If both Trunk Auto Answer and RCOC are enabled on the same
trunk, the call will first be answered as per Trunk Auto Answer, and then placed on the destination as per
RCOC.

• CLI Based Routing: If both Trunk Auto Answer and CLI Based Routing are enabled on the same trunk,
the call will first be answered as per the Trunk Auto Answer, and then placed on the destination as per the
CLI Based Routing Table.
• DISA - PIN Authentication: If both Trunk Auto Answer and DISA PIN Authentication are enabled on the
same trunk, only DISA - PIN Authentication will work. The call will be treated as a DISA call and routed as
per the DISA logic.
• Auto Attendant: If both Trunk Auto Answer and Auto Attendant are enabled on the same trunk, only Auto
Attendant will work. The call will be processed by the Auto Attendant.
Routing Calls based on CLI Received

It is possible to land calls from a particular telephone number on a particular extension or group of extensions using
“CLI Based Routing” on the trunk. For this, you need to configure the CLI of the calling parties and their
corresponding landing destinations in the CLI Based Routing Table.
• When CLI Based Routing is configured for the current time zone, the system checks the entries of the CLI
Based Routing Table.
• If the calling party’s number is found in the CLI Based Routing Table, the call is placed on the
corresponding landing destination.
• If the calling party’s number does not exist in the CLI Based Routing Table, the call will be routed
according to the incoming call routing option you have configured.
For a detailed feature description, see “CLI Based Routing”.
Routing Calls on the basis of DDI Numbers

Direct Dialing-In (DDI) is a feature of ISDN (T1E1PRI and BRI) and SIP trunks which enables calls to be directly
placed to the desired extensions without operator intervention. The DDI Numbers given by the Service Provider are
mapped to extensions. Callers can reach the extensions directly by dialing the DDI number of the respective
extension.
• DDI Routing works on the basis of the IC Reference Table and the DDI Routing Table.
• The T1E1PRI, BRI and SIP trunks ports are assigned IC Reference ID, which may be different for each
time zone. The IC Reference ID is the reference number that acts as an identifier to the translation logic
programmed in the IC Reference Table. When a call lands on the ISDN/SIP trunk, the system checks the
IC Reference ID assigned to it. It checks the IC Reference Table for the corresponding DDI Routing
Reference ID for call resolving.
• Using the DDI Routing Reference ID, the system checks the DDI Routing Table for mapping the received
DDI number to a flexible number (target extension).
• The system first compares the received DDI number (called party number) with the DDI numbers
programmed in the DDI Routing Table.
• Once a perfect match is found, the system checks for the Route on First Destination flag in the IC
reference table. If the flag is enabled, the call lands on the first extension of the MSN number in the DDI
Routing Table. If the flag is disabled, the call is routed to the identified target extension.

• Once the target extension is identified, the system checks the DDI IC routing flag of the extension in its
“Station Advanced Feature Template”. If the flag is enabled, the call lands on the extension else the call is
routed to the TLG assigned to the trunk.
• When the call lands on the DDI extension, the caller gets the Ring Back Tone. The extension rings for
duration of the DDI Ring Timer.
• If the call is not answered within the DDI Ring Timer, the system checks for the When No reply option in
the IC reference Table for action to be taken.
• Similarly, if the DDI extension is busy, the caller gets Busy Tone. The system checks the When Busy
option in the IC Reference Table for action to be taken.
• The options for action to be taken the DDI extension does not No Reply or is Busy are:
• Disconnect the call
• Route the call to Trunk Landing Group
• Answer the call automatically, greet the caller with a voice message, and on completion of message
• Answer the call, greet the caller with a voice message, and on completion of the message route the call
to Trunk Landing Group.
• Route the call to Voice Mail; to the DDI extension’s Mail Box.
To know more, refer the topics “Direct Dialing-In (DDI)” .
DDI Routing will not work if Auto Attendant or DISA are enabled on a trunk.
Routing Calls to Outgoing Trunk Bundle Groups

To use SARVAM UCS’s gateway functionality, you can route incoming calls on a trunk to a group of outbound
trunks, referred to as “OG Trunk Bundle Group”. The Called Number (number dialed by the caller) received in the
incoming call is out dialed from Outgoing Trunk Bundle Groups.
Routing to Outgoing Trunk Bundle Groups is supported only on SIP, ISDN T1E1PRI, ISDN BRI and E&M
Trunks (except Express line), as they support Called Number. CO and Mobile trunks support only CLI.
To do this, you must configure the desired outbound trunks in an “Outgoing Trunk Bundle” and create an “Outgoing
Trunk Bundle Group (OGTB)”. You must then include this OGTB as a member of the Trunk Landing Group you
assign to the SIP/ISDN T1E1PRI/ISDN BRI/E&M trunk.
When there is an incoming call on the SIP/ISDN T1E1PRI/ISDN BRI/E&M trunk,

• the system checks the trunk features and the routing option configured for the current time zone on the
trunk in a sequence as mentioned earlier.
• On finding OGTB as member in the TLG of the current time zone, the call is routed to the OGTB number
assigned.
• The system out dials the Called Number received in the incoming call from the OGTB.

How to configure
Incoming call routing options for a trunk are to be configured on the Trunk Feature Template assigned to the trunk.
Configuring Trunk Landing Group

To land calls on a group of extensions, you must do the following:
• First configure a Trunk Landing Group. You may configure the same Trunk Landing Group for working,
break and non-working hours or you different Trunk Landing Group for all three time zones.
• To land calls on the Operator extension first, make sure the Operator extension is included in the TLG you
configure. Also make sure that the Operator extension is the first member in the group.
You can also keep the same Routing Group you configured as Operator group as the Trunk Landing
Group.
• Assign the Trunk Landing Group to the trunk in the Trunk Feature Template for each time zone.
• To have a number of extensions in the group ring simultaneously, enable Continuous Ring on these
extensions and set the Ring Timer for these extensions to ‘00’ seconds.
• To set equal distribution of incoming calls on all extensions in the group, enable Rotation for the entire
group (default: disabled).
For detailed instructions, see “How to configure” in Trunk Landing Group.
If you want incoming calls to be routed to Trunk Landing Group only, do not enable any other incoming call
features on the trunk.
Configuring Auto Attendant

To route calls to the Built-In-Auto Attendant, you must do the follow:
• Make a list of the trunks by their port type (CO, Mobile, SIP, BRI, T1E1PRI) and port number on which you
want to use the Built-In Auto Attendant.
• Configure a Trunk Feature Template with Built-In Auto Attendant enabled for the desired time zones.
• Assign this Trunk Feature Template to the desired trunks. The calls landing on this trunk will be answered
by the Built-In Auto Attendant. See “Trunk Feature Template”, for more information.
• Set the Start Time for the Morning, Afternoon and Evening Greeting Messages. Refer “Greeting Message
Time” in System Parameters for instructions.
• Assign Voice Modules for Built-In Auto Attendant Messages. To play to callers pre-recorded voice
messages as Built-In Auto Attendant greetings and to play voice prompts at each stage of the call, you
need to assign Voice Modules for the following Built-In Auto Attendant Messages:
For detailed instructions, see “How to configure” in “Auto Attendant”.
• If you want the incoming calls to first land on extensions, you must configure the Auto Attendant Delayed
Timer (sec). The call will be placed fist on the extensions in the Trunk Landing Group. If the call remains

unanswered till the expiry of the timer, it shall be answered by the Built-In Auto Attendant. This is known as
Delayed Auto Attendant on trunks.
To route the call to the Voice Mail Auto Attendant, you need to configure:
• Make a list of the trunks by their port type (CO, Mobile, SIP, BRI, T1E1PRI) and port number on which you
want to use the Voice Mail Auto Attendant.
• Configure a Trunk Feature Template with Voice Mail Auto Attendant. enabled for the desired time zones
and assign the Voice Mail Auto Attendant (VMAA) Menu.
• Assign this Trunk Feature Template to the desired trunks. The calls landing on this trunk will be answered
by the Voice Mail Auto Attendant. See “Trunk Feature Template”, for more information.
• Complete the VMS related configuration.
• Configure Welcome and Greeting messages. You may either use the default, pre-recorded welcome
messages of the VMS, or record the custom welcome messages that meet your requirements, in .WAV
file format.
• Configure the parameters of the Voice Mail Auto Attendant (VMAA) Menu.
For more information and instructions, see the “Configuring Voice Mail System”.
• If you want the incoming calls to first land on extensions, you must configure the Auto Attendant Delayed
Timer (sec). The call will be placed fist on the extensions in the Trunk Landing Group. If the call remains
unanswered till the expiry of the timer, it shall be answered by the Voice Mail Auto Attendant. This is known
as Delayed Auto Attendant on trunks.
Configuring CLI Based Routing

To route the call to a specific extension on the basis of the CLI received:
• Select the trunk on which to want to enable CLI Based Routing.
• Configure the CLI Based Routing Table. Enter the numbers of the calling parties and the numbers of the
corresponding destination extensions.
• Enable CLI Based Routing on the desired trunks according to time zones in their “Trunk Feature
Template”.
For more details, see “How to configure” in CLI Based Routing.
Configuring Trunk Auto Answer

• Select the trunk on which you want to enable Trunk Auto Answer.
• Enable Trunk Auto Answer in the Trunk Feature Template of the desired trunk.

• Select the Trunk Auto Answer Greeting message, the Trunk Auto Answer Ring Back Tone Message,
and the Trunk Auto Answer Busy Bye Message for the Working Hours, Break Hours and Non-Working
Hours.Create a Trunk Landing Group with the desired extensions.
• Configure the Trunk Auto Answer related Timers, if required. The following Timers are of relevance to the
Trunk Auto Answer Feature:
• The DID Inactivity Timer (default: 60 seconds)
• The Ring Back Tone Timer (default: 45 seconds)
• The Busy Tone Timer (default: 7 seconds)
• You may change the duration of these timers from the“System Timers and Counts” page.
The Ring Back Timer and the Busy Tone Timer are also applicable for the Ring Back Tone and the Busy
Tone played for internal calls.
• Record and assign Voice Modules for the following Voice Messages related to this feature:
• Trunk Auto Answer Greeting Message
• Trunk Auto Answer Ring Back Tone Message
• Trunk Auto Answer Busy Bye Message
For each of these messages, you can record four different messages.
See the topic “Voice Message Applications” for instructions on recording and assigning voice modules to
greeting messages.
For detailed instructions, see “How to configure” in Trunk Auto Answer.
Configuring Direct Dial-In (DDI)

• DDI can be enabled on ISDN T1E1PRI, BRI as well on SIP Trunks. Select the trunks on which you want to
enable DDI.
• Make a list of the Extensions to whom you want to assign DDI Numbers.
• Assign an Incoming (IC) Reference ID in the respective port parameters of the trunks. For instructions, see
“Configuring E1 Trunks”, “Configuring T1 Trunks”, “Configuring BRI Trunks” and “Configuring SIP Trunks”.
• Configure the DDI Routing Table and Incoming Reference Table. For instructions, see “Direct Dialing-In
(DDI)”.
Configuring Routing to OGTB
To route the call to an outgoing trunk, when using the gateway functionality of SARVAM UCS,
• You must make the Outgoing Trunk Bundles. See “Outgoing Trunk Bundle”
• Make an Outgoing Trunk Bundle Group by assigning these Outgoing Trunk Bundles. See “OG Trunk
Bundle Group” for detailed instructions.
• Select OTBG as member in the “Routing Group”.
• Assign this Routing Group Number in the Trunk Landing Group.

• You may configure the same Trunk Landing Group for working, break and non-working hours or you
different Trunk Landing Group for all three time zones.
• Assign the Trunk Landing Group to the trunk in the Trunk Feature Template for each time zone.
• For detailed instructions, see “How to configure” in Trunk Landing Group.

Outgoing Call Routing
What's this?
SARVAM UCS provides you a platform wherein you can integrate and use the services from different networks -
VoIP, ISDN, GSM, CO.
The extensions users (UC/SIP Users, DKP, ISDN Terminals, SLT) of SARVAM UCS can make outgoing calls to
external numbers or to internal numbers. The process of identifying the call and then routing the call through the
most economical network, constitutes Outgoing Call Routing.
An outgoing call can be a call:

• to another extension number, that is a number dialed without grabbing a trunk line.
• to an external number, that is a number dialed after grabbing a trunk line.
• to a number in a closed user group, that is a number which is a part of a private network. A private network
is when a few Systems are connected to each other using SIP, T1/E1 QSIG, T1/E1 PRI, E&M etc.
When an outgoing call is made by the extension user, the system simultaneously checks for:
• the features and facilities enabled for the extension user.
• the features enabled on the trunk assigned in the OG Trunk Bundle Group, for the Trunk Access Code
dialed by the user.
Features and Facilities that can be enabled in the Station Basic Features Template assigned to the extension user:
• The Class of Service assigned to the extension user. Class of Service (CoS) defines the permission an
extension will have on a System. It defines the set features of the System that the extension is to be
allowed access to.
Feature requirements vary among users and with time. Similarly, certain features that are required during
working hours may not be required during break or non-working hours.
For details, see “Class of Service (COS)”.
• The Call Budget assigned to the extension user: The system keeps a tab on the total cost of phone calls
made by extension users. If the budget assigned to the user has been consumed, the system will not allow
the user to make further outgoing calls, but will be able to make internal calls.
The extension user can be assigned a fresh budget, after which s/he can resume making calls.
Call Budget can be enabled on all the extensions as well as on selected extensions. Each extension can
be assigned a different amount depending on the user requirement.
For details, see “Call Budget on Extension”.
• The Call Privilege assigned to the extension. In Call Privilege you can define the Toll Control Levels. Toll
Control (or Toll Restriction) is an expense control feature of SARVAM UCS. It enables you to program the
system so that each extension has a designated calling permission referred to as 'Call Privilege'.
Each type Call Privilege allows the extension to call certain areas and restricts it from calling others. The
extension can also be restricted from the dialing of specific telephone numbers.

For details, see “Toll Control”.
Features enabled on the trunk:
• A Trunk Access Code is a short digit sequence dialed from an extension phone instructing the System to
assign a trunk line or any trunk line from a group of trunks (“OG Trunk Bundle Group”) to the user to dial an
external number. For making outgoing calls each extension user must dial a Trunk Access Code.
• The outgoing calls are routed through the OG Trunk Bundle Groups assigned to the extensions in the
Station Basic Features Template. All the trunks connected to the system can be bunched in different
groups called OG Trunk Bundle Groups and these OG Trunk Bundle Groups can be allotted to each
extension.
Each OG Trunk Bundle Group consists of a single OG Trunk Bundle or multiple OG Trunk Bundles. The
Outgoing (OG) Trunk Bundle is set of parameters that completely define the grouping of similar channels/
trunks. Bundles of similar trunks/channels only can be formed. The system can hunt for a free trunk within
the bundle as per the set option - Ascending, Descending or Cyclic. For more information, see “Outgoing
Trunk Bundle”.
An Extension can be allotted different OG Trunk Bundle Group during different timings of the day.
• Since there are different trunk lines for making calls and the service providers of these trunks offer different
tariffs for calls made to certain locations or numbers or during a particular time of the day, you can enable
Least Cost Routing on the trunks.
When a call is made from an extension of the SARVAM UCS, using LCR the system selects the lowest
cost trunk from among all the trunks allotted to that extension to make the outgoing call, depending upon
the type of LCR configured.
For more information, see “Configuring LCR”.
• For LCR to work, all trunks that are allotted to extensions for making outgoing calls, must first be assigned
a Cost Factor.
Cost factor is used for grading trunks in the order of increasing cost of routing calls, from 01 to 99, where
01 signifies least cost and 99 signifies the highest cost. Thus you can grade up to 99 trunks according to
the increasing cost of routing calls.
After assigning Cost Factor to Trunks, you must configure the Type of LCR to be used on Trunks in the
Outgoing Trunk Bundle Group (OGTBG) allotted to the extensions for making calls.
If LCR is not enabled, the outgoing call will be routed through the free trunk from the OG Trunk Bundle
Group.
For more information, see “Cost Factor” in Configuring LCR.
• The Call Budget assigned to the Trunk. Each trunk can be allotted a 'budget' limit for outgoing calls. This
budget limit can be programmed to be reloaded manually each time it is exceeded or at a scheduled date,
either daily or at a particular date of the month.
For more information, see “Call Budget on Trunk”.

How it works
When the user dials a number without grabbing a trunk, that is, without dialing a Trunk Access Code, the system
checks:
• The Closed User Groups table, if a match is found the call is routed through the OGTBG assigned in the
table. For more information, see “Closed User Group (CUG)”.
• If no match is found, the checks for an internal extension number. For the Extension user it checks for the
features enabled in the Class of Service (CoS). If Basic Features are disabled the extension user will not
be able to make internal calls. If in the CoS the Basic Features are enabled and a match is found in the
internal group, the call is placed on the dialed extension.
• If still the system is unable to find a match, the systems plays an error tone to the caller.
When the user dials a number after grabbing a trunk line, that is after dialing a Trunk Access Code, the system
checks:
• The Station Basic Feature Template assigned to the user. In this template it checks for the type of Toll
Control and Call Budget.
• The Outgoing Trunk Bundle Group assigned to the user for making outgoing calls, for the Trunk Access
Code dialed.
• The features enabled on the Trunks in the OGTBG, that is Least Cost Routing and Call Budget.
• If LCR is not enabled the system allots a free trunk from the OG Trunk Bundle Group to route the call.
• If LCR is enabled, the system checks for the type of LCR enabled and routes the call using the
cheapest free trunk from the group. If the cheapest trunk is not free, the system hunts for the second
cheap trunk in group and routes the call. If none of the trunks are free, the system plays a busy tone to
the user.
• After checking LCR, for the free trunk the system checks the type of Call Budget enabled on the trunk.
The calls will be routed using this trunk as long as the budget limit set for the trunk (i.e. the Amount or
Minutes or the maximum number of Calls) is not crossed. As soon as the limit is crossed the Trunk is
automatically disabled.
How to configure
For the extension user configure the following in the Station Basic Features Template:
• The Basic Features are enabled in the Class of Service (CoS). If you want to restrict dialing of internal calls
by the user, disable the Basic Features. Select a CoS group number, for example 19 and disable the Basic
Features. In the Station Basic Features Template assigned to the user, enter 19 as the CoS for each time
zone.
• To restrict calling, enable Call Budget and configure the following parameters for the feature to work:
• select the Call Budget check box to enable
• select the Toll Control-Call Budget Consumed option
• set the Preset Call Budget Amount

• Set the Call Privilege for the different time zones.
• For each the Trunk Access Code in each time zone, assign an Outgoing Trunk Bundle Group.
For detailed instructions to assign a CoS, Call Budget, Call Privilege and Outgoing Trunk Bundle Group to an
extension user, see “Station Basic Feature Template” and “Configuring DKP Extensions”, “Configuring SLT
Extensions”, “Configuring ISDN Terminals”, “Configuring SIP Extensions”.
You need to provide access of the trunks to the users for making calls, to do so, follow the steps given below:
• To create an Outgoing Trunk Bundle put similar trunk types are put together.
• Assign the Outgoing Trunk Bundles as members in an Outgoing Trunk Bundle Group. For detailed
instructions, see “OG Trunk Bundle Group”.
• For the Outgoing Trunk Bundle Group, enable Least Cost Routing, if required. Select the type of LCR
required and also configure the respective LCR tables. For detailed instructions, see “Configuring LCR”.
• Enter this Outgoing Trunk Bundle Group number, as per the time zone in the Station Basic Features
Template assigned to the extension user.
• Determine the cost factor you want to assign each trunk. Assign the Cost Factor to each trunk type in their
respective trunk parameters.
• Enable Call Budget on Trunks. Select the type of Call Budget you want to set on the trunk, in their
respective trunk parameters.
For detailed instructions on assigning the Cost Factor and Call Budget to the trunk, see “Configuring CO
Trunks”, “Configuring Mobile Trunks”, “Configuring E1 Trunks”, “Configuring T1 Trunks”,“Configuring BRI
Trunks”, “Configuring SIP Trunks”.

CHAPTER 14 Features and Facilities
If you have CDMA Mobile Card installed in your system, it is recommended to avoid using features that
Abbreviated Dialing
What's this?
Abbreviated Dialing is the use of short codes (abbreviated numbers), to dial out long-digit numbers. It is also
referred to as Memory Dialing.
Abbreviated Dialing allows you to dial quickly and easily, frequently called, long-digit numbers.
This feature requires you to store the frequently called, long-digit numbers199 and their corresponding short codes
in special lists, known as 'directories'. These directories may be 'personal' or 'global'.
SARVAM UCS supports two types of Abbreviated Dialing based on the type of directory used: Personal
Abbreviated Dialing and Global Abbreviated Dialing.
Abbreviated Dialing forms the basis of two other features of the SARVAM UCS: “Dialed Number Directory” and
“Quick Dial”.
Personal Abbreviated Dialing

This variation of Abbreviated Dialing makes use of the Personal Directory.
Personal Directories can be programmed and assigned to groups of extensions. The use of Personal Directories is
limited to the extensions to which they are assigned.
A personal directory accommodates 25 numbers. Each number may be up to 16 digits long. A personal directory
has Index numbers from 0001 to 0025 against which the frequently dialed telephone numbers are stored along with
their corresponding names and trunk access code ID.
199.These may be numbers of your branch offices, your clients, as also numbers of emergency services such as fire, police.

As many as 50 different personal directories, numbered from 01 to 50 can be created and assigned to SLT, DKP
and SIP extensions.
With a personal directory assigned to an extension, the extension user simply dials out the Feature Access Code
for Abbreviated Dialing and the Index Number at which the desired number is stored in the personal directory.
For example: personal directory number 02 is assigned to extension 2001. The number 02652630555 is stored at
Index number 16 of this directory. The user of extension 2001 can call this number by simply dialing '8' (feature
access code) followed by '16' (the index number).
The system will automatically dial out the number using the trunk access code ID specified for this number in the
personal directory.
• When an extension user dials an abbreviated number from the Personal Directory, the system first
checks OG Trunk Bundle Group (OGTBG) and Toll Control Level (Call Privilege) of that extension and
then dials out the number.
• Each extension can access only the personal directory assigned to it.
• Personal Directory can be programmed by the System Engineer, as well as extension users. Extension
users can add contacts to the Personal Directory assigned to them from their extensions phones (DKP/
SLT/ISDN phone/IP Phone).
Global Abbreviated Dialing

This variation of Abbreviated Dialing makes use of a system-wide list of numbers stored in the memory of the
SARVAM UCS, known as the Global Directory.
Being a system-wide list, the Global Directory can be accessed by any extension connected to the system.
The Global Directory has the capacity to store up to 900200 numbers of a maximum of 16 digits each. The Global
Directory is divided into three parts:
• Part 1 - contains Memory Location codes 100 to 799.
The Global Directory has Memory Location codes starting from 100 to 999. The telephone numbers along with their
corresponding names are stored against Memory Location codes.
Whenever extension users of the system want to use Global Abbreviated Dialing, all they needs to do is dial the
feature access code ('8' or '6') and the Memory Location code at which the desired number is stored.
For example: the number 02652630566 is stored at Memory Location 102 of the Global Directory. Now, extension
users of the system can call this number by simply dialing the '8' or '6' (feature access code for Abbreviated Dialing)
followed by '102' (Memory Location code at which the desired number, 02652630566, is stored).
The system will dial out the number using any of the trunks in the “OG Trunk Bundle Group” assigned to it in the
Memory Location.
200. ETERNITY LENX stores up to 2900 numbers in the Global Directory; Part 1 - 100 to 2399, Part 2 - 2400 to 2699 and Part 3 - 2700
to 2999.

Extensions can use Global Abbreviated Dialing only if this feature is included in the “Class of Service
(COS)” allowed to them.
• Further, an extension can access only that part of the Global Directory which is allowed to it in the CoS.
For instance, if extension 2001 is allowed Global Directory Part 1 in its CoS, the user of extension 2001
can dial out only those numbers contained in Global Directory Part 1.
• So, to be able to access the entire Global Directory, extensions must be assigned all three parts of the
directory in their Class of Service. By default, only Global Directory Part 1 is included in the CoS of all
extensions.
• Global Directory can be programmed by the System Engineer, and by Digital Keyphone extension
users who have Global Directory Programming allowed to them in the Class of Service.
• While the System Engineer can program all three parts of the Global Directory, digital keyphone
extension users who are allowed Global Directory Programming in their Class of Service can configure
only Global Directory Part 1.
How to configure
For both Personal and Global Abbreviated Dialing to work, the System Engineer must:
• Program the Personal Directories and the Global Directory.
• Assign Personal Directory to the desired extensions (which may be different: SLT, DKP, ISDN Terminals,
IP Phones).
• Enable Global Directory Part 1/2/3 as desired in the Class of Service (CoS) group allowed to the
extensions.
• Enable ‘Global Directory Part 1’ and ‘Global Directory Programming’ in the CoS group of the digital key
phone extension users, who are to be allowed to program (add, delete, edit) contacts in the Global
Directory Part 1 from their digital key phones.
All the above parameters can be programmed by the System Engineer using Jeeves as well as a telephone.
Preparing Numbers Lists for Personal and Global Directories

In consultation with the extension users, you may:
• Find out the number of personal directories that need to be programmed.
• Make a list of numbers frequently dialed by the extension.
• Ask the extension users the numbers they would like to be included in the personal directory of their
extension.
• Make separate lists of numbers along with their corresponding names, email id, group, trunk access
codes, for each personal directory. You may draw five-column tables on paper and enter the Numbers and
corresponding names, email ids, groups and trunk access codes against each Index number. For
example:

Personal Directory 01
Email ID Group
Index No. Number Name TAC ID
(optional) (optional)
01
02
: : : :
: : : :
25
Personal Directory 02
Group TAC ID
Index No. Number Name Email ID
01
02
: : : :
: : : :
25
• Compile the numbers to be included in the global directory. Numbers that are commonly dialed by all
extensions can be included in the global directory.
• Draw a five-column table on paper and enter the telephone numbers along with their names, the Outgoing
Trunk Bundle Group (OGTBG), email id and group at each Memory location. For example:
Global Directory
Email ID Group
Memory Location OGTBG Number Name
100
101
: : : :
: : : :
999
• Prepare the Global Directory keeping in mind that is divided into three parts: Part 1 (100 to 799), Part 2
(800 to 899), and Part 3 (900 to 999). As Part 1 is allowed to all extensions in their default CoS, you may
include the numbers allowed to all extensions in this part of the directory.
Uploading Personal and Global Directory Contacts

If you have personal and global directory contacts database in an excel sheet, you can convert the same into a
CSV file and upload it through Jeeves.

For Personal directory contacts, the format of the CSV file must be as follow: For example 21, 9867985489, Sean
Gilbert, Sean@hotmail.com, 1, 0
Where,
21 is the Index Number at which you want the entry to be stored (mandatory)
9867985489 is the Contact Number (mandatory)
Sean Gilbert is the Contact Name (optional)
Sean@hotmail.com is the Email ID (optional)
1 is the Group Index number (optional), see “SMS/Email Group”
0 is the Trunk Access Code (TAC) to route the call (optional)
To upload Personal Contacts CSV file using Jeeves, see “Upload Personal Directory CSV files” .
For Global directory contacts, the format of the CSV file must be as follow: For example 121, 9867985489, Sean
Gilbert, Sean@hotmail.com, 1, 01, 000
Where,
121 is the Index Number at which you want the entry to be stored (mandatory)
9867985489 is the Contact Number (mandatory)
Sean Gilbert is the Contact Name (optional)
Sean@hormail.com is the Email ID (optional)
1 is the Group Index number (optional), see “SMS/Email Group”
01 is the Outgoing Trunk Bundle Group (OTBG) to route the call (optional)
000 is the Alternate Number Group (optional)
To upload Global Contacts CSV file using Jeeves, see “Upload Global Directory CSV file”.
Update Personal Directories via Email

You can update the Personal Directory contacts via Email, if you have enabled the SMS Server Application. You
can Add, View, Edit and Delete contact/s via Email. For more details, see “Configuring Personal Directory via
Email”.
Configuring Personal Directory using Jeeves

• Under Configuration, click Abbreviated Dialing.
• Click Personal Directory to open the page.

You can configure upto 50 Personal Directories. Select the directory number by clicking the required
number tab above the table.
For each directory configure the following parameters,
• Enter the Number you wish to store against an Index Number. and enter the contact's Name against
the number.
The length of the Number field is limited to 16 digits. The length of the Name field is limited to 12
alphanumeric characters. All ASCII characters except < > and “ (double quote) are allowed. Ensure
that the number and the name are programmed within this limit.
Each directory has a limit of 25 entries. You may enter up to 25 Numbers and Names in each Personal
Directory.
Change the TAC Index (TAC ID), if required.
• Click Advance to configure the Email ID and select the Group for the contact.
• Enter the Email ID of the contact you wish to store. The Email ID can be a maximum of 64 characters.
• You can assign the contact to a Group. Select the desired Group Type from the list. The system clubs
together contacts assigned the same Group. Default: None. For details, see “SMS/Email Group”.
• Click Submit to save the entries.

• Repeat these steps to program each Personal Directory.
• Click Submit at the bottom of the page to save your entries.
Keep a print of each personal directory for your record and for the record of the extension user to whose
phone the personal directory is assigned. This will also help you take care of overlaps and include some of
the numbers that are dialed by all users in the Global Directory instead of the Personal Directory.
Assigning Personal Directories to Extensions

To assign Personal Directory to SLT extensions,
• Click the Advance button.Go to Personal Directory column of the SLT to which the directory is to be
assigned.
• Enter the number of the Personal Directory. For example, to assign Personal Directory No. 02 to SLT 2001
(software port 001, connected on hardware slot 03, hardware port 09), enter '02' in the 'Personal Directory'
column for SLT 2001.
• Similarly, enter the Personal Directory number to be assigned to each SLT.
To assign Personal Directory to DKP extensions,
• Click the Advance button. Go to Personal Directory column of the DKP to which the directory is to be
assigned.
• Enter the number of the Personal Directory. For example, to assign Personal Directory No. 01 to DKP
3001 (software port 001, connected on hardware slot 17, hardware port 01), enter '01' in the 'Personal
Directory' column for DKP 3001.
• Similarly, enter the Personal Directory number to be assigned to each DKP.
To assign Personal Directory to ISDN Terminal extensions,
• Click the Advance button. Go to Personal Directory column of the ISDN Terminal to which the directory
is to be assigned.

• For each ISDN Terminal that is to be assigned a Personal Directory, enter the number of the directory in
this column.
To assign Personal Directory to SIP Extensions extensions,
• Click SIP Extension Settings to open the page.
• Select the software port number of the SIP extension you want to assign the Personal Directory.
• Click the Advance button.
• Scroll to Personal Directory, and select the Personal Directory number you want to assign to this
extension.
It is possible to assign the same personal directory to multiple extensions.
Upload Personal Directory CSV files

• Click Upload/Download to open the page.

• Click Upload Personal Directory Contacts to expand.
• In Personal Directory Number, select the number of the personal directory in which you want to
upload the contacts from the CSV file.
• Select the Clear all other indices of the selected Personal Directory, which are not specified in
the .csv file being uploaded check box, to overwrite the existing contacts in the Personal directory
with the contacts of CSV file. Default: Disabled.
• Click the Browse button to Select the .csv file to be uploaded from the location on the local disk.
• Click the Upload button.
• All the contacts of the CSV file will be uploaded in the selected Personal Directory. To view, click the
respective Personal Directory link.
Download Personal Directory CSV files


• Click Download Personal Directory Contacts to expand.
• In Personal Directory Number, select the number of the personal directory from which you want to
download the contacts.
• Click the Download button.
• You will get a prompt with an option to open the Opening Personal_Directory_01.csv file or save the
file to a location. Save the file on the local disk.
• To Open the Personal_Directory_01.csv file from the location on the local disk, follow the steps given
below:
• DO NOT OPEN THE CSV FILE DIRECTLY WITH EXCEL!
• Open a New worksheet in Excel.

• Open the Data tab and select From text button in the Get External Data.
• Select your CSV file from the location on the local disk to import.
• In Original data type section, select Delimited radio button and click Next.

• In Delimiters select the Comma check box (column dividers will appear in preview) and click Next.

• Select the column with leading zeros and in Column data format select the Text radio button.
You will have to do this for each column where the data contains leading zeros.
• Click Finish.
• The leading zeros will still be there in the new worksheet with the imported data.
Configuring Personal Directory via Email

You can configure the Personal Directory via Email, if you have enabled the SMS Server Application. To know
more, see “SMS Server”.
If you have enabled SMS Server Application, you can configure contacts in the Personal Directory via Email.
The SMS Server Application of SARVAM UCS allows you to Add, View, Edit and Delete contact/s via Email.
When the SMS Server receives an Email for configuring the Personal Directory, the system searches for the Email
ID in the system users lists.
If a match is not found, the system sends a reply mail to the sender, with the Subject: SMS Server Personal
Directory Configuration' and the message in the body: Not a valid user!.
If a match is found, the system checks for the Personal Directory assigned to the user.
If a Personal Directory is not assigned, for the extension user then the system sends a reply mail to the sender with
the subject: SMS Server Personal Directory Configuration and the message in the body: Personal Directory is not
assigned, can't add contact Name: XXXXX Number: XXXXX. Where name/number is as received in mail for
directory configuration.

If a match is found for the user and the user is assigned a Personal Directory, the system accepts and takes the
necessary action as requested by the user. The request may be for adding, viewing, deleting or editing a contact.
The following validations are applicable when you want to configure contacts via Email:
The Number can be a maximum of 16 digits.
The Name can be a maximum of 12 characters.
The Email ID can be a maximum of 64 characters.
The Group can be a maximum of 16 characters.
While configuring Personal Directory via Email, make sure you have configured the Type of Group in the
“SMS/Email Group” topic under SMS Server.
Adding a contact
You can add a single/ multiple contacts via Email. To add a contact you must send an Email to the SMS Server in a
specific format. Given below are the various examples and formats of adding a contact.
To add a single contact
Given below is the format of the mail to be sent to the SMS Server, to add a single contact.
To: smssever@domain.com (this is the Email ID of the SMS Server)

Subject: James Smith=+919898985400 (The name and number of the contact you wish to add)
Here,
The To field contains the Email ID of the SMS Server.
The Subject contains the details of the contacts.
The name of the contact is James Smith
The number of the contact is +919898985400
The contact is added at a first free index, between 0001 to 0025 in the Personal Directory and a confirmation mail is
sent to the sender, with the Subject: SMS Server Personal Directory Configuration and the message in the body:
New entry is added successfully at index-XXXX.
To add multiple contacts
To add multiple contacts via single mail, you must enter the contacts by separating them with a comma(,). Given
below is the format of the mail.
To: smssever@domain.com
Subject: James Smith=+919898985400, Steve D=898954045
Here,
The Subject contains the details of the contacts.
The names to be added are James Smith and Steve D and their numbers are +919898985400 and
898954045 respectively.
• Make sure you do not enter any space before and after "=".
The contact is added at a first free index, between 0001 to 0025 in the Personal Directory and a confirmation mail is
sent to the sender, with the Subject: SMS Server Personal Directory Configuration and the message in the body:

New entry is successfully added at index-XXXX, Name: XXXXX, Number: XXXXX. Where XXXXX=is the
information received in the mail for configuration. A separate mail is sent for each contact that is added.
View details of existing contact

You can view the numbers saved against a Name or Group in the Personal Directory. Given below is the format of
the mail sent to the SMS Server.
Given below is the format of the mail to be sent to the SMS Server, to get the details of the existing contacts.
Subject: HDFC Customer Care? (this is the name/group)
Here,
The Subject contains the Name/Group Name whose details are required by you.
The Group Name is HDFC Customer Care.
The server will send a return mail with the number(s) as given below:
HDFC Customer Care:
James: +919898985400
John: +919897894512
Steve: +91898954045
If multiple entries exist, the return mail will contain the details of all. For example, if the query is made for the name
James, that is Subject: James?. If the system finds two entries (James D and James S) then reply email contains
both entries.
James D: 9426712345
James S: 9426921345
Similarly, if you want the details of all the names/groups beginning with J, then return mail will contain all names
starting with character "J" and each name will be displayed in a separate row, followed by their numbers.
James D: 9426712345
James S: 9426921345
John S :9996565123
If no match is found for the Name/Group Name, the system sends a reply mail with the Subject: SMS Server Error
Cause and the message in the body: X not found in Personal Directory, where X is the actual name.
Deleting a Contact
To delete an entry from the Personal Directory, the mail to be sent to the SMS Server must be in the format as
given below.
Subject: HDFC Customer Care=
Here,
The Subject contains the Name/Group Name that you want to delete.
In this case, there are three contacts +919898985400 (James), +919897894512 (John), +91898954045
(Steve) that will be deleted.

If the system is able to delete the entry, a reply mail with Subject: SMS Server Personal Directory Configuration and
the message in the body: X is deleted from Personal Directory', where X is the particular name which is received for
deletion of the contact/group name. In this case, the reply Email will be HDFC Customer Care is deleted from
Personal Directory.
If the SMS Server received a delete request for a contact and the same is not configured in the Personal Directory,
then a reply mail is sent to the sender, with the Subject: SMS Server Error Cause and the message in the body: X
not found in Personal Directory. Where, X is the actual name.
Modify/Edit a Contact
You cannot edit any contact in the Personal Directory via Email.
To modify any contacts details, you must first delete the existing contact from the Personal Directory. Then you can
add the same contact with new contact details.
Configuring Personal Directory using a Telephone

• Enter SE mode.
To program a telephone number in a personal directory, dial:
• 1902-Personal Directory-Location Code-Number

Where,
Personal Directory is from 01 to 50.
Location Code is from 01 to 25.
Number is the telephone number, max. 16 digits. If the number has fewer than 16 digits, you must dial
#* to terminate the command.
To clear a telephone number from a location in a personal directory, dial:

• 1902-Personal Directory-Location Code-#*
To program a name in the personal directory, dial:

• 1903-Personal Directory-Location Code-Name
Where,
Name is a string of alphanumeric characters, max. 12 characters. If the name has fewer than 12
characters, you must dial #* to terminate the command.
To clear a name from a location in a personal directory, dial:

• 1903-Personal Directory-Location Code-#*
To program the TAC Index for personal directory, dial:

• 1904-Personal Directory-Location Code-TAC Index
Where,
PM Group is from 01 to 50.
TAC Index is from 1 to 6.
By default, the TAC Index 1 is assigned to Personal Directory.

• To assign a Personal Directory to an extension, dial the following commands:
If the extension is an SLT, dial:

• 1905-1-SLT-Personal Directory to apply personal directory on a single extension.
• 1905-2-SLT-SLT-Personal Directory to apply the same personal directory on a range of extensions.
• 1905-*-Personal Directory to apply the same personal directory on all extensions.
Where,
SLT is the Software Port Number of SLT from 001 to 512.
By default, 00 is the Personal Directory assigned to all SLTs.
To clear the personal directory assigned to the SLT, dial:

• 1905-1-SLT-00 to clear personal directory from a single SLT.
• 1905-2-SLT-SLT-00 to clear personal directory from a range of SLTs.
• 1905-*-00 to clear personal directory from all SLTs.
If the extension is a DKP, dial:

• 1906-1-DKP-Personal Directory to assign personal directory to a single DKP.
• 1906-2-DKP-DKP-Personal Directory to assign the same personal directory to a range of DKPs.
• 1906-*- Personal Directory to assign the same personal directory to all DKPs.
Where,
DKP is the Software Port Number of the DKP from 001 to 128.
By default, 00 is the Personal Directory assigned to all DKPs.
To clear the personal directory assigned to the DKP, dial:

• 1906-1-DKP-00 to clear personal directory from a single DKP.
• 1906-2-DKP-DKP-00 to clear personal directory from a range of DKPs.
• 1906-*-00 to clear personal directory from all DKPs.
If the extension is an ISDN Terminal, dial:

• 1907-1-ISDN-Personal Directory to assign personal directory to a single ISDN Terminal.
• 1907-2-ISDN-ISDN-Personal Directory to assign the same personal directory to a range of ISDN
Terminals.
• 1907-*-Personal Directory to assign the same personal directory to all ISDN Terminals.
Where,
ISDN is the Software Port Number of the ISDN Terminal from 01 to 64.
By default, 00 is the Personal Directory assigned to all ISDN Terminals.
To clear the personal directory assigned to an ISDN Terminal, dial:

• 1907-1-ISDN-00 to clear personal directory of a single ISDN Terminal.
• 1907-2-ISDN-ISDN-00 to clear personal directory of a range of ISDN Terminals.
• 1907-*-00 to clear personal directory of all ISDN Terminals.
Where,
ISDN is the Software Port Number of the ISDN Terminal from 01 to 64.
By default, 00 is the Personal Directory assigned to all ISDN Terminals.
• Exit SE mode.

Configuring Global Directory using Jeeves
• Click Global Directory to open the page.
Each page of has 100 entries. To go to the next 100 entries click the links above the table '200-299'
• Enter the Number you wish to store against a Memory Location Code. Enter the contact's Name
against the number.
The length of the Number field is limited to 16 digits. The length of the Name field is limited to 12
alphanumeric characters. All ASCII characters except < > and “ (double quote) are allowed. Ensure
that the number and the name are programmed within this limit.
• Change the OG Trunk Bundle Group, if required. Default: 01. See “OG Trunk Bundle Group”.

While routing the Global Directory number, the system will use the OGTBG configured in the Station
Advanced Feature Template assigned to the Extension that is dialing a Global Directory number. So, if you
want the system to route Global Directory numbers using OGTBG you configured here, make sure you
have selected OGTBG configured in the Global Dir. in the Route Global Directory Calls using
parameter of the Station Advanced Feature Template.
• Click Advance to configure the Email ID and select the Group for the contact.
• Enter the Email ID of the contact you wish to store. The Email ID can be a maximum of 64 characters.
• You can assign the contact to a Group. Select the desired Group Type from the list. The system clubs
together contacts assigned the same Group. Default: None. For details, see “SMS/Email Group”.
Applying Global Directory to Extensions

To apply the Global Directory to extensions,
• Make sure that the feature Global Directory is enabled in the CoS of the extensions to which you are
assigning the Global Directory.
By default, Global Directory Part 1 is allowed to all extensions in their CoS.
If the entire directory is to be assigned to all extensions, you may simply enable 'Global Directory Part 2
and Part 3 in the default CoS group 01 in the default Station Basic Feature Template 01 assigned to the
extensions.
However, if selected extensions are to be allowed Global Directory Part 2/Part 3, follow these steps:
• Define a CoS group with Global Directory Part 2/Part 3 enabled.

• Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
• Assign this template to the extensions to which Global Directory Part2 /Part 3 are to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for further instructions.
• Decide which of the DKP Extension users are to be allowed ‘Global Directory Programming’ (of Global
Directory Part 1) and allow this feature in their Class of Service.
By default, Global Directory Programming is disabled in the default CoS group 01 in the default Station
Basic Feature Template 01 assigned to all extensions of the system. This means none of the extensions
can program Global Directory.
If you want to allow Global Directory Programming to all DKP extension users, simply enable this feature in
the CoS group of the Station Basic Feature Template assigned to them.
If you want to allow Global Directory Programming to only selected extensions, then follow these steps:
• Define a CoS group with Global Directory Programming enabled.
• Make sure this CoS also has Global Directory Part 1 enabled.
• Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
• Assign this template to the DKP extensions to which Global Directory Programming is to be allowed.

Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for programming
instructions.
• If Global Directory Part 1, 2 or 3 is assigned to an extension user, the system will not check for Toll
Control.
• When you assign Global Directory Programming to a DKP extension user, the user can program any
number in Global Directory Part 1, this includes numbers denied to the extension user in the Call
Privilege defined in the Toll Control level of this extension user.
• Since the system does not check for Toll Control for numbers dialed out from Global Directory Part 1,
there is a possibility of extension users programming numbers not allowed to them in their Toll Control
level in the Global Directory Part 1, inadvertently or intentionally.
• Hence, the System Engineer is advised to exercise caution when allowing this feature to DKP
extension users.
• Click Submit at the bottom of the pages on which you make changes to save your settings.
• If you have finished configuration, you may log out of Jeeves. Or you may continue, as required.
Upload Global Directory CSV file

• Click Upload Global Directory Contacts to expand.
• Select the Clear all other indices of the Global Directory, which are not specified in the .csv file
being uploaded check box, to overwrite the existing contacts in the Global directory with the contacts
of CSV file. Default: Disabled.
• Click the Browse button to Select the .csv file to be uploaded from the location on the local disk.

• All the contacts of the CSV file will be uploaded in the Global Directory. To view, click the Global
Directory link.
Download Global Directory CSV files

• Click Download Global Directory Contacts to expand.
• Click the Download button.
• You will get a prompt with an option to open the Opening Global_Directory.csv file or save the file to
a location. Save the file on the local disk.
• To Open the Global_Directory.csv file from the location on the local disk, make sure you follow the
steps given below:
• DO NOT OPEN THE CSV FILE DIRECTLY WITH EXCEL!
• Open a New worksheet in Excel.

• Open the Data tab and select From text button in the Get External Data.
• Select your CSV file from the location on the local disk to import.
• In Original data type section, select Delimited radio button and click Next.

• In Delimiters select the Comma check box (column dividers will appear in preview) and click Next.

• Select the column with leading zeros and in Column data format select the Text radio button.
You will have to do this for each column where the data contains leading zeros.
• Click Finish.
• The leading zeros will still be there in the new worksheet with the imported data.
Configuring Global Directory using a Telephone

• Enter SE mode.
To program a telephone number in the global directory, dial:

• 1801-Location Code-Number
Where,
Number is the telephone number, max. 16 digits. If the number has fewer than 16 digits, you must dial
#* to terminate the command.
To clear a telephone number from a location in the global directory, dial:

• 1801-Location Code-#*
To program a name in global directory, dial:

• 1802-Location Code-Name
Where,
Name is a string of alphanumeric characters, max. 12 characters. If the name has fewer 12 characters,
you must dial #* to terminate the command.

To clear a name from a location in the global directory, dial:
• 1802-Location Code-#*
To program the OGTBG for a contact in the global directory, dial:

• 1803-1-Location Code-OGTBG to assign OGTBG to a single Location code.
• 1803-2-Location Code-Location Code-OGTBG to assign OGTBG to a range of Location Codes.
• 1803-*-OGTBG to assign OGTBG to all Location Codes.
Where,
By default, the OGTBG assigned is 01.
To clear a location code in the global directory:

• 1800-1-Location Code
• 1800-2-Location Code-Location Code
• 1800-*
• Exit SE mode.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for instructions on how to use
SE commands to
• enable a feature (Global Directory) in the CoS Group.
• assign the CoS group to a Station Basic Feature Template
• apply the template with the newly programmed CoS group to the extensions.
Personal Directory Configuration by Extension Users

Extension users can program their own Personal Directories using their extension phones. The personal directory
may be programmed using DKP or SLT.
If you are using an SLT, you will not be able to program the Name of the contact in the directory.
For EON and Extended IP Phone Users

• Press DSS Key assigned to Personal Directory (if programmed).
OR
• Dial 1071.
• Enter Personal Memory Index (01 to 25)
• Enter Number of the contact (max. 16 digits).
• Press 'Enter' key.
• Enter Name of the contact.
• Enter Trunk Access Code.
• You get confirmation tone and the message on your phone's display.

For SLT Users
• Pick up handset.
• Dial 1071.
• Dial Personal Memory Index (01 to 25).
• Dial Number of the contact (max. 16 digits).
• Press #*.
• Dial Trunk Access Code.
• You get confirmation tone.
• Replace handset
Global Directory Configuration by Extension Users

Extension users can add, delete and edit contacts in Global Directory Part 1 using their extension phones, provided
that:
• their phone is a digital key phone;
• Global Directory Part 1 is allowed to them in their Class of Service.
• Global Directory Programming is allowed to them in their Class of Service.
• Extension users can only add, delete and edit names and numbers of contacts in Global Directory Part
1. However, they cannot program the Outgoing Trunk Bundle Group (OGTBG) for the contacts in the
directory.
• When an extension user programs Global Directory Part 1, the system will automatically assign the
number and name to a free Memory Location. The system will use the OGTBG assigned to that
Memory Location by the System Engineer to dial out the number added by the extension user.
• By default, OGTBG 01 is assigned to all Memory Location Codes in the Global Directory.
• To simplify configuration for both the System Engineer and the extension users, the System Engineer
is recommended to assign the same OGTBG number uniformly to all Memory Location Codes, and
enable Least Cost Routing on this OGTBG.
• If no OGTBG has been assigned to a Memory Location in the Global Directory (that is, the field is
blank), and an extension user adds a contact to this Memory Location, the number will not be dialed
out.
To program Global Directory Part 1 from the Digital Key Phone or Extended IP Phone, follow these steps:
• Press Enter key to enter Phone Menu.

• Scroll to ‘Contacts’ and press Enter Key.
• You will get the following options:
• Add
• Edit
• Delete
Adding a contact
To add a contact, select ‘Add’ and press Enter key.
• Enter your contact’s name on the prompt: ‘Name:’
A maximum of 12 characters are allowed.
• Press Enter key to save name.
• Enter your contact’s number on the prompt: ‘Number:’

A maximum of 16 digits are allowed.
• Press Enter key to save number.
• You will get the confirmation tone and the confirmatory message: “Stored at Index xxx”.
Editing a contact
To edit a contact,
• Scroll to ‘Contacts’ in the phone menu and press Enter key.
• Select ‘Edit’ and press Enter key.
• You get the prompt: 'Name:'
• Enter the initial letters of the contact's name.
• The number of matching entries that will appear at a time on your phone's display will vary according to
your phone's LCD display capacity.
• Scroll with the Up/Down navigation keys to reach the desired contact's name on the list.
• Press 'Enter' key to select the name.
• The system displays the name you selected.
• To delete a character, use the Back/Forward navigation key to place the cursor under the character you
want to delete.
• Press the ‘Cancel’ key to delete the character you selected with the cursor.
• To enter a character, use the Back/Forward navigation key to place the cursor in the position you want to
enter the character.
• Enter the desired character by pressing the relevant digit pad keys in quick succession.
• After you have finished editing the name/ number, press Enter key.
• The number of the contact whose name you edited will be displayed.
• Repeat the same steps as you did for editing the name.
• After you have finished editing the number, press Enter key.
• You will get the confirmation tone and the confirmatory message: “Stored at Index xxx”.
Deleting a contact
To delete a contact,
• Scroll to ‘Contacts’ in the phone menu and press Enter key.
• Select ‘Delete’ and press Enter key.
• You get the prompt: 'Name'
• Enter the initial letters of the contact's name.
• Press 'Enter' key to delete the name.
• You will get the confirmation tone and the confirmatory message: ’Deleted’.
How to use
Personal Abbreviated Dialing

• Press DSS Key assigned 'Abbreviated Dialing' function (if programmed).
OR
• Dial 8 (users world wide)

OR
• Dial 6 (users in USA)

• You get the message 'Give Index'
• Enter Personal Directory Index number: 01 to 25.
• The desired number will be dialed out.
For SLT Users

• Pick up the handset.
OR

• Dial Personal Directory Index number: 01 to 25.
Global Abbreviated Dialing
For EON Users

• Press DSS Key assigned 'Abbreviated Dialing' function (if programmed).
OR
OR

• You get the message 'Give Index'
• Enter Global Directory Index number: 100 to 999.
For SLT Users

OR

• Dial Global Directory Index number: 100 to 999.

Alternate Number Dialing for EON74201
What's this?
Matrix EON74 is a special phone designed for use in trading environment, where call traffic is very high and each
trader handles multiple calls simultaneously to trade for and to communicate with customers. Alternate number
dialing feature helps you to dial an alternate phone number of the person or organization you are trying to reach
quickly by just pressing a DSS Key.
Central Office (CO) provides various facilities to its users such as—Call Hold, Call Transfer, Call waiting, Call
Forward, Hotline along with specific codes to access them.
CO allows you the facility to set Hotline for any one number on the CO Trunk. Using Alternate Number Dialing
feature, you will be able to dial an alternative number for CO Hotline number. See “Abbreviated Dialing” for more
details.
Matrix EON74 will work only with DKP8 and DKP16 cards of ETERNITY LENX/ MENX system and you
must have a Dealer Board license.
How it works
For this feature to work, you must:
• configure the Alternate Number table.
• assign a DSS key for the CO Trunk with the Hotline enabled on it.
• assign a DSS key for Abbreviated Dialing.
To assign DSS keys, see “DSS Keys Programming” for instructions.
This is how Alternate Dialing works:
• A EON user presses a DSS key assigned to the CO trunk with Hotline enabled on it.
• The Hotline number set for the CO Trunk will be dialed out.
• If the dialed number is busy or not responding, then EON user presses the DSS key assigned to the
Abbreviated Dialing followed by the CO trunk DSS key whose alternate number need to be dialed out.
Instead of pressing the DSS key assigned to the Abbreviated Dialing, a EON user may dial the Feature
Access Code ‘8’ to invoke Abbreviated Dialing feature.
• SARVAM UCS will check the Alternate number table to find out the Quick Dial Index number
corresponding to the respective CO trunk and will dial out the number configured at that index in the Global
Directory.
• In case the alternate number dialed by the system is also busy or not responding, a EON user may set
Auto Redial to redial the alternate number.
201. Check for availability. This feature will be visible only if your system supports EON74.

How to configure
1. Make a list of CO Trunks with Hotline enabled on it.
2. Make a list of Alternate numbers to be dialed out from the CO Trunks.
3. Configure the Alternate numbers in the Global Directory. For instructions, see “Abbreviated Dialing”.
4. Configure the Alternate Number table. For instructions, see “Configuring Alternate Number table using
Jeeves”.
Configuring Alternate Number table using Jeeves

• Under Configuration, click Alternate Number to open the page.
• In the Quick Dial Index (Global Directory), enter the Global Directory index number at which you have
configured the alternate number for the respective trunk.
How to use
To dial an alternate hotline number:
• Press the DSS Key assigned to the Abbreviated Dialing.
• Then press the DSS Key assigned to the CO Trunk
• The alternate number will be dialed out.
• Talk when the remote party answers.

Access Codes
What's this?
Access codes are short digit sequences dialed from an extension phone to instruct the System to perform a
function such as:
• Calling an extension.
• Calling a group of extensions (“Department Call”).
• Grabbing a trunk line or any trunk line from a group of trunks (“OG Trunk Bundle Group”).
• Invoking a feature. Activating or deactivating a feature.
Accordingly Access Codes are classified into:
• Station Codes: Codes used for calling extensions. These codes are also commonly referred to extension
numbers, phone numbers. For the purpose of this document, station codes are referred to as Flexible
Numbers.
Default station codes: the factory-set default values for SLT extensions are from 2001 to 2512; for DKP
extensions from 3001 to 3128.
• Logical Group Codes: codes used for calling a group of stations as in a Department group, a group of
trunks as in Outgoing Trunk Bundle Group.
Default logical group codes: the factory-set codes for Department Numbers start from 3901, 3902….
Outgoing Trunk Bundle Groups from 61, 62, etc.
• Feature Codes: codes used for invoking a feature.
Default feature codes: there are different feature codes for every feature/function of the SARVAM UCS,
e.g.: '2' for Auto Call Back, '5' for Raid, '13' for Call Forward, etc.
Access codes may consist of single digits or a sequence of a maximum of 6 digits.
You can change the default access codes to codes of your choice. For example: the default Operator code
'9' can be changed to '0'; the default trunk access code for dialing Trunk Group1 can be changed from '61'
to '5'.
How it works
Whenever an access code is dialed from an extension, the system matches each digit in the code with the access
codes programmed within the system to determine the instruction, that is, whether it is an extension it must call, or
a trunk line it must grab, a port it has to activate, etc. The system processes the instruction when a match is found.
For example:
• An extension user dials 131 to set Call Forward.

• When the first digit '1' is dialed, the system finds a match. As several default access codes begin with '1'
the system waits for the next digit to be dialed.
• When the second digit '3' is dialed, the system finds a match for '13'.
• As '13' is common for all Call Forward options202, the system waits for the next digit to be dialed
• When the user dials the third digit '1', the system finds a match for '131'.
• If there is more than one access codes matching with '131', e.g. '1311', '1314', '1315' the system will wait
for the next digit to be dialed.
• If no further digit is dialed on expiry of the Inter Digit Wait Timer, the system understands the instruction as
'Call Forward - Unconditional' and waits for the destination phone number to be dialed.
Access Codes are related to various phases of a call. When a call is processed by a System, it goes through
a number of pre-defined phases.
Typically a call passes through the different phases as shown below:
Idle Dial Routing Blocked Placed Matured Matured Denied

2-Way 3-Way
No Digits are The system The dialed The dialed Connected Connected No reply
activity. pressed on is processing extension extension is with the with two from dialed
the phone the call. The is busy. ringing. dialed extensions. extension.
keypad/dialed call is neither extension.
from the placed nor
rotary. blocked.
Dial tone is Beeps are Busy tone Ring Back Two-way Three-way Error Tone
played. played. is played. Tone is speech. speech. is played.
played.
Different access codes are dialed at different call phases. Station Codes and Logical Group Codes are dialed in the
'Dial' phase.
As different features are invoked in each call phase, Feature Access Codes are dialed at different call phases. For
example:
• Call Forward code is dialed at the 'Dial' phase'.
• DND Override code is dialed at the 'Routing' phase'
• Auto Call Back code is dialed at the 'Blocked' phase' as well as 'Placed' phase.
• Three-party Conference code is dialed at the 'Matured 2 way' phase.

'Idle' phase is when no code is dialed. In the 'Denied Phase' no code is allowed to be dialed.
202. Call forwarding options: Unconditional, When Busy, When No Reply, When Busy or No Reply.

Each access code in a single call phase may be of different lengths, but must be unique. For example, the
same access code cannot be used for two different features like Call Forward and Redial, since both these
features are invoked in the 'Dial' phase.
However, the same access code can be used for features in different call phases. For example, '4' is the
default feature access code for DND Override (Routing Phase), Call Pick-Up-Group (Dial Phase) and Barge-
In (Blocked Phase).
Similarly, Station and Logical Group Codes too must be unique and should not match with any of the features
invoked in the 'Dial' phase. Refer the topics “Flexible Numbers” and “OG Trunk Bundle Group” to know more.
How to configure
SARVAM UCS provides default Access Codes for stations, logical groups - department and trunk groups - and
features.
It also provides country-specific default Access Codes which are applied automatically when you select the
'Region' to configure the system.
The default Access Codes for India are presented in the table below. The default Access Code tables also indicate
the call phase in which each feature is invoked.
Call Phases
Feature Access
Feature Matured Matured
Number Code Dial Routing Blocked Placed
2-way 3-way
Enter SE Programming 1 1#91 Y Y

Mode
Enter SA Programming 2 1#92 Y Y

Mode
Call Pickup - Group 3 4 Y Y
Call Pickup - Selective 4 12 Y Y
Auto Call Back - Set 5 2 Y Y
Auto Call Back - Cancel 6 102 Y Y
Redial 7 7 Y Y
Auto Redial - Set 8 17 Y Y
Auto Redial - Cancel 9 1070 Y Y
Personal Directory 10 1071 Y Y

Programming
Abbreviated Dialing 11 8 Y Y
Operator 12 9 Y Y Y
Call Forward 13 13 Y Y
Dynamic Lock 14 14 Y Y
Hotline 15 15 Y Y
Alarm 16 161 Y Y

Call Phases
Feature Access
2-way 3-way
Do Not Disturb 17 18 Y Y
Interrupt Request 18 3 Y
Barge-In 19 4 Y
Raid 20 5 Y
Trunk Reservation 21 6 Y
Call Toggle 22 1 Y
Conference 23 *3 Y
PIN Dialing 24 *2 Y Y
Dial-In Conference 25 *19 Y Y
Call Park 26 115 Y
Call Park - Retrieve 27 116 Y Y
Room Monitor 28 1073 Y Y
Last Caller Recall 29 1092 Y Y
Voice Help 30 1090 Y Y
Walk-In Class of Service 31 111 Y Y
Change User Password 32 114 Y Y
Paging 33 1074 Y Y
DISA Login 34 1079 Y
Trunk to Trunk Call Release 35 ## Y
Cancel all Feature Settings 36 1051 Y Y
Selective Port Access 37 69 Y Y
Flashing on Trunk 38 * Y Y
User Absent/Present 39 104 Y Y
Account Code by Number 40 1058 Y Y
Account Code by Name 41 1059 Y Y
Meet Me Paging 43 1093 Y Y
Hot Desk 44 1091 Y
Do Not Disturb Override 45 4 Y
Presence 46 1097 Y
Live Call Screening 47 1094 Y Y
Conversation Recording 48 1095 Y
Forced Release 49 #* Y
Transfer 50 F Y
Live Call Supervision 51 1098 Y Y

Call Phases
Feature Access
2-way 3-way
Forced Answer 52 5 Y
Change Room Clean Status 53 1054 Y
Guest Number Prefix 54 1055 Y
Minibar Details 55 1056 Y
Mute 56 1052 Y Y
Emergency Conference 57 1177 Y
Self Ring Test 58 1057 Y
Call Chaining 59 1050 Y
SA Command Prefix 60 1072 Y Y
COSEC Door Open 61 *7 Y
Floor Service 62 38 Y Y
Keypad Lock 63 - Y
CLI Restriction 64 103 Y
Call Cost Display 65 1075 Y
Reminder 66 162 Y Y
Alarm - Voice Guided 67 163 Y
Reminder - Voice Guided 68 164 Y
Blind Transfer to Voice Mail 69 1078 Y
Message Wait Set/Cancel 70 1076 Y Y
Retrieve New Message 71 1077 Y
PMS - User Defined Fields 72 1096 Y
Magneto Ring Enable 77 # Y
Invoke RCOC 78 ** Y
Scheduled Call Forward 79 1075 Y Y
E&M Manual Priority 80 *37 Y

Intrusion
E&M Forced Release Order 81 *38 Y
Department Call Forward 82 1179 Y Y
General Mailbox 83 1176 Y Y
Intercom 84 *5 Y Y
Terminate Conference 86 190 Y
Leave Temp. / Rejoin Conf. 87 191 Y
Call Forward - When Not 88 *13 Y

Registered

Call Phases
Feature Access
2-way 3-way
Trunk Access Code 1(TAC1) - 0 Y
Voice Mail - 3931 Y
You can either use the default Access Codes or change them to suit your preferences.
Changing Feature Access Codes using Jeeves

• Under Configuration, click Access Codes.
• Click Feature Access Code to open the page.
• Change the Access Codes as per your preference. A maximum of 6 digits.

If you enter a code that is already assigned to a station or a feature, the system will not accept the
duplicated code. The value will remain unchanged.

• To disable an access code, delete the existing code and leave the field blank.
• Click Default all to default all Access Codes.
To change Station Access Codes (for extensions), Department Groups, Trunks and Trunk Groups, and
General Mailbox Access Code refer the topics “Flexible Numbers”, “Department Call”,“OG Trunk Bundle
Group” and “General Mailbox Settings”.
Changing Feature Access Codes using a Telephone

• Enter SE mode from an SLT/DKP.
To program the access code for a feature, dial:

• 3111-1-Feature Number-Access Code-#*
Where,
Feature Number is from 01 to 88.
Access Code is a string of maximum 6 digits
• E.g.: To change the prefix of SA command '1072' to '107' dial: 3111-1-60-107-#*

Where,
60 is Feature Number for SA command.
107 is the new Access Code for SA command.
If you try to assign a number string that is already used to access an extension or use a feature then
the system will not accept the command and will play error tone.
To default access codes, dial:

• 3161-1-Feature Number to default access codes for a single feature.
• 3161-2-Feature Number-Feature Number to default access codes of a range of features.
• 3161-* to default access codes of all features
Where,
E.g.: To default the prefix of SA command '107' to '1072' dial: 3111-1-60
A default trunk access codes table is given below:
OGTBG Index Default Trunk Access Code
1 0
2 5
3 61
4 62
5 63
6 64
To assign access code for a trunk access index, dial:

• 3112-1-OGTBG Index-Access Code-#*
Where,
TAC Index is from 1 to 6.
Access Code is maximum 6 digits (Generally access code for trunk is of two digits).

To clear the access code for a trunk access index, dial:
• 3112-1-OGTBG Index-#*
• 3112-2-OGTBG Index-OGTBG Index-#*
• 3112-*-#*
To assign default access code for a TAC index, dial:

• 3162-1-OGTBG Index
• 3162-2-OGTBG Index-OGTBG Index
• 3162-*
To program the access code for a VMS Group, dial:

• 3114-1-VMS Group Index-Access Code-#*
• 3114-2-VMS Group Index-VMS Group Index-Access Code-#*
• 3114-*-Access Code-#*
Where,
VMS Group Index is 1.
Access Code is maximum of 6 digits. If it is less than 6 digits, terminate it with #*
By default, the Access Code of the VMS Group is 3931.
To default the access code for a VMS Group, dial:

• 3164-1-VMS Group Index
• 3164-2-VMS Group Index-VMS Group Index
• 3164-*
Where,
To clear the access code for a VMS Group, dial:

• 3114-1-VMS Group Index-#*
• 3114-2-VMS Group Index-#*
• 3114-*-#*
Where,
• Exit SE mode.

Account Codes
What's this?
Account Codes are very useful feature for organizations such as business consultants, law firms, advertising and
media agencies, and the like that cater to several clients, interacting with third parties on behalf of their clients.
Such organizations need to keep track of calls made to and on behalf of each client.
An 'Account Code' is a unique three-digit number that an organization can assign to each of its clients. Each
Account Code may be given a name and programmed in the Account Name List.
Doing so, whenever calls are made to the client or to a third party on behalf of the client,
• The extension user dials the Account Code or Name assigned to the client.
• The Account Code may be dialed,

• before dialing the external number.
Or
• when in speech with the client/third party
• Details of these calls are recorded by the Account Code dialed in the Station Message Detail Recording
Report (SMDR) for Outgoing Calls.
• The SMDR report can be printed using the Account Code as filter.
This way, the organization can know the details of calls made to and on behalf of each client.
• The SARVAM UCS supports as many as 999 Account Codes.
How it works
For example, an advertising media agency makes nearly 100 calls every day to and on behalf of its clients that
includes 'Midas Business Solutions', 'Jet-Set Holidays', 'Bacchus Vineyard'.
• Assign a three-digit account code to Midas Business Solutions, for instance '001' and the name code
'Midas Biz' in the Account Name List.
• Assign a three-digit account code to Jet-Set Holidays, for instance '002' and the name code 'Jet' in the
Account Name List.
Case 1: Applying same Account Code
• A, a person from advertising media agency makes a call to Midas Business Solutions and talks to the
secretary B. During an ongoing conversation A dials the account code 001. In between, A needs to consult
the manager C. Therefore, A presses the Transfer key to put B on Consultation hold and dials the
number of C. Account code 001 will be applicable to the second call made to C also.

Case 2: Applying different Account Codes
• A, a person from advertising media agency makes a call to Midas Business Solutions and talks to the
secretary B. During an ongoing conversation A dials the account code 001. In between, A needs to talk to
another client, say Jet-Set Holidays D. Therefore, A pressed the Call Hold key to put B on Exclusive/
Global hold and dials the number of D. While in speech with D, A dials account code 002. Here, Account
Code 001 will be applicable to Midas Business Solutions and Account Code 002 will be applicable to Jet-
Set Holidays.
Forced Account Code

The SARVAM UCS can be programmed to prompt extension users to dial the Account Code whenever they grab a
trunk to dial out a number. Hence the feature name Forced Account Code.
To apply Forced Account Code, this feature must be enabled on the extensions and trunks from which calls using
Account Codes are to be made. When Forced Account Code is enabled on an extension, and the extension user
dials out a number or the Trunk Access Code to grab a trunk, the system will play an error tone. If the extension is
a DKP, the system will flash a message on the phone’s display that Account Code is required to make the outgoing
call.
• Account Codes are applicable for external calls only.
• To use Account Codes, this feature must be included in the Class of Service (CoS) group allowed to
the extensions.
• If you want to use Account Names, you must program the Account Name List.
• When the Forced Account Code is enabled on an extension and trunk, the system will ask the user to
enter the account code irrespective of the method of dialing: Global Abbreviated dialing, Personal
abbreviated dialing, Least Cost Routing, or Selective Trunk Access.
• However, if Forced Account Code is enabled on the selected trunk, and the number is dialed using
Selective Trunk Access, the system will dial out the number using Store and Forward dialing.
In the case of Abbreviated Dialing or Direct Dialing, if the extension user fails to dial the Account Code,
an error message will be displayed on the extension user's DKP.
How to configure
For Account Code to work, you must:
1. Enable Account Codes feature in the Class of Service (CoS) of the extensions to which this feature is to be
allowed.
2. Prepare and program the Account Name List, if it is to be used.
3. If Forced Account Code is to be used, you must enable 'Forced Account Code' flag in
• the “Station Advanced Feature Template” applied to the extensions from which calls using account
codes are to be made.
• the “Trunk Feature Template” applied on the trunks through which calls using account codes are to be
made.

All the above feature parameters can be programmed using Jeeves or by dialing commands from a
Telephone.
Preparing Account Name List

In consultation with the Users, you may prepare the Account Name list. You may do the following
• Draw a two-column table on a paper.
• Write Account Codes on one column. Account codes may be any three-digit number between 001 and
999.
• Write the Account Names, that is, names of the clients on the second column, against their respective
Account Codes.
• The names must not exceed 12 characters. All ASCII characters except < > and “ (double quote) are
allowed. For example:
Account Code Client Account Name
001 Midas Biz
002 Jet Set
010 Bacchus
You need not follow a cardinal numbering sequence when assigning Account Codes.
You may assign any code to any client. For instance, you can assign code '111' to Midas Business
Solutions, '222' to Jet-Set Holidays, '333' to Bacchus Vineyard.
Configuring Account Codes using Jeeves

After preparing the Account Name list,

• Under Configuration, click Account Name.
• Enter the Names of the clients against the account codes you have assigned to them. Refer the paper with
the two-column table you created.
Now, include the feature Account Codes in the Class of Service of the extensions.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all extensions of the
system. Template 01 has the feature Account Codes in the default CoS Group (Number 01). So, all extensions of
the system can use this feature.
If Account Codes is to be allowed only to selected extensions, follow these steps:
1. Define a CoS group with 'Account Codes' enabled.
2. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
3. Assign this new Template to the extensions to which Account Codes is to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions.
If Forced Account Code is to be used, enable the 'Forced Account Code' flag in the “Station Advanced Feature
Template” of the extensions and in the “Trunk Feature Template” assigned to the trunks.

When you enable 'Forced Account Code' in a template, this feature will be enabled on all DKP, SLT and
SIP extensions that are assigned this template. If necessary, create a separate template with this feature
and assign this template only to those extensions that are to be assigned this feature.
To enable Forced Account Code flag on Trunks,
• Go to the parameters page of the type of trunk you want to program. For example: CO Parameters page
to program CO Trunk, Mobile Parameters page to program GSM Trunk, SIP Parameters page to
program SIP Trunks etc.
• Click the Trunk Feature Template link in the column. The feature template page will open.
By default Trunk Feature Template 01 is assigned to all Trunk Types.
If you want to enable this feature on all trunks, enable it in Trunk Feature Template 01.
If you want to enable this feature only on select trunks, program a different Template number with this
feature.
• Select the Forced Account Code check box in the template number assigned to the trunk.
• Return to the trunk parameters page.
• Now change the Trunk Feature Template number of the trunk you want to program. This number should
be the same as the template in which you have enabled the Forced Account Code check box.
Repeat the above steps to configure all other trunk types.
Configuring Account Codes using a Telephone

• Enter SE mode.
To create the Account List:

• Dial command 4851-1-Account Code-Account Name
Where,
Account Code is 001 to 999.
Account Name is a string of 12 characters.
Terminate Account Name with #*, if it is less than 12 characters.

For example: to program account code '001' with the name 'Midas Biz', dial
4851-1-001-MIDAS BIZ-#*
Press the digit key '1' twice to give space between characters.
To enable Forced Account Code flag in Station Advanced Feature Template:

• Dial command 5602-1-Template Number-Feature Number-Flag Code
Where,
Template Number is Station Advanced Feature Template from 01 to 50. Default: 01
Feature Number for Forced Account Code is 09.
Flag Code is
0 for Disable

1 for Enable
For example: To enable Forced Account Code Flag in Station Advanced Feature Template 02: Dial
5602-1-02-09-1
To apply the Station Advanced Feature Template now programmed with the Forced Account Code on
stations, refer the topic “Customizing Station Advanced Feature Template using a Telephone”.
To enable Forced Account Code flag on a Trunk:

• Dial command 5802-1-Template Number-Feature Number-Flag Code
Where,
Feature Number for Forced Account Code flag is 29.
Flag Code is
0 for Disable
1 for Enable
For example: To enable Forced Account Code Flag in Trunk Feature Template 01: Dial 5802-1-01-29-1
To apply the Trunk Feature Template now programmed with the Forced Account Code on different
types of trunks, refer the topic “Customizing Trunk Feature Template using a Telephone”.
• Exit SE mode.
How to use
Account Codes can be dialed in two ways: by Number and by Names.
Account codes, that is, number and names, can be dialed:
• before making the call,
• during the call,
• when grabbing a trunk (if Forced Account Code flag is enabled).
Print and hand out copies of the Account Code List to everyone in the organization for reference while
making calls.
Dialing Account Code by Number
For EON Users and Extended IP Phone Users

To enter Account Code Number before making the call:
• Press DSS Key assigned to 'Account Code by Number'.
OR
• Dial 1058
• Enter Account Code
• Dial Trunk Access Code

• Dial the number of the client.
To enter Account Code Number during the call:
• Press ‘Transfer’ Key

• Press DSS Key assigned to Account Code by Number.
OR
• Dial 1058
• Enter Account Code
• Speech will be resumed.
To enter Account Code Number when Forced Account Code Flag is enabled:

• Press DSS Key assigned to Account Code by Number.
OR
• Dial 1058
• Enter the Account Code Number.
• You get dial tone.
• Dial Trunk Access Code followed by the number of the client.
If you dial the Trunk Access Code to grab a trunk, without dialing the Forced Account Code, you will get an
error tone. Go ON-hook and then go OFF-hook. Now follow the same steps in the sequence mentioned
above.
For SLT Users

• Go OFF hook.
• Dial Account Code first.
• Dial Trunk Access Code followed by the number of the client.
If you dial the Trunk Access Code to grab a trunk, without dialing the Forced Account Code, you will get an
error tone. Go ON-hook and then go OFF-hook. Now follow the steps in the sequence mentioned above.
Dialing Account Code by Name
Dialing Account Code by Names is possible only if your phone is a DKP / Extended IP Phone.

To enter Account Code Name before making the call:
• Press DSS Key assigned to Account Code by Name.
OR
• Dial 1059.

• Enter the initial letter of the client's name.
The Account Name List will be displayed on your DKP, alphabetically with the corresponding account
codes.
• Scroll to select the desired client name and press Enter key.
• Dial the client's number.
To enter Account Code Name during the call:
• Press Hold key to put the called party on hold.

• Press DSS Key assigned to Account Code by Name.
OR
• Dial 1059.
• Dial the initial letter of the client's name.
The Account Name List will be displayed on your phone, alphabetically with the corresponding account
codes.
• Scroll to select the desired client name and press Enter key.
Speech will be resumed with the called party.
Dialed the wrong account code or name?
If you have dialed the wrong account code or name while in the middle of a call, you can correct it by
pressing 'Hold' again and following the steps described above. The system will override the previously
dialed account code or name.
Printing Call Reports of Clients

You can print the call details of your clients using their account codes as filter.
Refer the section “Station Message Detail Recording-Report”, for more detailed instructions on printing reports
using filters.

AC Impedance Test
What's this?
SARVAM UCS supports the AC Impedance Test for clear, audible and echo-free speech over the CO Trunks. This
test helps you to set the most appropriate values for the CO Trunk parameters —AC Impedance, CO Termination
and CO Line Type— to correct the line impedance mismatch between the AC Termination Impedance presented by
the CO port of SARVAM UCS to the line and the CO Termination Impedance presented by the Central Office to the
line.
Conducting AC Impedance Test

You can conduct an AC Impedance Test,
• by making an outgoing call

or
• on an ongoing incoming call.
To conduct the AC Impedance Test you will need,
• a telephone with a valid number. You are recommended to use -a mobile phone with Mute function.
• the CO Trunk which you want to test.
For the CO Trunk you want to test, make sure you select a CO Hardware Template number.
The DSS Key assigned to this CO Trunk will display status as busy to all the user, for the duration of the
test.
Configuring Using Jeeves

You can conduct the test after you have set the relevant test parameters. To do this,

• Under CO Configuration, click AC Impedance Test.
To conduct an AC Impedance Test by making an outgoing call

• In Enter Phone Number to which call should be made, enter the phone number on which you want
to make test call. The number can be a landline or a mobile number. We recommend you to use a
mobile number for the test call.
If you are using a mobile phone number, be sure the handset of the configured number supports the Mute
function.
• In Make call using CO Port, select the CO trunk using which you want to make the test call. This must
be the same CO trunk for which AC Impedance is to be set.
• Select the Test Mode. You may select Reliable (Recommended) or Accurate.
The Reliable Test mode suggests the AC Impedance settings on the basis of most commonly used AC
Impedances, CO Terminations and CO Line Types across the globe. The test using Reliable Test
mode takes approximately 5 minutes to complete.
The Accurate Test mode suggests the AC Impedance settings on the basis of all the possible AC
Impedances, CO Terminations and CO Line Types across the globe. The test using the Accurate Test
mode takes 1 hour and 20 minutes to complete.
• Click the Start Test button. The system calls the phone number you configured. The message ‘Setting
up the Test....’ appears on your screen.

While the test is being conducted, you will hear pulsating tone on all the ports of the same card.
• Answer the test call from your mobile phone. You will hear the Music-on-Hold as per the type of Answer
Supervision you have configured in the CO Hardware Template assigned to the CO trunk you are
testing.
By default, the Answer Supervision selected in the template is Pseudo Answer and the Pseudo Answer
Supervision Timer is set to 10 seconds. If you have not changed this default setting, you will hear
Music-on-Hold after 10 seconds of answering the call.
• As the Music-on-Hold begins to play, Mute the microphone of your mobile phone.
If you are making the test call on a landline number, mute the call using the Mute key of the phone. If
your phone does not have a Mute key, unplug the handset cable from the phone body. This is to
prevent test signals from reflecting back into the mic of the handset.
• After 5 seconds of Music-on-Hold, you will hear the test signals being transmitted by the system for the
duration of the test. The message ‘’Reliable’ Test running successfully...’ appears on your screen.
• On completion of the test, the system will automatically disconnect the call. The message ‘Test
completed’ appears on your screen.
However, if you wish to abort the test midway, you may click the Abort Test button.

To conduct an AC Impedance Test on an ongoing incoming call
• Click the Click Here link on the AC Impedance Test page.
• A new window opens. A list of ongoing calls will be displayed. Select a call on which you want to
perform the test.
• Select the Test Mode. You may select Reliable (Recommended) or Accurate.
The Reliable Test mode suggests the AC Impedance settings on the basis of most commonly used AC
Impedances, CO Terminations and CO Line Types across the globe. The test using Reliable Test
mode takes approximately 5 minutes to complete.
• The Accurate Test mode suggests the AC Impedance settings on the basis of all the possible AC
Impedances, CO Terminations and CO Line Types across the globe. The test using the Accurate Test
mode takes 1 hour and 20 minutes to complete.

• Click the Start Test button. The message ‘Setting up the Test....’ appears on your screen.
While the test is being conducted, you will hear pulsating tone on all the ports of the same card.
• After 5 seconds the message ‘Test running successfully...’ appears on your screen.
• On completion of the test, the system will automatically disconnect the call. The message ‘Test
completed’ appears on your screen.
However, if you wish to abort the test midway, you may click the Abort Test button.
Suggested Impedance Settings

• At the end of the test, the page displays the Suggested Impedance Settings for the AC Impedance,
CO Termination, CO Line Type and Return Loss.
• You may now apply the suggested AC Impedance settings to the CO Trunk. To apply these settings,
select the desired CO Trunks and click on the Apply button.
• Verify the settings by making a trial call from a DKP. There should be no echo and speech should be
audible and clear.
If you still hear echo during the trial call, you may re-run the test using the Accurate Test mode.
• After you have determined the best matching AC Impedance, CO Termination, CO Line Type and
Return Loss by running the tests, apply the same suggested settings on the CO Trunk you are testing.
You may configure the same settings to all other CO Trunks, you have subscribed from the same CO
exchange.
It is possible that the CO trunks subscribed from the same exchange differ in their AC Impedance settings,
in such a case, you must run the test for each CO trunk separately and configure a different CO Hardware
Template for each of these trunks.
• To generate the detailed test report, click the Generate Test Report button.

The detailed test report appears in a new window. The Suggested Impedance Setting will appear in
bold.
• You may print the report by clicking the Print in the test report window.

• Select your Printer in the Printer options.
• You can also save the report in PDF format by selecting the PDF creator in your Printer options.
• Close the window to return to the AC Impedance Test page.
• Repeat the above steps to conduct further tests as per your requirement.

Configuring Using Telephone
To run the AC Impedance Test using the Accurate Mode, dial:

• 3361-1-CO Port Number
where,
CO Port Number is 001 to 128.
You can view the details of the test report on the port configured as the Syslog Server IP Address:Port
on the System Debug page.
To view the report, dial:

• 3362

Alarms
What's this?
Alarms are an efficient and user-friendly feature available to all extensions of the SARVAM UCS.
Alarms can be set and canceled
• by the Operator for extension users.
• by the extension users for themselves.
Alarms can be set as:
• Once Only - A one-time call, where the extension phone rings at the set time.
• Daily - A repeat call, where the extension phone rings at the set time everyday.
Alarms can be served as:
• Personalized - The Operator greets the extension user to serve the alarm request.
• Automated - The system serves the alarm request by playing a voice message or music.
How it works
Personalized Alarm
When the Alarm serving mechanism is configured as 'Personalized',
• The Operator phone rings first203, displaying the number of the extension to which the alarm is to be
served.
• When the Operator answers this call, a call is placed on the extension on which the alarm is set.
• The extension rings for the duration of the Alarm Ring Timer.
• When the extension user answers the call, the Operator greets the extension user with the time and alarm
message.
• This event is recorded in the Hotel-Motel Activity Log as 'Wake-up Alarm of <HH:MM> Answered on
<phone number>'.
• If the extension user does not answer the call till the Alarm Ring Timer has elapsed, the Operator phone
will display a text message notifying 'No Reply' from the extension. The Alarm is now considered as
served.
203. The Operator phone rings for the duration of the Alarm Ring Timer. If the Operator does not answer the call, the system will make
two more Alarm Attempts at an Alarm Attempt Interval of 5 minutes to call the Operator.

• This event is recorded in the Hotel-Motel Activity log as 'Wake-up Alarm of <HH:MM> No Reply on <Phone
Number>'.
• If the extension is busy204, the Operator phone will display a text message notifying that the extension
number is 'Busy'.
• The Operator can now choose to
• inform the extension user about the alarm in person or send someone to do it.
• try the busy extension again.
• set “Auto Call Back (ACB)”.
Automated Alarm
When the Alarm serving mechanism is configured as 'Automated',
• The extension phone rings at the set time till the end of the Alarm Ring Timer. If the extension phone is
from the EON series, an Alarm message will appear on its display.
• When the extension user answers the call, s/he may be played music-on-hold, or a pre-recorded voice
message or be connected to a routing group, depending upon the Alarm Notification Type programmed by
the System Engineer.
The System Engineer may consult with the Enterprise to decide which of these options is to be
programmed as the Alarm Notification Type.
• If the extension user does not answer the alarm call, the system makes two more attempts (in all, 3
attempts) at an interval of 5 minutes between each attempt, to call the extension. (Each attempt is
recorded in the Hotel-Motel Activity log as 'Wake-up Alarm of <HH:MM> No Reply on <Phone Number>'.
• If all Alarm attempts go unanswered, the system places the call on the Operator phone. The Operator
phone rings till the end of the Alarm Ring Timer. The Operator phone displays the extension number with
the message 'No Reply'. The Alarm is now considered as served. (This event is recorded as "Alarm
Notification to Front Desk for <Phone Number>").
• If the extension phone is busy the system will continue to make Alarm Attempts at the Alarm Interval
programmed. When all Alarm Attempts go unanswered, the system will place a call on the Operator
phone. The Operator phone will display the number of the extension phone with the message 'Busy'.
The Snooze function can be added to Automated-Alarms to ensure that the extension user answers the
call. Snooze is a system-wide feature; when set, this function will be added to all Automated Alarms.
When Snooze is activated,
• The extension phone rings for the Number of Alarm Attempts programmed, at set Alarm Attempt
Intervals.
204. An improperly placed receiver may also be the cause for the busy tone on the extension phone. In that case, the system will notify
the Operator Phone with the 'OFF-Hook Alert'. This event is recorded in the Hotel-Motel Activity Log as "Alarm not Served, <phone
number> is Busy".

• The extension stops ringing, when the extension user answers the call and dials the Code '0' to
acknowledge the Alarm. Please note that this Alarm Acknowledgement Code is non-programmable.
Alarm Status Report

The Operator can know the status of Alarms (details of Alarms that have not been served) from the WakeUp
Alarms Reports from SA mode or by pressing the DSS key assigned to Wakeup Call Log.
The WakeUp Alarms Report is useful when Operators change shifts.
Using SA Mode
The Jeeves displays the status of Alarms set by Operator as well as extension users appears on this report, with
details of time (hours and minutes), type (once only, daily), and serving mechanism (personalized, automated). The
Alarm Report generated by the system can be printed or sent to a computer.
Using DSS Key

The DKP / Extended IP Phone users can also view the Wakeup Call Log Status on the phone LCD using the DSS
key assigned to Wakeup Call Log. For instructions to assign a DSS Key to Wakeup Call Log, see “Configuring
Alarm Parameters using Jeeves”.
The Wakeup Call Log205 is the log of:

• Unanaswered Alarm Calls - This log will display both Alarms Calls that have not been answered as well as
unacknowledged Alarm Calls.
• Pending Alarm Calls - This log will display Alarm Calls which have been set for a later time and/or date.
• Served Alarm Calls - This log will display Alarm Calls which have already been served.
You can check Alarms set (pending), served and unanswered for last 25 hours. Altogether maximum 500 entries
will be displayed. Each Alarm Call will display the details of time (hours and minutes), date and type (once only,
daily).
The LED of the DSS key assigned to Wakeup Call Log glows in Red to indicate Unanswered Alarm calls or it glows
in Blue to indicate Pending Alarm calls.
If there are both, Unanswered and Pending Alarm calls, the LED of the DSS key will glow in Red. After you view the
Unanswered Alarm Calls, the LED will glow in Blue to indicate Pending Alarm Calls. The LED will glow in Blue till all
the Pending Alarm calls have been served. If any Daily Alarm has been set the LED of the DSS key will glow in
Blue till the alarm is canceled.
To view the log from any DKP / Extended IP Phone user,
• Press the DSS Key assigned to Wakeup Call Log.
• The phone displays the logs — Unanswered Alarm Calls, Pending Alarm Calls, Served Alarm Calls.
• Select the desired log to view the details in the respective log.
• The acknowledged call is removed from the Alarms Log and is logged into the System Activity Log with the
details of the extension that acknowledged the call.
205. The logs will also display the Reminders.

• SARVAM UCS can register as many as 960 Alarm requests set by the Operator and extension users.
• Multiple Alarms can be set for an extension by the Operator and/or by the extension user. For example,
Daily Alarm at 09:00am is set for an extension. The extension user wants to change the alarm time to
08:30am for a day. The extension user/Operator can set another alarm, that is, a Once Only Alarm, at
08:30am without disturbing the daily alarm. Both the Alarms will ring at the set time.
• When multiple alarm requests have been set on an extension, if the Operator/extension user cancels
an alarm set for an extension, the system cancels all alarms set for the extension. It is not possible to
cancel any of these alarms selectively.
• It is not possible to modify an alarm request. Instead, the alarm request should be canceled and a new
one should be made.
• The duration of Alarm Ring Timer, the Number of Alarm Attempts and the Alarm Attempt Interval are
programmable.
• Alarms can be set for all extensions of the system, including the Operator phone also.
• All the Alarm events are logged in the "Hotel-Motel Activity Log".
• Alarm settings will be retained in the system during power down and system upgrades.
How to configure
The following parameters play an important role in the functioning of the Alarm feature. These parameters carry
default values. The default values have been selected keeping the larger user base in mind. However, these values
can be changed by the System Engineer at the time of installation or afterwards as per users' requirements.
1. Alarm Ring Timer - The duration for which the system rings the extension to serve an Alarm call. By
default, the Alarm Ring Timer is set to 45 seconds. This timer can be set between 001 to 255 seconds.
This timer also signifies the duration for which the Operator phone rings to notify that an Alarm call has not
been answered or the extension phone is busy.
2. Number of Alarm Attempts - Number of times the system attempts to place an Alarm call on the
extension phone before notifying the Operator that the call is not answered or the phone is busy. By
default, the Number of Alarm Attempts is set to '3'. The Number of Alarm Attempts can be set between 1
and 9.
3. Alarm Attempt Interval - The time period between each Alarm Call attempt. By default, the Alarm Attempt
Interval is set to 5 minutes. The Alarm Attempt Interval can be set between 1 and 9.
4. Use Alarm with Snooze - Snooze is a functionality which forces the extension user to acknowledge the
Alarm call. With snooze enabled, the system expects the user to answer the Alarm call by going OFF-Hook
and dial Acknowledgement code '0'. With snooze disabled, the system considers the Alarm as answered
when the extension user simply answers the alarm call by going OFF-Hook (dialing acknowledgement
code is not mandatory). Users may choose whether or not to enable snooze. By default, snooze is
disabled.
5. Configurable Alarm Type - When the Operator and extension user set an Alarm call request, the system
gives them the choice of setting 'Once Only' or 'Daily' Alarm calls.

User experience however, shows that 'Once Only' Alarm call requests are more common than 'Daily' Alarm
requests. So, the system allows you the flexibility of setting 'Once Only' as the default Alarm Type, by
disabling the 'Configuring Alarm Type' flag.
When this flag is disabled the system will prompt the Operator/Extension user to enter the Time of the
Alarm call and consider the Alarm Type as 'Once Only'.
By default, this flag is disabled.
6. Configurable Alarm Category - When the Operator sets an Alarm call for an extension, the system
prompts the Operator to select an Alarm Type (Once Only or Daily) and to select the alarm serving
mechanism - 'Automated or Personalized'.
If the Enterprise wishes to offer only 'Automated' Alarms to its extension users, the system allows the
flexibility to set 'Automated' as the default Alarm call serving mechanism. This can be done by disabling
the 'Configurable Alarm Category' flag.
When this flag is disabled, the system will consider the Alarm call serving mechanism as 'Automated' and
will prompt the Operator only for the Time of the Alarm call.
By default, this flag is disabled.
• When both flags 'Configurable Alarm Type' and 'Configurable Alarm Category' are disabled, the system
will set and serve 'Once Only - Automated' alarms only.
• If the 'Configurable Alarm Type' flag is disabled, but the 'Configurable Alarm Category' flag is enabled,
the system will set 'Once Only' alarm calls, but give the option of selecting 'Automated' or
'Personalized' as the serving mechanism.
• Similarly, if 'Configurable Alarm Type' is enabled, but the 'Configurable Alarm Category' flag is disabled,
the system will allow both 'Once Only' and 'Daily' alarms to be set, but the serving mechanism will be
'Automated'.
7. Voice Guided Alarm Verification: For Voice-guided Alarms, the VMS of the system allows you to enable/
disable the Alarm Verification for alarms and reminders, allowing extension users who want to use alarms
and reminders to confirm the Time set for an alarm and Date and time set as a reminder. By default, this
flag is enabled.
The flags ‘Configurable Alarm Type’ and ‘Configurable Alarm Category’ are not applicable for Voice-guided
Alarms. In the case of Voice-guided Alarms, the Operator/Extension user will be prompted to select the
Alarm type and serving mechanism, each time, even when both aforementioned flags are disabled.
8. Alarm Notification Type - This is the means of notifying the extension user about the Alarm call. The
extension user can be played Music-On-Hold, Live Music, Pre-recorded Voice Message, Weather
information, Date and Time, etc. The system supports four types of Alarm Notifications:
• Voice Message: Selecting this option would play a recorded message recorded to the extension user
when s/he answers the Alarm call.
• Music-On-Hold: Selecting this option would play music-on-hold to the extension user when s/he
answers the Alarm call.

• Routing Group: Selecting this option would connect the extension user to the stations programmed in
the Alarm Notification Group. The System Engineer may connect a device which can play customized
alarm greetings with date, time, weather conditions, traffic conditions, a marketing message, etc. on the
stations programmed in the Alarm Notification Group.
If Voice Mail Auto Attendant is selected as a Routing Group member, the system will place the call on the
Voice Mail System.
• Voice Mail: Selecting this option would connect the extension user to the Voice mail System. Use this
option only if you have the VMS module installed in the system.
9. Macros - This is a short code for simulating the Alarm call. The SLTs with special function keys send a
fixed string to the system, when each function key is pressed. The system interprets this string and
translates it into a string that can be understood by the system. For example, the SLT has a special
function key for Alarm calls which sends the string '51' to the system. The system can be programmed to
translate '51' into the feature access code for Alarm calls, '*161'.
All the above listed parameters can be programmed using Jeeves and a Telephone.
Configuring Alarm Parameters using Jeeves

• To configure Alarm parameters, go to “System Parameters”.
• To view the Alarm Report Status on your phone LCD, you must assign a DSS Key to Wakeup Call Log.
Refer the topic “DSS Keys Programming”, “DSS Key Settings” in “Configuring Matrix SPARSH VP330”,
“DSS Key Settings” in “Configuring Matrix SPARSH VP248”, “DSS Key Settings” in “Configuring Matrix
SPARSH VP310” and “DSS Key Settings” in “Configuring Matrix SPARSH VP510” for instructions.
• To select Alarm Notification Type for extensions, under Configuring Extensions, see“Station Advanced
Feature Template”.
• If you select Voice Message as Alarm Notification type, ensure that you assign a voice module to
'Alarm' voice message application. Please refer topic “Voice Message Applications” for more details.
• If you select Music-On-Hold as Alarm Notification type, no further configuration is required.
• If you select Voice Mail as the Alarm Notification Type, make sure you have installed the Voice Mail
module.
• If you have selected Routing Group as Alarm Notification Type, you must create a Routing Group and
assign this Routing Group number in the Station Advanced Feature Template of the extensions. See
“Routing Group” and “Station Advanced Feature Template” for instructions on applying the template to
SLTs, DKPs, ISDN Terminals, Virtual Extensions.
• To program SLTs with special Alarm function key and to create macro for a DKP key, see “Macros”.
• To use a customized alarm messaging device, see “Configuring Customized Alarm Messaging Devices”

Configuring the Alarm Parameters using a Telephone
To configure Alarm Ring Timer, dial:

• 2201-Seconds
Where,
Seconds is from 001 to 255. Default: 45 seconds.
To configure Number of Alarm Attempts, dial:

• 2202-Number of Alarm Attempts
Where,
Number of Alarm Attempts is from 1 to 9. Default: 3.
To configure Alarm Attempt Interval, dial:

• 2203-Alarm Attempt Interval
Where,
Alarm Attempt Interval is from 1 to 9 minutes. Default: 5 minutes.
To configure Snooze function, dial:

• 2204-Snooze
Where,
Snooze is 0 or 1.
Select '0' to disable Snooze and '1' to enable snooze.
To disable/enable Configurable Alarm Type, dial:

• 2208-Flag
Where,
Flag is 0 for Disable, 1 for Enable1. Default: Enabled.
To disable/enable Configurable Alarm Category, dial:

• 2209-Flag
Where,
Flag is 0 for Disable, 1 for Enable. Default: Enabled
To configure Alarm Notification Type, dial:

• 5602-1-Template Number-12-Alarm Notification Type
Where,
Template Number is Station Advanced Feature Template from 01 to 50.
Default: 01.
Alarm Notification Type is:
1 for Music on Hold
2 for Voice Message (Voice Modules)
3 for Routing Group
4 for Voice Mail
For example: To program Routing Group as Notification Type in Station Advanced Feature Template
02: Dial 5602-1-02 -12-3
To apply the Station Advanced Feature Template now programmed with the Alarm Notification Type to
extension phones, refer the topic “Customizing Station Advanced Feature Template using a
Telephone”.
To program Macros, dial the following commands:

• Dial command 1810-Macro Index-Number String to create a macro.

Where,
Macro Index is from 01 to 25 (as 25 macros can be created)
Number String is the feature access code for Alarm, in this case '*163'
Terminate the command with #* as the number string is less than 24 digits.
For example: to create a macro for Alarm on Index 2: dial 1810-02-*163-#*
• Dial command 1810-Macro Index-#* to clear a macro.
• Dial command 3115-1-Macro Index-Access Code to program Access code (that is, number sent to
the system by the SLT)
Where,
Macro Index is from 01 to 25 Access Code is a string of 4-digits.
If the length of the Access code is less than 4-digits terminate the command with #*
For example: To program access code '53' for the macro for Alarm dial: 3115-1-02-53-#*
• Dial command 3115-1-Macro Index to clear the Access code for the macro.
• Exit SE mode.
Configuring Customized Alarm Messaging Devices

Alarm Messaging devices play real-time updated information like the date and time, greetings, weather information,
Road highway status, specific event announcements, etc. when called upon. These devices should be connected
to the system on its SLT ports.
If the User wants to connect customized alarm messaging devices to the SARVAM UCS, the SE should configure
the system as instructed below:
• Connect the devices for customized alarm greetings to SLT ports only.
• Under Configuration, click Routing Group to open the page.

• Select the default Routing Group 31 used for Alarm Notification Group, or any other Routing Group
number.
• Select SLT as the Member Type and enter the SLT Port Number where the device is connected. It is
possible to configure 32 members in a single routing group.
If only one device is connected, disable all other members from 02-32 by setting Member Type to None.
• Open the Station Advanced Feature Template page. By default Template Number 01 is applied to all
stations. The template has Voice Message as default notification type. It is recommended that you
program another Template.
• Select Routing Group as the Alarm Notification Type in the template you have selected for configuration.

• Enter the number of the Alarm Notification Routing Group (default group: 31) in which you have
programmed the device (SLT port).
• Apply the Advanced Feature Template now configured with Routing Group as Alarm Notification Type and
the number of the Alarm Notification Routing Group to the stations.
• Refer the section “Station Advanced Feature Template” for instructions on applying this template to
extension phones.
Viewing and Printing Alarm Reports

The Operator can view the status of Alarms that are yet to be served from the System Administrator mode and if
required print them.
• Click the Reports link.
• Under Reports, click Wakeup Alarm.
• The unserved Alarm calls will appear on your screen.
• You can cancel any of the unserved Alarm calls by selecting the check-box and clicking the Cancel
Selected Alarms button on this page.
• You can also print this page by clicking the Print button on this page.
• Click Close to exit the page.

How to use
Alarms can be set by the extension users by themselves. The extension users can also ask the Operator to set the
alarm for them.
Alarms set/canceled by Operator

The Operator can set/cancel non-voice guided Alarms using EON, SLT and from System Administrator mode.
Operator using EON

Using DSS Key:
To set Alarm for the extension user,
• Press the key assigned the 'Remote Alarm' function.

• Enter the Extension Number.
• Enter Time in HH:MM
• Select 'Once Only' or 'Daily'.
• Select 'Personalized' or 'Automated'.
• Press 'Enter' key to set Alarm.
• You get a confirmation tone and a text message with the phone number for which the alarm is set.
• Go Idle or you get dial tone after 3 seconds.
To cancel Alarms,
• Press key assigned the 'Remote Alarm' function.

• Enter Extension Number.
• Select 'Cancel All'.
• Press 'Enter' Key.
Using Commands:
To set Alarm for the extension user,

• Dial 1072-003.
• Enter Extension number.
• Dial 1 for Once Only or Dial 2 for Daily
• Dial 1 for Personalized or Dial 2 for Automated.
• Press 'Enter' key to set Alarm.
• You get a confirmation tone and a text message with the phone number for which the alarm is set.
• Replace handset or you get dial tone after 3 seconds.
To cancel Alarms,

• Dial 1072-003.
• Dial #.
• You get a confirmation tone and a text message with the phone number for which the alarm is canceled.

Using Jeeves:
• Click the Extensions link.
• Search the extension on which you want to set the alarm by the Extension Number or the Extension
Name.
• The page of the Extension will open.
• Now set the desired type of alarm on this extension.

• Repeat the same to set alarm on another extension number.
• Click Logout to exit the System Administrator mode.
Alarms set/cancel by Extension Users
Extension Users using EON / Extended IP Phone

If the extension user has EON, alarms can be set using the DSS key as well as by dialing the command.
Using DSS Key:
To set Alarm,
• Press the key assigned the 'Alarm' function.

• Select 'Once Only' or 'Daily'.
• You get a confirmatory text message and confirmation tone.
To cancel Alarms,
• Press the key assigned the 'Alarm' function.

OR
• Dial 161
Dialing Commands:
To set Alarm,

• Dial 161.
• Enter Time in HH:MM (24-hours format)
• Dial 1 for Once Only or Dial 2 for Daily.
• Replace handset or you get dial tone after 3 seconds
To cancel Alarms,

• Dial 161.
• Dial #.

Extension Users using SLT
To set Alarms,

• Dial 161.
• Dial HH:MM
• Dial 1 for Once Only or Dial 2 for Daily.
• Replace the handset on the cradle.
To cancel Alarms,

• Dial 161.
• Dial #.
• Replace the handset.
• Extension users can set only automated alarms from their phones. For personalized alarms, they must
request the Operator.
• If there are multiple alarms set, alarms cannot be canceled selectively. Only the Operator can cancel
alarms selectively from SA mode.
• Alarms set on an extension will be served, even if DND is also set on the same extension.
Printing Alarm Reports

When Scheduled Alarm Report is enabled and the time is set, the system will automatically print the report at the
set time. The Operator can also print Alarm Status Report any time s/he using SA commands.
The Operator can issue SA commands from EON or an SLT to print Alarm Status Reports.

• Press the key assigned the 'Print Alarm Report' function (if programmed).
OR
• Dial 1072-913.
• You get a confirmatory text message and a confirmation tone.
• Go idle.
For SLT Users

• Dial 1072-913.
• Replace the Handset on the cradle.

Wakeup/Alarm Report AS ON 05-05-2016(Thu) AT 23:49
-------------------------------------------------------------------------------
Room# Phone# Wakeup D P Room# Phone# Wakeup D P
-------------------------------------------------------------------------------
3001 12:18 2017 14:12
3001 12:21 2017 12:14
-------------------------------------------------------------------------------
* indicates Daily Alarm and + indicates Personal Alarm Page : 1
---End of Report---

Alternate Number Dialing
What's this?
Alternate Number Dialing allows you to dial different phone numbers in an attempt to reach a person whose line is
busy.
Alternate Number Dialing is useful when the person or organization you are trying to reach has more than one
number, where they may be reached. The system dials out different phone numbers of the same party, saving you
time and effort of dialing each of these numbers manually.
How it works
This feature works as an extension of the features “Last Number Redial” and “Auto Redial”. It requires you to
program Alternate Number Groups in the Global Directory first. With the alternate numbers programmed in the
Global Directory, all you need to do is to use Last Number Redial or Auto Redial, every time you want the system to
try Alternate Number Dialing.
For example: Midas Business Solutions has four telephone numbers: 2640459, 2631235, 2635589 and 2565590.
To be able to use Alternate Number Dialing, you must first program all four numbers as Alternate Number Group in
the Global Directory.
Now, when you dial one of these numbers, '2640459', and get a busy tone, you can either initiate Last Number
Redial or set an Auto Redial request.
When you initiate Last Number Redial,
• The system will dial an alternative number for the dialed number.
• If the redialed number is busy, you can set Last Number Redial again.
• The system will dial a second alternative number.
• If the second alternative number is also busy, you can set Last Number Redial again.
• The system will dial a third alternative number.
• This process will be repeated each time you set Last Number Redial, until the call gets through.
When you set an Auto Redial request on busy tone,
• The system will dial an alternative number.
• If the alternative number is busy, the system will redial another alternative number.
• The system will dial a different (alternative) number on each auto redial attempt206, until the call gets
through.
206. The number of auto redial attempts depends on the Auto Redial Count programmed in the system. By default, the system will
make 5 redial attempts if Auto Redial 'normal' is set. If Auto Redial 'Priority' is set, the system will make 20 redial attempts.

• (for the number of redial attempts programmed), until the call gets through.
• when any of the alternate numbers gets through, the system will give a ring on your extension.
This process is illustrated in the graphic below:
(Busy) 2630555
(Bu
sy)
Calling Party
2630556
2630557
SARVAM UCS
• Alternate Number Dialing will work only on extensions that are allowed the features “Last Number
Redial” in their “Class of Service (COS)”
• Also, Alternate Number Dialing will work only for those numbers that exist in the Global Directory
assigned to each extension. The Global Directory is divided into three parts, 100-399 (Part 1), 400-699
(Part 2), and 700-999 (Part 3). If an extension is assigned only Global Directory Part 2, Alternate
Number Dialing will work only for those numbers grouped as Alternate Number Groups in Global
Directory Part 2.
• Alternate Number Dialing will work also with “Abbreviated Dialing”. For example, an extension user
dials the abbreviated code 8100, and the dialed out number is busy. When the extension user sets
Redial or Auto Redial, the system will try the alternate numbers related to 8100.
How to configure
For Alternate Number Dialing to work, the System Engineer must:
1. Make a List of Alternate Numbers.
2. Create Alternate Number Groups.
3. Program Alternate Number Groups in the Global Directory.
4. Enable the features 'Last Number Redial', 'Global Directory', in the Class of Service (CoS) group of the
extensions to which Alternate Number Dialing facility is to be provided. If desired, 'Auto Redial', 'Auto
Redial Priority' may also be enabled in the CoS of these extensions.
All of the above parameters can be programmed using Jeeves or dialing SE Commands from a telephone.
• To create Alternate Number Groups, the alternate numbers must exist in the Global Directory. If any of
the alternate numbers do not exist in the Global Directory, first program the numbers in the directory,
before you begin creating Alternate Number Groups. Refer the topic “Abbreviated Dialing” for
instructions on programming the Global Directory.

• As Alternate Number Dialing works only for the Alternate Number Groups in the Global Directory
assigned to each extension, ensure that the relevant Global Directory with the Alternate Number
Groups is allowed in the CoS of the extensions.
Preparing Alternate Number List

In consultation with the User, you may:
• Draw a two-column table on a paper.
• Write the name of the contact on one column and the Alternate Numbers for the contact on the other
column.
• Make a list of the numbers which need to be grouped as alternate numbers. For example:
Name of the Contact Alternate Numbers
Midas Business Solutions 2640459, 2631235, 2635589, 2565590
Jet Set Holidays 022281110001, 022281110002
Bacchus Vineyard 2640075, 2640076
GoodLife Inn 2788856, 2788896
You may include as many alternate numbers as required by the User.
Creating Alternate Number Groups

• Assign the alternate numbers of each of contacts to an 'Alternate Number Group'.
• Each group must be assigned a number between 000 to 255.
• Taking the above example further, the Alternate Number Groups on the list may be numbered as follows:
Name of the Contact Alternate Numbers Alternate Number Group (No.)
Midas Business Solutions 2640459, 2631235, 2635589, 2565590 001
Jet Set Holidays 022281110001, 022281110002 002
Bacchus Vineyard 2640075, 2640076 003
GoodLife Inn 2788856, 2788896 004
Programming Alternate Number Groups in Global Directory using Jeeves

Make sure that the numbers on the list are also programmed in the Global Directory. If any of these numbers do not
exist in the Global directory already, program them in the Global Directory first. Refer the topic “Abbreviated Dialing”
for instructions on programming the Global Directory.
To create Alternate Number Groups and program them in the Global Directory,

• Click Global Directory to open the page.
• Click the Advance button on the bottom of this page.
• Enter the number of the Alternate Number Group in the last column of the page.
For example, you have assigned Alternate Number Group '001' to all the numbers of the contact Midas
Business Solutions, enter this number against each number belonging to this contact.
Similarly, enter Alternate Group number '004' against the numbers belonging to the 'GoodLife Inn' to which
it is assigned.
Memory Outgoing Trunk Alternate

Number Name
Location Bundle Group Number Group
100 01 2640459 Midas Biz 001

Memory Outgoing Trunk Alternate
Number Name
Location Bundle Group Number Group
101 01 2631235 Midas Biz 001
102 01 2635589 Midas Biz 001
103 01 2565590 Midas Biz 001
104 01 2788856 GoodLife Inn 004
105 01 022281110001 Jet Set 002
106 01 022281110002 Jet Set 002
107 01 033298765432 R. Mendez 000
108 01 2640075 Bacchus 003
109 01 2640076 Bacchus 003
: : : : :
129 01 2788896 GoodLife Inn 004
The numbers of the contacts may not necessarily appear alphabetically or in a sequence. It is possible that
the numbers of the same contact may be programmed at different memory locations in the Global
Directory.
In the above example, one number of the GoodLife Inn is programmed at memory location Index 104 and
the other on Index 129. Since these two numbers are grouped and assigned the number alternate group
number '004', this number must be entered against the GoodLife Inn numbers at the respective memory
location Index.
• After assigning Alternate Number Groups, click Submit at the bottom of the page to save changes.
• Enable the features Last Number Redial and Global Directory, in the Class of Service (CoS) group of
the extensions to which Alternate Number Dialing facility is to be provided. If desired, Auto Redial, Auto
Redial Priority may also be enabled in the CoS of these extensions.

By default Station Basic Feature Template Number 01 is assigned to all extensions of the system. The
default CoS Group 01 in this template has 'Redial' enabled in the set of 'Basic Features', so all extensions
of the system can use Last Number Redial.
However, the default CoS Group 01 has only Global Directory Part 1 enabled.
Recall that Alternate Number Dialing will work only for those numbers that exist in the Global Directory
assigned to each extension. So, the Global Directory Part containing the Alternate Number Groups must
be allowed to the extensions in their Class of Service. For example, if Alternate Number Groups are
programmed in Global Directory Part 2, extensions must have Global Directory Part 2 in their Class of
Service.
If all extensions are to be allowed the Alternate Number Dialing facility, simply enable the Global
Directories containing Alternate Number groups in the default CoS group 01.
However, if Alternate Number Dialing is to be allowed to select extensions only, define a new CoS group
and prepare a new Station Basic Feature Template with this CoS group and apply it to the desired
extensions.
The Station Basic Feature Template 01 does not have the features Auto Redial and Auto Redial Priority in
the default CoS group 01. If these features are also to be allowed to the extensions, enable them in the
CoS you prepare.
Programming Alternate Number Groups in Global Directory using a Telephone

To assign Alternate Group Numbers, dial:

• 1804-1-Memory Location Code-Alternate Group Number
Where,
Memory Location Code = 100 to 999. This is the Index number, where the number and name of the
contact is programmed.
Alternate Group Number = 000 to 255.
For example: To assign the numbers of the GoodLife Inn to an Alternate Number Group, dial:
1804-1-104-004
The number '2788856' stored at Index 104 will be assigned to Alternate Number
Group 004, which is the number assigned to GoodLife Inn.
Similarly, to program the second number dial: 1804-1-129-004
The number '2788896' stored at Index 129 will be assigned to Alternate Number Group 004.
If you have a continuous sequence of numbers that need to be programmed in the same group, you may
dial:
• 1804-2-Memory Location Code-Memory Location Code-Alternate Group Number
Here you enter a sequence of Memory Location Codes, from 100 to 999, and the number of the
Alternate Number Group.
For example: To assign the numbers of Midas Biz to an Alternate Number Group, dial: 1804-2-100-
103-001

The numbers stored at Index 100 to 103 will be assigned to Alternate Number Group 001, which is the
number assigned to Midas Biz.
To clear the Alternate Number Groups, dial the following commands:

• 1804-1-Memory Location Code-000 to clear group of a single number.
• 1804-2-Memory Location Code-Memory Location Code-000 to clear group of a continuous
sequence of numbers.
• 1804-*-000 to clear all Alternate Number Groups
For example: To clear the Alternate Number Group assigned to a number of GoodLife Inn, dial: 1804-1-
104-000
The Alternate Number Group assigned to the GoodLife Inn number '2788856' stored at Index 104 will
be cleared.
For example: To clear the Alternate Number Group assigned to the sequence of numbers of Midas Biz,
dial: 1804-2-100-103-000
The Alternate Number Group assigned to the Midas Biz numbers stored in a continuous sequence
starting from Index 100 to 103 will be cleared.
• Exit SE mode.
How to use
Confirm with your System Engineer that
• Alternate Number Groups are programmed in the Global Directory allowed to your extension.
• 'Basic Features' (these include Redial) are enabled in the Class of Service allowed to your extension.
Now, follow the instructions for using the feature “Last Number Redial”.

Authority Codes
What's this?
Authority Code is a unique password-protected code with an associated Class of Service, Toll Control and Call
Budget, which can be assigned to extension users. With Authority Codes, extension users of the system can make
calls or access features from any other extension of the system as per the Class of Service, Toll Control and other
features / facilities assigned to their code.
This feature is useful when you want a group to extension users to use a single extension, but at the same time you
want to keep an account of the calls made by each user. If required, you can assign a call budget to each Authority
code to control call cost.
To make outgoing calls or access features, extension users must 'Walk-In' from any extension port: DKP, SLT, SIP,
ISDN Terminal, E&M (with Station as Orientation Type), and then dial their Authority Code and Password.
How it works
An 'Authority Code' is a unique three-digit number, protected by a four-digit password. The default Authority
Password is 1111. To be able to use an Authority Code, the password must be changed to another value. To avoid
unauthorized access, we recommend you to change the password regularly. Make sure it is strong and is kept
confidential.
The SARVAM UCS supports as many as 999 Authority Codes.
Each Authority Code has an associated Class of Service and Toll Control, which is configured in the Station Basic
Feature Template assigned to the Code.
To make calls using an Authority Code,
User A is assigned 222 as Authority Code and is provided a unique 4-digit Authority Password.
To access features or make calls as per the assigned Authority Code 222, User A must do the following:
• Dial the feature code for Walk-In Class of Service (default 111) from any extension of the system.
• Dial the code for Walk-In by Authority followed by the Authority Code and then the Authority Password.
To make calls or access features according to the Authority Code dial 111, the feature code for 'Walk-In
Class of Service' from any extension of the system.
• Select the option ‘Walk in by Authority Code’ or dial ‘2’.
• A dials the Authority Code assigned to him/her followed by the Password.
If the user enters a wrong password, to prevent any attempt to misuse the Authority Code, by default, the
system allows the user only three attempts to re-enter the password. If the user fails to enter the correct
password at the third attempt, the system will set the password to default, 1111. With the default password
the user will not be able to use the Authority Code.

The number of attempts to re-enter can be configured by setting the parameter Retry counts for
Authority Code password in the “System Timers and Counts”.
• User A can now make outgoing calls.
• If the Walk Out mode set for A is One Call A will automatically be logged out from the extension after one
call.
• If the Walk Out mode set for A is Multiple Calls, A can make as many calls as desired, and remains
'walked-in' until the A dials the feature code to 'Walk-Out', or until another extension user walks into the
same extension.
• If a call budget has been assigned to A’s Authority Code, A will be able to make calls till the assigned
amount is consumed.
• Details of the calls made by A are recorded by the Authority Code in the Station Message Detail Recording
Report (SMDR) for Outgoing Calls.
• The SMDR report can be printed using the Authority Code as filter.
This way, the organization can know the details of calls made by each user as well as have control over the
expenses.
You can also use Authority Codes from a remote location using DISA. To do so, you must make a DISA call. Then
you must ‘Walk In’ and dial your Authority Code and Password. To know more, see “Direct Inward System Access
(DISA)”.
How to configure
For Authority Codes to work, you must:
• Configure the Authority Code Table.
• Assign a Station Basic Feature Template to the Authority Code, with the desired Class of Service, Toll
Control and features/facilities.
• Enable the parameter Store Outgoing Calls in the Station Basic Feature Template assigned to the
Authority Code, if you want details of outgoing calls made using Authority Codes.
• Assign a Station Advanced Feature Template to the Authority Code with the desired Walk-Out Mode
• Assign each user a Call Budget, if required.
• Change the Retry Counts for the Authority Code, if required.
Configuring Authority Code Table

• Make a list of users you want to assign Authority Codes to and their respective passwords.

• Under Configuration, click Authority Code.
Against each Authority Code,
• Assign an Authority Password. The password can be a minimum of 4 digits and a maximum of upto 10
digits. Valid digits are 0 to 9, * and #. Default: 1111.
• Enter the Name of the user. The name acts as an identifier. Default: Blank.
• By default, Station Basic Feature Template number 41 and Station Advanced Feature Template
number 41 are assigned to all Authority codes.
If you want to change the calling permission, allow/deny features to users, you must customize the Class
of Service, Toll Control and other features/facilties in the Station Basic Feature Template according to your
requirement.
You may set the 'Walk-Out Mode’ in as One Call or Multiple Calls in the Advanced Feature Template.
Refer to “Station Basic Feature Template” and “Station Advanced Feature Template” for detailed
instructions.
You can assign/change the Name and Password from the SA mode also. To do this,

• Click Authority Code.
Against each Authority Code,
• Assign/change Authority Password. The password can be a minimum of 4 digits and a maximum of upto
10 digits. Valid digits are 0 to 9, * and #. Default: 1111.
• Enter the Name of the user against the authority code you have assigned to them. The name acts as an
identifier. Default: Blank.
Assigning a Call Budget to Authority Codes

Call Budget is a cost control feature that allows you to keep a tab on the total cost of each user. With this each user
can be allotted a 'budget' limit for outgoing calls, which is automatically reloaded at the start of every month.

To assign a Call Budget to the Authority Code,
• Under Reports, click Call Budget.
• Click the Authority Code tab.
For each Authority Code assigned to the user,
• In the Allot an Amount column, enter the amount you want to assign to the user as budget limit for
outgoing calls. The amount you allot here will be displayed as Allotted Amount.
If you are re-assigning a new amount before the previous balance is consumed, make sure you add the
available balance to the new amount. Enter this amount in Allot an Amount.
For example, you have allotted an amount of Rs.1000 and the consumed amount is Rs.600. The available
balance is Rs.400. Now, if you want to assign a new amount of Rs.500. In Allot an Amount you must enter
900 (Available Balance + New = 400 + 500).
• The Allotted Amount column displays the amount allotted to the user for making outgoing calls.
• The Consumed Amount column displays the call budget amount consumed by the user.

Setting the Retry Counts for Authority Code
By default, the system allows three attempts to re-enter the Authority Code Password. If you want to increase or
decrease the number of attempts, set the parameter Retry Counts for Authority Code to the desired value. For
instructions, see “System Timers and Counts”.
How to use
To use Authority Code from any extension:
• Go OFF-Hook.
• Press DSS Key assigned to ‘Walk-in COS’.
OR
• Dial 111
• Select the ‘Walk-in by Authority Code’ option and press the Enter key.
• Dial the Authority Code, followed by the Password.
• You will hear the confirmation tone, followed by dial tone.
• Dial the desired number.
To use Authority Code from a remote location:
• Dial the number of a DISA enabled Trunk.

• Dial DISA Login Code 1079-Extension Number-User Password, if PIN Authentication is required.
• After DISA Login beeps, dial 111.
• Press 2 to select ‘Walk-in by Authority Code.
• Dial the Authority Code followed by the Password.
• You will hear the Confirmation tone, followed by dial tone.
For SLT Users

• Go OFF-Hook.
• Dial 111
• Press 2 to select ‘Walk-in by Authority Code
• Dial the Authority Code, followed by the Password.
• You will hear the Confirmation tone, followed by dial tone.
Printing Reports of Outgoing Calls made using Authority Codes

You can print call details of users who made outgoing calls using Authority Codes. For this, you will need to:
• enable Store Outgoing Calls in the Station Basic Feature Template of the extension user.
• set the Calls made using Authority Code filter in Outgoing Call Report.
• configure the destination port for SMDR-Outgoing Call Report.
Refer the section “Station Message Detail Recording-Report”, for detailed instructions on printing reports using
filters.

Apple Push Notification Service Support
What's this?
Apple Push Notification Service (commonly referred to as Apple Notification Service or APNs) is a platform
notification service created by Apple Inc. that enables third party application developers to send notification data to
their applications installed on Apple devices.
Previously, VoIP applications needed to maintain a persistent connection in order to receive calls. Keeping a
connection open in the background, drains the battery as well as causes all kinds of problems when the application
crashes or is terminated by users.
In iOS 8 Apple has introduced PushKit as part of their effort to improve battery life, performance and stability for
VoIP applications such as Skype, WhatsApp, etc. PushKit offers high-priority push notification with a large payload.
The VoIP application receives the notification in the background, sets up the connection and displays a local
notification to the user.
SARVAM UCS supports PushKit for VARTA AMP100 Application only. Push Notifications will be sent for calls, new
messages as well as for voicemail. Push Notifications will be sent to the MATRIX VARTA AMP100 Application only
if it is in the background and when there is persistent internet connection. When the application user exits the
application Push Notifications will not be served.
How it works
Pre-requisites for Push Notifications:
• Make sure that the server has a persistent internet connection.
• Make sure the Date and Time of the server is synchronized with the NTP Server.
• To receive IM and IM notifications make sure the application is registered at Location 1. For more details,
refer “Configuring Matrix VARTA ADR100/AMP100 UC Clients”
Let us see how the notifications will be sent by the server when MATRIX VARTA AMP100 application is registered
with the server as a SIP Extension and it is in the background. There is an incoming call or message:
• You can check the status of the SIP Extension user. It will display Registered and under the respective
Contact 1, 2, 3 it will display Active (as the device is in the background) along with the time remaining for
the expiry of the Device Inactivity Timer.The default value of the Device Inactivity Timer is 2 days.
• The server will send a Push Notification to the MATRIX VARTA AMP100 application (client).
• The server will wait for 15 seconds after sending the Push Notification:
• if the client registers with the server within this time, the call will be connected or the message will be
delivered. The status of the SIP Extension will display Registered and under the respective Contact 1,
2, 3 it will display the SIP ID, IP Address and the Registration Expiry Timer.
• if the client does not register with the server within this time, the call will be disconnected or the
message will be rejected.The status of the SIP Extension will display Registered and under the
respective Contact 1, 2, 3 it will display Active and the time remaining for the expiry of the Device
Inactivity Timer.

• The server maintains an inbuilt timer, Device Inactivity Timer which is set as 2 days. Till the expiry of
the timer the server will send Push Notifications to the application.
• If for this duration, the server does not receive any registration request from the application and the
timer expires, the server will consider the application as unregistered and will stop all Push Notifications
to the application.The status of the SIP Extension will display Not Registered and under the respective
Contact 1, 2, 3 the details will be cleared. Calls and messages will be rejected.
The server will start sending notifications to the application in the background after the application is
bought in the foreground once and a registration request is received by the server.
Feature Interactions when the Application is in the Background

Call Forward when not Registered:
If the application receives an incoming call from a QSIG caller, the Call Forward functionality will not be applicable.
If the application user is a member of any Routing Group or Department Group, the Call Forward functionality will
not be applicable.
To know more, see “Call Forward-When Not Registered”.
Handover and Handoff:
VARTA AMP100 users will be able to use Handover but Handoff will not be possible. To know more, see “Handover
and Handoff”
System Restart
After System Restart the Device Inactivity Timer will be reset.
Redundancy:
Push Notification will not be supported in case of Redundancy.

Auto Attendant
What’s this?
Auto Attendant allows external callers to reach an extension directly without the intervention of the Operator.
If Auto Attendant is enabled on a trunk, whenever an external call lands on that trunk, the Built-In Auto Attendant or
the Voice Mail Auto Attendant (if VMS module is installed) of SARVAM UCS greets the caller and prompts the
caller to dial the desired extension number. The call is then placed to the extension number dialed by the caller.
SARVAM UCS offers Delayed Auto Attendant, whereby incoming calls routed to the Operator or the Trunk Landing
Group, can be answered by the Built-In Auto Attendant or the Voice Mail Auto Attendant, if none of the landing
extensions answers the call within a certain time period.
Regular callers who know the extension numbers, can use the Auto Attendant to reach the desired extensions
without Operator assistance. Thus, this reduces call traffic on the Operator extension, saves callers the time for call
set-up and transfer. The Auto Attendant is particularly useful during non-working hours and holidays, and it helps
project a professional image of the organization.
Built-In Auto Attendant will not work, when the dialed extension has Privacy from Built-In Auto
Attendant enabled in its Class of Service. So, if you want to prevent external callers from accessing
certain extensions, you must enable Privacy from Built-In Auto Attendant in their Class of Service. To
know more, see “Privacy”.
How it works
Auto Attendant can be configured on all trunk types, for the three time zones (working hours, break hours and non-
working hours).
When configuring Auto Attendant on a trunk, you may choose to have calls answered by the Built-In Auto Attendant
or the Voice Mail Auto Attendant.

When the Built-In Auto Attendant of SARVAM UCS is selected as the destination for incoming calls on a trunk, this
is how it will work:
• A call lands on a Trunk.
• The system waits for the period of the Built-In Auto Attendant Answer Wait Timer (default: 05 seconds) to
answer the call during this period. The caller gets Ring Back Tone from the CO.
• The system greets the caller with the pre-recorded Time-based Greetings. These are: Built-In Auto
Attendant— Morning Greetings/Afternoon Greeting/ Evening Greeting. This is then followed by the Built-In
Auto Attendant-Welcome Greeting for the current time zone (working hours, break hours, non-working
hours). A Voice Module must be assigned for the Built-In Auto Attendant Time-based as well as the
Welcome Greeting.
• The Built-In Auto Attendant Time-based and Welcome Greeting messages are played once.

If no voice module is assigned as Welcome Greeting, the system will play music-on-hold after answering
the call. It will play music-on-hold until the end of the Built-In Auto Attendant Music Timer (default: 5
seconds).
• On the completion of the Welcome Greeting or music-on-hold at the end of the Built-In Auto Attendant
Music Timer, the system plays the Built-In Auto Attendant Dial Message to prompt the caller to dial the
The Built-In Auto Attendant Dial Message is played once and the caller gets Beeps. The system waits for
the Built-In Auto Attendant Beeps Timer (default: 10 seconds) to expire.
• If the caller does not dial any number before the Built-In Auto Attendant Beeps Timer expires, the system
plays the Built-In Auto Attendant Call Transfer to Operator message and transfers the call to the landing
destination as configured by you—Extensions or Operator.
The system waits for the duration of the Built-In Auto Attendant Inactivity Timer (default: 60 seconds) for
the Operator to answer the call. If there is no answer at the end of this timer, the system releases the trunk.
If the caller fails to dial digits, you can have the call disconnected instead of having it routed to the
Operator. For this, you need to enable the Disconnect Built-In Auto Attendant call, when caller does
not dial any digit flag in the System Parameters. When this flag is enabled, the system will play the
Built-In Auto Attendant No Dial Voice message to the caller. If the caller fails to dial a digit within the Built-
In Auto Attendant Beeps Timer, the system will disconnect the call.
• If the caller dials the extension number, the system checks if the number is valid.
If the dialed digits are invalid, the system plays the Wrong Dial voice message to the caller. This message
is played once. The system waits for the duration for the Built-In Auto Attendant Error Tone Timer (default:
5 seconds).
If the Wrong Dial Voice Message is not programmed, the system plays Error Tone to the caller for the
duration of the Built-In Auto Attendant Error Tone Timer, followed by the Built-In Auto Attendant Dial
Prompt.
• If the number dialed by the caller is valid, the system checks if the dialed extension is free.
• If the dialed extension is busy, the system plays the Built-In Auto Attendant Busy Message to the caller.
The message is played once.
• If no Built-In Auto Attendant Busy Message is programmed, the caller will hear Busy Tone. The Busy Tone
is played for duration of the Built-In Auto Attendant Busy Tone Timer (default: 15 seconds), followed by the
Built-In Auto Attendant Dial Prompt.
To have the call disconnected if the dialed extension is busy, you may enable the Disconnect Built-In
Auto Attendant Call, when dialed number is busy flag in the System Parameters.
• The dialed extension is free. The system calls the extension and plays Built-In Auto Attendant Ring Back
Tone Message (if programmed) or Ring Back Tone to the caller. This message is played until the dialed
extension is ringing.
• The system waits for the period of the Built-In Auto Attendant Ring Timer for the dialed extension to
answer the call.

• When the dialed extension answers the call, the caller gets connected to the extension.
If the dialed extension does not answer before the expiry of the Built-In Auto Attendant Ring Timer, the
system prompts the caller to dial again with the Built-In Auto Attendant Dial Prompt message to the caller.
• The system diverts the call to the Operator. When the call is transferred to the Operator, the system plays
the Built-In Auto Attendant Call Transfer to Operator voice message (if programmed) or plays Ring Back
Tone to the caller.
If there is no reply from the dialed extension, you can have the call disconnected instead of having it routed
it to the Operator by enabling the Disconnect Built-In Auto Attendant call, when dialed number is not
responding flag in the System Parameters.
Voice Mail Auto Attendant
SARVAM UCS supports 64 simultaneous VMS calls207. These calls can be made by extension users to access
their mailbox or incoming calls landing on a trunks that has Voice Mail Auto Attendant enabled.
If all channels are used by extension users, the external incoming calls on the Voice Mail Auto Attendant enabled
trunks remain unanswered. To make sure that the incoming calls on the trunks are answered you can reserve
channels of the VMS as per your requirement.
The extension users will not have access to the channels reserved for incoming calls on the trunks to access their
mailbox.
If all the reserved channels are busy and there is an incoming call on the trunk, it will land on the unreserved
channel if free.
To reserve a channel, see “Configuring VMS General Parameters”.
When the Voice Mail Auto Attendant of SARVAM UCS is selected as the destination for incoming calls on a trunk,
this is how it will work:
• A call lands on a Trunk.
• The Voice Mail System (VMS) answers the call.
• The VMS greets the caller with the Welcome message and the Greeting Message selected for the current
time zone (working hours, break hours and non-working hours).
• If the system detects the day as a holiday, the VMS plays the Holiday Message. To know more, see
“Holiday Table”.
• The VMS plays prompts to the caller to process the call further.
Delayed Auto Attendant

You can use Delayed Auto Attendant to have incoming calls that are not answered by the landing destinations—the
Operator and the Trunk Landing Group—within a certain time period, to be handled either by the Built-In or the
Voice Mail Auto Attendant.
207. For the VMS calls, the number of channels that will be supported would be as per the license you purchase.

When you use Delayed Auto Attendant,
• as a call lands on a trunk, the system checks the incoming call routing configured for the current time zone
for the trunk.
• on finding Delayed Auto Attendant enabled, the system rings on the destination extensions (Operator and
Trunk Landing Group) for the duration of time defined for ringing the extensions (default: 10 seconds).
• if no reply is received from the extensions, the system routes the call to the auto attendant you selected,
which may by the Built-In Auto Attendant or the Voice Mail Auto Attendant.
• the call is processed further by the auto attendant you selected.
How to configure
To use the Built-In Auto Attendant on trunks, do the following:
1. Make a list of the trunks by their port type (CO, Mobile, SIP, T1E1PRI, BRI) and port number on which you
want to use the Built-In Auto Attendant.
2. In the “Trunk Feature Template” assigned to these trunks, enable Auto Attendant for the desired time
zones by selecting Built-In Auto Attendant.
3. Set the Start Time for the Morning, Afternoon and Evening Greeting Messages. Refer “Greeting Message
Time” in System Parameters for instructions.
4. Assign Voice Modules for Built-In Auto Attendant Messages. To play to callers pre-recorded voice
messages as Built-In Auto Attendant greetings and to play voice prompts at each stage of the call, you
need to assign Voice Modules for the following Built-In Auto Attendant Messages:
• Built-In Auto Attendant-Time-based Greetings (Morning, Afternoon and Evening Greetings):

These are played to the caller as soon as the call is answered by the Built-In Auto Attendant.
Different messages can be recorded for Morning, Afternoon and Evening Hours. You can also set the
time during which you want to play these greetings. For detailed instructions, see “Greeting Message
Time” in System Parameters.
• Built-In Auto Attendant-Welcome Greeting: Played to callers when answering the call. Different
welcome greetings can be programmed for Working Hours, Break Hours and Non-working Hours. The
Built-In Auto Attendant Welcome Greeting message is played once.
• Built-In Auto Attendant-Dial Prompt: Played after the Welcome greeting message to prompt the
caller to dial the desired extension number. This message is played once.
• Built-In Auto Attendant-Ring Back Tone: Played after the caller has dialed the number and the
system is ringing the dialed extension. This message is played continuously as the dialed extension
rings.
• Built-In Auto Attendant-Wrong Dial message: Played when the caller dials a wrong number or the
number dialed by the caller does not match with any extension number of SARVAM UCS. This
message is played once.
• Built-In Auto Attendant-Destination Busy: Played when the dialed extension is busy. This message
is played once.

• Built-In Auto Attendant-Destination No Reply: Played when the dialed extension does not respond.
This message is played once.
• Built-In Auto Attendant-No Dial: Played when the caller has not dialed any number. This message is
played once.
• Built-In Auto Attendant-Call Transfer to Operator: Played to the caller when the call is being
transferred to the Operator. This message is played once.
The default Voice Module numbers assigned to Built-In Auto Attendant messages and the messages
recorded on each module are:
Voice
Module Voice Message Application Voice Message
Number
02 Built-In Auto Attendant - Morning Greeting Good Morning!
03 Built-In Auto Attendant - Afternoon Good Afternoon!

Greeting
04 Built-In Auto Attendant - Evening Greeting Good Evening!
05 Built-In Auto Attendant - Welcome Welcome!

Greeting for Day Time (Working Hours)
06 Built-In Auto Attendant - Welcome Welcome! I am sorry, we are closed.

Greeting for Night time (Non-working hours)
07 Built-In Auto Attendant - Dial prompt Please dial the desired number.
08 Built-In Auto Attendant - No Dial message Sorry! You have not dialed any number.
09 Built-In Auto Attendant - Wrong Dial Sorry! The number is not valid.
message
10 Built-In Auto Attendant - Destination Busy The person you dialed is busy.
message
11 Built-In Auto Attendant - Destination The number you have dialed is ringing.
Ringing message
(Ring Back Tone)
12 Built-In Auto Attendant - Destination No The person you dialed is not responding.
Reply message
13 Built-In Auto Attendant - Call Transfer to Please hold, transferring your call to the
Operator message Operator.
You may customize these Built-In Auto Attendant voice messages by recording messages of your
choice and assigning them to the voice modules. For instructions on recording messages on the voice
modules and assigning voice modules to different functions, see “Voice Message Applications”.
If you do not use any of the above voice modules, the system will play the Call Progress Tone for each call
state.
To use the Voice Mail Auto Attendant on trunks, do the following:
1. Make a list of the trunks by their port type (CO, Mobile, SIP, BRI, T1E1PRI) and port number on which you
want to use the Voice Mail Auto Attendant.

2. In the “Trunk Feature Template” assigned to these trunks,
• enable Auto Attendant by selecting Voice Mail Auto Attendant for the desired time zones.
• select the desired Voice Mail Auto Attendant (VMAA) Menu.
3. Complete the VMS related configuration.
• Configure Welcome and Greeting messages. You may either use the default, pre-recorded welcome
messages of the VMS, or record the custom welcome messages that meet your requirements, in .WAV
file format.
• Configure the Voice Mail Auto Attendant (VMAA) Menu parameters.
For more information and instructions, see the “Configuring Voice Mail System”.
To use Delayed Auto Attendant on trunks, do the following:
1. Make a list of the trunks by their port type and port number on which you want to enable Delayed Auto
Attendant.
2. In the “Trunk Feature Template” assigned to these trunks,
• set the Auto Attendant Delayed Timer.
• if you want to use the VMS Auto Attendant for Delayed Auto Attendant, select Voice Mail Auto
Attendant and complete the voice mail related configuration. For more information and instructions,
see the “Configuring Voice Mail System”.
• if you want to use the Built-In Auto Attendant for Delayed Auto Attendant, select Built-In Auto
Attendant, and assign the Voice Modules for Built-In Auto Attendant Messages, as described
earlier.
If required, you may also change the default values of the following Built-In Auto Attendant related Timers and set
them to the desired values.
• Built-In Auto Attendant Inactivity Timer (default: 60 seconds)

• Built-In Auto Attendant Answer Wait Timer (default: 5 seconds)
• Built-In Auto Attendant Music Timer (default:5 seconds)
• Built-In Auto Attendant Beeps Timer (default: 10 seconds)
• Built-In Auto Attendant Ring Timer (default: 30 seconds)
• Built-In Auto Attendant Busy Tone Timer (default: 15 seconds)
• Built-In Auto Attendant Error Tone Timer (default: 5 seconds)
To know more about these timers and for configuration instructions, see “System Timers and Counts”.
You may also configure the following Built-In Auto Attendant related flags, as required:
• Disconnect Built-In Auto Attendant call, when dialed number is busy: When this flag is enabled, if
the dialed extension is found busy, the system will disconnect the call instead of routing it to the
Operator. Default: disabled.

• Disconnect Built-In Auto Attendant call, when dialed number is not responding: When this flag is
enabled, if there is no reply from the landing destination extensions, the system will disconnect the call
instead of routing it to the Operator. Default: disabled.
• Disconnect Built-In Auto Attendant call, when caller does not dial any digit: When this flag is
enabled, if the caller fails to dial a digit within the Built-In Auto Attendant Beeps Timer, the system will
disconnect the call instead of routing it to the Operator. Default: disabled.
These flags may be used in Hotels that provide 'Limited Services' and do not want to receive unanswered/busy
calls on the guest phones. For instructions on enabling or disabling these flags, see “System Parameters”.

Auto Answer
What's this?
Auto Answer allows incoming calls to be answered without any manual interventions by the extension users.
This feature is particularly useful for Operators in high call traffic settings, as it saves them the effort of picking up
the handset or pressing the speaker key repeatedly.
This feature works on digital key phones (DKP) as well as in Extended IP Phones.
How it works
With Auto Answer set on an extension DKP/Extended IP Phone, whenever a call lands on the DKP/Extended IP
Phone extension,
• the extension rings for the duration of the Auto Answer Timer208. This timer is programmable, and by
default it is set to 1 second.
• on the expiry of the Auto Answer Timer the system plays a beep to the user.
• the phone goes OFF-Hook to answer the call, without any intervention by the extension user such as
picking up the handset or pressing the speaker or the headset key.
• If a headset is connected, and headset connectivity is enabled, the incoming speech audio will be diverted
to the headset automatically.
• the extension user can talk to the caller.
Auto Answer works only if the phone is in idle state; the phone must not be busy with an active call or using a
feature.
How to configure
For Auto Answer to work, you are required to do the following:
1. Enable Auto Answer in the DKP Parameters/Extended IP Phone Extensions.
2. Change Auto Answer Timer, if required. The range of this timer is 1 to 9 seconds. By default, the Auto
Answer Timer is set to 1 second.
3. Enable Headset Connectivity, if headset is to be used for Auto Answer.
All of the above can be programmed by the System Engineer using Jeeves and a Telephone.
The extension users can also program the above parameters using the Phone Menu. See "How to use" Auto
Answer later in this topic.
208. This timer defines the time in seconds that the phone should wait before going OFF-Hook to answer incoming calls.

Programming Auto Answer using Jeeves
• If the DKPs are already installed and configured, identify the DKP you wish to provide Auto Answer feature
by its Hardware Port/Slot Number, Access Code or Name.
• Click the Advance button.
• Go to the column Auto Answer of the DKP parameters using the horizontal scroll bar. Enable the flag by
selecting the check box.
To cancel Auto Answer, disable the flag by selecting the check box.

• Now, go to the column Auto Answer Timer, and set the Timer as required. By default the Timer is set to 1
second.
• If Headset is to be used by the DKP, go to the column Headset Connected? and click the check box to
enable the flag.
• Repeat the above steps to program Auto Answer parameters for each DKP that is to be provided this
feature.
• Similarly configure these parameters for Extended IP Phones, see “Configuring Matrix SPARSH VP248”
and “Configuring Matrix SPARSH VP310”.
Programming Auto Answer using a Telephone

• Enter SE mode.
To set auto answer on DKP, dial:

• 1214-1-DKP-Auto Call Answer Mode to set Auto Answer on a single DKP.
• 1214-2-DKP-DKP-Auto Call Answer Mode to set Auto Answer on a range of DKPs.
• 1214-*-Auto Call Answer Mode to set Auto Answer on all DKPs.
Where,
DKP is the Software port number of the DKP, from 001 to 128.
Auto Call Answer Mode is
0 for Manual mode
1 for Auto Answer mode.
To set Auto Answer Timer, dial:

• 1215-1-DKP-Auto Call Answer Timer to set Timer for a single DKP.
• 1215-2-DKP-DKP-Auto Call Answer Timer to set the same Timer for a range of DKPs.
• 1215-*-Auto Call Answer Timer to set the same Timer for all DKPs.
Where,
Auto Call Answer Timer is from 1 to 9 seconds. Default: 1 second
To enable/disable Headset Connectivity flag, dial:

• 1213-1-DKP-Headset Connectivity Flag to enable/disable the flag on a single DKP.
• 1213-2-DKP-DKP-Headset Connectivity Flag to enable/disable the flag on a range of DKPs.
• 1213-*-Headset Connectivity Flag to enable/disable the flag on all DKPs.
Where,
Headset Connectivity Flag is
0 for Disable
1 for Enable
• Exit SE mode.
Similarly to configure these parameters for Extended IP Phones using a telephone, see “Configuring Matrix
SPARSH VP248”, “Configuring Matrix SPARSH VP310”, “Configuring Matrix SPARSH VP510” and for SPARSH
VP330 refer the SPARSH VP330 User Guide.

How to use
Extension users can set/cancel Auto Answer and enable Headset connectivity from their DKP/ Extended IP Phone
or by navigating the Phone Menu.
To set Auto Answer:
• Press the DSS Key assigned to Auto Answer function209.
OR

• Scroll down to select 'Phone Settings'; press Enter Key.
• Enter Your User Password (default: 1111).
• Scroll down to select 'Call Answer Type'; press Enter Key.
• You get the options:
Manual Call Answer
Auto Call Answer
• Scroll to select Auto Answer and press Enter key.
• Now select the Timer for Auto Answer from any of the options:
Answer After 1 sec (default)
Answer After 2 sec
:
Answer After 9 sec
• Press Enter Key.
To cancel Auto Answer:
• Repeat the above steps.

• Select 'Manual Answer' as the Call Answer Type.
It is recommended that Auto Answer Timer be set to at least 2 seconds.
To enable Headset Connectivity:
• Press the DSS Key assigned to Headset function210.
OR

• Scroll down to select 'Phone Settings'; press Enter Key.
• Enter Your User Password (default: 1111).
• Scroll down to select 'Headset Connectivity'; press Enter Key.
• You get the options:
Headset Not Connected
Headset Connected
• Scroll to select 'Headset Connected' and press Enter key.
209. This function must have been programmed by the System Engineer on a DSS Key of the phone. Refer "Digital Key Phone - Keys
Programming" for instructions.
210. This function must have been programmed by the System Engineer on a DSS Key of the phone. Refer "Digital Key Phone - Keys
Programming" for instructions.

Auto Call Back (ACB)
What's this?
If the extension number you have dialed is busy or is not responding, you may use the Auto Call Back feature,
instead of repeatedly dialing the number. Similarly, when you dial a code to access a trunk and the trunk is busy,
you may set Auto Call Back. SARVAM UCS allows you to set a maximum of 300 Auto Call Backs.
How it works
When you set Auto Call Back,
• the SARVAM UCS will queue your call attempt.
• As soon as both extensions, yours and the remote extension, are available, the system will ring first on
your extension for the duration of the Auto Call Back Ring Timer. This timer is set by default to 30 seconds
and is programmable.
• When you go OFF-Hook, the system will ring on the remote extension (provided it is also available at that
moment) for the duration of the Auto Call Back Ring Timer.
• When the remote extension user goes OFF-Hook, your call will get connected.
However, if the remote extension gets busy before the system can ring on it, the system will continue to try
again.
Auto Call Back set for a busy trunk works the same way. As soon as the busy trunk port you are trying to
access is available, the system will ring your extension. When you go OFF-Hook you will be connected to
the trunk port.
• Each extension of the SARVAM UCS can set only one Auto Call Back request at a time. If you set
another Auto Call Back request, before the first one has been served, the system will override the first
request and serve the second.
• The SARVAM UCS has the capacity to serve 300 Auto Call Back requests from its extensions at a
time. The service duration for each request is 60 minutes. Requests that are not served within 60
minutes are automatically canceled by the system. Also, the system will not serve any more requests if
all the 300 requests are pending. In such a case, the system will play an error tone, when an extension
attempts to make a request.
Auto Call Back request set by you will be cleared by the system if:
• it was successfully served, that is, your extension was connected to the remote extension or the trunk you
were trying to reach.
• you do not answer the Auto Call Back ring, before the expiry of the Ring Timer, that is, within 30 seconds
(default setting).
• the remote extension does not answer the Auto Call Back ring before the expiry of the Ring Timer.

• it has not been served within 60 minutes.
• Auto Call Back works for internal calls and for accessing trunk ports only.
• Internal calls include calls between Systems that are networked using Q-SIG.
How to configure
Auto Call Back is a Class-of-Service dependant feature. An extension user can set/cancel Auto Call Back only if it
is enabled in the extension's Class of Service.
The only programming involved in this feature is enabling/disabling Auto Call Back in the Class of Service and
changing the duration of the Auto Call Back Ring Timer, if required.
Both these can be programmed using Jeeves and a Telephone.
Programming Auto Call Back using Jeeves

In the default factory settings, Station Basic Feature Template Number 01 is assigned to all extensions of SARVAM
UCS. Template 01 has the features 'Auto Call Back Busy' and 'Auto Call Back No Reply' enabled in the default CoS
group 01. So, all extensions of the SARVAM UCS can set/cancel Auto Call Back if the called number is busy or
does not reply.
However, if Auto Call Back Busy/No Reply is to be denied to any of the extensions, follow these steps:
1. Define a CoS group with Auto Call Back Busy/No Reply disable.
3. Assign this new Template to the selected extensions to which Auto Call Back is to be denied.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for instructions on how to
enable/disable a feature in a CoS group, how to prepare a Station Basic Feature Template with a new CoS
group and assign the new template to SLT, DKP, SIP and ISDN Terminal extensions using Jeeves.
If the User wants to increase or decrease the duration of the of the Auto Call Back ring on both extensions,
that is, the extension requesting Auto Call Back and the destination extension, program the 'Auto Call Back
Ring Timer', according to User preference.
• Under Configuration, click System Timers and Counts to open the page.

• Scroll to reach Other Features.
• Set the Auto Call Back Timer (sec) to the desired duration.
Programming Auto Call Back using a Telephone

Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for instructions on how to enable/
disable a feature in a CoS group, how to prepare a Station Basic Feature Template with a new CoS group and
assign the new template to SLT and DKP extensions using SE commands.
To program Auto Call Back when Busy/No Reply in a CoS group, dial:
• 1302-1-COS Group-Feature Number-Code
Where,
CoS group is from 01 to 20
Feature Number for Auto Call Back when Busy is '04'
Feature Number for Auto Call Back when No Reply is '05'.
Code is
0 for Disable
1 for Enable.
For example: To enable Auto Call Back when Busy in CoS group 02, dial 1302-1-02-04-1
To enable Auto Call Back when No Reply in CoS group 02, dial 1302-1-02-05-1
To assign the CoS group with Auto Call Back Busy/No Reply to a Station Basic Feature Template, dial:
• 5502-1-Template Number-Feature Number-Code
Where,
Feature Number is

03 for CoS group for Working Hours
04 for CoS group for Break Hours
05 for CoS group for Non-Working Hours
Code is CoS group number from 01 to 20.
For example: To apply CoS group 02 with Auto Call Back Busy and No Reply for each Time zone in
Station Basic Feature Template 02, dial the following commands:
5502-1-02-03-02 for Working hours
5502-1-02-04-02 for Break hours
5502-1-02-05-02 for Non-working hours
• To apply the Station Basic Feature Template now programmed with Auto Call Back when Busy and No
Reply, to the extensions, dial the following commands:
If extension is an SLT, dial:

• 5503-1-SLT-Template Number to apply on a single SLT.
• 5503-2-SLT-SLT-Template Number to apply on a range of SLTs.
• 5503-*-Template Number to apply on all SLTs.
Where,
SLT is the Software port number of the SLT, from 001 to 512
Template Number is the number of the Station Basic Feature Template (01 to 50) you have
programmed with the Auto Call Back feature.
If extension is a DKP, dial:

• 5504-1-DKP-Template Number to apply to a single DKP.
• 5504-2-DKP-DKP-Template Number to apply to a range of DKPs.
• 5504-*-Template Number to apply to all DKPs
Where,
To program the Auto Call Back Ring Timer:

• Dial command 3801-Seconds
Where,
Seconds = 001 to 255 seconds.
• Exit SE mode.
How to use
Extension users can set two types of Auto Call Back:
• Auto Call Back on Busy - when the extension/trunk they are trying is Busy.
• Auto Call Back on No Reply - when there is no reply from the extension they are trying.Auto Call Back on
Busy

Using DSS Key:
To set Auto Call Back on Busy:
• Press the 'Call Back' Key on EON48 on Busy Tone or press the DSS Key assigned to Auto Call Back on
EON310 on Busy Tone.
• You get confirmatory message "Auto Call Back Set" on the phone's display. The LED of the DSS Key will
be turned on.
• Go idle or you get dial tone after 3 seconds.
To cancel Auto Call Back on Busy:
• Press the 'Call Back' Key on EON48 again or press the DSS Key assigned to Auto Call Back on EON310
again.
• You get confirmatory message "Auto Call Back Canceled" on the phone's display. The LED of the DSS
Key will be turned off.
Using Command:
• Dial 2 on Busy Tone.

• You get confirmatory message "Auto Call Back Set" on the phone's display. The LED of the DSS Key
assigned to Auto Call Back will be turned on.
To cancel Auto Call Back on Busy:
• Dial 102.
• You get confirmatory message 'Auto Call Back Canceled' on the phone's display. The LED of the DSS key
assigned to Auto Call Back will be turned off.
For SLT Users

• On Busy Tone.
• Dial 2.
• You get confirmatory tone
To cancel Auto Call Back:

• Dial 102.
• You get confirmatory tone.

Auto Call Back on No Reply

Using DSS Key:
To set Auto Call Back on No Reply:
• Press the 'Call Back' Key / DSS Key assigned to Auto Call Back on Ring Back Tone.
• You get confirmatory message "Auto Call Back Set" on the phone's display. The LED of the DSS Key will
be turned on.
To cancel Auto Call Back on No Reply:
• Press the 'Call Back' Key / DSS Key assigned to Auto Call Back again.
• You get confirmatory message "Auto Call Back Canceled" on the phone's display. The LED of the DSS
Key will be turned off.
Using Command:
• Dial 2 on Ring Back Tone.

• You get confirmatory message "Auto Call Back Set" on the phone's display. The LED of the DSS Key
assigned to Auto Call Back will be turned on.
To cancel Auto Call Back:
• Dial 102.
• You get confirmatory message 'Auto Call Back Canceled' on the phone's display. The LED of the DSS key
assigned to Auto Call Back will be turned off.
For SLT Users

• On Ring Back Tone

• Dial 2.
• You get confirmatory tone
To cancel Auto Call Back on No Reply:

• Dial 102.
• You get confirmatory tone.
If you hear an error tone while setting an Auto Call Back request, it is likely that the system already has 300
pending requests and is unable to accept yours.

Auto Redial
What's this?
The Auto Redial feature retries a call automatically if the dialed number is busy. It repeatedly checks the busy line
till it is free. When the called number is no longer busy, the extension of the caller rings.
Auto Redial saves time and the effort of repeatedly dialing the entire phone number over and over until the called
party gets off the phone.
The Auto Redial feature is supported for external numbers only. Maximum 50 Auto Redials can be set by extension
users.
How it works
When an extension user dials a number and gets a busy tone, s/he may set Auto Redial. When Auto Redial is set,
• SARVAM UCS checks for a free trunk to dial the number.
• SARVAM UCS will dial out the requested number and will wait until the 'Ring Back Tone Wait Timer211'
expires to sense the Ring Back Tone from the requested number. This timer is programmable and is set to
60 seconds as default.
• If the system does not detect Ring Back Tone for 60 seconds, it releases the trunk and tries again after
some time. If the system detects a busy tone, it releases the trunk and redials the number automatically
after some time. This process is repeated until the system detects the Ring Back Tone.
• When the SARVAM UCS detects the Ring Back Tone instead of the Busy Tone, it will ring on the
extension that set Auto Redial. The extension will ring for the duration of the 'Redial Ring Timer212'. This
timer is programmable and is set to 45 seconds as default.
• The extension must go OFF-Hook to get connected to the remote party.
• If the extension is in the middle of any activity such as dialing, ringing or speech, the SARVAM UCS will
suspend Auto Redial until the extension becomes idle again. After which it dials the requested number
again.
Two types of Auto Redial are supported by the SARVAM UCS - Auto Redial (normal) and Auto Redial 'Priority' -
that differ from each other in terms of the number of redial attempts and the interval between attempts.
• Auto Redial (normal): The system is programmed by default to make 5 attempts to redial at an interval of
45 seconds (default) between each attempt. Both, the number of attempts as well as the duration of the
interval can be changed match User preference, like decreasing the number of attempts to 3 and
increasing the interval to 60 seconds.
211. Time for which SARVAM UCS waits to sense the RBT from the PSTN/CO Network after dialing the requested number. This timer
is particularly relevant to CO ports. Valid range of the timer: 000 to 255 seconds. Default: 060 seconds.
212. Time for which the extension that has requested Auto Redial should ring. Valid range of the timer: 000 to 255 seconds. Default:
045 seconds.

• Auto Redial 'Priority': the system makes a greater number of attempts to redial and the duration of the
interval between each attempt is less. By default the system is programmed to make 20 redial attempts at
intervals of 20 seconds. The number of attempts as well as duration of the interval are programmable; for
instance, the number of attempts can be set to 30 and the interval to 15 seconds.
To change the number of redial attempts and the interval between them, the SE must Auto Redial Count and the
Auto Redial Timer respectively. In addition to these, the system has three other related timers, which can be
programmed to match User preference:
• Auto Redial Ring Back Tone (RBT) Wait Timer
• Auto Redial Ring Timer
• An extension user can request Auto Redial for multiple numbers at a time from the same extension and
more than one extension can attempt auto redial simultaneously.
• The system uses the same OG Trunk Bundle Group you used. If you dialed the number on group code
60, the system grabs one of the free trunks from group code 60 for Auto Redial.
• If the number was dialed the first time using selective trunk access, the system will use the same trunk
to execute Auto Redial.
• If the extension is programmed for 'Dynamic Lock', and you have set the 'Auto Redial', the system will
check the Toll control as per dynamic lock level.
Auto Redial may not work well on Two-wire Trunk lines, as its functioning greatly depends on line
condition. Unlike ISDN, GSM and VoIP trunks, the line condition of CO trunks may not always measure up
to the standard requirement for Auto Redial to function.
How to configure
For Auto Redial to work, the System Engineer must:
1. Enable the features 'Auto Redial' and 'Auto Redial Priority' in the Class of Service (CoS) group of the
extensions to which this feature is to be allowed.
2. Change the 'Auto Redial Normal/Priority Count' and the 'Auto Redial Normal/Priority Timer' to match User
preference. This will change the number of redial attempts made by the system and the interval between
them.
3. If required, also change other related Timers such as Auto Redial Dial Tone Wait Timer, Auto Redial Ring
Back Tone (RBT) Wait Timer, Auto Redial Ring Timer.
All the above parameters can be programmed using Jeeves and by dialing SE commands from a
telephone.
Programming Auto Redial using Jeeves

In the default factory settings, Station Basic Feature Template Number 01 is assigned to all extensions of the
system. The Station Basic Feature Template 01 does not have the feature Auto Redial and Auto Redial Priority in
the default CoS group 01. Thus none of the extensions of the SARVAM UCS have this feature.

If the User wants to allow all extensions the Auto Redial and /or the Auto Redial Priority feature, the you can simply
enable this feature in the default CoS group 01.
However, if Auto Redial/Auto Redial Priority is to be allowed on only select extensions, follow these steps:
1. Define a CoS group with Auto Redial/Auto Redial Priority enabled.
3. Assign this new Template to the extensions to which Auto Redial/Auto Redial Priority is to be allowed.
To change Auto Redial Counts and Timers:
• Under Configuration, click System Timers and Counts.
• Below Auto Redial, change the Count and Timer of the type of Auto Redial - Normal or Priority - as per
your requirement.
• You may change any of the related timers - Auto Redial Dial Tone Wait Timer, Auto Redial Ring Back Tone
(RBT) Wait Timer, Auto Redial Ring Timer - as per your preferences on this page.
• If the SARVAM UCS is installed in Australia, please ensure that:

i. The Timer for Auto Redial Normal as well as Priority must be set to more than 5 seconds.
ii. The Auto Redial Priority Count should be set to less than 15.

Programming Auto Redial using a Telephone
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for instructions on how to use SE
commands to:
• enable a feature (in this case Auto Redial and Auto Redial Priority) in the CoS group.
• assign the CoS group to a Station Basic Feature Template.
• apply the Template with the on DKP and SLT extensions, using SE commands.
To program the relevant Timers and count for Auto Redial:
To set Auto Redial Normal - Timer:

Where,
Seconds is from 000 to 255.
To set Auto Redial Normal - Count:

• Dial command 1705-Count
Where,
Count is from 000 to 255.
To set Auto Redial - Priority:

Where,
To set Auto Redial - Count:

• Dial command 1707-Count
Where,
To change Auto Redial RBT Wait Timer:

Where,
Seconds is from 000 to 255 seconds.
Do not set this timer to less than 2 seconds.
To change Redial Ring Timer:

Where,
• Exit SE mode.

How to use
Auto Redial can be set/canceled from EON/Extended IP Phone as well as SLT.

Using DSS Key
To set Auto Redial:
• When the external number you are trying is busy,

• On Busy tone, press the 'Redial' Key.
• Go Idle.
To cancel Auto Redial:
• Press the ‘Redial’ Key again.
Using Command:
To set Auto Redial:

• Go ON-Hook on Busy Tone.
• Dial 17.

• Dial 1070.
For SLT Users

To set Auto Redial

• Go ON-Hook on Busy Tone.
• Dial 17.

• Dial 1070.

Automatic Number Translation
What's this?
SARVAM UCS offers connectivity to different networks - PSTN, GSM, ISDN T1E1PRI, BRI, VoIP - each having a
different numbering plan. For example, the GSM network requires area codes to be dialed also for local numbers,
whereas the PSTN requires dialing of area codes for long distance calls.
When SARVAM UCS is connected with multiple networks, outgoing calls may be routed through any of these
networks, depending on the routing pattern configured in the system. However, as extension users do not know
through which telecom network their calls will be routed, they cannot be expected to dial numbers according to the
numbering plan of the destination networks.
The Automatic Number Translation feature of SARVAM UCS takes care of this. It modifies/manipulates dialed
numbers or part thereof to match with specific route numbering plan understood by the destination network (PSTN,
GSM, VoIP). This includes adding or stripping off of country codes and area codes.
For example, when an extension user dials a local landline number, if Automatic Number Translation is so
programmed, the SARVAM UCS will prefix the number with the appropriate country-area code before routing the
call through the GSM network.
How it works
Automatic Number Translation makes use of the Automatic Number Translation (ANT) Table. The ANT Table
consists of three columns:
• Dialed Number: This column contains the numbers you expect the users to dial.
• Strip Digit: This column contains the number of digit(s) to be stripped off by the system from the Dialed
Number string before dialing it out.
• Add Prefix: This column contains the digit(s) which are to be added as prefix to the Dialed Number string
by the system before dialing it out.
This table is applied on the desired trunk, through which outgoing calls are made.
Upto 8 different ANT tables can be configured and each table can accommodate upto 32 strings.
Here is an example of how this table is to be configured and used:
You want
• All 10-digit numbers to be dialed out after adding the prefix ‘1’.
• All 7-digit numbers, starting with 2 to be dialed out after adding the prefix ‘1315’.
• All numbers beginning with 91 to be stripped off the first 2 digits and ‘0’ to be added as prefix.
When you do not want to specify any numeric digits in the numbers to be modified, use the character X. This
character represents any numeric digit from 0 to 9. For example, a 10-digit number (having the numeric digits from
0 to 9) can be represented using this character as XXXXXXXXXX.

Thus, the entries you will need to make in the ANT table will be as follows:
Dialed Numbers Strip Digit Add Prefix
XXXXXXXXXX 0 1
2XXXXXX 0 1315
91 2 0
• The entry for 10-digit numbers to be dialed out after adding the prefix ‘1’ will be as shown in the first row of
this table. The 10-digit number is represented with the X character in the Dialed Numbers column. Since
no digit is to be stripped off, ‘0’ is entered in the Strip Digit column. As the prefix ‘1’ is to be added, this
number is entered in the Add Prefix column. The system will add 1 as prefix before dialing out numbers
from 0000000000 to 9999999999.
• Similarly, the entry for 7-digit numbers starting with 2 to be dialed out after adding the prefix ‘1315’ will be
as shown in the second row of this table. The system will add 1315 as prefix before dialing out numbers
from 2000000 to 2999999.
• The entry for all numbers beginning with 91 to be stripped off the first 2 digits and ‘0’ to be added as prefix
will be as shown in the third row of this table. When users dial numbers beginning with 91, the system will
strip off the first two digits and add 0. For example, when a user dials the number 919925801882, the
system will dial out 09925801882.
Automatic Number Translation also forms the basis of “Multi-Stage Dialing”.
How to configure
To apply Automatic Number Translation on a trunk,
• Decide which of the trunks types are to be assigned the Automatic Number Translation (ANT) feature.
• Decide the number of ANT tables you need. You can program 8 different tables, with a maximum of 32
entries in each.
• Make the tables on a piece of paper by drawing three-column tables. In the first column of a the table, write
the dialed numbers that need to be modified before being dialed out from the trunk. For each dialed
number in the first column, enter the number of digits you want the system to strip off (if required) from this
number in the second column. In the third column enter the number you want the system to add as prefix
(if required) before dialing out the number.
• Configure the Automatic Number Translation Table in the system using the tables you prepared.
• Enable the Automatic Number Translation flag in the Outgoing Trunk Bundle (OGTB) of the trunk.
• Assign the Automatic Number Translation Table you configured to the OGTB of the trunk.
This can be done using Jeeves and a Telephone.
Configuring Automatic Number Translation using Jeeves

• Click Configuration.

Configuring the ANT Table
• Click Automatic Number Translation.
• Choose any table number from 1 to 8.
• In the Dialed Number column of the table you chose, configure the numbers you expect the extension
users to dial. The Dialed Numbers can be a maximum of 16 characters. Default: Blank.
When you want to specify number-length without specifying the numeric digits, use the character X to
represent the digit. X represents any number from 0 to 9.
• In the Strip Digit column, enter the number of digits you want the system to strip off from the Dialed
Number before the dialing out this number. The valid range is from 00 to 16. Default: 0.
• In the Add Prefix column, enter the number you want the system to add as prefix to the Dialed Number
before the system dials out this number. The Prefix can be a maximum of 40 characters. Default: Blank.
Enabling Automatic Number Translation on Trunks

• Under Configuration, click Outgoing Trunk Bundle to open the page.
• In the OG Trunk Bundle of the desired trunk, enable the Automatic Number Translation (ANT) Apply
flag by selecting the check box.

• In the ANT Table No. list, select the table number you configured for this trunk.
• Repeat the steps to apply the feature on another OGTB.
Configuring Automatic Number Translation using a Telephone

The ANT Table can be configured using Jeeves only.
To enable Automatic Number Translation Flag on OGTB, dial:

• 6702-1-OG Trunk Bundle Number-Feature Number-Code to enable the flag in a single trunk bundle.
• 6702-2-OG Trunk Bundle Number-OG Trunk Bundle Number-Feature Number-Code to enable the
flag in a range of trunk bundles.
• 6702-*-Feature Number-Code to enable the flag in all trunk bundles.
Where,
OG Trunk Bundle Number is from 001 to 128.
Feature Number for Automatic Number Translation flag is '5'.
Code is:
0 for Disable
1 for Enable
Default: Disabled.

For example:
To enable Automatic Number Translation flag in OGTB number 1, dial: 6702-1-001-5-1
To enable the same flag in OGTB numbers 1 to 8, dial: 6702-2-001-008-5-1
To enable the same flag in all OGTBs, dial: 6702-*-5-1
To assign an Automatic Number Translation Table, dial:

• 6702-1-OG Trunk Bundle Number-Feature Number-Code to assign a Table to a single trunk bundle.
• 6702-2-OG Trunk Bundle Number-OG Trunk Bundle Number-Feature Number-Code to assign the
same Table to a range of trunk bundles.
• 6702-*-Feature Number-Code to assign the same Table to all trunk bundles.
Where,
Feature Number for Automatic Number Table is '6'.
Code is ANT Table number from 1 to 8.
For example:
To assign ANT Table 3 to OGTB number 1, dial: 6702-1-001-6-3
To assign ANT Table 3 to OGTB numbers 1 to 8, dial: 6702-2-001-008-6-3
To assign ANT Table to all OGTBs, dial: 6702-*-6-3
• Exit SE mode.

Barge-In
What's this?
Barge-In allows you to break into an on-going conversation between two extension users, between an extension
user and an external caller as well.
Barge-In can be used by Operators to transfer Incoming calls to busy extensions. The Operator can put the caller
on hold, barge into the busy extension to inform about the call, and then transfer the call.
Barge-In can be used by a Boss to interrupt the secretary's busy extension.
SARVAM UCS offers flexibility to allow/deny Barge-In feature to an extension user, that is, allow the extension user
to barge into on-going conversations. It also provides the flexibility to prevent conversations of extension users from
being barged in, referred to as Privacy against Barge-In.
How it works
• A, B and C are users of the system.
• A and B are talking to each other.
• C calls A.
• C gets busy tone.
• C dials Barge-In feature code.
• C gets Ring Back tone (RBT) and A gets beeps indicating a new call. If A is using EON, C's name and
number appear on C’s phone display.
• C gets RBT and A gets beeps for Barge-in timer. (By default, 10 seconds)
• During the beeps, A may press 'Flash' to answer the call.
• If A does not respond till the end of the Barge-In Timer (set to 10 seconds, by default), A gets connected to
C. B is put on hold and is given hold-on music.
• Call States:
• Barge-In works only if the dialed extension is busy. The dialed extension may be busy with another
extension or trunk (external number).
• Barge-In cannot be used when accessed trunk is busy.
• Barge-In works only if the user about to be barged in is in a two-way normal speech with another user
or external party.

• It will not work if the busy signal is due to the user being OFF-Hook, or in the middle of dialing, or
accessing a feature of the System.
• “Call Toggle”: Once A and C comes in speech with each other, A can toggle between B and C using Call
Toggle feature.
• Privacy against Barge-In: If the feature 'Privacy against Barge-in is enabled for an extension, it cannot be
barged into.
• “Priority”: No Interaction with Barge-In. If 'A' has lower priority than 'B' but has Barge-In enabled; A can
barge in B.
• “Do Not Disturb (DND)”: Barge-In will not work if the called user has set DND. If 'A' has set DND. A is
busy with C. B calls A. B cannot barge in A.
• DND Override: Barge-In will work if the calling user is allowed DND-Override. If 'A' has set DND. A is busy
with C. B calls A. On busy signal, B dials the Barge-In code. Barge-In will be successful only if B has DND-
Override enabled.
How to configure
The functioning of this feature is controlled by three parameters, 'Barge-In', 'Privacy against Barge-In' and 'Barge-In
Timer'.
Barge-In and Privacy against Barge-In

First decide which of the extensions are to be allowed Barge-In and the extension that are to be protected against
Barge-In.
In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the extensions of
SARVAM UCS. The Station Basic Feature Template 01 is assigned CoS group 01. The default CoS group 01 has
both Barge-In and Privacy from Barge-In are disabled. So, none of the extensions of the SARVAM UCS can use
these features.
If you want to allow Barge-In to the all extensions, simply enable Barge-In in the default CoS group 01.
However, if Barge-In is to be allowed on only selected extensions then follow these steps:
a. Define a CoS group with Barge-In enabled.
b. Prepare an extension Basic Feature Template with this CoS group applicable in all the “Time Zones”.
c. Assign this new Template to the extensions to which Barge-In is to be allowed.
Repeat the above steps to allow 'Privacy from Barge-In' in the CoS of extensions that are to be exempted from
Barge-In.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions on
programming.

Barge-In Timer
Barge-In Timer is the time after which the caller gets connected to the called party. By default the Timer is set to 10
seconds.
Changing Barge-In Timer using Jeeves

• Scroll to reach Other Features.
• Go to the parameter Barge-In Timer (sec).
• Set the desired value for the Timer.
• Log out of Jeeves or continue, as required.
Changing Barge-In Timer using a Telephone

• Enter SE mode.

Where,
Seconds is from 001 to 255 seconds. Default is 10 seconds.
• Exit SE mode.

How to use

• Dial an extension.
• If the extension is busy, you get Busy Tone.
• Press DSS Key assigned to 'Barge-In' function.
OR
• Dial 4213.
• You get Ring Back Tone.
• Wait for the system to connect you to the called extension.
• Talk.
• Replace the handset after the conversation has ended.
For SLT Users

• Dial an extension.
• If the extension is busy, you get Busy Tone.
• Dial 4.
• Wait for the system to connect you to the called extension.
• Talk.
• Replace the handset after the conversation has ended.
213. This default feature access code can be changed to suit your preference. Refer the topic “Access Codes”.

BCCH Selection
What's this?
BCCH Selection feature enables you to lock the Mobile Port of SARVAM UCS to a particular cell or channel or BTS
(Base Transceiver Station) for various reasons such as:
• better network availability
• minimum call drop due to bad signal/ network failure, etc.
To use this feature, make sure GSM214 is selected as the Preferred Network Mode of the Mobile Port. For
instructions, see “Mobile Port Parameters” in “Configuring Mobile Trunks”.
How it works
In the GSM network, each BTS is assigned one particular channel called as ARFCN (Absolute Radio Frequency
Channel Number), which is transmitted by BTS in BCCH (Broadcast Control Channel).
Now, when SARVAM UCS is switched on, the Mobile Port gets registered with the network on a particular BTS
which has the highest signal strength. However, the signal strength is not consistent. It keeps fluctuating, resulting
in call drop or poor voice quality.
Therefore, to avoid this, SARVAM UCS enables you to lock the Mobile Port to a particular cell or channel manually
after checking Signal Strength and Signal Quality of each cell.
How to configure
You can lock Mobile Port to a cell or a channel only through Jeeves.
214. BCCH Selection will not be supported if QUEC TEL UC20 module is installed in your system.

• Click BCCH Selection to open the page.
• The page displays the following parameters:
• Mobile Port Number: This is number of the Mobile port for which BCCH Selection status is displayed.
You can choose a different Mobile Port number from the drop down list. The page will display the
BCCH Selection related parameters for the selected mobile port.
• Mobile Port Status: The current state of the Mobile Port is displayed in this field. Given below is the
description of the various status indication messages that will appear in this field.
STATUS DESCRIPTION
Disabled Displayed when Mobile Port is disabled.
GSM Displayed when GSM module is in initialization state, that is, before SIM
Initialization detection.
SIM Absent Displayed when SIM Card is not detected by the system.
SIM PIN wrong Displayed when wrong SIM PIN is issued.
SIM PUK Displayed when SIM PUK is required.

required
Registering Displayed when the Mobile Port is in registration process with the Network.
Idle Displayed when the Mobile Port is registered with the Network and it is
free.
Busy Displayed when any active call is present on the Mobile Port.

• BCCH Locking Status: The current BCCH Locking status of the mobile port is displayed in this field.
Given below is a description of the various BCCH Locking status indication messages that will appear
in this field.
STATUS DESCRIPTION
Trying to Lock Displayed when user selects Manual BCCH Locking as 'No' from 'Yes' and
module is in initialization process after system or module restart.
Trying to lock on Displayed when BCCH Locking is selected as Manual and the Mobile Port is in
BCCH xxxxx the registration process with the Network. xxxxx is the BCCH selected by the
user for locking the cell.
Manually Locked Displayed when BCCH Locking is selected as Manual and Mobile Port is
on BCCH xxxxx successfully registered with the Network. xxxxx is the BCCH selected by the user
for locking the cell.
Auto Locked on Displayed when BCCH Locking is selected as Auto and Mobile Port is
BCCH xxxxx successfully registered with the Network. xxxxx is the BCCH of the Main Cell.
xxxxx is updated as per the changes in the Main Cell's BCCH.
• Main Cell- Bit Error Rate (%): Bit Error Rate of the Main Cell is displayed in this field. Bit Error Rate
(BER) is the percentage of received bits on a digital link that are in error relative to the number of bits
received. Bit Error Rate is calculated from the received signal quality.
• Manual BCCH Locking: This parameter allows you to lock the Mobile Port to a particular cell of your
preference. By default, manual BCCH locking is set to 'No'. When manual BCCH locking is set to 'No',
Mobile Port gets locked to the cell as per the highest signal strength. Select 'Yes' if you want to lock the
Mobile Port to the particular cell selected by you.
• Auto Refresh: Click this button to refresh BCCH Selection page. All parameters on this page will be
downloaded automatically after every 15 seconds. By default, Auto Refresh button is enabled.
• Stop Auto Refresh: By clicking this button, you can stop the system from automatically refreshing the
BCCH Selection page every 15 seconds. When you stop Auto Refresh, you must click 'Refresh' at the
bottom of this page to refresh the page whenever you want
• Cells: Indicates the cells with which the Mobile Port can be locked. You can decide to lock the Mobile Port
with a particular cell after considering the following cell related parameters, which appear on the page:
• MCC-MNC: In this field, MCC-MNC of a cell is displayed. Mobile Country Code (MCC) is a three digit
number uniquely identifying a country and Mobile Network Code (MNC) is either a two or three digit
number used to identify a given network from within a specific country.
• LAC (Location Area Code): In this field, LAC (Location Area Code) is displayed. LAC uniquely
identifies a location area within a GSM PLMN (Public Land Mobile Network). The maximum length of
LAC is 16 bits ranging from 0 to 65535. LAC is displayed in hexadecimal characters for SIMCOM-2G
and Wavecom-2G engines which ranges from 0000 to FFFF. For SIMCOM-3G engine, LAC is
displayed in decimal digits which ranges from 00000 to 65535.
• Cell ID: In this field, Cell ID is displayed. It is a 16-bit identifier that identifies the cell. Cell ID is
displayed in hexadecimal characters for SIMCOM-2G and Wavecom-2G engines which ranges from
0000 to FFFF. For SIMCOM-3G engine, Cell ID is displayed in decimal digits which ranges from 00000
to 65535.

• BSIC (Base Station Identification Code): In this field, BSIC (Base Station Identification Code) is
displayed. BSIC allows a mobile station to distinguish between different neighboring base stations.
BSIC is a three-digit value ranging from 0 to 255.
• BCCH (Broadcast Control Channel): In this field, the BCCH value of the cell is displayed. BCCH
defines the frequency channel number.
• Receive Level: In this field, the Receive Signal Strength level of the cell is displayed. It is the average
Receive Signal Strength of the cell. Its value ranges from -110 dBm to -47 dBm.
• Manual Cell Locking: This radio button is for locking a Mobile Port to a selected cell manually.
Now, to lock a Mobile Port to a particular cell,
• Select the desired Mobile Port Number from the drop down list.
• Set the parameter Manual BCCH Locking to Yes.
• Go to the Cell to which you want to lock the Mobile Port you selected.
• Select the radio button Manual Cell Locking of that Cell.
• The BCCH Locking for the selected Mobile Port will appear on this page, if Auto Refresh is enabled.
• If you have stopped Auto Refresh, click 'Refresh' at the bottom of the page to refresh the page and view
the current BCCH Locking settings of the selected Mobile port.
• You may now log out of Jeeves.
Example:
Consider the following example when using this feature:
Problem:
• SARVAM UCS is installed in roaming area, where more than one network is available, say A and B.
• Mobile Network Selection is set to 'Manual' mode and the first priority is programmed as network A and the
second priority is programmed as network B.
• The Mobile Port gets registered with A network. After registration, the user locks the Mobile Port to one of
the cells of A network.
• After registration, if the module or the system restarts or gets de-registered from the network, module
starts registration process again.
• While re-registering, SARVAM UCS tries to lock the Mobile Port to the last selected cell of network A.
• If network A is unavailable then the Mobile Port will not get registered with the network.
Solution:
• In this situation, user should set Manual BCCH locking mode to 'No' to register Mobile Port with the
suitable network automatically.
• Later, the user can set the Manual BCCH locking mode to 'Yes' and lock the Mobile Port to the desired cell
after assessing the cell information.

Behind the System Application
What's this?
It is common for small and medium Systems to be connected to larger System systems, where the trunks of the
larger System are connected to the stations of the smaller system. This is usually done for the purpose of
expanding the capacity of the large System already in use.
Such configurations are referred to as 'Behind the System Application'.
How it works
Consider the following illustration.
System-A is connected behind SARVAM UCS. In this 'Behind the System' configuration, the Trunk Lines T5, T6, T7
of SARVAM UCS are connected to the Stations (SLT) S4, S5, S6 of System-A.
However, Trunk lines T1 and T2 of System-A are connected directly to the PSTN.
In such application scenarios, implementing toll control restrictions for the trunks is a difficult task for SARVAM
UCS.
For example: Extension number 21 of SARVAM UCS in the above illustration is not allowed the facility of long
distance dialing. It has access to all the CO trunks.
When the user of Extension 21 wants to access T1, T2 or T3 (which are direct trunks from the PSTN to SARVAM
UCS) the user dials '0' (Trunk Access Code programmed), gets PSTN dial tone. When the user dials the number,
SARVAM UCS applies Toll Control.
When the user of Extension 21 tries to grab a trunk T5, T6 or T7 (which are connected to stations of System-A) by
dialing Trunk Access Code, for example, '0', the user gets the dial tone of System A. This means, the user of
Extension 21 must dial '0' again to grab PSTN dial tone of the T1/T2 connected to System-A.
But when the user dials '0' again, SARVAM UCS plays an Error Tone, because SARVAM UCS has applied Toll
Control and since Extension 21 is not allowed long distance dialing, SARVAM UCS rejects dialing on trunk and
plays error tone.

This would not have been a problem if Extension 21 were allowed long distance dialing. Since Extension 21 cannot
be allowed long distance dialing, SARVAM UCS provides a solution for this in the form of a programmable Pre-
PSTN Digit Count (PPDC) for each CO trunk.
The Pre-PSTN Digit Count defines the number of digits to be dialed to reach the PSTN. The system will apply Toll
Control check for the extension only after the programmed PPDC.
PPDC is to be programmed only for trunks that are connected to another System, and not for Trunks connected
directly to the PSTN. To take the above illustration further, PPDC must be programmed only for T5, T6, and T7.
PPDC count programmed should have the same number of digits as the Trunk Access Codes programmed for
System-A. For example, if the Trunk Access Code is a single digit number, such as '0', the PPDC will be '1'. If Trunk
Access Code is a two-digit number, such as 61, the PPDC will be '2'.
Since PPDC is not applicable on trunks directly connected to the PSTN, it must be programmed as '0' for T1, T2,
T3, T4 of SARVAM UCS.
How to configure
The 'Pre-PSTN Digit Count' (PPDC) is to be programmed in the “CO Hardware Template” applied to the CO trunks
of the System that are connected to station ports of the other System as well as to CO trunks that are directly
connected to the PSTN.
• For CO Trunks that are directly connected to the PSTN, PPDC must be programmed as '0'.
• For CO Trunks that are connected to the stations of another System, PPDC must be programmed as per
the number of digits in the Trunk Access Codes defined for the second System.
This can be done using Jeeves as well as from a Telephone.
The PPDC should be programmed only for 'Behind the System Applications'. For all normal applications,
this count must be set to '0' for all the trunks. Otherwise, external number dialing may be hampered.
Features like Least Cost Routing and Station Message Detail Recording will also be affected.
Configuring Pre-PSTN Digit Count (PPDC) using Jeeves

• Click CO Hardware Template to open the page.

• Scroll with the horizontal scroll bar to reach the PPDC column of the template.
• By default CO Hardware Template Number 01 is assigned to all trunks. The default 'PPDC' in this template
is '0'.
• For all trunks that are to be assigned PPDC '0' (that is, trunks connected directly to the PSTN), you may
retain this template.
• For trunks that are to be assigned a PPDC count from 1 to 6 (that is, trunks connected to the stations of
another System), prepare another CO Hardware Template by selecting another template number, for
instance Template 02.
• From the drop down list, select the appropriate value. This would depend on the number of digits in the
Trunk Access Code defined for the trunks in the other System. If the TAC is single digit, select '1'. If TAC is
double or triple digit, select '2' or '3' as applicable as the PPDC.
• Apply the CO Hardware Template to the CO ports. To do this,
• Click CO Parameters to open the page.
• Enter the number of the template you prepared (Template 02) in the field CO Hardware Template for
each port you want to assign this template.
• For Trunks to be assigned PPDC Count 0, retain CO Hardware Template Number 01.
• Log out or continue programming.
Configuring Pre-PSTN Digit Count (PPDC) using Telephone


To program PPDC Count in CO Hardware Template:
• Dial command 5902-1-Template Number-08-Code
Where,
Template Number is the CO Hardware Template, from 01 to 50. Default: 01
38 is the Feature Number for PPDC
Code is the PPDC Count, from 0 to 6.
For instance: To program PPDC

'1' in Template Number 02, dial: 5902-1-02-38-1
'2' in Template Number 03, dial: 5902-1-03-38-2
To assign the CO Hardware Template now programmed with the PPDC to trunks, dial:
• 5903-1-CO-Template Number to apply the template on a single CO trunk port.
• 5903-2-CO-CO-Template Number to apply the same template on a range of CO trunk ports.
• 5903-*-Template Number to apply the same template on all CO trunk ports.
Where,
CO is the number of the Software port of CO Trunks, from 001 to 128.
Template Number is CO Hardware Template number programmed with the PPDC, from 01 to 50.
For instance: To apply Template Number 02 to CO 003 to 005, dial 5903-2-003-005-02.
• Exit SE Mode.
Also, refer the topic “CO Hardware Template” to know more.

Building Intercom
What's this?
SARVAM UCS offers the Building Intercom solution by integrating Telecom and Security systems, for commercial
and residential buildings, such as malls, shopping complexes, residential apartment blocks and gated-
communities.
With the Building Intercom solution, you can also connect private networks, that is few Systems with the intercom
application can be connected to each other using VoIP.

To use the Building Intercom solution, the Intercom Line Card must be installed in the ETERNITY LENX/MENX/
GENX to connect extension phones in the building.
The Intercom Line Card is available in the following configuration.
ETERNITY LE Card ILC48 48-port card to connect 48 Single Line Telephones
ETERNITY GE CARD ILC20 20-port card to connect 20 Single Line Telephones
For installation instructions, under Installing ETERNITY LENX, see “The Intercom Line Card”, Installing ETERNITY
MENX, see “The Intercom Line Card”, Installing ETERNITY GENX, see “The Intercom Line Card”.
Caution: When installing ETERNITY LENX/MENX/GENX for Building Intercom, you must install the
Intercom Line Cards. The Vocoder and the VMS Modules can also be installed in the system. Make sure
other Trunk Cards or Extension or Combination Cards are not present in the system, else the Intercom
Line Card will not function.
Building Intercom Features

When used as Building Intercom, SARVAM UCS supports the following features and facilities:
• Alarms
• Auto Call Back
• Call Follow Me
• Call Forward
• Call Forward Remote
• Call Progress Tones
• Calling Line Identification and Presentation (CLIP)
• Calling Line Identification Restriction (CLIR)
• Communication Port
• Conference 3-Party, Multiparty
• Communication Port
• Conflict Dialing
• Customer Name
• Distinctive Rings
• Do Not Disturb

• Do Not Disturb (Remote)
• Field Programmable (Onsite upgradation)
• Flash Timer
• Flexible Numbers
• Help Desk
• Hot Desking
• Hotline
• Internal Call
• Internal call Restriction
• Interrupt Request
• Last Caller Recall
• Meet Me Paging
• Music On Hold
• Name Programming for Extensions
• Programmable Call Progress Tones and rings
• Programming the System — using SLT, DKP, Ethernet (LAN/WAN) Port
• Real Time Clock
• Remote Alarm
• Remote Call Forward
• Remote Programming
• Security Dialing and Reporting
• Self Ring Test
• Station In-Service/Out-service
• Station Message Detail Recording (internal calls made/received)
• System Activity Log and Display
• System Debug
• System Fault Log and Display
• System Administrator Mode
• System Security (Passwords)
• Time Tables
• Upgrading the Software
• User Security (User Password)
• Voice Help
• Voice Prompts for Tones
• Walk-In Class of Service
• Voice Mail System
• Web-based Programming
How to use
Building Intercom has two applications:
• Standalone Application
• Extended Application

Standalone Application
In this case the system is installed at the site where internal communication is required between internal members
in a same complex only.
Block A
System installed
in Block A
4051 4001
4052 4002
4053 4003
4100 4050
Here Block A is a residential complex with 100 flats. SARVAM UCS with ILC Cards is installed for internal
communication between the residents.
All the extensions 4001 to 4100 can communicate with each other.
Extended Application
Let us understand how to use the Building Intercom Application with the following illustration:
A residential society has three buildings. Each building has 50 flats. In each building a System (having Intercom
Line Cards) is installed. The residents in the same building only, can communicate with each other.
The residents of all the three buildings want to communication with each other.
To overcome this, you must install a Vocoder Module in each system. All the three buildings will be connected
using the IP network as shown in the figure below.
For installing the Vocoder Module in:

• ETERNITY LENX, see “Installing the VOCODER Module”.
• ETERNITY MENX, see “Installing the VOCODER Module”.
• ETERNITY GENX, see “Installing the VOCODER Module”.

In this case, System A, System B and System C will be connected to form a single group.The residents in Wing 1,
2 and 3 will be able to call each other using extension numbers.
Wing 1 Wing 2
2001 3001
2002 3002
2003 3003
System A System B
192.168.1.1 192.168.1.2 3050

2050
IP Network
Wing 3
192.168.1.3
4001
4002
4003
System C
4050
Intercom calls can be made between Wing 1, 2 and 3 with suitable configurations of the System.
• At each location, you need to do the following configuration in the System:
• Select a SIP trunk to be used for this application and enable it. For example, SIP Trunk 1.
• Set the SIP Trunk Mode of this trunk to Peer-to-Peer.
• In Treat Incoming call as select Station.
• Keep the SIP ID field of this SIP trunk blank.
For detailed instruction, see “Configuring SIP Extensions”.
• In System-A, to route the calls configure the following:

• The CUG table
• The Peer-to-Peer table.
To configure the CUG table in System A,
• In the Route Code field of the CUG table, enter the Number that will be dialed to call the users of System
B and C. In this case, 3 and 4 respectively. As the system uses the best match logic to match number
strings in the CUG table, you may configure only the prefix of the number to be dialed, instead of
configuring the complete number string.
• For the number you entered, in the Dialed Digit Count field, enter 4.
• Select the OG Trunk Bundle Group. Configure SIP Trunk 1 as the only member in this group. The calls will be
routed through this SIP Trunk only.

• Configure the Strip Digit Count as 0 and clear the Self Route check box.
The CUG table you configure in System A would look like this:
OG Trunk
Strip Digit Dialed Digit
Index Route Code Bundle Self Route
Count Count
Group
1 3 01 0 4
2 4 01 0 4
For detailed instructions, see “Closed User Group (CUG)”.
• To configure the Peer-to-Peer table in System A,
• In the Number field of the Peer-to-Peer table, enter the numbers of the extension users of System B
and C. In this case, 3 and 4, respectively. As the system uses the best match logic to match number
strings in the Peer-to-Peer table, you may configure only the prefix of the number to be dialed, instead
of configuring the complete number string.
• For the number 3001 enter the Domain Address, enter the Domain Name/IP Address of System B. In
this case, 192.168.1.2 and for 4001 the Domain Address is 192.168.1.3
• You can assign a Name to the numbers you have configured.
• The SIP messages can be transported using UDP, TCP or TLS. Select the Default Transport for
Outgoing Message as per your requirement for each index.
The Peer-to-Peer table you configure in System A would look like this:
Domain Default Transport for

Index Number Name
Address Outgoing Message
1 No Match
Found
2 3 192.168.1.2 System B TCP
3 4 192.168.1.3 System C TCP
• When 2001 from System A dials 3001, the system compares it with the CUG table configured in
System A. When a match is found in the CUG table, the system uses SIP Trunk 1 to route the call.
• SIP Trunk 1 is configured as a Peer-to-Peer trunk, hence the system will check the Peer-to-Peer table.
• As 3001 is configured in the Peer-to-Peer table, the system fetches Destination address and transports
the SIP messages using the protocol selected as the Default Transport for Outgoing Message.
• The system places the call on System B, using SIP Trunk 1.

• When there is an incoming call on System B, the system checks the CUG table first and as 3001 is not
programmed in the CUG table, it checks the flexible number of the extensions.
• As 3001 is found in the flexible number list, the call is routed to the extension 3001.
When 2001 dials 4001, the call will be routed as per the above logic.
Similarly you must configure the parameters in System B and C.
In certain cases different building may have the same extension numbers, in these cases you must configure the
CUG Table with Exchange ID. For detailed information, see “Closed User Group-With Exchange ID”.

Busy Lamp Field for Trunks
What is this?
On SIP extensions, SARVAM UCS supports the feature Busy Lamp Field (BLF) Subscription for Trunks, enabling SIP
extension users to monitor the status of desired trunks.
Using BLF, SIP extension users can monitor the status of different Trunk types, namely CO, BRI, T1E1, Mobile and SIP. In
the case of T1E1 and BRI Trunks, the state of calls on each channel can be monitored using BLF.
To provide SIP extension users this feature, you must enable this feature on their SIP phones and configure the trunk to
be monitored on the BLF key of their SIP phones. The number of trunks that can be monitored will depend on the number
of BLF keys supported on the SIP Phones.
With BLF subscription and BLF key configured, whenever, there is a change in the state of the monitored trunk,
SARVAM UCS sends a NOTIFY message to the SIP Extension. The NOTIFY message contains the Call State. On
receiving the NOTIFY message, the SIP Extension updates the LED indication of the BLF key on the SIP phone.
The SIP extensions will indicate the following calls states for the outgoing and incoming calls on the monitored
trunks:
Outgoing Calls
Call State Description
Trying When an outgoing call is made through the monitored trunk.
When the external party answers the call and speech is established with the
Confirmed
extension user, that is, the call is matured.
Hold When the call on the trunk has been put on hold by the extension.
Available/Idle When the SIP EXtension user disconnects the call.
Incoming Calls
Call State Description
Early When an indication is received from the Network that the external party is ringing.
When the incoming call is placed on the SIP Extension as the destination and
Confirmed
speech is established with the extension user, that is, the call is matured.
Hold When the call on the trunk has been put on hold by the SIP extension.
Available/Idle When the SIP Extension user disconnects the call
SIP phones may differ in the BLF indication (LED color and cadence, text message display) they provide
for the Call States. Refer to the manufacturer’s documentation for BLF Indication supported on the SIP
phones.

How it works
• An Open SIP Phone is registered as a SIP Extension, with the extension number 3301.
• As the user of extension 3301 wants to monitor the trunk CO-001, BLF subscription is enabled on
extension 3301 and CO-001 is assigned to the BLF key on the SIP phone.
• Extension 3301 makes an outgoing call to an external number 2630555. The BLF key will indicate the
current call state of the CO-001 Trunk as “Trying” according to the LED indication supported by the SIP
phone for this call state.
If the SIP phone supports text message display for call states, each call state will be displayed on the
phone.
• When the external party answers the call, the call between the CO-001 and SIP Extension 3301 gets
matured. The BLF key will indicate the current call state of the CO Trunk as “Confirmed” according to the
LED indication supported by the SIP phone for this call state.
• When the SIP Extension 3301 disconnects the call, the SARVAM UCS will disconnect the call of external
number and the BLF key will display the call state of the CO as “Terminated” according to the LED
indication supported by the SIP phone for this call state.
Similarly, the BLF key configured on the SIP Extension 3301 will display the call states of the CO-001 trunk
for incoming calls from external numbers.
Since multiple calls can be made through a single trunk, the BLF key will indicate the status of the first call
detected by the system. When the first call is terminated, the status of the second call (if ongoing) will be
indicated. Similarly the status of all subsequent calls will indicated after the previous call is terminated.
How to configure
To provide BLF to SIP extension users, you must do the following:
• Enable Busy Lamp Field Subscription on the SIP Extensions you want to provide this feature. For
instructions, see “Configuring SIP Extension using Jeeves” under “Configuring SIP Extensions”.
• Assign a BLF Key for the trunk to be monitored on the SIP Phones registered as extensions. For
instructions refer to the manufacturer’s documentation (Installation Guide/User Guide) for the respective
SIP Phones.
• To monitor the trunks, configure the BLF Key as per the table given below:
Trunk User ID part in SUBSCRIBE Remarks
CO COxxx Here, xxx should be a valid CO port number, that is, from 001
to 128.
BRI BRIxxCHyy Here, xx should be a valid BRI port number, that is, from 01 to
32 and the yy should be a valid channel no., that is, 01 and 02
T1E1 T1E1xxCHyy Here, xx should be a valid T1E1 port number, that is, from 01
to 08 and the yy should be a valid channel number, that is, 01
to 30.

Trunk User ID part in SUBSCRIBE Remarks
Mobile MOBxxx Here, xxx should be a valid Mobile port number, that is, from
001 to 064.
SIP SIPxx Here, xx should be a valid SIP Trunk number, that is, from 01
to 32.
E&M ENMxxx Here, xxx should be a valid ENM port number, that is, from 001
to 128.

Call Back on Trunk Ports
What's this?
The feature Call Back on Trunk Ports is used to respond to missed calls from particular numbers on the different
trunk ports of SARVAM UCS: SIP Trunks, T1E1PRI trunks, BRI trunks, Mobile trunks and CO Trunks.
When Call Back feature is enabled on a trunk port, and there is a missed call on that trunk port, the SARVAM UCS
determines if the calling number is eligible for a call back or not. It calls back the same number or an alternative
number programmed for that number, either from the port on which it was received or from a different port,
depending on the programming. SARVAM UCS can be programmed to choose the most cost effective line to call
back the missed call numbers.
Employees at remote locations can use this feature to have the SARVAM UCS installed in their office call them
back, thereby saving on charges (for example, roaming charges on mobile calls), where applicable.
How it works
For this feature to work:
• Call Back must be enabled on the desired Trunk Ports.
• The CLI of those callers whom the system should call back must be programmed in the ‘Call Back
Incoming Number List’.
• The ‘Call Back Timer’ may be programmed. When the caller disconnects within the Call Back Timer, the
Call Back will be applied for that number.
• You must define ‘Call Back on’, that is, you must select whether the number which must be called back
should be the same CLI number which the call was received or an alternative number.
• The number on which call back is to be made must be programmed in the ‘Call Back Outgoing Number
List’, if it is not the same CLI number or if it is an alternative number.
• You must select whether the call back should be made using the same trunk port on which the call was
received or an Outgoing Trunk Bundle Group (OGTBG). If you select OGTBG, you must also program the
OGTBG.
• You may enable Least Cost Routing (LCR) on the OGTB if you want the system to select the least cost
trunk for calling back the missed call number. Program LCR accordingly.
• Select a ‘Call Back Mode’, that is, how the call should be routed when the call back is answered by the
remote party; whether it should be routed through Built-In Auto Attendant, DISA or Operator.
Following is an example of a Call Back on a mobile port, when the above parameters are programmed.
• Caller A calls mobile port 01.

• The system checks if the Call Back flag is enabled on mobile port 01.
• The flag is enabled.

• The system matches the CLI of A with the Call Back Incoming Number List assigned to mobile port 01 to
determine if the calling number is eligible for a call back.
• A match is found on Index 15 of the Call Back Incoming Number List.215
• The system waits for the period of the Call Back Timer (programmable, default: 10 seconds).
• A must disconnect before the expiry of the Call Back Timer so that the system can treat it as a Missed Call.
• If A disconnects within the Call Back Timer, the system applies Call Back for A’s number.
• The system checks the ‘Call Back on’ parameter, whether it has to call back the same number or an
alternative number.
• If an alternative number is programmed as ‘Call Back on’, the system checks the Outgoing Call Back
Number List for the alternative number. As the CLI of A matches with the number on Index 15 of the Call
Back Incoming Number List, the system checks Index 15 of the Call Back Outgoing Number List for the
corresponding alternative number to this number.
• The system checks if the number is to be called from the same port or an OGTBG.
• If the same port is programmed, the system will make a call to the number using mobile port 01.
• If OGTBG is programmed, the system will check if Least Cost Routing is enabled in the OGTBG and make
the call back accordingly.
• When A answers the call,
• The system checks the type of Call Back Mode enabled on mobile port 01 (the port on which the call back
request was made).
Four scenarios are possible:
1. “Auto Attendant” is enabled as Call Back Mode on mobile port 01.
• A gets dial tone of SARVAM UCS.
• A can now use Built-In Auto Attendant feature.
2. 'Pin Authentication - Multiple Calls' or 'CLI Authentication - Multiple Calls' is enabled as Call Back Mode
on mobile port 01.
• A gets dial tone of SARVAM UCS.
• A can now reach any station or trunk of SARVAM UCS from DISA Mode.
3. 'CLI Authentication - Single Call Answer Signaling' is enabled as Call Back Mode on mobile port 01.
• A gets dial tone of another trunk of SARVAM UCS.
215. If the system does not find a match for the CLI of the caller in the Call Back Incoming Number List, the 'Call
Back' feature will not be applicable and the call will be processed according to the normal incoming call logic.

• A can make calls from the trunk.
4. ‘Operator’ is enabled as Call Back Mode on mobile port 01.
• A gets Ring Back Tone.
• The system lands the call on the Operator extension assigned to mobile port 01.
Read the topics “Auto Attendant”, “Direct Inward System Access (DISA)” and “Configuring 'Operator'” to
know more about the call respective call logic.
• Since this feature is essentially for callers, they must be aware of its functioning to be able to use it, that
is, disconnect the call within the Call Back Timer. If the caller does not disconnect within the Call Back
Timer, the call will be processed according to the normal incoming call logic.
• SARVAM UCS supports only one call back request at a time, for one trunk port. The second incoming
call on that trunk port will be processed by the system as per normal incoming call routing.
• For call back requests made from an OGTBG, if any of its trunks is busy, SARVAM UCS will support
only the last call back request in the OGTBG. Previous requests will be processed as per the normal
incoming call management logic.
How to configure
For this feature to function, you must program the following parameters on each Trunk port type (CO, BRI, T1, E1,
Mobile, SIP) on which you want to use this feature:
• Enable Call Back: This flag must be enabled on the desired trunk port on which you want to activate the
Call Back on Trunk Port feature. By default, this flag is disabled on all trunk port types.
• Call Back Timer: This is the duration for which the system waits for the caller to disconnect the call after
the system has found a matching number for the caller’s CLI in the Call Back Incoming Number List.
When the caller disconnects within Call Back Timer, the system applies Call Back on the port. If the caller
does not disconnect within the Call Back Timer, the incoming call management logic is applied for the call
on the trunk port.
The range of this timer is from 01 to 99 seconds. By default, it is set to 10 seconds.
• Call Back Incoming Number List: This is the list of numbers that are eligible for Call Back. The system
checks the CLI of the caller with this list to determine if the caller is eligible for a call back.
The system compares the number string programmed in the Call Back Incoming List with the number
string received as CLI.
Number string programmed in the 'Call Back Incoming Number List' shall be compared with the actual
received CLI.
The number string programmed in the Call Back Incoming Number List may be shorter than the number
string received as CLI, but only if the programmed number string completely matches with the received
CLI from the right towards left, the system will consider it as a complete match.

For example, if the programmed string is 263055 and the number string received in the CLI is
2652630555, the system will consider it a complete match. If the received CLI 912652630555, the system
will consider this caller too as eligible for a call back. Thus any CLI received with 263055 as the last 7 digits
will be considered as match found.
By default, ‘Number List’ 15 is assigned to all trunk port types as Call Back Incoming Number List. You
may program this list for all port types, or you may program another Number List and assign it to the
particular trunk port type.
Refer the topic “Number Lists” for instructions on how to configure the Number List.
• Call Back on: For each Trunk port type you have set the Call Back feature, you must define ‘Call Back on’,
that is, you must select whether the number which must be called back should be the same number from
which the call was received or a different number.
When missed call is eligible for call back (matches with Incoming Number list), the 'Call Back on'
parameter determines the number on which the call back is to be made, that is, whether on the same
number from which the missed call is received or on a different number.
In countries where CLI received on trunks can be dialed out without any modification, you may select ‘CLI
Number’ as ‘Call Back on’ option.
In countries where CLI received on trunks can be dialed only after appropriate modification, you may
select “Alternate Number’ as the ‘Call Back on’ option. You may also select ‘Alternate Number’ as Call
Back on when you want the call back to be made to a different number.
• Call Back Outgoing Number List: When the system finds a missed call eligible for a call back, it will make
the call back on the basis of the Call Back on option you selected and the Outgoing Number List you
programmed.
If you selected ‘CLI Number’ as “Call Back on’ option, you do not need to program the corresponding
outgoing number for the CLI received.
However, if the CLI received needs to be modified before being dialed out, then program the modified CLI
in the Outgoing List as the corresponding outgoing number for the CLI received.
The modified CLI or the Alternate number should be programmed at the same index number as the index
number at with the received CLI is programmed in the Call Back Incoming Number List. For example, for
the received CLI number string programmed at Index 15 in the Call Back Incoming Number List, the
corresponding modified CLI/Alternate number string should be programmed at the same Index, 15, in the
Call Back Outgoing Number List.
When the CLI received matches with the number string programmed at Index 15 of the 'Call Back
Incoming Number List', the call back will be made using the (modified/Alternate) number programmed at
Index 15 of the 'Call Back Outgoing Number List'.
By default, ‘Number List’ 16 is assigned to all trunk port types as Call Back Outgoing Number List. You
may program this list for all port types, or you may program another Number List and assign it to the
particular trunk port type.
Refer the topic “Number Lists” for instructions on how to configure the Number List.

If you have selected ‘Alternate Number’ as ‘Call Back on’ option, but do not want to provide alternative
numbers to call back particular callers (that is, CLI received), in such a case, program the CLI of these
callers in the Incoming Number List but keep the corresponding index numbers in the Outgoing Number
Lists blank.
• Call Back from: This parameter determines the trunk port to be used to make call back.The call back can
be made using the same port or an Outgoing Trunk Bundle Group (OTGTBG). Select ‘Same port’ if you
want the call back to be made using the same port on which the missed call was received. If you select
OGTBG, the call back will be made using the OGTBG, which you have defined.
• OGTB Group for Call Back: If you selected OGTBG for making the call back in the previous parameter,
you must assign the OGTBG that must be used in this parameter.
By default, OGTBG 01 is selected for Call Back.
If you want the system to select the lowest cost trunk for making the call back, enable Least Cost Routing
on the OGTBG that you define here for Call Back.
should be routed:
• Built-In Auto Attendant: The system will process the call as per the Built-In Auto Attendant call logic -
give a dial tone to the remote party, who can now call any extension. Refer the feature description for
“Auto Attendant”.
• PIN Authentication-Multiple Calls: The system will process the call as per DISA call logic - allow
remote party to enter DISA mode with PIN-Authentication. On successful authentication (DISA Login)
the user is allowed to make calls or use features as allowed to him/her.
• CLI Authentication-Multiple Calls: The system will process the call as per DISA call logic, allowing
the remote party to enter DISA mode with CLI Authentication-Multiple calls as authentication method
and level of access.
• CLI Authentication-Single Call: The system will process the call as per DISA call logic, allowing the
remote party to enter DISA mode with CLI Authentication-Single call as authentication method and
level of access. Refer the feature description for “Direct Inward System Access (DISA)”.
• Operator: When the remote party answers the Call Back call, the system will route the call to the
Operator216.
All these parameters may be programmed using Jeeves or by dialing SE commands from a telephone.
To program Call Back on different port types, refer the relevant topics mentioned below:
• For Call Back on SIP Trunks, refer the topic “Configuring SIP Trunks”.
• For Call Back on T1E1 Ports, refer the topic ‘Call Back on T1E1 Trunk Ports’, under “Configuring E1
Trunks” and “Configuring T1 Trunks”.
• For Call Back on BRI Ports, refer the topic ‘BRI Parameters’ under “Configuring BRI Trunks”.
216. 'Operator' is the station which is assigned to the Mobile port in the Trunk Feature Template. Refer “Trunk Feature Template” to
know more.

• For Call Back on Mobile Ports, refer the topic “Configuring Mobile Trunks”.
• For Call Back on CO Ports, refer the topic “Configuring CO Trunks”.

Call Budget on Extension
What's this?
Call Budget is a cost control feature that allows you to keep a tab on the total cost of phone call made by extension
users.
With this feature, each extension can be allotted a 'budget' limit for outgoing calls, which is automatically reloaded
at the start of every month.
Long distance calls form a major part of the increased cost of telephone calls. Though excessive use or misuse of
long distance dialing can be restricted using Toll Control, there may be extension users whose nature of work
requires them to make long distance calls. Instead of denying them the facility, their telephone bill can be limited to
a certain amount using Call Budget.
With a Call Budget allotted to the extension, the user is free to make calls as long as s/he does not cross the budget
limit. Once the user exceeds the budget limit, the extension can be denied access to long distance dialing.
The extension user can be assigned a fresh budget, after which s/he can resume making long distance calls.
Call Budget can be enabled on all the extensions as well as on selected extensions. Each extension can be
assigned a different amount depending on user requirement.
How it works
When an extension allotted Call Budget makes a call,
• The system checks the current call budget amount of the extension.
• If the consumed amount is within the budget limit allotted to the extension,
• The system allows the extension to make the call as per the “Toll Control Levels” assigned to it.
• After the call ends, the system calculates and adds the call amount to the extension's account. Thus it
calculates and updates the total cost of calls made from the phone.
• If the consumed amount exceeds the budget limit allotted to the extension,
• The system considers this as Call Budget exhausted.
• The system allows the extension to make the call as per the Toll Control-Call Budget Consumed
assigned to the extension.
• After the call ends, the system calculates and adds the call amount to the extension's account.
• Until a new Call Budget is allocated to the extension user, the extension user can make calls only as per
Toll Control assigned for the Call Budget Consumed state.
• Once a new Call Budget is allocated, the extension user can make calls as per the “Toll Control” assigned
to the extension.

• If the budget exceeds anytime during the month, and if no fresh budget amount is allotted, the system
allows calls to be made as per the Allowed and Denied List of Toll Control-Call Budget Consumed till the
end of the month. From the 1st day of the following month, the system automatically reloads the budget
amount. The extension can now make calls.
• The Call Budget allotted to extension is valid for one month. The system automatically reloads the budget
at the start of every month.
• The budget amount can be changed or allotted afresh to extensions from the System Administrator (SA)
mode, at any time. The Call Budget allotted by the SA will be reloaded in the following month.
• Call Budget is not based on real time (online) call cost calculation. The SARVAM UCS calculates the
call cost only after the call has ended.
• So, if the Call Budget allotted to an extension user gets exhausted in the middle of a call, the call will
not get disconnected, though the budget exceeds. To prevent this from occurring, the System Engineer
may program the “Call Duration Control (CDC)” feature.
• Call Budget is dependent on precise Call Cost Calculation. So, SMDR parameters and long distance
codes must be programmed properly to prevent errors in calculation.
• This feature works independent of any Call Accounting Software (CAS) installed with the SARVAM
UCS.
• The SARVAM UCS will calculate cost of phone calls made by extension phones even when no call
budget is allocated217.
How to configure
The working of this feature is controlled by three parameters: Call Budget flag, Toll Control-Call Budget
Consumed and Preset Call Budget Amount (this parameter is applicable only for the Hotel Mode).
These parameters can be programmed using Jeeves and Telephone.
Call Budget flag and Toll Control-Call Budget Consumed

To enable Call Budget feature on an extension, the System Engineer must enable the Call Budget flag and define
the Toll Control-Call Budget Consumed in the Station Basic Feature Template assigned to the extensions.
In the default Station Basic Feature Template 01 assigned to all extensions of the SARVAM UCS, the Call Budget
flag is disabled and the Toll Control-Call Budget Consumed is set to 'No Calls'.
If Call Budget is to be allowed to all extensions, simply enable the flag in the default Station Basic Feature Template
01 and select the Toll Control for Call Budget Consumed state.
However, if Call Budget is to be allowed to selected extensions, then prepare a separate Station Basic Feature
Template with the Call Budget flag enabled and the Toll Control-Call Budget Consumed set. Now, apply this
template on extensions that are to be allowed this feature.
Refer the topic “Station Basic Feature Template” for detailed instructions for programming a feature in the template
and assigning templates to extensions.
217. Based on the feature 'Call Cost Calculation'.

Preset Call Budget Amount218
The amount of '9999' is set as default Call Budget in the system. This value may be changed by the System
Engineer according to the organization's practices. For example, if the organization wants to allocate a fixed
amount of $10 to all extension users, the Call Budget value can be set to '10'.
The new Call Budget set by the System Engineer will be considered as the Preset Call Budget amount. This
amount will be allocated at the start of every month to all extensions having Call Budget feature in their Station
Basic Feature Template.
Further, the System Administrator (SA) can override the Preset Call Budget amount set by the System Engineer,
and allot call budgets on an extension-by-extension basis. For example: allotting higher amount to extensions of
senior managers, Marketing, Sales, Exports departments, and lower amount to extensions that are less likely to
make long distance calls frequently.
The amount may be greater or lesser than the default amount set by the System Engineer. The Call Budget amount
allotted by the SA will be reloaded at the start of every month on the extension. For instructions refer 'How to Use'
later in this section.
Changing the Preset Call Budget Amount using Jeeves

• Under Configuration, click Hotel Settings.
• Click Hotel Parameters.
• Scroll to Preset Call Privilege and change the Preset Call Budget Amount to the required value.
Changing Preset Call Budget Amount using a Telephone

• Dial command 3710-Preset Call Budget Amount
Where,
Preset Call Budget Amount is any amount of 6-digits.
If the amount is less than 6 digits, use leading zeros.
For example: To assign Call Budget = $10, dial 3710-000010
• Exit SE mode.
218. Applicable only for the Hotel Mode.

• Dial '00' to exit from an SLT
• Dial '00' and press 'Enter' key to exit from a DKP.
• The amount programmed as Preset Call Budget is to be considered as the local currency.
• At the time of installation, when the SE selects the Region Code (country code) and defaults the
system, the related Currency Code is applied.
• The currency symbol will not be displayed on the Operator's phone, on account of the limited number of
characters that can be displayed.
• The local currency symbol will appear at the relevant places in the outgoing SMDR reports.
How to use
Call Budget amount can be allotted to extensions from the System Administrator mode, using Jeeves or by dialing
SA commands from an extension phone.
Assigning Call Budget to Extensions using Jeeves

• Log into Jeeves as System Administrator.
• Click Extension.
• Now, Search Extension on which you want to set this feature. You search by entering either Extension
Number or the Extension Name.
• Click Submit.
• The searched extension user details appear on your screen.

• Click Phone Properties to expand.
• In Allot Call Budget, enter the amount you want to assign to the user as budget limit for outgoing calls.
To re-assigning a new amount before the previous balance is consumed, make sure you add the
available balance to the new amount. Enter this amount in Allot a Call Budget.
For example, if you have allotted an amount is Rs.1000 and the consumed amount is Rs.600. The
available balance is Rs.400. Now, if you want to assign a new amount of Rs.500. In Allot a Call Budget
you must enter 900 (Balance + New = 400 + 500).
• The Allotted Amount/Used displays the amount allotted to the user as well as the call budget amount
consumed by the user for making outgoing calls.
• Click Submit button to save.
• To allot call budget to another extension, follow the same instructions as above.

To assign Call Budget to an Extension:
• Press DSS Key assigned to ‘Set Call Budget for Remote Extension' function.
OR
• Dial 1072-004 (from SA mode).

• Enter Extension Number that is to the assigned a budget.
• Enter Call Budget Amount.
Use leading zero if amount is fewer than 6 digits.
• You get confirmatory message.
• Go idle.
To view Call Budget assigned to an Extension:
• Press DSS Key assigned to the ‘Call Budget Extension’ function.
OR

• Enter Extension Number
• Call Budget assigned to the extension appears on your phone display.
For SLT Users

• Lift the handset.
• Dial Extension Number.
• Dial Call Budget Amount.
Use leading zero if amount is less than 6 digits.
• Replace handset.
Call Budget cannot be viewed on SLT.

Call Budget on Trunk
What's this?
Call Budget on Trunks is an expense control feature of SARVAM UCS that allows you to keep track of the cost of
phone calls made from the different Trunk ports of SARVAM UCS.
With this feature, each trunk can be allotted a 'budget' limit for outgoing calls. This budget limit can be programmed
to be reloaded manually each time it is exceeded or at a scheduled date, either daily or at a particular date of the
month.
There are three types of Call Budget limit that can be set on the trunks:
• Amount: In this type of Call Budget, a fixed amount is assigned to the trunk. By default the amount of
999999 (to be considered in the local currency) is set as Call Budget Amount on trunks. With Amount-
based Call Budget you can control the actual expense incurred on making calls from a trunk.
• Minutes: In this type of Call Budget, a fixed number of Minutes are assigned to the trunk. By default,
999999 minutes are assigned as Call Budget Minutes on trunks. This type of Call Budget is useful when
the Service Provider offers 'Free' minutes. For example, the Service Provider allows the customer to make
calls for the first 1000 minutes every month. This offer can be availed of by programming Minutes-based
Call Budget on the trunk port.
• Number of Calls: In this type of Call Budget, you can define the maximum number of calls that can be
made from a trunk. By default, the maximum number of Call Budget - Calls is set to 9999 calls on the
trunks. This type of Call Budget is useful when the Service Provider offers a certain number of free calls or
a certain number of free calls for a fixed period. For instance, the Service Provider offers 150 free calls per
month.
With a Call Budget allotted to a trunk, the users can make calls from the trunk as long as the budget limit set for the
trunk—Amount or Minutes or Maximum number of Calls—is not crossed. Once the budget limit exceeds calls will
not be routed through this trunk.
The consumed Budget can be reset, after which the trunk becomes functional again and allows outgoing calls to be
made. The consumed Call Budget can be reset manually, that is, anytime, as required/desired, or on a scheduled
date either daily or on a particular date of the month.
How it works
Call Budget can be enabled on trunk port types - SIP, T1E1PRI, BRI, Mobile, CO - all at once or on selected trunk
port types from among them. Each trunk can be assigned a different Call Budget, depending on the requirement of
the users.
When Call Budget is enabled on a trunk port, for each outgoing call,
• The system checks the type of Call Budget set on the trunk - Amount, Minutes or number of Calls.
• It checks the Call Budget consumed.

Call Budget- Amount
• When Amount-based Call Budget is selected, the Amount should be assigned to the trunk.
• At the end of each outgoing call made from the trunk, the system will calculate the cost of the call on the
basis of the Pulse Rate Type programmed. The system will thus calculate the total amount consumed after
the end of each call. Refer the topic “Call Cost Calculation (CCC)” to know more.
Call Budget - Minutes

• When Minutes-based Call Budget is set, the total minutes for which calls will be allowed from the trunk port
must be defined.
• With the number of Minutes defined, at the end of each call, the system will calculate the duration of the
call on the basis of the units programmed in the Pulse Rate. The system will calculate the consumed
minute on the basis of the duration of the call. Refer the topic “Call Cost Calculation (CCC)” to know more.
Call Budget - Number of Calls

• When the Call Budget is based on 'Number of Calls', the maximum number of calls to be allowed from the
trunk port is to be defined.
• With the number of calls programmed, the system will maintain a count for the number of matured
outgoing calls made from that trunk port.
• Thus for each matured call, the Number of Calls-Count is incremented, irrespective of the actual duration
of the matured call.
When the assigned 'cost' or 'minutes' or 'number of calls' assigned to trunk is exhausted, SARVAM UCS will:
• print 'system activity log'.

• bar or limit outgoing calls from such trunks.
• play an Error Tone to the extension users who attempt to access such trunks using Selective Trunk
Access.
• However, incoming calls will remain unaffected, and will be allowed on these trunks.
The consumed Call Budget Amount/Minutes/Calls can be reset manually at any time from the System
Administrator mode or the System Engineer mode or can be programmed to be automatically reset either daily or
on a particular date of the month.
The current Call Budget Amount/Minutes/Calls limit can be changed from the System Administrator (SA) mode, at
any time. If scheduled reset of consumed Call Budget is programmed, then the Call Budget allotted by the SA will
be reloaded on the scheduled date.
Once a new Call Budget is allocated to the trunk, outgoing call facility is resumed on the trunk.
• Call Budget on Trunks is not based on real time (online) call cost calculation. The SARVAM UCS
calculates the call cost only after the call has ended.
• If the Call Budget allotted to a Trunk Port gets exhausted in the middle of a call, the call will not
disconnected, though the budget is exceeded.
• Call Budget on Trunks is dependent on precise Call Cost Calculation. So, SMDR parameters and long
distance codes must be programmed properly to prevent errors in calculation.

• This feature works independent of any Call Accounting Software (CAS) installed with the SARVAM
UCS.
• The SARVAM UCS will calculate cost of phone calls made by the trunks even when no call budget is
allocated219.
How to configure
Call Budget on Trunks is to be programmed in the Trunk Port Parameters of the trunk type on which you want to
enable this feature. This can be done using Jeeves as well as a telephone
Programming Call Budget on Trunks using Jeeves

• Call Budget parameters must be programmed in the Port Parameters page of the trunk type you want to
configure. For instance, to configure Call Budget on a CO Trunk,
• Under Configuration, click CO Configuration link.
• Click CO Parameters to open the page.
219. Based on the feature 'Call Cost Calculation'.

• Click the Advance button and configure the following parameters:
• Call Budget: If you want to enable Call Budget on Trunk feature, configure the following parameters for
this CO trunk port:
• Type: Select the type of Call Budget on Trunk, that is, Amount or Minutes or Calls to be applied on this
CO trunk port. By default, no Call Budget type is selected.
so.
• You may program the same Call Budget parameters as listed above for other trunk types:
• Click VoIP Configuration, click SIP Trunk Parameters to program Call Budget on SIP trunks. Click
the Advance button on this page to reach Call Budget parameters.
• Click Port Parameters under T1E1 Configuration to program Call Budget on T1E1PRI trunks. Click
the Advance button on this page to reach Call Budget parameters.

• Click BRI Parameters under BRI Configuration to program Call Budget on BRI trunks. Click the
Advance button on this page to reach Call Budget parameters.
• Click Mobile Port Parameters under Mobile Configuration to program Call Budget on Mobile Ports.
Click the Advance button on this page to reach Call Budget parameters.
• You may log out of Jeeves.
• The consumed Call Budget on trunk can be reset from the System Engineer mode as well as the
System Administrator mode manually at any time, referred to as Manual Reset.
• Manual Reset of Call Budget on Trunks by the System Engineer can be done either from Jeeves or
using a Telephone.
Manual Reset of Call Budget on Trunk using Jeeves

• You may perform manual reset of the consumed Call Budget on the above listed trunk types from the
Status page of each of these trunk types.
• Click the Status page under CO Configuration. Select the Reset Consumed Amount/Minutes/Calls
check box of the CO port for which you want to reset the consumed Call Budget.
• Similarly, click Status page under VoIP Configuration and enable the parameter Reset Consumed
Amount/Minutes/Calls to manually reset consumed Call Budget on the desired SIP trunks.
• Click Status page under T1E1 Configuration to enable the same parameter Reset Consumed
Amount/Minutes/Calls of the T1E1 port for which you want to reset the consumed Call Budget.
• Click Status page under BRI Configuration, and enable the parameter Reset Consumed Amount/
Minutes/Calls for the BRI port for which you want to reset the consumed Call Budget.
• Click Status page under Mobile Configuration to enable the same parameter Reset Consumed
Amount/Minutes/Calls of the Mobile port for which you want to reset the consumed Call Budget.

Similarly, you can reset Call Budget on Trunks from the SA mode.
Programming Call Budget on Trunks using a Telephone

• Enter SE mode.
Call Budget on CO Trunks

To program Call Budget Type on CO, dial:
• 3301-1-CO-Budget Type to select Call Budget Type for a single trunk port.
• 3301-2-CO-CO-Budget Type to select the same Call Budget Type for a range of trunk ports.
• 3301-*-Budget Type to select the same Call Budget Type for all trunk ports.
Where,
Budget Type
0 for None
1 for Amount
2 for Minutes
To program Call Budget Amount on CO, dial:

• 3302-1-CO-Budget Amount to program amount for a single trunk port.
• 3302-2-CO-CO-Budget Amount to program the same amount for a range of trunk ports.
• 3302-*-Budget Amount to program the same amount for all trunk ports.
Where,
Budget Amount is of 6 digits max. Use leading zeros if amount to be programmed has fewer than 6
digits.
To program Call Budget Minutes on CO, dial:

• 3303-1-CO-Minutes to program minutes for a single trunk port.
• 3303-2-CO-CO-Minutes to program the same minutes for a range of trunk ports.
• 3303-*-Minutes to program the same minutes for all trunk ports.
Where,
'Minutes' is of 6 digits max. Use leading zeros if Minutes to be programmed has less than 6 digits.
To program Call Budget - Number of Calls on CO, dial:

• 3309-1-CO-Number of calls to program number of calls for a single trunk port.
• 3309-2-CO-CO-Number of calls to program the same number of calls for a range of trunk ports.
• 3309-*-Number of calls to program the same number of calls for all trunk ports.
Where,
Number of Calls is of 4 digits from 0001 to 9999. Use leading zeros if number of calls to be
programmed has fewer than 4 digits.
To program Call Budget Reset Mode for CO, dial:

• 3304-1-CO-Call Budget Reset Mode to program reset mode for a single trunk port.
• 3304-2-CO-CO-Call Budget Reset Mode to program the same reset mode for a range of trunk ports.
• 3304-*-Call Budget Reset Mode to program the same reset mode for all trunk ports.

Where,
Reset Mode is
2 for Manual reset
To program the Date for Scheduled Reset mode, dial:

• 3305-1-CO-Date to program date for a single trunk port.
• 3305-2-CO-CO-Date to program the same date for a range of trunk ports.
• 3305-*-Date to program the same date for all trunk ports
Where,
Date is
Call Budget on Mobile Trunks

To program Call Budget Type on Mobile trunk port, dial:
• 8019-1-Mobile-Budget Type to program budget type for a single trunk port.
• 8019-2-Mobile-Mobile-Budget Type to program the same budget type for a range of trunk ports.
• 8019-*-Budget Type to program the same budget type for all trunk ports.
Where,
Mobile is the number of the Mobile software port from 001 to 064.
Budget Type
0 for None
1 for Amount
2 for Minutes
To program Call Budget Amount on Mobile trunk port, dial:

• 8020-1-Mobile-Budget Amount to program amount for a single trunk port.
• 8020-2-Mobile-Mobile-Budget Amount to program the same amount for a range of trunk ports.
Where,
Budget Amount is of max. 6 digits. Use leading zeros when programming an amount with fewer than 6
digits.
By default, Budget Amount is 999999.
To program Call Budget Minutes on Mobile trunk port, dial:

• 8021-1-Mobile-Minutes to program minutes for a single trunk port.
• 8021-2-Mobile-Mobile-Minutes to program the same minutes for a range of trunk ports.
Where,
Minutes is of max. 6 digits. Use leading zeros if Minutes to be programmed has fewer than 6 digits.
By default, Free Minutes is 999999.

To program Call Budget Number of Calls on Mobile trunk port, dial:
• 8033-1-Mobile-Number of calls to program number of calls for a single trunk port.
• 8033-2-Mobile-Mobile-Number of calls to program the same number of calls for a range of trunk
ports.
• 8033-*- Number of calls to program the same number of calls for all trunk ports.
Where,
To program Call Budget Reset Mode for Mobile trunk port, dial:
• 8022-1-Mobile-Call Budget Reset Mode to program reset mode for a single trunk port.
• 8022-2-Mobile-Mobile-Call Budget Reset Mode to program the same reset mode for a range of trunk
ports.
Where,
2 for Manual reset
To program the Date for Scheduled Reset mode for Mobile trunk port, dial:
• 8023-1-Mobile-Date to program reset date for a single trunk port.
• 8023-2-Mobile-Mobile-Date to program the same reset date for a range of trunk ports.
• 8023-*-Date to program the same reset date for all trunk ports.
Where,
Date is
Call Budget on BRI Trunks

To program Call Budget Type on BRI, dial:
• 6214-1-BRI-Budget Type to program budget type for a single trunk.
• 6214-2-BRI-BRI-Budget Type to program the same budget type for a range of trunks.
• 6214-*-Budget Type to program the same budget type for all trunks.
Where,
BRI is the number of the BRI software port from 01 to 32.
Budget Type
0 for None
1 for Amount
2 for Minutes
To program Call Budget Amount on BRI, dial

• 6215-1-BRI-Budget Amount to program amount for a single trunk.
• 6215-2-BRI-BRI-Budget Amount to program the same amount for a range of trunks.
• 6215-*-Budget Amount to program the same amount for all trunks.
Where,

Budget Amount is of max. 6 digits. Use leading zeros when programming an amount with fewer than 6
digits.
By default, Budget Amount is 999999.
To program Call Budget Minutes on BRI, dial:

• 6216-1-BRI-Minutes to program minutes for a single trunk.
• 6216-2-BRI-BRI-Minutes to program the same minutes for a range of trunks.
• 6216-*-Minutes to program the same minutes for all trunks.
Where,
Minutes is of max. 6 digits. Use leading zeros when programming an amount with fewer than 6 digits.
To program Call Budget Number of Calls on BRI, dial:

• 6205-1-BRI-Number of calls to program number of calls for a single trunk.
• 6205-2-BRI-BRI-Number of calls to program the same number of calls for a range of trunks.
• 6205-*-Number of calls to program the same number of calls for all trunks.
Where,
To program Call Budget Reset Mode for BRI, dial:

• 6217-1-BRI-Reset Mode to program reset mode for a single trunk.
• 6217-2-BRI-BRI-Reset Mode to program reset mode for a range of trunks.
• 6217-*-Reset Mode to program reset mode for all trunks.
Where,
2 for Manual reset
To program the Date for Scheduled Reset mode for BRI, dial:
• 6218-1-BRI-Date to program date for a single trunk.
• 6218-2-BRI-BRI-Date to program the same date for a range or trunks.
• 6218-*-Date to program the same date for all trunks.
Where,
Date is
Call Budget on T1E1 Trunks

• 6122-1-T1E1-Budget Type to program call budget type for a single trunk port.
• 6122-2-T1E1-T1E1-Budget Type to program the same call budget type for a range of trunk ports.
• 6122-*-Budget Type to program the same call budget type for all trunk ports.
Where,
Budget Type is
0 for None

1 for Amount
2 for Minutes

• 6123-1-T1E1-Budget Amount to program call budget amount for a single trunk port.
• 6123-2-T1E1-T1E1-Budget Amount to program the same amount for a range of trunk ports.
Where,
Budget Amount is of 6 digits max. Use leading zeros if amount to be programmed has fewer than 6
digits.

• 6124-1-T1E1-Minutes to program minutes for a single trunk port.
• 6124-2-T1E1-T1E1-Minutes to program the same minutes for a range of trunk ports.
Where,

• 6125-1-T1E1-Number of calls to program number of calls for a single trunk port.
• 6125-2-T1E1-T1E1-Number of calls to program the same number of calls for a range of trunk ports.
• 6125-*-Number of calls to program the same number of calls for all trunk ports.
Where,

• 6138-1-T1E1-Call Budget Reset Mode to program reset mode for a single trunk port.
• 6138-2-T1E1-T1E1-Call Budget Reset Mode to program the same reset mode for a range of trunk
ports.
Where,
2 for Manual reset
.
• 6139-1-T1E1-Date to program date for a single trunk port.
• 6139-2-T1E1-T1E1-Date to program the date for a range of trunk ports.
• 6139-*-Date to program the date for all trunk ports.
Where,
Date is

• Exit SE mode.
Resetting Consumed Call Budget on Trunks using a Telephone

• Enter SE mode.
To manually reset consumed Call Budget of CO trunks, dial:

• 3306-1-CO to manually reset a single trunk port.
• 3306-2-CO-CO to manually reset a range of trunk ports.
• 3306-* to manually reset all trunk ports.
Where,
To manually reset consumed Call Budget of Mobile trunks, dial:

• 8024-1-Mobile to reset a single trunk port.
• 8024-2-Mobile-Mobile to reset a range of trunk ports.
• 8024-* to reset all trunk ports.
Where,
To manually reset consumed Call Budget of BRI trunks, dial:

• 6219-1-BRI to reset a single trunk.
• 6219-2-BRI-BRI to reset a range of trunks.
• 6219-* to reset all trunks.
Where,
To manually reset consumed Call Budget of T1E1 trunks, dial:

• 6140-1-T1E1 to reset a single trunk port.
• 6140-2-T1E1-T1E1 to reset a range of trunk ports.
• 6140-* to reset all trunk ports.
Where,
• Exit SE mode.

Call Chaining
What's this?
Call Chaining is when an external/internal call transferred by the Operator to another extension or external number
is made to return to the Operator's extension after the conversation between the caller and the extension/external
number to which it is transferred has ended.
Call Chaining is useful situations where the Operator intervention is required after the transferred call has ended.
For instance:
• The caller needs to take an appointment or requires some information from the Operator after talking to the
desired extension.
• A marketing executive who calls his supervisor to consult on a technical problem needs to be informed
about his travel itinerary and ticket booking by the Operator. The Operator can transfer the call to the
supervisor, and use Call Chaining to retrieve the call once the conversation has ended to give the
information to the executive.
Call Chaining can be set for multiple calls.
Also refer “Call Transfer”, “Call Park”.
How it works
• A is an External Caller
• B is an extension.
• A calls a Trunk of SARVAM UCS.

• The Operator answers the call.
• The Operator sets Call Chaining and transfers the call to B.
• B disconnects the call with A, but A is still connected.
• The call comes back to the Operator, if the Operator is free.
• Now A is in speech with the Operator.
• If A disconnects the call with B, the call will be released. It will not return to the Operator.
• If the Operator is busy, A will be played music on hold for the duration of the Call Park Release Timer.
• If the Operator is busy and the Timer elapses, the call will be released.
Call Chaining can be performed when call is transferred from any extension to another extension or external
number.
How to configure
The only programming involved in the functioning of this feature is assigning this feature to a DSS key which has
LED and programming, if necessary, the Call Park Release Timer.
Refer the topic “DSS Keys Programming” and “Call Park” for instructions.

How to use
For EON & Extended IP Phone Users

• Go OFF-Hook to answer the incoming call.
• The caller requests for an extension/trunk.
• Press DSS Key assigned to Call Chaining.
OR
• Press the ‘Transfer Key’ and dial 1050.
You get confirmation tone and the message 'Party Chained' on your phone's display. If DSS Key is used,
the LED of the key will glow.
• Press the ‘Transfer Key’.

• Caller gets on-hold music.
• Dial the requested extension/external number.

• The called party answers.
• Go ON Hook to transfer the call.
• When extension/trunk disconnects, your extension rings.
• Go OFF Hook. You get connected with the caller.
• Go Idle after the conversation ends, or you get dial tone after 3 seconds.
For SLT Users
• Answer the incoming call.

• The caller requests for an extension/trunk.
• Dial Flash-1050.
• Dial Flash.
• Caller gets on-hold music.
• Dial the requested extension/external number.
• The called party answers.
• Go On-Hook to transfer call.
• When extension/trunk disconnects, your extension rings.
• Lift handset to answer. You get connected with the caller.
• Go Idle after the conversation ends.

Call Cost Calculation (CCC)
What’s this?
SARVAM UCS can calculate the cost in amount for the external calls made by the extension users.
The call cost is calculated according to the tariffs offered by the Service Providers. Different types of tariffs are
provided by different Service Providers. These may be:
• Time Based Tariffs, for example, free calling from 11.00 p.m. to 6.00 a.m.
• Unit Based Tariffs, for example, first 60 seconds at the rate of 50 paise and additional at the rate of 10
paise.
• Special Day Tariffs, for example, subsidized calls rates on August 15.
• Tariffs according to the Geographical Distances, for example, calling rates for UAE differ from calling rates
to Canada.
Using the Call Cost Calculation feature, the system calculates the cost of the call according to the duration of the
call interpreted in terms of units and the charge applicable for the duration as per the service provider.
How it works
• you must get the tariff details from the Service Providers and configure the same.
• determine the outgoing trunk for the calls according to the type of calls, namely, local calls, national calls or
international calls.
SARVAM UCS will calculate the cost of the call as follows:
• When the call is made from a trunk, the system checks the Call Cost Calculation Pulse Rate Option, 1 to
4 assigned to the trunk, on the basis of the Call Cost Calculation Time Schedule configured for the
outgoing call.
Each Call Cost Calculation Pulse Rate option contains a Pulse Rate Type for the Pulse Rate, which is
assigned in the Area Code Table.
• The system matches the Number dialed by the extension user with the Area Code Table configured in the
system. When the area code matches with an entry in the table, the system obtains the Pulse rate type
configured for the Call Cost Calculation Pulse Rate option assigned to the trunk.
• This Pulse Rate type obtained from the Area Code Table is checked in the Pulse Rate table to obtain
the corresponding duration and cost to be applied for the call duration. The Pulse Rate Table may be the
Normal Pulse Rate Table or Discounted Pulse Rate Table, depending on the day of the call. The
system uses the built-in RTC to determine the day.
• The Pulse Rate Type applied (duration and cost) is divided into two parts for each time zone:
• First unit.
• Additional units.

• Number of Units is derived from the pulse rate at the time of the call and duration of the call. System
acquires the pulse rate type and call duration with the help of in-built RTC.
Total Units = First Unit + Additional Unit.
If the call duration is less than the pulse rate of the first unit then additional unit is zero.
Call Units = (Call duration in seconds)/(Pulse rate in seconds).
• If the duration of the call is less than or equal to one unit,
Cost of Call = Cost of First Unit + Service Charge as applied.
• If the duration of the call is more than one unit,
Cost of Call = [Cost of First Unit + (Number of Additional units x Cost for Additional units)] + Service
Charge as applied.
• The system applies the rates as configured in the Normal Pulse Rate Table for all the days, except when it
detects a day configured in the Discounted Pulse Rate Schedule. These are special days when special
Tariffs are offered.
For the days configured in the Discounted Pulse Rate Schedule as special days, the system checks the
the duration and cost of the First unit and the Additional units configured in the Discounted Pulse Rate
Table.
SARVAM UCS uses the Cost of the Call for SMDR. This cost is deducted from the Call Budget (Amount), if
allotted to the trunk and also from the Call Budget, if assigned to the extension users.
The logic for call cost calculation is explained with the help of an example:
• An Outgoing Call is made using trunk, CO-1.
• On CO-1, Call Cost Calculation is configured as follows:
• Call Cost Calculation Pulse Rate option = 1

• Call Cost Calculation Schedule
• Time Zone 1 = Start Time: 00:00, End Time: 22:00
• Time Zone 2 = Start Time: 22:01, End Time: 23:59
• Assign Trunk Feature Template 1 to CO-1. Configuring the Call Cost Calculation parameters in the
Trunk Feature Template as follows:
Call Call Cost Calculation Time Schedule

Cost
T1 T2 T3
Calculat
Template
ion Start Time End Time Start Time End Time Start Time End Time
No.
Pulse
Rate
Option HH MM HH MM HH MM HH MM HH MM HH MM
001 1 00 00 22 00 22 01 23 59

• Area Code Table is configured as:
Ignore Pulse Rate Type for Pulse Rate

Area
Index Name Digit
Code Option - 1 Option - 2 Option - 3 Option - 4
Count
001 26 Local 03 06 09 10
002 09 Mobile 05 03 07 08
• The Normal Pulse Rate Table is configured as:
Time Zone T1 Time Zone T2 Time Zone T3 Time Zone T4

Pulse Rate Type First Add. First Add. First Add. First Add.
Unit Unit Unit Unit Unit Unit Unit Unit
Duration (sec) 180 180 60 30 90 30 120 60

01
Cost 002.00 002.00 002.00 002.00 002.00 002.00 002.00 002.00
Duration (sec) 300 300 300 300 300 300 300 300
02
Cost 001.00 001.00 001.00 001.00 001.00 001.00 001.00 001.00
Durations) 30 30 30 30 30 30 30 30
03
Cost 001.00 001.00 001.00 001.00 001.00 001.00 001.00 001.00
Duration (sec) 45 45 45 45 45 45 45 45
04
Cost 001.00 001.00 001.00 001.00 001.00 001.00 001.00 001.00
Duration (sec) 180 180 180 180 180 180 180 180
05
Cost 003.00 003.00 003.00 003.00 003.00 003.00 003.00 003.00
Durations) : : : : : : : :
:
Cost : : : : : : : :
Duration (sec)
32
Cost
• With this configuration, SARVAM UCS will calculate the call cost as follows:
• An Outgoing Call is made by an extension user, to the number ‘2630555’ through the trunk, CO-1 at
20:10 hours. SARVAM UCS will check the Call Cost Calculation parameters assigned on the trunk
and determine the Time Zone as per time of the call.The system will also check the corresponding
Pulse Rate Option configured on the trunk.
In this example, Time Zone for CO-1 at 20:10 Hours would be Time Zone 1, and the Pulse Rate
Option for CO-1 is ‘1’.
• SARVAM UCS will match the dialed number ‘2630555’ in the Area Code table. A best match is
found with the entry configured at index 001 in the Area Code Table.
As per the Area Code Table, the Pulse Rate Type ‘03’ is programmed in ‘Pulse Rate Option 1’ for
the matching entry (at Index 001).

(However, if CO-1 would have been assigned Pulse Rate Option ‘2’, the Pulse Rate type ‘06’ would
have been selected as shown in Area Code Table)
• Finally, for Pulse Rate Type ‘03’ SARVAM UCS will check the Normal Pulse Rate Table. SARVAM
UCS will consider the Cost for the First Unit as 001.00 (As ` or $ as per applicable currency) for the
duration of 30 seconds and for the additional unit also, the cost will be considered as 001.00 for the
duration of 30 seconds. This data will be used for calculating the total cost of call based on the total
duration of the call.
Similarly, when there are Special Tariffs offered on certain days, the system will check the Discounted Pulse Rate
Schedule and the Discounted Pulse Rate Table.
The days on which the special rates are to be applied must be configured in the Discounted Pulse Rate Schedule.
The duration and cost of the First unit and the Additional units must be configured in the Discounted Pulse Rate
Table according to the Time Zones.
How to configure
To be able to use Call Cost Calculation, you must do the following:
• Define the Unit and Service Charge on the basis of which call cost is to be calculated.
• Assign Call Cost Calculation Pulse Rate Option and the Call Cost Calculation Schedule on the trunks on
which you want to apply this feature.
• Configure the Pulse Rate Types for the Pulse Rate Option you assign to the trunk.
• You may configure different Pulse Rate Types:
• For Normal Days configure the pulse rate in the Normal Pulse Rate Table.
• For Special days configure the pulse rate in the Discounted Rate Table. If you configure the Discounted
Pulse Rate Table, you must also configure the Discounted Pulse Rate schedule.
• Configure the Area Code Table.
Configuring Call Cost Calculation using Jeeves

To configure call cost calculation parameters,

• Under Configuration click the Call Cost Calculation link.
Service Charge
By default, no Service Charge is applied on call cost by the system. Service Charge on call cost is generally applied
in Hotels and other organization which charge users for the calls made by them.
If you want to apply Service Charge,

• Click General Parameters to open the page.
• In the Service Charge Type field, select the type of service charge you want to apply from the options:
• Fixed for a call: A fixed amount is added as service charge to every call regardless of the cost of that
call.
If you select this option, you must define the Amount to be added as service charge in the Specify
Service Charge field.
• per Unit: service charge is added to each unit of the call. For example, if a call worth 10 units was
made, the service charge will be applied on each of the 10 units, instead of the one time service
charge as in the case of Fixed service charge.
If you select this option, you must define the amount to be added as service charge on each unit in the
Specify Service Charge field.
• Percentage of call cost: A percentage of the cost of the call is added as a service charge for that call.
If you select this option, you must define the percentage in the Specify Service Charge in % field
which appears.
Applying Call Cost Calculation on Trunks

• To assign Call Cost Calculation Pulse Rate Option and configure the Call Cost Calculation Schedule
on the trunks, go to Configuration, and configure these in the “Trunk Feature Template” assigned to the
different trunk port types. See “Configuring CO Trunks”, “Configuring Mobile Trunks”, “Configuring BRI
Trunks”, “Configuring E1 Trunks”, “Configuring T1 Trunks”, “Configuring Trunks” and “Configuring SIP
Trunks” under Configuring SARVAM UCS for instructions.
Configuring Pulse Rate Types

Generally service providers offer discounted call rates for special days. To take care of this, SARVAM UCS
provides two types of Pulse Rate Tables: Normal and Discounted.
If your service providers do not offer any special rates, you may skip configuring the Discounted Pulse Rate table.
To configure the Normal Pulse Rate table,
• Click Call Cost Calculation under Configuration.

• Click the Normal Pulse Rate Table link.
• Configure the Pulse Rate Type with rates for the First Unit and the Additional Unit.
Generally service providers offer different call rates for different types of calls, for example: local, national,
international. You can configure different Pulse Rate Types for different types of calls. Thus, each Pulse
Rate Type can have different rates for the First and the Additional unit.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases, you
will need to configure the Call Cost Calculation Schedule for the trunk, by dividing the day into Time
Zones, from 1 to 4, as required, to match the time of the pulse rates offered by your service provider.
If you have configured Time Zones for the Call Cost Schedule on a trunk, you may define the different
Pulse Rate Types for each Time Zone.

To configure the Discounted Pulse Rate table,
• Click the Discounted Pulse Rate Table link.
• Configure the Discounted Pulse Rate Type with rates for the First Unit and the Additional Unit, as
required.

• Click the Discounted Pulse Rate Schedule link.
• Define Weekly and Yearly special days on this page.
Configuring Area Code Table

The pulse rate of a call depends on the destination number dialed. Generally, pulse rate varies according to the
type of the call: Local, national, international. SARVAM UCS enables you to configure Pulse Rates according to
Area codes.
To configure Pulse Rates for Area Codes,
• Click the Area Code link.

• In the Area Code column, enter the number strings (prefix) of the Area Codes, country codes, local
numbers. You can configure as many as 999 area codes in the table.
• You may enter a Name to tag each Area Code.
• Do not configure the Ignore Digit Count. This parameter is relevant only for Service Provider Based Least
Cost Routing.
• Different service providers offer different pulse rates for the same type of calls. To take care of this,
SARVAM UCS allows you to assign different Pulse Rate Options for each area code.
• For each area code, configure the Pulse Rate to be followed for the desired Pulse Rate Options in the
Pulse Rate Type for Pulse Rate (Option 1 to 4) column.
For “Default Area Code Table for the Region-USA”’ at end of the chapter.
Configuring Call Cost using Telephone

Unit Charge
When call cost is to be calculated on the basis of 16 KHz metering pulses, user can select different unit charge for
first unit and different unit charge for the additional units.
To program the unit charge for first unit when 16 KHz metering is used, dial:
• 2600-Unit Charge for First Unit
Where,
Unit charge is the amount in XXX.XX format in any currency.
By default, unit charge for first unit is Rs.1.10.

To program the unit charge for additional units when 12/16 KHz metering is used, dial:
• 2601-Unit Charge for Additional Unit
Where,
Unit charge is the amount in XXX.XX format in any currency.
By default, Unit charge for additional unit is Rs.1.10.
Example: To program unit change for first unit to Rs. 1.50, dial 2600-0150
To program unit charge for additional unit to US$0.75, dial 2601-0075
Service Charge
To select service charge type, dial:

• 2602-Service Charge Type
Where,
Service Charge Type Meaning
0 Fixed Service Charge
1 Unit wise Service Charge
2 Percentage wise Service Charge
By default, the service charge type is fixed.
To program the service charges for Fixed or Unit wise Service Charge, dial:
• 2603-Service Charge
Where,
Service Charge is the amount in XXX.XX format in any currency.
By default, Service Charge is Rs.00.00.
Example: Let us program the service charge to Rs.2.00, dial 2603-0200

Let us program the service charge to US$1.75, dial 2603-0175
To set the percentage, if the service charge is selected based on the percentage, dial:
• 2604-Percentage
Where,
Percentage is from 000 to 100.
By default, percentage is 000.
Example: To set the percentage to 10% of the cost of each call, dial 2604-010
Number of Units
• Number of Units is derived from the pulse rate at the time of the call and duration of the call. System
acquires the pulse rate type and call duration with the help of in-built RTC.
Total Units = First Unit + Additional Unit.
• If the call duration is less than the pulse rate of the first unit then additional unit is zero.
Call Units = (Call duration in seconds)/(Pulse rate in seconds).

Assign parameters; 'CCC Pulse Rate Option' and program 'CCC Time Schedule' for the Trunk Feature Template
which is assigned to the specific trunk used for OG calls.
• Assign 'CCC Pulse Rate Option' from 1 to 4.
• Program four 'CCC Time Schedule', T1, T2,T3 and T4. Program Start Time and End Time for each.
• Refer chapter “Trunk Feature Template” for more details.
Pulse Rate Types
Pulse rate can differ for normal and holidays. Maximum 32 entries can be made in the pulse rate type. Each pulse
rate type can have different rate and different cost for first and additional unit. The table below displays the format
of the pulse rate for normal day.
Time Zone 1 Time Zone 2 Time Zone 3 Time Zone 4

Pulse Rate
Type First Add. First Add. First Add. First Add.
Duration 180.00 180.00 180.00 180.00 180.00 180.00 180.00 180.00

01
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration : : : : : : : :
02
Cost : : : : : : : :
: : : : : : : : : :
Duration 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00

32
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
To program duration of first unit for a pulse rate type on normal days, dial:
• 2607-Pulse Rate Type-Time Zone-Duration of First Unit
Where,
Pulse Rate Type is from 01 to 32.
Time Zone is from 1 to 4.
Duration of First Unit is from 000.00 to 999.99.
To load default normal pulse rate type, dial:

• 2606
To program duration of additional unit for a pulse rate type on normal days, dial:
• 2608-Pulse Rate Type-Time Zone-Duration of Additional Unit
Where,

Duration of Additional Unit is from 000.00 to 999.99.

Pulse Rate
Type First Add. First Add. First Add. First Add.
Duration 180.00 180.00 180.00 180.00 180.00 180.00 180.00 180.00

01
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 300.00 300.00 300.00 300.00 300.00 300.00 300.00 300.00

02
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 90.00 90.00 90.00 90.00 90.00 90.00 90.00 90.00

03
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 120.00 120.00 120.00 120.00 120.00 120.00 120.00 120.00

04
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 30.00 30.00 30.00 30.00 30.00 30.00 30.00 30.00

05
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 30.00 30.00 30.00 30.00 30.00 30.00 30.00 30.00

06
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 16.00 16.00 16.00 16.00 16.00 16.00 16.00 16.00

07
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 3.30 3.30 3.30 3.30 3.30 3.30 3.30 3.30

08
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 2.10 2.10 2.10 2.10 2.10 2.10 2.10 2.10

09
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 1.70 1.70 1.70 1.70 1.70 1.70 1.70 1.70

10
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 2.10 2.10 2.10 2.10 2.10 2.10 2.10 2.10

11
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00

12
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration : : : : : : : :
:
Cost : : : : : : : :
Duration 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00

32
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
To program the cost of first unit of a pulse rate type for normal days, dial:
• 2609-Pulse Rate Type-Time Zone-Cost of First Unit
Where,

Cost of first Unit is from XXX.XX.
To program the cost of additional unit of a pulse rate type for normal days, dial:
• 2610-Pulse Rate Type-Time Zone-Cost of Additional Unit
Where,
Cost of Additional Unit is from XXX.XX.
Discounted Pulse Rate Schedule
To load default Discounted Pulse Rate type:

• 2611

CCC Pulse
Rate Type First Add. First Add. First Add. First Add.
Duration 180.00 180.00 180.00 180.00 180.00 180.00 180.00 180.00

01
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 300.00 300.00 300.00 300.00 300.00 300.00 300.00 300.00

02
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 90.00 90.00 90.00 90.00 90.00 90.00 90.00 90.00

03
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 120.00 120.00 120.00 120.00 120.00 120.00 120.00 120.00

04
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 30.00 30.00 30.00 30.00 30.00 30.00 30.00 30.00

05
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 30.00 30.00 30.00 30.00 30.00 30.00 30.00 30.00

06
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 16.00 16.00 16.00 16.00 16.00 16.00 16.00 16.00

07
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 3.30 3.30 3.30 3.30 3.30 3.30 3.30 3.30

08
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 2.10 2.10 2.10 2.10 2.10 2.10 2.10 2.10

09
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 1.50 1.50 1.50 1.50 1.50 1.50 1.50 1.50

10
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration 2.10 2.10 2.10 2.10 2.10 2.10 2.10 2.10

11
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10

CCC Pulse
Rate Type First Add. First Add. First Add. First Add.
Duration 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00

12
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
Duration : : : : : : : :
:
Cost : : : : : : : :
Duration 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00

16
Cost 001.10 001.10 001.10 001.10 001.10 001.10 001.10 001.10
To program duration for a first unit of a pulse rate type on Special Days, dial:
• 2612-Pulse Rate Type-Time Zone-Duration of First Unit
Where,
Pulse rate type is from 01 to 32.
Time Zone from 1 to 4.
Duration of First Unit is from 000.00 to 999.99.
To program duration for additional unit for a pulse rate type on Special Days, dial:
• 2613-Pulse Rate Type-Time Zone-Duration of Additional Unit
Where,
Pulse rate type is from 01 to 32.
Time Zone from 1 to 4.
Duration of Additional Unit is from 000.00 to 999.99.
To program the cost of first unit of a pulse rate type for holidays, dial:
• 2614-Pulse Rate Type-Time Zone-Cost of First Unit
Where,
Cost of First Unit is from XXX.XX.
To program the cost of additional unit of a pulse rate type for Special Days, dial:
• 2615-Pulse Rate Type-Time Zone-Cost of Additional Unit
Where,
Cost of Additional Unit is from XXX.XX.
Area Code Table
To program an area code, dial:

• 2620-Area Code Index-Area Code-#*
Where,
Area Code is a number string of maximum of 4 digits.
To clear the area code for an index, dial:

Example: Program area code 022 for Mumbai at area code index 001, dial 2620-001-022-#*

To program Pulse Rate Type for Pulse Rate Option of area code index, dial:
• 2621-Area Code Index-Pulse Rate Option-Pulse Rate Type
Where,
Pulse Rate Option is from 1 to 4.
To delete the complete Area Code Table, dial:

• 2622-Reverse SE Password
Example: To delete the area code table, dial 2622-4321 (The SE password is assumed to be 1234).
There is no command to delete a single entry from the area code table. However, these can be cleared or
overwritten.
To program ignore digit count when Service Provider based LCR is to be used, dial:
• 2623-Area Code Index-Ignore Digit Count
Where,
Area Code Index is from 001 to 999. Refer “Default Area Code Table for the Region-USA”’ at end of the
chapter.
Ignore Digit Count is from 0 to 9.
By default, Ignore digit is 0.
Please refer topic “Configuring LCR” for more details on Service Provider base LCR.
Special Days
To assign a Pulse Rate Table to any day of the week, dial:

• 2630-Day-Code
Where,
Index 1 2 3 4 5 6 7
Day Sun Mon Tue Wed Thu Fri Sat
Code Meaning
0 Normal Pulse Rate
1 Discounted Pulse Rate
Example: To program Discounted Pulse Rate for Tuesday, dial 2630-3-1
To program a special date, dial:

• 2631-Special Date Index-Date-Month
Where,
Special Date Index is from 1 to 5 (Five dates can be programmed).
Date is from 01 to 31.
Month is from 01 to 12.
By default, Special is shown below:
Special Date Index Day
1 26-01
2 15-08

Special Date Index Day
3 02-10
4 Blank
5 Blank
To clear a Special date index, dial:

• 2632-Special Date Index
Example: To program 1st May as a Special Day, dial 2631-1-01-05
To program Area code name, dial:

• 2633-Area Code Index-Name
Where,
Name is character string is of maximum 12 characters.
Use following command to clear area code name, dial:

Default Area Code Table for the Region-USA
Index Area Code Area Name Pulse Rate Type Ignore Digit Count
1 1201 NJ 2 0
2 1202 DC 2 0
3 1203 CT 2 0
4 1204 Manitoba 2 0
5 1205 AL 2 0
6 1206 WA 2 0
7 1207 ME 2 0
8 1208 ID 2 0
9 1209 CA 2 0
10 1210 TX 2 0
11 1212 NY 2 0
12 1213 CA 2 0
13 1214 TX 2 0
14 1215 PA 2 0
15 1216 OH 2 0
16 1217 IL 2 0
17 1218 MN 2 0
18 1219 IN 2 0
19 1224 IL 2 0

20 1225 LA 2 0
21 1226 Ontario 2 0
22 1228 MS 2 0
23 1229 GA 2 0
24 1231 MI 2 0
25 1234 OH 2 0
26 1239 FL 2 0
27 1240 MD 2 0
28 1242 Bahamas 2 0
29 1246 Barbados 2 0
30 1248 MI 2 0
31 1250 BC 2 0
32 1251 AL 2 0
33 1252 NC 2 0
34 1253 WA 2 0
35 1254 TX 2 0
36 1256 AL 2 0
37 1260 IN 2 0
38 1262 WI 2 0
39 1264 Anguilla 2 0
40 1267 PA 2 0
41 1268 Antigua 2 0
42 1269 MI 2 0
43 1270 KY 2 0
44 1276 VA 2 0
45 1281 TX 2 0
46 1284 BVI 2 0
47 1289 Ontario 2 0
48 1301 MD 2 0
49 1302 DE 2 0
50 1303 CO 2 0
51 1304 WV 2 0
52 1305 FL 2 0
53 1306 Saskatchewan 2 0

54 1307 WY 2 0
55 1308 NE 2 0
56 1309 IL 2 0
57 1310 CA 2 0
58 1312 IL 2 0
59 1313 MI 2 0
60 1314 MO 2 0
61 1315 NY 2 0
62 1316 KS 2 0
63 1317 IN 2 0
64 1318 LA 2 0
65 1319 IA 2 0
66 1320 MN 2 0
67 1321 FL 2 0
68 1323 CA 2 0
69 1325 TX 2 0
70 1330 OH 2 0
71 1331 IL 2 0
72 1334 AL 2 0
73 1336 NC 2 0
74 1337 LA 2 0
75 1339 MA 2 0
76 1340 USVI 2 0
77 1345 Cayman 2 0
78 1347 NY 2 0
79 1351 MA 2 0
80 1352 FL 2 0
81 1360 WA 2 0
82 1361 TX 2 0
83 1386 FL 2 0
84 1401 RI 2 0
85 1402 NE 2 0
86 1403 Alberta 2 0
87 1404 GA 2 0

88 1405 OK 2 0
89 1406 MT 2 0
90 1407 FL 2 0
91 1408 CA 2 0
92 1409 TX 2 0
93 1410 MD 2 0
94 1412 PA 2 0
95 1413 MA 2 0
96 1414 WI 2 0
97 1415 CA 2 0
98 1416 Ontario 2 0
99 1417 MO 2 0
100 1418 Quebec 2 0
101 1419 OH 2 0
102 1423 TN 2 0
103 1424 CA 2 0
104 1425 WA 2 0
105 1430 TX 2 0
106 1432 TX 2 0
107 1434 VA 2 0
108 1435 UT 2 0
109 1438 Quebec 2 0
110 1440 OH 2 0
111 1441 Bermuda 2 0
112 1443 MD 2 0
113 1450 Quebec 2 0
114 1456 NANParea 2 0
115 1469 TX 2 0
116 1473 Grenada 2 0
117 1478 GA 2 0
118 1479 AR 2 0
119 1480 AZ 2 0
120 1484 PA 2 0
121 1500 NANParea 2 0

122 1501 AR 2 0
123 1502 KY 2 0
124 1503 OR 2 0
125 1504 LA 2 0
126 1505 NM 2 0
127 1506 NewBrunswick 2 0
128 1507 MN 2 0
129 1508 MA 2 0
130 1509 WA 2 0
131 1510 CA 2 0
132 1512 TX 2 0
133 1513 OH 2 0
134 1514 Quebec 2 0
135 1515 IA 2 0
136 1516 NY 2 0
137 1517 MI 2 0
138 1518 NY 2 0
139 1519 Ontario 2 0
140 1520 AZ 2 0
141 1530 CA 2 0
142 1540 VA 2 0
143 1541 OR 2 0
144 1551 NJ 2 0
145 1559 CA 2 0
146 1561 FL 2 0
147 1562 CA 2 0
148 1563 IA 2 0
149 1567 OH 2 0
150 1570 PA 2 0
151 1571 VA 2 0
152 1573 MO 2 0
153 1574 IN 2 0
154 1575 NM 2 0
155 1580 OK 2 0

156 1585 NY 2 0
157 1586 MI 2 0
158 1600 Canada 2 0
159 1601 MS 2 0
160 1602 AZ 2 0
161 1603 NH 2 0
162 1604 BC 2 0
163 1605 SD 2 0
164 1606 KY 2 0
165 1607 NY 2 0
166 1608 WI 2 0
167 1609 NJ 2 0
168 1610 PA 2 0
169 1612 MN 2 0
170 1613 Ontario 2 0
171 1614 OH 2 0
172 1615 TN 2 0
173 1616 MI 2 0
174 1617 MA 2 0
175 1618 IL 2 0
176 1619 CA 2 0
177 1620 KS 2 0
178 1623 AZ 2 0
179 1626 CA 2 0
180 1630 IL 2 0
181 1631 NY 2 0
182 1636 MO 2 0
183 1641 IA 2 0
184 1646 NY 2 0
185 1647 Ontario 2 0
186 1649 T&CIsland 2 0
187 1650 CA 2 0
188 1651 MN 2 0
189 1660 MO 2 0

190 1661 CA 2 0
191 1662 MS 2 0
192 1664 Montsrat 2 0
193 1670 CNMI 2 0
194 1671 GU 2 0
195 1678 GA 2 0
196 1682 TX 2 0
197 1684 AS 2 0
198 1700 NANParea 2 0
199 1701 ND 2 0
200 1702 NV 2 0
201 1703 VA 2 0
202 1704 NC 2 0
203 1705 Ontario 2 0
204 1706 GA 2 0
205 1707 CA 2 0
206 1708 IL 2 0
207 1709 Newfoundland 2 0
208 1710 US 2 0
209 1712 IA 2 0
210 1713 TX 2 0
211 1714 CA 2 0
212 1715 WI 2 0
213 1716 NY 2 0
214 1717 PA 2 0
215 1718 NY 2 0
216 1719 CO 2 0
217 1720 CO 2 0
218 1724 PA 2 0
219 1727 FL 2 0
220 1731 TN 2 0
221 1732 NJ 2 0
222 1734 MI 2 0
223 1740 OH 2 0

224 1754 FL 2 0
225 1757 VA 2 0
226 1758 St.Lucia 2 0
227 1760 CA 2 0
228 1762 GA 2 0
229 1763 MN 2 0
230 1765 IN 2 0
231 1767 Dominica 2 0
232 1769 MS 2 0
233 1770 GA 2 0
234 1772 FL 2 0
235 1773 IL 2 0
236 1774 MA 2 0
237 1775 NV 2 0
238 1778 BC 2 0
239 1779 IL 2 0
240 1780 Alberta 2 0
241 1781 MA 2 0
242 1784 St. V&G 2 0
243 1785 KS 2 0
244 1786 FL 2 0
245 1787 PrtoRico 2 0
246 1800 NANParea 2 0
247 1801 UT 2 0
248 1802 VT 2 0
249 1803 SC 2 0
250 1804 VA 2 0
251 1805 CA 2 0
252 1806 TX 2 0
253 1807 Ontario 2 0
254 1808 HI 2 0
255 1809 DomRepub 2 0
256 1810 MI 2 0
257 1812 IN 2 0

258 1813 FL 2 0
259 1814 PA 2 0
260 1815 IL 2 0
261 1816 MO 2 0
262 1817 TX 2 0
263 1818 CA 2 0
264 1819 Quebec 2 0
265 1828 NC 2 0
266 1829 DomRepub 2 0
267 1830 TX 2 0
268 1831 CA 2 0
269 1832 TX 2 0
270 1843 SC 2 0
271 1845 NY 2 0
272 1847 IL 2 0
273 1848 NJ 2 0
274 1850 FL 2 0
275 1856 NJ 2 0
276 1857 MA 2 0
277 1858 CA 2 0
278 1859 KY 2 0
279 1860 CT 2 0
280 1862 NJ 2 0
281 1863 FL 2 0
282 1864 SC 2 0
283 1865 TN 2 0
284 1866 NANParea 2 0
285 1867 Yukon 2 0
286 1868 Tri&Tob 2 0
287 1869 St. K&N 2 0
288 1870 AR 2 0
289 1876 Jamaica 2 0
290 1877 NANParea 2 0
291 1878 PA 2 0

292 1888 NANParea 2 0
293 1900 NANParea 2 0
294 1901 TN 2 0
295 1902 N Scotia 2 0
296 1903 TX 2 0
297 1904 FL 2 0
298 1905 Ontario 2 0
299 1906 MI 2 0
300 1907 AK 2 0
301 1908 NJ 2 0
302 1909 CA 2 0
303 1910 NC 2 0
304 1912 GA 2 0
305 1913 KS 2 0
306 1914 NY 2 0
307 1915 TX 2 0
308 1916 CA 2 0
309 1917 NY 2 0
310 1918 OK 2 0
311 1919 NC 2 0
312 1920 WI 2 0
313 1925 CA 2 0
314 1928 AZ 2 0
315 1931 TN 2 0
316 1936 TX 2 0
317 1937 OH 2 0
318 1939 PrtoRico 2 0
319 1940 TX 2 0
320 1941 FL 2 0
321 1947 MI 2 0
322 1949 CA 2 0
323 1951 CA 2 0
324 1952 MN 2 0
325 1954 FL 2 0

326 1956 TX 2 0
327 1970 CO 2 0
328 1971 OR 2 0
329 1972 TX 2 0
330 1973 NJ 2 0
331 1978 MA 2 0
332 1979 TX 2 0
333 1980 NC 2 0
334 1985 LA 2 0
335 1989 MI 2 0
336 0117 Kazkhstan 3 0
337 01120 Egypt 3 0
338 01127 South Africa 3 0
339 01130 Greece 3 0
340 01131 Netherlands 3 0
341 01132 Belgium 3 0
342 01133 France 3 0
343 01134 Spain 3 0
344 01136 Hungary 3 0
345 01139 VaticanCity 3 0
346 01140 Romania 3 0
347 01141 Switzerland 3 0
348 01143 Austria 3 0
349 01144 UK 3 0
350 01145 Denmark 3 0
351 01146 Sweden 3 0
352 01147 Norway 3 0
353 01148 Poland 3 0
354 01149 Germany 3 0
355 01151 Peru 3 0
356 01152 Mexico 3 0
357 01153 Cuba 3 0
358 01154 Argentine 3 0
359 01155 Brazil 3 0

360 01156 Chile 3 0
361 01157 Colombia 3 0
362 01158 Venezuela 3 0
363 01160 Malaysia 3 0
364 01161 Australia 3 0
365 01162 Indonesia 3 0
366 01163 Philippines 3 0
367 01164 NZ 3 0
368 01165 Singapore 3 0
369 01166 Thailand 3 0
370 01181 Japan 3 0
371 01182 Korea 3 0
372 01184 VietNam 3 0
373 01186 China 3 0
374 01190 Turkey 3 0
375 01191 India 3 0
376 01192 Pakistan 3 0
377 01193 Afghanistan 3 0
378 01194 Sri Lanka 3 0
379 01195 Myanmar 3 0
380 01198 Iran 3 0
381 011212 Morocco 3 0
382 011213 Algeria 3 0
383 011216 Tunisia 3 0
384 011218 Libya 3 0
385 011220 Gambia 3 0
386 011221 Senegal 3 0
387 011222 Mauritania 3 0
388 011223 Mali 3 0
389 011224 Guinea 3 0
390 011225 IvoryCoast 3 0
391 011226 BurkinaFaso 3 0
392 011227 Niger 3 0
393 011228 Togolese 3 0

394 011229 Benin 3 0
395 011230 Mauritius 3 0
396 011231 Liberia 3 0
397 011232 SierraLeone 3 0
398 011233 Ghana 3 0
399 011234 Nigeria 3 0
400 011235 Chad 3 0
401 011236 CenAfrica 3 0
402 011237 Cameroon 3 0
403 011238 CapeVerde 3 0
404 011239 SaoTome 3 0
405 011240 Equtl_Guinea 3 0
406 011241 Gabonese 3 0
407 011242 Congo 3 0
408 011243 CongoDem 3 0
409 011244 Angola 3 0
410 011245 GuineaBissa 3 0
411 011246 DiegoGarcia 3 0
412 011247 Ascension 3 0
413 011248 Seychelles 3 0
414 011249 Sudan 3 0
415 011250 Rwandese 3 0
416 011251 Ethiopia 3 0
417 011252 SomalianRep 3 0
418 011253 Djibouti 3 0
419 011254 Kenya 3 0
420 011255 Tanzania 3 0
421 011256 Uganda 3 0
422 011257 Burundi 3 0
423 011258 Mozambique 3 0
424 011260 Zambia 3 0
425 011261 Madagascar 3 0
426 011262 Reunion 3 0
427 011263 Zimbabwe 3 0

428 011264 Namibia 3 0
429 011265 Malawi 3 0
430 011266 Lesotho 3 0
431 011267 Botswana 3 0
432 011268 Swaziland 3 0
433 011269 Comoros 3 0
434 011290 StHelena 3 0
435 011291 Eritrea 3 0
436 011297 Aruba 3 0
437 011298 FaroeIsland 3 0
438 011299 Greenland 3 0
439 011350 Gibraltar 3 0
440 011351 Portugal 3 0
441 011352 Luxembourg 3 0
442 011353 Ireland 3 0
443 011354 Iceland 3 0
444 011355 Albania 3 0
445 011356 Malta 3 0
446 011357 Cyprus 3 0
447 011358 Finland 3 0
448 011359 Bulgaria 3 0
449 011370 Lithuania 3 0
450 011371 Latvia 3 0
451 011372 Estonia 3 0
452 011373 Moldova 3 0
453 011374 Armenia 3 0
454 011375 Belarus 3 0
455 011376 Andorra 3 0
456 011377 Monaco 3 0
457 011378 SanMarino 3 0
458 011379 VaticanCity 3 0
459 011380 Ukraine 3 0
460 011381 Yugoslavia 3 0
461 011385 Croatia 3 0

462 011386 Slovenia 3 0
463 011387 Bosnia 3 0
464 011389 Macedonia 3 0
465 011420 Czech Repub 3 0
466 011421 Slovakia 3 0
467 011423 Liechtenstein 3 0
468 011500 Falklands 3 0
469 011501 Belize 3 0
470 011502 Guatemala 3 0
471 011503 El Salvador 3 0
472 011504 Honduras 3 0
473 011505 Nicaragua 3 0
474 011506 CostaRica 3 0
475 011507 Panama 3 0
476 011508 St.Pierre 3 0
477 011509 Haiti 3 0
478 011590 Guadeloupe 3 0
479 011591 Bolivia 3 0
480 011592 Guyana 3 0
481 011593 Ecuador 3 0
482 011594 FrenchGuyana 3 0
483 011595 Paraguay 3 0
484 011596 Martinique 3 0
485 011597 Suriname 3 0
486 011598 Uruguay 3 0
487 011599 NethAntilles 3 0
488 011670 East Timor 3 0
489 011672 Antarctic 3 0
490 011673 Brunei 3 0
491 011674 Nauru 3 0
492 011675 PapuaNewGuin 3 0
493 011676 Tonga 3 0
494 011677 SolomonIslnd 3 0
495 011678 Vanuatu 3 0

496 011679 Fiji 3 0
497 011680 Palau 3 0
498 011681 Wallis Island 3 0
499 011682 Cook Islands 3 0
500 011683 Niuel Island 3 0
501 011684 AmerSamoa 3 0
502 011685 WSamoa 3 0
503 011686 Kiribati 3 0
504 011687 NewCaledonia 3 0
505 011688 Tuvalu 3 0
506 011689 FrenchPolyne 3 0
507 011690 Tokelau 3 0
508 011691 Micronesia 3 0
509 011692 MarshalIslnd 3 0
510 011850 Korea North 3 0
511 011852 Hong Kong 3 0
512 011853 Macau 3 0
513 011855 Cambodia 3 0
514 011856 Laos 3 0
515 011870 SatIndlOcn 3 0
516 011871 SatEastAtl 3 0
517 011872 SatPacific 3 0
518 011873 SatIndianOcn 3 0
519 011874 SatWestAtl 3 0
520 011880 Bangladesh 3 0
521 011960 Maldives 3 0
522 011961 Lebanon 3 0
523 011962 Jordan 3 0
524 011963 SyrianArab 3 0
525 011964 Iraq 3 0
526 011965 Kuwait 3 0
527 011966 SaudiArabia 3 0
528 011967 Yemen 3 0
529 011968 Oman 3 0

530 011971 UAE 3 0
531 011972 Israel 3 0
532 011973 Bahrain 3 0
533 011974 Qatar 3 0
534 011975 Bhutan 3 0
535 011976 Mongolia 3 0
536 011977 Nepal 3 0
537 011992 Tajikistan 3 0
538 011993 Turkmenistan 3 0
539 011994 Azerbaijani 3 0
540 011995 Georgia 3 0
541 011996 Kyrgyzstan 3 0
542 011998 Uzbekistan 3 0
543
544
545
546
547
998
999

Call Cost Display
What is this?
With Call Cost Display, you can view the cost of the last 10 external calls made from your extension. These external
calls may have been made from Trunk Ports and on the Tie line network.
The system will display the dialed numbers and the call cost for each number that it has calculated on the LCD of
EON / Extended IP Phone.
How to configure
For this feature to work, it must be enabled on the extension by the System Administrator (SA).
To enable 'Call Cost Display' for an extension:
• Enter SA mode.
• Dial command 1072-181-Station-Flag

Where,
Station is Number of the Extension
Flag is
0 for Disable
1 for Enable
By default Call Cost Display is enabled.
• Exit SA mode.
How to use
For EON and Extended IP Phone Users Only

• Press DSS Key assigned to Call Cost Display feature.
OR
• Dial 1075.
• Scroll with the up/down navigation keys to view the cost of the last 10 calls.
• The display shows the last 10 dialed numbers and their corresponding call cost.
For example: If the call charge is for the dialed number 0014034545247 is $2 and 80 cents' then the
display will show:
001403454 5247 2.8 0

F ri 22 JAN 12:19

• The display is retained till the extension remains OFF-Hook.

Call Duration Control (CDC)
What's this?
Call Duration Control (CDC) allows a maximum time limit to be set on internal and external (both incoming and
outgoing) telephone calls. When the maximum call duration is reached, the calls are disconnected, after a warning
tone indicating to the user that the calls in progress will be disconnected.
By limiting the duration of the conversations, CDC helps increase availability of trunks for making outgoing calls
and for receiving incoming calls, which is important in high call traffic situations. Besides increasing trunk
availability, CDC curbs unrelated and unproductive conversations.
How it works
• A is an extension user. B is an external number.
External-Outgoing Calls
• A dials B's number.
• The system checks the CDC Table applied to A.
• SARVAM UCS will check whether the CDC is applicable on Extension of A as well as the Trunk used for
making the outgoing call.
To check whether CDC is applicable on Extension A
• The system checks the CDC Table selected in the Station Advanced Feature Template assigned to A. In
the Table,
• It checks whether the check box, Apply CDC for outgoing calls made from trunk, is enabled.
• It matches B's number with the entries on the Apply CDC for calls matching with numbers list and
the Do Not Apply CDC for calls matching with numbers list in the CDC table.
To check whether CDC is applicable on the Trunk
• The system checks the Call Duration Control - For OG Calls check box in the Trunk Features Template
assigned to the trunk.
The following results are possible:
• In the CDC Table, the Apply CDC for outgoing calls made from trunk check box is enabled and a
match is found for the number in the Apply CDC for calls matching with numbers List. The Call
Duration Control - For OG Calls check box is enabled in the Trunk Features Template assigned to
the trunk. CDC is applied on the call.
match is found for the number in the Apply CDC for calls matching with numbers List. The Call

Duration Control - For OG Calls check box is disabled in the Trunk Features Template assigned to
the trunk. CDC is not applied on the call.
match is found in the Do Not Apply CDC for calls matching with numbers list. The Call Duration
Control - For OG Calls check box is enabled in the Trunk Features Template assigned to the trunk.
CDC is not applied on the call.
match is found in the Do Not Apply CDC for calls matching with numbers list. The Call Duration
Control - For OG Calls check box is disabled in the Trunk Features Template assigned to the trunk.
CDC is not applied on the call.
match is found in both the number lists. The system gives precedence to the Do Not Apply CDC for
calls matching with numbers list. The Call Duration Control - For OG Calls check box is enabled in
the Trunk Features Template assigned to the trunk. CDC is not applied on the call.
match is found in both the number lists. The system gives precedence to the Do Not Apply CDC for
calls matching with numbers list. The Call Duration Control - For OG Calls check box is disabled in
the Trunk Features Template assigned to the trunk. CDC is not applied on the call.
• When CDC is applied to the call, the CDC Timer starts as soon as B has answered the call. This timer is
set to 160 seconds as default, but can be programmed to the desired time limit.
• At the end of the default/programmed time limit of the CDC Timer, the Beep Timer starts (5 seconds; non-
programmable) then the CDC Goodbye Timer starts. This Goodbye timer provides a grace period of 20
seconds for the user to finish the call. This Timer is non-programmable.
• At the end of the Goodbye Timer, the call is disconnected, if the Disconnect CDC after Time check box is
enabled.
• If this flag has not been programmed, the call will not be disconnected.
• Instead, the CDC Warn Timer will be loaded again for the default/programmed duration. The user can
know how long s/he has been talking.
• A is played Warning Beeps. B cannot hear the beeps. This continues until either party disconnects.
External-Incoming Calls
CDC works similarly for incoming calls.
• B calls A.
• The system checks whether the following conditions are fulfilled,
• the Apply CDC for incoming calls received from trunk check box is enabled in the CDC Table
assigned to A.
• the Disconnect Call after CDC Timer check box is enabled in the CDC Table assigned to A.
• a match is found for the B’s number in the Apply CDC for calls matching with numbers List.

• the Call Duration Control - For IC Calls check box is enabled in the Trunk Features Template
assigned to the trunk on which the call is received.
CDC is applied on the call.
• A will get beeps.
Internal Calls
A and B are extension users.
• A calls B.
• The system checks whether the check box, Apply CDC to Internal Calls, is enabled in the CDC Table.
• If the check box is enabled, CDC is applied on the call.
• Warning beeps are played to A, who made the call.
Warning Beep
Default: 160 secs. Default: 20 secs.
Total time after which call will be disconnected
In this case the total time after which the call is disconnected will be 160 seconds (default CDC Timer) + 5 seconds
(Beep Timer; non-programmable) + 20 seconds (Goodbye Timer; non-programmable).
Feature Interactions:
• Call Transfer: In case of transferred call, the CDC timer gets reset and starts again afresh on the
transferred extension.
• Conference, Conversation Recording and Call Taping: CDC is treated as turned OFF.
• Call Park and Call Hold (Exclusive and Global): CDC is treated as turned ON.
• Interrupt Request, Barge-In: CDC is treated as turned ON.
• Raid: CDC is not applicable when you raid an extension as the conversation is converted into a 3 party
conference.
• Emergency Number Dialing: Emergency calls are not affected by this feature, i.e. CDC will not be
applied on the dialing of Emergency Numbers.
For Inter PINX or Intra PINX calls (QSIG Calls), the CDC will work only if it is enabled on the source port
(calling extension) irrespective of whether CDC is enabled or disabled on the called extension.

How to configure
To use this feature on extensions,
• Configure Call Duration Control Table. You may configure up to 8 different Tables.
• Assign a CDC Table in the Station Advanced Feature Template of those extensions on which Call
Duration Control is to be applied.
• Enable CDC for incoming and outgoing calls from the Trunks on which you want to apply this feature.
To program the Call Duration Control Table,
• decide the types of calls - Outgoing, Incoming and Internal - on which CDC is to be enabled.
• make a list of numbers on which CDC is to be applied, that is, the Apply CDC to Numbers List.
• Make a list of numbers on which CDC is not to be applied, the Do Not Apply CDC to Number List.
The Call Duration Control Table can be programmed using Jeeves and a Telephone.
Configuring CDC using Jeeves
CDC Table
• Under Configuration, click the Call Duration Control to open the page.
• The CDC Table will open. There are 8 CDC Tables. By default CDC Table No. 1 is assigned to all
extensions of SARVAM UCS. If the same CDC is to be assigned to all extensions, program this table.
If different CDC is to be applied to different extensions, program separate CDC tables for these
extensions.
• Now program the following parameters in the table you have selected:

• Apply CDC to Internal Calls: This check box is to be enabled if CDC is to be applied on internal calls.
By default the flag is disabled.
• Apply CDC to Incoming Calls received from Trunk: This check box is to be enabled if CDC is to be
applied to incoming calls external calls. By default this flag is disabled.
• Apply CDC to Outgoing Calls made from Trunks: This check box is to be enabled if CDC is to be
applied to outgoing external calls. By default this flag is disabled.
• Do Not Apply CDC for calls matching with numbers: This is the list of numbers on which CDC is not
to be applied. By default, Number List 08 is assigned to this parameter. You must program this list with
numbers which you want to be exempt from CDC.
To program the list, click Do Not Apply CDC for calls matching with numbers.

This will lead you to the “Number Lists” page. Click '001-250' of Number list 07-08.
By default, Number List 08 is assigned to this parameter. You can also program any other Number List
you want. Enter the list of numbers on which CDC is not be applied (refer to the list you prepared).
Click Submit at the bottom of the page to save your list.
Return to the Call Duration Control page. If you have prepared a Number List other than the default
08, then enter the number of that list in the Do Not Apply CDC to Number List column.
• Apply CDC for calls matching with numbers: This is the list of numbers on which CDC is to be
applied. By default, Number List 07 is assigned to this parameter. You must program this list with
numbers on which you want CDC to be applied.
To program the list, click Apply CDC for calls matching with numbers. The 'Number Lists' page
opens.
Click '001-250' of Number list 07-08. Follow the same steps as described above for programming the
Do Not Apply CDC Number List.
• CDC Timer: This is the time for which the warning beeps are to be played before the system
disconnects the call. The range of the timer is 0001 to 9999 seconds. By default this Timer is set to 160
seconds. Set the CDC Timer to the desired time limit.
• Disconnect Call after CDC Timer: This check box is to be enabled if you want the call to be
automatically disconnected on the expiry of the CDC Timer. By default the check box is disabled, which
means that calls will not be disconnected on expiry of the CDC Timer. Enable the check box if required.

• Under Configuration, click Station Advanced Feature Template. Ensure that the CDC Table Number
(in this case 01) you have programmed is assigned in the Template you want to apply to the extensions.
• To assign the Station Advanced Feature Template with the CDC Table on SLT, DKP and ISDN Terminal
extensions, SIP Extensions go to the respective pages SLT Parameters under SLT Configuration, DKP
Parameters under DKP Configuration and ISDN Terminal Parameters under ISDN Configuration and
SIP Extensions Settings under VoIP Configuration.
Refer “Station Advanced Feature Template” for instructions on customizing the templates and assigning
them to extensions.
• If selected extensions are to be allowed CDC or if different CDC parameters are to be allowed to selected
extensions (for example, 160 seconds duration timer for a few extensions, 360 duration timer for some
other extensions), then follow these steps:
a. Define a new CDC table.
b. Program the different CDC parameters in this table, as required for the extensions.
c. Apply this CDC table on a separate Station Advanced Feature Template.
d. Apply the new Station Advanced Feature Template now programmed with a different CDC table on the
selected extensions which are to be allowed this feature.
CDC on Trunk
• You must enable CDC in the Trunk Feature Template assigned to the trunks. For instructions, see
“Configuring Trunks”.

• Log out of Jeeves or continue programming if required.
Configuring CDC using a Telephone
CDC Table
• Enter SE mode.
To enable CDC for Outgoing Calls in a CDC Table, dial:

• 4202-1-CDC Table-Code to enable the flag in a single table.
• 4202-2-CDC Table-CDC Table-Code to enable the flag in a range of tables.
• 4202-*-Code to enable the flag in all tables.
Where,
CDC Table is from 1 to 8.
Code is
0 for Disable
1 for Enable.
To enable CDC flag for Incoming Calls in a CDC Table, dial:

Where,
Code is
0 for Disable.
1 for Enable.
To enable CDC flag for Internal Call, dial:

Where,
Code is
0 for Disable.
1 for Enable.
To assign a number list to Apply CDC to Number List, dial:

• 4205-1-CDC Table-Number List to enable the flag in a single table.
• 4205-2-CDC Table-CDC Table-Number List to enable the flag in a range of tables.
• 4205-*-Number List to enable the flag in all tables.
Where,
Number List is from 01 to 16.
Default Number List is 07.
To assign a number list to Do Not Apply CDC to Number List, dial:

• 4206-1-CDC Table- Number List to assign Number List to a single table.
• 4206-2-CDC Table-CDC Table- Number List to assign Number List to a range of tables.
• 4206-*-Number List to assign Number List to all tables.
Where,

Default, Number List is 08.
You must also program the Number Lists (default: 07 and 08) before assigning them to a CDC table. Refer
the topic 'Number Lists' for programming instructions.
To change CDC timer in a CDC table:

• 4207-1-CDC Table-CDC Timer to set Timer value in a single table.
• 4207-2-CDC Table-CDC Table-CDC Timer to set the same Timer value in a range of table
• 4207-*-CDC Timer to set the same Timer value in all tables.
Where,
CDC Timer range is from 0001 to 9999 seconds.
Default CDC Timer value is 160 seconds.
To enable the Disconnection Flag in a CDC table:

• 4208-1-CDC Table-Disconnection Flag to enable the flag in a single table.
• 4208-2-CDC Table-CDC Table-Disconnection Flag to enable the flag in a range of tables.
• 4208-*-Disconnection Flag to enable the flag in all tables.
Where,
Disconnection Flag is
0 for Disable.
1 for Enable.
To default a CDC table220 (see Table with Default Values):

• 4201-1-CDC Table to default a single table.
• 4201-2-CDC Table-CDC Table to default a range of tables.
• 4201-* to default all tables.
Where,
CDC Table is 1 to 8.
Example: Apply Call Duration Control on SLT 202 (connected at software port number 001) to disconnect all
calls starting with '0' after 240 seconds, except calls starting with '022'.
Solution: Since only one SLT is to be programmed, it is recommended that you use a CDC table other than
the default (CDC Table No. 1). So, select another CDC table to be assigned to SLT extension 202, for
example CDC Table No. 5. Follow these steps:
1. First, program the Apply CDC to Number List and Do Not Apply CDC to Number List. For example, take
Number List 04 as the Apply CDC to Number List and program the number '0' in this list. Take Number List
05 as the Do Not Apply CDC to Number List and program '022' in this list. Refer the topic “Number Lists”
for programming instructions.
2. Enable CDC for Outgoing call in the table. If using SE Commands, dial 4202-1-5-1.
3. Assign Number List 04 as allowed list and Number List 05 as denied list in table 5. If using SE commands
dial 4205-1-5-04 to assign List 04 and dial 4206-1-5-05 to assign List 05.
4. Change the CDC timer to 240 seconds. If using SE Commands, dial 4207-1-5-240.
5. Enable CDC disconnection Flag in CDC Table 5. If using SE Commands, dial 4208-1-5-1.
220. Refer Call Duration Control Table (Default Values).

6. Assign CDC Table 5 to SLT 202. To do this, change the CDC number in a Station Advanced Feature
Template of SLT 202.
7. Program a different Station Advanced Feature Template for SLT 202. Ensure that all other features and
parameters in the template are relevant for SLT 202.
8. Change the CDC Table number in the Template to 5.
9. Assign the Station Advanced Feature Template now programmed with CDC Table No. 5 to SLT 202. Refer
the topic “Station Advanced Feature Template” for programming instructions.

Call Duration Display
What's this?
By invoking Call Duration Display, extension users can view the duration of the current call instantly.
The system displays the duration of the current call on the LCD display of the phone.
The current call may be incoming or outgoing, internal or external call.
How it works
• The user goes OFF-Hook
• Dials an external number.
• The remote party answers the call.
• The dialed external number with duration (5-digits in the format of MM:SS) is displayed on the LCD of the
phone, when the call is answered.
6 1 6 A M I T PAT E L 02:52
F ri 22 JAN 12:19

Call Forward
What's this?
During a typical workday, it is common for people in an organization to move from one place to another. For
instance, a manager might go on the production floor or remain in the conference room for a few hours; a field
engineer may spend half of the day on site. So, they need to be able to attend their calls even when they are not
present at their desks. The 'Call Forward' feature of SARVAM UCS ensures this.
Using this feature, calls landing on an extension can be forwarded to another extension, an external number, Voice
Mail, or a Department Group. This way, extension users can ensure that callers can reach them and that they do
not miss calls when they are not present at their extension.
You can also set Call Forward for all the extensions from the SA Mode. See “Settings Call Forward for All
Extensions using SA Jeeves”.
The Call Forward feature of SARVAM UCS offers the following forwarding options:
• Unconditionally - calls are forwarded to the destination phone number automatically without waiting for a
response from the called party's phone.
• If Busy - calls are forwarded to the destination phone number only when the called party's phone is busy.
• If No Reply - calls are forwarded to the destination phone number only when the called party does not
answer the phone. Each extension can set a different time after which the call should be forwarded, in
case of no reply. The default time is 30 seconds for all extensions and can be changed by programming
the Call Forward No-Reply Timer.
• If Busy or No Reply - calls are forwarded to the destination phone number when the called party's phone
is either busy or does not reply.
• Dual Ring221 - when calls are forwarded to another phone number. Both phones, that is, the source phone
(whose calls are forwarded) as well as the destination phone (on which call is forwarded) will ring and the
user can answer from either extension.
Dual Ring is useful to users who may have to be present frequently at two different places. As it is
cumbersome to forward the calls from one extension to another and cancel it repeatedly, extensions users
can set Dual Ring, so they can attend to their calls at either place they are present.
If you do not want to set/cancel Call Forward manually, you can set Preset Call Forward. Call will be
forwarded automatically to the selected destination according to the type of Preset Call Forward set.
You can set a different type of forward and destination for each time zone. To know more, see “Preset Call
Forward”.
221. This feature is not supported in Q-Signaling.

How it works
A has set Call Forward to extension B unconditionally.
• The system forwards all calls for A to B, without checking for Busy Tone and without waiting for the Call
Forward No-Reply Timer to expire.
A has set Call Forward to external number on No-Reply.
• The system waits for the Call Forward No-Reply Timer to expire and forwards all external incoming calls to
the external number.
A has set Call Forward to extension B on Busy.
• The system forwards the call for A to B on detecting Busy signal from A.
B belongs to a Department Group and has set Call Forward-If Busy to C within the Department Group.
• If the system detects Busy signal on B, it forwards the call for B to C in the Department Group.
• However, if the caller has called the Department Group instead of calling B directly, the call will land in the
sequence on all Department group extensions. When it is B's turn, the call will not be forwarded to C, B will
ring instead.
C belongs to a Department Group and has set Call Forward-No Reply to D within the Department Group.
• The system waits for the Call Forward No-Reply Timer to expire, and forwards the call for C to D in the
Department Group.
D has set Call Forward to Voice Mail on Busy or No Reply.
• Whenever there is a call for D, if the system does not detect a busy signal from D, it waits for the Call
Forward No-Reply timer to expire.
• The system forwards the call to the Voice Mail System.
E has set Call Forward Dual Ring on extension F.
• When there is a call for E, the system rings on both E and the destination F.
• E can answer the call at E or at F.
Call Forward set by member extensions in a routing group will be ignored by the system if, the Ignore call
forward set by member extension, when call is routed on Routing/Dept. Group flag is enabled. See
“System Parameters” for more information.
Call Forward when set/canceled from the SA mode, will not depend on the assigned CoS.

Feature Interaction:
• Do Not Disturb (DND):
When DND or DND with Intercept Destination is set along with Call Forward-Unconditional on an
extension, Call Forward is given priority.
If any other type of Call Forward and DND are set on an extension, DND is given priority. However, DND
with Intercept Destination will not work.
If an extension has set both Call Forward and DND, then Feature Tone will be played to the extension
user.
• You can select the types of calls, that is, internal calls only, or trunk calls, or both, to be forwarded to
external numbers. You can program the system to forward internal calls only, or trunk calls only or both
trunk calls and internal calls, to the external number. For this, the parameter 'Allow External Call
Forward for' must be programmed in the “Station Advanced Feature Template” of the extensions that
are allowed Call Forward.
• The system supports only single-point Call Forward, which means, if the destination extension is also
forwarded, the call will not follow the forwarding path. For example: Calls for extension A are set to be
forwarded to extension B. Call Forward is also set on extension B with C as the destination number.
Calls for A will land on B only and calls for B will land on C only.
• Only one Call Forward Type can be set from an extension. Every new Call Forward Type set overrides
the previous one.
• When the calls are forwarded the extension user gets the feature tone on lifting the handset to indicate
that Call Forward is set on his/her extension.
How to configure
The functioning of this feature is controlled by three parameters: 'Class of Service' and 'Call Forward No-Reply
Timer' and ‘Allow External Call Forward for’.
Call Forward must be enabled in the Class of Service (COS) group of the extensions to which this feature is to be
allowed.
When Call Forward No-Reply is set, if required the Call Forward No-Reply Timer needs to be programmed.
You may select the types of calls, that is, internal, external, both internal and external calls to be forwarded by
programming the ‘Allow External Call Forward for’ parameter.
You can set Call Forward for All the Extensions from the SA mode only, see “Settings Call Forward for All
Extensions using SA Jeeves”.
Call Forward in Class of Service

SARVAM UCS. Template 01 is assigned COS group 01 in which Call Forward is enabled. So, all extensions of
SARVAM UCS can use Call Forward.
If you want to deny Call Forward to certain extensions, follow these steps:

a. Define a CoS group with Call Forward disabled.
b. Prepare a Station Basic Feature Template with this CoS group applicable in all the “Time Zones”.
c. Assign this new Template to the extensions to which Call Forward is to be denied.
Refer the topics Class of Service and Station Basic Feature Template for programming instructions.
Call Forward No Reply Timer

When using Call Forward -No Reply, each extension can set a different Time after which the incoming call on the
extension should get forwarded when there is no reply from the extension. For this, the Call Forward No-Reply
Timer must be programmed. By default, this timer is set to 30 seconds.
The Call Forward No-Reply Timer is to be programmed in the “Station Advanced Feature Template” applied on the
extensions which are allowed Call Forward in their COS.
If you want to set this timer to the same duration for all extensions, simply set the Call Forward No-Reply Timer in
the default Station Advanced Feature Template 01 which is assigned to all extensions.
If you want to set different Timer duration for different extensions, then prepare separate Station Advanced Feature
Templates with the desired Timer durations and assign different Templates (with different Timer durations) to the
extensions as desired.
Allow External Call Forward for

The types of calls to be forwarded to the external number may be selected in the parameter "Allow External Call
Forward for" in the “Station Advanced Feature Template” applied on the extensions which are allowed Call
Forward.
You may select from 'Internal Calls', 'Trunk Calls' and 'Internal + Trunk Calls'. By default, only trunk calls are
forwarded to external numbers in the default Station Advanced Feature Template 01 which is assigned to all
extensions.
If you want to set different call types for different extensions, then prepare separate Station Advanced Feature
Templates with the desired Call Types and assign these different Templates to the extensions as appropriate.
Refer the topic “Station Advanced Feature Template” for instructions on customizing the template and applying the
template to extensions using Jeeves and from a Telephone.
Changing Call Forward No-Reply Timer using Jeeves


• Select an Advanced Feature Template number. (by default Template 01 is assigned to all extensions)
• Go to the column Call Forward No-Reply Timer (Sec.).

• Change to the desired value.
• Apply the Template now configured with the Call Forward No-Reply Timer to the extensions.
Changing Call Forward No-Reply Timer using a Telephone

• Dial command 5602-1-Station Advanced Feature Template Number-02-Call Forward No-Reply Timer
Where,
Station Advanced Feature Template is from 01 to 50. Default: 50.
Timer is from 001 to 255 seconds.
02 is the parameter number for "Call Forward No-Reply Timer" in the Template.
For example: To program Call Forward No-Reply Timer as '60' secs.' in Template number 02, dial 5602-1-
02-02-060
• Exit SE mode.
Refer the topic “Station Advanced Feature Template” for instructions on applying the template to extensions
using Jeeves and from a Telephone.
When Call Forward No-Reply is set on a phone that is programmed in a Trunk Landing Group, the calls will
be forwarded on expiry of 'Call Forward No-Reply Timer' programmed in the routing group for this member
phone. Call Forward No-Reply Timer, programmed in Station Advanced Feature Template will not be
applied in this case.

Settings Call Forward for All Extensions using SA Jeeves
• Log in to Jeeves as System Administrator.
• Click Call Forward- All Extension.
• To set Call Forward of all Extensions to the Voice Mail, select Forward Calls of all Extensions to
Voice Mail. Select the type of Call Forward—Unconditionally, When Busy, When No Reply, When Busy
or No Reply you want to set.
• To set Call Forward of all Extensions to an Extension, select Forward Calls of all Extensions to
Extension Configure the Extension Number to which the calls are to be forwarded. Select the type of
Call Forward—Unconditionally, When Busy, When No Reply, When Busy or No Reply you want to set.
• To Cancel Call Forward, select Cancel Call Forward of all Extensions.
• Click Submit to save the option you select.
How to use
Call Forward can be set/canceled by extension users who are allowed this feature. It can be set/canceled by an
extension user for another extension (refer “Call Forward-Remote” to know more).

To set Call Forward,
• Press the 'Forward' key.

OR
• Dial 13.
• Scroll to select the desired Call Forward Type.
• Enter destination Phone Number/Voice Mail System/ Department Group Number.
To set Call Forward-Dual Ring,
• First, set the desired Call Forward type.

• Press the 'Forward' key.
OR
• Dial 13.
• Scroll to select Dual Ring.

• Select Dual Ring ON
• Press ‘Enter’ key.
• If the call is to be forwarded to an extension, dial the extension number.
• If the call is to be forwarded to an external number, dial Trunk Access Code, then the external phone
number and terminate the command with #*.
• For users world wide, Trunk Access Code (TAC) for dialing external numbers are: 0, 5, 61, 62, 63,
64.
• For users in USA, TAC for dialing external numbers are: 9, 5, 81, 82, 83, 84.
• If call is to be forwarded on voice mail, dial the Access Code for the Voice Mail System. The default
Access Code is 3931. Verify with the System Engineer if the default VMS Access Code has been
changed and use the new code to dial the VMS.
To cancel Call Forward,

• Press 'Forward' Key again.
OR
• Dial 13.
• Select 'Cancel'.
• Replace Handset on the cradle or you get dial tone after 3 seconds.
For SLT Users

• Dial 131 for Call Forward - All Calls
• Dial 132 for Call Forward - If Busy
• Dial 133 for Call Forward -If No Reply
• Dial 134 for Call Forward -If Busy or No Reply
• Dial destination Phone Number/Voice Mail System/Department Number.

• Set the desired Call Forward type.
• Dial 136-1 for Call Forward -Dual Ring

• Dial 130.
To disable Dual Ring,

• Dial 136.
• Dial 0.
• Replace Handset.

Call Forward-Remote
What's this?
An extension user can set Call Forward for another ('remote') extension from his/her own extension. Thus, Call
Forward set for an extension from another extension is called 'Call Forward-Remote'.
This feature can be used by the Operator or the Receptionist to forward the calls for the Managers and other
extension users to the destinations where they will be available.
This feature is also useful in Hotels, where the Front Desk can set Call Forward for guests. Refer the SARVAM
UCS Hospitality System Manual to know how this feature can be used in hotels.
• Call Forward-Remote is possible only from the System Administration (SA) mode.
• Call Forward-Remote is not supported by Q-Signaling (QSIG).
How it works
This feature works in the same way as Call Forward. The only difference is that it is set by one extension user for
another extension.
For example:
• A and B are extension users.
• A needs to forward calls for B's extension to another extension 'C' or an external number or a Voice Mail
System or a Department Number.
• A dials the Call Forward-Remote feature code followed by B's extension number, the destination number
where the calls for B should land.
• The system routes all incoming calls for B to the destination number.
How to configure
As Call Forward-Remote can be invoked only from the SA mode, either the feature 'SA Mode' or 'SA Extension'
must be enabled in the Class of Service of extensions that are to be allowed this feature.
It can also be set from Jeeves by the SA.
The feature 'SA Mode' requires a password to be dialed. Users must be provided a password to use this
feature from their extensions. The feature 'SA Extension' allows entry into SA mode, without a password.
SARVAM UCS. This Template is assigned CoS group 01 by default. The default CoS group 01 has both 'SA Mode'
and SA Extension disabled.

You may decide which extensions should be allowed Call Forward-Remote feature. In general practice only very
few extensions are allowed this feature.
So, to allow this feature to a few extensions only:
a. Define a CoS group with either 'SA Mode' or 'SA Extension' enabled. Recall that the facility 'SA Mode' is
password protected, so the extensions allowed access to this feature must also be provided an SA
Password.
c. Assign this new Template to the extensions to which Call Forward-Remote is to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed instructions and
programming.
How to use
Settings Call Forward-Remote using SA Jeeves

• Now, Search Extension, by entering either extension number as Extension Number, or by entering the
name of the extension as Extension Name.
• Click Submit.
• The searched extension users details appear on your screen.

• Click Call Forward to expand.
• Select the type of Call Forward you want to set for the extension:
• To forward calls to voice mail, select Forward Calls to Voice Mail and the type of call forward. Default:
Unconditionally.
• To forward calls of this extension to another extension, select Forward Calls to Phone and the type of
call forward. Default: Unconditionally.
Enter the extension number to which calls must be forwarded.
• To forward calls of this extension to an external number, select Forward Calls to External Number
and select the type of call forward. Default: Unconditionally.
Enter the external number to which calls must be forwarded.
• Click the Call Forward button to set Call Forward.
The message “Call Forward is set” appears.
• Click the Dual Ring button to set Call-Forward Remote with Dual Ring.
The message “Dual ring is On” appears.
• To set Call Forward on another extension, follow the same instructions as above.
• You can also forward the calls of all extensions at one go to the same destination. To do this,

• Click Call Forward-All Extensions.
• Select the Forward Calls of all Extensions and select the Call Forward type.
• In to Extension, enter the Call Forward destination number.
• To cancel, click Cancel Call Forward of All extensions.

To set Call Forward-Remote:
• Press the DSS Key assigned to 'Call Forward-Remote'.
OR
• Dial 1072-006.
• Enter the Destination Phone Number.
• Scroll to select the desired Call Forward Type:
• All Calls.
• If Busy.
• If No Reply.
• If Busy or No Reply.
• Dual Ring.
• Enter Destination Phone Number222/Voice Mail System223/Department Group.
• You get a confirmation tone and a text message for the Call Forward type set.
222. If call is to be forwarded to an extension of the SARVAM UCS, dial the extension number. If call is to be forwarded on an external
number, dial Trunk Access Code, then dial the external phone number and terminate the command with #*.
For users world wide, Trunk Access Code (TAC) for dialing external numbers are: 0, 5, 61, 62, 63, 64. For users in USA, TAC for
dialing external numbers are: 9, 5, 81, 82, 83, 84.
223. If call is to be forwarded on voice mail, dial the Access Code for the Voice Mail System. The default Access Code is 3931.

To cancel Call Forward set for an extension:
• Press the DSS Key assigned to 'Call Forward-Remote'.
OR
• Dial 1072-006.
• Scroll to select 'Cancel'.
• You get a confirmation tone and text message for Call Forward canceled.
For SLT Users

To set Call Forward-Remote:
• Lift handset.
• Dial 1072-006.
• Dial 1 for All Calls
• Dial 2 for If Busy
• Dial 3 for If No Reply
• Dial 4 for If Busy or No Reply
• Dial 5 for Dual Ring
• Dial destination Phone Number/Voice Mail System.
To cancel Call Forward Remote:

• Dial 1072-006.
• Dial the Extension Number.
• Dial 0.

Call Forward-Scheduled
What's this?
Extension users may want their calls to be automatically forwarded to a desired destination number during working
hours or non-working hours. To cite an example, a Support Technician spends working hours on the field and
wants all incoming calls on his extension in the office to be forwarded to his cell phone during working hours. During
non-working hours, he wants call calls to be forwarded to his voice mail.
Remembering to set and cancel Call Forward and changing the destination number for each Time Zone, that is,
working hours, non-working hours, break hours, every day proves to be cumbersome for such extension users.
In addition to “Call Forward”, SARVAM UCS supports 'Call Forward - Scheduled', which allows extension users to
set call forward for desired Time Zones at one time, and the system automatically forwards the calls to the
destination defined for each Time Zone.
How it works
Call Forward-Scheduled supports all the forwarding options as Call Forward: Unconditionally, If Busy, If No Reply, If
Busy or No Reply, Dual Ring.
Any of these options can be set for the three Time Zones: working hours, break hours and non-working hours.
The destination for Call Forward-Scheduled can be an internal (extension) number or an external number.
Both 'Call Forward' and Call Forward-Scheduled can be set on the same extension. In this case, priority is given to
'Call Forward' over Call Forward-Scheduled.
The logic for forwarding calls to the destination number remains the same as described in the topic “Call Forward”,
illustrated in the following example.
• Extension user A sets Call Forward-Unconditionally to extension B for Non-working hours.
• When there is a call on extension A, the system first checks if there is any 'Call Forward' type (that is,
Unconditional, Busy, No Reply, Busy/No Reply, Call Follow Me) set on extension A.
• If 'Call Forward' is set on extension A, the system will follow the logic described in 'How it works' under the
topic 'Call Forward".
• If no 'Call Forward' is set on extension A, the system will check if Call Forward-Scheduled is set on A.
• Since Call Forward-Scheduled is set on extension A, the system will compare the Time Zone for which the
Call Forward is scheduled with the current Time Zone of extension A.
• If the current Time Zone of extension A is the same as the Time Zone set for Call Forward Scheduled, that
is, non-working hours, the call will be forwarded to extension B as per the call forward type set.
• As the Call Forward Type set by A is Unconditional, the system will forward the call to B, without checking
for the Busy Tone and without waiting for the Call Forward No-Reply Timer to expire.

• If the current Time Zone of extension A is not the same as Time Zone set for Call Forward-Scheduled, the
call will not be forwarded. The system will consider that no call forward has been set.
• Call Forward - Scheduled can be set simultaneously for more than one Time Zone from the same
extension. For example, extension A can set Call Forward-Scheduled for working hours, then again set
Call Forward-Scheduled for non-working hours, and again for break hours.
• A different Call Forward Type can be set for a different Time Zone. For example, extension A can set
Call Forward -Unconditional for non-working hours, and Call Forward -Busy for working hours. Also, a
different destination number can be set for forwarding calls in each Time Zone. For example, extension
A can set Call Forward-Unconditional for non-working hours to a mobile number and set extension B as
destination number for working hours.
• When more than one Call Forward type is set on the same extension for the same Time Zone, the
latest Call Forward type set for the Time Zone will override the previous Call Forward type set for that
Time Zone. For example, extension A sets Call Forward -Busy for working hours, then sets Call
Forward Busy or No Reply for working hours, the latter will override the former. The system will
consider the latest, that is, Busy or No Reply as the Call Forward type for forwarding calls during
working hours.
• Call Forward-Scheduled can be cancelled individually for a desired Time Zone or all at once for all
Time Zones.
• Call Forward-Scheduled can be set by extension users as well as for extension users from the System
Administrator mode.
• It is also possible to select the types of calls, that is, internal calls only, or trunk calls, or both, to be
forwarded to external numbers. You can program the system to forward internal calls only, or trunk
calls only or both trunk calls and internal calls to the external number. For this, the parameter 'Allow
External Call Forward for' must be programmed in the “Station Advanced Feature Template” of the
extensions that want to use Call Forward-Scheduled.
• Call Forward-Scheduled when set/canceled from the SA mode, will not depend on the assigned CoS.
How to configure
The programming of this feature involves the same parameters as in “Call Forward”.
'Call Forward' must be enabled in the Class of Service (CoS) group of the extensions to which this feature is to be
allowed. Refer the topic "Call Forward".
If Call Forward No-Reply is to be set, and if required, the Call Forward No-Reply Timer may be programmed in the
“Station Advanced Feature Template” applied on the extensions which are to be allowed this feature. Refer the
topic “Call Forward”.
The types of calls to be forwarded to the external number may be selected in the parameter "Allow External Call
Forward for" in the “Station Advanced Feature Template” applied on the extensions which are allowed Call
Forward-Scheduled. You may select from 'Internal Calls', 'Trunk Calls' and 'Internal + Trunk Calls'. By default, only
trunk calls are forwarded to external numbers.
Extensions that are to be allowed to set Call Forward-Scheduled for other extensions must be allowed either the
feature 'SA Mode' or 'SA Extension' in their COS. Refer the topic “Call Forward-Remote”.

How to use
Call Forward-Scheduled can be set/canceled by users for their own extension, or for any other extension from the
SA mode.
Setting Call Forward-Scheduled for Extension Users

The Operator or any extension user having access to System Administrator mode can set or cancel Call Forward-
Scheduled for other extension users using Jeeves or by dialing SA commands from a telephone.
Setting Call Forward-Scheduled using Jeeves

• Click Submit.

• Click Call Forward - Scheduled to expand.
• You can set Call Forward - Scheduled for Working Hours, Break Hours as well as for Non-working Hours
To set Call Forward for the extension for the Working Hours, under Working Hours, select the type of Call
Forward:
• To forward calls to voice mail, select the radio button Forward Calls to Voice Mail and the type of call
forward from the drop down list. Default: Unconditionally.
• To forward calls of this extension to another extension, select Forward Calls to Phone radio button
and the type of call forward for this option from the drop down list. Default: Unconditionally.
Enter the extension number to which calls must be forwarded in the empty field provided for this option.
• To forward calls of this extension to an external number, select Forward Calls to External Number
radio button, and select the type of call forward for this option from the drop down list. Default:
Unconditionally.
Enter the external number to which calls must be forwarded in the empty field provided for this option.
• Click the Apply Call Forward button to set Call Forward -Scheduled.
The message “Call Forward is set” appears.
• Click the Dual Ring button to set Call Forward-Scheduled with Dual Ring.
The message “Dual ring is On” appears.
• To set call forward for Break Hours and Non-working Hours, follow the same instructions as above.
• To set Call Forward - Scheduled for another extension, follow the same instructions as above.

Setting Call Forward-Scheduled using a Telephone
• The destination number for forwarding calls can be a maximum of 16 digits. Terminate the command
with #* if destination number has fewer than 16 digits.
• If the destination number is an external number, enter the Trunk Access Code followed by the
destination number.
Call Forward-Scheduled by Extension Users
To set Call Forward-Scheduled:
• Press DSS key assigned to Call Forward-Scheduled.

OR
• Dial 1175.
• Scroll to the desired Time Zone.
• Press Enter key to select Time Zone.
• Scroll to the desired Call Forward type for the selected Time Zone.
• Press Enter key to select Call Forward type.
• Enter Destination Number on prompt.
• You get confirmation tone and message showing extension to which Call Forward is set.
To cancel Call Forward-Scheduled for a Time Zone:

OR
• Dial 1175.
• Scroll to select Cancel.
• Press Enter key.
• You get confirmation tone and message.
To cancel Call Forward-Scheduled in all Time Zones:

OR
• Dial 1175.
• Scroll to 'Cancel Call Forward'.
Call Forward-Scheduled from SA mode
To set Call Forward-Scheduled:
• Press DSS key assigned to Call Forward-Scheduled - Remote.

OR
• Dial 1072-223.
• Enter extension number (from which calls are to be forwarded)

• Scroll to the desired Call Forward type for the selected Time Zone.
• Press Enter key to select Call Forward type.
• Enter Destination Number on prompt.
• You get confirmation tone and message showing extension to which Call Forward is set.
To cancel Call Forward-Scheduled for a Time Zone:

OR
• Dial 1072-223
• Enter extension number (for which it is to be canceled)
• Scroll to select Cancel.
To cancel Call Forward-Scheduled in all Time Zones:

OR
• Dial 1072-223
• Enter extension number (for which it is to be canceled)
• Scroll to 'Cancel Call Forward'.
For SLT Users

Call Forward (CF)-Scheduled by Extension Users
To set/cancel CF-Scheduled for working hours:
• Lift handset.
• Dial 1175-1-1-Destination Number for CF-Scheduled-Unconditional.
• Dial 1175-1-2-Destination Number for CF-Scheduled -Busy.
• Dial 1175-1-3-Destination Number for CF-Scheduled -No Reply.
• Dial 1175-1-4-Destination Number for CF-Scheduled-Busy/No Reply.
• Dial 1175-1-5-1 for CF-Scheduled -Dual Ring.
• Dial 1175-1-5-0 to cancel CF-Scheduled -Dual Ring.
• Dial 1175-1-0 to cancel CF-Scheduled for Working Hours.
To set/cancel CF-Scheduled for break hours:
• Lift handset.
• Dial 1175-2-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1175-2-4-Destination Number for CF-Scheduled -Busy/No Reply.
• Dial 1175-2-0 to cancel CF-Scheduled for Break Hours.

To set/cancel CF-Scheduled for non-working hours:
• Lift handset.
• Dial 1175-3-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1175-3-4-Destination Number for CF-Scheduled -Busy/No Reply.
• Dial 1175-3-0 to cancel CF-Scheduled for Non-working Hours.
To cancel CF-Scheduled for all Time Zones:

• Lift handset.
• Dial 1175-0.
Call Forward (CF)-Scheduled from SA mode
To set CF-Scheduled for working hours:
• Lift handset.
• Dial 1072-223-Extension number-1-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1072-223-Extension number-1-2-Destination Number for CF-Scheduled -Busy.
• Dial 1072-223-Extension number-1-3-Destination Number for CF-Scheduled -No Reply.
• Dial 1072-223-Extension number-1-4-Destination Number for CF-Scheduled -Busy/No Reply.
• Dial 1072-223-Extension number-1-5-1 for CF-Scheduled -Dual Ring.
• Dial 1072-223-Extension number-1-5-0 to cancel CF-Scheduled -Dual Ring.
• Dial 1072-223-Extension number-1-0 to cancel CF-Scheduled -for working hours.
To set Call Forward-Scheduled for break hours:
• Lift handset.
• Dial 1072-223-Extension number-2-5-1 for CF-Scheduled -Dual Ring.
• Dial 1072-223-Extension number-2-5-0 to cancel CF-Scheduled -Dual Ring.
• Dial 1072-223-Extension number-2-0 to cancel CF-Scheduled -for break hours.
To set Call Forward-Scheduled for non-working hours:
• Lift handset.
• Dial 1072-223-Extension number-3-5-1 for CF-Scheduled-Dual Ring.
• Dial 1072-223-Extension number-3-5-0 to cancel CF-Scheduled-Dual Ring.
• Dial 1072-223-Extension number-3-0 to cancel CF-Scheduled - for break hours.

To cancel Call Forward-Scheduled for all Time Zones:
• Lift handset.
• Dial 1072-223-Extension Number-0

Call Forward-When Not Registered
SIP Phones connected as extensions may fail to register with SARVAM UCS when the network link is down or
when there is power failure. Using the Call Forward-When Not Registered feature, the extension users can have
their calls forwarded even when their extension phone is not registered with SARVAM UCS.
The destination for ‘Call Forward-When Not Registered’ can be an internal number, an external number or the
Voice Mail.
It is also possible to select the types of calls—internal calls only, or trunk calls, or both—to be forwarded to external
numbers.
Call Forward-When Not Registered can be set/canceled by,

• the System Administrator mode.
• SIP phone users from their phones.
Call Forward- When Not Registered can also be set for each Time Zone—Working Hours, Break Hours, Non-
working Hours, by setting Call Forward-When Not Registered - Scheduled.
Call Forward - When Not Registered-Scheduled can be set for more than one Time Zone at a time on the same SIP
phone. It can be canceled individually for a desired Time Zone, or all at once for all Time Zones. A different
destination number can be set for forwarding calls in each Time Zone. For example, the destination number for
non-working hours can be a mobile number and the destination number for working hours can be another
extension number.
If the SIP Extension user is a VARTA AMP100 client (the application is in the background) and,
• if the incoming call is received from a QSIG caller, Call Forward functionality will not be achieved. To
know more refer to the MATRIX VARTA AMP100 User Guide.
• if the client is a member in any Routing Group or Department Group, Call Forward functionality will not
be achieved. To know more refer to the MATRIX VARTA AMP100 User Guide.
• If ‘Call Forward-Unconditional’ and ‘Call Forward-When Not Registered’, have been set on the same SIP
phone. ‘Call Forward-Unconditional’ will have priority over ‘Call Forward-When Not Registered’.
• If ‘Call Forward-Scheduled-Unconditional’ and ‘Call Forward-When Not Registered-Scheduled’, have been

set on the same SIP phone. ‘Call Forward - Scheduled - Unconditional’ will have priority over ‘Call
Forward-When Not Registered-Scheduled’.
How to configure
The Call Forward-When Not Registered feature does not require any specific programming except:
• ensuring that 'Call Forward' in the “Class of Service (COS)” group in the “Station Basic Feature Template
Parameters” applied to the SIP phones.
• if required, selecting the types of calls to be forwarded to the external number. By default, only trunk calls
are forwarded to external numbers. If you want to select a different type of call, configure the parameter

“Allow External Call Forward for” in the Station Advanced Feature Template applied to the SIP phones.
Refer the sub-topic “Station Advanced Feature Template”, under Configuring Extensions.
• If you want to allow Call Forward-When Not Registered to be set only by the System Administrator (SA) for
the extension users, the System Engineer (SE) must disable ‘Call Forward’ feature in the Class of Service
(CoS) group in the Station Basic Feature Template applied to the SIP phones.
If you disable ‘Call Forward’ in the CoS of a SIP phone, the user will not be able to set any other type of
Call Forward.
Setting Call Forward-When Not Registered

Call Forward-When Not Registered can be set from
• the SA mode from Jeeves.
• the SIP phones connected as extensions.
Call Forward-When Not Registered set by SA

• Now, Search Extension on which you want to set this feature. You search by entering either Extension
Number or the Extension Name.
• Click Submit.
• The searched extension user details appear on your screen.

• Click Call Forward When Not Registered.
• Select the destination for forwarding calls when the SIP Extension fails to register from the following:
• Forward Calls to Voice Mail.
• Forward Calls to Extension Number. If you select this option, you must enter the desired
Extension Number in the corresponding box.
• Forward Calls to External Number. If you select this option, you must enter the desired external
number in the corresponding box. Also, assign a trunk to route the call by selecting the Trunk
Access Code from the using TAC list.
• Click the Apply Call Forward button. The message “Call Forward is set” appears.
• To set time-zone based Call Forward - When Not Registered, click Call Forward When Not Registered-
Scheduled to expand.
• To set Call Forward When Not Registered for working hours, under Working Hours, select the desired
destination from the following options:

• Forward Calls to Voice Mail.
• Forward Calls to Extension Number. If you select this option, you must enter the desired
Extension Number in the corresponding box.
• Forward Calls to External Number. If you select this option, you must enter the desired number in
the corresponding box, and assign a trunk to route the call by selecting the Trunk Access Code in
the using TAC list.
• Click the Apply Call Forward button. The message “Call Forward is set” appears.
• To set call forward for Break Hours and Non-working Hours, follow the same instructions as above.
• To set Call Forward When Not Registered - Scheduled for another extension, follow the same
instructions as above.
Call Forward-When Not Registered set/canceled by SIP Phone Users

SIP extension users can set/cancel Call Forward-When Not Registered from their SIP phones. The SIP phone may
be a Matrix Extended IP Phone or any open Open SIP phone.
Using Matrix Extended IP Phone

• Lift handset.
• Press DSS key assigned to Call Forward-When Not Registered (if programmed).
OR
• Dial *13.
• Scroll to the desired option.
To set Call Forward - When Not Registered regardless of time-zone,

• Select ‘Always’ and press 'Enter' key.
• Select ‘Set’ and press ‘Enter’ key.
To set Call Forward When Not Registered - Scheduled,

• Select ‘Working Hours’/’Break Hours’/’Non-Working Hours’, and press ‘Enter’ key.
• Select ‘Set’ and press ‘Enter’ key.
• On the prompt, ‘Forward to Number’, enter the Destination Number—Extension Number/External Number/
Voice Mail System.
• If the you want to route the calls to the Voice Mail, enter the VMS Access Code as the destination
number.
destination number and #*.
To cancel Call Forward - When Not Registered,

• Lift handset.

OR
• Dial *13.
• Select ‘Always’ and press 'Enter' key.
• Select ‘Cancel’ and press ‘Enter’ key.
To cancel Call Forward When Not Registered - Scheduled for each Time Zone,
• Lift handset.
OR
• Dial *13.
• Select the desired time-zone ‘Working Hours’/’Break Hours’/’Non-Working Hours’, and press ‘Enter’ key.
• Select ‘Cancel’ and press ‘Enter’ key.
To cancel All Call Forward When Not Registered,

OR
• Dial *13.
• Select ‘Cancel Call Forward’ and press ‘Enter’ key.
Using Standard IP Phone

• Lift handset.
To set Call Forward - When Not Registered regardless of time-zone,

• Dial *13-1-1-Destination Number
• If the you want to route the calls to the Voice Mail, enter the VMS Access Code as the destination
number.
destination number and #*.
To set Call Forward - When Not Registered - Scheduled,

• Dial *13-2-1-Destination Number for working hours.
• Dial *13-3-1-Destination Number for break hours.
• Dial *13-4-1-Destination Number for non-working hours.
To cancel Call Forward - When Not Registered,

• Lift handset.
• Dial *13-1-0.
To cancel Call Forward When Not Registered - Scheduled,

• Dial *13-2-0 for working hours
• Dial *13-3-0 for break hours.
• Dial *13-4-0 for non-working hours.
To cancel All Call Forward-When Not Registered,

• Lift handset.
• Dial *13-0.


Call Hold
What's this?
Call Hold enables you to put an on-going conversation (with an internal or external number) on hold. SARVAM UCS
offers three types of Call Hold:
• Exclusive Hold: An on-going conversation is put on hold from a DKP/Extended IP Phone and is retrieved
from the same DKP/Extended IP Phone that put it on hold.
• Global Hold: An on-going conversation is put on hold from a DKP/Extended IP Phone and is retrieved
from any DKP/Extended IP Phone connected to SARVAM UCS.
• Consultation Hold: An on-going conversation is put on hold in order to perform any further activity, such
as Call Transfer, Conference, Call Toggle.
Exclusive Hold and Global Hold are supported on SIP and DKP extensions.
Consultation Hold is supported on the SIP, DKP and SLT extensions.
SARVAM UCS supports interoperability with the Polycom IP Phones. When any extension of SARVAM
UCS puts a SIP Extension on hold (Exclusive, Global or Consultation Hold), SARVAM UCS will send Re-
Invite message to the SIP Extension put on hold.
How it works
Exclusive Hold using the Hold Feature Key

When a call is put on Exclusive Hold,
• SARVAM UCS starts the Exclusive Hold Retrieval Timer (programmable; default: 2 minutes).
• The call remains on hold for the duration of this timer.
• The extension user can retrieve the call within this timer.
• If the call is not retrieved before the expiry of this timer, it will return back to the DKP/Extended IP Phone
that has put the call on hold. The DKP/Extended IP Phone rings and the user may answer the call.
• The returned call is disconnected, if the DKP/Extended IP Phone is not in the idle state, or if the call is not
answered by the DKP/Extended IP Phone user.
A call placed on Exclusive Hold can be retrieved in the following ways:
• Pressing the Hold key again (when the DKP/Extended IP Phone is idle).
• Pressing the Call Appearance key of the call put on hold (when the DKP/Extended IP Phone is busy).
• Pressing DSS key assigned to the Trunk/extension you put on hold.

• Answering the call, when it returns at the end of the Exclusive Hold Retrieval Timer.
To be able to place calls on Exclusive Hold, you must select Exclusive Hold as the Default Call Hold Type in the
System Parameters. See “System Parameters” for instructions.
If multiple calls have been put on Exclusive Hold and you press the Hold key, then the last call that was put
on hold will be retrieved.
Exclusive Hold using DSS Key
DKP and/or Extended IP Phone users can configure upto 8 DSS keys for Exclusive Hold, namely Exclusive Hold1
to 8. Using DSS key assigned to Exclusive Hold calls can be put on Exclusive Hold only. The functioning of the
DSS key does not depend on the Default Call Hold Type you select.
For instructions to assign a DSS Key to Exclusive Hold, see “DSS Keys Programming”.
When a call is put on Exclusive Hold using DSS key assigned to Exclusive Hold 1,
• SARVAM UCS starts the Exclusive Hold Retrieval Timer (programmable; default: 2 minutes).
• The call remains on hold for the duration of this timer.
• The extension user can retrieve the call within this timer.
• If the call is not retrieved before the expiry of this timer, it will return back to the DKP/Extended IP Phone
that has put the call on hold. The DKP/Extended IP Phone rings and the user may answer the call. The
LCD displays the message ‘Held X Recall’, where X is the hold position number.
• The returned call is disconnected, if the DKP/Extended IP Phone is not in idle state, or if the call is not
answered by DKP/Extended IP Phone user.
A call placed on Exclusive Hold can be retrieved in the following ways:
• Pressing the DSS key (when the DKP/Extended IP Phone is idle).
• Pressing the Call Appearance key of the call put on hold (when the DKP/Extended IP Phone is busy).
• Pressing DSS key assigned to the Trunk/extension you put on hold.
• Answering the call, when it returns at the end of the Exclusive Hold Retrieval Timer.
Global Hold
When a call is placed on Global Hold,
• The call remains connected in the system. The call remains on hold for the duration of the Global Hold
Retrieval Timer (programmable; default: 60 seconds).
• Any DKP/Extended IP Phone connected to the SARVAM UCS can pick up the call put on Global hold by:
• Pressing DSS key assigned to the Trunk put on Global Hold.

• Pressing the DSS key assigned to the extension put on Global Hold.
• If this call is not retrieved before the expiry of the Global Hold Retrieval Timer, the call is returned to the
DKP/Extended IP Phone which put it on hold. The DKP/Extended IP Phone rings and the user may answer
the call.
• The returned call is disconnected, if the DKP/Extended IP Phone is not in idle state, or if the call is not
answered by DKP/Extended IP Phone.
To be able to place calls on Global Hold, you must select 'Global Hold' as the Default Call Hold Type in the System
Parameters of SARVAM UCS. The DKP/Extended IP Phone (which picks up the call) must have a DSS Key to
access the Trunk or the Extension which is put on hold. See “System Parameters” for instructions.
SARVAM UCS provides the flexibility to use Exclusive Hold and Global Hold at the same time. You can put
calls on Exclusive Hold even when Global Hold is enabled in the system using the Hold Feature key only.
SARVAM UCS does not support Global Hold on SIP Trunks.
You must first retrieve the call that is put on Exclusive or Global Hold, if it is to be transferred or included in
a Conference.
Consultation Hold
During an on-going conversation, any SLT, DKP or Extended IP Phone can place a call on Consultation Hold to
perform any of the following:
• “Call Transfer”
• “Call Toggle”
• “Conference-3 Party”, “Conference-Multiparty”, “Conference Dial-In”
• “Call Park”
• “Mute”
• “Call Chaining”
• “Conversation Recording”
• “Flashing on Trunks (Continued Dialing)”
• “Priority Calls in E&M MFCR2 Signaling”
The call is released from the held state once the operation has been performed or canceled.
For instructions on using the above mentioned features, refer How to Use of the respective feature.
How to configure
For Exclusive and Global Hold, you must configure the following parameters:
• Class of Service: Call Hold must be enabled in the Class of Service (CoS) of the DKPs/Extended IP
Phones you want to allow this feature.
In the default Station Basic Feature Template 01 assigned to all extensions of SARVAM UCS, Call Hold is
included in the 'Basic Features' assigned to all Class of Service groups, including the default CoS group
01. So, all extensions of SARVAM UCS can use this feature.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” to know more.

• Call Hold Type: Enable the desired option, that is, Exclusive Hold or Global Hold in the “System
Parameters”.
• Send Re-INVITE over SIP Trunk on Hold: When an external call over a SIP Trunk is put on hold by any
extension, and you want SARVAM UCS to send Re-INVITE message over SIP Trunk to the remote end,
you must enable this flag on the SIP Trunk. See “Configuring SIP Trunks” to know more.
• DSS Keys: Program DSS Keys for Trunks and Stations on the DKPs which are allowed to retrieve calls on
Global Hold. Program the DSS Keys for Exclusive Hold, if required. Refer the topic “DSS Keys
Programming”, “DSS Key Settings” in “Configuring Matrix SPARSH VP330”, “DSS Key Settings” in
“Configuring Matrix SPARSH VP248”, “DSS Key Settings” in “Configuring Matrix SPARSH VP310” and
“DSS Key Settings” in “Configuring Matrix SPARSH VP510” for instructions.
• Global Hold Retrieval Timer: Change the default setting of this timer to the desired duration, if required.
• Exclusive Hold Retrieval Timer: Change the default setting of this timer to the desired duration, if
required.
For instructions to change the Timers, see “System Timers and Counts”.
For Consultation Hold to work, Call Hold must be enabled in the “Class of Service (COS)” of the SLTs, DKPs/
Extended IP Phones you want to allow this feature.
How to use
Exclusive Hold

To put a call on Exclusive Hold, when Exclusive Hold is selected as the Default Call Hold Type:
• You are in speech with a Trunk/with an extension.

• Press ‘Hold’ Key.
• Go idle.
• Call with Trunk/extension is put on 'Exclusive Hold'.
To retrieve the call:
• Press ‘Hold’ Key again.

• Press Call Appearance key of your DKP/Extended IP Phone
• Press DSS Key of the Trunk/extension you put on hold from your DKP/Extended IP Phone.
To put a call on Exclusive Hold, when Global Hold is enabled:

• Press ‘Hold’ Key twice in quick succession within 2 seconds.
• Go idle.
To retrieve the call:
• Press ‘Hold’ Key again.

To put a call on Exclusive Hold using DSS key assigned to Exclusive Hold:
• You are in speech with on a Trunk/with an extension.

• Press DSS key assigned to Exclusive Hold.
• Go idle.
To retrieve the call put on hold using DSS key:
• Press DSS Key again.

Using DSS key assigned to Exclusive Hold calls can be put on Exclusive Hold only. The functioning of the
DSS key does not depend on the Default Call Hold Type you select in the System Parameters.
Global Hold

To put a call on Global Hold, when Global Hold is selected as the Default Call Hold Type:

• Press ‘Hold’ Key.
• Go idle.
• Call with Trunk/extension is put on 'Global Hold'.
To retrieve a call on Global Hold:
• From any DKP/Extended IP Phone, press the DSS Key of the Trunk/extension put on Global Hold.
To put a call on Global Hold, when Exclusive Hold is enabled:

• Press ‘Hold’ Key twice in quick succession within 2 seconds.
• Go idle.
• Call with Trunk/extension is put on 'Global Hold'.
To retrieve a call on Global Hold:
• From any DKP/Extended IP Phone,

• Press the DSS Key of the Trunk/extension put on Global hold.

Consultation Hold

To put a call on Consultation Hold:
• Press the DSS Key assigned to the feature (if configured).

OR
• Press Transfer key.
For SLT Users

To put a call on Consultation Hold:
• Press Flash.
OR
• Tap the Hook switch of your phone.

Call Park
What is this?
Call Park allows you to place a call on Hold, so it can be retrieved from the same or another extension of the
system.
A call is 'parked' when the extension user temporarily places the call into a location in the system called 'Orbit'. The
user can attend to other calls. The parked call can be retrieved on completion of the current call by dialing the Orbit
number.
Call Parking is useful in offices housed in different parts of a building or multi-storied offices. It is useful in situations
like:
• the person who picked up the call is not the desired called party or the desired party is at an unknown
location. The person who picked up the call can then either go to find the desired called party or call other
numbers to find him/her. When found, the desired called party can pick up the call from the same or any
extension by dialing the Orbit number.
• the person who picked up the call may have to go to another part of the office to look up a file or consult a
colleague. The person can park the call and continue the conversation from the other part of the office.
The SARVAM UCS offers two types of Call Park facility:
• Call Park-General Orbit: The extension user can park calls in any of the 8 'general' Orbits, which are like
fictional extensions located in the system. The calls parked in the General Orbit can be picked up from any
extension by dialing the General Orbit Number. At a time, only one call can be parked in each General
Orbit.
• Call Park-Personal Orbit: Each telephone instrument (EON/SLT/IP Phone) connected as extension has
one Personal Orbit. Calls parked in personal orbit can be picked up only from where the call is parked. So,
no other person can pick up this call. Multiple calls can be parked in the Personal Orbit at a time.
Extension users can park the call either in the General Orbit or the Personal Orbit by dialing an Orbit Number from
1 to 9, where:
• 1 is the Personal Orbit Number.
• 2 to 9 are General Orbit Numbers.
After parking a call, the extension user can continue to make and answer other calls and use other system features.
For Open SIP phones, SARVAM UCS supports Call Park and Retrieve using REFER Message. For a list
of IP phones on which this feature has been tested, see “SARVAM UCS Features tested on IP Phones of
different Brands”in the Appendix.

How it works
A and B are extension users. C, D and E are callers.
Parking Calls in General Orbit

• C calls B.
• A picks up the call.
• As B is not present at his extension, A parks the call in General Orbit Number 2 by dialing the Access
Code.
• C is played on-hold music.
• A tries to find B (by either calling several numbers or by going in person or sending someone).
• The parked call remains in orbit for the duration of the Call Park Timer, which is set to 2 minutes by default.
• A finds B.
• B retrieves the call from another extension by dialing the feature access code for retrieving Call Park and
Orbit number 2.
• If A cannot locate B or if B cannot attend the call, A can also retrieve the call from his extension.
However,
• If neither A nor B retrieves the parked call within the Call Park Timer, the system will hunt for the extension
that parked the call (A) on the expiry of the Call Park Timer.
• Meanwhile, if A is busy, the system again keeps the call parked in orbit number 2 for the period of the Call
Park Timer. This process continues for the duration of the Call Park Release Timer, which is set to 3
minutes by default.
• If A is free, the system will ring on A's phone. A gets connected to C again.
• If A does not retrieve the parked call till the end of the Call Park Release Timer, C gets disconnected.
Parking Calls in Personal Orbit

Parking calls in the Personal Orbit works the same way as in General Orbit. The only difference is that A can park
multiple calls by dialing the Personal Orbit Number. But calls can be retrieved from A's phone only.
When there are multiple calls to be retrieved from the Personal Orbit, they are retrieved one by one, without
following any particular sequence like FIFO or LIFO.
To be able to use 'Call Park', this feature must be enabled in the COS of the requesting extension.
However, for retrieving parked calls, the system does not check COS. So any extension can retrieve
parked calls.
How to configure
To provide this feature to extensions,

• Enable the Call Park in the “Class of Service (COS)” of the “Station Basic Feature Template” of the
extensions. By default, this feature is enabled in the COS of all extension types for all the time zones.
• If required, you may change the duration of the Call Park Timer and the Call Park Release timer. See
“System Timers and Counts” for instructions.

How to use

To park a call:
• You are in speech with extension/external caller.

• Press DSS Key assigned to 'Call Park'.
OR
• Press Transfer Key and dial 115
• Enter Orbit Number (1-9)
(Personal Orbit:1; General: 2-9).
To retrieve a parked call from your phone, when your phone is in idle state:
• Press DSS Key assigned to 'Call Park - Retrieve'.

OR
• Dial 116
• Enter Orbit Number where you parked the call (1-9)
• You are in speech with the extension/external caller.
To retrieve a parked call from your phone, when you are in speech with someone:
• Press Transfer Key.

• Press DSS Key assigned to 'Call Park - Retrieve'.
OR
• Dial 116
• Enter Orbit Number where you parked the call (1-9)
For SLT Users

To park a call:
• You are in speech with extension/external caller.

• Dial-Flash-115-Orbit Number
• Call is parked.
To retrieve a parked call from your phone, when your phone is in idle state:

• Dial 116-Orbit Number
To retrieve a parked call from your phone, when you are in speech with someone:
• Dial Flash -116-Orbit Number


Call Logs
What's this?
SARVAM UCS stores the details of 20 each, of the following types of calls:
• Missed calls: incoming calls that were not answered by extension users.
• Answered calls: incoming calls answered by extension users.
• Dialed calls: calls made by extension users.
The call history of each of the above types of calls is stored by Name, Number, and Date-Time of the Call.
If there is no name in the CLI of the above types of calls, the system stores and displays the Number and the Date-
Time. In case there is no number in the CLI, the system will display the Port number on/from which the call was
received/made.
The Call Logs contain details of both internal as well as external calls made or received by the extension users.
The Call Logs feature is supported on EON and Extended IP Phones.
Using call logs you can:
• view call history: you can see the calls you missed, answered or dialed.
• make calls: you can call any number that you have missed, answered, or dialed.
• edit the numbers: you can change or modify the number in the call log. This is useful when the CLI
received and stored in the call log is not in the same format that is to be used to make calls.
• save the numbers: you can store the external numbers in your call logs in the "Personal Directory" and
use them for “Personal Abbreviated Dialing”.
The maximum number of calls that can be stored under each Call Log type is 20. The logs will be cleared
automatically using the First-In, First-Out method, that is, the latest call detail will replace the record of the oldest
call detail.
Given the limited Call Log capacity, the system also allows you to choose if you want internal calls to be displayed
or not in the Missed, Answered and Dialed Call Logs. And accordingly it will store internal calls in the logs.
The system stores each Missed, Answered and Dialed call individually even if the same number is received
multiple times.
How to configure
This feature does not require any specific programming, except:
• Selecting whether internal calls should be logged in the Missed, Answered and Dialed Call Logs. This can
be done on the 'System Parameters' page of Jeeves or by using a Telephone.
• Programming of a DSS key for the Call Logs feature. For instructions please refer the topic “Configuring
DKP Extensions”, “DSS Keys Programming”, “DSS Key Settings” in “Configuring Matrix SPARSH VP330”,
“DSS Key Settings” in “Configuring Matrix SPARSH VP248”, “DSS Key Settings” in “Configuring Matrix
SPARSH VP310” and “DSS Key Settings” in “Configuring Matrix SPARSH VP510” for instructions.

Programming Internal Call Logging using Jeeves
• Under Configuration, click the System Parameters to open the page.
• You may enable any or all of the following flags by selecting the respective check box:
• Store Internal Calls in Missed Call Log
• Store Internal Calls in Answered Call Log
• Store Internal Calls in Dialed Call Log
• Store Internal Calls in Redial Call Log
• Log out of Jeeves or continue, as required.
Programming Internal Call Logging using a Telephone

• Enter SE mode.
To enable/disable Log Internal Calls in Missed Calls, dial:

• 5361-Code
Where,
Code is
0 for Disable (Do not store internal calls in "Missed Calls" log)
1 for Enable (Store internal calls in "Missed Calls" log)
By default, Internal Call Logs in Missed Calls is enabled.
To enable/disable Log Internal Calls in Answered Calls, dial:

• 5362-Code
Where,
Code is
0 for Disable (Do not store internal calls in "Answered" log)
1 for Enable (Store internal calls in "Answered Calls" log)
By default, Internal Call Logs in Answered Calls is enabled.

To enable/disable Log Internal Calls in Dialed Calls, dial:
• 5363-Code
Where,
Code is
0 for Disable (Do not store internal calls in "Dialed Calls" log)
1 for Enable (Store internal calls in "Dialed Calls" log)
By default, Internal Call Logs in Dialed Calls is enabled.
• Exit SE mode.
How to use
The Call Logs feature allows you to view calls and edit numbers, make calls to any number logged, and store
numbers.
Viewing Call Logs

There are two ways to view call logs:
• From the Phone Menu

• Using the Feature Key assigned to Call Logs.
If you are using a DSS Key for the Call Logs feature, whenever there is a missed call, the LED of the DSS
key will glow. If you press the Call Logs key, the system will display the last missed call details.
To view Call Logs from Phone Menu:
• Press Enter key when the phone is idle.

• Place your cursor on Call Logs option, press Enter key.
• Scroll with the Up/Down Navigation Key to reach the desired Call Log: Missed, Answered, Dialed.
• Press Enter key to select the desired Call Log.
• The phone will display the call log details.
• To view another call log, scroll with the Back Navigation key to return to the previous option.
• You may exit the Phone Menu by going ON-Hook or pressing the Cancel key.
• If there is no name in the CLI, the Call Log will only display the number.
• If you press the 'Enter' key, the system will dial out the number you just viewed.
To view Call Logs using DSS Key:
• Press DSS Key programmed for Call Logs, when the phone is idle.
• Scroll to select the desired Call Log: Missed, Answered, Dialed.
OR
• Press the DSS Key assigned the Call Logs feature, when it glows.
• The Phone will display the Call Logs: Missed, Answered, Dialed.
• Press Enter key to select the Missed Call Log.

• The LED of the Call Logs DSS key will be turned off once you have viewed the missed call.
Editing Numbers in Call Logs

• Go to Call Logs from the Phone Menu or by pressing Call Logs DSS Key. (see instructions given above).
• Scroll with the Up/Down Navigation Key to reach the desired number.
• To edit the number, move the cursor with the Forward (>) navigation key.
• Place the cursor under the digit you want to delete.
• Press 'Cancel’' key to delete a digit.
• To insert a digit, place the cursor where you want to insert the digit, and enter the digit using the dial pad.
The digit will be inserted in the number string accordingly.
• Repeat the same to delete/insert another digit.
• After editing the number, you may store it in the Personal Directory or dial the edited number by pressing
the Enter key.
The original number (you now changed) will remain unaffected in the Log. However, if you make a call to
the new number (you changed), it will be logged in the “Dialed” call log and the Last Number Redial list.
Making calls using Call Logs

• Go to Call Logs from the Phone Menu or by pressing Call Logs DSS Key.
• The system will dial out the selected number using the Outgoing Trunks assigned for dialing ‘0’.
• The dialed number will be logged in the "Dialed" call log and the Last Number Redial List.
Storing numbers of Call Logs in the Personal Memory of the Phone

To store any external call record (trunk call) from Received, Missed and Dialed call logs to Personal Directory,
• Go to Call Logs from the Phone Menu or by pressing Call Logs DSS Key.
• Edit the number (following instructions given above), if required.
• Press 'v' (Down Navigation key).
• You will get the prompt: "Enter Name"224.
• Enter the name of the contact.
• You will get the confirmation message: "Stored at Memory: <XXX>".
224. Only if there is a free Memory Index in the Personal Directory.

• When you store the number in the Personal Directory, the system will automatically assign Trunk
Access Code "0".
• If all 25 Location Index Numbers of the Personal Directory are already programmed, the message
"Memory Full" will appear on your phone's display and you will get an Error Tone. Refer the topic
“Abbreviated Dialing” to know more.

Call Pick Up
What's this?
Call Pick-Up allows extension users to answer calls ringing on other extensions from their own extension; without
physically going to the ringing extensions.
Extension users can 'pick-up' both internal and trunk calls ringing on other extensions.
As extension users can answer calls of their colleagues or co-workers without physically going to their extensions,
this feature ensures that all incoming calls are answered.
SARVAM UCS offers two types of Call Pick-Up:
• Call Pick Up-Group - extensions are assigned to Pick-Up Groups. Any extension in a Pick-Up Group can
answer calls ringing on other extensions within the same group only.
• Call Pick-Up Selective - calls ringing on any extension of the system can be answered.
On SIP extensions, SARVAM UCS supports Call Pickup-Selective and Call Pickup-Group using
Temporary Subscription. For a list of IP phones on which this feature has been tested, see “SARVAM UCS
Features tested on IP Phones of different Brands” in the Appendix.
SARVAM UCS will send only first 3 ringing call's information in NOTIFY message to the SIP Extension,
which has requested Group Call Pickup. This feature has been supported in SIP Phones of CISCO and
POLYCOM.
SIP Extension which has subscribed for BLF of DKP / SIP Extension of SARVAM UCS, the SARVAM UCS
will send information for the call present in the first call loop only.
How it works
Call Pick-Up Group
• Extensions must be assigned to Call Pick-Up Groups. The extensions in a Call Pick-Up group may be SLT,
DKP and ISDN Terminal.
• As many as 99 such groups may be formed.
• Each group is assigned a number 01 to 99
• For example, extensions 2007, 2008, 2009, 2010. 2011, 2012, 2013 are assigned to Pick-Up Group
number 03.
• When an extension in this group rings, any extension in the group can pick up the call by dialing the
feature access code for “Call Pick-Up Group” (default: 4).
• The ringing extension should be in the same Pick-Up Group.

Call Pick-Up Selective
• Extensions need not be in Call Pick-Up Groups.
• Whenever an extension in the system rings, the call can be picked up by any extension of the system by
dialing the feature access code and the number of ringing extension.
When more than one extension in a Pick-Up Group is ringing, you can choose which one to answer first,
using Call Pick-Up Selective.
• Call States: Call Pick-Up will fail if the ringing extension goes into idle state just when you are dialing the
pick-up access code.
• Auto Call Back: Call Pick-Up will fail if the call ringing on the extension is an Auto Call Back request.
• Alarms: Call Pick-Up will fail if the call ringing on the extension is an Alarm Call.
How to configure
For this feature to function, Call Pick-Up should be enabled in the Class of Service of extension that are to be
Call Pick-Up Groups
On a sheet of paper, list the extensions that are to be grouped into a Call Pick-Up Group. Make as many Call Pick-
Up Groups as required. Assign each group a number.
Call Pick-Up
SLT Extensions DKP Extensions ISDN Terminals
Group Number
01 2002, 2003, 2006, 2014 3003, 3004, 3005, 3201, 3201, 3203
3019, 3020, 3021, 3022, 3023,

02
3024
2007, 2008, 2009, 2010.

03 3205, 3206
2011, 2012, 2013
99
The numbering of Call Pick-Up Groups must start from 01 and end at 99.
Do not assign '00' as Call Pick-Up Group. '00' is the command to de-assign from a Call Pick-Up Group.
To program these groups, you may use Jeeves or issuing SE commands from a telephone.

Assigning Extensions to Call Pick-Up Groups using Jeeves
• In the column Call Pick-up Group, assign the group number for SLT extensions. Refer to the sheet of
paper you prepared.

• Assign Call Pick-Up Group number to DKP extensions. Refer to the sheet of paper you prepared.
• Click the Advance button on the bottom of the page.
• Assign Call Pick-Up Group numbers to ISDN Terminals. Refer to the sheet of paper you prepared.
Assigning Extensions to Call Pick-Up Groups using a Telephone

• Enter SE mode from DKP/SLT.
To assign SLT extension to a group, dial:

• 3901-1-SLT-Call Pick-Up Group to assign a single SLT to a group.
• 3901-2-SLT-SLT-Call Pick-Up Group to assign a range of SLTs to the same group.
• 3901-*-Call Pick-Up Group to assign all SLTs to the same group.
Where,
SLT is the Software Port of the SLT, from 001 to 240.
Call Pick-Up Group is from 00 to 99.
By default, Call Pickup Group for SLTs is 99.
To remove SLT extension from a group, dial:

• 3901-1-SLT-00 to remove a single SLT extension.
• 3901-2-SLT-SLT-00 to remove a range of SLT extensions.
• 3901-* to remove all SLT extensions from a group.
To assign DKP extension to a group, dial:

• 3902-1-DKP-Call Pick-Up Group to assign a single DKP to a group.
• 3902-2-DKP-DKP-Call Pick-Up Group to assign a range of DKPs to the same group.
• 3902-*-Call Pick-Up Group to assign all DKPs to the same group.
Where,

DKP is the Software Port of the DKP, from 001 to 96.
Call Pick-Up Group is from 00 to 99.
By default, Call Pickup Group for DKPs is 99.
To remove DKP extension from a group, dial:

• 3902-1-DKP-00 to remove a single DKP extension.
• 3902-2-DKP-DKP-00 to remove a range of DKP extensions.
• 3902-* to remove all DKP extensions from a group.
To assign an ISDN Terminal to a group, dial:

• 3903-1-ISDN Terminal-Call Pick-Up Group to assign a single ISDN Terminal to a group.
• 3903-2-ISDN Terminal-ISDN Terminal-Call Pick-Up Group to assign a range of ISDN Terminals to
the same group.
• 3903-*-Call Pick-Up Group to assign all ISDN Terminals to the same group.
Where,
ISDN is the Software Port of the ISDN Terminal, from 01 to 64.
Call Pickup Group is from 00 to 99.
By default, Call Pickup Group for ISDN Terminals is 99.
To remove an ISDN Terminal from a group, dial:

• 3903-1-ISDN Terminal-00 to remove a single terminal.
• 3903-2-ISDN Terminal-ISDN Terminal-00 to remove a range of terminals.
• 3903-* to remove all terminals.
• Exit SE mode.
Call Pick-Up in Class of Service

Decide which extensions are to be assigned to Call Pick-Up groups. Ensure that the feature Call Pick-Up is enabled
in the Class of Service (COS) of these extensions.
SARVAM UCS. The Station Basic Feature Template 01 is assigned COS group 01 which has Call Pick-Up feature
enabled. Thus, all the extensions of the system can use Call Pick-Up by default.
If you want to deny Call Pick-up feature to all extensions, you can simply disable Call Pick-Up in the default CoS
group 01.
However, if Call Pick-Up is to be denied on only selected extensions then:
1. Define a CoS group with Call Pick-Up disabled.
3. Assign this new Template to the extensions to which Call Pick-Up is to be denied.
programming.

How to use

To pick up a ringing extension in your Group:
• Press DSS Key assigned to Call Pick-Up Group.
OR
• Dial 4.
• Talk.
• Go idle.
To pick up any one of several ringing extensions ringing or the extension that is not in your group:
• Press DSS Key assigned to Call Pick-Up Selective.
OR
• Dial 12.
• Dial number of the Extension you want to pick up.
• Talk.
• Go idle.
For SLT Users

To pick up a ringing extension in your Group:

• Dial 4.
• Talk.
To pick up any one of several ringing extensions ringing or the extension that is not in your group:

• Dial 12.
• Dial number of the Extension you want to pick up.
• Talk.

Call Progress Tones
What's this?
Call Progress Tones (CPT) are audible tones sent from switching systems such as PSTN or PBX to calling parties
to show the status of phone calls, like dial tone, error tone, ringing error in number dialed, ringing called party, busy
line, etc.
Each CPT has a distinctive tone frequency and cadence assigned to it, for which some standards have been
established by the International Telecommunication Union (ITU).
On the basis of specific frequency, modulating frequency and cadence, the CPTs generated by SARVAM UCS are
categorized as:
CPT Event Sound Duration Timer
Dial Tone 1 Played on lifting the Toooooooooooo Played for 7 seconds. Dial Tone Timer
handset. After which Error Tone
starts
Dial Tone 2 Played on lifting the Toooooooooooo Played for 7 seconds. Dial Tone Timer
handset, when 'Store After which Error Tone
and Forward Dialinga' is starts
done.
Ring Back Tone Played when the Turroo... Turrroo Played for 45 seconds Ring Back Tone
internal number you Timer
have dialed is free.
Busy Tone High pitch beeps with Tooooooo......... Played for 7 seconds. Busy Tone Timer
(Engaged Tone) equal ON and OFF Toooooooo
periods, played when
the dialed extension is
busy. Busy tone
continues for 7
seconds. This Busy
Tone Timer is
programmable.
Error Tone Fast beeps, played on a Too…Too…Too Played for 30 seconds Error Tone Timer
(Congestion/ wrong operation being …Too
Refusal Tone as performed or a feature
per ITU invoked without access.
Internal Call Short beep followed by Beep.……….… Played for duration of Interrupt Request
Waiting Tone longer OFF duration Beep the Interrupt Request Timer, Barge-In
(Intrusion Tone repeated every second; Timer or the Barge-In Timer
as per ITU) played to the busy Timer.
extension when another
extension attempts
Interrupt Request/
Barge-In

CPT Event Sound Duration Timer
External Call Two ticks followed by a Beep...Beep… Played for the duration Transfer-On Busy
Waiting Tone longer OFF time of ……......Beep... of the Transfer-On Busy Timer.
(Call Waiting approx. 3 seconds; Beep Timer.
Tone as per ITU) played to a busy
extension when there is
a new incoming Trunk
call.
Confirmation Continuous, fast beeps, Beep... Beep... Played for 7 seconds. Confirmation Tone
Tone played to confirm Beep Timer
(Acceptance successful use of
Tone as per ITU) features.
Feature Tone Short beep followed by Beep................. Played until user goes
a longer off duration Beep ON-Hook or dials a
repeated every second; feature code.
played when dialing
feature access codes
Programming Short beep followed by Beep................. Played until user goes

Tone a longer off duration Beep ON-Hook or dials a
repeated every second; command.
played to prompt
entering of fresh
commands during
programming.
Programming Continuous, fast beeps; Beep... Beep... Played for 3 seconds. Programming
Confirmation played to indicate that Beep Confirmation Timer
Tone system has received a
valid command and is
processing it.
Programming Fast beeps, played on a Too…Too…Too Played for 3 seconds. Programming Error
Error Tone wrong programming …Too Tone Timer
command being dialed.
a. In Store and Forward dialing, the digits are first stored in a memory location and then these are dialed on the trunk.
For example: When Least Cost Routing (LCR) is enabled, the system will store the dialed digits first, check the trunk
through which the call is to be routed and then dials the number on the appropriate trunk.
Tone standards vary with the country of application. For example, as per ITU standard, the Dial Tone for India
consists of 400Hz modulated by 25Hz, whereas it is 350+440Hz, without modulation, for USA/Canada. Further,
many countries use different frequencies and cadences for the same tone. For example, in the US, five different
frequency and cadence are used for Dial Tone.
SARVAM UCS offers the flexibility of setting the Call Progress Tone Generation (CPTG) type to match the country-
specific CPT standards established by ITU.
India being the default 'Region' for SARVAM UCS, the CTPG for India is set as default in the system.

How it works
At the time of installation, when you select the Region (according to the geographical location of the site where the
SARVAM UCS is installed), SARVAM UCS sets the country-specific CPTG type defined for the selected Region.
To see default CPTG types applicable for each region, see “CPTG Region Codes”.
For countries that use different frequencies and cadences for the same tone, e.g. USA, only one
frequency/cadence among the group is considered. See “Default CPTG Type” at the end of this topic.
How to configure
Programming of Call Progress Tones involves configuration of three parameters: CPTG Type (Region), CPT
related Timers, and Dial Tone Type.
The country-specific CPTG type is set automatically by the system when the 'Region' is selected. However, if
required, the System Engineer can change the CTPG type set by the system.
Programming CPTG Parameters using Jeeves

• Click Call Progress Tones to open the page.
• To set the Call Progress Tone Type, select the desired Region from the list.
• Select the desired Dial Tone Type from Type 1 or Type 2.
To change CPT-related Timers

• Go to Call Progress Tones.
• Change the values of the CPT-related Timers as desired.
Programming CPTG Parameters using a Telephone

To select the Region, dial:

• 3501-Region Code
For Region Code, see “CPTG Region Codes”.
To select the Dial Tone Type, dial:

• 5307-Flag
Where,
Flag is
1 for Dial Tone 1
2 for Dial Tone 2
To change CPT-related Timers, dial:

• 3502-Seconds to program the Dial Tone Timer.
Where,

Default: 007 seconds
• 3503-Seconds to program the Ring Back Tone Timer.

Where,
• 3504-Seconds to program the Busy Tone Timer.

Where,
• 3505-Seconds to program the Error Tone Timer.

Where,
• 3506-Seconds to program the Feature Confirmation Tone Timer.

Where,
• 3509-Seconds to program the Programming Confirmation Tone Timer.

Where,
• 3508-Seconds to program the Programming Error Tone Timer.

Where,
• 3542-Seconds to change Tone Demo Timer225.

Where,
• Exit SE mode.
For SE commands to change Interrupt Request Timer and the Barge-In Timer, Transfer-On Busy Timer,
refer the relevant topics: “Interrupt Request (IR)”, “Barge-In” and “Call Transfer”.
How to use
It is important that users of SARVAM UCS also get acquainted with the different Call Progress Tones played by the
system, so that they understand the meaning of the terms used for various tones. Therefore, SARVAM UCS makes
it possible for users to listen to the various Call Progress Tones.
225. Time for which the system demonstrates the tone/ring to the user.

Demonstration of Tones
It is possible to demonstrate Call Progress Tones to users by dialing the SE commands from EON or an SLT.
By default, the system will play each tone as demonstration for 30 seconds. The duration of demonstration can be
changed by setting the 'Tone Demo Timer' to match user preference (see "Changing CPT-related Timers using a
Telephone" above).
To demonstrate call progress tones:

• Dial command 3541-Code.
Where,
Code is
01 Dial Tone 1
02 Dial Tone 2
03 Ring Back Tone
04 Busy Tone
05 Error Tone
06 Confirmation Tone
07 Feature Tone
08 Routing Tone
09 Programming Tone
10 Intrusion Tone (ICWT)
11 External Call Waiting Tone (CCWT)
• Exit SE mode.
CPTG Region Codes
CPTG Dial tone 1 Dial Tone 2 Ring Back Tone Busy Tone
Region Region Cadence Cadence Cadence Cadence
Code Freq. Freq. Freq. Freq.
(sec) (sec) (sec) (sec)
1 Region1 440 Continuous 350+440 Continuous 350+440 0.4on 0.2off 440 0.75on 0.75off
0.4on 2.0off
2 Region2 400 Continuous 400 Continuous 400 0.6on 0.2off 400 0.5on 0.5off
0.2on 2.0off
3 Region3 350+440 Continuous 350+440 Continuous 440+480 2.0on 4.0off 480+620 0.5on 0.5off
4 Argentina 425 Continuous 425 Continuous 425 1.0on 4.0 off 425 0.3on 0.2off
5 Australia 425*25 Continuous 425*25 Continuous 400*25 .4on .2off 425 0.375on
.4on 2.0off 0.375off
6 Brazil 425 Continuous 425 Continuous 425 1.0on 4.0 off 425 0.25on 0.25off
7 Canada 350+440 Continuous 350+440 Continuous 440+480 2.0on 4.0off 480+620 0.5on 0.5off
8 China 450 Continuous 450 Continuous 450 1.0on 4.0off 450 0.35 on
0.36off
9 Egypt 425*50 Continuous 425*50 Continuous 425*50 2.0on 1.0off 425*50 1.0on 4.0off
10 France 440 Continuous 440 Continuous 440 1.5on 3.5off 440 0.5on 0.5off
11 Germany 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.48on 0.48off
12 Greece 425 0.2on 0.3off 425 0.2on 0.3off 425 1.0on 4.0off 425 0.3on 0.3off
0.7on 0.8off 0.7on 0.8off
13 India1 400*25 Continuous 400*25 Continuous 400*25 .4on .2off 400 0.75on 0.75off
.4on 2.0off
14 Indonesia 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.5on 0.5off

CPTG Dial tone 1 Dial Tone 2 Ring Back Tone Busy Tone
Region Region Cadence Cadence Cadence Cadence
Code Freq. Freq. Freq. Freq.
(sec) (sec) (sec) (sec)
15 Iran 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.5on 0.5off
16 Iraq 400 0.4on 0.2off 400 0.4on 0.2off 400 Continuous 400 1.0on 1.0off
0.4on 1.5off 0.4on 1.5off
17 Israel 400 Continuous 400 Continuous 400 1.0on 3.0off 400 0.5on 0.5off
18 Italy1 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.5on 0.5off
19 Japan 400 Continuous 400 Continuous 400*25 1.0on 2.0off 400 .5on .5off
20 Kenya 425 Continuous 425 Continuous 425 0.67on 425 0.2on 0.6off
3.0off 1.5on 0.2on 0.6off
5.0off
21 Korea 350+440 Continuous 350+440 Continuous 440+480 1.0on 2.0off 480+620 0.5on 0.5off
22 Malaysia 425 Continuous 425 Continuous 425 0.4on 0.2off 425 0.5on 0.5off
0.4on 2.0off
23 Mexico 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.25on 0.25off
24 New 400 Continuous 400 Continuous 400+450 0.4on 0.2off 400 0.5on 0.5off
Zealand 0.4on 2.0off
25 Phillippines 425 Continuous 425 Continuous 425+480 1.0on 4.0off 480+620 0.5on 0.5off
26 Poland 425 Continuous 425 Continuous 425 1.0on 4.0off 425 0.5on 0.5off
27 Portugal 425 Continuous 425 Continuous 425 1.0on 5.0off 425 0.5on 0.5off
28 Russia 425 Continuous 425 Continuous 425 0.8on 3.2off 425 0.4on 0.4off
29 Saudi 425 Continuous 425 Continuous 425 1.2on 4.6off 425 0.5on 0.5off
Arabia
30 Singapore 425 Continuous 425 Continuous 425*24 0.4on 0.2off 425 .75on.75off
0.4on 2.0off
31 South 400*33 Continuous 400*33 Continuous 400*33 0.4on 0.2off 400 .5on.5off
Africa 0.4on 2.0off
32 Spain 425 Continuous 425 Continuous 425 1.5on 3.0off 425 0.2on 0.2off
33 Thailand 400*50 Continuous 400*50 Continuous 400 1.0on 4.0off 400 0.5on 0.5off
34 Turkey 450 Continuous 450 Continuous 450 2.0on 4.0off 450 0.5on 0.5off
35 UAE 350+440 Continuous 350+440 Continuous 400+450 0.4on 0.2off 400 0.375on
0.4on 2.0off 0.375off
36 UK 350+440 Continuous 350+440 Continuous 400+450 0.4on 0.2off 400 0.375on
0.4on 2.0off 0.375off
37 USA 350+440 Continuous 350+440 Continuous 440+480 2.0on 4.0off 480+620 0.5on 0.5off
38 Italy2 400 Continuous 400 Continuous 400 1.0on 2.0off 400 0.5on 0.5off
39 Belgium 425 Continuous 425 1.0on 425 1.0on 3.0off 425 0.5on 0.5off
0.25off
40 India2 350+440 Continuous 350+440 Continuous 350+440 0.4on 0.2off 400 0.75on 0.75off
0.4on 2.0off
CPTG Error Tone Confirmation Tone Feature Tone CCWT ICWT

Region Region Cadence Cadence Cadence Cadence Cadence
Code Freq. Freq. Freq. Freq. Freq.
(sec) (sec) (sec) (sec) (sec)
1 Region1 440 0.25on 350+440 0.1on 350+ 0.1on 350+440 0.1on 440 0.1on
0.25 off 0.1off 440 0.9off 0.1off 2.9off
0.1on
2.7off
2 Region2 400 0.25on 400 0.1on 400 1.5on 400 0.2on 400 0.2on
0.25 off 0.1off 0.1off 4.8off 4.8off

3 Region3 440 0.25on 350+440 0.1on 350+ 0.1on 440+480 0.1on 440 0.1on
0.25 off 0.1off 440 0.9off 0.1off 2.9off
0.1on
2.7off
4 Argentina 425 0.3on 425 0.1on 425 0.1on 425 0.3on 425 0.1on
0.4off 0.1off 0.9off 10.0off 2.9off
5 Australia 425 0.375on 425*25 0.1on 425* 0.1on 425 0.2on 425 Continuous
0.375off 0.1off 25 0.9off 0.2off
0.2on
4.4off
6 Brazil 425 0.25on 425 0.1on 425 0.1on 425 0.05on 425 0.1on
7 Canada 480+ 0.25on 350+440 0.1on 350+ 0.1on 440 0.3on 480+ 0.5on
620 0.25off 0.1off 440 0.9off 10.0off 620 0.5off
8 China 450 0.7on 450 0.1on 450 0.1on 450 0.4 on 450 0.2on
0.2on
0.6off
9 Egypt 450 0.5on 425*50 0.1on 425* 0.1on 425*50 0.1on 450 0.5on
0.5off 0.1off 50 0.9off 0.1off 0.5off
0.1on
2.7off
10 France 440 0.25on 440 0.1on 440 0.1on 440 0.3on 440 0.1on
11 Germany 425 0.24on 425 0.1on 425 0.1on 425 0.2on .2off 425 0.1on
0.24off 0.1off 0.9off .2on 5.0off 2.9off
12 Greece 425 0.15on 425 0.1on 425 0.1on 425 0.3on 425 0.15on
0.3on 0.15on
10.0off 1.45off
13 India1 400 0.25on 400 1.0on 400* 0.1on 400 0.2on 400 0.15on
0.2on
7.5off
14 Indonesia 425 0.25on 425 0.1on 425 0.1on 425 0.15on 425 0.1on
0.15on
10.0off
15 Iran 425 0.25on 425 0.1on 425 0.1on 425 0.2on 425 0.1on
0.2on
10.0off
16 Iraq 400 0.25on 400 0.1on 400 0.1on 400 0.1on 400 0.1on
0.1on
2.7off
17 Israel 400 0.25on 400 0.17on 400 0.1on 400 0.5on 400 0.1on
0.34on
5.0off
18 Italy1 425 0.2on 425 0.1on 425 0.1on 425 0.4on 425 0.1on
0.25on
0.1off
0.15on
5.0off

19 Japan 400 0.25on 400 0.1on 400 0.1on 400*25 0.5on 400* 0.1on
0.25off 0.1off 0.9off 2.0off 25 2.9off
0.05on
0.45off
0.05on
3.45off
20 Kenya 425 0.2on 425 0.1on 425 0.1on 425 0.1on 425 0.1on
0.1on
2.7off
21 Korea 480+ 0.3on 350+440 0.1on 350+ 0.1on 350+440 0.25on 350+ 0.1on
0.25on
3.25off
22 Malaysia 425 2.5on 425 0.1on 425 0.1on 425 0.2on 425 0.1on
0.2on
5.0off
23 Mexico 425 0.25on 425 0.1on 425 0.1on 425 0.1on 425 0.1on
0.1on
2.7off
24 New 400 0.25on 400 0.1on 400 0.1on 400 0.2on 425 0.1on
Zealand 0.25off 0.1off 0.9off 3.0off 2.9off
0.2on
5.0off
25 Phillippines 480+ 0.25on 425 0.1on 425 0.1on 440 0.3on 440 0.1on
620 0.25off 0.1off 0.9off 10.0off 2.9off
26 Poland 425 0.5on 425 0.1on 425 0.1on 425 0.15on 425 0.1on
0.15on
4.0off
27 Portugal 450 0.33on 425 1.0on 425 0.1on 425 0.2on 425 0.2on
0.2on
5.0off
28 Russia 425 0.25on 425 0.1on 425 0.1on 950 0.333on 425 0.1on
29 Saudi 425 0.25on 425 0.1on 425 0.1on 425 0.15on 425 0.1on
Arabia 0.25off 0.1off 0.9off 0.2off 2.9off
0.15on
10.0off
30 Singapore 425 0.25on 425 0.125on 425 0.1on 425 0.3on 425 0.25on
0.3on
3.2off
31 South 400 0.25on 400*33 0.1on 400* 0.1on 400*33 0.4on 400 0.15on
Africa 0.25off 0.1off 33 0.9off 4.0off 0.25off
0.15on
1.45off
32 Spain 425 0.25on 425 0.1on 425 0.1on 425 0.175on 425 0.1on
0.175on
3.5off

33 Thailand 400 0.3on 400*50 0.1on 400* 0.1on 400 0.1on 400 0.1on
0.1on
2.7off
34 Turkey 450 0.2on 450 0.04on 450 0.1on 450 .2on .6off 450 0.1on
0.2off 0.04off 0.9off .2on 8.0off 2.9off
.6on .2off
35 UAE 400 0.4on 350+440 0.1on 350+ 0.1on 350+440 0.1on 350+ 0.1on
0.35off 0.1off 440 0.9off 0.1off 440 2.9off
0.225on 0.1on
0.525off 2.7off
36 UK 400 0.4on 350+440 0.1on 350+ 0.1on 350+440 0.1on 400 0.2on
0.225on 0.1on
0.525off 2.7off
37 USA 480+ 0.25on 350+440 0.1on 350+ 0.1on 440 0.3on 480+ 0.5on
38 Italy2 400 0.25on 400 0.1on 400 1.75on 400 0.2on 400 0.2on
0.2on
2.5off
39 Belgium 425 0.167on 425 0.1on 425 0.1on 1400 0.175on 440 0.1on
0.175on
3.5off
40 India2 400 0.25on 350+440 0.1on 350+ 0.1on 350+440 0.1on 350+ 0.5on
0.25 off 0.1off 440 0.9off 0.1off 440 0.5off
0.1on 1.0on
2.7off 5.0off
Stuttered Dial Tone

Frequency: Same as Dial Tone 1 (Region wise)
Cadence: 400 ms On - 100 ms Off, 400 ms On - 100 ms Off, 400 ms On - 100 ms Off, 400 ms On - 100 ms Off, 400
ms On - 100 ms Off, 400 ms On - 100 ms Off (same for all Regions)
The meaning of frequency notation is as follows:

• f1*f2: f1 is modulated by f2.
• f1+f2: The juxtaposition of two frequencies f1 and f2 without modulation.

Default CPTG Type

Call Restriction based on IP Address
What's this?
When the CPU Card of SARVAM UCS is connected to a public IP network, it is may be necessary to allow traffic
from particular IP address only.
With the feature 'Call Restriction based on IP Address', SARVAM UCS makes it possible to entertain requests on
its LAN/WAN Ports from predefined IP Addresses only.
How it works
• the Trusted IP Address/es table must be configured for each SIP Trunk.
• with this table configured, incoming call traffic from all IP Addresses, other than those programmed in the
Trusted IP Address/es table, will be blocked.
All incoming traffic on the SIP Trunk will be rejected if the Trusted IP Address/es Table of that SIP Trunk is
blank.
How to configure
For each SIP Trunk, make a list of IP Addresses:Port from which you want to allow traffic. If you want to allow
incoming calls from all ports for a particular IP Address, configure only the IP Address.
You are allowed to configure a maximum of 10 IP Addresses.
Programming IP Address Based Call Traffic Restriction using Jeeves

• Login to Jeeves as System Engineer.

• Click SIP Trunk Parameters.
• Scroll to the Trusted IP Address/es parameter.
• Click IP Address Table.

The first entry in the table will display the Proxy/Registrar Server Address:Port or Outbound Proxy Address: Port as
configured for the “Configuring SIP Trunks”. For the Index numbers 1 to 10,
• Enter the IP Address and the corresponding Port from which you want to allow incoming calls.
Do not configure the port, if you want to allow incoming calls from all the ports for a particular IP Address.

Call Taping
What's this?
Call Taping allows extension users to record the telephone conversations they have with other extensions or
external numbers, without the opposite party coming to know about it.
Feature is useful for keeping records of important conversations. For this feature to work, make sure the VMS
Module is installed.
Call Taping can be done for:

• Incoming and outgoing external calls.
• Incoming and outgoing internal calls.
SARVAM UCS supports tapping of 20 calls simultaneously in a 3-Party Conference. Calls can be taped either in
the extension user’s personal mailbox or a common mailbox assigned to this feature. The taped calls are stored
along with the call details, that is, the time and date of the call, the calling number and the called number. If calls are
taped in a common mailbox, only the extension users having access to the common mailbox can retrieve and listen
to the recorded conversations.
To be able to record external incoming and outgoing calls,
• a list of phone numbers (both incoming and outgoing) must have been programmed in Number List-
Incoming Calls and Number List- Outgoing Calls respectively, in the Station Advanced Feature
Template.
• Call Taping must be enabled in the Trunk Feature Template assigned to the trunk used for making/
receiving external calls.
Incoming calls without Calling Line Identification (CLI) can also be taped. For this, the check box Tape calls
coming without CLI must be enabled in the Station Advanced Feature Template assigned to the extensions.
To be able to record internal calls, the Call Taping for Internal Calls check box must be enabled must be enabled
in the Station Advanced Feature Template assigned to the extension.
• Use this feature in accordance with the local privacy laws.
• Matrix Comsec is not responsible for any mis-/abuse of this feature by users.
How it works
A and B are extensions. Both have Call Taping parameters configured in the Station Advanced Feature Template
assigned to them. Also, the trunks assigned to both the extensions for making outgoing calls have Call Taping
enabled.
C and D are external parties.
E is the mailbox extension where the tapped calls are recorded.

A calls C
• The system matches the dialed number with the numbers in the Number List - Outgoing Calls. The
system finds a match.
• When speech is established, the system starts recording the conversation between A and C automatically
in E's mailbox.
• Call Taping Beeps will be played to A and C, if Play Beep when Raid/Call Taping/Conversation
Recording starts is enabled in the System Parameters.
D calls B
• The system matches the incoming number with the numbers in the Number List-Incoming Calls.
• On finding a match, when speech is established, system records the speech between D and B in E's
mailbox.
• Call Taping Beeps will be played to D and B only if Play Beep when Raid/Call Taping/Conversation
• If an incoming call does not have any CLI, the system checks the check box Tape calls coming
without CLI under the Call Taping in Station Advanced Feature Template assigned to the extension.
• If the check box is enabled, all calls without CLI are taped.
A calls B
• The system checks if the Call Taping for Internal Calls is enabled under the Call Taping in Station
Advanced Feature Template assigned to A.
• If the check box is enabled, the system records the speech between A and B in E's mailbox.
• Call Taping Beeps will be played to A and B only if Play Beep when Raid/Call Taping/Conversation
• If the check box is disabled, the speech between A and B will not be recorded.
• The same is done when B calls A. The speech will be recorded in E's mailbox.
To listen to the conversation, A and B must have access to the mailbox of E.
• Conversation Recording: If Call Taping and “Conversation Recording” both are enabled for an
extension, then priority is given to Call Taping.
How to configure
The functioning of this feature requires the following parameters to be programmed:
• Save Call Taping Files in: Specify the location to save Call Taping files. You can select Common
Mailbox or Personal Mailbox.
• Common Mailbox for Call Taping (Enter Extension Number): You must program the extension number
of the user in whose mailbox the calls are to be taped.

• Call Taping on Trunks: Call Taping must be enabled in the Trunk Feature Template assigned to the
Trunks. Make sure these trunks are used by extension users for making/receiving external calls.
• Tape calls coming without CLI check box: This check box must be enabled if you want calls without CLI
to be taped.
• Number Lists for Incoming and Outgoing Calls: The Call Taping Number List-Incoming Calls and Call
Taping Number List-Outgoing Calls are to be programmed in the Station Advanced Feature Template
assigned to the extension users so that the system can match the phone numbers of the incoming and
outgoing calls and initiate the recording of the speech.
On a sheet of paper, prepare the Call Taping List Incoming and Call Taping List Outgoing.
You can add as many as 999 numbers to each list. Each entry on these Lists is stored in a serial order
against a 'Location Number'. So, draw three columns and enter the numbers against a location number
from 01 to 999.
Location Number List-Incoming Calls Number List-Outgoing Calls
001
002
999
Use this table to program the Number lists. By default Number List 09 is assigned for numbers of incoming
calls, and Number List 10 is assigned to numbers of outgoing calls.
• Call Taping for Internal Calls check box: This check box is to be enabled in the Station Advanced
Feature Template applied on those extensions that are to be allowed Call Taping of internal calls, that is,
calls made or received by them to or from other extensions.
• Play Beep when Raid/Call Taping/Conversation Recording starts: This check box is to be enabled if
Call Taping Beeps are to be played to the two parties in speech. Enable Call Taping Beeps only when you
want indication of speech recording to the two parties in speech. By default, this check box is enabled.
Configuring Call Taping Parameters using Jeeves

• Under Configuration.
For Call Taping Mailbox,
• Click VMS General Parameters to open the page.

• Under General Settings,
• Specify the location where you want to Save Call Taping Files in. You can save the Call Taping files
either in the Common Mailbox or in your Personal Mailbox. By default, Call Taping files are saved in the
Common Mailbox.
• If you choose to save Call Taping files in the Common Mailbox, under Common Mailbox for Call Taping
(Enter Extension Number), enter the Access Code of any SLT, DKP, SIP Extension, Department Group
or General Mailbox, whose mailbox you want to assign for Call Taping.
To enable/disable Call Taping Beeps,
• On the same System Parameters page, go to Play Beep when Call Taping/Conversation Recording
Starts and enable/disable beeps by selecting/clearing the check box.

To enable Call Taping on Trunks,
• Select the Call Taping check box in the Trunk Feature Template assigned to the Trunks. For detailed
instructions, see “Configuring Trunks”.
To configure the Call Taping parameters and Number Lists,
• Click Station Advanced Feature Template to open the page.
By default, Station Advanced Feature Template 01 is assigned to all extensions of the SARVAM UCS. If
you want to assign Call Taping facility to all extensions, then program the Call Taping related parameters
and Number Lists, in Template 01.

However, if only selected extensions are to be assigned this feature, then:
1. Prepare a separate Station Advanced Feature Template.
2. Set the Call Taping Parameters in this template.
3. Apply this new template to desired extensions that are to be allowed this feature.
• Scroll with the horizontal bar to reach the Call Taping column of the Template Number assigned to the
extensions.
• If you want calls without CLI to be taped, select the Tape calls coming without CLI check box.
• To program the list of numbers of incoming calls, click the link Number List- Incoming Calls.
The Number List page will open.
• Click the default number list 9-10 link assigned to Call Taping, then click the 001-255 link of the default.
The Number Lists 9 and 10 will open.
• If the same incoming and outgoing numbers are to be programmed for all extensions, you may simply
program the default Number lists 09 and 10.
• If different incoming and outgoing numbers are to be programmed for different extensions, then
prepare different number lists.
• Enter the List of Incoming Numbers that the system should match in List No. 09.
• Enter the List of Outgoing Numbers that the system should match in List No. 10.

You can program as many as 999 numbers in each list. Each entry on these Lists is stored in a serial order
against a 'Location Index, starting from 001-999'. There are 250 Location Index on each page on your
screen. To go to the next set of Location Index, for instance, 251-500, click the link under 09-10.
• Click Submit at the bottom of the page to save your number lists.
• Follow the same steps to program a different Call Taping number list. But ensure that the different List
number you programmed is entered in the Station Advanced Feature Template applied to the extensions.
• If you want calls between extensions to be taped, click the Call Taping for Internal Calls check box.
• Click Submit at the bottom of the page to save changes to the template.
• Now, apply the programmed template to the desired extensions to which you want to provide the Call
Taping facility. Refer the topic “Station Advanced Feature Template” for programming instructions.
Programming the Station Software Port using a Telephone

To enable/disable Call Taping Beeps, dial:

• 5332-Code
Where,
Code is
0 for Disable
1 for Enable
To program a number in a List:

• 4302-List Number-Location Index-Number-#*
Where,
List Number is from 01 to 16. In this case, 09 for Incoming Number List and 10 for Outgoing Number
List.
Location Index is from 001 to 999.
Number is a number string of maximum 16 digit long terminated with #*.
The digits used are as given in Table below for digits: 0-9, #, *, A, B, C, D, F, P, +.
Special Digit Code
Flash (F) #2
Pause (P) #3
A #4
B #5
C #6
D #7
+ #8
Dot (.) #9
# ##
* **

To clear a number from a Location Index:
• 4302-List Number-Location Index-#*
Where,
List Number is from 01 to 16. In this case, 09 for Incoming Number List and 10 for Outgoing Number
List.
Location Index is from 001 to 999, where the number you want to clear is stored.
To program Number List - Incoming in Station Advanced Feature Template:

Where,
Feature Number is
18 for Number List-Incoming.
19 for Number List-Outgoing.
Code is the Number List from 01 to 16.
In this case, 09 for Incoming Number List and 10 for Outgoing Number List.
To enable Call Taping Internal Flag in a Station Advanced Feature Template, dial:
Where,
Feature Number for Call Taping Internal Flag is 20.
Code is
0 for Disable
1 for Enable
To enable Tape calls coming without CLI Flag in a Station Advanced Feature Template, dial:
Where,
Feature Number for Tape calls without CLI is 17.
Code is
0 for Disable
1 for Enable
• Exit SE mode.
For SE commands for applying the programmed template to DKP and SLT extensions, refer the topic
“Customizing Station Advanced Feature Template using a Telephone”.
Also refer the topic “Number Lists” to know more.
How to use
This feature works automatically on the extensions which have the Call Taping parameters configured.
Call Taping conversations can be recorded either in a common mailbox or in the extension user’s personal mailbox.
• If calls are taped in a common mailbox, only the extension users having access to the common mailbox
can retrieve and listen to the recorded conversations.
• If calls are taped in a personal mailbox, the extension user can listen to the recorded conversation by
accessing their personal mailbox.

Accessing Personal Mailbox
If you are an EON/Extended IP Phone user
• Press ‘Voice Mail’ Key.

OR
• Dial 3931-Your Mailbox Password226
• Follow Voice Mail Prompts to listen to new messages.
If you are an SLT user

• Dial 3931-Your Mailbox Password
• Follow Voice Mail Prompts to listen to new messages.
226. Only if the mailbox is password protected, you will be prompted to enter the password.

Call Toggle
What's this?
Call Toggle allows you to have two simultaneous telephone conversations, talking to two persons alternately.
Call Toggle is also referred to as Hold-Consult or Call Splitting,
You can toggle between:
• Two internal calls (that is, two extensions).
• An internal Call and an External Call (extension and trunk).
• Two external calls (two trunks).
How it works
• A, B, and C are extensions.
• D and E are trunks.
Toggling between two internal calls
• A is in speech with B and C is on Consultation Hold.
• To talk with C, A dials Flash-1. Speech with C.
• To talk with B, A dials Flash-1. Speech with B.
• A can toggle back to C by dialing Flash-1.
Toggling between internal call and external call
• A is in speech with B and D (external call) is on Consultation Hold.
• To talk with D, A dials Flash-1. Speech with D.
• To talk with B, A dials Flash-1. Speech with B.
• A can toggle back to D by dialing Flash-1.
Toggling between two external calls
• A is in speech with D and E is on Consultation Hold.
• To talk with E, A dials Flash-1. Speech with E.
• To talk with D, A dials Flash-1. Speech with D.

• A can toggle back to E by dialing Flash-1.
• The party put on Consultation Hold during Call Toggle cannot hear the conversation between the other
two parties.
• You can also toggle between an incoming internal/external call (indicated by call waiting tone) and an
internal/external call you are currently in speech with.
• You can also answer an incoming 'Interrupt Request' call and toggle between the interrupting extension
and the extension you were in speech with.
• You can convert a Call Toggle into a three-party conference by dialing Flash-*3.
• You can transfer the call you are currently in speech with to another extension.
• You can park the call you are currently in speech with.
How to configure
Call Toggle is a Class of Service (CoS) dependant feature.
In the default Station Basic Feature Template 01 assigned to all extensions of SARVAM UCS, Call Toggle is
included in the 'Basic Features' assigned to all CoS groups, including the default CoS group 01. So, all extensions
of SARVAM UCS can use this feature.
As Call Toggle is a part of the set of 'Basic Features', you cannot disable this feature selectively in the COS of
extensions, without disabling the entire set of features.
No specific programming is required for this feature, except for programming a DSS key for Call Toggle, if required.
Refer the topic “DSS Keys Programming” for instructions.
How to use

Call Toggle between two internal calls:
• Speech with Extension 1.

• Extension 2 on Consultation Hold.
• To talk with Extension 2, press DSS key assigned to Call Toggle.
• Speech with extension 2.
• Press DSS key assigned to Call Toggle again.
• Speech with Extension 1.
Call Toggle between an Internal Call and an External Call:
• Speech with extension.

• External party on Consultation Hold.
• To talk with the external party, press DSS key assigned to Call Toggle.
• Speech with external party.
• Speech with Extension.

Call Toggle between two External Calls:
• Speech with external party 1.

• External party 2 on Consultation Hold.
• Press DSS Key assigned to Call Toggle.
• Speech with External party 2.
• Speech with External party 1.
For SLT Users

Call Toggle between two internal calls:

• Extension 2 on Consultation Hold.
• To talk with extension 2, dial Flash-1.
• Dial Flash-1again.
Call Toggle between an Internal Call and an External Call:

• External party on Consultation Hold.
• To talk with external party, dial Flash-1.
• Speech with external party.
• Dial Flash-1 again.
Call Toggle between two External Calls:

• External party 2 on trunk 2 on Consultation Hold.
• To talk with external party 2, dial Flash-1.
• Dial Flash-1 again.

Call Traffic
What is this?
Call traffic measurement feature of SARVAM UCS gives a graphical representation of the time duration for which
extensions and trunk ports remained Off-hook. The data is represented in a bar graph format for the SLT, DKP,
Magneto, ISDN Terminals, CO, E&M, T1E1 and Mobile ports as well as SIP Extensions.
In the graph, the duration is shown in Hours along the Y-axis while the extension(s)/ trunk(s) port names are shown
along the X-axis. The following call traffic measurement data (for each Extension and Trunk port separately) is
displayed:
• Time for which each extension remained in speech for making/receiving Internal calls.
• Time for which each extension remained in speech for making/receiving Trunk calls (Incoming +
Outgoing).
• Time for which each trunk port remained in speech for Incoming calls.
• Time for which each trunk port remained in speech for Outgoing calls.
This time is measured in terms of the number of hours, and the traffic is measured for last 24 hours.
You can view this traffic information in graphical format on the Jeeves; see the illustration below for Call Traffic
information generated for SIP extension.

The two-color bars distinguish Internal calls (green) and Trunk calls (red)
You can also export this information in a database readable format like Microsoft Excel. The call traffic information
files can be saved on a local disk.
How to use
To view Call Traffic in graphical format, you need to log into the Jeeves as System Engineer.
Go to the configuration page of the desired trunk or extension port for which you want to view Call Traffic data.
On the page, click the Call Traffic button.
The call traffic will be generated as a graph.
If there is no call traffic usage data is available for the extension/ trunk you are currently viewing, the page
will not show any bars.
Click the Refresh button to get the latest 24hrs call traffic statistics. The application filters the SMDR records for
last 24hrs and graph is generated accordingly.

• Click the Export button to export the data you are currently viewing.
• The Opening SIP_CallTraffic.csv window will open.
You can choose whether to open this file or save it in .xls format as per your requirement.

• If you open this file in MS Excel then it may look as shown in the following image.
The file contains the date and time when the call traffic data is calculated and the names of the extensions/
trunks for which it is calculated.
• You can save this file on your local disk from Excel window also.
• After you are done, you may log out of Jeeves.

Call Transfer
What is this?
Call Transfer enables you to relocate an existing call from an extension or trunk to another extension or to an
external number. Calls can be transferred after notifying the other extension/external number about the impending
transfer or can be transferred directly without notification.
The types of Call Transfer SARVAM UCS offers are:
• Call Transfer - Screened: The Operator puts the caller on Consultation Hold, dials the desired party's
extension, and informs the desired party of the impending transfer. If the desired party chooses to accept
the call, the call is transferred over to them.
• Call Transfer - While Ringing: The Operator puts the caller on Consultation Hold, dials the desired
party's number and transfers the call when the desired party's extension starts ringing.
This feature is used when there are several other calls to be attended and the Operator cannot wait for the
desired party to answer.
• Call Transfer - On Busy: The Operator puts the caller on Consultation Hold, dials the desired party's
number and transfers the call even when the desired party is busy in speech with another person. The
busy extension gets intrusion tone and can choose to answer the intruding (transferred) call.
• Call Transfer - Trunk-to-Trunk: An external call is transferred on to another trunk line. The Operator puts
the external caller on Consultation Hold, dials the desired party's external number, and transfers the call
after or without notifying the desired party of the impending transfer.
Trunk-to-Trunk call transfer may be used to transfer incoming calls for out-of-office extension users to their
cell phones, or to connect personnel at remote or distant locations. For instance: an out-of-office executive
who does not have long distance dialing permission can call the office and request the operator to connect
him to the desired party on a trunk line.
• Blind Transfer to VMS: The Operator puts the caller on Consultation Hold, dials the feature access code
for Blind Transfer to VMS, dials the desired party's number, and transfers the call. The call is transferred to
the mailbox assigned to the desired party. The caller may leave a message in the mailbox.
• Call Transfer is not exclusively an Operator feature, though it is used mostly by Operators. Calls can be
transferred by any extension to another extension or external number, if "Basic Features" are allowed
in Class of Service of the transferring extension.
• SARVAM UCS enables SIP extensions to resume a transferred call before it has been answered by the
transfer target (which may be an extension or an external number). For a list of IP phones on which this
feature has been tested, see “SARVAM UCS Features tested on IP Phones of different Brands”in the
Appendix.
• SARVAM UCS allows Semi-attended Transfer and Transfer on Conference Hangup on SIP Trunks.
For a list of IP phones on which this feature has been tested, see “SARVAM UCS Features tested on IP
Phones of different Brands”in the Appendix.

How it works
A and B are extension users.
C is an external caller.
D is an external number.
Call Transfer from Trunk/Extension to Extension

Three scenarios are possible:
1. Screened Transfer:
• C calls a Trunk of SARVAM UCS.
• Operator puts C on Consultation Hold.
• Operator dials B's extension.
• When B answers the call, the Operator informs B about the call.
• If B accepts the call, the Operator transfers the call to B.
• Now, B and C are in speech.
• If B does not accept the call, Operator may dial Flash to retrieve the call and speak to C.
• The Operator can also abort call transfer while B's phone is ringing by dialing Flash. The Operator gets
connected to C.
2. Transfer While Ringing:

• The Operator puts C on Consultation Hold and dials B's extension.
• The Operator transfers the call when B's phone starts ringing.
• If B answers the call, B gets connected to C.
• If B does not answer the call at the end of the Transfer-While Ringing Timer (programmable; default: 30
seconds), the call is returned to the Operator. C gets Ring Back Tone.
• The Operator answers the call and is in speech with C.
• However, if the Operator is busy at the time of call return, the system waits for the Operator to become
free. When the Operator is free, C gets Ring Back Tone.
• The Operator answers the call and is in speech with C.
3. Transfer On Busy:
• The Operator puts C on Consultation Hold and dials B's extension.
• The Operator gets busy tone from B's extension. B is busy with A
• The Operator transfers the call to B on Busy tone.
• B gets intrusion beeps. The beeps are played for the duration of the Transfer on Busy Timer
(programmable; default: 30 seconds)
• B may dial Flash to answer the call. A is put on hold.
• B is now connected to C.
• If B does not answer the intrusion beeps at the end of the Transfer on Busy Timer, the call is returned
to the Operator. C gets ring back tone.
• If the Operator too is busy at the time of call return, C gets busy tone.

Call Transfer - Trunk to Trunk
• C calls a Trunk of the SARVAM UCS.
• B is out of office, but is available at external number D.
• The Operator puts C on Consultation Hold.
• The Operator dials trunk access code and calls the external number D.
• The Operator may:
Wait for D to answer the call, transfer the call if D accepts the call. (screened transfer)
OR
Transfer the call as soon as D's phone starts ringing. (transfer while ringing)
• C and D are now in speech for the duration of the Trunk-to-Trunk Inactivity Timer227.
• A warning tone is given at the end of the Trunk-to-Trunk Inactivity timer (programmable; default: 2
minutes). On expiry of this timer, the call is disconnected.
• To extend the call, either C or D must dial any digit in tone (DTMF), except '##'.
Dialing ‘##’ to extend the call will result in Call Disconnection.
In the case of Trunk-to-Trunk transfer, both parties in speech on trunk lines must be informed that their call
would be disconnected at the end of the Trunk-to-Trunk Inactivity Timer and that they must dial any digit,
except ‘##” to extend the call.
Call Transfer - Blind Transfer to VMS

• C calls a Trunk of SARVAM UCS. C wants to talk to A.
• The Operator attends the call.
• The Operator puts C on Consultation Hold.
• The Operator dials Feature Access Code for Blind Transfer to VMS and A's extension number.
• The system hunts for the mailbox assigned to A's extension.
• When the mailbox is found, the Operator gets confirmation tone. C gets connected to A's mailbox.
• C gets voice prompts. C may follow the prompts to leave a message.
• If A does not have a mailbox assigned, the Operator will get an error tone while transferring the call.
• The Operator may retrieve C's call by pressing Transfer Key/Flash /Call Appearance key.
• CLIP and Caller ID Presentation while Transfer: SARVAM UCS provides the flexibility to display either
the extension number that is transferring the call or the held party's number, that is, the number of the party
that is about to be transferred. Refer “Calling Line Identification and Presentation (CLIP)”.
• Privacy: Call Transfer-On Busy will not work if the busy extension has Call Privacy from intrusion Tone in
its Class of Service.
• DND: Call Transfer will not work if the destination extension has set DND.
• Call Hold: You must retrieve a held call, to Transfer it.
227. The process of Trunk-to-Trunk transfer takes place outside of the System. So, the System will not know which of the two trunks
have gone ON-Hook. Hence the call is automatically disconnected when the Trunk-to-Trunk Inactivity Timer expires.

How to configure
To be able to use Call Transfer, this feature must be enabled in the Class of Service group of the extensions to be
allowed this feature. The default values of the related Timers may be changed, if required.
To be able to use Blind Transfer to VMS, the extensions must be assigned a mailbox. For instructions, refer to
“Configuring Voice Mail System” .
Call Transfer in Class of Service

In the default Station Basic Feature Template Number 01 is assigned to all the extensions of SARVAM UCS, the
default CoS group 01 in this template has Call Transfer included in the 'Basic Features'. So, all extensions of
SARVAM UCS can use this feature.
You cannot disable 'Call Transfer' selectively without disabling the entire set of 'Basic Features'.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” to know more.
Call Transfer Related Timers

• Transfer While Ringing Timer: This timer is related to Call Transfer - While Ringing. It is the time for
which the system rings the extension. By default it is set to 30 seconds. At the end of the timer the call is
returned to the transferring extension.
• Transfer on Busy Timer: This timer is related to Call-Transfer on Busy. It is the time for which the system
waits for the busy extension to respond to the intrusion tone. By default the timer is set to 30 seconds. At
the end of the timer the call is returned to the transferring extension.
• Trunk to Trunk Inactivity Timer: This is the time duration after which the system disconnects the call
transferred from one trunk line to another. By default it is set to 2 minutes. At the end of the timer the call is
disconnected, if either party does not dial digits to extend the call. This Timer is relevant for CO to CO and
CO to E&M calls only.
Changing Call Transfer Related Timers using Jeeves


• Scroll to reach Other Features and change the values as required for Call Transfer related timers.
• Log out of Jeeves or continue with programming tasks, as required.
Changing Call Transfer Related Timers using a Telephone

• Enter SE mode.
To program Transfer While Ringing Timer:

Where,
Default: 030
To program Transfer on Busy Timer:

Where,
Default: 030
To program the Trunk-to-Trunk Inactivity Timer:

• Dial command 3808-Minutes
Where,
Seconds is from 001 to 255 minutes.
Default: 2 minutes
• Exit SE mode.

How to use

Extension to Extension:

• Press DSS Key assigned to desired party extension.
• Go ON-Hook or press ‘Transfer’ Key.
OR

• Press ‘Transfer’ Key. You get feature tone.
• Dial desired party's extension number.
Extension to Trunk:

• Press DSS Key assigned to Trunk.
• Dial External Number (transfer target)
OR

• Press ‘Transfer’ Key. You get feature tone.
• Dial-Trunk Access Code228- External Number.
Trunk to Extension Transfer:
• Speech with Trunk.

• Press DSS Key assigned to desired party extension.
OR
• Speech with Trunk

• Press ‘Transfer’ Key and dial desired party's extension number
Trunk to Trunk Transfer:

• Press DSS Key assigned to Trunk.
• Dial External Number.
• Speech with External Number.
228.Trunk Access Code: users worldwide may dial a code from 0, 5, 61, 62, 63, and 64. Users in USA may dial a code from 0, 9, 81, 82,
83, and 84.

OR

• Press ‘Transfer’ Key and dial Trunk Access Code229-External Number.
• Speech with the External Number.
Blind Transfer to VMS:

• Press DSS Key assigned to Blind Transfer to VMS.
• Dial desired party’s extension number.
OR

• Press Transfer key.
• Dial 1078.
For SLT Users

Extension to Extension Transfer:

• Press Flash.
Extension to External Number:

• Press Flash.
• Dial External Number.
External Number to Extension Transfer:
• Speech with External Number.

• Press Flash.
Trunk to Trunk Transfer:
• Speech with External Number 1.

• Dial Flash.
• Dial Trunk Access Code
229.Trunk Access Code: users worldwide may dial a code from 0, 5, 61, 62, 63, and 64. Users in USA may dial a code from 0, 9, 81, 82,
83, and 84.

• Dial External Number 2.
• Speech with the External Number 2.
Blind Transfer to VMS
• Speech with Trunk/Extension.

• Press Flash.
• Dial 1078.

Calling Line Identification and Presentation (CLIP)
What's this?
The SARVAM UCS provides the facility of detecting the caller's number and presenting it on the display of the
called extension phone. This feature is called Calling Line Identification and Presentation (CLIP).
The calling number can be presented on ISDN Terminals, EON and also on SLTs that support CLI protocols.
The signaling protocols for CLI supported by SARVAM UCS are: DTMF, FSK V.23, and FSK-Bellcore.
These protocols are supported on trunks as well as extensions. Any type of trunk line and supporting DTMF or FSK
signaling can be interfaced with the SARVAM UCS.
Similarly, any type of telephone instrument supporting DTMF or FSK signaling protocol can be connected to the
SLT port.
How it works
When CLIP is enabled on a trunk,
• SARVAM UCS senses the digits/codes sent by PSTN.
• It sends this information to the landing extension/Operator along with the ringing signal.
• In case of, Internal calls the calling extension's name and number both are presented to the called
extension.
• In the case of External calls, only the number will be displayed on the landing/Operator extension.
• When the landing extension/Operator transfers the incoming call to an extension, putting the external
caller on hold, the system sends this information to the extension to which the call is transferred.
• During the transfer, the number of the landing extension/Operator will be displayed on the transfer
destination extension.
• On successful call transfer, the caller's number will be displayed on the transfer destination extension.
In the case of Call Transfer, the system also provides the option of displaying to the destination extension either the
number of the party that is put on hold to be transferred, that is, the Held Party OR the number of the Transferring
Party, while the call transfer is taking place. This feature is called Caller ID Presentation while Transfer.
It is also possible to remove and replace the '+' character received as CLI on telephones that do not support CLIP
starting with this character.
For example, the GSM network sends the calling party number with '+' as the prefix. If the telephone connected as
extension does not support this, it will not present the CLI of the caller. To overcome this, SARVAM UCS provides
you the option of replacing '+' with an appropriate number string which these telephones can display.

• CLIR: CLIP and Caller ID Presentation while Transfer will work only if CLIR is not enabled on the
extension that has transferred the call. Refer the topic “Calling Line Identification Restriction (CLIR)”.
• Q-Sig: When two Systems - System A and System B are networked using Q-Sig, and an extension of one
System, for instance, System A transfers a call to an extension of System B by putting the caller on hold,
the CLI presented on the extension of System B will be according to the type of Caller ID Presentation
while Transfer set on the transferring extension of System A.
How to configure
The functioning of this feature is controlled by two parameters: CLIP Type and Caller ID Presentation while
Transfer.
If you want to replace '+' characters received as CLI on telephones that do not support CLI prefixed with this
character, you must program the relevant flag and the desired number string in the 'System Parameters'.
All these parameters can be programmed using Jeeves and a Telephone.
CLIP Type
If SLTs supporting CLI are connected to the SARVAM UCS, the System Engineer must select a signaling protocol
for CLI in the SLT Hardware Template applied on the SLT extensions. By default SLT Hardware Template 01 is
assigned to all SLT extensions. The default CLIP Type in Template 01 is 'DTMF'.
There is no need to select a CLIP Type in the default Hardware Template 01, if all the SLTs support DTMF protocol.
If all SLTs support a different CLIP Type say FSK-Bellcore, you may simply select this CLIP Type in the default
Hardware Template applied on all SLT extensions.
However, if certain SLTs support a particular CLIP type and some support a different CLIP type, then create
separate SLT Hardware Templates with different CLIP types and apply them to the appropriate SLTs.
For example, you may select SLT Hardware Template 02 with FSK V.23 and SLT Hardware Template 03 with FSK
Bellcore, and Template 04 with DTMF as the CLIP Type and apply each template to the SLTs as per the CLIP Type
they support.
Selecting CLIP Type using Jeeves


• Click SLT Hardware Templates to open the page.
• Select the CLIP Type supported by the SLT.
Selecting CLIP Type using Telephone

• Dial command 5702-1-Template Number-Parameter Number-Code

Where,
Template Number is SLT Hardware Template Number from 01 to 50
Parameter Number is 01 for CLIP Type.
Code is:
0 for None
1 for DTMF
2 for FSK-V23
3 for FSK Bellcore
For instance: To select FSK-V23 in SLT Hardware Number 01, dial 5702-1-01-01-2
• Exit SE mode.
Caller ID Presentation while Transfer

The Caller ID Presentation while Transfer gives you the choice of displaying to the destination extension, the CLI of
either the party put on hold to be transferred, that is, the Held Party or the Transferring Party, that is, the party
transferring the call, during a call transfer.
This feature is to be programmed in the Station Advanced Feature Template applied on the extension.

In the Station Advanced Feature Template 01 assigned to all extensions by default, Caller ID of the Held Party is
selected.
But if you want to show the Caller ID of the Transferring party, select this option in the default Station Advanced
Feature Template 01.
If all certain extensions are to be provided Caller ID of the Held Party and others the Caller ID of the Transferring
Party,
1. Create separate Station Advanced Feature Templates.
2. Select 'Held Party' or 'Transferring Party' as desired in each Template.
3. Apply this Template on the extensions.
Configuring Caller ID Presentation while Transfer using Jeeves

• Under Configuration, click Station Advanced Feature Templates to open the page.
• In Caller ID Presentation while Transfer field, select the desired option: Held Party or Transferring
Party.
• Log out of Jeeves or continue programming.
Configuring Caller ID Presentation while Transfer using a Telephone

• Dial command 5602-1-Template Number-Feature Number-Code

Where,
Template Number is Station Advanced Feature Template Number from 01 to 50

Parameter Number is 01 for Caller ID Presentation while Transfer.
Code is:
1 for Transferring Party
2 for Held Party
For instance: To select Transferring Party in Template Number 02, dial 5602-1-02-01-1
• Exit SE mode
Refer the topic “Station Advanced Feature Template” for instructions assigning Templates to extensions.
Replacing '+' in CLI

To replace '+' characters received as CLI on telephones that do not support CLI prefixed with this character, you
must program the flag 'Replace '+' from CLI?' and the desired number string that should replace this character.
Configuring Replace '+' in received CLI using Jeeves

• Under Configuration, click System Parameters to open the page.
• Enable the Replace '+' from CLI flag by selecting the check box.
• Enter the desired number string in the field Replace '+' from CLI with the number string.
Configuring Replace '+' using Telephone

To replace '+', dial:

• 5334-Code

Where,
Code is
0 for Disable (no modification in CLI)
1 for Enable (replace '+' in CLI with the substitute number string programmed)
Default: 0
To program Replacement string, dial:

• 5335-Replacement String-#*
Where,
Replacement string is any string with digits from 0 to 9. A maximum of 6 digits are permitted. Terminate
the command with #* if the replacement string is fewer than 6 digits.
To remove the replacement string, dial:

• 5335-#*
• Exit SE mode.

Calling Line Identification Restriction (CLIR)
What's this?
The SARVAM UCS allows extension users to suppress their identity, that is, extension number and name, when
they call other extensions. This feature is called Calling Line Identification Restriction (CLIR).
Extensions that have 'CLIR Override' facility can view the CLI of those that have suppressed it with CLIR.
This is a feature of the system and not the PSTN. It is applicable for internal calls only.
This feature will work only on the Matrix Extended IP Phone, VARTA UC Clients, proprietary digital key phone EON
and SLTs that support Caller Line Identification (CLI).
How it works
• Extension A has CLIR enabled.
• Extension B does not have CLIR Override enabled.
• Extension C has CLIR Override enabled.
• When A calls B, B cannot view the extension name and number of A.
• When A calls C, C can view A's extension name and number.
Now,
• Extension D calls extension E.
• A picks up the call.
• D will be able to view A's name and extension only if it has CLIR Override enabled and is a digital key
phone, EON.
• CLIP and Caller ID Presentation while Transfer: Both these features will not work if CLIR is enabled.
How to configure
CLIR and CLIR Override are Class-of-Service-dependent features. Extensions that are to be allowed these
features, must have them enabled in their Class of Service (CoS) group.
Decide which extensions should be allowed CLIR and which should be allowed CLIR Override.
SARVAM UCS. Template 01 is assigned CoS group 01 in which both CLIR and CLIR Override are disabled. Thus,
none of the extensions of the SARVAM UCS can suppress their CLI or force any other extension to display its CLI.
If you want to enable both features on all extensions, simply enable CLIR and CLIR Override in the default CoS
group 01.

If you want to allow CLIR to all extensions, but not allow CLIR Override to any extension, simply enable CLIR in the
default CoS group 01.
If you want to allow CLIR and/or CLIR Override to selected extensions, only, then follow these steps:
1. Define a new CoS group with CLIR/CLIR Override enabled.
3. Assign this new Template to the extensions to which CLIR/CLIR Override is to be allowed.
programming.
How to use

To enable CLIR
• Press DSS Key assigned to CLIR230.

OR
• Dial 1031 to enable CLIR.
• You get confirmatory tone and message on the phone's display.
To disable CLIR:
• Press DSS Key assigned to CLIR again231.

OR
• Dial 1030.
• You get confirmatory tone and message on the phone's display.
For SLT Users

To enable CLIR:
• Lift handset.
• Dial 103-1.
To disable CLIR:
• Lift handset.
• Dial 103-0.
230. System Engineer is recommended to assign a DSS Key with LED to this feature. When the assigned DSS key is pressed, it will
glow red indicating that CLIR is enabled.
231. If a DSS key with LED has been assigned, when you press the key again, the LED will be turned off indicating CLIR is now
canceled.


Cancel All Station Features
What's this?
For each feature of the SARVAM UCS that extension users have set/enabled on their extension, the system
provides access code for cancellation of the feature.
Extension users can also cancel all features set on their extension by dialing a feature access code. These set of
features can also be canceled for any extension user from the SA Jeeves.
When an extension user dials the 'Cancel All Station Features' access code, the following features, if set, are
canceled from the extension:
• Auto Answer
• Auto Call Back
• Call Forward
• Do Not Disturb
• Hot Line
• Trunk Reservation
How to use
The extension users can cancel all features set on their extension from their own extension or these can be
canceled from SA Jeeves for any extension user.
Cancel All Features by Extension Users

• Press DSS Key assigned to 'Cancel All Features' (if programmed).
OR
• Dial 1051.
• You get confirmation tone and confirmatory message on your phone display.
• Go idle or wait for dial tone.
For SLT Users

• Lift handset.
• Dial 1051.

Cancel All Features for Extension Users
The Operator or any extension user having access to System Administrator mode can cancel all features for
extension users using Jeeves.
• Click Submit.
• Click Cancel All Features to expand.
• Click the Cancel All Features button. The following features will get canceled if any is set on this
extension user:
• Walk-in Class of Service
• Do Not Disturb
• Call Forward
• Call Forward - Scheduled
• Hotline
• Auto Answer
• Auto Redial
• Auto Call Back

Class of Service (COS)
What's this?
Class of Service (CoS) defines the permission an extension will have on a System. It defines the set features of the
system that the extension is to be allowed access to.
Feature requirements vary among users and with time. Certain groups of extension users may have a need for
voice mail, while another group may need the ability to forward calls to a cell phone, and still others may have no
need to make calls outside the office.
Similarly, certain features that are required during working hours may not be required during break or non-working
hours.
SARVAM UCS offers the flexibility to allow or deny extension users access to features of the System, on the basis
of their requirement and according to time of the day. For users, access to various features from their extensions is
their CoS.
How it works
The list of all features allowed to an extension is referred to as 'CoS group'. There are 20 CoS groups numbered
from 01 to 20.
In each CoS group there are 60 features, which are identified by 2-digit numbers, from 01 to 60. These are referred
to as the 'CoS Feature Numbers'.
Each extension port of the System has an associated CoS group that indicates which features of the System the
port is allowed to access.
The CoS group of an extension port is defined in the “Station Basic Feature Template” applied to that extension
port. It is defined for each "Time Zone", namely, working hours, break hours, and non-working hours, in the
Template.
A feature can be allowed or denied to an extension by enabling or disabling it in the CoS group of the Station Basic
Feature Template applied to that extension.
The same CoS group uniformly to all extensions ports for all Time Zones. Doing so, all extensions can access the
same set of features in all time zones. For example: CoS group 03 is assigned to all extensions for Working, Break
and Non-Working hours.
A different CoS group for each Time Zone can be assigned to all extension ports. Doing so, all extensions can
access only those features allowed for the particular Time Zone.
For example: All extensions are assigned CoS group 03 for Working, CoS group 04 for Break hours and CoS group
05 for Non-Working Hours.
Different CoS groups can be assigned to different extension ports, for all or for different Time Zones. Doing so,
each extension can access a different set of features in each Time Zone.

For example: extensions 3001 to 3010 are allowed CoS group 03 for all Time Zones, while extensions 3011 to 3015
are assigned CoS group 03 during Working Hours, and for the Non-Working and Break Hours, they are assigned
CoS group 04 and 05 respectively.
The following features can be set/cancelled on extensions from the SA mode, regardless of whether these
features are allowed or denied in the CoS assigned to the extensions:
• Call Forward
• DND
• Dynamic Lock and Timer
• Hotline
Basic Features
A set of features including Internal Call, Call Hold, Call Toggle, Call Transfer, Department Call, Operator Access,
Redial, and Call Mute defined as Basic Features and allowed in all CoS groups.
It is not possible to enable or disable selectively any of the features included in "Basic Features".
How to configure
The table below presents the CoS groups from 01 to 20 with the list of 01 to 60 features supported on the
extensions.
Default CoS Groups
Feature Class of Service Group

Feature Name
Number 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20
01 Account Code                    
02 SA Mode
03 SA Extension                    
04 Auto Call Back-Busy                    
05 Auto Call Back-No Reply                    
06 Auto Redial                    
07 Auto Redial Priority                    
08 Barge-In                    
09 Call Forward                    
10 Call Park                    
11 Call Pickup                    
12 Change Room Clean Status                    
13 Global Directory Programming
14 CLIR                    
15 CLIR Override                    
16 Trunk Call Waiting                    
17 Conference                    
18 Continued Dialing                    
19 Conversation Recording                    
20 Decrement Dynamic Lock Timer                    
for Internal Calls
21 DISA                    
22 Do Not Disturb                    
23 Do Not Disturb - Override
24 Dynamic Lock                    
27 Forced Answer                    

Feature Class of Service Group
Feature Name
Number 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20
28 Forced Release                    
29 Forced Release Order
30 Global Directory Part-1                    
31 Global Directory Part-2                    
32 Global Directory Part-3                    
33 Hot Desk                    
34 Hotline                    
35 Interrupt Request                    
36 Live Call Screening                    
37 Live Call Supervision                    
38 Manual Priority Intrusion
39 Message Wait (Set/Cancel)                    
40 Mini Bar Details
41 Paging                    
42 Privacy from Built-In Auto                    
Attendant
43 Privacy from Interrupt Request,                    
Barge-In and DND-Override
44 Privacy from Raid                    
45 Raid                    
46 Return Call to Original Caller                    
(RCOC)
47 Room Monitor                    
48 Selective Port Access                    
49 Trunk Reservation                    
50 Trunk-Trunk Transfer                    
51 Basic Featuresa                    
52 General MailBox                    
53 Dept. Group-Call Forward                    
54 Emergency Conference                    
55 Intercom
56 DSS Call Pickup-Station
57 DSS Call Pickup-Trunk
58 Closed User Group (CUG)                    
59 PIN Dialing
a. Basic Features includes: Internal Call, Call Hold, Call Toggle, Call Transfer, Department Call, Operator Access,
Redial, Mute.
• CoS group number 01 is assigned for all Time Zones in the default Station Basic Feature Template 01
assigned to all extensions of the SARVAM UCS.
• CoS group number 19 and 20 are assigned when the Hospitality Application of SARVAM UCS is used.
When this application is being used in CoS group number 20, the SA Mode will be enabled. See
SARVAM UCS Hospitality System Manual.
Creating CoS Groups

• Take a pen and a paper pad.
• Prepare a list of extensions.

• Read the list of features supported on the extensions (see above Table 'Default CoS Groups for
Features').
• Against each extension name on the list, write the features needed for each Time Zone. You will notice
that the features needed by many extensions are identical.
• List the common features to be allowed to and features to be denied to all extensions. Assign a CoS Group
Number to this list.
• Are there any other features, in addition to those on the common list, which you want to allow to selected
extensions?
• If yes, extend the common list you prepared by adding the features to be allowed to selected extensions.
Assign a CoS Group Number to this extended list.
• You can prepare different CoS Groups for different Time Zones and assign a number to each group.
• For example, you may end up creating five different CoS groups. The First group may contain none of the
features. The Second group may contain the most common features like Call Forward, Call Transfer,
Internal Dialing, etc. The Third group may contain more advanced features, and the Fourth group may
contain even more advanced features. The Fifth group may contain all the features.
• When you are finished preparing the CoS groups you need, program the CoS groups using Jeeves or by
issuing SE commands from a Telephone (DKP or SLT). See below for instructions.
• Now, the CoS groups to be assigned to extensions must be programmed in the Station Basic Feature
Template applied to the extensions. This can be done using Jeeves or by issuing SE commands from a
Telephone (DKP or SLT). See below for instructions.
Enabling a feature in a CoS Group using Jeeves


• Under Configuration, Click the link Class of Service (CoS).
• The default CoS groups from 01 to 20 appear. The check boxes selected under each CoS group column
indicate that the feature is enabled in that CoS group. The default CoS groups meet the requirements of
most extension users. Check the default CoS groups whether the features you want to allow are enabled
and features you want to deny are disabled.
• To enable a feature in a CoS group, select corresponding check box in the CoS group. To disable a
feature simply clear the check box. For example: to enable DND-Override in CoS group 01, select the
check box against DND-Override in CoS group 01. To disable clear the check box again.
• Click 'Submit' to save changes.
Enabling a feature in a CoS Group using a Telephone

To enable a feature in a CoS group, dial:

• 1302-1-CoS Group-Feature Number-Code to enable a feature in a single CoS group.
• 1302-2-CoS Group-CoS Group-Feature Number-Code to enable the same feature in a range of CoS
groups.
• 1302-*-Feature Number-Code to enable the same feature in all CoS groups.
Where,
CoS Group from 01 to 20.
Feature Number from 01 to 60. Refer the Table “Default CoS Groups” for Feature Numbers.
Code is
1 for Enable
0 for Disable
To default a CoS group, dial:

• 1301-1-CoS Group to default a single CoS group.
• 1301-2-CoS Group-CoS Group to default a range of CoS groups.
• 1301-* to default all CoS groups.

Where,
CoS Group is from 01 to 20.
• Exit SE mode.
Assigning a CoS group in a Station Basic Feature Template using Jeeves

• Log in as System Engineer, if not already logged in.
• Under Configuration, click Station Basic Feature Template.
• The default CoS group assigned to each time zone, that is, working hour (WH), non-working hour (NH) and
break hour (BH), appears under Class of Service in each Template.
• To assign a CoS group to a Station Basic Feature Template, enter the CoS group number for each time
zone under Class of Service.
By default, Station Basic Feature Template 01 is applied on all extensions and CoS group 01 is the
assigned by default to this template in all time zones.
If all extensions to be allowed the same set of features during working hours, break hours, non-working
hours, enter the same CoS group number in all time zones in the Template Number applied to all
extensions. For example: to assign CoS group 04 to all time zones in Template Number 01; enter 04 under
WH, NH and BH.

If all extensions are to be allowed a different set of features in each Time Zone, enter the CoS group for
each Time Zone. For example: to assign CoS group 03 in working hours, 04 in Break Hours and 05 in Non-
Working hours in Template 01, enter 03, 04, 05 under WH, BH and NH respectively.

• If a set of features is to be allowed to select extensions only, assign the CoS group with these features
enabled to a separate Station Basic Feature Template. Apply this template to the select extensions which
are to be allowed this CoS.
For example: To assign all features to extensions, create a CoS group with all features enabled, CoS
group 07. Select a different Station Basic Feature Template, for example 05. Enter CoS Group 07 in all
Time Zones in Template 05. Apply Template 05 to the software ports of the extensions that are to be
assigned all features.
• Remember to click Submit to save the changes you make on every page.
Assigning CoS group to Station Basic Feature Template using a Telephone

To assign CoS group to a template, dial:

Where,
Template Number is Station Basic Feature Template Number, from 01 to 50.
Feature Number is
03 for CoS group for Working Hours
04 for CoS group for Break Hours
05 for CoS group for Non-Working Hours
Code is CoS group number from 01 to 20.
For Example: To program CoS group 04 in Station Basic Feature Template Number 06 for all Time
Zones, dial:
• 5502-1-06-03-04 to program CoS group 04 for Working Hours.
• 5502-1-06-04-04 to program CoS group 04 for Break Hours.
• 5502-1-06-05-04 to program CoS group 04 for Non-Working Hours.

Similarly, to program CoS Group 03 in working hours, 04 in Break Hours and 05 in Non-Working hours,
dial:
• 5502-1-06-03-03 to program CoS group 03 for Working Hours.
• 5502-1-06-04-04 to program CoS group 04 for Break Hours.
• 5502-1-06-05-05 to program CoS group 05 for Non-Working Hours.
• Exit SE mode.
After you have programmed the CoS group in the Station Basic Feature Template, you must assign this
template to the stations. Refer the topic “Station Basic Feature Template” for instructions on applying
templates on SLT, DKP, ISDN Terminal and SIP extensions.
Finally, test the CoS programmed for each extension by invoking the features from each extension.

CLI Based Routing
What is this?
SARVAM UCS offers the facility to detect the calling party's number and name. This is known as Calling Line
Identification.
On the basis of CLI, it is possible to land calls from a particular telephone number on a particular extension or group
of extensions. This is known as CLI Based Routing.
How it works
A, B, C are extensions. D and E are external callers.
Calls made by D are to be landed on A.
Calls made by E are to be landed on B and C.
The CLI of D and E and their corresponding landing destinations should be entered in the CLI Based Routing
Table.
CLI Based Routing should be enabled on the desired trunks for each Time Zone (working hours, break hours and
non-working hours).
The system can match the incoming call CLI with the numbers configured in the CLI table in two ways, that is,
Match from last digit of CLI or Match from first digit of CLI.
• If you select Match from last digit of CLI, this is how the call will be routed:
• D calls on a trunk of SARVAM UCS.
• The system checks if CLI Based Routing is enabled on the trunk for the current time zone.
• If CLI Based Routing is enabled on the trunk, the system checks the numbers stored in the CLI Based
Routing table.
• D's number is found in the CLI Based Routing Table.
• The system checks the destination number stored against D's CLI.
• A's number is found as the destination extension.
• The system lands the call on A.
• If you select Match from first digit of CLI, this is how the call will be routed:
• D calls on a trunk of SARVAM UCS.
• The system checks if CLI Based Routing is enabled on the trunk for the current time zone.
• Then the system checks if the parameter Replace '+' from CLI, is enabled.
• If enabled, the system will replace + sign received in the incoming CLI with the number string
configured in the Replace '+' from CLI with the number string. To know more, see “System
Parameters”.
• Now the system checks if Incoming CLI Modification is enabled.
• If enabled the system modifies the incoming number according to the parameters configured in the
Incoming CLI Modification. To know more, see “Incoming CLI Modification” in “System Parameters”.
• The system matches the modified number string with the numbers stored in the CLI Based Routing
table.
• D's number is found in the CLI Based Routing Table.
• The system checks the destination number stored against D's CLI.
• A's number is found as the destination extension.

• The system lands the call on A.
If D's number does not exist in the CLI Based Routing Table, the call will be routed according to the incoming
call management logic.
How to configure
For this feature to work, you must do the following:
• enter the numbers of the calling parties and the numbers of the corresponding destination extensions in
the CLI Based Routing Table. You can store up to 2000 numbers in the CLI Routing Table.
• enable CLI Based Routing on the desired trunks according to time zones in their “Trunk Feature
Template”.
Creating CLI Routing Table

To apply this feature,
• On a sheet of paper, create a 5-column table, as illustrated below. Each calling party number in the CLI
table is stored a location index in the system. Enter the telephone numbers and names of the calling
parties and the corresponding landing destinations, that is, the Port Type and Port Number. The Port Type
may be SLT, DKP, ISDN Terminal, SIP Extension, a Routing Group or Virtual Extension or a Voice Mail
Auto Attendant.
• The 'Name' field is for identifying the entry. When placing a call on the destination extensions, both the
number and the 'Name' are presented in the CLI.
• Determine the method which the system should use to match the incoming CLI with the numbers in the
table, that is, Match from last digit of CLI or Match from first digit of CLI Based Routing Table.
Index Telephone Number Name Port Type Port Number
1 2640459 MidasBiz Routing Group 02
2 022281110001 Jet Set SLT 004
: :
10 2640075 Bacchus Routing Group 03
Configuring CLI Based Routing using Jeeves


• Under Configuration, click CLI Based Routing.
• In Method for matching received CLI, select the method according to which you want the system to
match the received CLI with the numbers stored in the CLI table. You can select:
• Match from last digit of CLI
• Match from first digit of the CLI
The method you select will be applicable to all the numbers configured in the CLI Based Routing Table.
• In the CLI table each number is to be stored at a Location Index numbered from 0001 to 2000.
There are 100 entries on each page. To go to the next 100 Index numbers, click the tabs 0101-0200, 0201-
0300, 0301-0400 , 0301-0400.................................1901-2000.
• At each Location Index, enter the information for the following parameters:
• Calling Party’s Number: enter the number of the calling party, not exceeding 16 digits. You can also
enter '+' in the number string.
• Calling Party’s Name: enter the name of the calling party. You can enter a maximum of 8 characters
in this field.
• Destination Type: select the landing destination extension. It may be an SLT, a DKP, an ISDN
Terminal, SIP extension, a Routing Group, a Virtual Extension or Voice Mail Auto Attendant.
• Port Number: enter the port number (software port or routing group number) to which the landing
destination is connected. This is applicable for Destination Types — SLT, DKP, ISDN Terminal, SIP
Extension or Routing Group.
If you have selected Virtual Extension as the Destination Type, enter the software port of landing
destination of the Virtual Extension.
If you have selected a Routing Group as the Destination Type, enter the number of the Routing Group
(01 to 96) in this field.
Destination Type, select the VMAA Menu to be assigned to the respective calling party.

• Click Submit to save entries.
• Enable CLI Based Routing for the desired Trunks in their “Trunk Feature Template”. Refer the topic
“Customizing Trunk Feature Templates” for instructions.
Configuring CLI Based Routing Table using a Telephone

To add the calling party telephone number in the CLI Based Routing table, dial:
• 4101-Index-Telephone Number-#*
Where,
Index is from 0001 to 2000.
Telephone Number is the calling party's telephone number (Max. 16 digits).
Terminate the command with #*, if the number is less than 16 digits.
To enter '+' in the number string, dial #8
For Example: to enter 2640459 at Index Location 001, dial 4101-001-2640459-#*
To clear a telephone number from the CLI Based Routing table, dial:
• 4101-Index-#*
To enter the name of the calling party corresponding to the calling party's telephone number, dial:
• 4102-Index-Name-#*
Where, Index is from 0001 to 2000.
Name can be a maximum of 8 characters.
Terminate the name with #*, if less than 8 characters.
For Example: to enter the name Midas Biz for the number 2640459 at Index Location 001, dial 4102-
001-MidasBiz
To clear the name stored at a location index in the CLI table, dial:
• 4102-Index-#*
To assign the landing destination extension in the CLI table, dial:

• 4103-Index-Port Type-Port Number
Where,
Port Type is
00 for Null
01 for SLT
02 for DKP
20 for Routing Group
28 for ISDN Terminal
34 for SIP Extension
36 for Virtual Extension
Port Number is Software Port number of SLT/DKP/ISDN Terminal/SIP Extension.

Port Number of SLT is from 001 to 240.
Port Number of DKP is from 001 to 96.
Port Number of ISDN Terminal is 01 to 64.

Port Number of Routing Group is from 01 to 96.
Port Number of SIP Extension is from 001 to 999.
Port Number of Virtual Extension is from 01 to 64.
For Example: to assign an SLT connected at Software Port number 004 as the landing destination for
calls from Midas Biz 2640459, dial 4103-001-01-004
To clear a location index in the CLI Based Routing table, dial:

• 4104-Index to clear a single location Index.
Where,
By using these commands, the telephone number, name and the Port type-Port number will be cleared
from the location index.
• Exit SE mode.

Clock Synchronization
What's this?
When data is transmitted from the SARVAM UCS to external lines or when SARVAM UCS receives data from the
external lines, it is necessary that the transmitter and receiver be properly synchronized. If not clock slips can occur.
A clock slip can generate a loss or addition of data to the data stream.
How it works
• This can be done in three ways viz. using the data clock or using the external clock (clock is sent by the
network on a dedicated cable pair) or using the internal clock. SARVAM UCS does not support external
clock. When the SARVAM UCS is connected to the PSTN, then it is recommended to extract the clock
from the incoming data whereas if the SARVAM UCS is used to form a private network, you are
recommended to use the internal clock. For example, if a private network is formed by connecting three
SARVAM UCS systems, then one system should be programmed as master clock whereas other two
should be programmed in the slave mode.
• If two or more T1E1 Ports are connected to the PSTN (or a Private Network) then in such case, clock will
be extracted from the first T1E1 Software port whereas the transmit data on all other ports whether
connected to PSTN or private network will be clocked as per received.
How to configure
Configuring Clock Synchronization using Jeeves

• Click Clock Synchronization to expand.

• Clock Source - Priority 1to 4: In the Priority levels, Clock Source Priority-1 to Clock Source Priority-4,
select the Clock Source option as per your preference.
By Default: Priority 1 is T1E1-001, Priority 2 is T1E1-002, Priority 3 is T1E1-003 and Priority 4 is T1E1-004.
• Clock Synchronization Frequency: Select Clock Synchronization Frequency in this field. You can select
from the following options:
• 8 KHz Derived
• 8 KHz
• 2.048 MHz
• 1.54 MHz
By Default, Clock Synchronization Frequency is set to8 KHz.
The system will restart when the frequency is changed.
• PLL Locking Mode: Depending upon the speed required for clock synchronization, select the speed for
PLL Locking Mode. You can set PLL Locking Mode to either fast or slow. By default, it is slow.
• PLL TIE Control: You can enable or disable PLL TIE Control. By default, it is disabled.
• PLL Operating Mode: Select the PLL Operating Mode in this field. You can select from the following
options:
• Normal
• Hold Over
• Free Run
By default, PLL Operating Mode is Normal.
You can program PLL TIE Control and PLL Operating Mode only using Jeeves.

Configuring Clock Synchronization using a Telephone
Clock Source for 'Master Clock Synchronization'

The SARVAM UCS supports 4 clock sources which can be programmed for the specific port.
Use the following command to configure the clock sources:

5341-Clock Source Index-Port Type-Port Offset
Where,
Port Offset is 01 to 32.
Clock Source Index is from 1 to 4.
Port Type Meaning Port Offset
05 T1E1 01-08
04 BRI 01-32
00 Null 000
0 is a valid port to follow internal clock.
Default Table for the clock source, programmed in the system
Clock Source Index Port Type-Port Offset
1 T1E1-1
2 T1E1-2
3 T1E1-3
4 T1E1-4
The system checks this table for a master clock. If none of the ports is synchronized out of this table, the system
gives priority to the internal clock. If any one port is synchronized, the system selects that port as a system clock
master. Here index is given priority, that is, if the second port of this table is selected as clock master and suddenly
first port is synchronized, then the system changes its master from 2nd port to first port. Now if first port has lost its
synchronization then in this case again, the second port is selected as master clock of the system.
Clock Synchronization Frequency

Use following command to select Clock Synchronization Frequency:
5342-Clock Synchronization Frequency
Where,
Clock Synchronization Frequency is from 1 to 4.
System Clock Synchronization Meaning
1 8 KHz Derived
2 8 KHz
3 2.048 MHz
4 1.54 MHz

By Default, 'System Clock Synchronization' is 2.048 MHz for India and other countries except USA.
For USA, default 'System Clock Synchronization' is 1.54 MHz.
PLL Locking Mode

This parameter is used to select the suitable speed of the PLL locking in the system.
Use following command to select the 'PLL Locking Mode':

5343-PLL Locking Mode
Where,
PLL Locking Mode is from 1 to 2.
PLL Locking Mode Meaning
1 Fast
2 Slow
By default, PLL Locking Mode is 'Slow'.

Closed User Group (CUG)
What’s this?
To have private networks, few systems can be connected to each other using E&M, T1/E1, QSIG, etc. The
requirement demands that the systems connected to each other forming the network behave as a single group.
The users need not dial a separate code to access an extension user of other system. The entire network should
behave as a single unit. Users will not know whether they are dialing an extension number of their own system or of
the other system. This is called Closed User Group.
System-A System-B
T1 E&M1 T1
T2 T2
PSTN PSTN
E&M2 E&M3
Tn Tn
S1 S2 Sn S1 S2 Sn
2001 2002 2003 3001 3002 3003
System-C
S1 S2 Sn
4001 4002 4003
In the above figure, 3 Systems are connected through E&M connectivity.
• T1 to Tn are trunk lines from the local central office (CO).
• S1 to Sn are extensions.
• E&M1 to E&M3 are E&M lines between the three Systems.
How it works
For Closed User Group, you must:
• have unique extension number in all the systems, that is, one cannot have extension number 2001 in
System-A as well as in System-B or System-C.
• enable Closed User Group (CUG) in the Class of Service assigned to the extension users.

Few new words have been used to explain this application, each of these words have been explained below:
• Closed User Group Table: This table has the following parameters viz. Route Index, Route Code, OG
Trunk Bundle Group, Strip Digit Count, Self Route check box, Apply Toll Control check box and Apply Call
Cost check box. The closed user group programming works according to this table.
Route Strip Digit Self Dialed Digit Apply Toll Apply Call
Index OGTBG
Code Count Route Count Control Cost
001
002
003
250
• Index: Maximum 250 different routes (001-250) can be programmed.
• Route Code: Route code could be of maximum sixteen digits. Valid Digits: 0 to 9, * # A B C D F P, where
P is Pause, F is Flash, A to D is DTMF Digits. Generally route code will be a truncated number of the
extension numbers. For example in the figure given above, route code for System-B can be defined as ‘3’
and that for System-C can be defined as ‘4’.
If System-B were having extension numbers from 3100 to 3199 and System-C were having extension
numbers from 3200 to 3299 then route code for System-B can be defined as ‘31’ and that for System-C
can be defined as ‘32’. If System-B were having extension numbers from 301 to 399 and 401 to 499 then
two route codes can be defined for System-B viz. ‘3’ and ‘4’. Likewise for System-C.
• OG Trunk Bundle Group: An OG Trunk Bundle Group (OGTBG) is assigned to each route code.
Whenever a call is to be made on that route, a free trunk from the OGTBG is selected and the extension
number is dialed on it. The same logic of Rotation On/Off for trunk selection from the OGTBG is used. If
rotation is OFF then always the first trunk in the OGTBG is selected. If it is busy then the next trunk in the
group is selected. This helps to select an alternate route. Whereas if Rotation is ON then the trunks in the
OGTBG are selected in round robin fashion.
• Strip Digit Count: For the Closed User Group application, configure this parameter as 0. This parameter
is relevant when you are configuring “Closed User Group-With Exchange ID”.
• Self-Route: For the Closed User Group application make sure this is disabled. This parameter is relevant
when you are configuring “Closed User Group-With Exchange ID”.
• Dialed Digit Count: When digits are dialed on the trunk, the system waits for inter digit timer after the last
digit is dialed. In order to avoid this timer and number of digits dialed to be routed without further delay,
count for the number of digits to be programmed in this field. If the number of digits received are equal to
the parameters programmed then the number is dialed out immediately without waiting for the inter digit
timer. If the number of digits dialed by the user are not equal to the digits programmed, the number is
dialed after inter digit timer.
• Apply Toll Control: When Self Route check box is disabled, system will check this parameter.
By default, the Apply Toll Control check box is enabled. The system will apply toll control to all the outgoing
calls. Disable this check box, if you do not want to apply toll control to the CUG numbers dialed by you.

For detailed information, see “Toll Control”.
Apply Call Cost: By default, this check box is enabled and the system will calculate the cost of each call.
For certain calls (internal calls) you do not require the call cost calculation, clear the check box
corresponding to these entries. You can also set the filter Calls with units more than to generate a report
according to the Call Cost. For details, see “Station Message Detail Recording-Report”.
The SARVAM UCS has only one routing table. The same table is used for Closed User Group and Close
User Group-With Exchange ID applications. Hence the table has to be programmed keeping the
application in mind.
Configuring using Jeeves

To configure the Closed User Groups,
• Under Configuration, click Closed User Groups.
Against each Index configure the following parameters:
• Route Code: Route code could be of maximum sixteen digits. Valid Digits: 0 to 9, * # A B C D F P,
where P is Pause, F is Flash, A to D is DTMF Digits. Generally route code will be a truncated number of
the extension numbers. For example in the figure given above, route code for System-B can be defined
as ‘3’ and that for System-C can be defined as ‘4’.
If System-B were having extension numbers from 3100 to 3199 and System-C were having extension
numbers from 3200 to 3299 then route code for System-B can be defined as ‘31’ and that for System-C

can be defined as ‘32’. If System-B were having extension numbers from 301 to 399 and 401 to 499
then two route codes can be defined for System-B viz. ‘3’ and ‘4’. Likewise for System-C.
Whenever a call is to be made on that route, a free trunk from the OGTBG is selected and the
extension number is dialed on it. The same logic of Rotation On/Off for trunk selection from the OGTBG
is used. If rotation is OFF then always the first trunk in the OGTBG is selected. If it is busy then the next
trunk in the group is selected. This helps to select an alternate route. Whereas if Rotation is ON then
the trunks in the OGTBG are selected in round robin fashion.
• Strip Digit Count: It has no significance for Closed User Group application. But it has to be
programmed as 0.
• Self-Route: It has no significance for Closed User Group application. But it has to be programmed as
0.
• Dialed Digit Count: When digits are dialed on the trunk, the system waits for inter digit timer after the
last digit is dialed. In order to avoid this timer and number of digits dialed to be routed without further
delay, count for the number of digits to be programmed in this field. If the number of digits received are
equal to the parameters programmed then the number is dialed out immediately without waiting for the
inter digit timer. If the number of digits dialed by the user are not equal to the digits programmed, the
number is dialed after inter digit timer.
• Apply Toll Control: By default, this check box is enabled. The system will apply toll control to all the
outgoing calls. Disable this check box, if you do not want to apply toll control to the CUG numbers
dialed by you.
• Apply Call Cost: By default, this check box is enabled and the system will calculate the cost of each
call.
corresponding to these entries.
To configure a single port: XXXX-1
To configure a range of ports: XXXX-2
To configure all the ports: XXXX-*
Step 1
Use following command to configure route code:
4502-1-Route Index-Route Code-#*
4502-2-Route Index-Route Index-Route Code-#*
4502-*-Route Code-#*
Where,
Route Index is from 001 to 250.
Route Code is a sixteen digits string of numbers.

Use following command to clear a particular route code:
4502-1-Route Index-#*
4502-2-Route Index-Route Index-#*
4502-*-#*
Where,
By default, Program Route Code is Blank.
Step 2
Use following command to assign OG Trunk Bundle Group to the route code:
4503-1-Route Index-OG Trunk Bundle Group
4503-2-Route Index-Route Index-OGTBG
4503-*-OG Trunk Bundle Group
Where,
OG Trunk Bundle Group is from 01 to 32.
By default, OG Trunk Bundle Group is 01.
Step 3
Use following command to configure strip digit count for a route:
4504-1-Route Index-Strip Digit Count
4504-2-Route Index-Route Index-Strip Digit Count
4504-*-Strip Digit Count
Where,
Router Index is from 001 to 250.
Strip Digit Count is from 0 to 9.
By default, Strip Digit Count is 0.
Step 4
Use following command to configure self-route check box for a route:
4505-1-Route Index-Code
4505-2-Route Index-Route Index-Code
4505-*-Code
Where,
Code Meaning
0 Disable Self-route check box
1 Enable Self-route check box
By default, Self-route check box is disabled.
Step 5
Use following command to configure maximum dialed digits to select router for a route code:
4506-1-Route Index-Dialed Digit Count
4506-2-Route Index-Route Index-Dialed Digit Count
4506-*-Dialed Digit Count
Where,
Maximum dialed digits is from 00 to 99.

Step 6
Use following command to clear an entry in a routing table:
4501-1-Route Index
4501-2-Route Index-Route Index
4501-*
Where,
SARVAM UCS offers few features associated with closed user group. Each of these are discussed below:
• Alternate Route:This feature provides flexibility of accessing an extension of other exchange through
alternate routes if the normally used route (the shortest route) is not free. To achieve this, the trunks that
offer shortest routes should be programmed first in the OGTBG followed by the trunks that provide
alternate routes. Also the rotation within the OGTBG should be OFF. In figure the requirement is that if
E&M1 is busy then E&M2 should be used to call 3001 from System-A. In this case a OGTBG is to be so
formed that it has E&M1 as first trunk and E&M2 as second trunk and should be assigned to route code ‘3’.
However, the rotation within the OGTBG should be disabled. Similarly, if the call is to be made to 4001
then E&M2 should be used. Hence another OGTBG should be programmed with E&M2 as first trunk and
E&M1 as second trunk and it should be assigned to route code ‘4’. Also the rotation option for the OGTBG
should be selected.
• Transit Barring:This feature helps to bar the Transit calls through the exchange. Consider figure 1. It is
required that an extension user in System-A can access extension 3001 using alternate route through
System-C but an extension user in System-B cannot access extensions 2001 to 2010 using alternate
route through System-C. This can be accomplished using Transit Barring. To achieve this, a denied list
containing 10 numbers viz. 2001 to 2010 should be assigned as Toll Control for the SLT port programmed
for the E&M2. Doing so when an extension user from System-B dials 2001, if E&M1 is busy, the system
would try dialing through E&M2 but since E&M2 does not have requisite toll control, it will give error tone to
extension user. Transit Barring adds value to the Alternate Route by allowing a selective access to the
extension with alternate route.
• Each system in the network has a routing table. By default, the routing is blank and hence E&M works
as per normal E&M connectivity. After programming the routing table as per the requirement, when the
user dials an extension number, the system first searches for the dialed number in the same system by
considering Self Route check box. If it is not available in its own system, it checks for it in the routing
table, finds the best fit, selects a free E&M path and reaches the dialed port.
• Please refer figure 1. When an extension in System-A dials an extension number 3001, the system
searches for this number in System-A. Since there is no extension with flexible number 3001 in
System-A, the system checks the E&M routing table. The system follows the routing table, identifies
that the dialed number is in System-B, selects a free E&M path and reaches the dialed port.
• As shown in figure 1, the shortest path to reach extension 3001 from System-A is through E&M1. But if
E&M1 is busy then the network can be programmed to reach extension 3001 through E&M2 and
E&M3. For this both System-A and System-B have to be programmed to accomplish this.
• Forced Call Disconnection: Please refer “Forced Call Disconnection” for more details.

Closed User Group-With Exchange ID
What’s this?
To have private networks, few systems can be connected to each other using E&M, T1/E1, QSIG, etc. The
requirement demands that the systems connected to each other forming the network behave as a single group.
The users need not dial a separate code to access an extension user of other System. The entire network should
behave as a single unit. The extension users will not know whether they are dialing an extension number of their
own system or other system. This is called Closed User Group. However, it is possible that the Systems connected
to form a network may have same extension numbers. Also, all the exchanges within the network may have their
own identity (called Exchange ID). In such cases, the routing scheme (the routing table) has to be programmed
keeping the Exchange ID (SID) in mind. This is known as Closed User Group-With Exchange ID.
This facility is generally used in PLCC Applications wherein new power extensions (and hence Systems) are added
in the network. It is not feasible to have unique extension numbers throughout the network. In such cases, an
Exchange ID is assigned to the newly added System and a routing table is programmed in the exchange. Also, the
routing tables of other exchanges are modified to include the newly added exchange in the network.
Figure 1: Open User Group Application
In the below figure, 3 Systems are connected through E&M connectivity.
• S1 to Sn are extensions.
• E&M1 to E&M3 are E&M lines between the three Systems.
System-A System-B
T1 21 E&M1
22 T1
T2 T2
PSTN PSTN
E&M2 E&M3
Tn Tn
S1 S2 Sn S1 S2 Sn
2001 2002 2099 2001 2002 2099
System-C 23
S1 S2 Sn
2001 2002 2099
How it works
In this application, it is possible to have same extension numbers in two or more Systems of the network, but you
must enable Closed User Group (CUG) in the Class of Service assigned to the extension users.

Few new words have been used to explain Closed User Group-With Exchange ID application, each of these words
have been explained above.
• Closed User Group Routing Table: This table has following parameters viz. Route Index, Route Code,
OG Trunk Bundle Group, Strip Digit Count, Self Route check box, Apply Toll Control check box and Apply
Call Cost check box. The Closed User Group-With Exchange ID programming works according to this
table.
Route Strip Digit Dialed Digit Apply Toll Apply Call

Index OGTBG Self Route
Code Count Count Control Cost
001
250
P is Pause, F is Flash, A to D is DTMF Digits. Generally, route code will be a unique number. The route
code should not clash with any of the extension numbers of same System. For example in the figure given
above, route code for System-A can be defined as ‘21’, route code for System-B can be defined as ‘22’
and that for System-C can be defined as ‘23’. This means that no extension in System-A can start with ‘22’
or ‘23’. Similarly, no extension in System-B can start with ‘21’ or ‘23’ and no extension in System-C start
with ‘21’ and ‘22’.
number is dialed on it. The same logic of rotation On/Off for trunk selection from the OGTBG is used. If
group is selected. This helps to select an alternate route. Whereas if rotation is ON then the trunks in the
• Strip Digit Count: This count signifies the number of digits to be stripped off while dialing/decoding a
number. To elaborate: Consider figure 1. The requirement is that if extension 2001 of System-B dials
212002 and if E&M 1 is busy then the call should reach extension 2002 of System-A through alternate
route. In this case the strip digit count of System-A should be programmed as 2 and that of System-B and
System-C should be programmed as 0. Doing so, when extension 2001 of System-B dials 212002 and if
E&M1 is busy then the call is routed through System-C. In this case, System-B dials 212002 on E&M3,
System-C receive this code and dials out the same code, that is, 212002 on E&M2 without striping of any
digit. On receiving 212002, System-A strips of two digits as per the programming and routes the call to
extension 2002.
• Self-Route: This flag signifies that the digits being dialed are for the same System and are not to be dialed
on the E&M trunk.

• Apply Toll Control: This parameter is not relevant as Self Route flag is enabled. This parameter is
relevant when you are configuring “Closed User Group (CUG)”.
• Apply Call Cost: By default, this check box is enabled and the system will calculate the cost of each call.
Please note that the SARVAM UCS has only one routing table. The same table is used for Closed User
Group and Closed User Group-With Exchange ID. Hence the table has to be programmed keeping the
application in mind.
How to configure
Please refer topic “Closed User Group (CUG)” for more details.
SARVAM UCS offers few features associated with Closed User Group-With Exchange ID. Each of these are
discussed below:
• Alternate Route: Please refer “Closed User Group (CUG)” for more details.
• Transit Barring: Please refer “Closed User Group (CUG)” for more details.
• Strip Digit Count is of significance in a network in which few exchanges possess Exchange ID whereas
others do not. Refer figure 2. System-A and System-B are made to work as Closed User Group since they
have unique extensions. But since System-C possess extensions whose flexible number clashes with
extensions in System-A, it cannot be made a part of Closed User Group. In such case Closed User
Group-with Exchange ID can be used with the combination of System-A + System-B and System-C
forming the network.
In this case, if 2001 of System-A wants to access 3001 through E&M1, he must dial 3001. But if E&M1 is
busy then system will allot him E&M2 and since there would not be any programming done, it will give
error tone to the caller. To avoid this condition, the extension user of System-A should be asked to call
3001 by dialing 223001. For this, the strip digit count for the route index with route code ‘22’ in the routing
table of System-B should be programmed as ‘2’ and the strip digit count for the route index with route code
‘22’ in the routing table of System-C should be programmed as ‘0’. Doing so, if the call to 3001 is made
through E&M1, then System-B would strip of the first two digits on receiving 223001 and make the caller
reach 3001. If the call to 3001 is routed through E&M2, the System-C will not strip of any digit and would
dial out 223001 on E&M3. On receiving 223001, System-B as per the programming would strip of 22 and
make the call land on 3001.
Strip Digit Count is also of importance in following case:

For example in figure 1, if extension 2001 of System-A dials 212002 then also the call should go to 2002 of
System-A only. To accomplish this, the strip digit count should be programmed as ‘2’ and Self Route flag
should be enabled. Doing so when 2001 dials 212002, the system strips off first two digits and checks for
the remaining digits, which in this case would be 2002 and thereby the user reaches 2002 of the same
System.
• Forced Call Disconnection: Please refer “Forced Call Disconnection” for more details.

• Prefix String Feature: This feature is programmed only for few Exchanges like ET1, Genesis, or BPL
which does not send ‘0’ when user dials a number to call an extension number of the exchange. For other
Exchanges like ‘ET2’ it is not required to be programmed as it sends ‘0’ also if it is dialed.
Maximum of 4-characters can be programmed as ‘Prefix String’.
‘Prefix String’ is a string of characters which is prefixed to the string, dialed by the user and then CUG
Routing-Table is applied.
Example:
Detailed application for ‘Prefix String’ feature is explained by following steps:
• If this feature is not programmed for Exchange of type: ET1 or BPL, and ‘Prefix String’ is blank, then,
only ‘2223’ is dialed by the Exchange when ‘02223’ is dialed by the user, to call extension ‘23’.
• Now when ‘2223’ reaches Exchange B, since there is no entry in the CUG table, the feature-extension
flexible number table is checked. Now since a match starting with ’22..’ is found, the call is routed to
extension ‘22’ instead of ‘23’, because extension number ‘22’ is present on the Extension.
• To avoid this, an entry is made in the routing table containing ‘22’ as the route code with strip digit
count ‘2’. But then since the system checks CUG Routing-table first, the first two digits out of ‘2223’
always get striped off and the call is not routed to extension ‘22’, that is, the other extensions of
Exchange B will never be able to call extension ‘22’.
• To solve this problem ‘Prefix String’ feature is programmed in the E&M Feature Template with ‘0’ as
prefix string, so that ‘string with prefix’ (022) is matched with the entries of CUG Routing Table (as
shown in figure). When user dials ‘02223’, first 3-digits are stripped off and the required Extension ‘23’
can be called.
Please refer topic “E&M Feature Template” to configure this feature.

Communication Ports
What's this?
SARVAM UCS supports two communication ports for ETERNITY GENX platform. The first COM Port is inbuilt on
the CPU Card and the other can be used by connecting the USB to COM converter in the “External USB Port
(Device Port) 3.0” of the CPU Card.
SARVAM UCS supports only one communication port for ETERNITY LENX and MENX platform. This can be used
by connecting the USB to COM converter in the “External USB Port (Device Port) 3.0” of the CPU Card.
SARVAM UCS supports serial, asynchronous, DB-9 connector for the Communication Ports.
A Communication Port (COM or USB to COM) is used for the following facilities:
• Programming SARVAM UCS using a PC

• PMS Interface (see SARVAM UCS Hospitality System Manual)
• SMDR Reports
• SMDR Online
• SMDR Posting
• Hotel-Motel Activity Log (see SARVAM UCS Hospitality System Manual)
A Communication Port is necessary for programming SARVAM UCS using a PC, whereas for other above listed
facilities, Communication Port may or may not be used232.
How to configure
In order for each of the above listed facilities to work, a Communication Port (COM or USB to COM Port) must be
assigned first as the 'Destination Port' and the attributes of the COM or USB to COM Port of SARVAM UCS and the
Communication Port of the PC to which it is connected must programmed to match.
Communication Port as Destination Port

COM or USB to COM Port can be assigned as Destination Port using Jeeves as well as by dialing SE commands
from a Telephone. For instructions, refer the individual topics:
• PMS Interface
• “Station Message Detail Recording-Report”
• “Station Message Detail Recording-Online”
• “Station Message Detail Recording-Posting”
• “System Activity Log”
• “System Fault Log”
• Hotel-Motel Activity Log
• Print Check-out Report
• “System Debug”
• “Loop Back Tests”
232. PMS can be interfaced on the LAN/WAN Port. For SMDR-Posting LAN/WAN Port can be used.

You may assign the destination port as Communication Port (COM Port or USB to COM) to multiple
facilities as mentioned above, but at a time only one facility will be served by each Communication Port.
The facility assigned first will be served first.
Communication Port Attributes

The functioning of the COM or USB to COM Port is controlled by the following attributes:
• Speed in bps.
• Number of data bits.
• Number of stop bits.
• Parity
These attributes must be programmed keeping in mind the application for which the communication port is used
(for instance, Programming through PC, generating SMDR Reports, etc.)
The Communication Port attributes can be changed using Jeeves and dialing SE commands from a telephone.
Changing the Communication Port attributes using Jeeves

• Under Configuration, click Communication Ports to open the page.
• Set the desired values for COM Port and USB to COM Port:
• Speed (Bps)
• Data Bits
• Parity
• Stop Bits

When the USB to COM converter is connected to the External USB Port, the same will be notified on this page.
Any one of the following USB to COM status will be displayed:
Status Description
This will be displayed when the USB to COM converter is connected,

Up
the port is available and functioning.
This will be displayed when the USB to COM converter is not

connected or not detected.
Down
Try re-plugging the device. The status will be updated accordingly.
This will be displayed when the USB to COM converter is connected

but the internal registration fails.
Fail
You may change any of the above parameter values or set the page to
default to re-initiate the registration process. The status will be updated
accordingly.
Changing Communication Port attributes using a Telephone
The SE Commands mentioned below are applicable only for COM Port and not for USB to COM Port.
To set Data Transfer rate for COM Port, dial:

• 3201-Port-Speed
Where,
Port is
1 for COM Port
Speed is
0 for 1200 bps
1 for 2400 bps
2 for 4800 bps
3 for 9600 bps
4 for 19200 bps
5 for 38400 bps
6 for 57600 bps
7 for 115200 bps
By default, Data Transfer Speed is 115200 bps.
To set Data Bits for COM Port, dial:

• 3202-Port-Data Bits
Where,
Port is
1 for COM Port
Data Bits are

0 for 7 data bits
1 for 8 data bits

By default, Data Bits are 8.
To set Parity for COM Port, dial:

• 3203-Port-Parity
Where,
Port is
1 for COM Port
Parity is
0 for None
1 for Odd
2 for Even
3 for Mark
4 for Space
By default, Parity is set as 'None'.
To set Stop Bits for COM Port, dial:

• 3204-Port-Stop Bits
Where,
Port is
1 for COM Port
Stop Bits are

0 for 1 stop bit
1 for 2 stop bit
By default, Stop Bits are 1.
To assign default parameters to COM Port, dial:

• 3210-Port
Where,
Port is
1 for COM Port
• Exit SE Mode.
Example 1: Configure COM Port parameters with these values:
• Speed = 9600 bps.

• Data Bits = 8 bits.
• Parity = None.
• Stop Bits = 1.
To do this, dial the following SE commands:
3201-1-3
3202-1-1
3203-1-0
3204-1-0
For connecting SARVAM UCS to a Computer/programming SARVAM UCS through a computer set the
Communication Port parameters to the following values (recommended):
Speed = 9600 bps.

Data Bits = 8.
Parity = None.
Stop Bits = 1.
How to use
For SARVAM UCS to communicate with a PC through the Communication Port, it must be connected with the
Communication Port of the PC.
Connecting SARVAM UCS with a Computer over COM/USB to COM Port

To connect the SARVAM UCS with a PC over COM or USB to COM Port, use the communication cable supplied by
Matrix233. This communication cable is provided with DB-9 female connectors on both ends.
You may connect any end to the SARVAM UCS and the other end to the PC. If the PC supports only USB
connectivity, use a USB-to-DB-9 converter of any standard make.
Refer the following Table for pin-out details of the COM Port.
Pin No. Signal Name
1 NC
2 Receive Data (RXD)
3 Transmit Data (TXD)
4 NC
5 Ground (GND)
6 NC
7 NC
8 NC
9 NC
233. This cable is supplied as an optional item. Contact your Matrix Dealer or the company to obtain this cable.

Conference-3 Party
What's this?
SARVAM UCS offers three types of conference calls: Conference-3 Party, “Conference Dial-In”, and “Conference-
Multiparty”.
Conference-3 Party (also referred to as Three-Way Calling) is a telephone call, in which the calling party can have
two other persons participate in the call.
A 3-Party Conference is initiated by dialing the number of the first person one wishes to talk to. The first person is
informed about the conference and put on Consultation Hold. The number of the second person one wishes to talk
to is dialed. When the second person answers, s/he is informed about the conference. Three-way speech is
established by pressing Flash-*3.
An already connected two-way speech can be converted into a conference by adding a second person, without
disconnecting the call with the first person.
Thus, a 3-Party Conference may be planned or conducted on the spur of the moment.
A 3-Party Conference can be conducted with extensions of SARVAM UCS and between extensions and external
numbers.
It is also possible to conduct an Unsupervised 3-Party Conference, wherein the operator (assistant) connects two
trunks through the system and withdraws from the three-way speech. For confidential discussions where the
parties need to know that the operator has withdrawn, the system provides facility to play beeps.
Beeps will be played:

• when the operator is present in the conference at regular intervals
• when the operator has left the conference
The maximum number of simultaneous 3-Party conferences supported by each model is mentioned in the table
below:
Max. number of simultaneous 3-Party

Model
conferences supported
ETERNITY LENX 15
ETERNITY MENX 15
ETERNITY GENX 20
How it works
A, B, C are extensions.
D and E are external numbers.
3-Party Conference between extensions

• A is in speech with B.

• A and B want to include C in their conversation.
• A presses the ‘Conference’ Key. B is put on Consultation Hold.
• A gets feature tone. B gets on-hold music.
• A dials C's extension number. A gets ring back tone.
• A is in speech with C. B cannot hear their conversation.
• A presses the ‘Conference’ key a three-way speech is established.
• A, B, and C are now in speech.
• Any of them can disconnect to withdraw from the conference.

• If C disconnects, A and B will be in two-way speech.
• A and B can carry on the conversation or can have a conference with another trunk (external number) or
with another extension.
3-Party Conference between two extensions and a trunk

• A and B want to include D in their conversation.
• A presses the ‘Conference’ Key. B is put on Consultation Hold.
• A gets feature tone. B gets on-hold music.
• A grabs a Trunk and dials D's extension number. A gets ring back tone.
• A is in speech with D. B cannot hear their conversation.
• A presses the ‘Conference’ Key to enable three-way speech.
• A, B, and D are now in speech.

• If B disconnects, A and D will be in two-way speech.
• A can now conduct a conference with another extension or trunk.
3-Party Conference between an extension and two trunks

• A is in speech with D.
• A and D want to include E in their conversation.
• A presses the ‘Conference’ Key. D is put on Consultation Hold.
• A gets feature tone. D gets on-hold music.
• A grabs a Trunk and dials E's number.
• A is in speech with E. D cannot hear their conversation.
• A presses the ‘Conference’ Key to enable three-way speech.
• A, D, and E are now in speech.

• If A disconnects, D and E are now in two-way speech.
3-Party Confidential Conference established by Operator
• F is in speech with D.
• F wants to include E in their conversation.
• F presses the ‘Conference’ Key. D is put on Consultation Hold.
• F gets feature tone. D gets on-hold music.
• F grabs a Trunk and dials E's number.
• F is in speech with E. D cannot hear their conversation.
• F presses the ‘Conference’ Key to enable three-way speech.
• F, D, and E are now in speech.
• D and E will hear Assistant Present beeps234 as long as F is present in the conference.

• For confidential conversation between D and E to be initiated, F must leave the conference.
• D and E will hear Assistant Leave beeps as soon as F leaves the conference.
• D and E are now in two-way speech.
• A, B and C are in speech. When A disconnects, either B and C are also disconnected or speech is
established between them depending on the option you select in If the Extension creating 3 party
conference, disconnects during Conference in the System Parameters.
• The Conference can be broken only by the master DKP/Extended IP Phone that has initiated the
Conference.
• If a call put on hold is to be included in a Conference, it must be retrieved first.
• If all the parties to the conference are SIP Extensions/Trunks and if the initiator of the Conference goes
on-Hook during the conference, the other parties will still remain in conversation. This is known as
Transfer on Conference Hangup.
How to configure
For this feature to work, the feature 'Conference' must be enabled in the Class of Service group of the extensions
that are to be allowed this feature.
For confidential conference to be established by Operator(Assistant), you must:

• Enable the Play beeps when Assistant present in Multiparty Conference check box. For instructions,
see “System Parameters”.
• Enable the Play beeps when Assistant leaves the Conference check box. For instructions, see “System
Parameters”.
• Configure the desired interval time in Conference – Assistant Present Beep Interval (sec). For
• Make sure you select Station Type as Assistant for the Operator Extension. For instructions, see
“Configuring SLT Extensions”, “Configuring DKP Extensions”, “Configuring Matrix SPARSH VP248”,
“Configuring Matrix SPARSH VP310”, “Configuring Matrix SPARSH VP330”, “Configuring Matrix SPARSH
VP510”, “Configuring Matrix VARTA ADR100/AMP100 UC Clients” and “Configuring Matrix VARTA
WIN200 UC Client”.
If extension users at remote locations are to be allowed to initiate the 3-party conference, “Direct Inward System
Access (DISA)” or “Auto Attendant” must be enabled on the trunk on which their call lands.
3-Party Conference in Class of Service

In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the extensions of the
SARVAM UCS. The Station Basic Feature Template 01 is assigned CoS group 01 which has 'Conference' enabled.
So, all extensions of the SARVAM UCS can make Conference calls.
If you want to deny 3-Party Conference to selected extensions, follow these steps:
1. Define a CoS group with 'Conference' disabled.
3. Assign this new Template to the extensions to which Conference is to be denied.
234. The total Beep Time = Beep Cadence + Wait Timer. The Wait Timer(non-programmable) is fixed as 2 seconds by the system.

programming.
The feature 'Conference' in the Class of Service also includes Dial-In and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed all three types of
conferences - 3-Party, Dial-In and Multi-party Conference.
How to use

• Speech with Party 1 on trunk/extension.
• Press ‘Conference’ Key. Party 1 put on Consultation Hold.
• Dial the number of Party 2.
If Party 2 is a trunk,
• Dial Trunk Access Code, to grab a trunk.
• You get Trunk dial tone.
• Dial telephone number of Party 2. You get ring back tone.
• Speech with Party 2.
• Press the 'Conference' Key.
• Three-way speech is established.
When the Conference is established, the Conference Key LED will glow continuously and you will get a
message “Conference” on your phone display.
For SLT Users

• Speech with Party 1 on trunk/extension.
• Dial Flash. You get Feature tone. Party 1 put on Consultation Hold.
• Dial the number of Party 2.
If Party 2 is a trunk,
• Dial Trunk Access Code to grab a trunk. You get Trunk dial tone.
• Dial telephone number of Party 2. You get ring back tone.
• Speech with Party 2.
• Dial Flash-*3.
• Three-way speech is established.

Conference-Multiparty
What’s this?
Like the Dial-In Conference, a Multi-party conference allows speech between more than three participants.
The key difference between Dial-In and Multi-party conference is that in a Dial-In conference participants can
include themselves in the conference by dialing into it without assistance, whereas in a Multi-party Conference the
party initiating the conference must include the participants by dialing their numbers.
In a 3-party Conference, when you add the fourth participant, a Multiparty Conference is initiated.
A Multiparty conference may be

• between extensions
• between extensions and trunks, that is, external numbers.
Any participant in a Multiparty Conference can Include a party, Remove a party, Leave a conference temporarily or
can Cancel a conference. When any participant is included in the conference, the system plays a beep to indicate
the inclusion.
A conference may also include the Operator(Assistant). For confidential discussions, where the parties need to
know that the operator has withdrawn, the system provides facility to play beeps.
Beeps235 will be played:
External callers can initiate multiparty conference using “Direct Inward System Access (DISA)”.
Depending on the model you are using, refer below table for the details regarding the multiparty conferences
supported.
Maximum simultaneous Maximum parties included

Maximum conference conferences (If all the in a conference (If all the
Model
participants conferences involve 3 parties are in a single
parties) conference)
ETERNITY LENX 45 15 21
ETERNITY MENX 45 15 21
ETERNITY GENX 62 20 21
How it works
A, B, C, and D are extension users.
E and F are external numbers.
A decides to hold a teleconference with B, C, D, E and F.
G is the Operator.
235. The total Beep Time = Beep Cadence + Wait Timer. The Wait Timer(non-programmable) is fixed as 2 seconds by the system.

Initiating a Multiparty Conference
• A has initiated a 3-party conference with B and C.
• If A dials the D’s number followed by the 3-party conference code. D is included in the conference.
• If A dials the G’s number followed by the 3-party conference code. G is included in the conference. The
system plays a beep to indicate the inclusion.
Now, the 3-party Conference is converted into a Multiparty Conference.

• The system will play beeps at regular intervals indicating that one of the participants present in the
conference is the Operator (Assistant).
After the Conference has been initiated, conference participants can:
• Include a Party in an on-going Multiparty Conference.
Internal as well as external callers can be included in an ongoing conference by the any of the participants.
To include external callers, in this case, E and F. A must dial the Trunk Access Code followed by E’s
number and the 3-party conference code, Similarly F can be included.
• Remove a Participant from an on-going Multiparty Conference.
Only DKP or Extended IP Phone users can remove another participant from the Multiparty Conference.
The DKP/Extended IP phone displays the numbers of all the participants, select the number of the
participant to be removed from this list.
If the last Operator (Assistant) in the conference leaves the group, a beep will be played.
• Temporarily Leave a Multiparty Conference.
Any participant in a conference can dial the Temporarily Leave Conference code to leave the conference
for a short time period.
If all participants, Temporarily Leave the conference one-by-one, the system will start the 'Release
Conference if idle for more than (Minutes) Timer'. This Timer is programmable, and by default it is set to
002 Minutes. If no participant rejoins the conference before the expiry of this Timer, the system will free the
resource occupied by the conference on the conferencing circuit.
• Rejoin a Multiparty Conference.
The participants who leave temporarily can rejoin the conference at a later stage by going Off-Hook and
dialing the Rejoin Conference conference code.
• Permanently Leave from a Multiparty Conference.
Participants in a conference can exclude themselves by going ON-Hook. Once any participant goes ON-
Hook, he/she cannot rejoin the conference.
• Cancel a Multiparty Conference.
Any participant in a conference can dial the Cancel conference code to end the conference.

All participants will get Error Tone and the system resource occupied by the conference will be freed.
Examples of Multi-party Conferences:

A Multiparty conference between extensions.
Simultaneous Multi Party conferences between extensions.

Multiple Multiparty Conferences between trunks and extensions.
Multiparty Conferences between one trunk and a few extensions.
Multiparty Conference between a few trunks and one extension.

Simultaneous Multiparty Conference between a few trunks and extensions and between extensions.
How to configure
• You must enable the feature 'Conference' in the“Class of Service (COS)” of the extensions in their “Station
Basic Feature Template”. By default, this feature is enabled on all extensions, so all extensions can use
this feature.
The feature 'Conference' in the Class of Service also includes 3-Party and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed Dial-In as well as 3-
Party and Multi-party Conference.
For confidential conferences, you must:

Parameters”.
• If desired, you may also change default value of the Release Conference if Idle for more than (min.)
Timer. See “System Timers and Counts”.
• If external parties are to be allowed to initiate or join the Conference, “Direct Inward System Access
(DISA)” must be enabled on the trunk on which they call.
• You can program a DSS key for Terminating a Conference, Temporarily Leave/Rejoining a Conference, if
required. Refer the topic “DSS Keys Programming” for instructions.

How to use

To initiate multiparty conference:
• Dial the number of party 1.

• When you are in speech with party 1, press ‘Conference’ Key.
• Party 1 is put on hold and gets on-hold music.
• You get feature tone. Dial number of party 2.
• When you are in speech with party 2, press the ‘Conference’ Key.
• Party 2 is included in the conference. A 3-way speech is established.
• Press the ‘Conference’ Key. Dial the number of party 3.
• When in speech with party 3, press the ‘Conference’ Key.
• A Multiparty Conference is initiated. Press the ‘Conference’ Key.
• The Multiparty Conference menu appears on the LCD.
• Select the option Include Party and dial the number.
• When you are in speech with the party, press the ‘Conference’ Key.
• Repeat the above steps to add new participants in the conference (max. 21).
To include a party in a multiparty conference:
• After the Multiparty Conference has been initiated. Press the ‘Conference’ Key.
• Select the option Include Party and dial the number.
• When you are in speech with the party, press the ‘Conference’ Key.
• Repeat the above steps to add new participants in the conference (max. 21).
To temporarily leave multiparty conference:
• While in Conference, press the ‘Conference’ Key.

• Select the option ‘Leave Temporary’ and press the Enter Key.
OR
• Press the DSS key assigned to Temporary Leave Conference/Rejoin Conference.
To rejoin multiparty conference:
• Press the ‘Conference’ Key.
OR
• Press the DSS key assigned to ‘Temporary Leave’ Conference/Rejoin Conference.
To remove a party from multiparty conference:

• Select the option ‘Remove Party from Conf’
• The LCD displays the numbers of all the participants.
• Select the number of the participant you want to remove and press the Enter Key.

• The selected participant is disconnected.
To permanently leave from the multiparty conference:
• While in Conference, go ON-Hook.
To cancel multiparty conference:

• Select the option ‘Terminate Conference’ and press the Enter Key.
OR
• Press the DSS key assigned to Terminate Conference.

• All the participants will get an Error Tone and the system resource occupied by the conference will be
freed.
For SLT Users

To initiate multiparty conference:
• Lift handset.
• Dial number of party 1.
• Speech with party 1.
• Dial Flash.
• Speech with party 2. Dial Flash-*3. A 3-way speech is established.
• Dial Flash.
• Speech with party 3. Dial Flash-*3. A multiparty conference is established.
To include a party in a multiparty conference:
• Lift handset.
• Speech with party 1.
• Dial Flash.
• Speech with party 2. Dial Flash-*3. A 3-way speech is established.
• Dial Flash.
• Speech with party 3. Dial Flash-*3.
• Repeat the steps to include the desired number of parties (max. 21).
To temporarily leave from the multiparty conference:
• While you are all in speech, dial Flash-191.
To rejoin the multiparty conference:
• Go OFF-Hook.
• Dial 191.

To permanently leave from the multiparty conference:
• Go ON-Hook.
To cancel multiparty conference:
SLT users cannot Remove any participant from the Multiparty Conference.
To use Multiparty Conference from DISA mode, you may see the instructions “Dial-In Conference using DISA”
under “Conference Dial-In”.

Conference Dial-In
What’s this?
Dial-In Conference is a multi-party conference held at a pre-defined time. Extension users can schedule a Dial-In
Conference and inform other participants to join in the conference at the scheduled time. Dial-In Conference can be
used to conduct client meetings or sales presentations, project meetings and updates, regular team meetings, and
to communicate with coworkers who operate in different locations. Thus, this feature helps to increase productivity
by saving time and cost of travel for out-of-office meetings.
A conference may also include the Operator (Assistant). For confidential discussions, where the parties need to
know that the operator has withdrawn, the system provides facility to play beeps.
Beeps236 will be played:
Depending on the model you are using, refer below table for the details regarding the Dial-In conferences
supported.
Maximum simultaneous Maximum parties included

Maximum conference conferences (If all the in a conference (If all the
Model
participants conferences involve 3 parties are in a single
parties) conference)
ETERNITY LENX 45 15 21
ETERNITY MENX 45 15 21
ETERNITY GENX 62 20 21
How it works
• A Dial-In Conference can be scheduled by dialing the access code for Dial-In conference followed by the
conference number and a password.
• The Conference Number can be: 01 to 20.
The Conference Number must correspond with the number of simultaneous Dial-In Conferences
supported by the ETERNITY GENX (01 to 20) and ETERNITY LENX/MENX (01 to 15). If a user dials a
conference number other than this, system will play an Error Tone.
• The Conference Password is a four digit number string. The default conference password is 1111 and
must be changed before using this feature.
To avoid unauthorized access, make sure the password is strong and is provided to the participants
only.
• All the other participants must be informed about the conference.
You can also have external callers join the conference by providing them the DISA login.
236. The total Beep Time = Beep Cadence + Wait Timer. The Wait Timer (non-programmable) is fixed as 2 seconds by the system.

Let us understand how Dial-In Conference works with the following example.
• Extension user A wants to schedule a Dial-In Conference at 4:30 p.m. with B, C, D, E and F.
• B and C are extension users. C is an extension user who has been provided a DISA login to access an
extension of SARVAM UCS.
• D, E and F are external parties.
• Any extension user can initiate the conference, in this case A initiates the conference.
Scheduling a Dial-In Conference

• A schedules a Dial-In Conference for 4.30 pm.
• A informs B, C, D, E and F about the conference and provides them conference number, for example, '1'
and the password, '4040'.
If C wants to schedule a conference, C must log into his extension from DISA mode.
Initiating a Dial-In Conference

• Any extension user can initiate the Dial-In conference by dialing the feature access code for Dial-In
conference followed by the two-digit conference number and the password, for instance: '01' and '4040'. In
this case, A initiates the conference.
After the Conference has been initiated, conference participants can:
• Join a Dial-In Conference.
In this, example, B can join the conference by dialing the feature access code to join the conference
followed by the two-digit conference number and password.
C can join the conference from the DISA mode.
When a new party joins the conference, the system plays beeps to the existing participants, to inform them
of the new inclusion. Beeps are programmable (default: enabled).
• Include a Party in an on-going Dial-In Conference.
Internal as well as external callers can be included in an ongoing conference by the any of the participants.
To include external callers, in this case D, E and F, A must dial the Trunk Access Code followed by the
their numbers.
• Remove a Participant from an on-going Dial-In Conference.
Only DKP or Extended IP Phone users can remove another participant from the Dial-In conference. The
DKP/Extended IP phone displays the numbers of all the participants, select the number of the participant t
to be removed from this list.
• Temporarily Leave a Dial-In Conference.
Any participant in a conference can dial the Temporarily Leave Conference code to leave the conference
for a short time period.

If all participants, Temporarily Leave the conference one-by-one, the system will start the 'Release
Conference if idle for more than (Minutes) Timer'. This Timer is programmable, and by default it is set to
002 Minutes. If no participant rejoins the conference before the expiry of this Timer, the system will free the
resource occupied by the conference on the conferencing circuit.
• Rejoin a Dial-In Conference.

The participants who leave temporarily can rejoin the conference at a later stage by going Off-Hook and
dialing the Rejoin Conference conference code.
• Permanently Leave from a Dial-In Conference.

Participants in a conference can exclude themselves by going ON-Hook. Once any participant goes ON-
Hook, he/she cannot rejoin the conference.
• Cancel a Dial-In Conference.
Any participant in a conference can dial the Cancel conference code to end the conference.
All participants will get Error Tone and the system resource occupied by the conference will be freed.
The conference can also be can canceled by logging into the SA mode.
How to configure
• You must enable the feature 'Conference' in the“Class of Service (COS)” of the extensions in their “Station
Basic Feature Template”. By default, this feature is enabled on all extensions, so all extensions can use
this feature.
The feature 'Conference' in the Class of Service also includes 3-Party and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed Dial-In as well as 3-
Party and Multi-party Conference.
• If you want beeps to be played when any one joins the conference, enable Play Beep when Raid/
Conference/Dial-in Conference begins. See “System Parameters”, for instructions.
For confidential conferences, you must:

Parameters”.
• If desired, you may also change default value of the Release Conference if Idle for more than (min.)
Timer. See “System Timers and Counts”.

• If external parties are to be allowed to initiate or join the Conference, “Direct Inward System Access
(DISA)” must be enabled on the trunk on which they call.
• You can program a DSS key for Dial-In Conference, Terminating a Conference, Temporarily Leave/
Rejoining a Conference, if required. Refer the topic “DSS Keys Programming” for instructions.
How to use

To Schedule a Dial-In Conference:
• Go OFF-Hook.
• Press the DSS key assigned for Dial-In Conference.
OR
• Dial *19.
• The Dial-In Conference menu appears on the LCD.
• Select the option ‘Schedule a Conf’ and press the Enter Key.
• Enter Conference Number on the prompt.
• Enter Conference Password on the prompt.
• You get confirmation tone and the message 'Conf <number> Scheduled' on your phone's display.
• Go ON-Hook.
• Call all participants and inform them of the time of the Dial-In conference, the Conference Number and
Password.
To join the Dial-In Conference:

• Go OFF-Hook.
• Press the DSS key assigned for Dial-In Conference.
OR
• Dial *19.
• The Dial-In Conference menu appears on the LCD.
• Select the option ‘Include in Schd Conf’ and press the Enter Key.
• Enter the Conference Number on the prompt.
• Enter the Conference Password on the prompt. If you enter the wrong password, you get the message
'Check Conf P/w' on your phone's display.
• Conference is initiated.
• You will hear speech if any other participant has joined it. You will hear silence if no other participant has
joined it.
• If you hear silence, wait for others to join in.
OR
• Include the other participants in the conference, if you initiated the conference.
To include a party in the Dial-In conference:

• After initiating the conference and while you are all in speech, press the ‘Conference’ Key.
• Select the option ‘Include Party in Conf’ and dial the desired number to be included in the conference. You
get Ring Back Tone.
• When you are in speech with the party, press the ‘Conference’ Key The party is included in the conference.
To remove a participant from the Dial-In conference:

• Select the option ‘Remove Party from Conf’.

• The numbers of the participants appear on your phone's display.
• Scroll to select the participant you want to remove.
• You get confirmation tone. The extension user is now excluded from the conference.
To temporarily leave Dial-In conference:

• Select the option ‘Leave Temporary’ and press the Enter Key.
OR
• Press the DSS key assigned to Temporary Leave Conference/Rejoin Conference.
To rejoin Dial-In conference:

• Press the ‘Conference’ Key.
OR
• Press the DSS key assigned to ‘Temporary Leave /Rejoin Conference’.
If you have configured a DSS key for Temporary Leave and you leave the Conference by pressing the
DSS key, to Rejoin the Conference press the DSS key again.
To permanently leave from the Dial-In conference:

While in Conference, go ON-Hook.
To cancel Dial-In conference:

OR
freed.
To cancel a Dial-In Conference from the System Administrator (SA) Mode:

• Enter SA mode from a DKP/SLT/Extended IP Phone.
• Dial 1072-026-Conference Number.
• You get confirmation, the conference is released.
You can also cancel a Dial-In Conference from System Administrator (SA) Mode using Jeeves. To do this,
• Open Jeeves.

• Click the Dial-In Conference - Cancel link.
• Enter the two-digit conference number (01 to 20) which you want to cancel in the Cancel Dial-In
Conference Number field.
• Click Submit button.
• Log out of Jeeves.
For SLT Users

Make sure you enter the two-digit Conference Number — 01 to 20.
To schedule a Dial-In conference:

• Dial *191-Conference Number-Conference Password.
• Inform all intended participants about the time, conference number and password.
To initiate the Dial-In conference:

• Talk, if you hear speech, or wait for others to join in.
OR
• Include any participant in the conference.
To join the Dial-In conference:

• Talk, if you hear speech, or wait for others to join in.
OR
• Include any participant in the conference.

To include a participant in Dial-In conference:
• After initiating the conference, or when in speech,
• Dial Flash.
• Dial the number of the desired party.
• Speech with party, dial Flash-*3.
To temporarily leave from the Dial-In conference:

To rejoin the Dial-In conference:

• Go OFF-Hook.
• Dial 191.
To permanently leave from the Dial-In conference:

• Go ON-Hook.
To cancel a Dial-In conference:

SLT users cannot Remove any participant from the Dial-In Conference.
Dial-In Conference using DISA

To schedule a Dial-In Conference:
• Dial a DISA enabled Trunk
• Dial DISA Login Code 1079-Extension Number-User Password, if PIN Authentication is required.
• After DISA Login beeps, dial *191-Conference Number-Conference Password.
To initiate or join a Dial-In Conference:

• After DISA Login beeps, dial *192-Conference Number-Conference Password.
To go ON-Hook during Conference:

• Dial #0
To go OFF-Hook during Conference:

• Dial #1
To include a party midway of the Conference:

• Dial #0 to go ON-Hook.
• Dial #1 to go OFF-Hook.
• Dial the extension number to be included in the conference. You get Ring Back Tone.
• When you are in speech with the extension user, dial #2.
• You get feature tone, and the extension user is played music on hold.
• Dial Flash-*3. Both of you are now included in the conference.
To permanently leave the Conference:

• Dial #0 to go ON-Hook.
• Dial #0#9 to end DISA session.

When you enter DISA mode, you get beeps, dial digits before the DISA Inactivity Timer elapses.
Never dial 'Flash' when in DISA mode, you will get disconnected.
Keep dialing any digit to continue the conference.
See “Direct Inward System Access (DISA)” to know more.

Conflict Dialing
What's this?
You may recall that “Access Codes” are dialed at different call phases. No two Access Codes must be the same in
the same call phase.
For example, the same access code cannot be used for two different features like Call Forward and Redial, since
both these features are invoked in the 'Dial' phase. Similarly, Station (Extension) and Logical Group Codes too
must be unique and should not match with any of the features invoked in the 'Dial' phase.
However, SARVAM UCS allows overlaps within Feature Codes and “Flexible Numbers” (Station Codes). One
Feature Access Code can be a part of (subset) another code, for example, 4, 41, 412; Flexible Numbers of
extensions can be 201, 2011 etc.
So, when such overlapping access codes are dialed, the system matches the first digit. On finding more than one
Access code starting with the same digit, the system will not know how to interpret the instruction and act
accordingly.
Conflict Dialing feature resolves this confusion. When an access code that is a subset of any other access code is
dialed, the system waits for some time for the extension user to dial the next digit. If the user does not dial any digit
within that time, the system interprets it as the smaller Access Code, and invokes the associated feature.
The time for which the system waits for the next digit to be dialed before resolving the Access Codes is called
"Conflict Dialing Timer". This timer is set to 2 seconds and is programmable.
Refer the topics “Access Codes” to know more.
How it works
You may set,
• The Access code of Call Pick Up as '4'.

• The Access code for Alarms as '41'.
• The Access code for Department Group 01 as '412'.
• Extension user A dials '4'.

• The system finds three access codes starting with '4' (4, 41, 412).
• So, it waits for 2 seconds, which is the default duration of the Conflict Dialing Timer, for the next digit to be
dialed.
• If A does not dial any other digit before the Timer expires, the system interprets the code as '4' and invokes
Call Pick-Up.
• If A dials '1' before the Timer elapses,

• The system interprets it as '41'.
• The system detects another access code starting with '41'.
• So it waits for 2 seconds again for the next digit to be dialed.
• If A does not dial any other digit before the Timer elapses, the system interprets the code as '41' and
invokes the Alarm feature.

• If A dials '2' before the Timer elapses,
• The system interprets the code as '412' and invokes Department Call to Group 01, provided there are
no other access codes like 4121, 4123, etc.
• If such access codes exist, the system again waits for the duration of the Conflict Dialing Timer for another
digit to be dialed.
• Thus, only when the conflict in the access codes is resolved will the system respond accordingly.
How to configure
The working of this feature is controlled by the Conflict Dialing Timer, which is set by default to 2 seconds and can
be changed as desired.
If the duration of the Conflict Dialing Timer is long, it may cause delay in the system's response to the
feature. If the duration is less, the system may misinterpret the access codes. Ensure that the value of the
Timer is programmed optimally (that is, at least the default value).
Changing Conflict Dialing Wait Timer using Jeeves

• Scroll to Other Features and set the time in seconds as desired for the Conflict Dialing Timer.
• Log out of Jeeves or continue programming further.

Changing Conflict Dialing Wait Timer using a Telephone

Where,
• Exit SE mode.

Conversation Recording
What's this?
Conversation Recording allows extension users to record their talk with other extension users or external parties,
after or without informing the opposite party.
This feature can be used to record verbal agreements, important discussions, instructions, interviews, client
requirements, take or place orders, etc.
Extensions must have a mailbox assigned to them for recording conversations. So, a VMS Module must be
installed on the CPU Card of the system for this feature to work.
SARVAM UCS supports recording of 20 calls simultaneously.
• Use this feature in accordance with the local laws.
On SIP extensions, SARVAM UCS supports Conversation Recording using INFO Message. For a list of IP
phones on which this feature has been tested, see “SARVAM UCS Features tested on IP Phones of
It is not possible to pause the Conversation Recording in SIP Extension. So, when SIP Extension puts the
call on hold and then retrieves the call then for the hold duration, silence will be recorded in the recorded
file.
How it works
A and B are extensions. Both are assigned a mailbox each.
C and D are external parties.
• A calls C.
• C answers the call.
• A presses the Transfer Key.
• C is put on Consultation Hold.
• A dials the command for Conversation Recording.
• The system sends a string of digits to the Voice Mail System to initiate Conversation Recording.
• A and C are in speech again.
• The conversation recording starts in A's mailbox. The system plays beeps, if Conversation Recording
Beeps are enabled.
• A or C disconnects the call.
• Conversation recording ends.
• A can listen to the recorded conversation by invoking the Voice mail feature.
The same is repeated when B calls A. As both have mailboxes assigned, both can record the conversation.

How to configure
The functioning of this feature is controlled by three parameters: 'Class of Service', 'Mailbox', and 'Conversation
Recording Beeps'. These parameters can be programmed using Jeeves or by dialing SE Commands from a
telephone.
Conversation Recording in Class of Service

Conversation Recording must be enabled in the COS group of the extensions to which this feature is to be allowed.
In the default Station Basic Feature Template Number 01 is assigned to all the extensions of SARVAM UCS, CoS
group 01 is assigned as default. Conversation Recording is disabled in CoS group 01. So, none of the extensions
of the SARVAM UCS can record conversations.
Decide which extensions should be allowed Conversation Recording.
If you want to allow this feature to all extensions, simply enable Conversation Recording in the default CoS group
01.
If you want to allow Conversation Recording to selected extensions,
1. Define a CoS group with Conversation Recording enabled.
3. Assign this new Template to the extensions to which Conversation Recording is to be allowed.
programming.
Mailbox
Extensions that are to be allowed Conversation Recording must also have a mailbox. Refer “Configuring SLT
Parameters using a Telephone”, “Configuring DKP Extensions using a Telephone”, “Configuring ISDN Terminals
using a Telephone”, “Viewing SIP Extension Status” for more information and programming instructions.
Conversation Recording Beeps

Decide whether you want Beeps to be played during Conversation Recording. Follow the instructions given below.
Programming Conversation Recording Beeps using Jeeves


• Go to Play Beep when Call Taping/Conversation Recording starts. Click the check box to enable or
clear the check box to disable this feature.
• Log out of Jeeves or continue with further programming.
Programming Conversation Recording Beeps using a Telephone

• Dial command 5332-Code.

Where,
Code is
0 for Disable
1 for Enable
• Exit SE mode.
How to use

To record a conversation:
• You are in speech with another extension/external number.

• Press DSS Key assigned to Conversation Recording (if programmed).
OR
Press Transfer Key and Dial 1095.
• You get beeps (if enabled).

• Speech with party re-established.
• Recording starts.
• Go ON-Hook, after conversation ends.
To listen to a recorded conversation:
• Press ‘Voice Mail’ Key
OR
Dial 3931237
• Follow Voice Mail Prompts.

Conversations are recorded as New Messages. So, follow the voice mail prompts for listening to new
messages.
For SLT Users

To record a conversation:
• You are in speech with another extension/external number.

• Dial Flash-1095
• You get beeps (if enabled).
• Speech with party reestablished.
• Recording starts.
• Replace handset when conversation ends.
To listen to a recorded conversation:
• Lift the handset

• Dial 3931
• Follow Voice Mail Prompts.
237. This is the default Voice Mail Feature Access Code. Verify with you System Engineer if this has been changed and use the new
code.

Customer Name
What's this?
Customer Name is the name of the organization/enterprise that has deployed SARVAM UCS. As the User, you can
enter the name of your company/organization in the system.
When Customer Name is assigned in the system, this name will appear as header on the various System Reports
generated and printed by the SARVAM UCS like SMDR Incoming, Outgoing and Internal Call Reports, T1E1PRI
Performance reports, Alarm Status reports, etc.
The Customer Name may consist of a maximum of 80 alphanumeric characters, including punctuation marks. So,
you can enter the organization's address along with the Customer Name.
How to configure
Customer Name can be programmed using Jeeves and dialing SE commands from a Telephone at the time of
installation, or any time thereafter. It can also be corrected or changed any time.
Programming Customer Name using Jeeves

• Enter the name (and address, if desired) of the organization/enterprise in the field Customer Name. For
example: Prudent Investment, 701 Sunshine Boulevard, Bannerghatta, Bangalore.

Programming Customer Name using a Telephone
• Enter SE mode from a DKP.
To enter Customer Name, dial:

• 5401-Customer Name-#*
To clear the Customer Name, dial:

• 5401-#*
• Exit SE mode
• Use EON to assign Customer Name, as SLT does not support alphanumeric dialing.
• The method of entering Customer Name from EON is similar to typing text messages from the mobile
phone.

COSEC Integration
SARVAM UCS supports integration with COSEC for “Building Intercom” solutions, that is, few Systems with the
intercom application can be connected to each other using VoIP and these are also integrated with the Matrix
COSEC Door Controllers. However, COSEC Integration can also be used in VIP Apartments and Villa’s, wherein
ETERNITY LENX/MENX/GENX is installed.
With this integration the users — SLT, DKP and Extended IP— of SARVAM UCS can unlock the COSEC Door
Controller.
Matrix COSEC is an enterprise-grade people mobility management solution for organizations covering Time-
Attendance, Access Control, Visitor Management, Employee Self Service Portal, Roster Management, Contract
Workers Management and Cafeteria Management. To know more about Matrix COSEC, refer to COSEC
documentation or visit our website: www.MatrixComSec.com.
How it works
• Group together extensions of SARVAM UCS that need to access the same COSEC Door Controller.
Assign a Group ID to each group. You can create 50 groups. Each group must have a unique Group ID.
• In SARVAM UCS for integration a User Name and Password is configured. Make sure the same User
Name and Password is configured in each COSEC Door Controller you have installed. You can integrate
50 such Door Controllers with SARVAM UCS.
• Each COSEC Door Controller has a unique IP Address and port.
• Each Group ID is mapped to the IP Address and Port of the COSEC Door Controller. The extensions in the
same Group ID can only unlock the COSEC Door mapped with their Group ID.
• The extension users can open the door by dialing *7 (access code to unlock the COSEC Door,
programmable, see “Access Codes”)or by pressing the DSS key assigned to COSEC Door Open.
Open SIP Phone users can put a call on hold by dialing #2. These users can also unlock the COSEC Door
by dialing *7.
Let us understand how this feature works with the help of an example.
• ABC Limited has SARVAM UCS with Building Intercom Application installed, where there are 10
residential towers. Each tower has around 100 flats. Each flat has an SLT extension.
• The users in Tower 1 are grouped together, that is extensions from 200 to 300. They are assigned a Group
ID 01.
• Each tower has its own gate with COSEC Door Controller installed. At each gate there is an extension for
visitors.
• Group 01 is mapped to the COSEC Door Controller IP Address 192.168.101.200:8080
When there is a visitor at the main gate the security guides the visitor to the Tower 1. The visitor makes a call to the
flat owner, extension 205. After confirming the identity of the visitor, the tower door needs to be opened.

The flat owner puts the caller on hold and dial *7 (access code to unlock the door) or press the DSS key assigned
to COSEC Door Open.
The system checks the Group ID of the flat owners extension and the corresponding Door Controller mapped to it.
The system sends the request along with the User Name and Password to the COSEC Door Controller.
If within 10 seconds (Response Timer, not programmable) a success response is received from the COSEC Door,
the door is unlocked. If a response is not received or if failure response is received the user will get an error tone.
The COSEC Door access code can be dialed in Idle state also, that is, if the flat owner has received the
call on his/her mobile from the visitor and needs to open the door, the flat owner can lift the handset and
dial *7.
How to configure
• On a piece of paper make four columns and enter the details as given below (taking the above example
further):
Extension COSEC Door they COSEC Door IP

User Name Group ID
Number need to access Address and Port
James William 200 01 Tower 1 192.168.101.200:8080
Jessica James 201 01 Tower 1 192.168.101.200:8080
Kelly 302 02 Tower 2 192.168.101.205:8080
. . . .
. . . .
. . . .
. . . .
William 505 05 Tower 5 192.168.101.250:8080
You can create 50 groups. Assign each group a Group ID. Range of the Group ID is from 00 to 50.
• Configure the COSEC Integration table.
• Assign each user the Group ID.
To configure the COSEC Integration table,

• Under Configuration, click COSEC Integration to open the page.
• For COSEC Integration, enter the User Name and Password. Default User Name: admin, Password:
1234.
Make sure the same User Name and Password is configured in the COSEC Door Controllers you have
installed.
• Against each Group ID, enter the COSEC Door Controller IP Address and Port.
The same COSEC Door Controller IP Address and Port cannot be assigned to different Group ID’s.
• Click Submit.
To assign Group IDs to SLT users, see parameter COSEC Door Group in “Configuring SLT Extensions”.
To assign Group IDs to DKP users, see parameter COSEC Door Group in “Configuring DKP Extensions”.
To assign Group IDs to SIP users, see parameter COSEC Door Group in “Configuring Matrix SPARSH VP248”,
“Configuring Matrix SPARSH VP310”, “Configuring Matrix SPARSH VP330”, “Configuring Matrix VARTA ADR100/
AMP100 UC Clients”, “Configuring Matrix SPARSH VP510”, “Configuring Matrix VARTA WIN200 UC Client” and
“Configuring Open SIP Phones”
To assign a DSS key to COSEC Door Open, see “DSS Keys Programming”.

Day Night Mode
What's this?
Certain features of the SARVAM UCS like Operator, Class of Service, Toll Control, Outgoing Trunk Bundle Access
Groups, Trunk Landing Group, Auto Attendant, Direct Inward System Access (DISA), Security Alarms, etc, require
extensions and trunks to behave differently according to the working hours, non-working hours and break hours,
which are referred to as Time Zones.
These Time Zone-dependant features and facilities are operated automatically according to the Time Tables
programmed in the system. In a Time Table, the Time Zones - Working Hours, Non-Working Hours, Break Hours -
are defined for the entire week. Time Table is assigned to trunks, extensions and other time-zone dependant
features. The system executes the Time-Zone dependent features and facilities automatically according to the
Time Table.
To know more refer the topic “Time Tables”.
Day/Night Mode allows you to manually change the Time Zone of the system at any point in time, by issuing a
command. For example, the office is to be closed on account of an unplanned holiday or emergency. So, the Time
Zones of all extensions and trunks must be set to Non-working hours to route outgoing calls and land incoming
calls from/to the appropriate destination. You can set the SARVAM UCS to Night Mode until the office remains
closed and set it back to operate as per the Time Table, when work is resumed.
To cite another example, the office must work for extended hours. You can set the SARVAM UCS to Day Mode and
set it back to operate as per the Time Table.
When you set the system in Day/Night Mode, the system overrides the Time Tables assigned to Trunks,
Extensions and Operator. According to the mode you selected, it applies Working Hours/Non-Working Hours to run
all the Time-Zone dependent features of the system.
When the system is set to Day Mode, it applies Working Hours as the Time Zone for all extensions, trunks and time
zone dependant features and facilities. When the system is set to Night Mode, it applies Non-Working Hours as the
Time Zone on Time-Zone dependent features of the system.
Thus, Day/Night Mode forces the system to work in a particular Time Zone, until it is changed again, manually.
Day/Night mode can be set by the System Engineer (SE mode) as well as by the System Administrator (SA Mode).
It can be done using Jeeves or by dialing a command from a Telephone.
How to configure
Setting Day/Night Mode from SE mode using Jeeves


• Go to Day/ Night Mode. Select the desired option:
• Set System in Day Mode (Working Hrs)

• Set System in Night Mode (Non-Working Hrs)
• Set System in Break Hours Mode
• Operate System as per Timetable assignment (default).
• Log out of Jeeves or continue, as desired.
Setting Day/Night Mode from SA mode using Jeeves

• Log into Jeeves as System Administrator.
• Click Day/Night Mode to open the page.
• Select the desired option:
• Set System in Day Mode (Working Hrs)

• Set System in Night Mode (Non-Working Hrs)
• Set System in Break Hours Mode
• Operate System as per Timetable assignment (default).

Setting Day/Night Mode from SE mode using a Telephone

To set the system in Day/Night mode, dial:

• 4801-Code
Where,
Code is from 1 to 4.
1 is for Day Mode.
2 is for Night Mode.
3 is for Operate system as per Time Table.
4 is for Break Hours.
• Exit SE mode.
Setting Day/Night Mode from SA mode using a Telephone

• Enter SA mode from a DKP/SLT.
• Dial 1072-018-Code
Where,
Code is from 1 to 4
1 is for Day Mode
2 is for Night Mode
4 is for Break Hours
• Exit SA mode.
If you are setting Day/Night Mode from a DKP using a DSS key, refer the LED indication in table below.
LED Indication on DSS Key assigned to Day/Night Mode
Model Event Color Cadence
Day Mode Set Blue Continuous ON
EON48/EON310/ Night Mode Set Red Continuous ON

EON510 Break Hours Mode Violet Continuous ON
System set to work as per Time Table -- OFF

Daylight Saving Time (DST)
What's this?
Daylight Saving Time (DST) is the practice of advancing clocks so that afternoons have more daylight and
mornings have less. Typically clocks are adjusted forward one hour near the start of spring and are adjusted
backward in autumn.
Many countries of the world use DST, though the start and end dates of DST vary with location and year. Even
within countries, uniform DST may not be observed. For example the states of Arizona and Hawaii do not observe
DST. Certain countries may observe DST in certain years, for instance Guatemala, while in most countries of Asia
and Africa, and in certain countries of South America, DST is not observed at all.
When SARVAM UCS is installed in a country/region where DST is used, it is necessary to synchronize the Real
Time Clock of SARVAM UCS with the local time.
So, if you are installing SARVAM UCS in a country where DST is used, find out the DST convention currently in use
in that country, and adjust DST accordingly.
How it works
The forward and backward adjustment of clocks can be Scheduled or Manual.
• Scheduled DST Adjustment: The Real Time Clock of the SARVAM UCS is advanced and set backward
automatically according to the DST convention of the country/region where the SARVAM UCS is installed.
Scheduled DST Adjustment is useful in countries/regions where DST Time is fixed, such as in Europe,
USA and Canada, without yearly variations.
The table below gives describes the DST conventions followed in the different countries for which
SARVAM UCS will automatically adjust DST.
SARVAM UCS supports 18 DST Types for Scheduled DST Adjustment.
DST DST Timings

Typ Applicable in Countries
e Start Time End Time
01 Last Sun MAR Last Sun OCT Austria, Poland, Russia, Spain
From 01:59 to 03:00 From 02:59 to 02:00
02 Last Sun OCT Last Sun MAR Australia, Australia-Tasmania, Belgium,

From 01:59 to 03:00 From 02:59 to 02:00 France, Germany, Greece, Hungary, Italy,
Sweden, Switzerland
03 Second Sunday MAR First Sunday NOV Bahrain, Mexico, Turkey, United States
From 01:59 to 03:00 From 01:59 to 01:00
04 First Sun NOV Third Sun FEB Brazil

From 23:59 to 01:00 From 23:59 to 23:00

DST DST Timings
Typ Applicable in Countries
e Start Time End Time
05 Second Sunday MAR First Sunday NOV Canada

From 01:59 to 03:00 From 01:59 to 01:00
06 Second Sat OCT Second Sat MAR Chile

From 23:59 to 01:00 From 23:59 to 23:00
07 Last Sun MAR Last Sun OCT Denmark, Ireland, Portugal, United Kingdom
00:59 02:00 01:59 01:00
08 Last Sun MAR Last Sun OCT Finland

02:59 04:00 03:59 03:00
09 First APR First OCT Iraq

02:59 04:00 03:59 03:00
10 Last Sun MAR Last Sun OCT Kyrgyzstan

02:29 03:30 02:29 01:30
11 Last Fri APRIL Last Thu SEP Egypt

23:59 01:00 23:59 23:00
12 Last Sun MAR Last Sun OCT Lebanon

23:59 01:00 23:59 23:00
13 First Sun SEP First Sun APRIL Namibia

01:59 03:00 01:59 01:00
14 Last Sun SEP 01:59 First Sun APR 02:59 New Zealand
03:00 02:00
15 Last Sun MAR Last Sun OCT Norway

01:59 03:00 02:59 02:00
16 First Sun OCT First Sun APRIL Paraguay

23:59 01:00 23:59 23:00
17 First APRIL First OCT Syria

23:59 01:00 23:59 23:00
18 First APRIL Last Sun OCT Cuba

23:59 01:00 23:59 23:00
The DST Type is to be selected according to the country/region where the system is installed.
When DST Mode is set to 'Scheduled' and the DST Type is selected, the system will automatically adjust
DST at the preset dates and time for the country/region where the system is installed.
For example, if SARVAM UCS is installed Spain, the DST Type 01 applicable to this country should be
programmed as Scheduled DST. The system will automatically advance the clock on the last Sunday of
March at 01.59.03:00 am every year (the start date of DST) and set the clock backward on the last Sunday
of October at 02.59.02:00 am of the same year.
• Manual DST Adjustment: The Real Time Clock of the SARVAM UCS is advanced and set backward
manually according to the DST convention of the country/region where the SARVAM UCS is installed.

Manual DST Adjustment is to be used in regions/countries that have no fixed DST Convention and where
yearly variations in DST practices are likely.
When DST Mode is set as 'Manual', you must set the start and the end time, that is, the time at which the
clock is to be advanced and the time at which the clock is to be delayed.
There are two ways to adjust DST manually:
1. The 'Day of Month' method, which specifies a day of the month DST will start or end. For example:
starting on the 2nd Sunday of March and ending on 1st Sunday of November.
2. The 'Date and Month' method, which specifies a date of the month that DST will start or end. For
example: starting on March 11 and ending on November 4.
DST is not applicable in certain regions/countries, like Asia and South America. In such cases, the DST
Mode is to be 'Disabled'.
How to configure
Adjusting DST using Jeeves

• Under Configuration, click Date and Time.
• Click Day Light Saving Time to expand.
• Set the DST Mode to Manual or Scheduled as per your requirement.
Scheduled DST Adjustment

• If you have selected Scheduled as DST mode, in the Region list, select the name of the country/region
where your system is installed.
• Click Submit at the bottom of the page to save your DST setting.
• If you do not find your region on this list, you are recommended to set DST Mode to 'Manual' and adjust
DST manually.
Manual DST Adjustment

• If you have selected Manual as DST Mode, set the Forward and Backward Time Adjustments.
• Go to the option Forward Time Adjustment to advance the time when DST starts.
• To Forward Time, select the desired option:

• Day-Month Wise to specify the day of the month DST will start.
OR
• Date-Month Wise to specify the date of the month DST will start.
If you select 'Day-Month Wise' option, the 'Date-Month Wise' option will be disabled, and vice versa.
Day-Month Wise
• If you select the 'Day-Month Wise' option, you should now select the desired options in each of the
following:
• Ordinal number: Select the Ordinal number of the day of the month, that is, the 1st, 2nd, 3rd, 4th, 5th
day, when DST begins.
• Day: Select the day of the month - Sunday, Monday, Tuesday, Wednesday, Thursday, Friday, Saturday
- when DST begins.
• Month: Select the month when DST beings (January-December).
• Change Time From: Select the time when DST will begin to change. The time mode is 24 hours, with
options from 00 to 23 hours and 00 to 59 minutes.
• To: Select the time to which the DST is advanced. The time mode is 24 hours, with options from 00 to
23 hours and 00 to 59 minutes.
Date-Month Wise
• If you select 'Date-Month Wise' option, you should now select the desired options in each of the following:
• Date: The date on which DST begins (1-31).
• Month: The name of the month when DST begins (January-December).
• Change Time From: The time when DST will begin to change. The time mode is 24 hours, with
options from 00 to 23 hours and 00 to 59 minutes.
• To: The time to which the DST is advanced. The time mode is 24 hours, with options from 00 to 23
hours and 00 to 59 minutes.

• Now, go to the option Backward Time Adjustments to set the time back (that is, end DST and begin
standard time).
• Follow the same steps described above (step no. 4 to 6) to set the day/date, month, hours and minutes
except, here you must set these parameters according to the time when DST ends.
• Click Submit at the bottom of the page to save your DST settings.
• SARVAM UCS gives you the flexibility to set the 'Forward DST Adjustment' according to Date-Month,
while the Backward DST Adjustment according to Day-Month. Similarly, the reverse is also possible,
that is, Forward DST may be set according to Day-Month, while the Backward DST may be set as
Date-Month. This is flexibility is particularly useful for setting DST of countries where the start of DST is
defined by date and month, like the First of April, but the end of DST is defined by Day and Month, such
as the last Sunday of October (as observed in Cuba).
• When the DST of a particular country starts or ends on the Last Sunday or any other day, for example,
the last Tuesday, last Friday of the month, always set the Ordinal Number as '5th'.
• Wherever time adjustments are made at 00:00 hours, use the previous date and set DST start time
(that is, "from" time) at 23:59 hrs.
• If you synchronize the RTC with the SNTP Server and the Date and Time changes, the DST will be
applicable as per the new Date and Time.

Adjusting DST using a Telephone
• Enter SE mode.
To set the DST Mode, dial:

• 1010-DST Mode
Where,
DST Mode is
0 for Disabled
1 for Manual
2 for Scheduled
If DST Mode is selected as 'Manual'.
To set the start time of DST, dial:

• 1011-Date-Month-Current Time-Advance Time
Where,
Date is from 01 to 31. Use leading zero in case of single digit date (Default 01).
Month is from 01 to 12. Use leading zero in case of single digit month (Default 01).
Current Time238 is in HH:MM format,

HH from 00 to 23 (use leading zero)
MM from 00 to 59 (use leading zero) (Default 00:00).
Advance Time239 is in HH:MM format,

HH is from 00 to 23 (use leading zero)
MM from 00 to 59 (use leading zero) (Default 00:00).
Please note that the advance time will be greater than the current time.
To set the end time of DST, dial:

• 1012-Date-Month-Current Time-Delay Time
Where,
Date is from 01 to 31 (use leading zero for single digit date) (Default 01).
Month is from 01 to 12 (use leading zero for single digit month) (Default 01).
Current Time is in HH:MM format (use leading zero),

HH is from 00 to 23
MM is from 00 to 59 (Default 00:00).
Delay Time240 is in HH:MM format (use leading zero must)
HH is from 00 to 23
MM is from 00 to 59 (Default 00:00).
Please note that the delay time will be less than the current time.
If DST Mode is selected as 'Scheduled'
To select the DST Type, dial:

• 1013-DST Type
238. The current time is the time that is presently followed by the system.
239. This is the time to which the Real Time Clock should be advanced.
240. This is the time to which the Real Time Clock should be set back to.

Where,
DST
Applicable in Countries
Type
01 Austria, Poland, Russia, Spain
02 Australia, Australia-Tasmania, Belgium, Cote d’Ivoire, France, Germany, Greece,

Hungary, Italy, Sweden, Switzerland
03 Bahrain, Mexico, Turkey, United States
04 Brazil
05 Canada
06 Chile
07 Denmark, Ireland, Portugal, United Kingdom
08 Finland
09 Iraq
10 Kyrgyzstan
11 Egypt
12 Lebanon
13 Namibia
14 New Zealand
15 Norway
16 Paraguay
17 Syria
18 Cuba
For example: to select DST Type for Brazil, dial: 1013-04.
• Exit SE mode.

Department Call
What's this?
Department Call enables you to group together extensions of a particular department so that callers can reach
anyone in the department by dialing a common access code assigned to the department.
Calls made to such groups of extensions are called Department Calls and the access code used to make
department calls is called Department Number.
This feature is useful in situations where any member of a department may interact with callers, as for instance in a
information counter, a customer care cell, a technical support team, etc.
Callers can also reach individual extensions in a Department group by dialing the extension number.
SARVAM UCS supports the formation of 16 department groups. The member extensions of a department group
may be single line telephones (SLT), digital key phones (DKP), SIP Extensions, ISDN Terminals or Virtual
Extensions.
Each Department Group can also be assigned a mailbox for voice mail, which any member extension can access.
Each Department Group can forward its calls to an extension or to its voice mail, or to another Department Group.
How it works
Extensions A, B, C, D are grouped as a Department with the access code 3901.
Internal Calls
• Extension E dials 3901 to call the Department.
• The system checks E's Class of Service for the Department Call feature.
• The feature is enabled. The system checks if Rotation is enabled in the routing group assigned to the
Department.
• The Rotation flag is enabled. The system lands the call on the extension which is set to ring first.
• Extension A, configured as the first landing destination rings for the duration of the Ring Timer
(configurable; default: 15 seconds).
• A answers the call. Speech established between A and E.
• If A does not answer, the system hunts for the next extension in the group to land the call, say B.
• B starts ringing for the duration of the Ring Timer.
• If Continuous Ring is enabled on A, A will continue to ring even as B is ringing.
• If B does not answer the call at the end of the timer, the system hunts for the next extension, C.
• If B has Continuous Ring enabled, B will continue to ring even as C is ringing.
• If the call is not answered even after hunting the last extension, the system will loop back and start from
the first extension once again.

External Calls
Department Calls can be made using Auto Attendant - Built-In Auto Attendant or VMS Auto Attendant. For
example, a company may use the Built-In Auto Attendant to have callers who want information only to dial the
Information Department instead of waiting for the Operator.
• An external caller places a call to Department 3901 using the Built-In Auto Attendant.
• The system checks if Rotation is enabled in the routing group assigned to the Department.
• As the Rotation flag is enabled, and the first call was landed on A, the system lands the call on the next
extension B.
• Extension B rings for the duration of the Ring Timer (configurable; default: 15 seconds). If the Continuous
Ring flag is enabled for B, it will continue to ring, even as the system hunts for another extension in the
group to land the call.
• A third call internal/external made to Department 3901.

• The same process as described above will be repeated.
• But the system will land the call on extension C first, because Rotation flag is enabled on this routing
group.
• The subsequent incoming calls will land on the extension which is next to the one that received the last
call. So the next call to the Department will land on extension D, the one thereafter on A, and so forth.
Thus for each call, the system will hunt for a landing extension as per the Rotation set for the routing group. The
extensions will ring for the duration of the Ring Timer, either continuously or one-by-one (as per the Continuous
Flag configured), and according to the sequence in which the extensions in the group are arranged.
Rotation ensures equal distribution of call traffic. If Rotation is disabled, the fresh call will always land on first
extension of the Department group.
Department Group and ISDN Terminal: When more than two ISDN terminals connected to the same BRI
port are configured as members of a Department group, if a call is made to this group using the department
group access code, only two ISDN terminals connected to the BRI port will ring. This limitation is because
of the BRI protocol.
Voice Mail
A Department Group can be assigned a common mailbox for Voice Mail, called the Department Group Mailbox. For
this a VMS module must be installed in the system. This common mailbox for the group is called Department Group
Mailbox. You can assign Department Group Mailbox to selected extensions or to all extensions in the Department.
To take the example of Extensions A, B, C, D with the Department Access Code 3901 further,
• Extensions A, B, C and D are all members of Department Group 1 with the Access Code 3901.
• Department Group Mailbox is assigned to all the four extensions.
• When there is a new message in the Group Mailbox, all four extensions - A, B, C, D - will get the Message
Wait Notification.
• The message wait indication may be a Stuttered Dial Tone or a Voice Message when the extension user
goes OFF-Hook, or blinking of the LED Lamp on the extension, or a Ring.241
241. This will depend on the type of Message Wait Indication configured for the Extension’s Voice Mail Settings.

• To the first extension that answers the notification call, for example, Extension A, the Voice Mail System
informs about the new message(s) waiting in the Department Group Mailbox and in the Personal Mailbox.
"You have <x> new Message in your Personal Mail Box. You have <y> new Messages in your Department
Group Mail Box."
• If there is no new message in both mailboxes, the VMS will play the message: "You have Zero new
Message."
• If there is a new message in the Department Group Mailbox, but none in the Personal Mailbox, the
VMS will play the message:"You have <x> new Message in your Department Group Mailbox."
• If there is no new message in the Department Group Mailbox, but new message in the Personal
Mailbox, the VMS will play the message: "You have <x> new Message in your Personal Mailbox."
• The VMS prompts Extension A to access the Group mailbox: "To go to Personal Mailbox, press 1. To go to
Department Group Mailbox, press 2."
• The user of Extension A presses 2, and is taken to the Department Group Mailbox,
• VMS prompts A: "Enter your mailbox password". Enter your department group mailbox password.
• The VMS checks the utilized mailbox memory,
• if 80% of the mailbox memory has been consumed, the VMS prompts the caller: “Your Mailbox is 80%
Full. Please Delete few messages.”.
• if 100% of the mailbox memory has been consumed, the VMS prompts the caller: ”Your Mailbox is Full.
Please Delete few messages.”.
• VMS prompts: "To listen to new messages press '1', to listen to old messages press '2', to send a message
press '3', to change your mailbox settings press '4'."
• Extension A presses 1.
• VMS plays the new messages.
• After playing the new messages, the VMS cancels Message Wait Notification set for extensions B, C and
D.
Call Forward
Just as calls can be forwarded to a Department Group, a Department Group can also forward its calls to:
• an extension
• its own Department Group Mailbox
• another Department Group
For Department Groups, SARVAM UCS does not support Call Forward to an external destination number.
You can set Call Forward for Department Group from the SA Mode only.
SARVAM UCS supports the following Call Forward options for Department Groups:
• Call Forward - unconditionally: calls are forwarded to the destination number, without checking the
status or waiting for a response from the Department Group.

• Call Forward- if Busy: calls are placed on the Department Group as per the Rotation configured for it and
are forwarded to the set destination, only when all the member extensions of the Department Group are
found to be busy.
• Call Forward- if No Reply: when a call is made to the Department group, SARVAM UCS will place the
call as per the Rotation configured for the Department Group for the duration of the 'Call Forward No Reply
Timer for Department’ (default: 30sec ). If none of the member extensions answers the call before the
expiry of this timer, the call is forwarded to the destination.
If you select this option, you may set the Call Forward No Reply Timer for Department to the desired value.
This Timer is commonly applied on all Department Groups which set Call Forward No Reply.
• Call Forward - if Busy/No Reply: calls made to the Department Group will be routed to the destination, if
all members of the Department Group are busy or when none of the member extensions answered the call
within the Call Forward No Reply Timer for the Department.
• Call Forward - Dual Ring is not supported for Department Groups.
• Member extensions of a Department Group can set Call Forward on their extensions. However, Call
Forward set for the Department Group will have precedence over Call Forward set by individual
member extensions.
• Call Forward set by member extensions in a routing group will be ignored by the system if, the Ignore
call forward set by member extension, when call is routed on Routing/Dept. Group flag is enabled. See
“System Parameters” for more information.
• Call Forward for a Department Group can also be set from SA Mode. See “Setting Call Forward for
Department Group”.
Again, taking the above example of Department Group 1 further, here’s how call forward will work:
• Extensions A, B, C, D of Department Group 1 are allowed Call Forward Department Group in their Class of
Service.
• Any of them can set Call Forward for Department Group 1. Extensions A, B, C and D can also set Call
Forward on their own extensions.
• When any extension or an external caller (also using Auto Attendant or Direct Inward System Access)
dials the Access Code 3901 to call Department Group1, SARVAM UCS will check the Call Forward option
set for the Department Group and route the call accordingly.
• If Call Forward - unconditionally is set, the call will be routed to the destination number, if the Ignore call
forward set by member extension, when call is routed on Routing/Dept. Group is enabled.
• If Call Forward - Busy is set, and the first extension in the Department Group is busy, the system will hunt
for the next free extension in the group. It will continue to hunt for a free extension. If all extensions in the
group are busy, the call will be forwarded to the destination number.
Call Forward unconditional, busy, or busy/No reply set by any member extension will not work.
• If Call Forward - No Reply is set, the system will start the Call Forward No Reply Timer Department Group
and place the call as per the Rotation set for the Department Group. If the call is not answered by any of
the extensions before the timer expires, the call will be forwarded to the destination number.

If a member extension that is offered the call has set Call Forward-Unconditional, and the Call Forward No
Reply Timer Department Group has not expired, the call forward set by the extension will be applied. If the
timer expires, the Call Forward No Reply set for the Department Group will be applied.
If a member extension that is offered the call has set Call Forward-No Reply, or No-Reply/Busy, the Call
Forward No Reply Timer (for individual extension) will start simultaneously with the Call Forward No-Reply
Timer Department Group. If the No Reply Timer for the extension expires first, the call will be forwarded to
the destination set for the extension. If the No Reply Timer of the Department Group expires first, before
the call is answered, the call will be forwarded to the destination set for the Department Group.
Call Forward-No Reply Timer can be set from the SE Mode only.
How to configure
The functioning of this feature requires you to do the following:
• create Department Groups.
• configure “Routing Group” (each routing group consisting of extensions related to a Department) and
assign the routing groups and appropriate access codes to the department groups.
If you want to provide voice mail facility to the Department Group, you must:
• assign a Mailbox to the Department Group.
• allow member extensions access to the Department Group Mailbox.
If you want to enable Call Forward to the Department Group, you must:
• enable 'Department Group Call Forward' in the Class of Service (CoS) of the member extensions.
• change, if required the default value of the Call Forward No Reply Timer for Department Group.
Creating Department Groups

• On a sheet of paper, draw a table.
• Decide the number of department groups you want to create, e.g.: 4 groups.
• Group all the extensions you want to put in each department group. You cannot group more than 16
extensions in a single department group.
• Decide in what sequence the extensions in each group should ring, that is, which extensions should ring
first, second, third, and so forth.
• Decide the access code you want to assign to each department group.
• You may also assign a name to the department group.
• Your table may look like this:
Access Extensions to be included as members

Department
Code
Group Name SIP ISDN
to be SLT DKP
Index Extension Terminal
assigned
2001, 2002, 3010, 3011,

1 3901 3301, 3302
2003 3012

Access Extensions to be included as members
Department
Code
Group Name SIP ISDN
to be SLT DKP
Index Extension Terminal
assigned
2015, 2020,
2 3902 3015, 3020 3305
2021
2023, 2024,
16 3916 3025 3315
2025
The access codes for the department groups and extensions in this table are default access codes.
• Now, with this information ready, you may configure the department groups using Jeeves or dialing the
relevant SE commands from a Telephone.
Enabling Call Forward for Department Call in Class of Service

To be able to set Call Forward for the Department Group, member extensions must have this feature enabled in
their COS.
In the default Station Basic Feature Template Number 01 is assigned to all the extensions of SARVAM UCS, the
default COS group 01 in the template has ‘Department Group Call Forward’ enabled. So, all extensions of
SARVAM UCS can set Call Forward Department Group.
If you wish to allow this feature to member extensions only, retain this feature ‘Department Group Call Forward’ in
the COS group of member extensions, and disable Call Forward feature in the COS of all the extensions. For this,
you may create separate templates for member extensions and other extensions.
Refer the topic “Class of Service (COS)” and “Station Basic Feature Template” for instructions.
Configuring Department Groups using Jeeves


• Under Configuration, click Department Groups.
Creating Department Groups

• To create a Department group, assign an Access Code for the department group against the Index
Number.
By default, the Access Codes assigned to Department groups are from 3901 to 3916.
If you decide not to use the default access codes, ensure that the access code you assign to each
department group is unique and does not match with any SLT, DKP, ISDN, SIP access code or any feature
access code of the Dial Phase. Refer the topic “Access Codes” to know more.
To assign Station Access Codes according to your preference and requirement to a range of Department
Groups, see “Assigning Access Codes to a Range of Extensions”.
• You may also assign a Name to the department group to facilitate identification. This name will appear in
the Dial by Name directory along with the department group number. The Name can be a maximum of 18
characters.
• Now, enter the Routing Group, that is, the number of the group you created for this department.
Where multiple departments exist, you must create separate routing groups for each department group.
This can be done in two ways:
• create the routing groups first and simply enter the relevant routing group number against the
Department Group Index (to which you have assigned the access code).
OR
• in Routing Group, click the Routing Group link. The Routing Group page opens. Now create the
Routing Group as per your requirement. For details, see “Routing Group”.

For example:
The Customer Care Department of a company has four extensions: 201, 202, 203 and 204 (on software ports 001,
002, 003, and 004 respectively), which needs to be grouped for Department Calls.
Extensions 201 and 202 are SLTs. Extensions 203 and 204 are DKPs.
Requirement: The company wants the following:

1. Call traffic should be distributed equally on all four extensions.
2. Ring sequence should be 201, 202, 203, and 204 (first to last).
3. First 201 should ring for 20 seconds.
4. If no reply, 201 should continue to ring and 202 should ring for 10 seconds.
5. If still no reply, 201 should continue to ring and 203 should ring for 15 seconds.
6. If still no reply, 201 should continue to ring and 204 should ring for 20 seconds.
7. 51 should be the access code for this department group.
Solution: Select a routing group e.g. 03, and configure as follows:

1. 201 as member 01, with member type SLT, and Port number 001.
2. 202 as member 02, with member type SLT, and Port number 002.
3. 203 as member 03, with member type DKP, and Port number 003
4. 204 as member 04, with member type DKP, and Port number 004.
Set ‘member type’ of members 5 to 32 in Routing Group 3 to 'None'.
1. Enable the 'Rotation Flag' on routing group number 03 to distribute call traffic.
2. Enable the 'Continuous Ring Flag' for member 01 (201) and set the 'Ring Timer' to '20 seconds.
3. Set the Ring Timer of member 02 (202) to 10 seconds. Disable 'Continuous Ring' flag.
4. Retain the Ring Timer of member 03 (203) as default 15 seconds. Disable 'Continuous Ring' flag.
5. Set the Ring Timer of member 04 (204) to 20 seconds. Disable 'Continuous Ring' flag.

6. Assign 51 as access code and routing group 03 to the Customer Care Department.
Remember to click Submit to save your settings.
Setting Call Forward for Department Group

You can set Call Forward for Department Group from SA Mode only.
• Click Department Group Properties.
• Click the desired Department Group Number tab for which you want to set Call Forward.
• Click Call Forward to expand.
• You have two options for Call Forward:

• To forward all department calls to Voice Mail, select the Forward Calls to Voice Mail and select the
Call Forward type from the combo box.
• To forward all department calls to a specific number, select Forward Calls-to Phone and enter an
extension number where the call is to be forwarded.
• Click the Apply Call Forward button.
The color of the text indicating that Call Forward is set will change to red.
• Click the Cancel Call Forward button.
The color of the text indicating that Call Forward is not set will change to black.
Setting Call Forward No Reply Timer for Department Group

If have enabled Call Forward Department Group, you may change, if required, the Call Forward No Reply Timer for
Department Group. You can change this timer only from SE Mode. By default the Timer is set to 30 seconds.

To change the Call Forward No Reply Timer,
• Under Configuration, click System Timers and Count.
• Scroll with the vertical bar to Call Forward No-Reply Timer for Department Group (sec).
• Set the timer to the desired value. The range of this timer is 1 to 255 seconds.
• Click Submit to save your timer setting.
Configuring Department Groups using a Telephone

To create a routing group with members, dial:

• 6502-1-Routing Group-Destination Index-Port Type-Port Number
Where,
Routing Group is the number of the Routing Group 01 to 96.
Destination Index is from 01 to 32.
Port Type is the 'Member type':
00 for None
01 for SLT
02 for DKP
28 for ISDN terminal
16 for OTBG
Port Number is the Software port number on which the member extension SLT, DKP, SIP Extension,
ISDN Terminal is attached.
Software port number of the SLT, from 001 to 240.
Software port number of the DKP, from 001 to 096.
Software port number of the ISDN Terminal, from 01 to 64.
Software port number of the SIP extension, from 001 to 999.

OGTBG Number is from 01 to 32.
E.g.: To include the four extensions in the above example in Routing Group 03, dial:
• 6502-1-03-01-01-001 to include extension 201.
To set the Ring Timer for each member extension in the routing group, dial:
• 6503-1-Routing Group-Destination Index-Ring Timer
Where,
Destination Index is number of the member extension in the routing group from 01 to 32.
Ring Timer is from 000 to 255 seconds. (Default: 015 seconds)
E.g.: To set the Ring Timer for the individual extensions of the Routing Group 03 in the above example,
dial:
• 6503-1-03-01-020 to make extension 201 ring for 20 seconds.

To set the Continuous Ring Flag for extensions in the routing group, dial:
• 6504-1-Routing Group-Destination Index-Flag
Where,
Destination Index is the number of the member extension in the routing group from 01 to 32.
Continuous Ring Flag is:

0 for disable continuous ring (each member extension in the group will ring for the duration of the 'Ring
Timer' for the group).
1 for enable continuous ring (the first extension in the group will ring till the call is answered).
E.g.: To enable/disable the Continuous Ring Flag for the individual extensions of the Routing Group 03
in the above example, dial:
• 6503-1-03-01-1 to enable the flag on extension 201.

• 6503-1-03-02-0 to disable the flag on extension 202.
To turn Rotation ON for the Routing Group, dial:

• 6505-1-Routing Group-Rotation Flag to enable rotation on a single routing group.
• 6505-2-Routing Group-Routing Group-Rotation Flag to enable rotation on a range of routing
groups.
• 6505-*-Rotation Flag to enable rotation on all routing groups.
Where,
Rotation Flag is:
0 for Disable (fresh call lands on the first extension within the group).
1 for Enable (fresh call lands on the extension next to the one that received the previous calls).

E.g.: To enable Rotation Flag for Routing Group 03 as in the above example, dial 6505-1-03-1.
To assign routing group to the department group, dial:

• 2001-1-Department Group Index-Routing Group
Where,
Department Group Index is from 01 to 16.
Routing Group is from 01 to 96. Default: 01.
E.g.: To assign Routing Group 03 to Department Group 01 (Customer Care) as in the above example,
dial 2001-1-01-03.
To clear the routing group assigned to department group, dial:

• 2001-1-Department Group Index-00 to clear the routing group of a single department group.
• 2001-2-Department Group Index-Department Group Index-00 to clear the routing groups of a range
of department groups.
• 2001-*-00 to clear the routing groups of all department groups.
To assign an access code (that is, Department Number) to a department group, dial:
• 3113-1-Department Group Index-Access Code
• 3113-2-Department Group Index-Department Group Index-Access Code
• 3113-*-Access Code
Where,
Department Group Index is from 01 to 16.
Access Code is a string of a maximum of 6 digits.
Terminate the command string with #* if the Access Code has fewer than 6 digits.
E.g.: To assign '51' as Access Code for the Department Group 01 as in the above example, dial 3113-
1-01-51-#*
To clear the access code assigned to a department group, dial:

• 3113-1-Department Group Index-#* to clear access code assigned to a single department group.
• 3113-2-Department Group Index-Department Group Index-#* to clear access codes assigned to a
range of department groups.
• 3113-*-#* to clear access codes of all department groups.
To default the access codes assigned to Department groups, dial:

• 3163-1-Department Group Index to default the access code of a single department group.
• 3163-2-Department Group Index-Department Group Index to default the access codes of a range of
department groups.
• 3163-* to default the access codes of all department groups.
• Exit SE mode.
How to use
Making a Department Call

Making a department call is the same as calling another extension.

• Press DSS Key assigned to the desired Department Group.
OR
• Dial the desired Department Group Number (default: 3901-3916).
• You get Ring Back Tone as the call lands on an extension within the department group.
• Talk when the call is answered.
For SLT Users

• Lift handset
• Dial the desired Department Group Number (default: 3901-3916).
• You get Ring Back Tone as the call lands on an extension within the department group.
• Talk when the call is answered.
Accessing Department Group Voice Mail

When LED of Voice Mail Key is turned on to indicate new message,
• Press Voice Mail key.

• VMS informs you about new message(s) in your Department Group mailbox and Personal Mailbox.
• Follow Voice Mail prompts to access Department Group mailbox.
• Press 2 to go to Department Group mailbox.
• Press 1 to listen to new messages.
• Go ON-Hook or follow voice prompts for the desired option.
For SLT Users

New messages in your mailbox will be notified according to the type of Message Wait Notification set on your
extension phone.
When the LED Lamp of the Message Waiting Key glows,

OR
When your phone plays Message Wait Ring242,
• Dial 3931.
• Follow voice prompts.
• Press 2 for Department Group mailbox.
• Replace handset after listening to messages. Or you may follow voice prompts.
If you hear Stuttered Dial Tone/Voice Message when you go OFF-Hook,

• Dial 3931.
• Follow voice prompts.
• Press 2 for Department Group mailbox.
• Replace handset after listening to messages or follow voice prompts to continue.
242. Refer “Distinctive Rings” for description.

Setting Call Forward for Department Group

• Press DSS Key assigned to the Call Forward Department Group.
OR
• Dial 1179 (users worldwide). Users in the Philippines, dial 1108.
• Enter the Department Group Number whose calls are to be forwarded.
• Scroll to select Call Forward Type from the following options:
• Cancel Call Forward
• Forward Unconditionally
• Forward when Busy
• Forward when No Reply
• Forward when Busy/No Reply
• Enter Destination number (extension number or voice mail)
• You get confirmation tone and message on your phone’s LCD.
• Go Idle.
For SLT Users

• Lift handset.
• Dial 1179 (users worldwide). Users in the Philippines, dial 1108.
• Dial the Department Group Number whose calls are to be forwarded.
• Dial Call Forward Type:
• 1 for Forward Unconditionally
• 2 for Forward when Busy
• 3 for Forward when No Reply
• 4 for Forward when Busy/No Reply
• 0 to Cancel Call Forward
• Dial Destination number (number of the desired extension or voice mail)

Dial By Name
What's this?
Dial By Name enables extension users to call another extension or an external party by dialing the name of the
person, instead of dialing their telephone number.
This feature is accessible only to users of the proprietary Digital Key Phones and the Extended IP phones of Matrix.
With Dial By Name users need not remember the desired party's telephone number or short codes, that is,
“Abbreviated Dialing” codes.
For each extension, the database for names used in Dial by Name is drawn from:
• the Personal Directory, which is assigned to each extension, wherein up to 25 external party numbers
along with their names may stored. The system uses the Personal Directory to dial external parties by their
names. See “Abbreviated Dialing” to know more.
• the Global Directory, which is assigned to the extension in its “Class of Service (COS)”. The Global
Directory is a system-wide list of external party numbers and names. Up to 999 numbers can be stored in
this directory, and parts of the Global Directory (Part 1, 2, 3) can be assigned to each extension in its Class
of Service. See “Abbreviated Dialing” to know more.
• Names of Extensions, which are names of users/departments groups. Their names are assigned to SLT,
DKP and SIP extensions to identify the extension users. Names of Extensions are necessary for making
internal calls using the Dial By Name feature.
How it works
• Extension user presses the DSS Key assigned to 'Dial By Name' feature.
• On EON48 models and on SPARSH VP248, press the 'Names' key. On EON310/SPARSH VP310, press
the ‘Contacts’ key.
• The prompt <Name: > appears on the phone display.
• User enters the name of the desired party243.
• For example, user wants to call Midas Biz, and enters the letter 'M' using the keypad.
• The system displays in alphabetical order, all names starting with 'M'. These numbers are drawn from the
Personal and Global Directories assigned to the extension and the Extension Names programmed in the
system.
• User scrolls the list using the Up/Down navigation keys to reach the desired contact's name.
OR
Instead of scrolling the entire list, the user enters more than one initial letter of the contact's name. The
search is narrowed down to more accurate matches. The phone displays the matching entries in the
directory.
• The user must select the desired name by pressing 'Enter' Key.
243. The process of entering the names is the same as when writing text messages (SMS) from a cell phone. The keys must be
pressed multiple times in quick succession to enter the desired alphabet.

• The system dials out the number stored under the selected name. The name and number are displayed on
the user's phone.
How to configure
For this feature to work, the following must be programmed:
1. DSS Key: A direct station selection (DSS) key must be programmed for the Dial by Name feature. Without
the DSS Key this feature will not be accessible.
The factory-default key map of EON48 and SPARSH VP248, phones have the DSS Key labeled as
'Names'. EON310/SPARSH VP310 have a fixed feature key ‘Contacts'
2. Global Directory: The names of the external parties must be programmed against their respective
telephone numbers in the directory. Refer the topic “Abbreviated Dialing” for instructions on programming
the Global Directories.
3. Personal Directory: The names of the external parties must be programmed against their respective
telephone numbers in the Personal Directory. Refer the topic “Abbreviated Dialing” for instructions on
programming the Personal Directories.
4. Extension Names: Extensions may be SLTs, DKPs, ISDN Terminals or SIP extensions. Refer the topics
related to configuration of extensions244.
5. Class of Service: Dial By Name is allowed to all DKP users. However, the use of this feature is related to
the following features, which must be enabled in the Class of Service of the DKP and SIP extension users:
• Internal Calls - This is a part of the Basic Features. By default these are enabled.
• Global Directory Part 1
• Global Directory Part 2
• Global Directory Part 3.
Global Directory Part 1 is assigned to the default CoS group 01 assigned to all extensions in the default
Station Basic Feature Template 01.
If you want the names to be drawn from Global Directory Part 2 and Part 3, provided these are
programmed, you must enable these two directories in the CoS of the DKP and SIP extensions.
Refer “Abbreviated Dialing”, for instructions on programming the Global Directory.
Refer “Class of Service (COS)” and “Station Basic Feature Template” for programming instructions on how
to enable a feature in the CoS and how to apply it on extensions.
The system will display the names exactly as they have been programmed in the Personal and Global
Directories and the SLT/DKP/ISDN Parameters.
244. You may also refer the instructions provided under the topic Configuring Extensions: “Configuring SLT Extensions”, “Configuring
DKP Extensions”, “Configuring ISDN Terminals” , “Configuring Matrix SPARSH VP330”, “Configuring Matrix SPARSH VP248” ,
“Configuring Matrix SPARSH VP310” , “Configuring Matrix SPARSH VP510”.

How to use
For EON48 and Extended IP Phone Users

• Press 'Names' key.
• You get the prompt: 'Name'.
• Enter the initial letters for the contact's name.
• Press 'Enter' key to select the name.
• The system displays the name and number being dialed out.
• You get Ring Back Tone or Busy Tone.
Entered the wrong alphabet?
• Go ON-Hook.
• Go OFF-Hook.
• Press the DSS Key labeled 'Names' again.
• Enter the name/initial letters of the contact's name.

Dialed Number Directory
What's this?
Dialed Number Directory feature is available only to the users of the proprietary DKP, EON and Extended IP
Phones.
It is the list of numbers dialed out from the phone, similar to the call history of recently dialed calls on a cell phone.
SARVAM UCS retains up to 16 numbers dialed out from a phone in a directory.
These numbers may have been dialed out using features like Abbreviated Dialing, Quick Dial, Redial, Walk-In
Class of Service, or may be a simple outgoing call made by directly dialing the external number.
How it works
• When a DKP/IP Phone extension user makes an outgoing call, the number is stored in the Redial Number
List.
By default the system stores only the external numbers in the Last Number Redial List. If you want the
system to store internal calls in this list, make sure you enable the Store Internal Calls in Redial Call Log
check box in the “System Parameters”.
• The list has a capacity of storing a maximum of 16 recently dialed numbers.
• The list is updated using the First-In First-Out logic, whereby the earliest dialed number is replaced with
the most recently dialed number.
• To use this feature, the phone user must invoke the “Last Number Redial” feature.
• Doing so, the Redial Number List will appear on the phone display.
• The user may now navigate the list, select the number to be dialed out.
• The system will dial out the selected number using the same Outgoing Trunk Bundle Group used to place
this call earlier.
• If the number had been dialed earlier using Abbreviated Dialing, the system will check for Toll Control
when dialing out the number again from the dialed number directory245.
How to configure
No specific programming required.
245. Recall that the system does not check for Toll Control when Abbreviated Dialing is used.

How to use

• Press the ‘Redial’ Key.
OR
Dial 7
• The list of last dialed calls appear on your phone's display.

• Scroll with up/down navigation key to reach the desired number.
• The desired number is dialed out and appears on your phone's display.
• You get ring back tone.
• Talk when speech is established.
• Go ON-Hook after the conversation has ended.

Dial Plan for SIP Extension
What's this?
SARVAM UCS supports 8 Dial Plans with total 32 entries in each table. The Dial Plan contains a series of digits
and/or wild card characters.
When a user dials a number, it is compared with the Rule configured in the Dial Plan. If a match is found, the IP
Phone routes the call immediately without waiting for End of Dialing and if a match is not found, the IP Phone will
wait for the End of Dialing and then route the call.
How to configure
Programming the Dial Plan involves the following steps:
• Selecting a Dial Plan and configuring the rules in the Dial Plan.
• Assigning the Dial Plan to the desired SIP Extensions.
Programming Dial Plan using Jeeves

• Under Configuration, click Dial Plan for SIP Extension.
• Click the desired Dial Plan number. The Dial Plan for SIP Extension - VP110/VP710 page opens.

• Against each Index configure the Rule according to which you want the system to process the call. You
can configure a maximum of 32 Rules in each Dial Plan.
For example, if you want that users should be able to dial extension numbers from 3000 to 3999 without
any delay, configure the Rule as 3XXX.
For more details to configure the rules, refer to the topic Dial Plan in the SPARSH VP110 User Guide.
• Click Submit.
Assigning Dial Plan to SIP Extensions

To assign the Dial Plan you configured for VP110,
• Click the location at which you have registered SPARSH VP110, for example, Location 1.
• Scroll to DSS Key Settings and Dial Plan.
• Select the Dial Plan number you configured.
• Click Submit.
To assign the Dial Plan you configured for VP710,

• Click the location at which you have registered SPARSH VP710, for example, Location 1.
• Scroll to Dial Plan.
• Select the Dial Plan number you configured.
• Click Submit.

Digest Authentication
What's this?
Digest Authentication is a challenge-based authentication service of SIP to authenticate the identity of the
originator of SIP request in the INVITE message. The recipient of the request can ascertain whether or not the
originator of the request is authorised to make the request. When the digest credentials of the originator—User
Name and Password—in the INVITE message are authenticated and accepted by the recipient, the originator and
the recipient are connected.
SARVAM UCS supports Digest Authentication. You may use Digest Authentication to
• restrict access to SARVAM UCS to specific callers.
• prevent unwanted or malicious calls.
How it works
The Digest Authentication feature works on the basis of the Digest Authentication Table, in which the credentials,
namely the User Name and Passwords of trusted/authorised calling party SIP devices are stored. You must
configure this table. The Digest Authentication Table is common for all SIP trunks on which this feature is enabled.
When you enable this feature on a SIP trunk, for all incoming calls (SIP requests),
• SARVAM UCS will challenge the identity of the calling party (the SIP device initiating the request) to send
its digest credentials.
• When the calling party sends its credentials, SARVAM UCS authenticates the credentials by matching it
with its Digest Authentication Table.
• If a match is found, the calling party will be authenticated and the call will be allowed on the SIP trunk.
• If no match is found, SARVAM UCS will consider it as invalid authentication information and reject the call.
How to configure
To use this feature on SIP Trunks, you must do the following:
• Enable Digest Authentication on the SIP trunks you want to use this feature.
• Configure the Digest Authentication Table.
You can configure the Digest Authentication Table using Jeeves and from a telephone.
Configuring Digest Authentication using Jeeves

• Under Configuration, click the VoIP Configuration link.
• Click Digest Authentication.

The Digest Authentication table opens.
• In the User ID field, enter the User ID to be authenticated. The User ID must be within 40 characters.
• In the User Password field, enter the corresponding Password. The Password must be within 16
characters.
To avoid unauthorized access, we recommend you to change the Password regularly. Make sure it is
• Now, enable Digest Authentication on the desired SIP trunks. For instructions, see “Configuring SIP
Trunks”.
Configuring Digest Authentication using a Telephone

To configure the User Password for the User ID, dial:

• 4119-Index-User Password
Where,
Index is from 01 to 99. The User Password should be configured at the same index as the User ID.
User Password may have a maximum for 16 characters. If the User Password has fewer than 16
characters, terminate the command string with #* if using an SLT or press the 'Enter' key if using a DKP.
Default: Blank.
• Exit SE mode.
For SE Command for enabling Digest Authentication on SIP trunks, see “Configuring SIP Trunks”.

Digital Key Phone-Operation
EON - the Proprietary Digital Key Phone

Matrix offers 'EON', the proprietary digital key phone (DKP). Matrix offers various models of EON. Each of these is
explained later. These are powerful stations, supporting a host of phone and System features, as listed below.
DKP Features
• Status of other ports (Tri-color LED indication)
• Programmable Direct Station Selection (DSS) Keys and Feature keys
• LCD notification messages
• Ringer Tune selection
• Adjustable Speech level
• Adjustable Ringer Volume
• Adjustable Backlight and Contrast levels
• Hands-free operation - Speaker key and headset connectivity.
• Call Logs - last 20 Missed, Answered and Dialed Calls.
• Operator, Executive, Hotel Attendant and Guest Functionality
• Message Paging
• Menu based operation of System features
• Multiple Language support.
System Features
Listed below are the features of SARVAM UCS that require a Digital Key Phone:
• Abbreviated Dialing
• Auto Answer
• Call Chaining
• Call Cost Display
• Call Duration Display
• Call Mute
• Dialed Number Directory
• Directory Dialing by Name
• Dynamic Lock
• Forced Answer
• Keypad Lock
• Live Call Screening
• Message Paging
• Off-Hook Alert
• Room Monitor
• Text Message Reply
• Time Zone Display
• User Status (Presence)

Models of EON at a Glance
Models
Feature
EON48S EON48P EON310 EON510
Total number of keys 48 48 42 49
Number of programmable keys 29 29 21 22
Capsense keys Yes Yes No No
Context Sensitive keys No No No Yes
Feature keys 12 12 9 8
DSS keys 16 16 12 16
2 lines x 24 6 lines x 24 2 lines x 24 240 x 64 Pixel

LCD display characters characters characters Graphical LCD
display
LCD with backlight Yes Yes Yes Yes
Headset Interface Yes Yes Yes Yes
Ringer Lamp (LED) Yes Yes Yes Yes
Speaker Phone Full duplex Full duplex Full duplex Full duplex
EON48
EON48S
2 lines and 24 characters LCD display, full duplex, capsense feature keys

EON48P
6 lines and 24 characters LCD display, full duplex, capsense feature keys.
LCD Display
The LCD display of EON48 is backlit and can be tilted at a convenient angle for a clear view of the text/characters
displayed.
The LCD backlight can be turned on and off as well as adjusted for contrast and brightness from the "Phone
Settings" of the DKP Phone Menu.
Ringer LED
The Ringer LED indicates incoming internal and external calls. The LED Cadence will match with the Ring
Cadence of the incoming internal/external call.
The Ringer LED cadence changes according to the type of call, as described in the table below.
Type of Call Cadence
Internal Call Short, very slow (750ms ON, 2250ms OFF)
External Call Double (400ms ON, 200ms OFF, 400ms ON, 2000ms OFF)
Alarm Long, fast (1500ms ON, 500ms OFF)
Auto Redial Call Long, very slow ( 2000ms ON, 4000ms OFF)
Auto Call Back Call Short, slow (750ms ON, 2250ms OFF)
Priority Triple (400ms ON, 200ms OFF, 400ms ON, 200ms OFF, 400ms ON, 2000ms OFF)
Programming mode Continuous

Navigation Keys
These are 5 capsense keys. The functions of each are described briefly below.
• Enter Key: To enter the Menu; when the phone is in the idle state (without any incoming or outgoing call
being made), if you tap the 'Enter' key, you will enter into the 'Menu'.
Enter key is also used to make a selection from the Menu/sub-menu options or to complete an action.
• Up Key: To scroll upwards while navigating the Menu/sub-menu.
• Down Key: To scroll downwards while navigating the Menu/sub-menu.
• Forward Key: To move forward when dialing a number.
• Back Key: To move backwards when dialing a number; to go back one level in the Menu.
Feature Keys
These are 12 capsense keys assigned to important or frequently accessed features of SARVAM UCS. Refer to the
table given below:
Sr.No. Description LED
1. Voice Mail Single Color - Blue
2. Call Back Single Color - Blue
3. Cancel No
4. Mute Single Color - Blue
5. Conference No
6. Transfer No
7. Forward Single Color - Blue
8. DND Single Color - Blue
9. Names No
10. Redial No
11. Release No
12. Hold No
These Feature keys are programmable. Refer the “DSS Keys Programming” topic for instructions on programming
these keys. You cannot change the labels of these keys and therefore it is recommended that you avoid
programming these keys.
A few of these Feature keys are equipped with an LED to indicate the status of the feature assigned to it. You may
re-assign other features to these keys. We recommend you to assign those features that require LED indication to
the Feature keys equipped with an LED.
Not all features require LED indication. For example, the feature Call Pick-Up does not require an LED. So when
you assign this feature to a feature key, the LED of the key remains inactive, when Call Pick-Up is accessed.

Feature like Auto Redial requires an LED to show that it has been set or canceled. So, the LED of the Feature key
to which the Auto Redial feature has been assigned will glow Blue, when Auto-Redial is set, and the LED is turned
off when the feature is canceled.
Direct Station Selection (DSS) Keys

These are 16 programmable keys that can be assigned to Stations and Trunks and important or frequently
accessed features of SARVAM UCS.
Refer the “DSS Keys Programming” topic for instructions on programming these keys.
DSS Key LEDs

Each DSS key is equipped with an LED which glows to indicate the status of the Trunk/Station or Feature assigned
to it.
• Status of Stations and Trunks: The LED of DSS keys assigned to Stations/Trunks glow in three colors to
indicate status of the call event on the Stations/Trunks and on the DKP.
Thus, the status of the DKP user's own Station as well as that of the other Stations and the status of Trunk
lines are indicated by the LED of the DSS keys assigned to those Stations and Trunks on the DKP.
The following table shows the relationship between the color of the LED and various events:
LED LED Mode

Color Continuously ON Slow Blink Fast Blink
Blue The key assigned to the The key assigned to the The key assigned to the
Station you are in speech Station you have kept on Station you are calling or from
with. hold. which you are being called.
Red The key assigned to the The key assigned to the The key assigned to the
Station that is now busy with Station which has put another Station/Trunk that is called or
another Station/Trunk. Station/Trunk on hold. being called by another.
Violet You are talking on a Trunk You have held a Trunk You have an incoming call on
(external call) (external call) the Trunk (external call)
• Blue indicates the state of the station/trunk you access. For example, when you make a call to another
Station 203, the LED of the DSS key assigned to Station 203 blinks Blue to indicate ringing at the
Station. If you have successfully established speech with Station 203 the LED glows Blue continuously.
• Red indicates the state of other Stations/Trunks. For example, if the LED of the DSS key assigned to
Station 201 is glowing Red continuously, it means Station 201 is busy with another Station or Trunk.
• Violet indicates the state of the trunk you are in speech with. For example, when you are in speech on
an outgoing call on Trunk 1 the LED of the DSS Key assigned to Trunk 1 will be continuously ON.
When you put the call on hold, the LED will blink slowly.
• Status of Features: The LED of a DSS key is activated when the feature assigned to this key is used.
Not all features require LED indication. For example, the feature Call Pick-Up does not require an LED. So
when you assign this feature to a DSS key, the LED of the key remains inactive, when Call Pick-Up is
accessed.

Feature like Auto Redial requires an LED to show that it has been set or canceled. So, the LED of the DSS
key to which the Auto Redial feature has been assigned will glow Red, when Auto-Redial is set, and the
LED is turned off when the feature is canceled.
The LEDs of the Call Appearance (CA) Keys will function in the same manner as the DSS Key LEDs.
Dial Pad
The dial pad consists of 12 fixed keys for the digits 0, 1-9, and the characters * and #. The dial pad is used for
dialing numbers of stations, external parties, and for dialing the programming and feature access codes.
Speaker Key
The speaker key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable,
you can assign any other feature/function on this key.
Speaker Key LED

The Speaker Key on EON48 is equipped with a single color LED which glows Blue when pressed for the speaker
mode and is turned off, when you exit the speaker mode.
Volume Keys
• "+" (plus): This is the increase key, to raise the volume of speech while talking and to increase the Ringer
volume, when the phone is ringing.
• "-" (minus): This is the decrease key, to lower the volume of speech while talking and to decrease the
Ringer volume when the phone is ringing.
Headset Connectivity
The EON48 provides two Headset interfaces: A 2.5mm Audio Jack and an RJ9 connector at the bottom of the
phone body.
So you can use any stereo headset of standard make with a 2.5 mm single connector or a headset with an RJ9
connector.
You can also assign Headset key function to any of the DSS keys. Refer the topic “DSS Keys Programming” for
instructions.

Key Maps
As EON48 may be the extension of the Operator/s and Executives in an enterprise, and the extension of the Front
Desk Attendant and Guest in hotels, to meet the varied requirements of each user group, SARVAM UCS provides
Key Maps for Operator, Executive1, Executive2, Executive3, Hotel Attendant and Hotel Guest.
The key maps of the Operator and Executive 1, 2, 3 are the same.
These key maps can be customized to match the exact requirement of individual users. Refer the topic “DSS Keys
Programming” for instructions on customizing the Key Maps.
Phone Menu
You can access the following System and phone features from the Menu of EON48:
Menu option Description
Call Logs To view call history of internal and external Missed, Answered and Dialed calls.
You can also edit numbers in the call logs and store them in the Personal Directory.
Call Forward To set and cancel Call Forward-Busy, Call-Forward No Reply, Call-Forward-
Unconditional, and Follow Me.
Dynamic Lock To change the Toll Control level of the phone.
User Status To set User Present or User Absent.
Keypad Lock To lock the keypad of the phone.
Do Not Disturb To set/cancel Do Not Disturb on the phone, that is, block incoming internal and external
calls.
Call Cost Display To view the cost of calls made from the phone.
Hotline To set/cancel Hotline and Delayed Hotline.
Alarm To set/cancel Personalized and Automated Alarms.
Change User To change User Password.

Password
One Touch Transfer To configure the fixed destination number for One Touch Transfer.
Phone Settings To customize settings of the phone such as Speech and Ringer Controls, LCD Display
settings (Brightness and Contrast, Backlight ON/OFF), Headset Connectivity, Call
Answering Mode (manual/auto answer).

Navigating the Phone Menu
• Tap on Enter key when the phone is idle.

• Scroll by tapping the Up/Down Navigation Key to reach the desired Menu option.
• Tap on Enter key to select the desired Menu option.
• Scroll by tapping on the Up/Down Navigation Key to reach the desired sub-menu option.
• Tap on Enter key to select the desired sub-menu option.
To exit menu,
• Press Cancel key.
Or
• Go ON-Hook.
Call Waiting Indication
For EON48 V2 and earlier

During an on-going call, if there is another incoming call, an indication will be provided to you for the waiting call.
• The call waiting indication depends on the Call Waiting Tone option you select — Beep Once, Beep until
answered or Off.
• the LCD will display the Caller ID of the extension/trunk.
For EON48 V3 and later

Ringer Mode — Ring if Idle, Ring Immediately or Ring After Delay
If you are in speech using the handset/headset and there is a waiting call for you,
• the ring will be played on the speaker.
If you are in speech on a speaker and there is a waiting call for you,
• the Call Waiting Tone will be played as per the option you select — Beep Once, Beep until answered or
Off.
Ringer Mode — Silent
If you are in speech using the handset/headset/speaker and there is a waiting call for you,
Operating EON48
Please refer the EON48_310_SPARSH VP248_310 User Guide for instructions on operating the features of
SARVAM UCS using EON.

EON310
2 lines and 24 characters LCD display, full duplex, fixed feature keys.
Ringer LED
LCD
Navigation
Keys
Feature Keys
(Programmable Keys)
01
02
03
04
05
DSS Keys
06 (Programmable Keys)
07
08
09
10
11
12
Dial Pad
Volume Speaker Volume
Decrease Key Increase
Keys Keys
LCD Display
The LCD display of EON310 is backlit for a clear view of the text/characters displayed.
Settings" of the DKP Phone Menu.
Ringer LED
The Ringer LED will glow in Blue to indicate incoming internal and external calls.

Navigation Keys
These are 6 keys. The functions of each are described briefly below.
• Enter Key: To enter the Menu; when the phone is in the idle state (without any incoming or outgoing call
being made), if you tap the 'Enter' key, you will enter into the 'Menu'.
Enter key is also used to make a selection from the Menu/sub-menu options or to complete an action.
• Cancel Key: To exit a menu; To abort a function/process; To delete a digit.
• Forward Key: To move forward when dialing a number.
• Back Key: To move backwards when dialing a number; to go back one level in the Menu.
Feature Keys
Sr.No. Feature Icon Assigned Feature LED
3. Hold No
4. Do Not Disturb (DND) Single Color - Blue
6. Conference No
7. Logs Single Color - Blue
8. Contacts No
9. Transfer No

These Feature keys are programmable. Refer the topic “DSS Keys Programming” for instructions on programming
these keys. You cannot change the labels of these keys and therefore it is recommended that you avoid
programming these keys.

These are 6 programmable keys that can be assigned to Stations and Trunks and important or frequently accessed
features of SARVAM UCS.
Refer the topic “DSS Keys Programming” for instructions on programming these keys.
DSS Key LEDs

Each DSS key is equipped with an LED which glows to indicate the status of the Trunk/Station or Feature assigned
to it.
• Status of Stations and Trunks: The LED of DSS keys assigned to Stations/Trunks glow in three colors to
indicate status of the call event on the Stations/Trunks and on the DKP.
Thus, the status of the DKP user's own Station as well as that of the other Stations and the status of Trunk
lines are indicated by the LED of the DSS keys assigned to those Stations and Trunks on the DKP.
LED LED Mode

Station you are in speech Station you have kept on Station you are calling or from
with. hold. which you are being called.
Station that is now busy with Station which has put another Station/Trunk that is called or
another Station/Trunk. Station/Trunk on hold. being called by another.
• Blue indicates the state of the station/trunk you access. For example, when you make a call to another
Station 203, the LED of the DSS key assigned to Station 203 blinks Blue to indicate ringing at the
Station. If you have successfully established speech with Station 203 the LED glows Blue continuously.
Station 201 is glowing Red continuously, it means Station 201 is busy with another Station or Trunk.

accessed.
Dial Pad
dialing numbers of stations, external parties, and for dialing the programming and feature access codes.
Speaker Key
The speaker key sets the phone in 'Speaker mode' for hands-free operation.
Speaker Key LED

The Speaker Key on EON310 is equipped with a single color LED which glows Blue when pressed for the speaker
Volume Keys
The EON310 provides two Headset interfaces: A 3.5mm Audio Jack and an RJ12 connector at the bottom of the
phone body.
So you can use any headset of standard make with a 3.5 mm single connector or a stereo headset with an RJ9
connector.
You can also assign Headset key function to any of the DSS keys. Refer the topic “DSS Keys Programming” for
instructions.

Key Maps
As EON310 may be the extension of the Operator/s and Executives in an enterprise, and the extension of the Front
Desk Attendant and Guest in hotels, to meet the varied requirements of each user group, SARVAM UCS provides
Key Maps for Operator, Executive1, Executive2, Executive3, Hotel Attendant, and Hotel Guest.
The key maps of Executive 1, 2, 3 are the same.

These key maps can be customized to match the exact requirement of individual users. Refer the topic “DSS Keys
Programming” for instructions on customizing the Key Maps.
Phone Menu
You can access the following System and phone features from the Menu of EON310:
Contacts To add, edit, delete names and numbers of contacts in the Global Directory Part 1.
calls.

Password


To exit menu,

Or
• Go ON-Hook.

The call waiting indication depends on the Call Waiting Tone option you select — Beep Once, Beep until answered
or Off.
Operating EON310
SARVAM UCS using EON.
EON510
240 x 64 Pixel Graphical LCD display, full duplex, fixed feature keys.

LCD Display
The LCD display of the phone is backlit. The LCD backlight can be turned on and off as well as adjusted for
contrast and brightness from the Phone Menu.
Ringer LED
Navigation Keys
The function of each Navigation key is described below.
• Forward Key: To move forward when dialing a number or scroll to view the Context Sensitive Key options.
• Back Key: To move backwards when dialing a number, to go back one level in the Menu or scroll
backwards to view the Context Key options.
• Menu or Select / OK Key : To enter the Menu; when the phone is in the idle state (without any
incoming or outgoing call being made).
Menu Key functions as the Select / OK Key to make a selection from the Menu/sub-menu options or to
complete an action. When there is an incoming call it functions as the Answer Key.
• Cancel Key : To Cancel all features set by you or exit the Menu/sub-menu.

Feature Keys
EON510 supports following feature keys. Default features assigned to these keys are as follows.
Assigned
Feature icon LED Programmable
Feature
Hold No Yes
Redial No Yes
Transfer No Yes
Conference No Yes
Voicemail Single Color - Blue Yes
Do Not Disturb Single Color - Blue Yes
Headset Single Color - Blue No
Mute Single Color - Blue No
All Feature keys are programmable except Headset and Mute key. Refer the topic “DSS Keys
Programming” for instructions on programming these keys. You cannot change the labels of these keys and
therefore it is recommended that you avoid programming these keys.

There are 16 programmable keys that can be assigned to Extensions and Trunks and important or frequently
DSS Key LEDs

Each DSS Key is equipped with an LED which glows to indicate the status of the Trunk/Extension or Feature
assigned to it.
• Status of Extensions and Trunks: The LED of DSS keys assigned to Extensions/Trunks glow in three
colors to indicate status of the call event on the Extensions/Trunks and on the phone.
Thus, the status of the phone user's own Extension as well as that of the other Extensions and the status
of Trunk lines are indicated by the LED of the DSS keys assigned to those Extensions and Trunks on the
phone.
LED LED Mode

Extension you are in speech Extension you have kept on Extension you are calling or
with. hold. from which you are being called.
Extension that is now busy with Extension which has put another Extension/Trunk that is called or
another Extension/Trunk. Extension/Trunk on hold. being called by another.
Violet You are talking on a Trunk You have held a Trunk (external You have an incoming call on
(external call) call) the Trunk (external call)
• Blue indicates the state of the extension/trunk you access. For example, when you make a call to
another Extension 203, the LED of the DSS key assigned to Extension 203 blinks Blue to indicate
ringing at the Extension. If you have successfully established speech with Extension 203 the LED
glows Blue continuously.
• Red indicates the state of other Extension/Trunks. For example, if the LED of the DSS key assigned to
Extension 201 is glowing Red continuously, it means Extension 201 is busy with another Extension or
Trunk.
accessed.

Dial Pad
The dial pad consists of 12 fixed keys for the digits 0, 1-9, and the characters * , #, Lock. The dial pad is used for
dialing numbers of extensions, external parties, and for dialing the programming and feature access codes.
Speaker Key
Speaker Key LED

The Speaker Key on the phone is equipped with a single color LED which glows Blue when pressed for the speaker
Volume Keys
The phone provides two Headset interfaces: A 3.5mm Audio Jack and an RJ9 connector at the bottom of the phone
body.
So, you can use any stereo headset of standard make with a 3.5 mm single connector or a stereo headset with an
RJ9 connector.
To use the Headset, a Headset Key is assigned on the phone. The Headset Key on the phone is equipped with a
single color LED which glows Blue when pressed for the headset mode and is turned off, when you exit the headset
mode.
Key Maps
As EON510 may be the extension of the Operator/s and Executives in an enterprise, to meet the varied
requirements of each user group, SARVAM UCS provides Key Maps for Operator and Executives. These key maps
can be customized to match the exact requirement of individual users. All the key maps of the EON510 are the
same.

Phone Menu
You can access the following System and phone features from the Menu of the phone:
calls.

Password


• Press the Select or OK Key to select the desired Menu option.
• Press the Select or OK Key to select the desired sub-menu option.
To exit menu,

or
Go ON-Hook.

The call waiting indication depends on the Call Waiting Tone option you select — Beep Once, Beep until answered
or Off.
Operating EON510
Please refer the EON510_SPARSH VP510 User Guide for instructions on operating the features of SARVAM UCS
using EON.
EONSOFT
The EONSOFT is a PC-based Digital Key Phone. Based on a graphic user Interface (GUI), the EONSOFT offers all
the features of EON48, making it a substitute for the Digital Key Phone. Its integration with the SARVAM UCS
obviates the need for a separate telephone instrument.
The EONSOFT can be installed on any personal computer with Windows or NT operating system.
Two PC-based DSS64 Consoles are available to be used with the EONSOFT. You can use either one or both
DSS64 Consoles.
The EONSOFT occupies only a single port, even when both PC-based DSS64 Consoles are used. Thus it supports
all the features of the Digital Key Phone and the DSS64 Console on a single window.

DKP Port
The DKP port connects EONSOFT to the DKP port of SARVAM UCS's DKP Card.
Handset Port
The Handset port connects the Receiver of the phone, to be used for speech.
The EONSOFT has the provision for attaching a Handset. A handset with spring cord is supplied by Matrix and is to
be connected to the handset jack (RJ12) on the Dongle.
Headset connectivity
EONSOFT supports headset connectivity, providing a MIC and a Speaker interface. Any stereo Headset of
standard make, with dual connectors can be connected to the MIC and the Speaker on the Dongle.
COM Port
The COM port connects EONSOFT to a PC (COM Port).
After EONSOFT has been successfully installed on a PC and the DKP parameters have been configured, each
time you open EONSOFT, the display and keypad of the phone will appear on your PC screen.
The illustration below shows EONSOFT with two DSS64 Consoles attached to it.

Phone Display
The EONSOFT has a 2-line and 24-character display. In the ON-Hook or idle condition, the first line displays the
Station Number and the Station name. The second line displays the Day, Date and Time.
When there is an incoming call, the calling party's number is displayed on Line 2 of the LCD246.
The LCD messages for various call events (dial, transfer, forward, hold, etc.), for prompts, alerts, confirmation,
errors, text messages, are displayed.

These are a set of 25 keys programmable keys, allowing you to assign Stations, Trunks or Features to them, so
that you can access any of these by pressing the respective key.
Refer the topic “DSS Keys Programming” for instructions on assigning stations, trunks, features to keys.
DSS Key LEDs

Each DSS key is equipped with an LED which changes the color to indicate the status of the Trunk/Station or
Feature assigned to it.
• Status of Stations and Trunks: The LED of DSS keys assigned to Stations/Trunks may change in three
different colors to indicate status of the call event on the Stations/Trunks on EONSOFT.
Thus the status of the Stations and the Trunk lines are indicated by the LED of the DSS keys assigned to
those Stations and Trunks on EONSOFT.
LED LED Mode

Green The key assigned to the The key assigned to the The key assigned to the Station
Station you are in speech Station you have kept on you are calling or from which you
with. hold. are being called.
Red The key assigned to the The key assigned to the The key assigned to the Station/
Station that is now busy Station which has put Trunk that is called or being
with another Station/ another Station/Trunk on called by another.
Trunk. hold.
Orange You are talking on a You have held a Trunk You have an incoming call on the
Trunk (external call) (external call) Trunk (external call)
• Green indicates the state of the station/trunk you access. For example, when you make a call to
another Station 203, the LED of the DSS key assigned to Station 203 blinks Green to indicate ringing at
the Station. If you have successfully established speech with Station 203 the LED is Green
continuously.
Station 201 is Red continuously, it means Station 201 is busy with another Station or Trunk.
246. Only if the Station, to which EONSOFT is connected, has been allowed CLIP facility in its Class of Service.

• Orange indicates the state of the trunk you are in speech with. For example, when you are in speech
on an outgoing call on Trunk 1 the LED of the DSS Key assigned to Trunk 1 will be continuously ON.
The LEDs of DSS Keys that are designated as Call Appearance (CA) Keys will function as follows:
LED LED Mode

Green When you are in speech When you have put a When any Station is calling
with a Station (internal call) Station on hold (internal call) (internal call)
Orange When you are in speech When you have put a Trunk When any Trunk is calling
with Trunk (external call) on hold (external call) (external call)
The LED of DSS keys assigned to Stations/Trunks turn Red to indicate status of the call event on the
Stations/Trunks.
• Not all features require LED indication. Hence the LED on a DSS Key is activated only if the feature
assigned to that key requires LED.
• For example, Call Pick-Up; this feature does not require an LED. So when a DSS key is assigned to
this feature, the LED of the key remains inactive, when Call Pick-Up is accessed.
• A feature like Auto Redial requires an LED to show that it has been set or canceled. So, the LED of the
DSS key to which the Auto Redial feature has been assigned will turn Red, when Auto-Redial is set,
and the LED is turned off when the feature is canceled.
• Thus the LEDs of the DSS keys function only if the LED is relevant for the feature/ function assigned to
the keys, otherwise it remains inactive.
• The LEDs of DSS keys to which features like Raid, Interrupt Request, Barge-In, Last Caller Recall are
assigned, will not change color.
Dial Pad
The dial pad consists of 12 keys (non-programmable), which include the digit keys for 0, 1-9, and character keys for
* and #.
Function Keys
These are non-programmable keys on the keypad of EONSOFT which have fixed functions.
• Redial: This key is used for redialing the last external number.
• Func Key: This key is used for accessing the Phone menu.
• Names: This key is used to dial by Name.
• : This key is used for accessing the Address Book. The EONSOFT provides the facility of an
Address Book that is integrated with the Standard Windows Address Book, for storing the numbers and

addresses of contacts. So, when a call is to be made, you can select and dial the desired number from the
directory.
• Rel: This key is used to disconnect an on-going call.
• Xfr: This key is used for transferring calls to another destination
• Hold: This key is used for putting the caller on hold. This key is also used to make a selection in the Phone
Menu.
• : This key is used for going OFF-Hook. It simulates lifting of the handset, pressing of the speaker
key to make or receive calls.
• : This key is used for going ON-Hook. It simulates replacing of the handset, pressing of the speaker
key to disconnect.
Volume Control Keys

The  and  keys are used for increasing or decreasing the phone volume. Volume can be increased by clicking 
and decreased by clicking .
Navigation Keys
The following keys are used for navigating the phone menu:
• 'Func' key: This key is used for entering the Phone menu and to go back one level in the menu.
• Up and Down keys: The  and  keys function as the Up and Down keys to scroll the Menu and sub-
menu options. You can scroll up down the menu by clicking  and scroll up the menu by clicking .
• 'Hold' key: This key is used to make a selection in the Menu.
Speaker key
The 'Spk' key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable; you
can assign any other feature/function to this key.
Shortcut keys
You can use the Keyboard of the PC to operate EONSOFT, with the help of "shortcut keys'. The following table
describes the functions performed when shortcut keys on the keyboard are pressed:
Short Cut Key Label Description
F1 Help
F2 Windows Address Book
F3 Spd
F4 Func
F5 Space Bar - Hook
F6 Alt+Enter - Hold

Short Cut Key Label Description
F7 Xfr
F8 Spk
Esc ON-Hook
. (dot/period) Flash
 (Up Arrow Key) Volume key. To increase volume of ringer and speech
 (Down Arrow Key) Volume key. To decrease volume of ringer and speech
Ctrl+C To open COM Port Settings Dialog Box
Ctrl+T To open Trunk Access Code Dialog Box
Tab  (Tab Forward) Forward shifting of selection
Tab  (Tab
Backward shifting of selection
Backward)
Shift + Enter  To press the button with selection
Num Pad Dial out a number
Key Maps
EONSOFT can function as a Station for the Operator, Executive, and Hotel Attendant, also Guest (though unlikely
to be used by guests).
Phone Menu
The Phone menu is the same as EON48.

• Click 'Func' key when the phone is idle.

• Scroll by clicking the Up/Down Key to reach the desired Menu option.
• Click 'Hold' key to select the desired Menu option.
• Scroll by clicking the with Up/Down Key to reach the desired sub-menu option.
• Click 'Hold' key to select the desired sub-menu option.
To exit menu,
• Press the 'Func' key repeatedly to go back one level in the menu, till you reach 'Menu'.
Or
• Press Cancel key, if programmed.
If you want to use the Keyboard, press the Shortcut key for the desired function.

Tool Tips
You can assign labels and tool tips for the DSS keys, which are displayed to the user on mouse over. You can
assign the function of each key as Tool Tip, to help user in intuitive operation of EONSOFT.
To assign a tool tip,
• right click the desired DSS key.

• The dialog box will show the Key Number.
• Enter the label for the key.
• Enter the Tool Tip for the key.
• Click 'OK'.
• Repeat the same steps to configure Tool Tip on another key.
Each time you move the mouse on the key, the Tool Tip you programmed for this key will be displayed.
Call Indication
Incoming Calls are indicated by:
• Popping up of the EONSOFT window, when the window is minimized.

• Flashing of the Title Bar of the EONSOFT window, when the window is maximized.
In order for the EONSOFT window to pop up, you must have enabled the 'PopUp When Ring' option. When
this option is enabled and the EONSOFT window is minimized a new incoming call causes the window to pop
up to its full size notifying the user about the new call. When this option is enabled and the EONSOFT window
is maximized, a new incoming call is indicated by the flashing of the Title bar of the window.
When the 'PopUp When Ring' option is disabled and the EONSOFT window is minimized, a new incoming call
is indicated by the flashing of the EONSOFT Title at the bottom bar of the PC screen.
By default, the 'PopUp When Ring' option is enabled.
Second Call Indication

When you are in an on-going conversation and there is a second incoming call, EONSOFT will indicate the second
incoming call. The second call indication depends on the type of Ringer Mode selected by you.
The tables below describes the second call indications for each Ringer Mode.
Ringer Mode - Ring if Idle
Second call indication, when the Second call indication, when the Second call indication, when the
first call is on the Speaker first call is on the Handset first call is on the Headset
Call Waiting Tone is played and the

Call Waiting Tone is played and the
LCD displays the Caller ID of the
LCD displays the Caller ID of the NA
station/trunk.
station/trunk.

Ringer Mode - Ring Immediately
Call Waiting Tone is played and the Ring on the speaker and the LCD
LCD displays the Caller ID of the displays the Caller ID of the station/ NA
station/trunk. trunk.
Ringer Mode - Ring After Delay
The LCD displays the Caller ID of The LCD displays the Caller ID of
the station/trunk and the Call the station/trunk and there is a Ring
NA
Waiting Tone is played after the on the speaker after the expiry of the
expiry of the Ring Delay Timer. Ring Delay Timer.
Ringer Mode - Silent
The LCD will display the Caller ID of the station/trunk..
Operating EONSOFT
EONSOFT can be operated using the keyboard and the mouse.
Making calls
To make calls,
• Click the desired digits on the dial pad.

• The number will be dialed out.
• Talk.
• Click ON-Hook key to disconnect.
To make calls from Headset mode,
• Click OFF-Hook key.

To make calls from Speaker mode,
• Click 'Spk' key.

• A headset must be connected and 'Headset Connectivity' must be enabled in the 'DKP Parameters'.
Refer “DSS Keys Programming” for instructions.
• If you are using the keyboard instead of the mouse, press the appropriate Shortcut Keys listed above
and use the Number pad on the keyboard to dial digits.

Receiving calls
• When window pops up to indicate a call,
• Click 'Spk' key or 'OFF-Hook' Key.
• Talk.
• Click 'ON-Hook' key to disconnect.
EON74247
EON74, also called Digital Turret or Dealer Board, is a special terminal for Trading Rooms of stock exchanges.
With two separate speech paths (two handsets) and 74 DSS keys, EON74, supports quick and efficient handling of
multiple calls by dealers/agents.
For detailed product description and operation instructions, refer to the EON74 User Guide.
247. Check for availability. Not supported by ETERNITY GENX.

Direct Dialing-In (DDI)
What’s this?
DDI is the service feature provided to the customers by the ISDN exchanges / ITSPs over digital trunks / SIP
Trunks. DDIs are Direct Dial-In numbers. These are additional numbers that can be associated with the main
number assigned to the ISDN line/SIP Trunk. They are normally used with the system to provide separate numbers
to extension users.
When there is an incoming call on the a specific DDI number it is routed through to the required extension without
operator intervention.
Before we understand how DDI routing works, we must understand the following terms:
• Pilot Number/ISDN Number: It is a single number assigned by the service provider to a single ISDN/IP
trunk. This is the combination of MSN Number and the first DDI Number. The Pilot Number is of maximum
16 digits. This is also known as ISDN Installation Number/ISDN Number. The MSN is a fixed series and is
given by the service provider whereas the DDI Numbers can be selected by the user and these series
vary. For example, if the Pilot Number is 2630555, the MSN is 2630 and the DDI Number is 555. However
the number of digits to be used as DDI Numbers is informed by the service provider as they may differ for
each service provider.
• Multiple Subscriber Number (MSN): In India, the MSN are the first four digits of the Pilot Number, which
remain fixed and are provided by the service provider. The number of fixed digits may vary depending on
the service provider.
• DDI Numbers: These are additional numbers, which can be selected by you but are allotted by the service
provider. These are sequential numbers. In the DDI number only the last 3 or 4 digits vary. These are
known a DDI numbers. However the number of DDI digits which vary are provided by the service
provider. These numbers are normally used with a PBX and are assigned to the extension user.
How it works
Let us understand this with the help of an example:
• A BRI line has been connected to the PBX.
• 50 DDI numbers have been given by the service provider.
• Incoming calls on the DDI numbers can be routed to different landing destinations as extension users,
department groups or other trunks within the organization.
• To assign DDI numbers to the extension users and to route calls as per the DDI numbers, you need to
configure the following:
• Incoming Reference Table
• Outgoing Reference Table
• DDI Routing Table

Handling Incoming Calls
Pre-requisites to route incoming calls as per the DDI logic,
• Assign an Incoming (IC) Reference ID to the BRI port. This Incoming (IC) Reference ID acts as a link
between the port settings and the Incoming Reference Table.
• In the Incoming Reference Table against the same Incoming (IC) Reference ID, configure the Start
Channel No., Total Channel Count, DDI Routing Ref. ID, Route on First Destination, Ring Timer (sec),
When No Reply, When Busy, Trunk Features Template as per your requirement.
• To route and place the call on the final destination, the system refers to the DDI Routing Ref. ID configured
in the Incoming Reference Table. As per this ID the system checks the DDI Routing Table. The DDI
Routing Ref. ID acts as a link between the Incoming Reference Table and the DDI Routing Table. The
system places the call on the final destination as configured against the DDI Routing Reference ID in the
DDI Routing Table.
When there is an incoming call on the BRI port,
• As per the time zone, the system checks the Incoming (IC) Reference ID assigned to the BRI port
• If the Incoming (IC) Reference ID has been configured, the system then hunts for the same ID in the
Incoming Reference Table.
If no channel match is found in the Incoming Reference Table, the call is routed as per the Trunk Feature
Template assigned to the BRI port.
If a match is found, the system verifies the channels on which the routing logic is to be applied. These are
defined in the parameters Start Channel Number and the Total Channel Count. Hence the system also
matches the channel number on which the incoming call landed and the channel configured.
• For the matching entry, the system checks the corresponding DDI Routing Reference ID. The DDI Routing
Reference ID acts as a link between the Incoming Reference Table and the DDI Routing Table.
• The system hunts for the same DDI Routing Reference ID in the DDI Routing Table. In the DDI Routing
Table, a range of DDI numbers and their respective landing destinations are configured. You can configure
as many as 224 DDI Reference ID’s with different DDI numbers and their corresponding landing
destinations. From this table the system fetches the extension number configured as the landing
destination.
• When a match is found in the DDI Routing Table for the incoming call on the respective DDI number, the
system places the call on the corresponding landing destination configured by you.
When the call is not answered by the landing destination or if the landing destination is busy, the system
again checks the Incoming Reference Table for the options you have selected for the parameters, When
No Reply and When busy. You may select from the following options - Disconnect, Route to a TLG, Greet
and Disconnect, Greet and Route to TLG or Route to VMS.
• If no match is found for the DDI number in the DDI Routing Table, the call is routed as per the Trunk
Feature Template assigned in the Incoming Reference Table.

Handling Outgoing Calls
Reverse DDI is when you want the DDI numbers assigned to the extension users to be displayed as the CLI to the
called party. For this you must configure the DDI Routing Table and the Outgoing Reference Table.
When an outgoing call is made by the extension user, the DDI numbers will be sent as CLI to the called parties only
if, the service provider supports this facility. If this facility is not supported by the service provider and if SARVAM
UCS sends the DDI Number, this number will be swapped by with the Pilot number by the service provider and then
the call will be routed further.
Pre-requisites to route outgoing calls as per the DDI logic,
• Assign an Outgoing Reference ID to the BRI port. This Outgoing Reference ID acts as a link between the
port settings and the Outgoing Reference Table.
• In the Outgoing Reference Table against the same OG Reference ID, configure the Start Channel No.,
Total Channel Count, DDI Routing Ref. ID and the ISDN Number as per your requirement.
• To fetch the DDI number the system refers to the DDI Routing Ref. ID configured in the Outgoing
Reference Table. As per this ID the system checks the DDI Routing Table. The DDI Routing Ref. ID acts
as a link between the Outgoing Reference Table and the DDI Routing Table.
When an outgoing call is made using the BRI port,
• The extensions user dials a trunk access code to grab the BRI line to out-dial a number.
• As per the time zone, the system checks for the Outgoing (OG) Reference ID assigned to the BRI port.
• If the OG Reference ID has been configured, the system then hunts for the same ID in the Outgoing
Reference Table.
If a match is found, the system verifies the channels assigned to the user for making an outgoing call.
These are defined in the parameters Start Channel Number and the Total Channel Count. Hence the
system also matches the channel assigned for making the outgoing call with the channel configured in the
table.
• The system the checks the corresponding DDI Routing Reference ID. The DDI Routing Reference ID acts
as a link between the Outgoing Reference Table and the DDI Routing Table.
• The system hunts for the same DDI Routing Reference ID in the DDI Routing Table. From this table the
system fetches the DDI number assigned to the extension user. You can configure as many as 224 DDI
Reference ID’s with different DDI numbers.
• When a match is found in the DDI Routing Table, the system sends the MSN Number + DDI number of the
extension in the calling party field to the called party.
• If no match is found in the DDI table, the ISDN Number, that is the MSN Number + the first DDI number is
sent in the calling party field to called party.

To route incoming calls as per the DDI logic you must configure the following:
• Assign the Incoming Reference ID on the respective port.
• Incoming Reference Table
• DDI Routing Table
To configure the Incoming Reference Table,
• Determine the ISDN ports/SIP trunk on which you want to apply the DDI logic, for example, BRI port.
• On the BRI port, assign an Incoming Reference ID number. To do this,
• Scroll to Incoming (IC) Reference ID.
• Assign an Incoming (IC) Reference ID number for each time zone.
• Under Configuration, click DDI Routing.
• Click Incoming Reference Table and configure the following parameters:
• IC Reference ID: You must configure the parameters to apply DDI Routing against the same ID as
assigned on the BRI port. This ID is a link between the port settings and the Incoming Reference Table.
• Start Channel Number: The system starts applying the routing logic from this channel of the trunk.
• Total Channel Count: This is the total number of channels of the trunk on which the system applies
the routing logic.
• DDI Routing Reference ID: Enter the DDI Routing Reference ID number. As per the this reference
number the system checks the DDI Routing Table and determines the final landing destination for
placing the call.
This ID is a link between the Incoming Reference Table and the DDI Routing Table.
• Route on First Destination: When the final destination extension is identified as per the DDI number
table, the system checks if this check box is enabled. If it is enabled, every incoming call will always be
placed on the first extension configured in the DDI Table. If the check box is clear, the call is routed to
the respective extension number.
• Ring Timer: This timer signifies the time for which the extension on which the incoming call is placed
must ring. On expiry of this timer if the call is not answered the call is routed as per the option you
configure for the parameter When No Reply.
• When No Reply: When the call is not answered by the DDI extension, the system checks the option
you have selected for this parameter and processes the call further. You may select any option from the
following:

• Disconnect the call
• Route the call to Trunk Landing Group
• Answer the call, greet the caller with a voice message, and on completion of the message route the
call to Trunk Landing Group.
• Route the call to Voice Mail, that is to the Mail box assigned to the DDI extension. For this, a mail
box must be assigned to the extensions. If the extension is not assigned mail box, the caller will
hear the welcome message of the VMS, but will not be able to access the mail box.
• When Busy: If the DDI extension is busy, the system checks the option you have selected for this
parameter and processes the call further. You may select any option from the following:
• Disconnect the call.
• Route the call to Trunk Landing Group.
• Answer the call, greet the caller with a voice message, and on completion of the message route the
call to Trunk Landing Group.
• Route the call to Voice Mail, that is to the Mail box of the DDI extension. For this, a mail box must be
assigned to the extensions. If the extension is not assigned mail box, the caller will hear the
welcome message of the VMS, but will not be able to access the mail box.
• Trunk Feature Template: A Trunk feature template is assigned to each Incoming Reference ID. You
can enable different features in the Trunk Feature Templates. A different Trunk Feature Template can
be assigned to each channel or a range of channels, such as set Auto Answer Time Zone wise, allow
DID according to the time zone. Hence incoming calls can be answered and routed in different ways.
For more details refer the topic “Trunk Feature Template”.
• Similarly, to assign IC Reference ID to T1E1 ports, under Configuration, click T1E1 Configuration.
Click Port Parameters.
• To assign IC Reference ID to SIP Trunks, under Configuration, click VoIP Configuration. Click SIP
Trunk Parameters.
• After the Incoming Reference Table has been configured you must configure the DDI Routing Table.
DDI numbers can be assigned to extensions users, Department Groups, Routing Groups or to different
port types, depending on the organizations requirement.
You can configure as many as 224 DDI Reference ID’s with different DDI numbers and landing
destinations. To configure the DDI Routing Table,
• Click DDI Routing Table and configure the following parameters:
• DDI Routing Reference ID: You must configure the parameters to apply DDI Routing against the
same ID as assigned in the Incoming Reference Table. This ID is a link between the Incoming
Reference Table and the DDI Routing Table.

DDI Routing Ref. ID’s in the DDI Routing Table act as identifiers. This Reference ID is assigned in the
Incoming Reference Table.
When there is an incoming call the system checks the Incoming Reference ID assigned to the port/
trunk. The system tracks this ID from the Incoming Reference Table.
• Start DDI Number: This is the first DDI Number from where the DDI range starts. You must only enter
the DDI number. For example, if the Pilot Number is 2630555. If 10 DDI numbers have been taken,
then the DDI Number range is from 555 to 564. The Start DDI Number in this case is 555.
• Total DDI Numbers: The total number of DDI numbers for you want to set the same landing
destination. For example, if the Start DDI Number is 555, Total DDI Number is 1, then the landing
destination selected will only be for the DDI Number 555. If Start DDI number is 556 and the Total DDI
Numbers is configured as 10, then the landing destination will be for the entire range of numbers, that
is from 556 to 565.
• DDI Number of Digit: The number of digits in a DDI number. Suppose 100 DDI numbers are supported
on an ISDN Trunk, then the Number of Digits for that Trunk should be configured as 3. Suppose 10 DDI
numbers are supported on another ISDN Trunk, then the Number of Digits for that Trunk should be
configured as 2.
• Route to Destination: The incoming calls on the DDI numbers can be placed on extensions users,
Department Groups, Routing Groups or to different port types, depending on the organizations
requirement.
Configure the following for the parameters for the landing destination:
• Port Type: Select the port type to place the DDI call from the following:
• BRI
• T1E1
• E&M
• AOP
• Department Group
• Quick Dial
• Routing Group
• Voice Mail Auto Attendant
• Mobile
• SIP Trunk
• Flexible Number
• Virtual Extension
For example, to use SARVAM UCS as a gateway select Port Type as Mobile, BRI, T1E1 etc.
To place incoming calls on the extensions, select Flexible Number as the Port Type.
• Port Number: For the Port Type you have selected, enter the Port Number/Group Number. The
Port Number range depends on the Port Type you have selected.
• Start DDI Flexible Number: If you have selected the Port Type as Flexible Number, enter the first
extension number to which you want to assign the first DDI Number. As per the Total DDI numbers
configured, the system automatically maps the DDI numbers with the extension numbers. Both the
DDI numbers and the extensions numbers are mapped sequentially. For example, if the Start DDI

number is 2630555, the Total DDI Numbers are 10 and Start DDI Flexible Number 2001, the
system maps the DDI numbers with the extensions numbers as follows:
Start
DDI Number Flexible
Number
555 2001
556 2002
. .
. .
. .
. .
. .
564 2010
Hence the call for the respective DDI Number is placed on the respective extension number.
• Voice Mail Auto Attendant (VMAA) Menu: if you have selected the Voice Mail Auto Attendant as
the Port Type, select the VMAA Menu to be assigned to the respective DDI Routing Table entry.
You may click the Voice Mail Auto Attendant (VMAA) Menu link to edit the parameters of desired
VMAA Menu. For details, see “Voice Mail Auto-Attendant Menu”.
If you want to apply Reverse DDI, that is if you want DDI numbers to be displayed as the CLI to the called party, you
must configure the Outgoing Reference Table. The system uses the same DDI Routing Table to trace the
extension users and their respective DDI numbers when an outgoing call is made.
To assign the Outgoing Reference ID and to configure the Outgoing Reference Table you must,
• Determine the ISDN ports/SIP trunk on which you want to apply the Reverse DDI logic, for example, BRI
port.
• On the BRI port, assign an Outgoing Reference ID number. To do this,
• Scroll to Outgoing (OG) Reference ID.
• Assign an Outgoing (OG) Reference ID number for each time zone.
• Click Outgoing Reference Table and configure the following parameters:

• Outgoing Reference ID: To apply DDI Routing configure the parameters given below against the
same ID as assigned on the BRI port. This ID is a link between the port settings and the Outgoing
Reference Table.
• Start Channel Number: The system starts applying the routing logic from this channel of the trunk.
• Channel Count: This is the total number of channels of the trunk on which the system applies the
routing logic.
• ISDN Number: Each ISDN Trunk is given an Pilot Number by the Service Provider. This is the
combination of MSN Number and the first DDI Number. The Number is of maximum 16 digits. This is
also known as ISDN Installation Number/ISDN Number. The MSN number is given by the service
provider whereas the DDI Numbers can be selected by the user. However the number of digits to be
used for the DDI Number is informed by the service provider.
• DDI Routing Reference ID: Enter the DDI Routing Reference ID number. As per the this reference
number the system checks the DDI Routing Table and determines the DDI number assigned to the
extension user. The system then swaps this DDI number with the ISDN number when an outgoing call
is made. This DDI number is displayed as the CLI to the called party.
The DDI numbers will be displayed as CLI to the called party only if the service provider supports this
facility over the ISDN Trunk/SIP Trunk.
• Similarly, to assign OG Reference ID to T1E1 ports, under Configuration, click T1E1 Configuration.
Click Port Parameters.
• To assign OG Reference ID to SIP Trunks, under Configuration, click VoIP Configuration. Click SIP
Trunk Parameters.
Configuring IC Reference Table using a Telephone

• To program IC Reference Table Index, dial:

• 6302-1-IC Reference Table Index-Feature Number-Value
• 6302-2-IC Ref. Table Index-IC Ref. Table Index-Feature Number-Value
• 6302-*-Feature Number-Value
Where,
IC Reference Table Index is from 01 to 64.
See the table at the end of this topic for Feature numbers and the codes.
• To restore default IC Reference Table Index, dial:

• 6301-1-IC Reference Table Index to restore default of a single index.
• 6301-2-IC Ref. Table Index-IC Ref. Table Index to restore defaults of a range of index.
• 6301-* to restore defaults of the entire table.
Where,
IC Reference Table Index is from 01 to 64.
• Exit SE mode.

Default values of the IC Reference Table
Feature No. 01 02 03 04 05 06 07 08 09
Feature IC Start Total DDI DDI Ring Trunk

Route on When When
Name/Table Ref. Channel Channel Routing Timer Feature
First Dest. No-Reply Busy
Index ID No. Count Ref. ID (Sec.) Template
01 00 01 00 00 Disable 045 Disconnect Disconnect 01
: : : : : : : : : :
Parameters Value:
Code 0 00- 01-30 00-30 00-99 Disable 001-255 Disconnect Disconnect 01-50
99
1 Enable Route to Route to

TLG TLG
2 Greet and Greet and

Disconnect Disconnect
3 Greet and Greet and

Route to Route to
TLG TLG
4 Route to Route to
Voice Mail Voice Mail
Auto Auto
Attendant Attendant
Configuring OG Reference Table using a Telephone

• To program OG Reference Table Index, dial:

• 6312-1-OG Reference Table Index-Parameter Number-Value
• 6312-2-OG Reference Table Index-OG Reference Table Index-Parameter Number-Value
• 6312-*-Parameter Number-Value
Where,
OG Reference Table Index is from 01 to 64.
Parameter Number is from 1 to 5.
• To default an OG Reference Table:

• 6311-1-OG Reference Table Index
• 6311-2-OG Reference Table Index-OG Reference Table Index
• 6311-*
Where,
OG Reference Table Index is from 01 to 64.
• Exit SE mode.

Default values of OG reference table as given below:
Parameter No. 1 2 3 4 5
Parameter Name/ OG Ref. Start Channel ISDN DDI Routing Ref.

Channel Count
Index ID No. Number ID
01 00 01 00 Blank 000
02 00 01 00 Blank 000
03-63 Same as 02
64 00 01 00 Blank 000
Parameter Value:
Code 01-99 01-02 01-30 16 digits 001-128
01-30
Configuring DDI Routing Table using a Telephone

• To configure the feature in a DDI Routing Table:

• 6322-1-DDI Routing Table ID-Parameter Number-Value
• 6322-2-DDI Routing Table ID-DDI Routing Table ID-Parameter Number-Value
• 6322-*-Parameter Number-Value
Where,
DDI Routing Table ID is from 001 to 224.
Parameter Value is from 01 to 06. Refer the DDI Routing Table for Parameter Values and their codes.
• To default a DDI Routing Table:

• 6321-1-DDI Routing Table ID
• 6321-2-DDI Routing Table ID-DDI Routing Table ID
• 6321-*
Where,
DDI Routing Table ID is from 001 to 224.
• Exit SE mode.
Following table shows default DDI Routing Table
01 02 03 04 05 06
Para. No./ DDI Start Total Start
Table ID DDI Number Port
Routing DDI DDI Port Type Flexible
of Digit Number
Ref. ID Numbers Numbers Number
001 00 000000 000000 0 None Blank Blank
002 00 000000 000000 0 None Blank Blank
003-223 Same as 224
224 00 000000 000000 0 None Blank Blank

Parameter Value:
Code 00 00-99 000001- 000000- 0-4 None -- Max. 6-

999999 999999 digits
04 BRI 01-32
05 T1E1PRI 1-8
06 E&M 01-32
11 Dept Grp 01-16
12 Quick Dial 001-999
20 Routing 01-32
Group
25 Mobile 01-40
26 SIP 01-32
27 Flex. No. 4-digits
36 Virtual 01-64
Extension
• Refer to above table. Whenever Flexible number is selected as the port type, the range concept of Start
flexible number becomes relevant. In all other cases, range concept is not relevant.
• Flexible Number includes SIP Extension Numbers.
• When a call is placed on SLT/DKP port, the calling party number is displayed on the terminal.
When the call is placed on BRI-NT or PRI-NT, the calling party number and the called party number both are sent to
the NT port. Doing so, the system connected to the NT port can resolve the DDI number and place the call on the
programmed extension.

Direct Inward System Access (DISA)
What’s this?
With Direct Inward System Access (DISA) remote users can access and use the system's features and facilities
using Trunks, on which this feature is enabled.
Using DISA, remote users can:

• call any extension.
• make external calls.
• make calls using Authority Codes.
• use features and facilities of the system.
• configure features and facilities of the system and administer the system.
All these can be done as if being done from a local extension of the SARVAM UCS.
DISA Variants
SARVAM UCS offers three types of DISA, each with a different method of authentication and level of access:
• PIN Authentication-Multiple Calls
• DISA with CLI Authentication-Multiple Calls
• DISA with CLI Authentication-Single Calls
PIN Authentication - Multiple Calls

Callers can access an extension of SARVAM UCS by dialing the DISA Login Code that consists of:
• the DISA Feature Access Code.
• the extension number they want to access.
• the User Password of that extension.
Callers are authenticated and allowed to use the extension on which they are logged in.
The callers must dial special digits or codes to go On-hook, Off-hook. They are allowed to make as many trunk calls
and internal calls for as long as they remain logged in to the DISA mode.
To end the DISA login session, callers must dial the Termination code or disconnect from the remote end.
Callers can access an extension to use DISA PIN Authentication-Multiple Calls only if the extension has DISA
feature enabled in its Class of Service.
DISA with CLI Authentication - Multiple Calls

The system authenticates the caller by matching the caller’s CLI with the entries of the DISA-CLI Authentication
Table and logs the caller in to the extension configured as ‘Auto Login’ extension for the CLI.
Callers are not required to dial any DISA Login Code or any password.
When a caller is authenticated on the basis of CLI, the system plays the ('internal' system) Dial Tone to the caller.
The callers must dial special digits or codes to go On-hook, Off-hook. They are allowed to make as many trunk calls
and internal calls for as long as they remain logged in to the DISA mode.

To end the DISA login session, callers must dial the Termination code or disconnect from the remote end.
For this type of DISA, the DISA CLI Authentication Table must be configured first.
DISA with CLI Authentication - One Call

This type of DISA is similar to the previous one. The system authenticates callers by matching the callers’ CLI with
the entries of the DISA-CLI Authentication Table and logs the callers in to the extension designated as ‘Auto Login’
extension for the CLI.
When the caller is authenticated on the basis of CLI, the system gives the caller direct access to the Outgoing
Trunks selected for TAC-1 for the current time zone (working hours, break hours, non-working hours)’ in the
“Station Basic Feature Template” assigned to the Auto Login extension. It plays the dial tone.
Callers are allowed to make a single external call. The system ends the DISA session on the completion of the call
by the caller or by the other remote party.
For this type of DISA, the DISA CLI Authentication Table must be configured first.
To make another call, the caller must enter the DISA mode again, by calling the SARVAM UCS from the remote
location.
DISA with CLI Authentication - One Call Answer Signaling is generally used when SARVAM UCS is deployed in the
Gateway Mode, where SARVAM UCS is configured to send an answer signal to the caller/calling device, receive
the DTMF digits dialed by the caller/calling device and dial out the digits dialed by the caller/calling device. To know
more, see “Gateway Application-Answer Signaling”.
How it works
For this feature to work, you must enable the desired DISA variant on the desired trunks: CO, Mobile, SIP,
T1E1PRI, BRI.
• A call lands on a DISA enabled Trunk.
• The system checks if a DISA variant is enabled on the trunk for the current time zone, that is, working
hours, break-hours and non-working hours.
• If a DISA variant is enabled on the trunk, the system processes the call according to the DISA variant
enabled on the trunk.
• If DISA with PIN Authentication - Multiple Calls is enabled,
• The system plays Welcome Greeting message to the caller248.
• The caller must dial the DISA Login Code consisting of:
• the DISA Feature Access Code.
• the number of the extension the caller wants to access.
• the user password of the extension.
• On successful login, the system starts the DISA Idle State Timer (configurable; default: 20 seconds).
The system waits for the caller to go Off-hook249.
248. If no voice message is recorded, the system plays music-on-hold to the caller.
249. If the caller does not go Off-hook within this timer, the system releases the call.

• When the caller goes Off-hook by dialing the Off-hook code #1, the system plays the internal dial tone
and waits for the caller to dial digits.
• If the caller dials an external number using a CO trunk, the system starts the DISA Inactivity Timer
(configurable; default: 2 minutes)250.
• The system waits for the caller to dial digits within the DISA Inactivity Timer.
• The system reloads this timer each time it receives digits from the caller. If the caller fails to dial any
digit within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed;
15 seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA
session. If digits are received before the end of the Warning Beeps, the system reloads the DISA
Inactivity Timer.
• The caller can make as many trunk calls and internal calls as the caller wants.
• The caller can terminate the DISA login session either by disconnecting from the remote end or by
dialing the Termination Code #9.
• If DISA with CLI Authentication - Multiple Calls is enabled,
• The system compares the CLI of the caller with the Calling Party Numbers configured in the CLI
Authentication Table.
• If the CLI matches with any of the Calling Party Numbers in the Table, the system provides access to
the extension configured as Auto Login extension for this Calling Party Number in the Table251.
• The caller gets logged into the Auto Login extension and gets the dial tone of SARVAM UCS.
• At the end of the call, the caller dials the On-hook code #0 to go On-hook. To make another call, the
caller dials Off-hook code #1 and dials the desired number. Thus the caller dials the On-hook and Off-
hook codes to make as many trunk and internal calls as desired.
(configurable; default: 2 minutes)252. The system waits for the caller to dial digits within the DISA
Inactivity Timer.
• The system reloads this timer each time it receives digits from the caller. If the caller fails to dial any
Inactivity Timer.
• If DISA with CLI Authentication - One Call Answer Signaling is enabled,
• The system compares the CLI of the caller with the Calling Party Numbers configured in the CLI
Authentication Table.
250. DISA Inactivity Timer is not applicable for T1E1PRI lines, BRI lines, SIP and Mobile trunks.
251. If no match is found for the CLI of the caller in the Table, the call will be routed as per the Incoming Call Routing configured in SAR-
VAM UCS.
252. DISA Inactivity Timer is not applicable for T1E1PRI line, BRI, SIP and Mobile trunks.

• If the CLI matches with any of the Calling Party Numbers in the Table, the system provides access to
the extension configured as Auto Login extension for this Calling Party Number in the Table253.
• The caller gets logged into the Auto Login extension and gets dial tone of the outgoing trunks selected
for TAC-1 for the current Time Zone (working hours, break hours, non-working hours).
(configured; default: 2 minutes).
• The system waits for the caller to dial digits within the DISA Inactivity Timer. If the caller fails to dial any
Inactivity Timer.
• After the external call is completed, that is, the caller disconnects from the remote end or the other
remote called party has disconnected, the caller is logged out.
• To make another external call, the caller must call the DISA enabled trunk of SARVAM UCS again.
In all the variants of DISA, the caller can use all the features allowed in the “Class of Service (COS)” of the
extension the caller is logged in to (using PIN Authentication or CLI Authentication).
• DISA calls in the SMDR report are marked as "O" in the remarks column. See “Station Message Detail
Recording-Report”.
• If DISA is disabled, SARVAM UCS will route the call by Auto Attendant logic, if Auto Attendant is
enabled.
• If DISA and Auto Attendant both are disabled, the incoming call will be routed as per the incoming call
routing configured. To know more, see “Auto Attendant”.
WARNING! This feature allows access to system resources to remote users, and therefore has serious
implications for your system's security. There is a risk of fraudulent calls being made from your system, if a
third party comes to know the authentication PIN or the User Password of an extension number. The cost
of such fraudulent calls will have to be borne by the owner of SARVAM UCS.
To avoid unauthorized access, we recommend you to change the PIN regularly. Make sure it is strong and
is provided to users who need to access the system using DISA only.
• If both, Built-In Auto Attendant and DISA are enabled on the trunk, SARVAM UCS supports all types of
DISA.
• If both, VMS Auto Attendant and DISA are enabled on the trunk, SARVAM UCS supports only PIN
Authentication-Multiple Calls. To know how the VMS handles a DISA call, see “VMS DISA Login”.
253. If no match is found for the CLI of the caller in the Table, the call will be routed as per the Incoming Call Routing configured in the
SARVAM UCS.

How to configure
To provide DISA to remote users you need to do the following configuration:
• Select the DISA variant for the Trunks on which you want to apply this feature in their “Trunk Feature
Template”.
• Enable DISA in the “Class of Service (COS)” of the extensions which you want to allow callers to access
using DISA.
• Change the User Password of the DISA extensions, if you selected DISA PIN Authentication-Multiple
Calls. If you selected DISA PIN Authentication-Multiple Calls on a trunk, the default User Password (1111)
will not work. See “User Password” and “System Security” more information and instructions.
• Configure the related timers, DISA Idle State Timer and DISA Inactivity Timer, if required. See “System
Timers and Counts” for instructions.
• If you have selected the DISA CLI Authentication-Multiple Calls or CLI Authentication-One Call Answer
Signaling on a trunk, you must configure the CLI Authentication Table.
Configuring DISA CLI Authentication Table using Jeeves
To configure the DISA CLI Authentication Table,
• Make a list of remote users and their numbers whom you want to allow DISA.
• For each remote user’s number on your list, write the Extension number of the SARVAM UCS you want to
allow this extension user to log in.
• Open Jeeves.

• Under Configuration, click DISA - CLI Authentication. The CLI Authentication Table page opens.
• You can configure as many as 999 numbers in this table, by clicking the tabs of the index on the top of the
table.
• Refer to the list of remote user numbers and the corresponding SARVAM UCS extension numbers you
made.
• In the Calling Party’s Number column, enter the number of the remote users whom you want to allow
access to DISA using CLI Authentication. The system will match the CLI of the callers with the numbers
you store here.
• For each Calling Party Number, in the Auto Login as field, select the extension Port Type (SLT, DKP, SIP
Extension, ISDN Terminal, Virtual Extension) and Port Number you want to allow access to after the
Calling Party Number is authenticated.
• Click Submit button to save your entries.
Configuring DISA CLI Authentication Table using a Telephone

To enter Calling Number in the CLI Authentication Table, dial:
• 4111-Index-Calling Number-#*

Where,
Calling Number may contain a maximum of 16 digits.The allowed digits are 0-9, #, *, A, B, C, D, +. Use
following codes to enter these digits:
Special Digit Code
A #4
B #5
C #6
D #7
+ #8
* **
# ##
To assign Port Type and Port Number, dial:

• 4112-Index-Port Type-Port Number
Where,
Port Type and Port Number are:
Port
Port Type Meaning
Number
00 000 None
01 001 to 512 SLT
02 001 to 128 DKP
28 01 to 64 ISDN Terminal
34 001 to 999 SIP Extension
36 01 to 64 Virtual Extension
For example, to configure extension '3001', which is a DKP with port number 001, as auto login station in
Index 001 of the Table, you must dial 4112-001-02-001.
• Exit SE mode.
How to use
If you are a Remote user, to be able to use DISA, you must know:
• the number of the Trunk on which DISA is enabled and the variant of DISA enabled on this trunk.
• the number of the extension and the user password which you want to access, if using DISA with PIN
Authentication.

• the duration of the DISA related Timers: The DISA Idle State Timer and the DISA Inactivity Timer, so that
you may dial digits accordingly, without delay.
• the special digits to be dialed during a DISA login session.
If you are a Remote user, to be able to use Authority Codes via DISA, you must know:
• the number of the Trunk on which DISA is enabled and the variant of DISA enabled on this trunk.
• your Authority Code and Password.
• the duration of the DISA related Timers: The DISA Idle State Timer and the DISA Inactivity Timer, so that
you may dial digits accordingly, without delay.
• the special digits to be dialed during a DISA login session.
For more information on Authority Codes, see “Authority Codes”.
Dialing Special Digits

After successful login, you will be required to go on-hook, go off-hook, use 'flash', use 'pause' or dial characters like
A, B, C, D, from your remote device to use the features and facilities of SARVAM UCS.
However, SARVAM UCS will not be able to understand the conventional way of dialing 'flash' key or going on-hook
with momentary make/break of loop current. Therefore, SARVAM UCS supports specific codes for specific
activities. If these codes are received during a DISA session, SARVAM UCS interprets it and performs the
associated activity.
When you are in DISA mode, use the following codes to indicate an activity:
Special Digit/activity Code to be dialed
on-hook #0
off-hook #1
Flash #2
Pause #3
A #4
B #5
C #6
D #7
+ #8
To Terminate the DISA #9
# ##
End of String #*
To program # when in
####
SE Mode

To use DISA,
• Dial the number of the Trunk on which DISA is enabled for the current time zone, Working, Break, Non-
working hours.
• SARVAM UCS answers the call. You will get music or Built-In Auto Attendant Voice Message, if
configured.
If DISA PIN Authentication-Multiple Calls is enabled,

• You will get beeps at the end of music/Built-In Auto Attendant Voice Message.
• Dial DISA Login Code 1079 during the beeps.
• Dial the extension number. You will get beeps.
• Dial the User Password for the extension during the beeps.
• You get feature tone on successful login.
• Dial the special digits #0 to go On-hook and then #1 to go Off-hook, you get the dial tone. Dial the number
to make your calls like any local extension. For example, making an external call, dial Trunk Access Code
'0' to grab a trunk and dial the number.
• To terminate DISA session dial #9 or disconnect the call from your (remote) end.
If DISA CLI Authentication-Multiple Calls is enabled,

• You will get system dial tone at the end of music/Built-In Auto Attendant Voice Message.
• Dial the special digits #1 to go Off-hook, #0 to go On-hook, and make your calls like any local extension.
• To terminate DISA session dial #9 or disconnect the call from your (remote) end.
If DISA CLI Authentication - One Call Answer Signaling is enabled,

• You will get Trunk dial tone at the end of music/Built-In Auto Attendant Voice Message.
• Dial the external number, without the Trunk Access Code.
• The DISA session will be terminated when you or the remote called party disconnects.
• To make another call, you must dial the number of the DISA enabled trunk again.
The features listed below are not supported in the DISA mode.
• Auto Call Back
• Auto Redial
• Call Park
• Call Chaining
• Self Ring Test
• Live Call Supervision

Direct Station Selection Console
The Direct Station Selection (DSS) Console is an add-on module with tri-color LEDs for the Digital Key Phones
(DKP) and SPARSH VP510. It provides you quick access to Stations, Trunks, Features/Functions of SARVAM
UCS; also making calling operations easy.
While the DSS Console is generally used by the Operator/receptionist in an organization, it is also meant to be
used by anyone who needs to access various features of SARVAM UCS at a touch of a single key.
Matrix offers two types of DSS Consoles:

• DSS64: This can be attached only with EON48 and EON310.
• DSS532: This can be attached only with EON510 and SPARSH VP510.
Maximum 32 DSS Consoles (DSS64 or DSS532) are supported by SARVAM UCS.
DSS64
A maximum of two DSS64 can be attached to a single DKP. Each DSS Console occupies a Digital Key Phone Port.
For example, if you attach two DSS consoles to a single DKP, three DKP ports would be occupied.The DSS
Console can be attached with SARVAM UCS in the same way as the DKP and is programmed as an attachment of
the DKP.
Each DSS Console that is attached to a DKP occupies a DKP port. Hence, the more DSS Consoles you attach to
DKPs, the lesser number of DKP ports will be available on SARVAM UCS.
When a single DSS64 is attached with a DKP, the DSS keys of the DKP as well as all the 64 keys of the DSS64 can
be used. Similarly, if two DSS64 are attached to a DKP, 128 additional keys are at your disposal to be used as DSS
keys.
To install and configure the DSS Consoles, refer “Installing DSS64” and “Programming DSS Console Keys”.

DSS532
A maximum of four DSS532 can be attached to EON510 or SPARSH VP510. Unlike DSS64, the DSS532 does not
occupy a DKP port. Although DSS64 and DSS532 work in a similar manner.
With each DSS532, 32 additional keys are at your disposal to be used as DSS keys. When all four DSS532 are
connected, you get 128 additional DSS keys at your disposal to be used.
For instructions:
• to install the DSS532 with EON510, see “Installing DSS532 with EON510”.
• to configure the DSS keys of the Console, see “Programming DSS Console Keys”.
DSS Keys
You can assign Station numbers or features/functions to the keys on the DSS Console, so that they can be
accessed easily simply by pressing a single key.
LEDs
Each DSS Key is equipped with an LED which glows to indicate the status of the Trunk/Extension or Feature
assigned to it.
Status of Extensions and Trunks
The LED of DSS keys assigned to Extensions/Trunks glow in three colors to indicate status of the call event on the
Extensions/Trunks and on the phone.
Thus, the status of user's own Extension, status of other Extensions and status of the trunk lines are indicated by
the LED of the DSS keys assigned to those Extensions and Trunks on the phone.
LED LED Mode

with. hold. from which you are being called.

LED LED Mode
Extension that is now busy with Extension which has put another Extension/Trunk that is called or
another Extension/Trunk. Extension/Trunk on hold. being called by another.
Violet You are talking on a Trunk You have held a Trunk (external You have an incoming call on
(external call) call) the Trunk (external call)
• Blue indicates the state of the extension/trunk you access. For example, when you make a call to another
Extension 203, the LED of the DSS key assigned to Extension 203 blinks Blue to indicate ringing at the
Extension. If you have successfully established speech with Extension 203 the LED glows Blue
continuously.
• Red indicates the state of other Extension/Trunks. For example, if the LED of the DSS key assigned to
Extension 201 is glowing Red continuously, it means Extension 201 is busy with another Extension or
Trunk.
• Violet indicates the state of the trunk you are in speech with. For example, when you are in speech on an
outgoing call on Trunk 1, the LED of the DSS Key assigned to Trunk 1 will be continuously ON. When you
put the call on hold, the LED will blink slowly.
Status of Features
The LED of a DSS key is activated when the feature assigned to this key is used.
you assign this feature to a DSS key, the LED of the key remains inactive, when Call Pick-Up is accessed.
Feature like Auto Redial requires an LED to show that it has been set or canceled. So, the LED of the DSS key to
which the Auto Redial feature has been assigned will glow Red, when Auto-Redial is set, and the LED is turned off
when the feature is canceled.

Distinctive Rings
What's this?
Distinctive Rings are ringing patterns used for distinguishing between different types of call events.
SARVAM UCS supports the following types of call events:
1. Internal Call
2. Priority Internal Call
3. External Call
4. Alarm Call
5. Auto Call Back Call
6. Auto Redial Call
7. Message Wait Call
8. SE Mode (Programming Ring)
9. Operator Alarm
10. Emergency
11. Self Ring
12. Call Supervision
13. Presence
14. Emergency Conference
15. Conference
With Distinctive Rings, it is possible to use ring cadence of user's choice for each of these call events. For instance,
Triple ring can be set for 'Priority Internal Calls' and long rings can be set for 'Alarm Calls'.
A set of ring types is called Distinctive Ring type. The default Distinctive Ring Types are:
Call Event Ring Type - T1 Ring Type - T2 Ring Type- T3
Internal Call Short Very Slow Double Double
Trunk Call Double Long Slow Long Slow
Auto Call Back Short Slow Short Slow Short Slow
Auto Redial Long Very Slow Long Very Slow Long Very Slow
Self Alarm Long Fast Long Fast Long Fast
Emergency Long Fast Long Fast Long Fast
Operator Alarm Long Fast Long Fast Long Fast
Message Wait
Short Fast Short Fast Short Fast
Call
Programming
Continuous Continuous Continuous
Ring
Self Ring Short Slow Short Slow Short Slow
Priority Triple Triple Triple
Call Supervision Continuous Continuous Continuous

Call Event Ring Type - T1 Ring Type - T2 Ring Type- T3
Presence Continuous Continuous Continuous
Emergency
Triple Triple Triple
Conference
Conference Triple Triple Triple
These ring types have the following ring cadence:
Ring Pattern Cadence (in milliseconds)
Short Fast 750-750
Short Long 500-1500
Short Very
750-2250
Slow
Long Fast 1500-500
Long Slow 1000-4000
Very Long Slow 2000-4000
Double 400-200-400-2000
400-200-400-200-400-
Triple
2000
Ring cadence is not programmable.
Distinctive Rings on SIP Extensions

On SIP Extensions, SARVAM UCS supports Distinctive Rings using Alert-INFO field in the INVITE message. To
indicate the different call events, SARVAM UCS sends the Ring Text for the respective Ring Type.
The default Distinctive Ring Types and their corresponding Ring Texts are given below.
Call Events Ring Type - T1 Ring Text
Internal Call Short Very Slow internal
Trunk Call Double external
Auto Call Back (ACB) Short Slow acb
Auto Redial (AR) Long Very Slow autord
Self Alarm Long Fast selfalram
Emergency Long Fast emergency
Operator Alarm Long Fast opratoralarm
Message Wait Short Fast msgwait
Programming Ring Continuous prog
Ring Test Short Slow test

Call Events Ring Type - T1 Ring Text
Priority Triple priority
Call Supervision Continuous callsup
Presence Continuous presence
Emergency Triple emergencyconf

Conference
Conference Triple conf
The Ring Text is sent in the Alert-INFO field of the INVITE message and the corresponding Ring Type is played on
terminal registered as the SIP Extension, if the terminal supports Distinctive Rings.
The Ring Text is programmable. You can change the Ring Text, if required.
How to configure
At the time of installation, when you select the “Region” (as per the geographical location of the system), and set
the system to default, SARVAM UCS loads the country-specific Distinctive Ring Type defined for the selected
Region.
Refer the topic “Default Settings” for the default Distinctive Ring Type applied to your country/region.
However, if required, you can change the default Ring Pattern and the Ring Text (for SIP Extensions) loaded by the
system.
When you change the Ring Texts for the Ring Types in SARVAM UCS, you must configure the same Ring Text in
the SIP phones.
Programming Distinctive Rings using Jeeves


• Click Distinctive Rings to expand.
• Select the desired Ring Pattern (Ring Type) for each call event that you want to customize.
• You can also customize the Ring Text of each call event on SIP extensions. The text can be a maximum of
20 alphanumerical characters.
• Log out of Jeeves or continue programming as required.
Programming Distinctive Rings using a Telephone

To change Ring Pattern for the call event, dial:

• 4002-Event-Ring Pattern
Where,
Event is:
01 for Internal Call

02 for Trunk Call
03 for Auto Call Back Call
04 for Auto Redial Call
05 for Self Alarm Call
06 for Emergency
07 for Operator Alarm
08 for Message Wait Call
09 for Programming Ring
10 for Self Ring
11 for Priority Call
12 for Call Supervision
14 for Presence

15 for Emergency Conference
16 for Conference
Ring Pattern is:
00 for OFF
01 for Continuous
02 for Short Fast
03 Short Slow
04 Short Very Slow
05 Long Fast
06 Long Slow
07 Long Very Slow
08 Double
09 Triple
To default ring patterns of events:

• Dial command 4001
• Exit SE mode.
Demonstration of rings
It is possible to demonstrate Ring Types to users by dialing the SE commands on DKP, Extended IP Phone and
SLT extensions of SARVAM UCS.
By default, the system will play each Ring Type as demonstration for 30 seconds.
Users of SARVAM UCS may be acquainted with the different Distinctive Rings played by the system so that they
can associate the terms used to describe the rings with the sound emitted by the system for each ring.

• Enter SE mode.
• Dial command 4003-Ring Pattern

Where,
Ring Pattern is
01 for Continuous
02 for Short Fast
03 for Short Slow
04 for Short Very Slow
05 for Long Fast
06 for Long Slow
07 for Long Very Slow
08 for Double
09 for Triple
• After dialing the desired Ring Pattern Code,
• Press ‘Enter' key.

• You get the prompt 'Go Idle for Ring' on your phone display.
• Go Idle.
• The Ring Pattern you selected will be played.
• You may repeat the SE commands to demonstrate another Ring Pattern.
• Exit SE mode.
For SLT Users

• Enter SE mode.
• Dial command 4003-Ring Pattern

Where,
Ring Pattern is
01 for Continuous
02 for Short Fast
03 for Short Slow
04 for Short Very Slow
05 for Long Fast
06 for Long Slow
07 for Long Very Slow
08 for Double
09 for Triple
• Go Idle, after dialing the desired Ring Pattern Code.
• The Ring Pattern you selected will be played.
• You may repeat the SE commands to demonstrate another Ring Pattern.
• Exit SE mode.

Do Not Disturb (DND)
What's this?
Extension users may restrict calls to their extensions in order to work uninterrupted by frequent phone calls. The
feature, Do Not Disturb, enables users accomplish this. This feature is useful to extension users who are in the
middle of a meeting or any important work that requires their undivided attention.
Using DND users can restrict—all calls, internal calls or external calls. However even if DND is set, users can route
their incoming calls to an Intercept Destination. This destination can be the extension users own mailbox or another
extension. In this way, extension users can ensure that they do not miss any important calls.
If required, when DND is set, a Stuttered Dial Tone can be played to the user for notification.
DND can be set and canceled by

• Extension users
• Operator for extension users, referred to as DND-Remote.
Doing so, calls — all, internal or external— will be barred. However, the extension user would continue to receive:
• Alarm calls.
• Reminder calls.
• Auto Call Back calls (Auto Callback as well as Auto Redial)
• Emergency Reporting Calls.
Also, the extension user can:

• use all the features of the System.
• make Outgoing calls and
• make Internal calls to other extensions.
DND has two supplementary features— DND-Override and Privacy from DND-Override.
The 'Do Not Disturb' feature bars calls to the phone on which DND is set. The 'DND-Override' feature breaks this
bar and allowing the calls to land on the phone on which DND is set. Protection is also given to the phone on which
DND-Override is attempted. If the phone on which DND-Override is attempted has 'Privacy from DND-Override'
enabled, the calling phone shall not be able to Override the DND.
When a caller calls a phone on which DND is set, he/she gets Routing tone (Feature tone). The caller can dial
DND-Override code. On dialing DND-Override code the call is placed on the called phone and the called phone
starts ringing.
The DND-override feature works only if the calling phone has 'DND-Override' feature enabled in its CoS group.
DND-Override will not work if the called phone has ‘Privacy from DND-Override’ enabled in its Class of Service or if
the called extension has opted for intercept routing.
So, using DND-Override feature, the users can be reached in case of some Emergency despite the DND set on the
phone.

• DND when set/canceled from the SA mode, will not depend on the assigned CoS.
• The system supports only single-point DND with Intercept Destination, which means, if the destination
extension has also set DND with Intercept Destination, the call will not follow the forwarding path.
How it works
• you must select the “DND Call Type”.
• you may select the “Intercept Destination for DND”.
• you may select the “DND Text Message” as per your requirement.
• you may assign a voice module for DND Notification. See “Voice Message for DND Notification”.
DND Call Type

The extension user/Operator can select the type of calls to be restricted while enabling DND. They can select either
All, Internal or External Calls.
Intercept Destination for DND

If the user wants that the calls are attended to even if DND is set, System Engineer must configure the Intercept
Destination for the user. Incoming calls landing on the extension that has set DND will be routed to the Intercept
Destination. This destination can be the users own mailbox or another extension (SLT, DKP, SIP).
DND Text Message

A DND Message is a short Text Message such as 'In a Meeting', 'In a Conference', 'On Vacation'.
When setting DND (also DND-Remote), the extension user/Operator can select an appropriate text message to be
displayed to the calling extension.
This DND text message is displayed on the calling extension, only if the calling extension is the proprietary digital
key phone, EON or an Extended IP Phone.
The SARVAM UCS supports 9 different DND Text Messages, out of which 8 messages can be changed as per
user requirement by the System Engineer. User can select and set on their phones any of the DND messages
programmed by the System Engineer.
Voice Message for DND Notification

Using this feature, a pre-recorded Voice Message can be played to the caller informing him/her about the DND set
on the called extension. For example, "The dialed extension has activated Do Not Disturb".
When DND is set on an extension, callers who try to reach that extension will be played an error tone. Callers who
are using EON/Extended IP Phones are displayed the DND Text Message set by the called extension, and thus
come to know the cause of the error tone. Such a facility is not available to callers who are using SLTs, who can
hear only the error tone and have no way of knowing the cause of the error tone.
Using Voice Message for DND Notification, a pre-recorded Voice Message can be played to the callers to notify
them of the DND set on the called extension.

You must record and assign a Voice Module to play the pre-recorded voice messages as DND Notification to the
callers.
Let us understand this feature with an example:
A, B and C are extension users.

B has EON, while C has an SLT.
B has DND-Override in his Class of Service, C does not have this feature.
DND Text messages as well as Voice Message Notification for DND have been programmed by the System
Engineer.
• A has set DND on his extension with the DND Text message 'In Meeting'254.
• B calls A.
• As B has DND-Override, the Voice Message for DND Notification is played to B once, and the DND
message 'In Meeting' set by A appears on B's phone display. B gets routing Beeps.
• To exercise DND-Override, B must dial '4' the feature access code for 'DND-Override' during either during
the Voice Message or during the routing Beeps.
• B gets Ring Back Tone, if A's extension is free.
• B gets Busy Tone, if A's extension is busy.
• However, if A has Privacy from DND Override, B will get error tone and the DND message set by A
appears on B's phone.
If B fails to dial the DND-Override code before the end of the routing beeps, error tone will be played to
him.
Now, to take another example,
• C calls A.
• As C has an SLT, C will get only the Error tone.
• But as Voice Message for DND Notification is programmed in the system, C will be played the pre-
recorded message once.
• Since C is not allowed 'DND-Override' in his Class of Service, he cannot exercise this feature during the
Voice Message.
• At the end of the voice message, C will be played error tone.
In the above examples, if A has set E’s extension as the Intercept Destination, then calls from B and C will be
routed to E’s extension.
Call Forward: When DND and Call Forward-Unconditional are set on an extension, Call Forward is given priority. If
any other type of Call Forward and DND are set on an extension, DND is given priority. However, if DND with
Intercept Destination is set, it will not work. If an extension has set both Call Forward and DND, then Feature Tone
will be played to the extension user.
If an extension A has set DND with Intercept Destination as E, incoming calls on A’s extension will be routed to E’s
extension.
If E sets Call Forward or DND with Intercept Destination as B, incoming calls from A’s extension will be still routed
to E’s extension. Only incoming calls on E’s extension will be routed to the B’s extension.
254. While DND and DND Text Message can be set from any phone, DND Text Message can be viewed on EON /Extended IP Phone
only.

How to configure
For this feature to work, 'DND' must be enabled in the Class of Service of the group of the extensions which is to be
Also, 'DND-Override' and 'Privacy from DND-Override' can be enabled in the Class of Service of the extensions to
whom this features are to be provided.
Besides these, the System Engineer may program the DND Text Message, Stuttered Dial Tone when DND is set,
Intercept Destination for DND and the Voice Message for DND Notification, as per user requirements.
The user can then select the type of calls for which he wants to set DND.
Programming DND in Class of Service

In the default Station Basic Feature Template 01 assigned to all extensions of the SARVAM UCS, the default CoS
group 01 has DND enabled. So, all extensions of SARVAM UCS can set and cancel DND. DND-Override and
Privacy from DND-Override are disabled in the default CoS group 01. So, none of the extensions can use DND-
Override, or be exempt from DND.
While it makes sense to offer all extensions DND, providing DND-Override and Privacy from DND also to all
extensions will not serve the purpose of DND.
Decide which extensions are to be allowed 'DND', which are to be allowed 'DND-Override', and which are to be
allowed 'Privacy from DND-Override'. Generally, DND-Override is allowed to the Operator extension. It may be
allowed to extensions of persons in senior positions in the organization. Similarly, Privacy from DND-Override may
be allowed to persons in senior positions in the organization.
If you want to allow DND to all extensions, retain the default CoS group 01 in Station Basic Feature Template 01.
However, if you want to allow DND only to selected extensions, disable this feature in the default CoS group 1.
Now, to assign DND to selected extensions, follow these steps:
1. Define a CoS group with DND enabled.

2. Prepare a Station Basic Template with this CoS group applicable in all the time zones.
3. Assign this newly prepared Station Basic Feature Template to the extensions on which 'DND' is to be
enabled.
Similarly, if 'DND-Override' is to be to be allowed to the Operator and a few other extensions, follow these
steps:
1. Define a CoS group with DND-Override enabled. If DND is also to be allowed, enable both DND-Override
and DND in this CoS group.
3. Assign this newly prepared Station Basic Feature Template to the Operator extension on which 'DND-
Override' is to be enabled.
Repeat the same steps to allow 'Privacy from DND-Override' to selected extensions. For extensions that are to be
allowed 'DND' as well as 'Privacy from DND-Override', enable both features in the CoS group in the Station Basic
Feature Template applied on these extensions.
Similarly, for extensions that are to be allowed 'DND', 'DND-Override' and 'Privacy from DND-Override', enable all
three features in the CoS group that you prepare for these extensions.

Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for detailed programming
instructions on how to prepare a CoS in the Station Basic Feature Template and how to apply this template on
extensions.
Programming DND Text Messages
By default, 9 DND Text Messages are programmed in the SARVAM UCS as listed below:
Message # DND Message
1 Do Not Disturb
2 Unavailable
3 In a Meeting
4 In a Conference
5 Try on Mobile
6 On Vacation
On Business
7
Trip
8 Out of Office
9 With a Guest
You can use these default message options or program messages from 2 to 9 as per user preferences.
Programming DND Messages using Jeeves

• Click DND Text Message to expand.
• All the default text messages appear in the DND message field. You may change the DND text messages
2 to 9 as per your requirement. Click the field and enter your custom DND text message.
Programming Stuttered Dial Tone

If required by users, then the system will play Stuttered Dial Tone on the extension when DND is set.
For instructions, see “System Parameters”.
Programming Intercept Destination for DND

By default, the Station Advanced Feature Template 01 is assigned to all extensions of the SARVAM UCS and in
this template the Intercept Destination is configured as None.
You can either configure the Intercept Destination in this template or you may select another template and
customize it as per your requirement. For instructions see “Station Advanced Feature Template”.
Programming Voice Message for DND Notification

To be able to play a voice message to callers for DND notification, you must first record a Voice Module with the
desired message.
ETERNITY LENX/MENX supports 16 Voice Modules and ETERNITY GENX supports 15 Voice Modules. The
duration of each module can be 16 seconds.
Record a Voice Module with the message “The dialed extension has activated Do Not Disturb” (recommended).

Assign the Voice Module to the Voice Message Application number 44 defined for 'DND Notification'. Refer the
topic “Voice Message Applications” for programming instructions.
How to use
DND set/canceled by Extension Users

To set DND:
• Press the 'DND' Key.
OR
• Dial 18
• Scroll to select the type of call:
• All calls
• Internal calls
• External calls

• You get a text message 'DND Set' on the phone's display and confirmation tone.
To select a DND Message
• Press the 'DND' Key.
OR
• Dial 18
• Scroll to select the Set DND Message option.
• The list of DND messages appear on the phone's display:
• Do Not Disturb
• Unavailable
• In a Meeting
• In a Conference
• Try on Mobile
• On Vacation
• On Business Trip
• Out of Office
• With a Guest
• Scroll to the desired option and press 'Enter' key.
• Go Idle or you get dial tone after the confirmation tone.
To cancel DND:
• Press the DND Key again.

• The following options appear on the phone's display
• All calls

• Internal calls
• External calls
• Cancel DND
• Select Cancel DND and press 'Enter' key.
OR
• Dial 18-0
• You get a text message 'DND Canceled' on the phone's display and confirmation tone.
For SLT Users

To set DND:
• Lift handset.
• Dial 18-1 to set DND for All calls
• Dial 18-2 to set DND for Internal calls
• Dial 18-3 to set DND for External calls
To select DND Message:

• Lift handset.
• Dial 18-4-1 for 'Do Not Disturb'
• Dial 18-4-2 for 'Unavailable'
• Dial 18-4-3 for 'In a Meeting'
• Dial 18-4-4 for 'In a Conference'
• Dial 18-4-5 for 'Try on Mobile'
• Dial 18-4-6 for 'On Vacation'
• Dial 18-4-7 for 'On Business Trip'
• Dial 18-4-8 for 'Out of Office'
• Dial 18-4-9 for 'With a Guest'
To cancel DND:
• Lift handset.
• Dial 18-0
DND-Remote

To set DND for an extension user:
• Press the DSS Key assigned to DND-Remote.
OR
Dial 1072-001.
• Enter the extension number.
• Scroll to select the type of call:
• All calls
• Internal calls
• External calls

• You get a text message 'DND Set on <Extension Number>' and confirmation tone.
• Go Idle or you get dial tone after confirmation tone.
To select a DND Message:

• Press the DSS Key assigned to DND-Remote.
OR
Dial 1072-001.
• Enter the extension number.
• Scroll to select the option Set DND Message
• The list of DND messages appear on the phone's display:
• Do Not Disturb
• Unavailable
• In a Meeting
• In a Conference
• Try on Mobile
• On Vacation
• On Business Trip
• Out of Office
• With a Guest
• Scroll to the desired option and press 'Enter' key.

• Go Idle or you get dial tone after the confirmation tone.
To cancel DND-Remote:
• Press the key assigned Remote-DND function.
OR
Dial 1072-001.
• Scroll to select the message 'Cancel DND'.
• You get a text message 'DND Canceled on <Extension number>' and confirmation tone.
• Go Idle or you get dial tone after confirmation tone.
For SLT Users

To set DND for an extension user:
• Dial 1072-001. You get feature tone.
• Dial Extension number. You get feature tone.
• Dial 1 for All Calls
• Dial 2 for Internal Calls
• Dial 3 for External Calls
To select DND Message:

• Dial Extension number. You get feature tone.
• Dial 4-1 for 'Do Not Disturb'
• Dial 4-2 for 'Unavailable'
• Dial 4-3 for 'In a Meeting'

• Dial 4-4 for 'In a Conference'
• Dial 4-5 for 'Try on Mobile'
• Dial 4-6 for 'On Vacation'
• Dial 4-7 for 'On Business Trip'
• Dial 4-8 for 'Out of Office'
• Dial 4-9 for 'With a Guest'
To cancel DND set for an extension user:

• Dial Extension Number. You get feature tone.
• Dial '0' to cancel DND.
DND-Override
For EON, Extended IP Phone Users and SLT Users

• Dial an extension number.
• You get routing beeps and a DND Notification message, if programmed (and a DND Text message, if
using EON/Extended IP Phone)
• Dial '4', the DND-Override Code, during the message or the routing beeps.
• The called extension will start ringing.
• You will get Ring Back tone.
• If the dialed phone is busy, you will get busy tone.

DSS Call Pick-Up
What's this?
DSS Call Pick-Up allows DKP/Extended IP Phone users to answer calls ringing on other extensions or incoming
calls on trunks by pressing the DSS Keys assigned to those extensions/trunks on their phones.
There are two types of DSS Call-Up:
• DSS Call Pick-Up-Station - internal or external calls ringing on any extension, can be picked-up by
pressing the DSS Key assigned to that extension on your phone.
• DSS Call Pick-Up-Trunk - incoming calls on any trunk for any extension can be picked-up by pressing the
key assigned to that trunk on your phone.
For T1E1 and BRI Trunks, DSS keys can be assigned for each channel. Similarly for SIP Trunks, DSS
keys can be assigned for each Call Appearance.
If you have assigned a DSS key to All the Channels for T1E1/BRI Trunks or All the Call Appearances for
SIP Trunks, you will only be able to grab the trunk to make outgoing calls. You will not be able to pick up
incoming calls on these trunks.
How it works
• enable the desired Call Pick-up in the COS of the extension user.
• assign DSS Keys with LED to the desired extensions/trunks on their DKP/Extended IP Phone.
This is how DSS Call Pick-Up works:
• Extension user A has configured a DSS Keys for extension 2007 and CO trunk 1 on his/her DKP/Extended
IP Phone.
• When a call lands on extension 2007 and it rings, the DSS Key assigned to 2007 blinks fast in Blue color
to indicate that the extension is ringing. A presses the DSS Key to pick-up the call ringing on extension
2007.
If DSS Call Pick-Up-Station is not enabled in the COS assigned to extension user A, the DSS Key blinks
fast in Red color to indicate that the extension is ringing. However, A will not be able to pick-up the call
ringing on extension 2007.
• Similarly when there is an incoming call on CO trunk 1, the DSS Key assigned to CO trunk 1 blinks fast in
Violet color to indicate that there is an incoming call on the trunk. A presses the DSS Key to answer the
call on the trunk.
If DSS Call Pick-Up-Trunk is not enabled in the COS assigned to extension user A, the DSS Key blinks
fast in Red color to indicate that the there is an incoming ringing call. However, A will not be able to pick-up
the incoming ringing call on that trunk.

• Call States: DSS Call Pick-Up-Station and DSS Call Pick-Up-Trunk are possible only when the calls are in
ringing state.
• Priority: If multiple calls are ringing on an extension, when you press the DSS Key assigned to that
extension, you will be connected to the ringing call with the highest priority. To know more, see “Priority”.
How to configure
To provide this feature to extension users, you must
• enable these features in their Class of Service to be assigned to the users. For instructions, see “Class of
Service (COS)” and “Station Basic Feature Template”.
• configure the DSS Keys for the desired extensions and trunks on their DKP/Extended IP Phone. To know
more about assigning DSS Keys, see “DSS Keys Programming”.
How to use
To use DSS Call Pick-Up-Station:
• When the DSS key assigned to the station blinks fast in blue color to indicate that the station is ringing,
press the DSS Key.
• You are in speech with the calling party.
• You may talk.
To use DSS Call Pick-Up-Trunk:
• When the DSS key assigned to the trunk blinks fast in violet color to indicate that there is an incoming call
on the trunk, press the DSS Key.
• You are in speech with the calling party.
• You may talk.

Dynamic Lock
What's this?
Dynamic Lock allows extension users to change the Toll Control Levels (Calling Permissions) of their extensions on
their own by dialing a code.
The System Administrator/Operator can also change the Toll Control Levels of extensions using Dynamic Lock.
With this feature, extension users can prevent misuse of outgoing call facility from their extensions, especially in
their absence.
Dynamic Lock also forms the basis of 'Call Privilege', which is feature of the Hotel Application of SARVAM UCS.
There are four types of Toll Control Levels, starting from Level 0 to Level 3 that can be set for extension phones.
For each Toll Control Level from 0 to 3, a 'Call Privilege’255 is to be assigned and corresponding numbers strings to
be allowed and number strings to be denied for each Call Privilege are to be programmed.
• Toll Control - Level 0 is Time Zone based, wherein the Call Privilege Type must be defined for each Time
Zone, that is, Working Hours, Break Hours and Non-Working Hours. For instance, you may define 'All
Calls' as Call Privilege for Working Hours, 'Local Calls' as Call Privilege for Break Hours and 'No Calls' as
Call Privilege for 'Non-Working' Hours.
By default, Call Privilege 'All Calls' is selected for all three Time Zones.
• Toll Control - Level 1 is not based on Time Zones. By default, the Call Privilege Type for this level is
'Local Calls'.
• Toll Control - Level 2 is not based on Time Zones. By default, the Call Privilege type set for this level is
'National Calls'.
• Toll Control - Level 3 is not based on Time Zones. By default, Call Privilege 'No Calls' is selected for this
level.
The Call Privilege for each of the above Toll Control Levels can be redefined according to user
requirements. For example, Toll Control Level 3 can be programmed for allowing all types of calls by
selecting 'All Calls' as Call Privilege Type and Level 0 can be programmed to allow only Local Calls, by
programming the strings of 'Local Numbers'.
Refer the feature description for “Toll Control” to know more.
Extension users who are allowed the Dynamic Lock feature in their Class of Service, can set the Toll
Control Level in two ways:
• Manually: the extension user changes the Toll Control Level of the extension whenever s/he wants by
dialing the feature access code.
255. The Call Privilege types are: No Calls, Local Calls, Regional Calls, National Calls, International Calls, All Calls and Limited Calls.

For example, an extension user having Toll Control Level 2 (default: National calls) can restrict long
distance dialing on his/her extension by setting the Toll Control Level to 1 (default: Local calls) before
leaving the workplace. On return, the user can restore the previous Toll Control Level, by setting it back
to Level 2.
Thus the extension user sets Dynamic Lock s/he manually selects the desired Toll Control Level for
his/her extension and restores the original Toll Control Level assigned to the extension.
• Automatically: the extension user changes the Toll Control Level of the extension using the Dynamic
Lock Timer. The user sets the Timer to the desired number of minutes. On the expiry of this Timer, the
system restores the original Toll Control Level assigned to the extension.
For example, an organization has defined Toll Control Level 0 as Local Calls, and Level 3 as All Calls.
An extension user of this organization is assigned Level 0. When this extension user wants to make
international calls, he sets the Dynamic Lock Timer and selects Toll Control Level 3. At the end of the
timer, Level 3 gets locked and Toll Control Level 0 is reapplied on the extension phone.
• The changing of Toll Control level requires the user to dial the 4-digit User Password. The system will
not accept the default User Password (1111). The extension user must first change the default User
Password.
• The Dynamic Lock Timer can be set to a maximum of 99 minutes.
• The Dynamic Lock Timer must be set to '00' when using Manual Dynamic Lock.
• Dynamic Lock when set/canceled from the SA mode, will not depend on the assigned CoS.
How it works
The Pre-requisites
• The Toll Control Levels 0 to 3 are programmed in the Station Basic Feature Template applied on the
extension.
• Dynamic Lock is allowed to the extension in its Class of Service.
The Process
For Dynamic Lock - Manual

• The user of extension A sets the Dynamic Lock manually by entering the User Password and selecting the
desired Toll Control Level.
OR
• The Operator sets Dynamic Lock manually for an extension by entering the extension number and
selecting the Toll Control Level.
For Dynamic Lock - Automatic

• The user of extension A sets the Dynamic Lock by entering the User Password, setting the Dynamic Lock
Timer, and selecting the desired Toll Control Level.

OR
• The Operator sets Dynamic Lock for an extension by entering the extension number, setting the Dynamic
Lock Timer, and selecting the Toll Control Level.
• Now, whenever a call is made from extension A, the system checks for Toll Control Level.
• The system then checks the associated Lists of allowed and denied numbers.
• If the Toll Control Level is 0, then Toll control is time zone based, that is, working hours, break hours
and non-working hours. The outgoing call is allowed/denied as per the Call Privilege and the
corresponding Allowed and Denied Number List programmed for that time of the day by the System
Engineer.
• If the Toll Control Level is 1, 2, 3 the outgoing call is allowed/denied as per the Call Privilege and the
corresponding number list programmed for each level.
• If Dynamic Lock - Automatic has been set by user/Operator, the system waits for the duration of the
Dynamic Lock Timer set for the extension. At the end of each outgoing call made during the period of this
Timer, the system will restart the Timer again. The system will change the Toll Control back to the previous
Level when no outgoing call is made till the expiry of this Timer.
• If Dynamic Lock - Automatic has been set by user/Operator, and an internal call is made during the period
of the Dynamic Lock Timer, the system will check for the 'Decrement Dynamic Lock Timer Internal Calls'
feature in the Class of Service of allowed to the extension. If this feature is enabled, the system will start
the decrement of the Dynamic Lock Timer. The system will change the Toll Control back to the previous
level on the expiry of this Timer. However, if the 'Decrement Dynamic Lock Timer' feature is disabled in the
Class of Service, the system will reset the Toll Control as described in the previous step.
• If Dynamic Lock - Manual has been set, the extension user/Operator must set the Toll Control Level back
to the previous Level.
• Redial and Auto Redial: The system will check for Toll Control Level when an extension on which
Dynamic Lock is set, attempts Redial. The system will not check the same for Auto Redial.
• Emergency Number Dialing: All extensions will be able to dial Emergency numbers always, regardless
of the Toll Control set on them.
SARVAM UCS provides for separate programming of Emergency Numbers, which remain unaffected by
Dynamic Lock set on the phones. Refer the topic “Emergency Dialing” to know more about this feature.
How to configure
For this feature to work, it must be enabled in the Class of Service of the extensions; Toll Control Level must be
programmed in the Station Basic Feature Template of the extensions. The user must change the default User
Password.
Dynamic Lock in Class of Service

In the default Station Basic Feature Template 01 assigned to all extensions of the SARVAM UCS, the default COS
group 01 has Dynamic Lock enabled. So, all extensions can set Dynamic Lock.

In the default COS group 01, 'Decrement Dynamic Lock Timer for Internal Calls' is disabled.
Retain the default template, if you want to allow this feature to all extensions and keep the Decrement Timer
disabled.
If you want to deny Dynamic Lock to all extension, simply disable this feature in the default COS group 01 of Station
Basic Feature Template 01.
If you want to allow Dynamic Lock and/or the Decrement Dynamic Lock Timer for Internal Calls only to selected
extensions, then follow these steps:
a. Define a new CoS group with Dynamic Lock and the Decrement Dynamic Lock Timer for Internal Calls
enabled.
b. Prepare a Station Basic Template with this CoS group applicable in all the time zones.
c. Assign this newly prepared Station Basic Feature Template to the extension on which 'Dynamic Lock' and
'Decrement Dynamic Lock Timer’ for Internal Call is to be allowed.
Similarly, if you want to deny Dynamic Lock/Decrement Dynamic Lock Timer to selected extension, prepare a new
Station Basic Feature Template with this feature disabled in the CoS group. Assign this feature to those extensions
which are to be denied this feature.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for programming instructions.
Toll Control Levels in Station Basic Feature Template

Program the Toll Control Levels in the Station Basic Feature Template that are assigned to the extensions which
are to be allowed the Dynamic Lock feature. Refer the topic “Toll Control” for instructions.
How to use
Dynamic Lock, Manual and Automatic, can be set by extension users as well as from their own extensions or from
the SA mode by the Operator.
The extension user/Operator must first set the Dynamic Lock Timer and then change the Dynamic Lock Level.
To set Dynamic Lock-Manual, the extension user/Operation must set the Dynamic Lock Timer to 00.
Recall that
• When the Dynamic Lock-Manual is set (Timer set to 00), the extension user/Operator must dial the
feature access code to restore the previous Toll Control Level.
• When Dynamic Lock-Automatic is set (Timer set to desired number of minutes), the system will restore
the previous Toll Control Level at the end of the Timer.
• The extension user must change the default User Password to be able to set the Dynamic Lock on his/
her extension. Refer the topic “User Password” for instructions on changing the password.

Changing Dynamic Lock by Extension Users

To set Dynamic Lock-Manual:
• Press DSS Key assigned to Dynamic Lock.
OR
• Dial 142.
OR
• Enter Phone Menu by pressing 'Enter' key.

• You get the prompt <Enter User Password>
• Enter your User Password.
• You get the prompt: <Lock Timer = XX Minutes>.
• Enter 00
• You get the Text message <Lock Timer = 00 Minutes> and confirmation tone.
To set Dynamic Lock-Automatic:
OR
• Dial 142.
OR

• You get the prompt <Enter User Password>
• You get the prompt: <Lock Timer = XX Minutes>.
• Enter the desired number of minutes (max. 99 minutes).
• You get the Text message <Lock Timer = xx Minutes> and confirmation tone.
To set Dynamic Lock Level:
OR
• Dial 141.
OR

• You get the prompt: <Enter User Password>.
• Select the desired Toll Control Level:

• Toll Control Level 0

• You get the text message 'OK!' and a confirmation tone.
If you are still working with the default User Password, the system will prompt you to 'Change User
Password' when you attempt to set Dynamic Lock. Change your User Password first, before you use this
feature.
For SLT Users


• Dial 142-User Password-00.
• You get Confirmation tone.

• Dial 142-User Password-Minutes (max. 99 minutes)
• You get Confirmation Tone.

• Dial 141-User Password-Toll Control Level.
• Dial 0 Level 0.
• Dial 1 for Level 1.
• Dial 2 for Level 2.
• Dial 3 Level 3.
• You get Confirmation tone.
Changing Toll Control Level for an Extension (SA mode)

• Press DSS key assigned to Dynamic Lock.
OR
• Dial 1072-002256.
256. Ensure that the feature 'SA Extension' is enabled in the “Class of Service (COS)” allowed to the extension from which this code is
being dialed.

OR
• Enter Phone Menu.

• You get a text message 'Enter Room/Phone Number'.
• Enter the extension number for which you want to set Dynamic Lock.
• Select the option 'Change Lock Timer'.
• You get the following prompt: 'XX Minutes'.
• Enter 00
• You get the text message <Lock Timer = 00 Minutes> and confirmation tone.
OR
• Dial 1072-002257.
OR

• Select the option 'Change Lock Timer'.
• You get the following prompt: 'XX Minutes'.
• Enter the desired number of minutes (max. 99)
• You get the text message <Lock Timer = xx Minutes> and confirmation tone.
OR
• Dial 1072-002258.
OR

• Select the option 'Change Toll Ctrl. Level'
• Scroll to select the desired Toll Control Level:
being dialed.
being dialed.

• You get the confirmatory text message 'OK!' and confirmation tone.
For SLT Users


• Dial 1072-002, you get feature tone.
• Dial Extension Number, you get feature tone.
• Dial 2, the code for 'Change Lock Timer'.
• Dial 00.
• Replace the Handset or you get dial tone after 3 seconds.

• Dial 2, the code for 'Change Lock Timer'.
• Dial the number of Minutes you want to set the Timer.
To change Dynamic Lock Level:

• Dial 1, the code for 'Change Toll Control Level'.
• Dial 0 for Level 0

E1/T1 Maintenance
What’s this?
The E1 Maintenance consists of Error Counts (Performance Statistics), Alarms and Loop Back Tests. This is as per
standards like G.704, G.706 and G.732. G.775 is also considered for detection of defect conditions like Loss of
Signal (LOS), Loss of Frame (LOF), Alarm Indication Signal (AIS), etc.
To elaborate, the Digital line can have transmission errors. All the errors will not generate an Alarm. Few severe
errors generate Alarms. However, all the errors are logged in the System Fault Log.
The SNIIC (Subscriber Network Interface Integrated Circuit), is used to interface E1 line to SARVAM UCS. It
supports error counters listed in the table given below. See “Error Counts (Performance Statistics)”.
Each error detected by the T1E1 port is sent to the CPU Card in the form of an event. The system counts these
errors and prepares a statistical record if the condition matches. For example, Severely Errored Seconds Count is
incremented when one OOF (Out of Frame) event detected by the system or more than 320 framing errors are
detected by the system. This statistical record is updated and maintained by the system.
Facility-Associated and Non-Facility Associated Signaling:

Signaling on ISDN PRI trunks consists of messages transported over the D-Channel, which is channel 24 on T1
interface or channel 16 on E1 interface. This signaling can be provided by two methods:
• FAS: The D channel can provide signaling for the other B channels on the same interface. This is called
‘Facility Associated Signaling’ (FAS).
• NFAS: The D channel can provide signaling for the other B channels on more than one interface. This is
called ‘No facility Associated Signaling’ (NFAS). The signaling arrangements, the capability is supported to
designate a D channel on one interface to be a backup to a D channel on another interface in case of
failure. This is called D channel backup.
Error Counts (Performance Statistics)
Error Counters supported by SNIIC in E1 Mode.
Errored Frame Alignment Signal This counter is incremented on receipt of each errored FAS.
E-bit This counter is incremented when either E1 or E2 bit is set in the transmit
frame.
CRC-4 Error This counter is incremented when the received frame has CRC-4 errors.
Line Code violation Error This counter is incremented when a line code violation error occurs.
Excessive Zeros Error This counter is incremented when excessive zeros are received or Line
code violation error occurs.
Positive Slip Buffer This counter is incremented every time a positive slip occurs.
Negative Slip Buffer This counter is incremented every time a negative slip occurs.

Following parameters form the statistical record. This can be generated in the form of a report as shown below:
Performance Parameter Seconds/Count
Error Seconds 000 to 255
Bursty Errored Seconds 000 to 255
Severely Errored Seconds 000 to 255
Severely Errored Framing Seconds 000 to 255
Unavailable Seconds 000 to 255
Positive Slip Seconds 000 to 255
Negative Slip Seconds 000 to 255
Loss of Frame Count 000 to 255
Line Errored Seconds 00000 to 65535
Excessive Zeroes Error Count 000 to 255
CRC-4 Error Count 00000 to 65535
Out Of Frame (OOF) - Out of Frame is the occurrence of a particular density of framing error events. OOF is
declared when three consecutive frame alignment signals have been received with an error. OOF ends when;
• In frame N, the FAS is correct.

• In frame N+1, the FAS is absent.
• In frame N+2, the FAS is present and is correct.
Errored Seconds - It is defined as a second with one of the following:
• One FAS Errors.

• One or more OOF defects.
• One or more Slip events.
• A detected AIS defect.
Bursty Errored Seconds
Without CRC - It is a second with:
• More than one but less than 320 Errored FAS.

• No OOF.
• No SES.
• No AIS.
With CRC - It is a second with:
• More than one but less than 320 CRC errors.

• No OOF.
• No SES.
• No AIS.

Severely Errored Seconds
With CRC - It is a second with one of the following:
• 832 or more CRC error events.

• one or more OOF defects.
Without CRC - It is a second with one of the following:
• 2048 or framing errors.

• Slips are not included. This is not incremented during Unavailable seconds.
Severely Errored Framing Seconds (SEFS) - It is a second with either one or more OOF defects or a detected
AIS defect.
Unavailable Seconds - It is defined as a second in which E1 service is unavailable. An unavailable state is

declared at the onset of 10 consecutive severely errored seconds and is cleared on onset of 10 consecutive
seconds with no severely errored seconds.
Positive Slip Seconds - It is defined as a second in which a frame is repeated to account for frequency drift
between ET2 and the network.
Negative Slip Seconds - It is defined as a second in which a frame is deleted to account for frequency drift
between ET2 and the network.
Loss of frame count - Loss of Frame is declared after 2.5 seconds of continuous loss of signal or OOF. LOF is
cleared after 10 seconds of continuous no loss of signal or OOF.
Line Errored Seconds - It is a second in which one or more than one line code violation error occurs.
Excessive Zeroes Error Count - This counter is incremented when excessive zeroes are received on the line or
when line code violation error occurs.
CRC-4 Error Count - This counter is incremented when a CRC-4 Error is detected. This is applicable for E1 ports
only.
CRC-6 Error Count - This counter is incremented when a CRC-6 Error is detected. This is applicable for T1 ports
only.
Parameters of Facility Data Link (FDL)
This parameter is applicable for T1 Ports only.
FDL is used for communicating general maintenance information or for transmitting user defined information within
the T1 link. General maintenance information is in the form of Performance Message Report which is generated by
the T1E1PRI Card and depending upon the T1 FDL Protocol, the Performance Message Report is sent every
second or on request.

How to configure
T1 FDL
T1 FDL can be enabled/disabled. This parameter is applicable only if Framing = ESF. If the Network (Public or
Private) to which the SARVAM UCS is connected does not support FDL then T1 FDL will be disabled.

6164-*-T1 FDL
Where,
T1 FDL Meaning
0 Disable
1 Enable
T1 FDL Protocol
SARVAM UCS will support both the protocols of reporting the performance monitoring. This parameter is applicable
only if T1 FDL is enabled and Framing = ESF. This parameter will match the protocol expected by the other end of
the link.
Where,
0 Disable
1 AT&T 54016
2 ANSI T1.403

ANSI T1.403
As per this standard, the receiving equipment transmits a performance report message (PRM) each second over
the FDL. This PRM is not sent to any specific remote location, but is broadcast so that any PRM receiving device
on the T1 line can intercept the message. The PRM contains error information pertaining to only the previous 4
seconds.
It is the responsibility of the PRM receiver to accumulate the information and store it for 24 hours or the time
desired. This method allows performance monitoring points at different locations along the T1 network so that error
localization is determined.
AT&T 54016
As per this standard, the receiving equipment collects the data but does not transmit it on its own based on time as
done by ANSI. Instead, the transmitting end sends a request to the receiving end to transmit the performance data.
The Performance Message Report Format is as per ANSI T1.403.
ALARMS
Alarms are indicated on the LEDs of the T1E1PRI Card. The T1E1PRI Card has four LEDs, namely L1 to L4. L1
and L2 indicate alarms for T1E1 PRI Port1 whereas L3 and L4 indicate alarms for T1E1 PRI Port2. During normal
conditions, the LED blinks green (1 sec ON, 1 sec. OFF).
RED Alarm
• This alarm is generated if Loss of Signal persists for 2.5 seconds.
• Received signal is more than 20 dB or 40 dB below nominal for at least 1ms.
• 32 consecutive zeroes are received.
• Loss of frame alignment occurs.
• This is indicated by flashing the LED Red (500ms ON, 500ms OFF).
• The system logs this event in the System Fault Log as RED Alarm <Slot No.> <Port No.> at HH:MM:SS.
• When RED Alarm is declared, Yellow Alarm is sent to the far end within 12ms of detection of LOS.
• RED Alarm is cleared when the signal is acquired back and persists for 10 seconds.
• The LED is turned OFF. The system logs this event in the System Fault Log as RED Alarm Cleared <Slot
No.> <Port No.> at HH:MM:SS.
YELLOW Alarm
• This Alarm is also known as Remote Alarm Indication or Distant Alarm.
• This Alarm is generated when Yellow Alarm is sent by the far end (Yellow Alarm is sent by the far end to
indicate that it has lost the incoming signal).
• Yellow Alarm is declared when the signal corresponding to Yellow Alarm persists for 0.5 seconds.
• This is indicated by flashing the LED Orange (500ms ON, 500ms OFF).
• The system logs this event in the System Fault Log as YELLOW Alarm <Slot No.> <Port No.> at
HH:MM:SS.
• This alarm is cleared when No Yellow Alarm signal persists for 0.5 seconds.
• The LED is turned OFF. The system logs this event in the System Fault Log as YELLOW Alarm Cleared
<Slot No.> <Port No.> at HH:MM:SS.
• Yellow Alarm is declared if Bit 3 of the receive NFAS is 1 on two consecutive occasions.
• Yellow Alarm is cleared if Bit 3 of the receive NFAS is 0.
If equipment is connected in downstream (Drop and insert mode, that is, NT mode) then on receipt of Yellow
Alarm, a Blue Alarm will be sent on the port, which is configured in NT mode.

BLUE Alarm
• It is also known as Alarm Indication Signal (AIS).
• This alarm indicates that the upstream equipment connected to the SARVAM UCS has lost receiving its
incoming signal.
• This alarm is generated when AIS persists for 2.5 seconds.
• This is indicated by flashing the LED Red (1 sec. ON, 1 sec. OFF).
• The system logs this event in the System Fault Log as BLUE Alarm <Slot No.> <Port No.> at HH:MM:SS.
• This alarm is cleared when clearance of AIS is detected for continuous 10 seconds.
• The LED is turned OFF. The system logs this event in the System Fault Log as BLUE Alarm Cleared <Slot
No.> <Port No.> at HH:MM:SS.
• Blue Alarm (AIS) is declared if unframed all ones signal is received as a string of 512 bits containing fewer
than three zero bits.
• When BLUE Alarm is declared, Yellow Alarm is sent to the downstream equipment connected to the
SARVAM UCS.
Port1 Status
LED Card Status Port Status
L1 Green Flashing @1 Sec. Card Heart Bit Port 1 Layer is established
L1 Red Flashing @1 Sec. Card Heart Bit Port 1 Layer is not established LOS-Red alarm detected.
L1 Red Flashing @500 Card Heart Bit Port 1 Layer is not established LBFA -Loss of Basic Frame
msec. Alignment.
L1 Red Flashing @100 Card Heart Bit Port 1 Layer is not established- LMFA loss of multi-frame
msec. alignment.
L1 Yellow Flashing @1 Sec. Card Heart Bit Port 1 Layer is established -RAI (Yellow Alarm) is detected.
L2 Green Port 1 achieves CRC4 synchronization.
L2 Yellow Port1 does not achieve CRC4 synchronization.
L2 Green Flashing @1 Sec. Port 1 achieves CRC4 synchronization-AIS (Blue Alarm) is

detected.
L2 Yellow flashing @1 Sec. Port 1 doesn’t achieves CRC4 synchronization-AIS (Blue Alarm)
is detected.
Port2 Status
L2 Green Steady Port2 Layer 1 is established.
L3 Red Flashing @1 Sec. Port 2 Layer is not established-LOS-Red alarm detected.
L3 Red Flashing @500 Port 2 Layer is not established-LBFA-Loss of Multi Frame

msec. Alignment.
L3 Red Flashing @100 Port 2 Layer is not established-LMFA Loss of Multi Frame
msec. Alignment.
L3 Yellow Flashing @1 Sec. Port 2 Layer is established-RAI (Yellow Alarm) is detected.
L4 Green Port 2 achieves CRC4 synchronization-AIS (Blue alarm) is

detected.

L4 Yellow Port2 doesn’t achieves CRC4 synchronization-AIS (Blue Alarm)

is detected.
Refer chapter “Loop Back Tests” for more details.

E&M Connectivity
What’s this?
E&M connectivity feature of SARVAM UCS offers seamless connectivity in PLCC network and also in between
various communication products like PBX, Router, Lease Line. E&M interface is widely used interface to connect
such diverse equipment. For example, in a PLCC network, number of PLCC System needs to be connected. As
shown in the figure 1 of PLCC network, number of Systems are connected with each other through E&M tie lines.
Say, an existing System capacity needs to be expanded beyond the configuration limit of a System. Installing one
more System and connecting both the Systems through E&M interfaces can get us the desired expansion.
Four different applications of E&M connectivity are shown below:
1. E&M Connectivity on PLCC-Network
E1 E2 En
PLCC-System (SID-61)
E&M1 E&M2 E&M3
PLCC-System
(SID-70)
E&Mn
PLCC-System
E&M1
(SID-34)
E&M2
E&M1 E1 E1
PLCC-System
(SID-52)
E&M2 E2 E2
E&M3
E&Mn En En
E&Mn
E&M1 E&M2 E&M3 E&Mn
E1 E2 En

How it works in PLCC Network
E&M interface is achieved using the E&M Card. An E&M port of the System has dual personality, both of a
subscriber and a trunk. An E&M port works like a subscriber interface for any incoming call to it and works
like a trunk interface when any subscriber makes an outgoing call through it. However, please note that a
trunk line cannot be connected to an E&M port. Also a subscriber cannot be connected to an E&M port.
Incoming call on an E&M port in PLCC network

When any subscriber of PLCC System with SID-51 grabs an E&M2 port, s/he gets dial tone of SID-61.
Now he can call any subscriber of SID-61 and use features like priority, conference, just like a normal
subscriber would do. However, this E&M port needs to be programmed for the same.
Outgoing call using E&M port in PLCC network

When any subscriber of PLCC System-34 dials network access code (as per programming) and dials SID
say 52, an E&M1 port is grabbed (as it is programmed for the same). Now he can call any subscriber of
SID-52 and establish speech. However, this E&M port needs to be programmed as any other normal trunk
like enabling the port, programming the dial type.
All E&M ports can be put together in one single group depending on the requirement. A separate access
code can be assigned to it. This makes the operations easy. Now the subscribers have at least two
different access codes for making outgoing calls using trunk lines and other for making outgoing calls
using E&M tie lines.
2. System EXPANSION (2-SYSTEMS)
Two Systems are connected using E&M interface.
System-A System-B
T1 E&M1 E&M1 T1
T2 E&M2 E&M2 T2
PSTN E&M3 E&M3
PSTN
Tn Tn
S1 S2 Sn S1 S2 Sn
In the above figure,

• S1 to Sn are stations.
• E&M1 to E&Mn are E&M lines between two Systems.

3. Systems are connected through E&M connectivity
System-A E&M1 E&M1 System-B

(8x8x8) E&M2 E&M2 (16x8)
T1 E&M3 E&M3 T1
E&M4 E&M4
T2 E&M5 E&M5
T2
PSTN E&M6 E&M6 PSTN
E&M7 E&M7
E&M8 E&M8
Tn Tn
S1 S2 S8 S1 S2 S16
2001 2002 ....... 2008 3001 3002 ........ 3016
E&M1 ............. E&M8

System-C
(24x8)
S1 S2 S24
4001 4002 ........... 4024
4. Two Systems located far from each other connected to each other using E&M connectivity.
System-A
T1 Router
PSTN T2 E&M E&M
T3
S1 Sn
Lease
circuit,
VSAT
2001 2099
System-B
T1 Router
PSTN T2 E&M E&M
T3
S1 Sn
2001 2099
How it works
E&M interface is achieved using the E&M Card. An E&M port has dual personality: both of a station and a
trunk. An E&M port works like a station interface for any incoming call to it and works like a trunk interface
when any station makes an outgoing call through it. However, please note that a trunk line cannot be
connected to an E&M port. Also a SLT or DKP cannot be connected to an E&M port.
Incoming call on an E&M port

When any station of System B grabs an E&M port, he gets dial tone of System A. This port of System A
behaves as a SLT port. Now he can call another station, grab a trunk, or use a feature like raid, just like a
normal station would do. However, this E&M port needs to be programmed as any other normal station.
Features like Class of Service, Toll Control, will apply to this port, if programmed.

Outgoing call using E&M port
When any station from System A makes an outgoing call through an E&M port, s/he dials trunk access
code. The E&M port behaves as trunk interface and gives the station, the dial tone of System B. Now s/he
can dial any station of System B or grab a trunk line of System B to make a call or use any feature
depending on the station programming done by the SE of System B. Hence the E&M port of System A
should be programmed as any other normal trunk by enabling the port, programming the dial type, etc.
All E&M ports can be put together in one single group or few groups depending on the requirement. A
separate access code can be assigned to it. This makes the operations easy. In such case the stations can
have at least two different access codes for making outgoing calls, one for making outgoing calls using
trunk lines and other for making outgoing calls using E&M lines. Generally, E&M connectivity is used to
expand the System capacity or connect two or more remotely located Systems. This forms a network of
Systems. The requirement is that, the so formed network should work as one Group. This is commonly
known as Closed User Group. Please refer “Closed User Group (CUG)” and “Closed User Group-With
Exchange ID” for more details.
How to configure
Please refer “Configuring E&M Lines”, “Station Basic Feature Template”, “Station Advanced Feature Template”,
“E&M Feature Template” for more details.

Emergency Calls (911) - Reporting to PSAP
What's this?
Make sure:
• you have selected the Region as USA or Canada.
• in the Emergency Table for the Number 911, you have assigned the outgoing trunk bundle group with
only T1 PRI lines. For instructions, see “Configuring Emergency Number Dialing”.
The SARVAM UCS allows you to assign Customer Emergency Services Identification (CESID) number to each
extension user. When the user dials the Emergency number 911, the system sends the caller’s CESID number to
the network and the call is routed to the Public Safety Answering Point (PSAP).
The PSAP stores information, such as name, location, and CESID number of the users in an Automatic Line
Information (ALI) database. The PSAP uses the CESID number that you have assigned to an extension to locate
the user's information in the ALI database. Hence, the PSAP can provide user’s the desired service without delay.
The CESID numbers are provided by the Service Provider. In case these are not provided, you can assign
DDI numbers to extension users. See “Direct Dialing-In (DDI)” to know more.
Configuring CESID List using Jeeves

• After you have selected the Region and configured the Emergency Number table.
• Under Configuration, click Emergency.

• Click CESID to open the page.
• Click the desired tab —SLT, DKP, SIP.
• The list of Extension Number and Extension Name of the users appear as configured by you.
• For each Extension Number assign the CESID Number and enter his/her Location.
The CESID Number can a maximum of upto 12 digits and the Location can be a maximum of upto 18
alphanumeric characters.
• Click Submit.

What’s this?
Emergency Conference enables you to establish a Conference between a pre-defined group of extensions using a
feature access code.
This feature can be used to call and consult with a group of people in emergency situations.
The number of parties that can be included in an Emergency Conference group depends on the Multiparty
Conference capacity of the SARVAM UCS. For details, see “Conference-Multiparty”.
How it works
• First decide the key persons in the organization who should be parties to the Emergency Conference.
• Form a Department Group with the extensions of these key persons as members. A single Department
Group can have up to 32 extensions. For more information on forming Department Groups, see the topic
“Department Call”.
For example, you have formed a Department Group for Emergency Conference, with the extensions A to G as
members. The Access Code assigned to the Department Group is 3901. H is the initiator of the conference as
Emergency Conference is enabled in the Class of Service assigned to his/her extension.
Another Emergency Conference is formed with Department Group 3902, having extensions J to P as members and
J as the initiator of the Conference as Emergency Conference is enabled in the Class of Service assigned to his/her
extension.
Now, extension H wants to initiate an Emergency Conference.
This is how the feature will work:
Initiating an Emergency Conference

• H dials the feature access code for Emergency Conference, followed by access code of the Department
Group (3901).
• All extensions in the Department Group (extensions A to G) which are free will start ringing. The system
will play Emergency Conference ring (default: Triple Ring) on the Extensions. Extensions that are busy will
not be included in the call.
If there are DKP/Extended IP Phone extensions in the group, and these phones have a Call Appearance
free, the system will ring these extensions on the free Call Appearance, but will not wait for the extensions
to become free.

When the Emergency Conference is initiated, the system will ring on the extensions that are not occupied
in any other single multi-party or 3-party conference.
For more information regarding the number of single multi-party or 3-party conferences supported, refer
“Technical Specifications - SARVAM UCS”.
• Extension A goes Off-Hook to answer the call first. A gets connected to the initiator of the conference,
extension H.
• Two-way speech is established with extension A and H. All other extensions continue to ring.
• When another extension, B goes Off-Hook to answer the call, A and H get a beep, and three-way speech
is established between A, B, H.
• Thus, whenever a new member joins the conference, all other extensions already in conference will get a
beep, if the flag Play Beep when Conference/Dial-In Conference Starts is enabled in the “System
Parameters”.
• If the conference initiator, extension H, goes idle, all other extensions in the conference will still be in
conversation.
Merging Emergency Conferences

• Only the initiator of the conference, extension H, can merge the conference using the phone menu.
• While in Conference, H presses the ‘Conference’ Key.
• From the Multiparty Conference Menu, H selects Merge Conference to merge with another ongoing
Emergency Conference.
• H selects 3902 from the list of ongoing Emergency Conferences. Both the Emergency Conferences
(3901 and 3902) are merged.
Cancelling an Emergency Conference

• Only the initiator of the conference, extension H, can cancel the conference. The initiator of the conference
can cancel the conference at two stages:
• When speech is established with one or more member extensions of the Emergency Conference
department group.
Or
• During Ring Back Tone, as the system rings on the extensions of the group, after the initiator of the
conference has dialed the feature access code.
• To cancel the Emergency Conference, extension H must dial the feature access code for Cancel
Conference, 190 (default).

How to configure
• You must enable the feature Emergency Conference in the“Class of Service (COS)” of the extensions in
their “Station Basic Feature Template”. By default, this feature is enabled on all extensions, so all
extensions can use this feature.
• If the extension you are providing this feature is a DKP or an Extended IP phone, you may program a DSS
key on the phone with this feature.
• You must also create a Department Group as Emergency Conference group. For instructions, see
“Department Call”.
• By default, the system plays a beep when the Emergency Conference starts. If you do not want the beep
to be played, you must disable the flag Play Beep when Conference/Dial-In Conference Starts. For
instructions, see “System Parameters”.
This flag is common for other features like “Conference-Multiparty”, “Conference Dial-In” and “Raid”.
• By default, the system plays Triple Ring as Ring Type for Emergency Conference. If necessary, you may
configure a different ring type. For more information and for instructions, see “Distinctive Rings”.
How to use

To initiate an Emergency Conference:
• Press DSS Key assigned to Emergency Conference.

Or
• Dial 1177
• Dial Department Group Number.
• All free extensions in the group will ring.
• You get connected to the extensions that answer.
If no resources are free, you will get the ‘Conf. Resource full’ message on your LCD.
You can initiate an Emergency Conference also using “Direct Inward System Access (DISA)”.
To merge Emergency Conferences:

• Select the option ‘Merge Conferences’
• The LCD displays list of Department Number and Group Name of ongoing Emergency Conferences.
• Select the desired Department Group number and press the Enter Key.
• Both the conferences will be merged.
Emergency Conferences can be merged by DKP or Extended IP Phone users only through the Phone
Menu.

To cancel an Emergency Conference while in speech with one or more extensions:

OR

freed.
To cancel an Emergency Conference while the extensions are ringing:
• Go ON-Hook during Ring Back Tone.

• All extensions in the conference group will stop ringing.
You can cancel the conference only if you have initiated it.
For SLT Users

To initiate an Emergency Conference:
• Dial 1177
• Dial Department Group Number.
To cancel an Emergency Conference while in speech with one or more extensions:
• Go ON-Hook and then go OFF-Hook.

• Dial 190
To cancel an Emergency Conference while the extensions are ringing:
• Go ON-Hook during Ring Back Tone.

Emergency Detection and Reporting
What’s this?
When an emergency call is made from an extension, the system dials out the number using any of the free trunks
selected for routing Emergency Numbers. Since the number is dialed out by the System, the Emergency Service
that attends to the call will be able to locate the System, but not the extension that made the call.
Similarly, the Operator too has no way of knowing which of the extensions made the call, thus making it difficult to
quickly reach and provide help to the extension that made the emergency call.
With the Emergency Detection and Reporting feature, the Operator can know from which extension the emergency
call is being made. Whenever an Emergency call is made by an extension user, the system detects and reports it to
the Operator extension.
How it works
When an extension of SARVAM UCS makes an emergency call by dialing an Emergency Number,
• the system hunts for a free trunk in the OGTBG selected for routing the emergency number, and dials out
the number from a free trunk.
• simultaneously, the system informs the Operator by ringing on the Operator extensions for the duration of
the Emergency Reporting Call-Ring Timer (configurable; default: 10 minutes).
• If Operator is a DKP or an Extended IP Phone, it will ring continuously, and an emergency message will be
displayed on the LCD.
Mon 21 MAR 10:38

3002 Recepti on 2
2003 Emer gency
The emergency message shows the number of the extension which has made the emergency call, in this
case, extension 2003.
• To acknowledge the Emergency call the operator must press the enter key. The acknowledged Emergency
calls are logged into the System Activity Log.
• If the Emergency Call is not acknowledged by the operator, the emergency call is logged into the
Emergency Alarms Log. To know more about the Emergency Alarms Log, see “Emergency Alarms Log” at
the end of the topic.
Also see the topics “Configuring Emergency Number Dialing” and “Emergency Dialing”.

How to configure
To be able to use this feature, the Emergency Dialing Reporting flag must be enabled in the System Parameters.
See “System Parameters” for instructions. By default, this flag is enabled.
If required, you can change the Emergency Reporting Call-Ring Timer, see “System Timers and Counts”.
Emergency Alarms Log

Emergency Alarms Log is the log of unacknowledged Emergency Calls.
When an Emergency call is made by any extension user and is not acknowledged, that call is logged into the
Emergency Alarms Log.
The log can be viewed by any DKP / Extended IP Phone user, using the DSS Key assigned to Emergency Alarms
Log only. When an Emergency call is made by any extension user, the LED of the DSS Key glows continuous RED.
To view the log from any DKP / Extended IP Phone user,
• Press the DSS Key assigned to Emergency Alarms Log.
• A list of the last 20 unacknowledged Emergency calls appears with the following details:
• Extension number from which the Emergency call was made.
• Date and Time when the Emergency call was initiated from that Extension.
• Press the enter key to acknowledge the Emergency Call. The message "Emergency Acknowledged"
appears on the screen.
• The system plays the Confirmation Tone followed by the Dial Tone.
• The acknowledged Emergency call is removed from the Emergency Alarms Log and is logged into the
System Activity Log with the details of the extension that acknowledged the call.

Emergency Dialing
What's this?
The SARVAM UCS supports dialing of Emergency number immediately without any blocking.
When an extension user dials an Emergency number, the system will hunt for a free trunk from the outgoing trunk
bundle group selected for the emergency number. See “Configuring Emergency Number Dialing”.
The system will not apply any of the following on the extension dialing the Emergency number:
• Toll Control (Allowed Denied Numbers, Dynamic Lock)
• Call Budget (even when call budget is consumed)
• Call Duration Control
• Automatic Number Translation
The system will allow the extension to dial the Emergency number even in the following conditions:
• the extension is in Off-Hook state.
• the extension is in Standby Mode.
• the extension has grabbed the trunk line (using Trunk access code or selective access)
• the call state is in any state: Ringing, Busy, Error, Confirmation.
• SIM card is not present in the Mobile port.
• Mobile port is not registered with the network.
• SIM PIN is not valid.
• the keypad of the extension phone is locked.
Emergency Numbers will always be out dialed through the OGTBG you have selected for the numbers, except
when you have grabbed a trunk using Selective trunk access code/Selective Trunk Access DSS key. In this case,
the number will be dialed only from the trunk you have grabbed.
Emergency Number will not be out dialed in the following cases:

• If the trunk port from which number is to be routed (CO, Mobile, T1E1PRI, BRI, SIP) is disabled.
• If the hardware related to dialing of Emergency number is not present.
• If the Emergency number is dialed from the SE Programming mode.
Emergency dialing will not work if Mains Power to the SARVAM UCS fails.
How to configure
The Emergency numbers are fixed as per the Region where SARVAM UCS is installed, you can add emergency
numbers, as required. For instructions, see “Configuring Emergency Number Dialing”.
How to use
To dial an Emergency number,
• Go Off-Hook
• Dial the Emergency Number
OR
• Dial Trunk Access Code-Emergency Number

For example:
Dial 0-112
• Wherever the Trunk Access Code conflicts with the Emergency Number, the emergency number
should be dialed after dialing the Trunk Access Code.
• Let us take the example of Australia, where the emergency number is 000 and the trunk access code is
0. Now, when an extension user of SARVAM UCS located in Australia dials ‘0’ of the emergency
number, the system will consider it as trunk access code and will apply the trunk access code logic.
Therefore, in such cases, the extension user must first dial the Trunk Access Code and then the
Emergency Number. In this case, the extension user must dial 0-000 for emergency number dialing, so
that the system will not wait for the Conflict Timer to apply the Trunk access code logic.

Extended IP Phone/VARTA UC Client - Operation
Matrix offers the following proprietary Extended IP Phones/Mobile UC Clients:

• SPARSH VP248, The High-Definition Edge to your IP Communication. For Detailed description, see
“Matrix SPARSH VP248”.
• SPARSH VP310, The Executive IP Phone. For Detailed description, see “Matrix SPARSH VP310”.
• SPARSH VP330, Intuitive Touchscreen IP Phone. For detailed description, see “Matrix SPARSH VP330”.
• SPARSH VP510, The Premium IP Phone. For detailed description, see “Matrix SPARSH VP510”.
• Mobile UC Clients - VARTA ADR100, Mobile UC Client for Android Smart Phones and VARTA AMP100,
Mobile UC Client for iPhones. See “Matrix VARTA ADR100 UC Client” and “Matrix VARTA AMP100 UC
Client”
• MATRIX VARTA WIN200, UC Client for Windows. For detailed description, see “MATRIX VARTA
WIN200”.
Matrix SPARSH VP248

The Matrix SPARSH VP248, the proprietary Extended IP Phone is a feature-rich, VoIP (Voice over Internet
Protocol) phone, providing voice communication over IP network. It looks and works like any normal phone, having
all the traditional phone features such as redial, speed dial, call transfer, call hold, call forward, conference, and so
forth. It allows you to make and receive calls using the handset, the headset (if connected) and the speaker.
The models of SPARSH VP248 are:
• SPARSH VP248S - the standard model, with a 2-line x 24-character LCD display.
• SPARSH VP248P - the premium model, with a 6-line x 24-character LCD display.
It is a powerful extension, supporting a host of phone and SARVAM UCS features, as listed below.
IP Phone Features
• Status of other ports (Tri-color LED indication)
• Programmable Direct Station Selection (DSS) Keys and Feature keys
• LCD notification messages
• Ringer Tune selection
• Adjustable Speech level
• Adjustable Ringer Volume
• Adjustable Backlight and Contrast levels
• Hands-free operation - Speaker key and headset connectivity.
• Call Logs - last 20 Missed, Answered and Dialed Calls.
• Message Paging
• Menu based operation of SARVAM UCS features
• Multiple Language support.
SARVAM UCS Features

The SPARSH VP248 supports SARVAM UCS features. A few of these are listed below:
• Abbreviated Dialing
• Auto Answer

• Call Chaining
• Call Cost Display
• Call Duration Display
• Call Mute
• Dialed Number Directory
• Directory Dialing by Name
• Dynamic Lock
• Forced Answer
• Keypad Lock
• Message Paging
• Off-Hook Alert
• Room Monitor
• User Status (Presence)
SPARSH VP248, Front View
1 Ringer LED 7 Speaker key with LED

2 Date and Time 8 Volume decrease key
3 C u rs o r 9 Dial Pad
4 Touch sense feature keys 10 Programmable feature keys
5 Touch sense navigation keys 11 Handset
6 Volume increase key 12 4P4C Spring Cord

Models of SPARSH VP248 at a Glance
Model
Feature
SPARSH VP248S SPARSH VP248P
Total number of keys 48 48
Number of programmable keys 29 29
Capsense keys Yes Yes
LCD display capacity 2 lines x 24 6 lines x 24

characters characters
LCD with backlight Yes Yes
Headset Interface Yes Yes
Ringer Lamp (LED) Yes Yes
Speaker Phone Full duplex Full duplex
SPARSH VP248S
2 lines and 24 characters LCD display, full duplex, capsense feature keys

SPARSH VP248P
6 lines and 24 characters LCD display, full duplex, capsense feature keys.
LCD Display
The LCD display of SPARSH VP248 is backlit and can be tilted at a convenient angle for a clear view of the text/
characters displayed.
Settings" of the SPARSH VP248 Phone Menu.
Ringer LED

Navigation Keys
The phone has 5 touch sense navigation keys to be used to move the cursor and scroll through Menu options.
is the Enter key, used to make a selection or to complete an action.
is the Up key, used to scroll upwards while navigating the 'Menu'.
is the Down key, used to scroll downwards while navigating the 'Menu'.
is the Forward key, used to move the cursor.
is the Back key, used to move the cursor, return from the Sub-menu to the Main Menu.
Feature Keys
These are 12 capsense keys assigned to important or frequently accessed features of SARVAM UCS. Refer to the
table given below:
Sr.No. Description LED
2. Call Back Single Color - Blue
3. Cancel No
5. Conference No
6. Transfer No
9. Names No
10. Redial No
11. Release No
12. Hold No
These keys are programmable. However, as you cannot change the labels, avoid programming these keys.
For instructions on programming these keys, see “DSS Key Settings” under “Configuring Matrix SPARSH VP248”.

DSS Key LEDs

Each DSS key is equipped with an LED which glows to indicate the status of the Trunk/Extension or Feature
assigned to it.
colors to indicate status of the call event on the Extensions/Trunks and on the Extended IP Phone.
Thus, the status of the Extended IP Phone user's own Extension as well as that of the other Extensions
(i.e. Extended IP Phones and SLTs) and the status of Trunk lines are indicated by the LED of the DSS
keys assigned to those Extensions and Trunks on the Extended IP Phone.
LED LED Mode

with. hold. from which you are being
called.
Extension that is now busy Extension which has put Extension/Trunk that is called
with another Extension/ another Extension/Trunk on or being called by another.
Trunk. hold.
• Red indicates the state of other Extensions/Trunks. For example, if the LED of the DSS key assigned
to Extension 201 is glowing Red continuously, it means Extension 201 is busy with another Extension
or Trunk.

accessed.
Dial Pad
dialing numbers of extensions, external parties, and for dialing the programming and feature access codes.
Speaker Key
The speaker key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable,
you can program any other feature/function on this key.
Speaker Key LED

The Speaker Key on SPARSH VP248 is equipped with a single color LED which glows Blue when pressed for the
speaker mode and is turned off, when you exit the speaker mode.
Volume Keys
The SPARSH VP248 provides two Headset interfaces: a 2.5mm Audio Jack and an RJ9 connector at the bottom of
the phone body.
connector.
You can also program any of the DSS keys to function as the Headset key. For instructions on programming the
key, see “DSS Key Settings” under “Configuring Matrix SPARSH VP248”.
Key Maps
As SPARSH VP248 may be the extension of the Operators and Executives in an enterprise to meet the varied
requirements of each user group, these key maps can be customized to match the exact requirement of individual
users. For instructions on customizing the Key Maps, see “Customizing Extended IP Phone Templates using
Jeeves” . You can also personalize the key maps for each location, for instructions, see “DSS Key Settings” under

Matrix Extended IP Phone, SPARSH VP248 Key Template (default)
By using Key Templates you can prepare and assign common key maps to all or as many Extended IP Phones as
you want, at one go.
SARVAM UCS also offers the flexibility to personalize the Key Maps of each Extended IP Phone, instead of using
the Key Templates. For example, if you have assigned a common Executive Key Template to 12 Extended IP
Phones, but you want to reassign some of the keys on two of these Extended IP Phones, SARVAM UCS allows
you to selectively personalize the key maps of these two Extended IP Phones.
Phone Menu
You can access the following SARVAM UCS and phone features from the Menu of SPARSH VP248:
Do Not Disturb To set/cancel Do Not Disturb on the phone, i.e. block incoming internal and external
calls.

Password
One Touch Transfer To set/clear Transfer Number.



To exit menu,

Or,
• Go ON-Hook.

The call waiting indication depends on the Call Waiting Tone (for SPARSH VP248/VP310/VP510) option you
select in General Parameters under SIP Extensions. See “Configuring Matrix SPARSH VP248” for instructions.
Connecting SPARSH VP248

For detailed instructions to connect SPARSH VP248, see “Connecting SPARSH VP248 as Extended SIP
Extension” in SIP Extensions for ETERNITY LENX, “Connecting SPARSH VP248 as Extended SIP Extension” in
SIP Extensions for ETERNITY MENX and “Connecting SPARSH VP248 as Extended SIP Extension” in SIP
Extensions for ETERNITY GENX.
Configuring SPARSH VP248

For detailed instructions on how to configure SPARSH VP248, see “Configuring Matrix SPARSH VP248”.
Operating SPARSH VP248

SARVAM UCS.

Matrix SPARSH VP310
SPARSH VP310, The Executive IP Phone is engineered to offer a contemporary design with crystal-clear audio
and feature-rich capabilities at economical price. Elegant design, built-in programmable DSS Keys and plug-n-play
connectivity makes SPARSH VP310 an easy to use phone for executives. SPARSH VP310 works in tight
integration with SARVAM UCS for speed of operations and better workforce collaboration.
Key Features
• 2 Line LCD with Backlit
• Fixed Function Keys (With LED) - Voice Mail, Mute, Do Not Disturb, Forward, Logs, Speaker
• Fixed Function Keys (Without LED) - Hold, Conference, Contacts, Transfer
• Superior Voice Quality with HD Audio
• Full Duplex Speaker Phone
• PC and LAN Ethernet Ports
• Power over Ethernet (IEEE 802.3af)
• 12 DSS/BLF Keys for Feature, Line, Extension
• Message Wait and Ringer Lamp
• 3.5mm Headset Connectivity
• Adjustable Desk Stand
• Plug & Play
If any SPARSH VP310 extension is Off-Hook, the system will not be able to detect it.However, if any SLT/
DKP is Off-Hook , Off-Hook Alert will be provided to SPARSH VP310 if it is the Operator phone.

1 Ringer LED
2 LCD Screen
3 Navigation Keys
4 Fixed Function Keys
5 DSS (Direct Station Selection) Keys
6 Digit Keys/ Dial Pad Keys
7 Volume Increase Key/ "+" Key
8 Speaker Key
9 Volume Decrease Key/ "-" Key
10 Handset
LCD Display
The LCD display of SPARSH VP310 is backlit. The LCD backlight can be turned on and off as well as adjusted for
contrast and brightness from the Phone Settings of the SPARSH VP310 Phone Menu.
Ringer LED

The Ringer LED changes according to the type of call, as described in the table below.
Navigation Keys
The phone has the following navigation keys which are used to move the cursor and scroll through Menu options.
is the Enter key, used to make a selection or to complete an action.
is the Up key, used to scroll upwards while navigating the Menu.
is the Down key, used to scroll downwards while navigating the Menu.
is the Forward key, used to move the cursor.
is the Back key, used to move the cursor, return from the Sub-menu to the Main Menu.
is the Cancel key, used to exit a menu.
Feature Keys
There are 9 Feature keys assigned to important or frequently accessed features of SARVAM UCS.
Sr.No. Feature icon Description LED
3. Hold No
5. Call Forward Single Color - Blue

Sr.No. Feature icon Description LED
6. Conference No
7. Call Logs Single Color - Blue
8. Names No
9. Transfer No
These keys are programmable. However, as you cannot change the labels avoid programming these keys.

DSS Key LEDs

assigned to it.
(i.e. Extended IP Phones, SLTs) and the status of Trunk lines are indicated by the LED of the DSS keys
assigned to those Extensions and Trunks on the Extended IP Phone.

LED LED Mode

called.
Trunk. hold.
or Trunk.
accessed.
key to which the Auto Redial feature has been assigned will glow Red, when Auto Redial is set, and the
Dial Pad
The dial pad consists of 12 fixed keys for the digits 0-9, and the characters * and #. The dial pad is used for dialing
numbers of extensions, external parties, and for dialing the programming and feature access codes.
Speaker Key

Speaker Key LED
The Speaker Key on SPARSH VP310 is equipped with a single color LED which glows Blue when pressed for the
speaker mode and is turned off, when you exit the speaker mode.
Volume Keys
The SPARSH VP310 provides two Headset interfaces: a 3.5mm Audio Jack and an RJ9 connector on the left side
panel of the phone.
connector.
Key Maps

Phone Menu
Press the Enter key to access the Phone Menu.
You can access the following SARVAM UCS and phone features from the Menu of SPARSH VP310.
Keypad Lock To lock the keypad of the phone. (when the keypad is locked, the features Call Log,
Contact, Call Forward, Dynamic Lock, User Status, DND, Call Cost Display, Hotline,
Alarm, Change User Password will not be accessible.)

Do Not Disturb To set/cancel Do Not Disturb on the phone, i.e. block incoming internal and external
calls.

Password
One Touch Transfer To set/clear Transfer Number.
When the phone is in idle state,
• Press the Down key to access the Network Settings.
• Press the Up key , if you wish to change the Ringtone and Play Key Tone.

• Press on Enter key when the phone is idle.

• Press on Enter key to select the desired Menu option.
• Press on Enter key to select the desired sub-menu option.
To exit menu,

Or
• Go ON-Hook.




SARVAM UCS.
Matrix SPARSH VP330
SPARSH VP330, the next generation feature rich SIP phone of Matrix with an intuitive GUI (Graphical User
Interface) based touch-screen. It provides you an easy way of managing your communication needs to meet the
day to day business requirements. Designed to change the way you communicate and collaborate, SPARSH
VP330 delivers a seamless communication experience that is convenient and ready to use in real time, so that you
can focus on the task in hand. Once it is registered with the SARVAM UCS, you can start operating the phone.
Key Features
• Enhanced Call Management: Dedicated one-touch feature keys and intuitive user interface provides
quick access to full range of System call management features including Call Hold, Call Park, Call
Transfer, Conference and Voicemail.
• Access to Corporate Directory: Easy integration with the enterprise’s Corporate Directory (Global
Directory) which allows you to easily locate and dial corporate contacts at one click.
• Presence: Provides intuitive Presence status display and supports changing your Presence status which
is viewable to other extension users. You can also view the Presence Status of remote users.

• Busy Lamp Field (BLF) Indication: You can monitor the status of the extensions and trunks who are
assigned DSS Soft keys. You can also pickup ringing extensions/trunks using DSS Soft keys.
• Plug & Play: Integrated Plug & Play feature that enables to power up the phone and start using it. On the
other hand, it helps in mass deployment of the phones in your organization without requiring a lot of
manual intervention to configure each of the phones separately.
• Multiple Language Support: The phone can be operated in six different languages including English,
French, German, Spanish, Portuguese and Italian.
• Capacitive Touch Screen: 4.3 inch Capacitive Touch Screen LCD (Liquid Crystal Display) that delivers
easy access to advanced features and a unique experience beyond traditional desk phones. Supports
adjustable Brightness controls from the touch screen to suit your customized LCD requirement.
• Easy-to-Use Hard Keys: Supports following hard keys and LEDs:

• 6 Fixed Feature Keys
• 12 Alphanumeric Digit Keys (Dial pad Keys)
• 1 Speaker Key.
• 12 DSS (Direct Station Selection) keys.
• 1 Ringer LED.
• Improved Audio: High Definition (HD) Audio output that delivers crystal clear voice and life like
conversations over HD handset and hands-free speaker.
• High Speed Ethernet connectivity: Dual switched 10/100 Base-T auto-sensing Ethernet LAN
connectivity allows unconstrained bandwidth from the network to the phone. LAN port is used to connect
the phone to a Switch or a Hub or a Router or an xDSL Modem. You can connect your PC to the PC port
of the phone.
• Power over Ethernet (PoE): Integrated IEEE 802.3f Power over Ethernet allows easy deployment with
centralized powering without a need for external power adapter.
• Wi-Fi Support: Supports Wi-Fi (WLAN) connectivity using which the phone provides seamless
connectivity to the corporate Wi-Fi network and offers flexibility to work from anywhere in the office. If your
installation setup does not meet the requirements of suitable wired Ethernet connectivity due to any
reason, then you can connect the compatible Wi-Fi Adapter supplied by Matrix to the USB port to register
and use the phone.
• Improved Audio: High Definition (HD) Audio output that delivers crystal clear voice and life like
conversations over HD handset and hands-free speaker.
• Full-duplex Hands-free Speaker: High quality speaker with acoustic echo cancellation to deliver natural
and clear speech even for hands free operation without any distortion.

Front View
1 Ringer LED
2 LCD Screen
5 Digit Keys/ Dial pad Keys
6 Volume Up Key/ "+" Key
7 Speaker Key
8 Volume Down Key/ "-" Key
9 Handset

Bottom View
2 3 4
1 USB Port for Wi-Fi Adapter
2 LAN Port for a LAN Switch / Router
3 PC Port for a PC / Computer
4 Power Port for the Power Adapter
Left Side View
1 RJ11 Headset Port
2 Headset Port
3 Handset Port

All the key maps of the SPARSH VP330 are the same.
Feature Keys and DSS Keys

Feature Keys
There are 6 Feature Keys. Each Feature Key is accompanied by a feature icon that describes its function. Default
features assigned to these keys are as follows.
SR. No. Feature icon Assigned Feature LED
1 Voicemail Single Color - Blue
2 Mute Single Color - Blue
3 Headset Single Color - Blue
4 Hold No
5 Conference No
6 Transfer No
These Feature keys are programmable. You cannot change the labels of these keys and therefore it is
recommended that you avoid programming these keys. However, if you still decide to reprogram features on these
keys, keep a note of the changes you have made.
DSS (Direct Station Selection) Keys

There are 12 DSS Keys which can be used for quick access to different features and facilities. For example, to dial
an extension number at one key press, just press the DSS key assigned to that extension and the call will be
placed automatically. These Keys have dual color LEDs - blue and red. Features/facilities assigned to these keys
can be changed by the System Engineer.

Jeeves” . You can also personalize the key maps for each location, for instructions, see, “DSS Key Settings” under
For detailed information about SPARSH VP330, refer to the SPARSH VP330 User Guide.



Refer the SPARSH VP330 User Guide for instructions on operating the features of SARVAM UCS.
Matrix SPARSH VP510
SPARSH VP510, the Premium IP Phone sets the benchmark for quality performance with elegant design and
crystal-clear voice. SPARSH VP510 features a Vivid LCD Graphical Display, Direct Station Selection (DSS) Keys,
3.5mm Headset connectivity, High Quality speakerphone and high definition audio quality.
Engineered to deliver full feature access of SARVAM UCS, SPARSH VP510 acts as face of your communication
system covering wide array of business environments.

The State-of-the-art Deskphone is best suited to deployment for Reception, Supervisors, Managers, Executives,
Call Center Agents and Office professionals. These phones offer flexibility to streamline communications and attain
higher return over investment.
Key Features
• Minimum 240*64 pixel and above graphical LCD with Back light
• Message Wait and Ringer Lamp
• Built-in 16 DSS Keys for Feature, Line, Extension
• Add -on DSS module facility
• HD Voice,HD Handset, HD Speaker
• 4 Context Sensitive Keys
• 5 Features Keys (With LED ) -Voice mail ,Headset ,Mute, DND, Speaker
• Fixed Function Keys (Without LED) - Hold , Conference, Redial, Transfer
• Tight integration with Server over SIP protocol /Proprietary Protocol
• HTTP Auto Provisioning
• Dual Color LED illuminated LED for line status
• One Touch Transfer
• Call logs
• Ringtone selection
• Wideband Codec : G722
• Narrowband Codec: G.711,G.723,G.729ab,GSM
• VAD,CNG,AEC,AJB,AGC
• Full Duplex speaker phone with AEC ,VAD ,CNG
• IP Assignment : Static /DHCP
• TCP/DNS-SRV
• AEC encryption for config file
• IEEE802.1x
• 3.5 mm / RJ9 headset port
• Dual port 10/100 Mbps Ethernet
• Stand with 2 adjustable angles
• PoE (IEEE 802.af) class2
If any SPARSH VP510 extension is Off-Hook, the system will not be able to detect it. However, if any SLT/
DKP is Off-Hook, Off-Hook Alert will be provided to SPARSH VP510 if it is the Operator phone.

1 LCD Screen
2 Ringer LED
5 Navigation Keys
6 Volume Increase Key
7 Speaker Key
8 Volume Decrease Key
9 Dial Pad/ Key Pad Keys
10 Context Sensitive Keys
11 Handset
LCD Display
The LCD display of the phone is backlit. The LCD backlight can be turned on and off as well as adjusted for
contrast and brightness from the Display Settings of the Phone Menu.
Ringer LED
The Ringer LED will glow in Blue (1 sec ON – 500 msec OFF) to indicate incoming internal and external calls.

Navigation Keys
The functions of each are described briefly below.
• Up Key: To scroll upwards while navigating the Menu/sub-menu or to access Phone Settings and set the
Ringtone (when phone is in the idle state).
• Forward Key: To move forward when dialing a number or scroll to view the Context Sensitive Key options.
• Back Key: To move backwards when dialing a number, to go back one level in the Menu or scroll
backwards to view the Context Key options.
• Menu or Select / OK Key : To enter the Menu; when the phone is in the idle state (without any
incoming or outgoing call being made).
Menu Key functions as the Select / OK Key to make a selection from the Menu/sub-menu options or to
complete an action. When there is an incoming call it functions as the Answer Key.
• Cancel Key : To Cancel all features set by you or exit the Menu/sub-menu.
Feature Keys
There are 8 Feature keys assigned to important or frequently accessed features of SARVAM UCS.
Assigned
Feature icon LED Programmable
Feature
Hold No Yes
Redial No Yes
Transfer No Yes
Conference No Yes
Voicemail Single Color - Blue Yes
Do Not Disturb Single Color - Blue Yes
Headset Single Color - Blue No
Mute Single Color - Blue No
These keys are programmable. However, as you cannot change the labels avoid programming these keys.


DSS Key LEDs

assigned to it.
(i.e. Extended IP Phones, SLTs) and the status of Trunk lines are indicated by the LED of the DSS keys
assigned to those Extensions and Trunks on the Extended IP Phone.
LED LED Mode

called.
Trunk. hold.

or Trunk.
accessed.
key to which the Auto Redial feature has been assigned will glow Red, when Auto Redial is set, and the
Dial Pad
The dial pad consists of 12 fixed keys for the digits 0, 1-9, and the characters Star (*), Hash (#), Lock ( ), Plus (+)
and Dot. The dial pad is used for dialing numbers of stations, external parties, and for dialing the programming and
feature access codes.
Speaker Key
Speaker Key LED

The Speaker Key on the phone is equipped with a single color LED which glows Blue when pressed for the speaker
Volume Keys
The phone provides two Headset interfaces: A 3.5mm Audio Jack and an RJ9 connector at the bottom of the phone
body.
connector.
To use the Headset, a Headset Key is assigned on the phone. The Headset Key on the phone is equipped with a
single color LED which glows Blue when pressed to indicate that the Headset mode is turned on and is turned off,
when you press it again to indicate that you have exit the headset mode.

Key Maps
Phone Menu
To access the Phone Menu, press the Enter Key. You can access the following System and phone features from
the phone:

Call Forward To set and cancel Call Forward-Unconditional, Call Forward-Busy, Call Forward No
Reply, Forward On Busy/No Reply and Follow Me.
User Status To set User Present or User Absent and Presence Status.
calls.

Password
One Touch Transfer To set/clear the fixed destination number for One Touch Transfer.
Phone Settings To customize settings of the phone.
When the phone is in idle state,
• Press the Down key to access the Network Settings.
• Press the Up key , if you wish to change the Ringtone and Play Key Tone.


To exit menu,

or
Go ON-Hook.



Please refer the EON510_SPARSH VP510 User Guide for instructions on operating the features of SARVAM UCS.
Matrix VARTA ADR100 UC Client
Matrix VARTA ADR100 is a proprietary SIP (Session Initiation Protocol) based UC Client running on Android
Phones/Tablets, delivering full-array of Matrix SARVAM UCS features to the user on-the-go along with an added
advantage of video calling. Through tight integration with the enterprise mobility features of the SARVAM UCS,
Matrix VARTA ADR100 provides advance call capabilities including Conferencing, Corporate Directory Access
(Global Directory), Call Logs and Conversation Recording with one-touch access. Other than these you can take
the advantage of using UC features like Presence, IM, IM to SMS, Corporate VMS access to enhance your overall
mobile experience.
working from office, home or traveling to any location. An innovative and easy to understand user interface delivers
all productivity features at fingertips that enhance speed of communication and collaboration with office users and
customers.
Make sure the phone/tablet in which you install Matrix VARTA ADR100, runs on Android V2.3.3 or later.
To use MATRIX VARTA ADR100 make sure you have:

• Purchased and activated the VARTA Essential or VARTA Professional or VARTA Collaboration license.
For more details, see “License Management”.

Key Features
• System Extension: Matrix VARTA ADR100 becomes a mobile extension of the SARVAM UCS. As an
increased number of business professionals are using the collaborative tools found in smartphone devices
to help them in their work activities, this application offers businesses an easy way to integrate their
enterprises’ voice solutions within the Android OS family.
• Advanced Call Capabilities: Access to features such as Callback, Dial-in Conference, Conversation
Recording and many more.
• Mobility: Matrix VARTA ADR100 provides you the mobility that you need in today's highly competitive
business environment; with the ability to access SARVAM UCS features easily once you are connected to
either Wi-Fi or 3G network. Considering the case of roaming users, one can register Matrix VARTA
ADR100 with the SARVAM UCS using the enterprise Wi-Fi network when working within the office (that is
within the organization's dedicated Wi-Fi coverage area). While working out of the office (where Wi-Fi
network may not be available), one can register the application using the Mobile Data (3G) network.
• Single Number Reach: Retains the identity of the corporate phone system while working away from the
office; so enhances business collaboration and lowers communication delays.
• Dial by Extension: Flexibility to reach to office users with direct extension number dialing.
• Corporate Directory Access: Enhance business collaboration with one-touch access to the Corporate
Directory contacts using the SARVAM UCS Global Directory.
• Video Support: The application offers the added advantage of Video Calling.
• Presence: Supports changing your Presence status as well as you can view the Presence status of other
extension users.
• Busy Lamp Field (BLF): Using BLF you can monitor the status of another extension or trunk and confirm
whether it is available or busy or ringing or on hold.
• IM and SMS: The application allows you to send IM and SMS to remote users.
• One Touch Transfer: You can transfer an ongoing call to a fixed extension without entering the number of
that extension and without putting the call on hold. Similarly, you can also transfer a call from the fixed
extension to your application.
• Voicemail Access: Access to the corporate Voicemail System from any location ensures no opportunity is
lost.
• Multiple Call Support: With multiple call support, you can easily handle multiple incoming calls, merge
and split calls apart, and place users on hold with a simple tap. With this Android Application, it’s like taking
your deskphone on the road.
• Wi-Fi to Cellular Handover and vise versa: The application can automatically move an active call from
the application to your cellular number on the cellular network and vise versa, without disconnecting the
call and having to redial.

• Better Voice Quality: Using customized codec settings, enhanced voice output is available. If you are
aware of the bandwidth and the network criteria of your location, you can select the proper codec from the
application to get high quality voice output.
• Standard Telephone Features: Provides intuitive access to Keypad, Contacts, Call Logs and more,
based on the Native Android design. One-touch access to call feature options during VoIP (Voice over IP)
calls including Adding a New Call, Mute, Hold, Transfer and Speakerphone. Also provides DTMF support
to enter numbers using an Auto Attendant.
• Cost Effective Calling: If you are using the enterprise Wi-Fi network to register Matrix VARTA ADR100
with the SARVAM UCS; calls made from the application will be almost free. Even if you are using the
application via 3G network during roaming, external calls can be made using the SARVAM UCS trunks
and thus reducing mobile calling and roaming charges.
• Multiple Language Support: The application can be viewed in six different languages including English,
• Application diagnostics: Supports logging and sending of log files to concerned recipients by e-mail.
These logs are used by the system engineer and/ or Matrix support engineers for troubleshooting.
Installing VARTA ADR100

For detailed instruction to install VARTA ADR100, refer to the Matrix VARTA ADR100 User Guide.
Configuring VARTA ADR100

For detailed instructions on how to configure VARTA ADR100, see “Configuring Matrix VARTA ADR100/AMP100
UC Clients”.
Operating VARTA ADR100

Refer to Matrix VARTA ADR100 User Guide for instructions on operating the features of SARVAM UCS.

Matrix VARTA AMP100 UC Client
Matrix VARTA AMP100 is a proprietary SIP (Session Initiation Protocol) based UC Client running on iPhones,
delivering full-array of Matrix SARVAM UCS features to the user on-the-go along with an added advantage of video
calling. Through tight integration with the enterprise mobility features of the SARVAM UCS, Matrix VARTA AMP100
provides advanced call capabilities including Conferencing, Corporate Directory Access (Global Directory), Call
Logs and Conversation Recording with one-touch access. Other than these, you can take the advantage of using
UC features like Presence, IM, IM to SMS, Corporate VMS access to enhance your overall mobile experience.
all productivity features at fingertips that enhances speed of communication and collaboration with office users and
customers.
Make sure the phone in which you install Matrix VARTA AMP100, runs on iOS7.
Key Features
• System Extension: Matrix VARTA AMP100 becomes a mobile extension of the SARVAM UCS. As an
increased number of business professionals are using the collaborative tools found in smartphone devices
to help them in their work activities, this application offers businesses an easy way to integrate their
enterprises’ voice solutions within the iOS family.
• Advanced Call Capabilities: Access to features such as Callback, Dial-in Conference, Conversation
Recording and many more.
• Mobility: Matrix VARTA AMP100 provides you the mobility that you need in today's highly competitive
business environment; with the ability to access SARVAM UCS features easily once you are connected to
either Wi-Fi or 3G network. Considering the case of roaming users, one can register Matrix VARTA
AMP100 with the SARVAM UCS using the enterprise Wi-Fi network when working within the office (that is
within the organization's dedicated Wi-Fi coverage area). While working out of the office (where Wi-Fi
network may not be available), one can register the application using the Mobile Data (3G) network.

• Single Number Reach: Retains the identity of the corporate phone system while working away from the
office; so enhances business collaboration and lowers communication delays.
• Dial by Extension: Flexibility to reach to office users with direct extension number dialing.
• Corporate Directory Access: Enhances business collaboration with one-touch access to the Corporate
Directory contacts using the SARVAM UCS's Global Directory.
• Video Support: The application offers the added advantage of Video Calling.
• Presence: Supports changing your Presence status as well as viewing Presence status of other extension
users.
• Busy Lamp Field (BLF): Using BLF you can monitor the status of another extension or trunk and confirm
whether it is available or busy or ringing or on hold.
• IM and SMS: The application allows you to send IM and SMS to remote users. It also supports the Emoji
keyboard to add Emoticons (Smileys) in your messages.
• One Touch Transfer: You can transfer an ongoing call to a fixed extension without entering the number of
that extension and without putting the call on hold. Similarly, you can also transfer a call from the fixed
extension to your application.
• Voicemail Access: Access to the corporate Voicemail from any location ensures no opportunity is lost.
• Multiple Call Support: With multiple call support, you can easily handle multiple incoming calls, merge
and split calls apart, and place users on hold with a simple tap. With this iPhone application, it’s like taking
your deskphone on the road.
• Cellular to Wi-Fi Handover and vice-versa: You can move an active call from the Cellular number (on
the Cellular network) to your application (registered using Wi-Fi network) without disconnecting or redialing
the number. Similarly Wi-Fi to Cellular Handover is also possible where you can manually handover your
call from the Wi-Fi to the Cellular network.
• Better Voice and Video Quality: Using customized codec settings and video quality preferences,
enhanced voice output and video rendering are available. If you are aware of the bandwidth and the
network criteria of your location, you can select the proper codec and video quality option from the
application to get optimum audio and/or video output.
• Standard Telephone Features: Provides intuitive access to Keypad, Contacts, Call Logs and more. One
touch access to call feature options during VoIP (Voice over IP) calls including Adding a New Call, Mute,
Hold, Transfer and Speakerphone. Also provides DTMF support to enter numbers using an Auto
Attendant.
• Cost Effective Calling: If you are using the enterprise Wi-Fi network to register Matrix VARTA AMP100
with the SARVAM UCS; calls made from the application will be almost free. Even if you are using the
application via 3G network during roaming, external calls can be made using the SARVAM UCS trunks
which reduces calling and roaming charges to a significant amount.
• Multiple Language Support: The application can be viewed in six different languages including English,

• Application Diagnostics: Supports logging and sending of log files to concerned recipients by e-mail.
These logs are used by the system engineer and/or the Matrix support engineers for troubleshooting.
Installing VARTA AMP100

For detailed instruction to install VARTA AMP100, refer to the Matrix VARTA AMP100 User Guide.
Configuring VARTA AMP100

For detailed instructions on how to configure VARTA AMP100, see “Configuring Matrix VARTA ADR100/AMP100
UC Clients”.
Operating VARTA AMP100

Refer to Matrix VARTA AMP100 User Guide for instructions on operating the features of SARVAM UCS.
MATRIX VARTA WIN200
MATRIX VARTA WIN200, is a SIP (Session Initiation Protocol) based Unified Communication Client running on
Windows OS, delivering full-array of the System features to the user on-the-go along with an added advantage of
video calling. Through tight integration with the enterprise features of the System, UC Client provides advance call
capabilities including Conferencing, Corporate Directory Access (Global Directory), Call Logs and Conversation
Recording with one-touch access. Other than these you can take the advantage of using UC features like
Presence, IM, IM to SMS, Corporate VMS access to enhance your overall mobile experience.
To use MATRIX VARTA WIN200 make sure you have:

• Purchased and activated the VARTA Essential or VARTA Professional or VARTA Collaboration license.
For more details, see “License Management”.

Installing MATRIX VARTA WIN200
For detailed instruction to install MATRIX VARTA WIN200, refer to the MATRIX VARTA WIN200 User Guide.
Configuring MATRIX VARTA WIN200

For detailed instructions on how to configure MATRIX VARTA WIN200, see “Configuring Matrix VARTA WIN200
UC Client”.
Operating MATRIX VARTA WIN200

Refer to MATRIX VARTA WIN200 User Guide for instructions on operating the features of SARVAM UCS.

Flash Timer
What is Flash?
Pulse dialing is a type of signaling in which codes (digits) are dialed in pulses. A hook switch or a Flash key is
generally used to dial this code. Technically, Flash is breaking the loop current for 200 milliseconds to 900 ms.
Please note that since this code is not simulated in standard DTMF convention, one cannot dial it in DTMF mode.
Flash timer signifies the time period for which the loop current breaks. Flash timer is programmable. Flash timer
ranges from 083 ms to 999 ms. By default, Flash Timer is 600 ms.
Where is it used?
Extensions dial flash to use few System features and also to use few PSTN features. Flash is used in following
cases:
• While transferring a call from one extension to another.

• While making conference between few extensions.
• While toggling call between two extensions.
• While using call-waiting facility available from PSTN exchange.
Now a days, more number of basic service providers and different types of advanced electronic telephone
exchanges are prevailing. It is possible that one service provider interprets breaking of loop current for 300 ms as
flash, other service provider interprets breaking of loop current for 900 ms as flash and system interprets breaking
of loop current for 600 ms as flash. Hence if the system engineer sets the flash timer to 600 ms then he might not
be able to use features provided by the service provider interpreting 900 ms for flash. To take care of this situation,
SARVAM UCS offers Flexibility to program different flash timers for both extensions and trunks.
How to configure
To set the timer to the desired value:
• for the SLT ports, refer “SLT Hardware Template” .
• for CO Trunks, refer “CO Hardware Template”.
• for E&M ports, refer “E&M Feature Template”.
• Many times it happens that while transferring the call, the call either gets disconnected or is not
transferred.
• This happens due to mismatch of time for which the hook switch is pressed, if used for transferring
the call hence it is advisable to use Flash Key of the telephone instrument, instead of hook switch.
• This problem may occur with Flash Key also, if the timer for the Flash Key on the telephone
instrument and the flash timer of the system are not set properly.
• Few telephone instruments have flash timers set to 800 ms. In such case the call does not get
transferred because the flash timer of all extensions is set at 600 ms by default. In such cases the
flash timer of the extension where the phone is connected should be increased to 800 ms.

Flashing on Trunks (Continued Dialing)
What’s this?
Public exchanges support features like call waiting, call forward, etc. To be able to use these features, users need
to dial certain codes during speech.
When a system is connected between the user and the central office, the codes for dialing the features of the
central office may clash with the codes for accessing the features of the system, making it difficult for users to
access the features of the central office while in speech.
To overcome this, SARVAM UCS supports Flashing on Trunks (Continued Dialing), which informs the system
about the codes dialed for the features of the central office on trunks by extension users.
How to configure
To be able to use this feature, Continued Dialing must be allowed to the extension in its “Class of Service (COS)” .
How to use

While in speech on trunk,
• Press ‘Transfer’ Key, dial * and the Desired Service Provider Code.
Or
• Press DSS Key assigned to Flashing on Trunks (if programmed).
• Dial the Desired Service Provider Code.
For SLT Users
While in speech on trunk,
• Press ‘Flash’.
• Dial *
• Dial the Desired Service Provider Code.
Example:
To use Call Waiting facility of service provider exchange from an SLT extension, follow the steps given below:
1 Dial Flash-*. This informs the system to pass following code.
2 Dial Flash-1. Speech with second call.
4 Dial Flash-1. Speech with first call.
6 Dial Flash-1. Speech with second call.

Flexible Numbers
What’s this?
SARVAM UCS offers Flexibility to assign a code of your choice to access an extension. This code is called Flexible
number. For example, to access first SLT having software port 001, one has to dial 2001. It is possible to change
this code to any other number of your choice.
SARVAM UCS offers the following types of extensions, namely SLT, DKP, Magneto, ISDN Terminals, SIP
Extensions, Radio Extensions, Virtual Extensions and Department Groups. The system loads default access codes
to all extensions on first power ON. Later on the extensions can be assigned default Flexible numbers using a
command.
The Default Access Codes for the Extensions are given below:
Software Port Default Access Codes (Flexible Numbers)
001 to 240 2001 to 2240 for SLT Extensions
01 to 96 3001 to 3096 for DKP Extensions
01 to 16 Blank for Magneto Ports
01 to 64 Blank for ISDN Terminals
001 to 999 Blank for SIP Extensions (This is the SIP ID)
01 to 16 Bank for Radio Extensions
01 to 64 Blank for Virtual Extensions
01 to 16 3901 to 3916 for Department Groups
SARVAM UCS also allows you to assign a Flexible Number to each extension individually or to a range of
extensions simultaneously.
Configuring Flexible Numbers (Access Codes)
Assigning Access Codes to Extensions

You can configure the Flexible Numbers using Jeeves as well as commands. For instructions, see “Configuring
SLT Extensions”, “Configuring DKP Extensions”, “Configuring ISDN Terminals”, “Configuring SIP Extension
Settings as per the Extended Phone Type”, “Virtual Extension”, “Configuring Radio Interface”, “Configuring
Magneto Interface”and “Department Call”.
• It is possible to have maximum of 6 digit flexible numbers.
• It is possible to clear the flexible number of a extension, range of extension and all extensions.
• Flexible numbers are the codes dialed from dial phase to call another extension. These flexible
numbers should be unique and should not match with either other SLT extensions or DKP extensions
or any of the features available from the dial phase.

• Flexible number having common digits can be assigned to another extension. Please refer “Conflict
Dialing” for more details.
• Same flexible number cannot be assigned to two different extensions.
• Use flexible numbers for all the features used from User mode and SA mode. Software port numbers
are to be used only from the SE mode.
• When the access code of a extension is cleared; its flexible number becomes null or void.
• If access code of a extension is cleared, one cannot call that extension. However the extension with
NULL flexible number can make calls as usual.
Assigning Access Codes to a Range of Extensions

You can assign Access Codes to a range of Extensions using Jeeves only.
If you are assigning a range of Extension Numbers (Access Codes) to the desired ports, and a match is found for
the same extension numbers, the system will clear these extension numbers from the existing database. The new
extension numbers will be assigned according to the given range.
Assigning Extension Numbers through Extn. Numbers Range has a priority over Extension Numbers assigned on
individual ports.
To assign a range of Extension Numbers,
• Under Configuration, click Access Codes.
• Click Extn Numbers in Range to open the page.

• Against each Index configure the following:
• Extension Type: Select the Extension Type. You can select SLT,DKP, SIP, ISDN Terminal, Magneto,
Radio, Department Group or Virtual Extension.
• Start S/W Port Number: Enter the Software Port Number from which you want the system to start
assigning the desired extension numbers.
• Define the range of Station Access Codes(Extension Number/Flexible Number/Access Code) that you
wish to assign to the Extension Type you selected in Start Extension Number and End Extension
Number.
For the given range of extension numbers, the system will assign extension numbers from the software
port number specified in Start S/W Port Number for this particular entry. The range of extension
numbers will be assigned in ascending order of the Software Port Number.
For example:
Extension Type you selected is SLT
Start Software Port Number is 1
Start Extension Number is 2001
End Extension Number is 2100
The system will assign Extension Number 2001 to Software Port Number 1, 2002 to Software Port
Number 2 and so on. The system will assign the last Extension Number 2100 to Software Port Number
100.
If you want to assign access codes starting with # or * to a range of extensions, make sure both Start and
End extension numbers begin with # or *.
• Click Submit.
To check the Extension Numbers (Access Codes) assigned by the system,
• Click the Extension Type, in this case, under Configuration click SLT Configuration.
• Click SLT Parameters.
• Similarly you can assign the Access Codes (Flexible Numbers) to DKP, SIP, ISDN Terminal, Magneto,
Radio, Department Group or Virtual Extension.

Floor Service
What's this?
The Floor Service feature allows you to provide a common access code to extension users which they can dial to
call floor service.
Essentially a hospitality feature, Floor Service is also useful in offices. Floor service can be any administration or
service department in the building, such as a stationery room, back office, backroom, photocopy/ mail room,
secretarial assistance, concierge/janitor, Storeroom.
Just as all extension users can reach the Operator by dialing the common access code '9', they can reach the floor
service by dialing a common access code, '38'. This is the default Floor Service access code, for all geographical
regions where SARVAM UCS is installed.
This feature can be used in:
• Multi-storied buildings, which have floor service (pantry, mail sorting, house keeping, janitor, coffee room,
refreshment area) for each floor. The SARVAM UCS can be programmed to land calls made by extension
users dialing the common access code '38' on the floor service extensions of their respective floors.
• Offices that have a centralized floor service, instead of one on each floor. The SARVAM UCS can be
programmed to land calls made from all extension phones by dialing '38' on the common floor service
extensions.
This feature requires a license. To use this feature you must purchase the license for Hospitality. Refer the
topic “License Management” to know more.
How it works
For example, Midas Towers houses different departments on each floor. Each floor has Floor Service.
Extensions 2001 to 2010 are on the first floor, 2011 to 2020 on the second floor, and 3001 to 3010 on the third floor.
The floor service extensions are numbered as 2012 on the first floor, 2022 on the second floor and 3012 on the
third floor.
With the Floor Service programmed for each floor, when the extension user 2001 dials '38', the call will land on the
service extension 2012, assigned to room service on the first floor. Similarly, when the extension user 3008 on the
third floor dials '38', the call will land on the service extension 3012 on the third floor.
If Midas Towers had a single floor service extension 2012 for all floors, with Floor service programmed, calls made
from all extensions by dialing '38' would land on extension 2012 only.
How to configure
Programming the Floor Service feature involves the following steps:
1. Creating a routing group for each floor. Include Floor service extensions of a floor in a routing group
prepared for that floor.

2. Assigning a routing group (number) in the Floor Service feature in the Station Advanced Feature
Template. Prepare a different Station Advance Feature Template for each floor.
3. Applying the Station Advanced Feature Template (with the Floor service group programmed) to the
extension. This will assign the extensions to the routing group programmed in the Template.
If the Enterprise/Building has centralized floor service, you only need to create a single Routing Group with
service extensions, as required. This routing group number can be programmed on a common Station
Advanced Feature Template which will be applied to all extensions.
Floor Service parameters can be programmed using Jeeves and a Telephone.
Programming Floor Service using Jeeves

• Under Configuration, click Routing Group.
• Choose the Routing Group number (01-96) you want to use as floor service group.
You can program different routing groups for different floors. In each routing group you can program
maximum 32 service extensions as 'members'.
• For routing group to be used as floor service, program the following parameters:
• Rotation Flag: With this flag, you can enable or disable the rotation of calls in the routing group which
has multiple 'member' extensions. When enabled, each fresh call will land on the extension which is
next to the one that received the last call. This ensures equal distribution of incoming calls to all the
destinations within the routing group. The flag has no relevance if the routing group has only one
member extension.
• When member rejects the call, place the call again: If any SIP/DKP member extension rejects an
incoming call and the system again checks the routing group sequence, you can allow or restrict
placing the same call on this extension. Select the When member rejects the call, place the call again
check box, if you want the system to place the call again on the extension.

If this check box is cleared (disabled) and you have selected the Continuous check box, the extension
that rejects the call stops ringing while the system hunts for other extensions in the routing group to
land the call.
• Member Type: Select the 'Member Type'. You can select SLT, DKP, SIP, Virtual Extensions, ISDN
Terminal, OGTBG or the Voice Mail Auto Attendant.
Program only as many extensions as you want in the routing group and set the remaining Member
Types to 'None'.
For example: if you want to program only one extension in the routing group, set the Member Type in
the remaining columns (Member 02-Member 32) to 'None.'
• Port Number: Enter the software port number on which the SLT/DKP/SIP/Virtual Extension/ISDN
Terminal is connected.
If you have selected OTBG then enter the OTBG number here.
Port Type, select the VMAA Menu to assign to the respective routing group.
• Ring Timer(s): This timer defines the time for which the extension, on which the call lands, should ring.
By default, the ring timer is set to 015 seconds and can be changed.
• Continuous Ring Flag: With this flag, you can set an extension to ring continuously until the call is
answered. The first extension will continue to ring even as the system hunts for other extensions in the
routing group to land the call. If the call still remains unanswered, the system will return the call to the
first extension once again. This flag is of no relevance, if there is only one member extension in a
routing group.
• Repeat the above steps to include other floor service extensions in the routing group.
• Click Station Advanced Feature Template to open the page.
All extensions are assigned the Advanced Feature Template 01, by default. If the enterprise requires
separate floor-service group for each floor, program a separate Station Advanced Feature Template for the
extensions of each floor.
• Select a Station Advance Feature Template number to be assigned to the user extensions of a floor. For
example: Template number 02 for extensions 2001 to 2010 are on the first floor, Template number 03 for
extensions, 2011 to 2020 on the second floor, and Template number 04 for extensions 3001 to 3010 on the
third floor.

• Scroll with the horizontal bar to reach the column Floor Service of the selected Templates. Enter the
Routing Group number you want to use as floor service group for that particular Station Advance Feature
Template.
• Now, apply the Station Advanced Feature Templates (with floor service routing groups programmed) to the
extensions of the respective floors. For example: Template number 02 on extensions 2001 to 2010 are on
the first floor, Template number 03 on extensions, 2011 to 2020 on the second floor, and Template number
04 on extensions 3001 to 3010 on the third floor.
If extensions are SLT, assign the Template on the SLT Parameters page.
If extensions are DKP, assign the Template on the DKP Parameters page.
If extensions are ISDN Terminals, assign the Template on the ISDN Parameters page.
If extensions are Virtual Extensions, assign the Template on the Virtual Extensions page.
Refer the topic “Station Advanced Feature Template” for instructions.
Programming Floor Service using a Telephone

• Enter SE mode.
To program a routing group with member extensions, dial:

Where,
Destination Index is from 01 to 32

Port Type is the 'Member type':
00 for None
01 for SLT
02 for DKP
28 for ISDN terminal
Port Number is the Software port number259 on which the floor service member extension SLT, DKP,
ISDN Terminal is attached.
Software port number of the SLT, from 001 to 512.
Software port number of the DKP, from 001 to 128.
Software port number of the ISDN Terminal, from 01 to 64.
Software port number of the SIP Extension, from 001 to 999.
Software port number of the Virtual Extension, from 01 to 64.
To program the Ring Timer for the routing group, dial:

• 6503-1-Routing Group-Destination Index-Ring Timer
Where,
Routing Group is the number of the Routing Group 01 to 96
Ring Timer is from 000 to 255 seconds (default: 015 seconds).
To program the Continuous Ring Flag for the routing group, dial:
• 6504-1-Routing Group-Destination Index-Flag
Where,
Continuous Ring Flag is
0 for disable continuous ring (each member extension in the group will ring for the programmed 'Ring
Timer' for the group)
1 for enable continuous ring (the first extension in the group will ring till the call is answered)
To program the routing group in a Station Advanced Feature Template, dial:

• 5602-1-Template Number-11-Routing Group
Where,
Template Number is from 01 to 50
11 is the feature code for Floor Service
Routing Group is from 01 to 96260
To apply the Station Advanced Feature Template now programmed with the Routing Group to SLTs,
DKPs, ISDN Terminals, SIP extensions, refer the topic “Customizing Station Advanced Feature
Template using a Telephone”.
• Exit SE mode.
259. Refer the topic 'Software Port and Hardware ID' .

260. Enter the number of the routing group you programmed as Floor Service group.

How to use
To be able to use floor service, extension users may dial the default Access Code defined for Floor Service: 38.
Check with your System Engineer if this access code has been changed and dial the new access code
obtained from the System Engineer.

• Go OFF-Hook.
• Press DSS Key assigned to Floor Service (if programmed)
OR
• Dial Access Code '38'

• Talk
• Go ON-Hook
For SLT Users

• Lift handset
• Dial, Access Code '38'
• Talk

Follow Me
What’s this?
Using this feature, extension users can make your calls follow you wherever you go. Extension users can receive
their calls on another extension, whenever they want.
How it works
• A’s extension number is 2001.
• B’s extension number is 2003.
• A is currently at B’s extension.
• A wants to receive calls from extension 2001 on extension 2003.
• A sets Call Follow Me on extension 2003.
• All calls landing on A’s extension 2001 will be forwarded to extension 2003.
• When A returns to extension 2001, A cancels Call Follow Me.
• The extensions dial tone changes to feature tone if its calls are forwarded.
• Multiple users can use ‘Follow Me’ from the same extension.
• Follow Me can be overwritten. Extension A sets Follow-Me on extension B. After a period of time; goes
to extension C. A can receive calls on extension C by setting Follow Me on extension C. Follow Me set
by A on extension B will be cancelled.
• Follow Me cannot be chained. If extension A sets Follow Me to extension B. And extension B sets
Follow Me on extension C, Follow Me of extension A is automatically cancelled.
• DND is given priority over Call Follow Me feature.
Also see “Call Forward”, “Class of Service (COS)” and “Do Not Disturb (DND)”
How to configure
To be able to use Follow Me, extension users must have Call Forward feature enabled in their Class of Service for
the time zone. For instructions, see “Class of Service (COS)” and “Station Basic Feature Template”.
How to use
To set Follow Me from another extension,
• Press ‘Forward’ Key of the other extension.

OR
Dial 135.
• Enter your extension number.

• Enter your user password.
To cancel Follow me, from your extension,
• Press ‘Forward’ Key of your extension phone.

• Select ‘Cancel’
OR
Dial 130.
For SLT Users
To set follow me from another extension,
• Lift the handset of the other extension.

• Dial 135
• Dial your extension number.
• Dial your user password.
To cancel Follow me,
• Lift the handset of your extension.

• Dial 130

Forced Answer
What’s this?
Extension users can force other extension users to answer their calls when there is no response from the called
extensions.
How it works
Forced Answer can be requested by the calling extension. The calling extension may be an SLT, a DKP or an
Extended IP Phone. However, the called extension (being forced to answer) must be either a DKP or an Extended
IP Phone.
• Extension user A (SLT) calls extension user B (DKP/Extended IP Phone).

• B’s phone is ringing, but B does not answer.
• A dials Forced Answer feature code.
• The speaker of B’s phone’s is turned on (goes OFF-Hook).
• A may now talk to B.
Forced Answer can be used when the called extension idle or ringing.
How to configure
To be able to use Forced Answer, extension users must have this feature enabled in their Class of Service for the
time zone. For instructions, see “Class of Service (COS)” and “Station Basic Feature Template”.
How to use
To use forced answer on an extension:

• Dial the desired extension number.
• Press the DSS Key assigned to Forced Answer on Ring Back tone.
OR
Dial 5 on Ring Back Tone.
• The Ring Back Tone stops.
• The called extension’s speaker is turned on.
• You are in speech with the called extension.
• You may talk.
For SLT Users
To use forced answer on an extension:

• Dial 5 on Ring Back tone.
• The Ring Back Tone stops.
• The called extension phone’s speaker is turned on.

• You may talk.
You can also dial ‘5’, the feature code for Forced Answer, immediately after dialing the desired extension
number, instead of dialing it during the Ring Back Tone. This way, you can talk to the desired extension
user without waiting for the called extension user to answer your call.

Forced Call Disconnection
What’s this?
Forced Call Disconnection enables extension users to disconnect a busy extension or a trunk at will, and free the
system resources (access to extension and trunk) for themselves.
How it works
Forced Call Disconnection of an Extension:
• A, B and C are extensions.
• A and B are in speech.
• C calls B and finds it busy.
• C uses Forced Call Disconnection by dialing the feature command.
• C gets confirmation tone, while A and B get error tone.
Forced Call Disconnection of a Trunk:

• A and B are extensions. C is the external party.
• A is in speech with C on Trunk 1.
• B grabs Trunk 1 using Selective Port Access, but gets busy tone.
• B uses Forced Call Disconnection by dialing the feature command.
• B gets confirmation tone. A gets disconnected and gets error tone.
• B must grab Trunk 1 again to get the dial tone of the network.
To be able to use Forced Call Disconnection, the extension user must have a higher “Priority” than the
extension user whom he/she tries to forcibly disconnect.
To be able to use Forced Call Disconnection on a busy trunk, the extension user must have grabbed that
trunk using “Selective Port Access”. If the extension user has grabbed the trunk using a Trunk Access
Code, the feature code to dial Forced Call Disconnection will not work.
In PLCC applications, Forced Call Disconnection can be used in a chain to reach the last Exchange
through many tandem exchanges in between.
Forced Call Disconnection is not supported on SIP Trunks, BRI and T1E1 lines.
You are advised to restrict access to this feature only to important extension users. Extension Users who
are allowed this feature are advised to use it judiciously.
How to configure
To be able to use Forced Call Disconnection, the extension must have:
• Forced Release feature enabled in the Class of Service. For instructions see “Class of Service (COS)”
and “Station Basic Feature Template”.
• As Forced Call Disconnection on a busy trunk is possible only if the extension user has grabbed that trunk
using Selective Port Access, this feature must be enabled in the Class of Service of the extension. For
instructions see “Class of Service (COS)” and “Station Basic Feature Template”.

• Higher Priority assigned than other extensions. See “Priority” for instructions.
How to use
To forcibly disconnect a busy extension/trunk261:
• Press the DSS Key assigned to Forced Call Disconnection on Busy tone.
OR
Dial #* on Busy tone.

• You may now dial the extension number/grab the trunk.
For SLT Users
To forcibly disconnect a busy extension/trunk:
• Dial #* on Busy tone.

• You may now dial the extension number/grab the trunk.
261. Only if you have grabbed this trunk using Selective Port Access.

Gain Settings
What's this?
To avoid noise or echo during speech, you must set the speech volume levels on the ports. SARVAM UCS allows
you to set the speech volume levels for the following port types—CO, Mobile, SIP and SLT.
The speech volume levels can be adjusted by increasing or decreasing the Gain Settings provided on each port
type.
How it works
A call received on the CO port can be placed on any of the following ports—DKP, SLT, Mobile, T1E1, SIP, BRI,
E&M. The speech volume levels differ according to the port type. Hence, on the CO port you must set the speech
volume levels for each of these port types.
In this case, let us assume that the call on the CO Trunk is to be placed on the SLT Port.
Before placing the call on the SLT Port, the system applies the CO to SLT Gain Setting (Receive and Transmit Gain
settings) configured on the CO Trunk to adjust the speech volume level.
When the call is placed on the SLT Port, the SLT to CO Gain Settings (Receive and Transmit Gain settings) on the
SLT Port are applied to adjust the speech volume level.
Hence, you can set different speech volume levels for each port type and the system automatically detects and
applies these gain settings for each port type.
How to configure
• Click CO Gain Settings to open the page.
• Select the CO Gain Settings Template Number you want to assign in the CO Hardware Template and
configure the following Transmit and Receive Gain Settings:

• CO - System (Tx-Gain and Rx-Gain): Configure the Gain Settings that you want the system to apply
on the CO port with respect to the system (for example Call Conference). These will not be applicable
for any other port type. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
• CO - Voice Mail (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply
on the CO port when the incoming calls on the CO port are being answered by the VMS. Valid Range
for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
• CO - Voice Module (Tx-Gain and Rx-Gain): Configure the Gain Settings that you want the system to
apply on the CO port when incoming calls are answered using Auto Attendant or Trunk Auto Answer.
Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
• CO - Call Progress Tones (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the
system to apply on the CO port while playing Call Progress Tones. Valid Range for Tx gain: +10dB to -
15dB and Rx Gain +10dB to -15dB.
• CO - Radio (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on
the CO port when the CO port is connected to any Radio Port of the system during an incoming or
outgoing call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
• CO - SLT (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on the
CO port when the CO port is connected to any FXS Port of the system during an incoming or outgoing
call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
• CO - CO (Tx-Gain and Rx-Gain): Configure the Gain setting that you want the system to apply on the
CO port when the CO port is connected to another CO Port of the system during an incoming or
• CO - DKP (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on
the CO port when the CO port is connected to any DKP Port of the system during an incoming or
• CO - SIP (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on the
CO port when the CO port is connected to any SIP Trunk or SIP Extension of the system during an
incoming or outgoing call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
• CO - Mobile (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on
the CO port when the CO port is connected to any Mobile Port of the system during an incoming or
• CO - T1E1/BRI (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply
on the CO port when the CO port is connected to any T1E1/BRI Port of the system during an incoming
or outgoing call. Valid Range for Tx gain: +10dB to -15dB and Rx Gain +10dB to -15dB.
• CO - E&M (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on
the CO port when the CO port is connected to any E&M Port of the system during an incoming or
• Similarly, to configure the Gain settings for the Mobile Port, click Mobile Gain Settings under Mobile
Configuration.

• For SIP Trunk / Extension, click SIP Gain Settings under VoIP Configuration.
• For SLT Port, click SLT Gain Settings under SLT Configuration.
Assigning the Gain Settings Templates

To apply the Gain Settings you configured for each port type, you must assign these templates on the respective
ports
• Assign the CO Gain Settings Template you configured in the CO Hardware Template, see “CO Hardware
Template”.
• Assign the Mobile Gain Settings Template you configured in the Mobile Port Parameters, see “Mobile Port
Parameters”.
• Assign the SIP Gain Settings Template you configured in the SIP Hardware Template, see “SIP Hardware
Template”.
• Assign the SLT Gain Settings Template you configured in the SLT Hardware Template, see “SLT
Hardware Template”.

Gateway Application-Answer Signaling
What's this?
When SARVAM UCS acts as a Gateway, this feature is used to convey the call maturity information on source port,
when call made using destination port gets matured. This information can be useful for billing equipment connected
at calling party end.
When the system acts as a Gateway, the Source Port is the port on which call originates and the Destination Port is
the port on which the call terminates.
How it works
• This feature of answer signaling is applicable only for DISA option selected as 'CLI Authentication-one call-
Ans. Sig.' in the DISA-CLI Authentication Table. Refer chapter “Direct Inward System Access (DISA)”.
• The SARVAM UCS works as gateway for routing the call to the destination.
• If the calling party's number is programmed in Table - 'DISA-CLI Authentication', the SARVAM UCS will
consider the calling party as successfully logged in as station which is programmed as "Auto Login
station". The call will get answered by the SARVAM UCS. The caller will get dial tone.
• Caller can dial the desired number.
• SARVAM UCS will route the call from the destination port.
• When called party answers the call (that is, call on destination port gets matured), the Answer Signaling
will be done on source port, if enabled.
• Answer Signaling will be done in form of DTMF digit string as programmed on the source port.
• If Answer Signaling is disabled on the source port, the DTMF digit string will not be dialed out, even if it is
programmed.
• On receiving these digits, Billing equipment/System with which the calling party is connected, can consider
the call is matured and start billing.
How to configure
To configure 'Gateway Application-Answer Signaling' flag and 'DTMF String' on desired trunk refer following
chapters:
• For CO Trunk, Refer chapter “CO Hardware Template”

• For E&M Trunk, Refer chapter “E&M Feature Template”
• For Mobile Trunk, Refer chapter “Configuring Mobile Trunks”
• For T1E1 Trunk, Refer chapter “Configuring PRI Trunks”
• For BRI Trunk, Refer chapter “Configuring BRI Trunks”
• For SIP Trunk, Refer chapter “Configuring SIP Trunks”

By default, 'Gateway Application-Answer Signaling' is disabled for all ports. By default, DTMF Digit String' is
programmed as 'CCC' on all ports.
Application:
Mumbai Delhi
Mobile FXS Mobile FXS

GFX11 GFX11
PSTN T1E1-1 T1E1-2 PSTN

SARVAM
N/w UCS N/w
Kolkata Chennai
Mobile FXS Mobile FXS

GFX11 GFX11
• Multiple GFX11s are installed at different places for PCO Application.
• One SARVAM UCS is installed at central place.
• On FXS port of each GFX11, 'Pay Phone' is connected.
• On GSM port of each GFX11, SIM card is installed.
• GFX11s are configured for multi stage dialing, in which when ever caller dials the number using pay
phone, GFX11 will store the number, it will first make a call to SARVAM UCS's T1E1 line on which DISA -
'CLI Auth- One Call - Ans. Sig.' is enabled.
• GFX11's SIM Numbers are programmed in 'DISA - CLI Auth.' table in SARVAM UCS.
• SARVAM UCS compares the Calling Number received on T1E1 Port with DISA-CLI Auth. Table.
• As the number is programmed in the table, SARVAM UCS answers the call and offers the trunk assigned
in Station Basic Feature Template of the station used as 'Auto Login'.
• When SARVAM UCS answers the call, the GFX11 sends the stored called number (which is actually
dialed by the caller) in DTMF digits.
• SARVAM UCS routes the call on this number using the offered trunk.
• When called party answers the call, SARVAM UCS will send 'DTMF Digit Strings' programmed on the
T1E1 Port (Source Port, on which call originated) as a Gateway Answer Signaling'.

• The Pay Phone connected with FXS Port of the GFX11 is also configured to understand the same DTMF
string as call maturity.
• Pay Phone will start billing only on receipt of the desired DTMF digits.

GPAX Application
What’s this?
GPAX application is one of the applications provided by SARVAM UCS and is used in commercial establishment/
Society/Organization, etc. Group Systems are installed, operated and maintained by organizations/agencies. The
owner will be treated as the main hirer. The private exchange (PSTN)/service provider will provide junctions to the
System and the owner will pay the rental of the junctions, and the call charges. The number of junctions provided to
the System should be adequate to carry the traffic. The owner/agencies of the Systems will be responsible for the
payment of all charges to the private exchange/service provider.
In this application say a System ‘A’ is connected to PSTN. System ‘A’ is given an exchange ID (say 2837). A
number of stations (say 001-100) can be connected to System ‘A’. When a station 015 dials 2837025, System ‘A’
interpret this number to be dialed for the same system. However, for dialing a station number belonging to same
system, it is not necessary to prefix the station number with exchange ID. When a station 020 dials 2834537,
System ‘A’ does not interpret this to be dialed for the same system and hence will dial the digits on the trunk.
When a station user picks up the handset and dials any digit except the one programmed in the routing table will be
dialed on the trunk. If the user dials digit that is programmed in the routing table with Self-flag enabled, the system
will not dial the digits on the trunk since it would interpret these to be dialed for the same system.
For dialing the digits on the trunk it is required to program the routing table carefully. Route code should be
specified in route code column of routing table in association with Self Route Flag disable. This will make the call to
be routed on trunk which is specified in OG Trunk Bundle Group. For details refer “Closed User Group (CUG)” and
“Closed User Group-With Exchange ID”.
System A
T1
T2
PSTN 2837
Tn
S1 S2 Sn
001 002 100
In order to make internal calls, user is advised to program Routing Table such that one of the entries in routing table
should have route code as ‘#’, Strip digit count as ‘1’ and Self route flag as ‘1’ (enable). Doing so, when user picks
up handset and dials the required number with prefix ‘#’, the system interpret this number as an activity for the
internal users and waits for relevant access code.
Because of Strip digit count=1, the first digit dialed by the user (that is, # in this case) will be ignored and next digit
will be processed which could be a feature code or a station number.

In order to use system features, the feature codes should be prefixed by ‘#’. Whenever the user dials ‘#’ on picking
the handset, the System assumes it to be an activity for the internal users and waits for relevant access code.
In GPAX application, Answer Signaling on SLT and Answer Supervision on trunk shall be programmed properly.
Also the Disconnect Supervision parameter shall be programmed as appropriate for proper billing of calls.
Following figures shows how call is established between users using PSTN and SARVAM UCS-GPAX.
PSTN
Customer Site
C Makarpura Alkapuri
Subscriber
Card
SARVAM UCS - GPAX
FXO Subscriber B
(CO) Card
Inter Exchange
Trunk interface Inter Exchange
FXS card Trunk interface
(SLT) card
Inter
Exchange
Trunk card
1. Call from A to B is shown by
2. Call from B to A is shown by D
3. Call from C to A is shown by Sub Card
Customer Site PSTN
C
SARVAM UCS - GPAX
Trunk
(CO) Subscriber
Card
SLT
Card
Inter
A Exchange
Trunk
interface card
MFC Operator
1. Call from A to C is shown by

3. Call from C to A is shown by

Trunk Offer PSTN
SARVAM UCS - GPAX C
Trunk
(CO) Subscriber
Card
SLT
Card
Inter
Exchange
A Trunk
interface
card
MFC Operator
1. Call from A to C is shown by

2. Call from C to A is shown by
3. Call from Opt. to A is shown by

GPAX Billing
What’s this?
In commercial establishments/large societies/organizations etc. staff make calls from their rooms. It is required that
the cost of these calls is calculated so that amount of calls made by a staff member can be paid. GPAX provides a
facility which if enabled can calculate cost of each call if programmed properly.
GPAX has a dedicated memory space (commonly called buffer) to store details of each call. These calls are
retained in the buffer even during power failure. Once the buffer is 100% full, the new call overwrites the oldest one,
that is First In First Out (FIFO) method.
You can also generate various reports and the same can be routed on the computer from this buffer. It is
recommended that printing of various reports should be regularized on fixed dates. This should be done regardless
of whether the buffer is full or not.This will prevent spilling and subsequent loss of data.
You can enable or disable call logging for individual trunk. This gives flexibility of monitoring only important (STD)
trunks. This also prevents frequent spilling of the buffer, as new local calls will not be recorded.
How to configure
To calculate the total cost of a call, you must configure the relevant parameters in “Call Cost Calculation (CCC)”,
“Call Duration Control (CDC)”, “Call Budget on Extension”.
In order to bill internal calls between GPAX users, GPAX charge internal calls option must be enabled. For
instructions see, “Station Advanced Feature Template”. If GPAX charge internal calls option is enabled, this call will
be recorded in the Station message detail recording-outgoing buffer. If GPAX charge internal calls option is
disabled, the call made to an internal station will not be billed and will be recorded in the SMDR-Internal buffer as
normal internal call.

Handover and Handoff
What is this?
Handover allows you to move an active VARTA Mobile UC Client call from the Wi-Fi network to the cellular number
(in the cellular network). This is useful when you have an ongoing call and you leave the Wi-Fi network, or if there
are voice quality issues over the Wi-Fi network.
Handoff is when you are back into the Wi-Fi network you can move the call from the cellular number to the VARTA
Mobile UC Client. The call is moved without being disconnected and redialing the number.
SARVAM UCS will serve the Handover request only if:

• the Cellular Number is configured in the Application Settings in the Mobile Client. Refer to the
respective Mobile Client User Guide for details.
or
• the Mobile Number is configured in the system. See “Configuring Matrix VARTA ADR100/AMP100 UC
Clients”.
If the SIP extension user is a VARTA AMP100 client (the application is in the background), Handover will
be possible but Handoff will not be possible.To know more refer to the MATRIX VARTA AMP100 User
Guide.
How to configure
To use this feature, make sure the Basic Features are enabled in the Class of Service assigned to you. For
instructions, see “Class of Service (COS)” and “Station Basic Feature Template”.

Help Desk
What’s this?
An organization may have a Centralized Information Office which provides information related to different
departments such as HR, IT, or General information. For each department in the organization, an extension number
can be defined as a Help Desk.
How it works
• Extension 2002 is defined as Help Desk for HR policies and general rules.
• Extension 2016 calls the Help Desk extension 2002.
• If the Help Desk extension is busy/not responding, an Auto Callback request is set automatically on the
Help Desk extension.
• The system will serve the auto callback request as soon as the Help Desk extension is available.
• The Help Desk extension calls back extension 2016.
Also see “Auto Call Back (ACB)”
How to configure
You can define an extension as ‘Help Desk’ by enabling the ‘Help Desk’ flag in its “Station Advanced Feature
Template”.

Holiday Table
What’s this?
The Holiday Table feature of SARVAM UCS enables you to configure incoming call management for holidays.
Using the Holiday Table feature of SARVAM UCS you can,

• define the landing destination for incoming calls on trunks on holidays.
• greet callers with customized holiday messages.
• determine the way extensions must work on the holidays.
You can configure a list of holidays in a single table.
How it works
In the Holiday Table, you need define the following:
• The Start and the End Dates and the Time to be considered as Holiday.
• The Time Zone to be considered for operating the time-zone based trunk and extension features262 during
the Date and Time configured as Holiday. The Time Zone for Holiday can be defined as Non-Working
Hours, Working Hours, or As per Time Table.
If you have Voice Mail System and are using the Voice Mail Auto Attendant as the landing destination for calls, on
holidays, you can play customized greeting messages to callers. For each holiday, you can play a different
message.
For example, A company, ABC Ltd., has the following requirements:

• December 23 to December 31, all the employees will be on a holiday. The callers must be greeted with a
holiday message.
• January 1 to January 4, few employees will be attending the office.
In this case you must define the following in the Holiday Table:
At Index 1,
• In Start, enter the starting date and time of the holiday in DD-MMM-HH-MM format, that is 23 DEC, 00:00
and in End enter the last date and time of the holiday, that is 31 DEC, 23:59.
• In Time Zone for Holiday, select Non-working hours.
• In Holiday Message select the customized holiday message number. The system will greet the callers
with this message.
262. Trunk Landing Group, Auto-Attendant, DISA are time-zone based features of Trunks configured in the Trunk Feature Template
assigned to trunks. Class of Service, Toll Control, and OG Trunk Bundle Group, are time-zone based features of extensions that
are configured in the Station Basic Feature Template assigned to extensions.

At Index 2,
• In Start, enter the starting date and time of the holiday in DD-MMM-HH-MM format, that is 31 DEC, 00:00
and in End enter the last date and time of the holiday, that is 4 JAN 23:59.
• In Time Zone for Holiday, select As per Time Table.
• In Holiday Message select None.
Time
Holiday
Index Holiday Name Zone for
Message
Holiday
Start End
(DD-MMM-HH-MM) (DD-MMM-HH-MM)
1 23 DEC 00 00 31 DEC 23 59 Christmas Non- 01

Holidays Working
Hours
2 31 DEC 00 00 04 JAN 23 59 Christmas As per

Holidays Time
Table
After you have defined the above parameters, this is how the feature Holiday Table works,
• On the set date and time, when SARVAM UCS detects a day as a holiday, it checks the configured Time
Zone for Holiday and whether Holiday Message is configured.
• When the Time Zone for Holiday is defined as Non-working hours,

• the system checks for and routes the incoming call on the trunk as per the Trunk Landing Group and
the Auto Attendant defined for the trunk for Non-Working hours in the Trunk Feature Template.
• customized Holiday Message, if configured, is played to the caller. The Holiday message is played in
place of the Welcome message for Non-Working hours.
• Extensions work according to the Class of Service, Toll Control and Outgoing Trunk Bundle Group
assigned to them for Non-working hours in the Station Basic Feature Template.
Similarly, if the Time Zone for Holiday is defined as Working Hours, the system will operate the time-zone based
features of the trunks and extensions according to Working Hours.
• When Time Zone for Holiday is As per Timetable,

• the system routes the call as per the Trunk Landing Group and Auto Attendant defined for the current
time zone in the Trunk Feature Template.
• customized Holiday message, if configured, is played to the caller in place of the Welcome message for
the current time zone.
• extensions work according to the Class of Service, Toll Control and Outgoing Trunk Bundle Group
assigned to them for the current time zone in the Station Basic Feature Template.
Day/Night Mode: You can set the system in Day/Night Mode, even when you have configured the Holiday Table.
In that case, the mode you select, Day (Working Hours) or Night (Non-working Hours) will override the Time Zone
you have selected for Holiday.

In order for the Time Zone for Holiday to come into effect, you must set the Day/Night Mode in System Parameters
to Operate System as per Timetable Assignment.
See “Day Night Mode” and “System Parameters” to know more about this feature.
How to configure
You can configure the Holiday Table from the SE as well as the SA mode.
Configuring Holiday Table

• Click Holiday Table.
Against each Index configure the following parameters:
• In Holiday configure the Start Date, Month and Time of the holiday and End Date, Month and Time of
the holiday.
Default, Start and End Date is Blank. Valid Range is from 01 to 31.
Start and End Month is Blank. Valid Range is from January to December.
Start and End Time is 00:00 (Hours:Minutes). Valid Range is from 00:00 to 23:59
• You can assign a Name to each time period you have defined as Holiday. For example, Christmas,
Independence Day, Thanksgiving. Default: Blank
• As the Time Zone for Holiday select the time zone for the period you have defined as Holiday:
Working Hours, Non-working Hours or As Per Time Table. Default: As Per Time Table.
• If the incoming calls are routed to the VMS Auto Attendant, select the Holiday Message number that
you want the system to play to the callers. Default: 01.

You can also configure the Holiday Table from the SA mode also. To do this,
• Click Holiday Table.
• Follow the same steps as given above to configure the Holiday Table.
Configuring Holiday Messages

To greet callers with customized holiday messages, you need to do the following configuration:
• Ensure that Voice Mail Auto Attendant is selected as the Auto Attendant in the Trunk Feature Template.
For detailed information, see “Trunk Feature Template”.
• You can either use the default Holiday messages or customize your messages by recording messages of
your preference. To know more about the default Holiday Messages and how to record customized
Holiday Messages, see “Recording Voice Messages”.

Hot Desking
What’s this?
Hot Desking enables extension users to use all the properties of their own extension from another extension.
Hot Desking is useful for people who are often away from their own desks and must work from another. Hot
Desking allows them to use all the features and facilities of their own extension from another.
How it works
This feature is supported on DKP and SLT extensions only.
Hot Desking is possible only between extensions of the same type: SLT to SLT and DKP to DKP extensions.
The User Password of both extensions involved in Hot Desking must not be 1111.
Hot Desking can be performed only when both the extensions are idle.
To perform Hot Desking two extensions are required:
• The Host Extension - the extension whose user performs the Hot Desking.
• The Hot Desk Extension - the extension on which Hot Desking is performed.
• When Hot Desk is performed from the Hot Desking extension, all the properties of the Host Extension are
copied to the Hot Desk Extension.
• On the Host Extension, the user cannot perform any activity except Cancel Hot Desking.
• You must cancel Hot Desk from both the Hot Desk Extension and the Host Extension.
• After cancelling Hot Desk, the Host Extension and the Hot Desk Extension acquire their original properties.
How to configure
For this feature to work, the feature 'Hot Desk' must be enabled in the Class of Service of the Host Extension and
the Hot Desk Extension. See “Class of Service (COS)” and “Station Basic Feature Template” for instructions.
How to use
For EON users

To perform Hot Desk:
• Go to the EON extension (Hot Desk Extension) with which you want to swap your EON extension (Host
Extension) properties.
• Press DSS Key assigned to Hot Desk.
OR

• Dial 1091
• Enter Host Extension number
• Enter Host Extension User Password
• Go ON-Hook.
The User password of both the extensions involved cannot be default password.
To cancel Hot Desk:
• On the Host Extension, press DSS Key assigned to Hot Desk.

OR
• Dial 1091
• Enter Hot Desk Extension number
• Enter Hot Desk Extension User Password
• Go ON-Hook.
• On the Hot Desk Extension, press DSS Key assigned to Hot Desk.
OR
• Dial 1091
• Enter own extension number (Hot Desk Extension Number)
• Enter own extension User Password
• You get confirmation, Hot Desk cleared
• Go ON-Hook.
For SLT Users

To perform Hot Desk:
• Go to the SLT extension (Hot Desk Extension) with which you want to swap your SLT extension (Host
Extension) properties.
• Lift the handset of the Hot Desk extension.
• Dial 1091
• Dial Host Extension number
• Dial Host Extension User Password
To cancel Hot Desk:
• On the Host Extension, dial 1091

• Dial Hot Desk Extension number
• Dial Hot Desk Extension User Password
• Go ON-Hook.
• On the Hot Desk Extension, dial 1091

• Dial own extension number (Hot Desk Extension Number)
• Dial own extension User Password
• You get confirmation, Hot Desk cleared
• Go ON-Hook.

Hotline
What’s this?
The Hotline feature connects the extension user immediately to a particular number or trunk, whenever the
extension user goes OFF-Hook.
You can set Hotline to connect immediately to another extension, to a Department Group, to an external number or
to an outgoing trunk.
Hotline set for external numbers and outgoing trunks is referred to as Hot Outward Dialing.
SARVAM UCS offers two types of Hotline/Hot Outward Dialing:

• Immediate: As soon as the extension user goes Off-Hook, the user gets connected to the desired hotline
extension number, department group, external number, or outgoing trunk. For this the Hotline Timer must
be set to ‘00’ seconds (default: 3 seconds).
• Delayed: When the extension user goes OFF-Hook, the system plays Dial Tone to the extension user and
waits for the Hotline Timer (default: 3 seconds). On the expiry of this timer, it connects the extension user
to the desired hotline extension number, department group, external number or outgoing trunk.
How it works
• Hotline/Hot Outward Dialing can be set from an SLT, DKP or Extended IP Phone extension, if the
extension has Hotline in its Class of Service.
• To be able to use Hotline/Hot Outward Dialing, extension users must do the following:
• Select the type of Hotline they want to set on their extension; whether to an internal Extension Number,
a Department Group, or an External Number or Outgoing Trunk.
• Configure the Hotline Timer. For Immediate Hotline, extension users must set the Hotline Timer to ‘00’
seconds. For Delayed Hotline, extension users can set the Timer as per their requirement.
• Here is an example of how Hotline works:
A frequently dials the number of B. So, A sets Hotline for B’s number and also sets the Hotline Timer to 5
seconds (Delayed Hotline).
• A goes Off-Hook
• SARVAM UCS plays dial tone and waits for 5 seconds
• If A dials a number within the Hotline Timer, SARVAM UCS outdials the number dialed by A.
• If A does not dial any digit within this time, SARVAM UCS dials B’s number.
• A gets connected to B.
If ‘Dial Tone’ timer of the system is less than Hotline Timer, the Hotline Timer will override the ‘Dial Tone’
timer. To know more about these timers, see “System Timers and Counts”.
• If A had set the Hotline Timer to ‘00’ seconds (Immediate Hotline), A would be connected to B as soon
as A goes Off-Hook.

• If A sets delayed Hot Outward Dialing for a Trunk or an External Number, the system will play dial tone
to A and wait for the duration of the Hotline Timer for A to dial digits. If A does not dial any digits within
this timer, the system connects A to the Trunk/External Number.
• If A sets immediate Hot Outward Dialing (Hotline Timer set to ‘00’ seconds), A will be connected to the
Trunk/External number as soon as A goes Off-Hook.
• Delayed Hotline/Hot Outward Dialing allows extension users to dial out other numbers or grab another
trunk, without having to cancel the Hotline/Hot Outward Dialing they have set for a particular number or
trunk.
How to configure
To be able to use Hotline, extension users must have this feature enabled in their “Class of Service (COS)” for the
time zone, as required.
How to use
Hotline can be set/canceled by users for their own extension, or for any other extension from the SA mode.
Hotline when set/canceled from the SA mode, will not depend on the assigned CoS.
Set/Cancel Hotline for Extension Users

The Operator or any extension user having access to System Administrator mode can set or cancel Hotline for
other extension users.
Set Hotline using SA Jeeves

• Click Submit.

• Click Hotline to expand.
• Select the type of Hotline you want to set for the extension user from the following:
• To set Hotline for an Extension or Department Group, select the radio button Hotline to Station or
Department Group. Enter the Extension number or the Department Group Number in the
corresponding box. Default: Blank.
• To set Hotline for a group of trunks, select the radio button Hot Outward Dialing to Group of Trunks
using TAC and select the Trunk Access Code from the corresponding drop down list.
Each Trunk Access Code has a group of trunks for which Hotline will be set.
• To set Hotline for an external number, select the radio button Hot Outward Dialing to External
Number. Enter the external number in the corresponding box and in Using TAC select a TAC from the
drop down list. Using a free trunk from this TAC the external number will be dialled out by the system.
• Click the Set Hotline button to set Hotline.
The message “Hotline is set” appears.
• To set Delayed Hotline, in Hotline Timer enter the desired time in seconds.
On the expiry of this timer, it connects the extension user to the desired hotline extension number,
department group, external number or outgoing trunk.
• Click the Set button to set Delayed Hotline.
• To cancel Hotline, click the Cancel Hotline button.

Set/Cancel Hotline by Extension Users
To set Hotline on an Extension / Department Group,

• Press DSS key assigned to Hotline.
• Scroll to select ‘Set Hotline: Stn/ Dept’, press enter key
• Enter Extension / Department Group Number
OR
• Dial 151 and enter Extension Number / Department Group Number
To set Hot Outward Dialing for Trunk,

• Scroll to select ‘Set Hotline: OG Trunk’, press enter key
• Enter TAC
OR
• Dial 152-TAC
To set Hot Outward Dialing to External Number,

• Scroll to select ‘Set Hotline: Ext Num’, press enter key
• Enter TAC
• Enter External Number #*
OR
• Dial 153-TAC-External Number#*
You cannot set Hotline and Hot Outward Dialing on the same extension at the same time.
To set Hotline Timer,

• Press DSS key assigned to Hotline
• Scroll to select ‘Set Hotline Timer’, press enter key
• Enter Hotline Timer:000-255 seconds
OR
• Dial 154-seconds (000-255)
The default value of Hot Line Timer is 3 seconds.
To Cancel Hotline / Hot Outward Dialing,

• Press DSS key assigned to Hotline
• Scroll to select ‘Cancel Hotline’, press enter key.
OR
• Dial 150
The cancellation code must be dialed from the dial tone. You have to be very quick in dialing the
cancellation code, if the delay in the Hotline Timer is set to 1 or 2 seconds.
For SLT Users

To set Hotline on an Extension / Department Group,
• Dial 151-Extension Number / Department Group Number
To set Hot Outward Dialing for Trunk,

• Dial 152-TAC

To set Hot Outward Dialing for External Number,
• Dial 153-TAC-External Number#*
To set Hotline Timer,

• Dial 154-seconds (000-255)
To cancel Hotline / Hot Outward Dialing,

• Dial 150
When you set the Hotline Timer to ‘00’ seconds (for immediate Hotline), you will not be able to dial any
digits, not even the feature code to Cancel Hotline.
If you have set Immediate Hot Outward Dialing for a Trunk or External Number, you will not be allowed to dial any
feature code, not even the feature code to cancel Hot Outward Dialing. However, if you need to cancel, you must
follow the steps described below.
• Go OFF-Hook.
You get the CO network Dial Tone.
• Dial by Digit.
You will hear Pause/Silence.
• Wait for the duration of the Trunk Inter Digit Timer

You will hear Pause/Silence.
• Press Flash.
You will hear the Feature Tone.
• Dial the code to change the Hot Outward Dialing Timer (154) and change the duration of the timer
–or–
Dial the access code to cancel the Hot Outward Dialing (150).
You get Confirmation Tone.
• Go ON-hook.
You get the return ring of the trunk.
• Go OFF-Hook again.
You get connected to the held trunk.
• Go ON-Hook.

Incoming CLI Modification
What’s this?
For System users in countries, where the Calling Line Identification (CLI) received must be suitably modified before
it can be used to dial out the number, SARVAM UCS offers the feature ‘Incoming CLI Modification’.
The Incoming CLI received with the Country or Area Code, or both. However, the dialing pattern of the public
network may require the received CLI to be prefixed with additional digits, to dial out the same number. Or the
dialing pattern of the public network may require the CLI to be stripped off the prefixed digits to dial out the same
number.
With the feature ‘Incoming CLI Modification’ programmed, the SARVAM UCS detects whether the incoming CLI is a
local number, a national, or an international number. It modifies the incoming CLI accordingly, by adding or stripping
off the prefixed digits so that the number can be dialed out as per the dialing pattern supported by the public
network.
The modified CLI is presented to the extension phones and is stored in the “Call Logs”, and SMDR (see “Station
Message Detail Recording (SMDR)”). Extension users can call a number in the Call Logs without having to modify
the CLI manually.
How it works
• Incoming CLI Modification parameters must be programmed in the system considering the dialing pattern
supported by the local public network.
• Accordingly, SARVAM UCS matches the CLI received with the programmed parameters.
• It detects whether it is an international, national or local number.
• It modifies the CLI according as per the Modification parameters programmed.
• It presents the modified CLI to the extension; stores the modified CLI in the SMDR and in the Call Logs of
the extension, provided it is a digital key phone.
• When the received CLI is dialed out by the extension user from Call Log, SARVAM UCS dials out the
same number.
How to configure
For this feature to work, the parameter ‘Incoming CLI Modification’ must be programmed in the ‘System
Parameters’ of SARVAM UCS. This can be done from Jeeves or by dialing SE commands from a Telephone.
Programming Incoming CLI Modification using Jeeves

• Click Incoming CLI Modification to expand.

Now, program these parameters:
• Enable Incoming CLI Modification: Enable this flag if you want to use the Incoming CLI Modification
feature. By default, this flag is disabled.
If you receive CLI in dialable format, there is no need to use this feature. In such case, keep the flag
disabled. You do not need to program any of the CLI modification parameters.
• Country Code: Enter the Country Code of the country where SARVAM UCS is installed. The Country
Code helps SARVAM UCS detect whether the Incoming CLI received is a national or an international
number. Do not enter any prefix for the Country Code. For example, if your SARVAM UCS is installed in
USA, enter only ‘1’ as the Country Code. Do not enter ‘+’ or “00’ as prefix to the country code ‘1’. By default
the Country Code is ‘91’ (India).
• Area Code: Enter the Area Code of the place where the SARVAM UCS is installed. The Area Code helps
SARVAM UCS detect whether the Incoming CLI received is a local number. Do not enter any prefix for the
Area Code. For example, if you want to enter Area Code for Mumbai, enter only ‘22’. Do not enter the
prefix ‘0’ to the area code. By default, Area Code is ‘265’ (Vadodara city).
• International Prefix: Enter the digits that are required as Prefix for dialing International Numbers. The
prefix may be up to 5 digits, with numbers from 00000 to 99999. By default, ‘00’ is set as the prefix for
dialing International numbers.
• National Prefix: Enter the digits that are required as Prefix for dialing long distance, National (within the
country) numbers. The prefix may be up to 5 digits, with numbers from 00000 to 99999. By default, ‘0’ is
set as prefix for dialing national numbers.
• Area Code required to make local calls?: Depending on the dialing pattern of your local public
telephone network, you may choose from the following options:
• No (Area Code not required): select this option if your public telephone network does not require the
dialing of Area Code for local numbers.
• Yes (Area Code is required): select this option if your public telephone network requires you to dial
the Area Code for local numbers.

• Yes (Area Code with Prefix required): select this option if you public telephone network requires you
to dial Area Code with a particular Prefix for local numbers. If you select this option, you must also
program the Prefix digits for the Area Code.
By default, the option, ‘No (Area Code not required) is selected.
• Prefix Area Code: If you have enabled the ‘Area Code required to make local calls’ flag in the previous
parameter, enter the prefix digits for the area code for local calls in this field.
Programming CLI Based Routing Table using a Telephone

To enable/disable Incoming CLI Modification, dial:

• 5367-Flag
Where,
Flag is
0 for Disable
1 for Enable
Default: Disabled
To program the Country Code, dial:

• 5368-Country Code
Where,
Country Code is a number string up to 5 digits. The number string may consist of numbers from 00000
and 99999.
Default: 91 (India)
Program the Country Code without any prefix. For example, if you want to program USA as Country
Code, enter ‘1’ only (without the prefix ‘00’ or ‘+’)
To program the Area Code, dial:

• 5369-Area Code
Where,
Area Code is a number string up to 5 digits. The number string may consist of numbers from 00000 and
99999.
Default: 265 (Vadodara)
Program the Area Code without any prefix. For example, if you want to program Mumbai as Area Code,
enter ‘22’ only (without the prefix ‘0’)
To program International Call Prefix, dial:

• 5370-International Call Prefix
Where,
International Call Prefix is a number string up to 5 digits. The number string may consist of numbers
from 00000 and 99999.
Default: 00
To program National Call Prefix, dial:

• 5371-National Call Prefix
Where,

National Call Prefix is a number string up to 5 digits. The number string may consist of numbers from
00000 and 99999.
Default: 0
To enable/disable ‘Area Code required to make local calls?’, dial:

• 5372-Code
Where,
Code is
1 for No (Area Code is not required)
2 for Yes (Area Code is required)
3 for Yes (Area Code with Prefix required)
Default: 1
To program Prefix Digits to Area Code for Local Calls, dial:

• 5373-Prefix Digits
Where,
Prefix Digits to Area Code is a number string up to 5 digits. The number string may consist of numbers
from 00000 and 99999.
Default: Blank
• Exit SE mode.

Intercom
What is this?
The Intercom feature of SARVAM UCS enables extension users to connect quickly with any desired extension,
without waiting for the called extension to answer.
• SARVAM UCS will serve an Intercom call made by an extension only if:
• the called extension is a DKP or a SIP Extension (Matrix Extended IP Phone or Open SIP Phone).
• the called extension is in idle state.
• the called extension is able to identify the incoming call as an intercom call (applicable in the case of
Open SIP Phones).
• the calling extension has Intercom in its Class of Service.
• the Priority of the calling extension is higher than that of the called extension.
• On SIP extensions, SARVAM UCS supports Intercom using Call-INFO / Alert-INFO Message. For a list
of IP phones on which this feature has been tested, see “SARVAM UCS Features tested on IP Phones
of different Brands” in the Appendix.
How it works
• A’s extension number is 3001 with Priority Level ‘7’ and ‘Intercom’ feature enabled in the Class of Service.
• B’s extension number is 3003 with Priority Level ‘5’.
• A wants to quickly connect to B. A dials the Intercom feature code *5 followed by B’s number, 3003.
• SARVAM UCS places the Intercom call on B.
• B’s extension is idle at the time of the call, and the speaker of B’s phone goes OFF-Hook, creating a
speech path between A and B.
• A can now talk with B.
• Do Not Disturb (DND): If the called extension has set DND, SARVAM UCS will not place the intercom call
on the called extension.
• Privacy from DND Override: If DND as well as Privacy from DND Override is enabled in the Class of
Service of the called extension, SARVAM UCS will reject the Intercom call.
• Call Forward-Unconditional: If the called extension has set Call Forward-Unconditional, SARVAM UCS
will forward the intercom call to the forwarded destination number. The call placed on the forwarded
destination will not be an Intercom call.
• Call Forward-No-Reply: If the called extension has set Call Forward-No-Reply, SARVAM UCS will not
forward the intercom call to the forwarded destination number on the expiry of the No-Reply Timer.

• Call Forward-Busy: If the called extension has set Call Forward-Busy, SARVAM UCS will place the call
on the forwarded destination number. However, this call (placed on the forwarded destination) will not be
an Intercom call.
• User Absent/Present: SARVAM UCS will place the Intercom call on the called extension only if the status
of the called extension is ‘Present’.
• Auto Call Back: When the Intercom call is generated and the called extension is busy, the calling
extension can set Auto Call Back on the called extension. When the called extension is free, SARVAM
UCS will serve the Auto Call Back request set by the calling extension. The ACB call placed on the called
extension will be a normal call.
.
• DISA: An Intercom call can be generated also from DISA mode.
• Priority: The calling extension must have a higher Priority level than the called extension.
When the Intercom call is generated on SIP Extension having multiple call appearance and already a call
is present on the SIP Extension then the SARVAM UCS will place the Intercom call as normal call on the
SIP Extension.
How to configure
To provide this feature to extension users, you must enable this feature in their Class of Service. For instructions,
see “Class of Service (COS)” and “Station Basic Feature Template”.
How to use
To use intercom to call an extension:
• Press the DSS Key assigned to Intercom.

OR
Dial *5
• The called extension’s speaker goes OFF-Hook.
• You may talk.
For SLT Users
To use intercom to call an extension:
• Dial *5 followed by the desired extension number.

• The called extension’s speaker goes OFF-Hook.
• You may talk.

Internal Call Restriction
What’s this?
Using this feature, the operator will be able to allow/restrict internal calls. The operator has flexibility to allow calls
during day time and restrict calls during night time.
In SARVAM UCS this feature is implemented in the following ways:

• Assigning 'Guest Group' to the station
• Enable/disable Internal Call Barring (Call Block)
Assigning 'Guest Group' to the station

• Privacy of the user is very important in Enterprises and hospitals. In general, most of the users need to
communicate only with reception, operator, pantry and such other service stations. He is not required to
dial other user numbers. For example, in hospitals the patients need to call the nurse and not the doctor or
other patients.
• Sometimes, a group of users occupy multiple rooms in the Enterprise. In such cases members of the
group would like to communicate only among themselves and service stations.
Essentially we need following logic:
• Certain service stations should be able to dial any other service station or any guest (for example Nurses,
Operator).
• Certain stations should be able to dial only service stations (for example: patients, single user in the office).
• A group of station users need to dial amongst each other and the service stations (for example: doctors).
Above requirements are implemented as follows:
• Solo Group (for example patients, single user of the station). This is classified in group 00.
• Universal Group (Nurses and other services staff for example). This is classified in group 99.
• Friend's Group (for example, Group of visitors). Such a group of station users can be assigned to any
group number from 01 to 98.
Following diagram depicts this logic:
Universal Group Friends Group Friend Group Friend Group

99 00 A B
Universal Group Friends Group Friend Group Friend Group

99 00 A B

How to use
From SA mode:
Dial, 1072-904-Flexible Number-Guest Group
OR
Press DSS key for 'Guest Group'
Where,
Flexible Number is the code given to access a station.
Guest Group is from 00 to 99.
By default, the guest group assigned is 99.
• The user can call any body in his 'guest group' number '99'.
• The station users in one 'guest group', '99' can call each other within the group as well as the station users
in other groups.
• The default Guest group of all the stations = 99.
Enable/Disable Internal Call Barring (Call Block)

SARVAM UCS supports a feature, to allow or bar the Internal Calls for the users of SARVAM UCS. This is called
Call Blocking.
How to use
From SA mode:
Dial, 1072-045-Code
OR
Press DSS key for 'Enable/Disable Internal Call Block'.
Where,
Code Meaning
0 Disable Call Block, (that is, Internal Calls allowed)
1 Enable Call Block (that is, Internal Calls barred)
• When this feature is enabled, the LED on the DSS key assigned to 'Call Block' will glow Red and when
disabled, the LED will be turned Off.
• When the operator issues "Call Block”- Enable command, all the station users will be assigned guest
group = 00.
• When the operator issues "Call Block”- Disable command, all the station users will be assigned guest
group = 99.
• Even after issuing SA command for 'Call Block', the operator can use the SA command 1072-904 to
change the guest group of any station user.
• Please refer the Hospitality Manual for more details.

Interrupt Request (IR)
What’s this?
Interrupt Request allows you to break into an on-going conversation after intimating the extension user about the
interruption.
In case of an important or urgent trunk call the operator can put the call on hold, interrupt the busy extension user to
inform about the urgent call and then transfer the urgent call.
How it works
• A, B and C are extension users.
• A and B are talking to each other.
• C calls A.
• C dials Interrupt Request feature code.
• C gets Ring Back tone (RBT) and A gets beeps indicating a new call.
• To answer C’s call, A must dial Flash before the expiry of the Interrupt Request Timer. A will be in speech
with C. B will be put on hold and will get music on hold.
• If A does not dial Flash before expiry of the Interrupt Request Timer, C’s call will be disconnected.
• Call States:
• Interrupt Request works only if the dialed extension is busy. The dialed extension may be busy with
another extension or trunk (external number).
• Interrupt Request works only if the user about to be interrupted in is in a two-way normal speech with
another user or external party.
• It will not work if the busy signal is due to the user being Off-hook, or in the middle of dialing, or
accessing a feature of the SARVAM UCS.
• “Call Toggle”: Once A and C comes in speech with each other, A can toggle between B and C using Call
Toggle feature.
• Privacy against Interrupt Request: If the feature 'Privacy against Interrupt Request is enabled for an
extension, it cannot be interrupted. See “Privacy”.
• “Priority”: No Interaction with Interrupt Request. If 'A' has lower priority than 'B' but has Interrupt Request
enabled; A can interrupt B.

How to configure
To be able to use Interrupt Request, extension users must have this feature enabled in their “Class of Service
(COS)” in their “Station Basic Feature Template”.
For instructions on configuring the different extension port types:

• “Configuring SLT Extensions”
• “Configuring DKP Extensions”
• “Configuring ISDN Terminals”
• “Configuring SIP Extensions”
If required you may also change the default value of the Interrupt Request Timer. For instructions see “System
Timers and Counts”.
How to use

When dialed extension is busy,
• Press DSS Key assigned to Interrupt Request

OR
• Dial 3 on Busy Tone
For SLT Users


Last Caller Recall
What’s this?
SARVAM UCS offers a facility—Last Caller Recall— to trace the extension that last made the call to your
extension.
How it works
• A’s extension number is 2001.
• A wants to know who made the last call to extension 2001.
• A dials Last Caller Recall feature access code (Default:1092).
• The extension that last called 2001 rings.
• When the called extension answers, speech is established between A and the called extension user.
On SIP extensions, SARVAM UCS supports Last Caller Recall using text (lcr) as access code.For a list of
IP phones on which this feature has been tested, see “SARVAM UCS Features tested on IP Phones of
How to use
• Press DSS Key assigned to Last Caller Recall.

OR
• Go OFF-Hook
• Dial 1092
The system dials out the extension number that last called your extension.
For SLT Users

• Dial 1092
The system dials out the extension number that last called your extension.

Last Number Redial
What’s this?
The system redials the last number string (external/internal) dialed from an extension. By default the system stores
only the external numbers in the Last Number Redial List. If you want the system to store internal calls in this list,
make sure you enable the Store Internal Calls in Redial Call Log check box in the “System Parameters”.
How it works
• Extension A dials the feature access code for ‘Redial’.
• If Extension A is an SLT, the system dials the last external number dialed from Extension A using the same
trunk access code used for dialing that number.
• If Extension A is a DKP or an Extended IP Phone, last 16 external numbers dialed by Extension A are
displayed on the phone’s LCD.
• Extension A may select the number to be dialed out. The system will dial out this number using the same
trunk access code used for dialing this number.
• If Extension A has ‘Dynamic Lock’ set and uses Redial feature, the system will check for Toll Control as
per the Lock Level set for Extension A before dialing out the number.
How to configure
No particular configuration is required for this feature to work. Redial is included in the Basic Features allowed to all
extensions by default in their “Class of Service (COS)”. So, all extensions can use the Redial feature.
How to use
• Press Redial Key.
Or
• Dial 7
A List of last 16 external numbers dialed will appear on your phone’s display.
• Scroll to select the desired number.
• The system dials out the external number.
For SLT Users

• Lift Handset
• Dial 7
• The system dials out the external number last dialed from your extension.

Least Cost Routing-Carrier Pre-Selection
What’s this?
This type of Least Cost Routing is used in countries where the same service provider offers local call and long
distance calling services. These service providers allow subscribers to select the service provider or Carrier for long
distance calling.
For example,
Service
Provider B
Subscriber Service Service

System Provider A Provider C
Service
Provider D
The subscriber of Service Provider A must grab trunk lines of Service Provider A to call other subscribers in the
local area.
However, when the subscriber of Service Provider A wants to make a long distance call, the subscriber must dial a
prefix to select the a carrier (trunk) of the desired long distance, Service Provider B, C and D. Thus, the subscriber
accesses a secondary service provider by dialing a short code or prefix for long distance calling.
This feature works on the basis of “Automatic Number Translation”. Using Automatic Number Translation,
SARVAM UCS adds the code of the appropriate secondary Service Provider to the number string dialed by the
extension user to route the call to the desired secondary Service Provider.
How to configure
To use this feature, you must do the following:
• Select an Automatic Number Translation Table Number from 1 to 8.
• Configure the Automatic Number Translation Table. In the Automatic Number Translation Table, in the
Dialed Number String column, enter the long distance numbers that extension users will dial. In the Add
Prefix column, enter the digits which are to be added as prefix to the Dialed Number string by the system
before dialing it out and in the Strip Digits column enter the number of digit(s) to be stripped off by the
system from the Dialed Number string before dialing it out. For example, the code ‘961’ for Service
Provider B must be prefixed to the number ‘2630555’ dialed by extension users, you must enter ‘2630555’
in the Dialed Number String column, ‘961’ in the Add Prefix column and ‘0’ in the Strip Digits column.
• Enable“Automatic Number Translation” feature on the trunk (CO/Mobile/SIP/T1E1/BRI) of your Primary

Service Provider in the “Outgoing Trunk Bundle” and apply the ANT Table number you configured to this
OG Trunk Bundle.
• Create “OG Trunk Bundle Group” that includes the OG Trunk Bundle you created with Automatic Number
Translation as member.

• Assign the OG Trunk Bundle Group (containing the OG Trunk Bundle with ANT) to the extensions for the
Time Zones, that is, Working Hours, Non-Working Hours, Break Hours, in the “Station Basic Feature
Template” of the extensions.

License Management
What's this?
The application SARVAM UCS and the features that it supports, require the purchase of licenses. When you buy
the ETERNITY GENX Platform, you get a unique default license key for the platform.
SARVAM UCS Application License

To use ETERNITY GENX as the Unified Communication Server, you need to purchase the SARVAM UCS SME
Application software license.
To use ETERNITY LENX/MENX as the Unified Communication Server, you need to purchase the SARVAM UCS
ENT Application software license.
If you do not have the license for the SARVAM UCS SME/ENT Application and you do not start the Demo
Period, the system will disconnect all the connected calls263 (internal or external, incoming or outgoing)
from any port after 60 seconds264. For details, refer “Demo Provision”.
Described below are the features of SARVAM UCS for which you require licenses. To know the name of the license
you need to purchase, see “Supported Licenses”.
IP Subscribers (For SIP Extensions)

SARVAM UCS supports upto 999 SIP Extensions. With a license, you can register SIP-enabled devices with the
SARVAM UCS. Without a license you cannot register any SIP Extension.
MATRIX VARTA User Licenses

SARVAM UCS supports three types of user licenses for VARTA Users — VARTA Essential Users, VARTA
Professional Users and VARTA Collaboration Users.
Following table lists the features which will be supported in MATRIX VARTA WIN200 / VARTA ADR100 / VARTA
AMP100 when you activate the respective license.
VARTA Essential VARTA Professional VARTA Collaboration

Users Users Users
Features
WIN ADR AMP WIN ADR AMP WIN ADR AMP
200 100 100 200 100 100 200 100 100
Making Calls         
Receiving Calls         
Hold         
Transfer         
263. Connected calls means where speech is connected between calling party and called party even if the called party port is not
matured.
264. Make sure the NX DBM VOCODER64 module is installed for SIP calls.

Blind Transfer         
One Touch Transfer         
3-Party Audio
        
Conference
Video Call         
Intercom         
Voicemail         
Call Forward         
Do Not Disturb         
Presence         
IM and SMS         
Favorites         
Global Directory
        
Access
All Menu Features         
All Call Features         
Hotkeys         
Multiparty Audio
     
Conference
Handover      
Drag and Drop
Transfer
     
Drag and Drop
Conference
     
Contact Grouping      
BLF Subscription      
DSS Soft Keys      
DSS Soft Keys for
Mobile users      
Call Transfer to other

user's Voicemail
(Blind Transfer to
     
VMS)
Click to Call      
Outlook Integration   
Presence Contact
Card Integration
  
Calendar Integration   

To know more about Matrix VARTA WIN200 refer to the MATRIX VARTA WIN200 User Guide and for VARTA UC
Clients for Mobile refer to the respective User Guide — Matrix ADR100 User Guide and Matrix AMP100 User
Guide.
Expansion Slots
Expansion Slots License is required to expand the number of functional universal slots in the system. This license
is applicable only for Firmware version V1R3 and later.
If you have upgraded the system firmware to V1R3 and later in the old ETERNITY LENX/MENX/GENX system, the
Expansion Slots license will be applicable for the universal slots. No universal slots will be functional by default.
You must purchase the license to activate the universal slots as required.
If you have purchased the new ETERNITY LENX/MENX/GENX system with the firmware V1R3 and later, the
Expansion Slots license will be applicable for the universal slots.
• For ETERNITY LENX, the first eight universal slots after the power supply card in the first rack will be
functional by default.
• For ETERNITY MENX, the first eight universal slots after the power supply card will be functional by
default.
• For ETERNITY GENX, the first four universal slots after the power supply card will be functional by default.
If you require more functional universal slots, you must purchase the Expansion Slots License to activate
the same.
Matrix provides two Expansion Slots licenses:

• SARVAM EXP4 SME for ETERNITY GENX system
• SARVAM EXP4 ENT for ETERNITY MENX and LENX system.
Each SARVAM EXP4 SME/ENT license will provide the activation for next four universal slots in the
sequence.
VOCODER Channels
VOCODER Channels license is required for SIP calls. The maximum number of VOCODER channels that will be
supported would be as per the license you purchase. SIP calls uses the VOCODER Channels, if you select RTP
Mode for routing SIP calls. Refer “RTP Mode” for details of VOCODER Channels usage during SIP calls.
DBM VOCODER64 module supports a maximum of 64 VOCODER channels.
If you require more than 64 VOCODER channels, you must install another NX DBM VOCODER64 Module.
During a SIP call, VOCODER channel is required and if no free channel is available, the system will reject the call.

VMS Channels
VMS Channels license is required for allowing the VMS calls. The system allows simultaneous VMS calls as per
the license only.
The NX DBM VMS64 Module is an optional module. If required, you may purchase it separately. The system
supports a maximum of 64 channels out of which 4 channels are provided by default. If you require more channels,
you can purchase the channel licenses according to your requirement.
If VMS call request is received and no free VMS channels are available, the system will reject the call.
CTI
CTI stands for Computer Telephony Integration. CTI is a technology that integrates a telephone and a computer.
With the CTI license, the Matrix TAPI Service Provider (TSP) acts as a link to integrate the interactions between
your telephones and your computers. The computer in which you install this application functions as a Client and
SARVAM UCS functions as a Server. In an organization, it is possible that the data is stored in different computers.
In this scenario, the Matrix TAPI Service Provider can be installed in three (maximum) different computers, if
required.
Q-Sig
When SARVAM UCS is networked with another SARVAM UCS or with any other ISDN-System, Q-Sig or Q-
Signaling is supported to facilitate feature transparency between the systems in the network. Q-Signaling will be
activated when you buy the license for the same.
Hospitality Management System

This functional module contains a set of special telephone and guest/patient management features for hospitality
and accommodation establishments like hotels and hospitals, which SARVAM UCS supports when it is deployed in
a hotel or hospital. When you buy the license for this module, the following features will be activated:
• Room Shift
• Check-In, Check-Out
• Change Room Occupancy Status
• Floor Service
• Change Room Clean Status
• Front Desk User GUI (web interface)
Property Management System (PMS)

SARVAM UCS supports interface for PMS, the application software commonly used by hotels to manage their
administrative functions. The PMS Interface supports proprietary PMS protocols of Matrix (Matrix PMS Type 1 and
Type 2), Micros Opera and Softbrands Extended Starlight. To be able to select any of these PMS Protocols, you
must buy a license.
License for the Hospitality module is a pre-requisite for PMS.
Dealer Board
If you want to connect and use EON74265, the proprietary Digital Turret for ETERNITY LENX/MENX, you need to
buy a Dealer Board license.
265. Check for availability. Not supported by ETERNITY GENX.

Gateway
When SARVAM UCS is to be used with Gateway functionality, a license is required to activate this functionality.
PLCC
This functional module contains a set of special features supported by the SARVAM UCS when it is deployed in a
Power Line Carrier Communication Network of electric utilities. When you buy the license for this module, the
following features will be enabled:
• CCS Signaling when End Point and Transit.

• Express Signaling.
• Seizure Pulse and Release Pulse Signaling.
SMS Server
With the SMS Server license, you can:
• Send/ receive SMS to/from individuals or groups using the Mobile Port of SARVAM UCS.
• Forward SMS received on Mobile Port as Emails to users through the Email Client.
• Forward Email of the users as SMS to the Mobile users through the Mobile Port.
The SMS Server application works as an intermediary between the GSM Short Message Service and the SARVAM
UCS. The Server supports multipart, 7 bit text messages as well as UNICODE messages.
The Server functions as an SMTP Client to send emails and as a POP3 Client to receive emails. SMS Server
supports three types of Emails—Plain Text, HTML and MIME— from its mail clients.
SMS Gateway
With the SMS Gateway license, you can send/receive messages to/from individuals, selective groups or masses
using the Mobile Port of SARVAM UCS.
SARVAM UCS allows you to register multiple SMPP Clients (Software Applications used for sending/receiving
messages) with SARVAM UCS. SARVAM UCS functions as an SMPP Server. These Clients can send/receive
messages using the Mobile port/s of SARVAM UCS.
Supported Licenses
Refer to the table below to know the name of the respective licenses you need to activate for each feature.
License Name Description
Unified Communication Server License for Small-Medium

SARVAM UCS SME
Enterprise. This is a UC Server License for ETERNITY GENX.
Unified Communication Server License for an Enterprise. This is a

SARVAM UCS ENT common UC Server License for ETERNITY LENX and ETERNITY
MENX.
SARVAM EXP4 SME License for SARVAM UCS SME to activate four Expansion Slots.
SARVAM EXP4 ENT License for SARVAM UCS ENT to activate four Expansion Slots.
SARVAM IPSUB5 License for 5 IP Subscribers to create 5 SIP Users.

SARVAM IPSUB500 License for 500 IP Subscribers to create 500 SIP Users..
License for 5 VARTA UC Users with Essential features. This

SARVAM VARTA USER5E
license is required to register Android/iOS/Windows UC Clients.

SARVAM VARTA USER10E



License for 5 VARTA UC Users with Professional features. This

SARVAM VARTA USER5P

SARVAM VARTA USER10P



License for 5 VARTA UC Users with Collaboration features. This

SARVAM VARTA USER5C

SARVAM VARTA USER10C



SARVAM VOCODER CHNL4 License for activating 4 additional VOCODER Channels.
SARVAM VOCODER CHNL16 License for activating 16 additional VOCODER Channels.
SARVAM VMS CHNL4 License for activating 4 additional VMS Channels.
SARVAM VMS CHNL16 License for activating 16 additional VMS Channels.
License for SARVAM UCS SME to enable CTI functionality to

SARVAM CTI SME
connect to third party CTI application.
License for QSIG interface for SARVAM UCS SME to contact with
SARVAM QSIG SME other Matrix or third party System for seamless calling and inter-
working of certain System features.
License for Hospitality functions suite for SARVAM UCS SME to

SARVAM HOSPITALITY SME
support hospitality functions and features used in Hotel-Motel.

License for Property Management System interface for SARVAM

SARVAM PMS SME
UCS SME to contact with third party PMS used in Hotel-Motel.
License for SARVAM UCS SME to support Gateway functions in

SARVAM GATEWAY SME
addition to UCS functions.
License for SARVAM UCS SME to enable SMS Server

functionality to connect to any Email Client and send/receive Email
SARVAM SMS SERVER SME
to SMS and vice versa over GSM SIM installed on GSM Interface
Card.
License for SARVAM UCS SME to enable SMS Gateway

functionality to connect to third party SMS Gateway Clients and
SARVAM SMS GATEWAY SME
send/receive SMS over GSM SIM installed on GSM Interface
Card.
License for SARVAM UCS ENT to enable CTI functionality to

SARVAM CTI ENT
connect to third party CTI application.
License for QSIG interface for SARVAM UCS ENT to contact with
SARVAM QSIG ENT other Matrix or third party System for seamless calling and inter-
working of certain System features.
License for Hospitality functions suite for SARVAM UCS ENT to

SARVAM HOSPITALITY ENT
support hospitality functions and features used in Hotel-Motel.
License for Property Management System interface for SARVAM

SARVAM PMS ENT
UCS ENT to contact with third party PMS used in Hotel-Motel.
License for SARVAM UCS ENT to support Gateway functions in

SARVAM GATEWAY ENT
addition to UCS functions.
License for SARVAM UCS ENT to enable SMS Server

functionality to connect to any Email Client and send/receive Email
SARVAM SMS SERVER ENT
to SMS and vice versa over GSM SIM installed on GSM Interface
Card.
License for SARVAM UCS ENT to enable SMS Gateway

functionality to connect to third party SMS Gateway Clients and
SARVAM SMS GATEWAY ENT
send/receive SMS over GSM SIM installed on GSM Interface
Card.
Demo Provision
Demo provision for licensed features is useful, when the customer’s system having licensed features cannot be
repaired on-site and a standby system needs to be installed Or when end user demands to use certain licensed
features on trial basis before actually purchasing the license.
Demo Provision enables you to use the SARVAM UCS Application as well as all the licensed features it supports,
free of cost for a period of 60 days. In this case, the application starts even if you have not purchased the SARVAM
UCS SME/ENT license.
All the Universal Slots will be functional during the Demo period irrespective of the number of activated SARVAM
EXP4 SME/ENT licenses.
The Demo Provision lets you access and use all the licensed features and functionalities266 supported by the
application.
266. For VoIP and VMS, the number of channels that will be supported in demo period will be as per the license you purchase for the
same.

To avail this facility,
• Open Jeeves.
Under Configuration, click License Management link. The License Management page opens.
• Click the Demo Period Start button. The demo period for using licensed feature in your system starts and
the Start button will change to Pause. The demo period is of 60 days.
You may Pause the Demo Period, if required. When you pause the demo period, all licensed features will
work as per the license key installed in the system.
If you want to use licensed feature after the expiry of the demo period, you must purchase the license key
and activate it in your system.
When you default the SARVAM UCS, the demo period will not be reset.
How to activate your License

For the functional modules and features described above, you would need to activate a valid License Key.
Instructions for Matrix Channel Partners

Your license voucher may be a paper or a PDF (protected) file.
You may activate your License Online. For this, keep the following items ready:

• The License Voucher containing the 16-digit PIN.
• A valid, unique User ID and Password from the Matrix License Support Centre.
• Access to Internet.
• Current License Key of the system.
To activate the License Key online,
• open Jeeves.
• Under Configuration, click License Management link.
The License Management page opens.
• Note down or copy the current License Key on this page.
You can view the features and functions that are currently available to you under Service Profile.
• Keep your Current License Key and the License Voucher (paper or PDF) ready.
• Open a new window on your browser.

Enter http://www.matrixcomsec.com/MatrixLicense/ in the address bar.
• Click Login to Access.
The Login to Access page will open.
• Enter your User Name and Password provided by Matrix and click the Login button.

On successful login, the License Activation page will open.
• As Product Family, select the option SAPEX.
• In the Current License Key field, type the current product license key you noted or paste the key you
copied from the License Management page of Jeeves.

• Click the View button.
• The page displays the current License Profile267 on ETERNITY GENX. Click the Next button to continue.
267. When ETERNITY GENX is used as the Unified Communication Server, all the licenses except UMG are applicable. UMG License
is applicable when you run the ETERNITY GENX as the Universal Media Gateway.

The License Activation page opens.
In the License PIN field on this page, enter the 16-digit License PIN from the Voucher.

• Click Details. The details appear in the fields Product Family, Product Name, Product Variant.
• Click the Next button. Your Current License Profile and your New License Profile will appear on this
page.
• Click the Activate button and wait for a few seconds, as the activation is initiated.

On successful activation, the confirmation message will appear on your screen along with the activation
date and time.
A confirmation mail will also be sent to your e-mail ID (registered with Matrix).
You may Save, Print, or Email this information for your records, by clicking the relevant button.
• Note down or copy the New License Key generated on this page.
• Go back to the Jeeves window (or log in as System Engineer again, if your session has ended).

Click the License Management page again.
• In Enter License Key, paste or enter the new License Key generated.

• Click the Submit button.
The Service Profile on this page will be updated according to the license.
• To log off, click the Logout button.
If you are unable to use Online Activation of the License Key or have no internet access, contact the Matrix
License Support Centre for assistance in generating the new License key.
Instructions for Customers

To activate your License, you would need the License Voucher containing the 16-digit License PIN. Contact your
Dealer/Distributor in this regard. Your License Voucher may be a paper or a protected PDF file.
• Open Jeeves.

• Under Configuration, click License Management. The License Management page opens.
• Note down the current License Key on this page.
Under Service Profile, you may view the functional modules and features that are currently available to
you.
• Send your Current License Key and the License PIN (on the Voucher) to the Matrix License Support
Centre.
• You will receive a new License Key.
• Open Jeeves again.

• Click the License Management link.
• In Enter License Key, enter the New License Key you obtained from Matrix.

• Click the Submit button.
The Service Profile on this page will be updated accordingly.
• To log off, click Logout.
The current License Key and Service Profile will remain unchanged when the system is set to default or
the firmware is upgraded.

Live Call Screening
What’s this?
Live Call Screening enables extension users to screen callers before attending their calls on their extensions.
How it works
To be able to use this feature,
• the VMS module must be present in the SARVAM UCS.
• the extension users who are to be provided this feature must have a Personal Mailbox assigned, and have
this feature allowed to them in their Class of Service.
• the extension users who want to use this feature must set Call Forward to Voice Mail System on their
extension.
• the extension on which this feature is used must be a DKP.
With the above pre-requisites fulfilled, this is how Live Call Screening will work:
• Extension A has activated Live Call Screening.
• B calls A, and is transferred to A’s mailbox (as Call Forward to Voice Mail System is set).
• The VMS Auto Attendant offers B the option to leave a message in the mailbox of extension A.
• As B starts to record a message in A’s mailbox, the speaker of the DKP of A gets turned ON for the
duration of the Live Call Screening Timer (configurable; default: 10 seconds).
• A hears the message being recorded in the mailbox.
• If A wants to answer the call, A can go Off-Hook. A gets connected to the caller and the system stops
recording the message in the mailbox.
• If A does not answer the call, the speaker is turned Off automatically on the expiry of the Live Call
Screening Timer.
• Also, after listening to some part of the message, if A finds that the call is not important, A can ignore the
caller by dialing any digit. When A dials any digit, the speaker of the DKP is turned OFF, but the message
recording in the mailbox continues.
LCS works only if your extension is idle.

How it configure
To be able to use Live Call Screening, extension users must have:
• A mailbox assigned to them.
• Live Call Screening enabled in the “Class of Service (COS)” for the time zone in their “Station Basic
Feature Template”, as required.
If required, you may also change the duration of the Live Call Screening Timer. See “System Timers and Counts”
for instructions.
How to use
For EON Users only

To activate Live Call Screening,
• Press DSS key assigned to Live Call Screening.

OR
• Dial 1094-1
To answer a caller who is recording a message in your mailbox,

• Go Off-Hook by pressing the speaker key or lifting the handset.
• Speak to the caller.
To ignore the message of the caller,

• dial any digit.
To deactivate Live Call Screening,

• Press DSS key assigned to Live Call Screening, again.
OR
• Dial 1094-0

Live Call Supervision
What’s this?
Using Live Call Supervision, any extension can know the last external number dialed by another extension, even
when that extension is in speech with an external party.
This feature is useful for supervisors who want to know where their subordinates are calling.
This feature is supported on DKP and SIP extensions, and on SLT extension which have CLI phone.
How it works
• A is the supervisor of B.
• A wants to know where B is calling, A can use Live Call Supervision.
• When B dials an external number it is stored in the system’s memory.
• When A requests Live Call Supervision for B’s extension, the system retrieves the last external number
dialed by B and presents it on the display of A’s phone.
• If B has not dialed any external number, A will get error tone with the message ‘No Calls to Supervise’
displayed on the LCD.
• If the B has dialed internal as well as external numbers, the last external number dialed by B will be
displayed on LCD of A’s phone.
Live Call Supervision can be used also when the extension being supervised is in speech with an external
party.
How to configure
To be able to use Live Call Supervision, extension users must have this feature enabled in their “Class of Service
(COS)” in their “Station Basic Feature Template” for the required time zones.
How to use
• Press DSS key assigned to Live Call Supervision

OR
• Dial 1098
• Enter the Extension number to be supervised
For SLT Users
• Dial 1098-Extension number to be supervised.

Logical Partition
What's this?
Logical Partitioning is used to restrict the flow of call traffic between PSTN and Private Networks as well as
between PSTN and VoIP networks.
This feature may be used in countries where such restrictions are mandated by telecom regulations. For example,
in certain countries, calls from VoIP to Public Networks (PSTN, Public Land Mobile Network) are not allowed.
Thus, local telecom regulations may either disallow termination of lines from both networks on the same equipment
or may allow lines from both networks to be terminated on the same equipment, provided the equipment is
designed to restrict flow of call traffic from these networks. For example, the Telecom Regulatory Authority of India
allows termination of lines from the PSTN and VoIP Networks in the same equipment, only if these lines are
logically partitioned. Termination of lines from both these networks in the same equipment without a logical partition
constitutes an offence.
SARVAM UCS supports Logical Partitioning for this purpose.
How it works
Logical Partition is applied on the Trunk ports. Trunk ports are assigned to any of the following categories according
to their installation scenario:
• Category 1: Trunk ports interfaced with PSTN /PLMN (Public Land Mobile Network) are assigned this
category.
• Category 2: Leased lines terminated in the trunk ports are assigned this category.
• Category 3: Trunk ports used to interconnect two Systems are assigned this category. For example, QSIG
used on T1E1 port or the CO of SARVAM UCS is interfaced with FXS of other System for expanding
configuration.
• These are default Categories. You have the flexibility to define each category according to suit your
preference. For example, you may, if you so prefer, define Leased lines to Category 3 and Trunk ports
connecting two Systems to Category 1.
• As calls from VoIP to Public network trunks are always to be restricted, VoIP trunks cannot be assigned
to any of the three categories of the trunks as explained above and therefore do not require any
classification.
For each of the above categories, including VoIP trunks, the System Engineer can program the calling
permission, that is, whether to 'allow' or 'restrict' the calls across and within these categories and with the
VoIP trunks.
You can change the settings as per the regulations in your country.
Depending on the calling permission programmed between the trunk categories, the system will allow or
deny the calls on the trunks.

If within the office premises, you want to remove the logical partition for local SIP Extensions allowing the
Internal Extension (SLT, DKP, Local SIP Extension and Remote Extension) calls and External (Trunk)
calls, then you can configure the table defining the IP address range of those local SIP Extensions. These
local SIP Extensions will be considered similar to the System extensions such as a DKP or SLT.
The SIP Extensions other than those configured in the table will be considered as External/Remote SIP
Extensions. Though Remote SIP Extensions can make internal calls to the DKP/SLT/Local SIP
Extensions, but Logical Partition will be applied for the external SIP extension calls and trunk calls.
Matrix recommends you to enable the logical partition in order to avoid any toll bypass wherever
necessary. Matrix will not be responsible in case of any discrepancy related to this.
How to configure
To able to use this feature you must first assign the trunk port - CO, ISDN BRI, T1E1PRI, Mobile, VoIP to the
appropriate Category and then define the calling permission - allow or restrict calls - in Category 1 to 3 and VoIP
trunks (as described above).
Configuring Logical Partition using Jeeves

• Now, assign the categories to the different Trunk Type, by programming the parameter 'Category (Logical
Partition)' in the Trunk Port Parameters of the respective trunk types: CO, BRI, T1E1, E&M and Mobile
Port.
• For CO trunks, assign Category in the “CO Hardware Template”.

• For E&M ports, assign Category in the “E&M Feature Template”.
• For Mobile ports, assign Category in the “Mobile Port Parameters”.
• For T1E1PRI ports, assign Category in the “Configuring E1 Trunks” and “Configuring T1 Trunks”.
• For BRI ports, assign Category in the “Configuring BRI Trunks”.

• Open the desired Trunk Port Parameters page by clicking the link. For example, to assign a Category to
CO ports, open the CO Hardware Template page.
• Scroll with the horizontal bar to reach the column 'Category (Logical Partition).
• Select a Category number 1, 2, or 3 to assign Category to CO ports.
• Click 'Submit' at the bottom of the page to save your settings.
• Repeat the same steps to assign Categories to other Trunk ports.
If you have completed assignment of Category to the Trunk Type,

• Under Configuration, click Logical Partitioning to open the page.
• Now, define the call permission for each category, that is, 'allow' or 'restrict' calls. For example, allow/
restrict calls from Category 1 to Category 1, from Category 1 to Category 2, from Category 1 to Category 3
and so forth.
• In the table, configure the IP addresses and subnet masks of those local SIP Extensions for which you
want to allow the Internal Extension and External (Trunk) calls without checking the logical partition. These
local SIP Extensions will be considered similar to the System extensions such as a DKP or SLT. You can
configure upto 25 entries in this table.
• Log out of Jeeves or continue with the configuration.
Configuring Logical Partition using a Telephone

• First, program the parameter 'Category (Logical Partition)' in the Trunk Port Parameters of the respective
trunk types. For SE Commands, refer the topics:
• “CO Hardware Template”

• “E&M Feature Template”
• “Mobile Port Parameters”
• “Configuring E1 Trunks” and “Configuring T1 Trunks”
• BRI Parameters in “Configuring BRI Trunks”

To define call permission across and between categories:
• Dial 5317-Category-Category-Flag
Where,
Category is
1 for Category 1 trunks
4 for VoIP
Flag is
0 for restrict calls
1 for allow calls
Default: 0 (that is, restrict calls) for Category 1 to 3 and 1 (allow calls) for VoIP.
• Exit SE mode.
• If you select India as the Region, by default, calls are restricted for all categories, except within VoIP,
which includes both SIP trunks and SIP extensions.
• When call permission is restricted between two categories of trunks and/or between any category of
trunk to VoIP, following feature interactions will apply:
• Call Transfer: Trunk to Trunk Transfer between restricted categories of trunk will not be allowed. If
the user attempts trunk-to-trunk transfer between restricted trunks, Error Tone will be played.
• Raid: If a user using DISA attempts to Raid a conversation of an extension with a trunk to which call
permission is restricted, the Raid attempt will fail and the user will get an Error Tone.
• Conference: An extension user will not be able to include restricted trunks in a 3 party or multi-
party conference. An Error Tone will be played when s/he attempts it.
• Dial-In Conference: Participation in a Dial-In Conference from trunks with restricted call permission
is not allowed.
• External Call Forward: In the case of Auto Attendant, DISA or when transferring a trunk call to an
extension, if the extension has set call forward to an external number, the system will allow the call
only if the call permission between the source and destination trunk is allowed. Otherwise, an Error
Tone will be played to the user.
• Hotline: When a user has logged into DISA and the extension being used for the DISA login has
the Hotline - Trunk or Hot outward dialing (HOD) feature enabled, the system will allow the call
between the source trunk (from where the DISA login is made) and the destination trunk (which is
used as Hotline Trunk) only if calling is permitted between them. Otherwise an Error Tone will be
played to the DISA caller on the expiry of the Hotline Timer.

Loop Back Tests
What’s this?
SARVAM UCS supports following types of Loop Back Tests for T1 and E1 lines:
• Near End Loop Back: These are of two type viz.

• Line Loop Back test
• Payload Loop Back test
• Far End Loop Back Test: These are of two type viz.
• Line Loop Back test
• Payload Loop Back test
These tests are conducted when the T1E1 port is in NT mode and is connected to other System and wants
to test the line between itself and the far end. In this mode, SARVAM UCS acts as a network.
Near End Loop Back
Line Loop Back

This can be implemented by shorting of signaling wires 'RTIP' and 'RRING' to 'TTIP' and 'TRING' respectively at the
line side.
This is as good as shorting the Rx pair to Tx pair. In this case you do not need to sync the T1E1 port with the
network.
Payload Loop Back

This is effectively a physical connection of payload channels (Timeslot) with framing generated by T1E1 port itself
for the transmit signal. In this case you must sync the T1E1 port with the network.
When the other end connected with the T1E1 port of the SARVAM UCS wants to perform the Loop Back tests, the
T1E1 port will form the Loop Back depending upon the type of test the other end wants to perform (that is, Line
Loop Back or Payload Loop Back).
Loop Back Process
Method of forming loop back

Method of forming loop back at the T1E1 port side is different for the E1 and T1 carrier. These are explained in the
following.
When line type is configured as ISDN_E1_PRI or ISDN_E1_CAS on the T1E1 port,
• The protocol doesn't support the facility that the remote end can close/open the loop at the T1E1 port side
automatically.
• Hence when the remote end wants to perform the loop back test, the SE must be informed, to form the
desired type of loop back (that is, Line Loop Back or Payload Loop Back) on the T1E1 port.

• When the SE forms the loop back at T1E1 port side (by issuing appropriate SE Command), the remote
end can start the test.
• On completion of the testing, the remote end must inform the SE to release/open the loop back formed on
the T1E1 port side.
• On receipt of the request from the remote end, the SE will issue SE command to open the loops on the
T1E1 port.
When line type is configured as ISDN_T1_PRI or ISDN_T1_RBS on the T1E1 port,
• The protocol supports loop back Activation and deactivation message, whereby the remote end can send
the loop activation code to the T1E1 port and the T1E1 port decodes the message and forms the loop back
automatically.
• On completion of the testing, the remote end can send the loop deactivation code and the T1E1 port can
open the already formed loop back.
• So in this case the SE's intervention is not required to form and release the loop back.
• Incase the remote end doesn't support the facility to automatically form/release the loop back for the T1E1
port though the carrier is T1, the SE can use the commands (6141-1-T1E1-Type of Loop Back) on request
of the remote end to form and release the loop backs.
Loop back Activation

When the system receives the SE command to form Line/Payload loop back (in case of E1):
• It will inform the SARVAM UCS Card T1E1 about the received command SARVAM UCS Card T1E1 will
release all the calls supported by the T1E1 Port under test.
• The SARVAM UCS Card T1E1 will form the required type of loop back.
• System will put the T1E1 port in maintenance mode. It will release all active calls supported by T1E1 port
and restrict the usage of T1E1 port for IC/OG calls.
When the loop back activation code is received from far end on the T1E1 port, (incase of T1 carrier):
• SARVAM UCS Card T1E1 will inform the system about the received information and the T1E1 port
number.
• SARVAM UCS Card T1E1 will release all the calls supported by the T1E1 Port under test.
• System will put the T1E1 port in maintenance mode. It will release all active calls supported by T1E1 port
and restrict the usage of T1E1 port for IC/OG calls.
Release Loop Back

On receiving the loop back release code or receiving the command to release the loop back (either Payload or
Line), the system will take the T1E1 port out from the maintenance mode and now the T1E1 port will function
normally.

Far End Loop Back Test
Far End Loop Back test gives facility to SARVAM UCS to check the health of the line towards the other end
connected on its T1E1 port using loop back tests.
The types of the far end loop back test are:

• Line Loop Back
• Payload Loop Back
Line Loop Back

This can be implemented by shorting of signaling wires 'RTIP' and 'RRING' to 'TTIP' and 'TRING' respectively at the
line side.
This is as good as shorting the Rx pair to Tx pair. In this case you do not need to sync the T1E1 port with the
network.
Payload Loop Back

This is effectively a physical connection of payload channels (Timeslot) with framing generated by T1E1 port itself
for the transmit signal. In this case you must sync the T1E1 port with the network.
When the other end connected with the T1E1 port of the SARVAM UCS wants to perform the loop back tests, the
T1E1 port will form the loop back depending upon the type of test the other end wants to perform (that is, line loop
back or payload loop back).
The option 'Release All Loop Backs' is required when the far end requires the universal loop back release
code to release the Loop Back.
Loop Back Process
Method of forming loop back

Method of forming loop back at the T1E1 port side is different for the E1 and T1 carrier. These are explained below.
When line type is configured as ISDN_E1_PRI or ISDN_E1_CAS on the T1E1 port,
• The protocol doesn't support the facility that the SARVAM UCS can perform the loop back tests
automatically.
• The SE of the SARVAM UCS will inform the far end connected with the T1E1 port, to form the loop back as
required.
• Once the far end has formed the desired loop back the SE can issue command to start the Far end loop
back test.
Start Loop Back Test:
• On receiving the command to start Far End loop back test (either the Line Loop Back or Payload Loop
Back test) the system will put this T1E1 port in maintenance mode and informs the T1E1 Card about the
test to being performed on the T1E1 port.
• All the calls supported by the T1E1 Port under test will be released.

• The T1E1 Card will start the PRBS generator and counter.
• The T1E1 Card will send the PRBS count every 1 second to the system.
• The T1E1 Card will increment PRBS Counter for every error encountered during the test every one
second.
• The T1E1 Card will reset the PRBS counter to zero, after sending the PRBS Counter to the system, every
second.
• The system will store the received PRBS count (received every second) in Performance Report, which can
be captured from Serial port/Ethernet port.
End Loop Back Test/Release All Loop Backs
• When the SE wants to end the loop back test he will issue the command to end the Far End Loop Back
Test, for the T1E1 port under the test.
• SE will inform the other end's person that loop back test is finished and now the remote end can open the
loop formed.
• On receiving the command to end loop back test for the T1E1 port, the system will take the T1E1 port out
of maintenance state and inform the T1E1 port about the received command.
• Normal functioning of the T1E1 port will resume.
When the line type is configured as "ISDN_T1_PRI" or "ISDN_T1_RBS" on the T1E1 port,
• Loop back activation/deactivation is automatic.
• The activation methods are different for the D4 and ESF framing.
• D4 framing supports only Line loop back
• ESF framing supports both Line loop back and Payload loop back.
• When D4 is selected as Framing,
Start Line Loop Back Test
• The test will start for the T1E1 port when SE issues the command to start the far end line loop back
test. When SE command (6142-1-T1E1-1) is issued the system will inform the T1E1 Card about the
command and this T1E1 port will be in maintenance mode.
• The T1E1 Card will send the line loop back "Activation Code" and will start the PRBS generator and
counter.
• The T1E1 Card will send the PRBS count every 1 second to CPU Card.
second.

• The T1E1 Card will reset the PRBS counter to zero, after sending the PRBS Counter to the system,
every second.
• The system will store the received PRBS count (received every second) in Performance report, which
can be captured from Serial port/Ethernet port.
Stop Loop Back Test
• When SE command (6142-T1E1-2) or (6142-T1E1-5) to end the Far end loop back test is issued the
system will inform the T1E1 Card about the received command and will remove the T1E1 port from the
maintenance mode.
• The T1E1 Card will send the inband "Deactivation Code".
• When ESF is selected as Framing,
Start Loop Back Test
• Depending upon the SE command issued to start the type of Loop Back (Line or Payload) test the
system will put the T1E1 port under test in Maintenance mode and will inform the T1E1 Port about the
received command.
• The T1E1 Card will send the Loop back "Activation Message" for the line/payload loop back as
informed by the system.
• The T1E1 Card will start the PRBS generator and counter.
• The T1E1 Card will send the PRBS count every 1 second to CPU Card.
second.
• The T1E1 Card will reset the PRBS counter to zero, after sending the PRBS Counter to Master, every
second.
• The system will store the received PRBS count (received every second) in Performance report, which
can be captured from Serial port/Ethernet port.
Stop Loop Back Test/Release All Loop Backs
• When SE issues command (6142-T1E1-2) or (6142-T1E1-4) or (6142-T1E1-5) to end the Far end loop
back test the system will inform the T1E1 Card about the received command and will remove the T1E1
port from the maintenance mode.
• The T1E1 Card will send the "Deactivation Message" for the line/payload loop back test as required.
Performance Report
A Performance Report of the tests can be generated. It contains the log of all the errors. The Performance Reports
stores upto 50 entries on FIFO basis.

Configuring Loop Back Test using Jeeves
To configure the Near End Loop Back Test parameters,
• Click Loop Back Test: Near End.
• For each T1E1 port,
• Select the Apply? check box to enable loop back.
• From the Activity list, you can select to Activate or Release the loop back test.
To configure the Far End Loop Back Test parameters,

• Click Loop Back Test: Far End.
• Select the Communication Port/Ethernet Port/ USB to COM to be assigned as destination port for
Online and Report logs.
• If you select Ethernet Port as the destination, enter the IP Address of the Ethernet Port and the Port
(Listening Port for the packets) for the Online and Report logs respectively. Both IPv4 and IPv6
addresses are supported. Valid port range: 1025 to 65535;514.
• For each T1E1 port,
• Select the Apply? check box to enable loop back.
• From the Activity list, you can select to Start, End or Release the loop back test.
Configuring Loop Back Test using Telephone
Near End Loop Back

Use the following command to activate/release Near End Loop Back for T1E1:
6141-1-T1E1-Loop back
Where,
Loop back Meaning
1 Activate Line Loop back

Loop back Meaning
2 Activate Payload Loop back
3 Release Loop back
Far End Loop Back

Use the following command to start/stop far end loop back test for T1E1:
6142-1-T1E1-Code
Where,
Code Meaning
1 Start Line Loop Back Test
2 End Line Loop Back Test
Start Payload Loop Back

3
Test
4 End Payload Loop Back Test
5 Release All Loop Backs
Performance Reports
Online Performance Report Printing
Use following command to assign the port for online Performance Report Printing:
6143-Port
Where,
Port Meaning
0 None
1 COM Port
2 Ethernet Port
3 USB to COM Port
Default, None.
Offline Performance Report Printing

Use following command to assign the port for offline Performance Report Printing:
6144-Port
Where,
Port Meaning
0 None
1 COM Port
2 Ethernet Port

Port Meaning
3 USB to COM Port
Default: None.
Performance Reports From SA Mode

Online Performance Reports
Use the following command to start/abort online printing of T1E1 performance report:
1072-030-Flag
Where,
Flag Meaning
0 Disable
1 Enable
By default, Flag is 0.
Offline Performance Reports
User the following command to start/abort offline printing of T1E1 performance report:
1072-031-Flag
Where,
Flag Meaning
0 Disable
1 Enable
• PRBS counter = 0 indicates the 'healthy' condition.
• During loop back test the PRBS counter may be greater than zero at initial stage of the loop back
stage, but it will be zero afterwards consistently for the healthy condition.
• PRBS counter = greater than zero, indicates the 'faulty' condition for the loop back test.

Macros
What’s this?
Extension users often have to dial access codes for specific functions like dialing a feature code, making an internal
call, making an external call, etc.
SARVAM UCS supports Macros, using which, you can abbreviate long number strings for regularly used functions
in to macros and assign them to a DSS key on a DKP/Extended IP Phone extension.
You can also assign Macros on SLTs that have special keys.
How it works
• Extension 2001, frequently sets Call Forward-All Calls to an external number 26550333.
• To do this, each time, Extension 2001 must dial 131-Trunk Access Code-26550333-#*.
• Instead of having to dial this lengthy number string, a Macro can be created for Call Forward-All Calls to
External number.
• If Extension 2001 is an Extended SIP Phone or a DKP, the Macro can be assigned to a DSS key on the
phone.
• Instead of dialing this number string, the user of Extension 2001 can simply press the DSS key on which
this Macro is assigned.
• Thus when the DSS key on which a Macro is assigned is pressed, the corresponding access code is
executed.
• The system sets call forward to the external number automatically.
SARVAM UCS also supports Macros for SLT which have special keys. When each of these keys is pressed, a
special number string, which you can program is dialed.
For example, an SLT instrument has 5 special purpose keys. When these keys are pressed, the strings *50, *51,
*52, *53, *54 programmed on these keys are dialed out.
You can create Macros for the strings dialed out using the special keys, whereby the string dialed by each of these
special keys is associated with a particular function. For example, the special key for dialing *50 is associated with
Call Forward -All Calls to an external number. So, when the extension user presses *50, the system receives this
string and takes appropriate action, that is, interprets it as call forward to the external number, and sets call forward.
Thus, each time the extension user presses the special key *50, the system considers that the extension user has
dialed 131-Trunk Access Code-26550333-#*.

How to configure
You can create as many as 50 Macros using Jeeves and by dialing system commands from a telephone connected
to the SARVAM UCS.
Creating Macros using Jeeves

• Under Configuration, click Macros.
The Macros page opens. Each macro is stored against an index number. By default the Macros Number
String and Access Code are blank when the system is operated in the Enterprise Mode.
• When creating a Macro for a DKP/Extended IP Phone,
• In the Number String field, enter the strings the system should consider as command when the DSS
Key on the DKP/Extended IP Phone.
When SARVAM UCS is operated in the Hotel Mode (see Customer Profile under “System Parameters”),
Number Strings and Access Codes are assigned for features such as Front Desk, Room Service, Voice
Guided Alarm, Reservation Desk, Voice Mail System, Retrieve Message. These Macros will appear on
this page. To know more, see the topic Customer Profile in the SARVAM UCS Hospitality System Manual.
• Now, assign this Macro to a DSS Key on the DKP/Extended IP Phone.
• When you assign a Macro to a DSS Key on a DKP or Extended IP Phone,
• you must select Macro as Function Type

• As Offset, you must select the Macro Index number (1-50) at which the number string is stored. In
this example, it would be Macro Index 1.
For detailed instructions on assigning a macro to a DSS key of a DKP, see “DSS Keys Programming”.
For instructions on assigning a macro to a DSS key of the Extended IP Phone, see “DSS Key Settings”
in “Configuring Matrix SPARSH VP330”, “DSS Key Settings” in “Configuring Matrix SPARSH VP248”,
“DSS Key Settings” in “Configuring Matrix SPARSH VP310” and “DSS Key Settings” in “Configuring
Matrix SPARSH VP510” for instructions.
• When creating a Macro for a special Key on an SLT,
• In the Number String field, enter the strings the system should consider as command when the special
key on the SLT is pressed.
• In the Access Codes field, enter the strings sent by the SLT on pressing the special function key.
For example, if the SLT sends the string ‘*53’ to the SARVAM UCS when the function key for Alarms is
pressed, enter the string 163 (the feature access code for Voice-guided Alarms) in the Number String
field, and enter the string *53 in the corresponding Access code field.
The Access Code that you assign here in Macros must not conflict with any other Access Codes in the
Dial Phase. See “Access Codes”.
Creating Macros using a Telephone

To create a Macro, dial:

• 1810-Macro Index-Number String
Where,
Macro Index is from 01 to 25.
Number String is a string of 24-digits.
If the length of the Number String is less than 24-digits terminate it with #*.
To clear a Macro, dial:

• 1810-Macro Index-#*
You can program a special digit for 'Pause' in the Key Board Macro Number string using the code #3.
To assign an Access Code for the Macro, dial:

• 3115-1-Macro Index-Access Code
Where,
Access Code is a string of 4-digits.
Access Code is maximum of 6 digits. If it is less than 6 digits, terminate it with #*.

To clear Access Code assigned to a Macro, dial:
• 3115-1-Macro Index
Where,
To default the access code for the Macro, dial:

• 3165-1-Macro Index
Where,
When SARVAM UCS is operated in Enterprise mode, the default Macros Number String and Access
Codes are blank. So, when you dial this command, the values in the Macros will become blank.
When SARVAM UCS is operated in the Hotel Mode, the Number Strings and Access Codes assigned by
default will be restored when you dial this command. See the topic Customer Profile in the SARVAM UCS
Hospitality System Manual to know more.
• Exit SE mode.

Meet Me Paging
What’s this?
While a Paging announcement is being made, any extension user of SARVAM UCS can get connected to the
Paging extension, by dialing the Meet Me Paging feature code and the number of the Paging extension.
This feature is useful to Operators. Using this feature they can locate extension users who are away from their
desks and get connected to them at their current location.
How it works
• A calls B’s extension, but B is away from the desk.
• A uses Paging and makes an announcement asking B to call A’s extension.
• To get connected to A’s extension, B may use Meet Me Paging from any extension, while the
announcement is being made.
• B dials Meet Me Paging code and A’s number during the announcement.
• B gets connected to A.
• Paging is an announcement made to a group of extensions within a Page zone. Extension users,
(including those who are outside the Page zone) who want to use Meet Me Paging to answer the
Paging call, will need to know the extension number they must call. Therefore, extension users who are
paging are advised to announce their extension number.
• Meet Me Paging can be used only if the Paging call is active. Therefore, extension users who are
paging must keep their call active, if they want their call to be answered using Meet Me Paging.
How to configure
No configuration is required for Meet Me Paging. However, extension users who are using Paging to make their
announcement, must have the feature “Paging” allowed in their Class of Service.
How to use
You can use Meet Me Paging to answer a Paging call from any extension, if you know the number of the Paging
extension, and if the call is still active.

To answer a Paging call from any extension other than the Paged extensions,
• While the announcement is being made,

• Press the DSS key assigned to Meet Me Paging.
OR
• Dial 1093
• Dial the number of the paging extension.
• You get connected to the Paging extension user.

To answer a Paging call from the same extension that is paged,

• Press the DSS key assigned to Meet Me Paging.
OR
• Dial 1093
• Dial the number of the paging extension.
For SLT Users
To answer a Paging call from any extension other than the Paged extensions,

• Dial 1093-Paging Extension Number.
To answer a Paging call from the same extension that is paged,

• Dial 1093-Paging Extension Number.

Message Wait
What’s this?
The Message Wait feature of SARVAM UCS enables extension users/Operator to set Message Wait on other
extensions to deliver important messages.
If the extension user has a mailbox assigned, the Message Wait feature indicates to the extension user, the arrival
of new messages in the user’s mailbox.
Thus, Message Wait can be set by extension users as well as by the Voice Mail System.
You can set multiple Message Wait, but on different extension users. However, only one Message Wait can be set
on one extension. A Maximum of 4 Message Wait can be set on an extension.
How it works
Message Wait set by Extensions/Operator

The Operator/any extension user can set Message Wait on another extension.
• The Operator calls Extension A.
• Extension A is not at the desk to attend the call.
• The Operator has an important message to communicate. So, the Operator sets Message Wait on
Extension A, using the Message Wait key (if configured) or by dialing the feature access code.
• Extension B tries to reach Extension A, and sets Message Wait on Extension A, using the Message Wait
key (if configured) or by dialing the feature access code.
• Message Wait will be indicated to Extension A according to the Type of Message Wait Notification set for
Extension A. This may be in the form of a Stuttered Dial Tone, a Voice Message, Ring, or LED Lamp.
• If Extension A is a DKP or an Extended IP Phone and has DSS key assigned for Retrieve New Message,
the LED of this key will glow to indicate new message wait.
• Now, Extension A can dial the feature access code to retrieve Message Wait, or press the Retrieve
Message Wait Key, if assigned.
• The system will call the extension that first set Message Wait on Extension A. In this case, the Operator. If
the Operator is busy, the system will place the call on Extension B. The system will try to call the
extensions that set Message Wait until the call is answered.
• The extension that set Message Wait on A gets the CLI of A as Message Wait. A can now deliver the
message.
• The LED of the Retrieve Message Wait key, if assigned, on Extension A will be turned off after all message
wait set by other extensions on Extension A have been served.

Message Wait set by the Voice Mail System
The Voice Mail System (VMS) sets Message Wait on the extension, whenever a new message arrives in its
personal Mailbox.
• Extension A is assigned a Personal Mailbox.
• There is a new message in A’s Mailbox. The VMS indicates this to Extension A as per the Type of
Message Wait Indication set for Extension A. This may be in the form of a Stuttered Dial Tone, a Voice
Message, Ring or LED Lamp.
• If Extension A is a DKP or an Extended IP Phone, the Voice Mail key on the phone will also glow to
indicate the arrival of a new message.
• If the Retrieve Message Wait key is assigned on the DKP/Extended IP Phone of Extension A, the LED of
this key will also glow simultaneously to indicate arrival of the new voice mail.
• To listen to the new message, Extension A can

• press the Voice Mail key
or
• the Retrieve New Message key (if assigned)
or
• dial the feature access code for Retrieve New Message.
The VMS answers the call. After Extension A has listened to the new messages, the LED of the Voice Mail
key is turned off.
The LED of the Retrieve Message Wait key, if assigned, will also be turned off.
Types of Message Wait Indications

The system gives Message Wait indication to extensions according to the type of Message Wait Indication selected
for the extension. The types of Message Wait Indications offered by SARVAM UCS are:
Stuttered Dial Tone/Voice Message

• When the extension user goes OFF-Hook, the user will hear a voice message, if a pre-recorded Voice
Module has been assigned for Message Wait Notification. If no voice module is recorded and assigned,
the extension user will hear a stuttered dial tone instead.
• If you want voice message to be played as message wait notification, record and assign a Voice Module.
Refer “Voice Message Applications” for instructions.
SARVAM UCS can play only 9 Voice Modules simultaneously. The Voice Module for Message Wait
Notification will not be played if there are already 9 being played simultaneously. In this case, Stuttered
Dial Tone will be played for Message Wait Notification, when the extension user goes OFF-Hook.
LED Lamp
• When the extension user has an SLT with 'Message Wait' lamp, you can set this type of Message Wait
Indication. When Message Wait is set, the lamp will blink continuously either using High Voltage or Polarity
Reversal. The lamp will be turned off when the extension user has retrieved all the waiting messages.

Ring
• When a new Message Wait is set on the extension, the system will play Message Wait Ring (Short, Fast)
on the extension. See “Distinctive Rings”.
• The extension will ring for the duration of the Message Wait Ring Timer (configurable; default: 30
seconds). If the call is not answered within this timer, the system will ring on the extension again for as
many times as the Message Wait Ring Count (configurable; default: 10 times), and at the interval set as
the Message Wait Ring Timer Interval (configurable; default: 30 minutes).
• When the extension user answers the call, the user gets connected to the VMS or the extension that set
Message Wait.
Message Wait Signal generated by SARVAM UCS
Particulars Value
Peak Voltage 82-85 V
ON Time 100ms
OFF Time 150ms
Frequency 4Hz
DC Offset 48V
How to configure
To provide this feature to extensions, you must do the following configuration on the extensions:
• Enable the Message Wait feature in the “Class of Service (COS)” of the “Station Basic Feature Template”
of the extensions. This allows the extensions to set and cancel Message Wait on other extensions. Only
those extensions that have this feature in their COS can set or cancel Message Wait on other extensions.
By default, this feature is enabled in the COS of all extension types for all the time zones.
• Select the desired Message Wait Indication in the “Extension Voice Mail Settings”of the SLT, DKP, ISDN
Terminal, SIP extensions and Department Groups.
• If you selected Voice Message as Message Wait Indication Type for an extension, you must also record
the desired Voice Message in a Voice Module and assign it to the Message Wait application. See “Voice
Message Applications” for instructions.
• If you selected Ring as Voice mail/Message Wait Indication Type for an extension, you may configure the
following Ring Parameters:
• Message Wait Ring Timer (default: 30 seconds)
• Message Wait Ring Count (default: 10 attempts)
• Message Wait Ring Timer Interval (default: 30 minutes).
See “System Timers and Counts” for instructions.
• You may also assign DSS Keys to ‘Message Wait’ and ‘Retrieve Message Wait’ on DKP and Extended IP
Phone extensions.

For instructions on assigning these features to DSS keys of a DKP, see “DSS Keys Programming”.
For instructions on assigning these features to DSS keys of the Extended IP Phone, see “DSS Key
Settings” in “Configuring Matrix SPARSH VP330”, “DSS Key Settings” in “Configuring Matrix SPARSH
VP248”, “DSS Key Settings” in “Configuring Matrix SPARSH VP310” and “DSS Key Settings” in
“Configuring Matrix SPARSH VP510”
How to use
For EON & Extended IP Phones
To set Message Wait:
• Press DSS Key assigned to 'Message Wait'.

OR
• Dial 1076
• Dial the extension number on which you want to set Message Wait.
• Select the option ‘Set Message Wait’
To cancel Message Wait:
• Press DSS Key assigned to 'Message Wait'.

OR
• Dial 1076
• Dial the extension number on which you want to cancel Message Wait.
• Select the option ‘Cancel Message Wait’
To retrieve Message Wait:
• Press DSS Key assigned to 'Retrieve Message Wait' or the Voice mail Key when the LED glows.
OR
• Dial 1077
For SLT
To set Message Wait:

• Dial 1076
• Dial extension number on which you want to set Message Wait.
• Dial 1
To cancel Message Wait:

• Dial 1076
• Dial extension number on which you want to cancel Message Wait.
• Dial 0

To Retrieve Message Wait:

• Dial 1077

Mobility Extension
What's this?
SARVAM UCS offers mobility to its extension users whose nature of work keeps them away from their desks
frequently and for longer durations.
Using mobility extensions, the extension users of SARVAM UCS can make and receive their calls from their current
(remote) location, placing calls through the system, and can access the system just as any other normal extension
of the SARVAM UCS.
SARVAM UCS offers mobility using:

• VARTA UC Clients — VARTA ADR100 or VARTA AMP100. For details refer to the respective User Guide.
• DISA. For details, refer to the explanation given below.
How it works
SARVAM UCS supports two types of users:
• Station Users: they are extension users of SARVAM UCS to whom a dedicated physical station is
assigned on their desk.
• Virtual Users: they are extension users of SARVAM UCS who share a physical station, or may not have
any physical station allotted to them.
The facility of Mobility Extension is provided to both virtual and normal extension users using the features
“Direct Inward System Access (DISA)” and “Call Forward”.
How to configure
To provide Mobility Extension to users, the follow the steps described below.
• List out the Station Users and Virtual Users and configure them first. The software ports of SLT, DKP and
ISDN Terminals which are not assigned a physical slot - port, can be used as Virtual stations.
• Make sure that stations which are to be provided Mobility Extensions have the features Call Forward and
DISA enabled in their “Class of Service (COS)”268. Class of Service is to be programmed in the “Station
Basic Feature Template” assigned to the Mobility Extensions.
• Make a list of External numbers to which the Mobility Station users will forward their calls. Program these
numbers in the 'Allowed List' of Local, Regional, National and International Numbers, as appropriate.
• The Toll Control assigned to the station will be applied when a call is forwarded to an external number.
Make sure that the stations which are to be provided Mobility Extension have the required “Toll Control”
level for Call Forward to the External Numbers (the numbers you have programmed in the Allowed List).
268. It is possible to map the virtual user's flexible number (extension number) to a physical station. With this mapping, whenever a vir-
tual user's number is dialed, the call is placed to the assigned physical station. The physical station can be assigned by defining it
as the 'Landing Destination for Virtual Users' in the Station Advanced Feature Template assigned to the virtual user.

Toll Control level is to be programmed in the “Station Basic Feature Template” assigned to the Mobility
Extensions.
• Program the parameter 'External Call Forward for' in the “Station Advanced Feature Template” assigned to
the Mobility Extensions. This parameter defines the types of call for which the External Call Forwarding is
to be applied. Select any one of the options, Internal Calls Only, Trunk Calls Only, Internal + Trunk Calls as
required.
• Program the parameter 'DISA' in the “Trunk Feature Template” of the trunk lines which Mobility Extensions
users are to be provided access to. Make sure you select the option 'CLI Auth. Multiple Calls' in the 'DISA'
parameter of the 'Trunk Feature Template'.
• Make a list of numbers which the Mobility Extension users will use to access the SARVAM UCS from DISA
mode. Program this list of numbers in the "DISA-CLI Authentication Table".
Program this list of numbers in the 'Calling Number' field of the Authentication Table. Program the 'Port
Type' and 'Port Number' of the Station assigned to Mobility Extension Users in the 'Auto Login As' field for
the respective 'Calling Number' field. Refer the topic “Direct Inward System Access (DISA)” to know more.
How to use
The Mobility Extension Users of SARVAM UCS can use the features of SARVAM UCS from a remote location as
described below.
Receiving calls
To receive calls, the Mobility Extension User must set Call Forward on his station with an external number (mobile
number, landline number, etc.) as the destination number.
To make calls ring on the station and the external number simultaneously, the Mobility Extension User must
activate the Call Forward-Dual Ring feature on his station.
The Mobility Extension User can also choose where he wants to receive the calls during a particular time of the day.
For example, he can receive calls during a particular time of the day, that is, Time Zone on his external number and
have his calls received by his Voice mail or the Operator or any other number during another Time Zone. To do
this, he must set “Call Forward-Scheduled" on his station. Dual Ring can also be set for Call Forward-Scheduled.
Making calls
The Mobility Extension User should make a call on the DISA enabled trunk of SARVAM UCS from the external
number and the system will provide the dial tone to the user after authenticating the external number with the help
of the DISA-CLI Authentication table.
On getting the dial tone, the Mobility Extension User can make internal as well as external calls as per the “Toll
Control” and “Class of Service (COS)” assigned to his Station.
The Mobility Extension User can also dial codes of the Personal directory and Global directory numbers to use the
feature Abbreviated Dialing.
Accessing Features
The Mobility Extension User can access the system features by dialing specific codes after making calls on the
DISA enabled trunk of SARVAM UCS, or after answering the calls received on his external number.

These codes are listed below.
Activity Code to be dialed
On-Hook #0
Off-Hook #1
Flash #2
Pause #3
Terminate the call #9
Described below are instructions for Mobility Extension users on using different call management features.
Call Hold
To put a call on hold,
• When in speech with Party A, Mobility Extension User dials #2.

• The call with Party A is put on Hold.
• Press Flash (#2) to retrieve the held call and talk to Party A again.
Call Transfer
To conduct a screened Call Transfer,
• When in speech with Party A, Mobility Extension User dials #2.

• The Call of Party A is put on Hold.
• Mobility Extension User gets Feature Tone.
• Dial Trunk Access Code (TAC) followed by the number of Party B. [To make external call]
OR
• Dial Internal station number [to make call on a station]

• When in speech, dial #0 to go on hook.
• Party A and Party B will get connected.
To perform Call Transfer-While Ringing,
• Mobility Extension User is in speech with Party A.

• During speech, Mobility Extension User dials #2.
• Mobility Extension User will get Feature Tone.
• Dial TAC - number of Party B. [To make external call]
OR
• Dial Internal station number [to make call on station]

• Dial #0 on receiving Ring Back Tone (RBT).
To perform Call Transfer-On Busy Station,

• Dial number of Station user.
• Dial #0 on receiving Busy Tone.
To perform a Call Transfer-Trunk-to-Trunk,

• Dial Trunk Access Code to grab the Trunk
• After receiving the Dial Tone of the trunk, dial number of Party B.
• After Party B answers the call, dial #0.
• Party A will get connected with Party B.
Making a Second Call

To make a second call by putting the current call on hold,

• The call of Party A is put on hold.
• Mobility Extension will get Feature Tone.
• To make an external call: dial TAC - number of Party B.
OR
• To make an internal call: dial extension number.
Call Toggle (Call Splitting)

To toggle between two calls,

• The call of Party A is put on hold.
• Mobility Extension will get Feature Tone.
• Dial number of Party B.
• Make speech with Party B.
• Dial #2-1 to put Party B in hold and speech with Party A.
• Again dial #2-1 to put Party A in hold and speech with Party B.
Call Pick Up
To pick up the call of same Call Pick Up-Group,
• Make a call on DISA enabled Trunk from Mobility Extension User.

• After receiving dial tone, dial 4.
• The call ringing on the station of the same Call Pick up-Group will get connected with Mobility Extension
User.

To pick up the selective call,

• After receiving dial tone, dial 12-Station Number.
• The call ringing on the dialed station of will get connected with Mobility Extension User.
3-Party Conference
To conduct a 3-Party Conference,
• Make the first call.

• Dial #2.
• Make the second call.
• Dial #2-*3 to establish conference.
Multiparty Conference
To create a Multiparty Conference,
• Mobility Extension User has generated three-party conference with B and C.

• Dial #2 from Mobility Extension user.
• Dial number of Party D.
• After speech, dial #2-*3.
• Mobility Extension User, B, C and D are in Multi-party Conference.
• Dial number of Party E.
• Mobility Extension User, B, C, D and E are in Multi-party Conference.
• Dial number of Party F.
• Mobility Extension User, B, C, D, E and F are in Multi-party Conference.
• All Parties will get connected with each other.
To terminate the Multiparty Conference,
• When in the middle of a Multiparty Conference,

• Dial #2.
• Dial 190 after getting feature tone.
To temporarily leave the Multiparty Conference,
• When in the middle of a Multiparty Conference,

• Dial #2.
• Dial 191 after getting feature tone.
To rejoin the Multiparty Conference,
• Dial #1 to go Off Hook.

• Dial 191.
To permanently leave the Multiparty Conference,

• While in multi-party conference, dial #0 to go Off Hook.
When Dialed Station is busy

To make a call on an internal station which is busy,
• During busy tone

• Dial 3 for interrupt request.
• Dial 4 to Barge-In
• Dial 5 to Raid.
Call Forward-Unconditional
To set Call Forward-Unconditionally,

• After getting Dial Tone, dial 131-Station/Department Group/VMS.
• The system will give confirmation tone.
To set Call Forward-Unconditionally on external number,

• After getting Dial Tone, dial 131-TAC-external number.
Call Forward-Busy
To set Call Forward-Busy,

To set Call Forward-Busy on external number,

Call Forward-No Reply

To set Call Forward-No Reply,

To set Call Forward-No Reply on external number,


Call Forward-Busy/No Reply
To set Call Forward-Busy/No Reply,

• After the Dial Tone, dial 134-Station/Department Group/VMS.
To set Call Forward-Busy/No Reply on external number,

• After the Dial Tone, dial 134-TAC-external number.

• After the Dial Tone, dial 130.
Call Forward-Dual Ring


• After the Dial Tone, dial 136-1.
To cancel Call Forward-Dual Ring,

• After the Dial Tone, dial 136-1.
Call Forward-Scheduled
To set/cancel Call Forward-Scheduled,

• After the Dial Tone, dial 1175-Time Zone-Call Forward Type-Number.
• Mobility Extension users can have Call Forward and Call Forward-Scheduled set on their extension by
the Operator or by another extension user.
• Using “Call Forward-Remote” and by setting “Call Forward-Scheduled” from the SA mode (using SA
commands or Jeeves), the extension user/Operator can set Call Forward and Call Forward-Scheduled
for any Mobility extension user. Refer the respective topics to know more.

Multi-Stage Dialing
What's this?
The Multi-Stage Dialing feature of SARVAM UCS is typically used in applications like Calling Card, where
extension users are required to dial digits in stages when making a call using the calling card.
The Multi-Stage feature enables extension users to directly dial the number they want to call, and the system dials
out the number at different stages of the call by suitably modifying the number.
How it works
A typical example of Multi-Stage Dialing is the use of prepaid Calling Cards. Here, the person using a calling card
must dial a fixed number string before dialing the actual number. When using a calling card,
• Users must first dial the number of the Calling Card server, for example: 1602233 (7 digits).
• After the call is answered by the Calling Card server, users must dial the PIN provided by the calling card
service provider, for example 1132121234.
• After dialing the PIN number, users can dial the number they want to call, for example 0014125126508.
Thus, when using a Calling Card, users must dial a very lengthy number string, each time they need to make a call
using the Calling Card.
The use of Multi-Stage Dialing saves the time and effort of dialing out lengthy digits in stages.
The Multi-Stage Dialing makes use of the “Automatic Number Translation” table. This table must be configured on
the trunk from which extension users will make calls using Calling Cards.
To take the above example further,
• Say, extension users are allowed to make international calls using Calling Card from the trunk CO1.
“Automatic Number Translation” table must be configured on CO1 trunk.
• The Automatic Number Translation table consists of Dialed Number Strings, Strip Digit and Add Prefix
Number strings.
• In Dialed Number, you must configure ‘00’, the prefix for international numbers.
• In Add Prefix you must configure the Calling Card server number and the PIN Number.
• Keep Strip Digits as 00.
• As the system must wait for the Calling Card server to answer before dialing the PIN, you must configure
Wait for Answer between the Calling Card server number and the PIN number.
You must insert a delay by configuring the Pause Timer after the PIN number and the destination number.

• The Automatic Number Translation table would look like this:
Index Dialed Number String Strip Digits Add Prefix
1 00 00 1602233W1132121234P
32
• When the Automatic Number Translation table is configured, the Extension user can simply dial the Trunk
Access code and the destination number: 0/5/61/62/63/64 - 0014125126508.
• The system matches the dialed number with the Dialed Number String of the ANT table, the number
matches with the entry ‘00’ stored in the table.
• The system dials the Add Prefix Number string 1602233 (number of the calling card server). It waits for the
calling card server to answer the call.
• When the call is matured, i.e. the calling card server has answered the call, the system dials the PIN
number 1132121234 and waits for the Pause Timer before dialing the destination.
Thus, the extension user directly dials the desired destination number and the system substitutes this number
by adding the Calling Card server number and PIN number and dials these numbers in two stages.
How to configure
Configuring Multi-Stage Dialing using Jeeves

To be able to use Multi-Stage Dialing, you must configure the following:
• Automatic Number Translation table on the trunk you want to use this feature. For instructions on
configuring ANT on different Trunk types, see “Automatic Number Translation” and to assign the ANT
Table number see “Outgoing Trunk Bundle”.
• Configure the DTMF Out Dial on the Trunk. Set the DTMF ON Time and the DTMF Inter Digit Pause
Timer and Pause Timer to the required values.
For instructions on configuring DTMF Out Dial on different Trunk types, see “Configuring CO Trunks”,
“Configuring E1 Trunks”, “Configuring T1 Trunks”, “Configuring BRI Trunks”, “Configuring Mobile Trunks”,
“Configuring SIP Trunks”, “Configuring E&M Lines”.
• If required you may configure the Call Proceeding Tone for Multi-stage Dialing as Network tone,
Pseudo Tone, or Silent. For instructions, see “System Parameters”.

Configuring Multi-Stage Dialing using a Telephone
Notes for programming Number String:
• It is recommended to SE, to not to program Pause character "P" before "W" character for the number
string to be out dialed from Mobile port. Else, the GSM Module may get restart while out dialing the DTMF
digits without call maturity signal.
• Use following Codes to program the special digits:
Special Digits Code for Programming through Command
Flash (F) #2
Pause (P) #3
A #4
B #5
C #6
D #7
+ #8
. (dot) #9
# ##
* **
W *1
Enable 'ANT' for the trunk port using command 6702. Refer chapter “Outgoing Trunk Bundle” for more details.
Assign the ANT Table to the trunk port using command 6702. Refer chapter “Outgoing Trunk Bundle” for more
details.
Refer chapter “CO Hardware Template” for command '5902' to program Pause Timer on CO Trunk.
Refer chapter “Configuring PRI Trunks” for command '6109' to program Pause Timer on T1E1 Trunk.
Refer chapter “Configuring BRI Trunks” for command '6209' to program Pause Timer on BRI port.
Refer chapter “Configuring Mobile Trunks” for command '8014' to program Pause Timer on Mobile port.
Refer chapter “E&M Feature Template” for command '6002' to program feature Pause Timer on E&M trunk.
Refer chapter “CO Hardware Template” for command '5902' to program DTMF ON Time on CO Trunk
Refer chapter “Configuring PRI Trunks” for command '6117' to program DTMF ON Time on T1E1 Trunk.
Refer chapter “Configuring BRI Trunks” for command '6210' to program DTMF ON Time on BRI port.
Refer chapter “Configuring Mobile Trunks” for command '8015' to program DTMF ON Time on Mobile port.
Refer chapter “CO Hardware Template” for command '5902' to program Inter Digit Pause Time on CO Trunk
Refer chapter “Configuring PRI Trunks” for command '6118' to program Inter Digit Pause Timer on T1E1 Trunk.
Refer chapter “Configuring BRI Trunks” for command '6211' to program Inter Digit Pause Timer on BRI port.
Refer System Parameters for command '5311' for programming of Call Proceeding Tone.

Music on Hold (MOH)
What’s this?
The music played to extension users and external callers who are put on hold is called Music on Hold (MoH).
How it works
SARVAM UCS supports Music on Hold from an Internal Music Source.
A Voice Module assigned the Music on Hold application serves as the source of music for this feature. By default,
Voice Module 1 is assigned for Music on Hold.
When a caller is put on hold, SARVAM UCS plays the music recorded in Voice Module 1.
You can play a voice message of your choice instead of music to the callers. The message may contain any
promotional information about your company or services provided by your organization, etc. For this, you must
record a Voice Module with the custom message and assign the Voice Module to the Music on Hold application.
If the option Routing Group is selected as the Alarm Notification Type for an extension, when the
extension goes Off-hook to answer an alarm call, and the extensions in the Routing Group for Alarm
Notification are busy, Music-on-Hold will also be played to the extension answering the alarm call.
If your DKP/SIP Extension supports multiple call appearance and you want the system to play MOH to
internal callers, when your extension is busy, enable the Play MOH to Queued Internal Calls on DKP/
Extended IP Phone check box in “System Parameters”. As soon as your extension is free, Ring Back
Tone will be played to the caller.
How to configure
You can record a piece of music or a message of maximum 16 seconds duration and assign it to Voice Module 01,
which is reserved for Music-on-Hold.
Refer the topic “Voice Message Applications” for instructions on recording and assigning voice modules.

Mute
What’s this?
This feature helps the extension user to disconnect the speech transmission path in the middle of a conversation.
The extension user can still listen to the opposite party because the receiving path remains connected. Mute is
useful when you want to consult someone in the middle of a conversation, but do not want the opposite party to
listen to your discussion. You can Mute a call before making a call or during speech.
How it works
• A wants to consult to C in the room, but does not want B to hear their conversation.
• A presses the Mute Key.
• The transmit speech path from A to B is disconnected. The receive path remains connected.
• So, A will be able to hear B, but B will not be able to hear the conversation between A and C.
• When A has finished consulting C, to resume speech with B, A presses the Mute key again.
• The transmit speech path from A to B is restored. A and B are in speech again.
How to use

To mute a call before making the call:
• Press the ‘Mute’ Key
The LED of the key glows.
• Dial a number on confirmation tone.
OR
• Dial 1052
• Dial desired number.
To mute a call during speech:

• Press the ‘Mute’ Key to silence outgoing speech.
OR
• Dial 1052
To resume outgoing speech:

• Press the ‘Mute’ Key.
The LED of the key is turned off.
OR
• Dial 1052
For SLT Users
To mute a call before making the call:

• Dial 1052
• Dial desired number.

To mute a call during speech:
• Press Flash Key.
• Dial 1052
To resume outgoing speech:

• Press Flash Key.
• Dial 1052

Number Lists
What’s this?
A ‘Number List’ is a group of number strings. SARVAM UCS uses Number Lists to support different features as Toll
Control, Call Duration Control, Call Taping, Call Back on Trunk Ports, Station Message Detail Recording (SMDR).
SARVAM UCS supports 16 Number Lists. Each Number List can contain upto 999 number strings. Each number
string consist of a maximum of 16 characters.
The number strings are stored against Location Index numbers in the Number List. The Location Index numbers
start from 001 to 999.
The default values of the Number Lists in the system are shown below:
Default Number Lists
Location List Number

Index 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16
001 00 0 00 * * * * * * * * * *
002 0 * * # # # # # # # # # #
003 1 # # F F F F F F F F F F
004 2 F F
005 3
006 4
007 5
008 6
009 7
010 8
011 9
012 *
013 #
014 F
015 +
016
: : : : : : : : : : : : : : : : :
999

Use of Number Lists in various Features of SARVAM UCS

When Call Back on set on any of the Trunk port types (CO, Mobile, BRI, T1E1, SIP), you must program two lists:
• Call Back Incoming Number List: this list defines the numbers which are eligible for a call back.
• Call Back Outgoing Number List: this list define the number to which the call back is to be made.
By default, Number List 15 is assigned to Call Back Incoming Number List, and Number List 16 is assigned to Call
Back Outgoing Number List.
Call Duration Control

To set Call Duration Control (CDC) feature on an extension, you must program the Call Duration Control Table, in
which you must program the numbers on which CDC must be applied as well as the numbers on which CDC must
not be applied.
This requires you to program and assign two Number Lists: the Apply CDC to Number List and the Do Not Apply
CDC to Number List.
By default, Number List 07 is assigned to Apply CDC Number List, and Number List 08 is assigned to Do Not Apply
CDC Number List.
Refer “Call Duration Control (CDC)” to know more.
Call Taping
Call Taping allows you to record conversations of incoming and outgoing internal and external calls. When you set
Call Taping on an extension for external calls, the SARVAM UCS uses two Lists: Number List - Incoming Calls: this
list has the list of numbers of external callers whose conversation is to be recorded.
Number List - Outgoing Calls: this list has phone numbers of external called parties whose conversation is to be
recorded. The system matches the incoming and outgoing numbers with the respective lists to apply Call Taping.
By default, Number List 09 is assigned to Incoming Calls Number List, and Number List 10 is assigned to Outgoing
Calls Number List.
Refer the feature description “Call Taping” to know more.
Toll Control
The Toll Control Call Privilege Type “Limited Calls” allows and restricts dialing of telephone numbers starting with a
particular digit or a particular area code or certain telephone numbers only. To apply Call Privilege type ‘Limited
Calls’ you must program an Allowed Number List with numbers that are to be allowed, and a ‘Denied List’ with
numbers that are to be restricted.
By default, Number List 01 is assigned as Allowed List as well as Denied List.
Refer the feature description for “Toll Control” to know more.

Station Message Detail Recording (SMDR)
SARVAM UCS supports ‘Destination wise’ storage of outgoing calls as an SMDR Filter, using which it is possible to
store calls made from and received on selected destination numbers. To set Destination wise storage of calls in the
SMDR Buffer, you must program and assign a Number List containing the phone numbers whose call details need
to be tracked and stored. SMDR of an outgoing call will be tracked and stored only if it matches with an entry in the
Number List assigned.
Similarly, SMDR Reports of Incoming and Outgoing Calls can be generated and printed for selected numbers by
programming and assigning a Number List.
By default, Number List 02 is assigned to Destination Wise storage of SMDR and SMDR Report Printing.
Refer the topics “Station Message Detail Recording (SMDR)”, “Station Message Detail Recording-Storage”,
“Station Message Detail Recording-Report”.
SMS on No Reply
SMS on No Reply feature is used to send an SMS to the mobile user only, when an extension user calls the mobile
user and the mobile user doesn't answer the call. The SMS that is sent to the mobile user contains the extension
user’s contact information and the message to be conveyed in brief, so that the mobile user can call back if
necessary. You must enter only the first digit of the number string, or a part of the string, or the complete number
string in the Number list for sending SMS. The system compares the dialed numbers with the numbers in this list to
confirm if the outdailed numbers are mobile numbers. Only if a match is found the SMS is sent.
By default, Number List 16 is assigned for SMS on No Reply.
Refer the feature description for “SMS on No Reply” to know more.
How to configure
Take a pen and a paper. Decide which of the above-mentioned seven features are to be used. Number List
according to the feature for which it is to be used.
Number Lists can be programmed from Jeeves or by dialing SE commands from a telephone.
Programming Number List using Jeeves


• Under Configuration, click Number List to open the page.
• Select the desired List Number. For example, Number List 03-04. Now click ‘001-250' of Number list 03-04.
The location Index 001 to 250 will appear for both lists on the page.

• Enter each number string against a Location Index (refer to the list you prepared).
• Click Submit at the bottom of the page to save your list.
• Assign the Number List you prepared to the relevant feature.
• Log out of Jeeves if you have no other configuration tasks.
Programming Number List using a Telephone

• Enter SE mode.
To program a number in a Number List, dial:

• 4302-Number List-Location Index-Number-#*
Where,
Where
List is from 01 to 16.
Number is a number string of maximum 16 digits. The digits allowed are: 0-9, #, *, A, B, C, D, F, P, +.
Refer the table below for codes for dialing the special digits. Terminate the number string with #*
Special Digit Code
Flash (F) #2
Pause (P) #3
A #4
B #5
C #6
D #7
+ #8
Dot (.) #9
# ##
* **
To clear the number programmed at a Location index in a Number list, dial:

• 4302-Number List-Location Index-#*
Where,
To default Number Lists, dial:

• 4301-1-Number List to default a single number list.
• 4301-2-Number List-Number List to default a range of number lists.
• 4301-* to default all number lists.
Where,
• Exit SE mode.

OFF-Hook Alert
What's this?
When the handset of an extension is not placed correctly, it will not be possible for the Operator or any other
extension to call the extension. Also, incoming calls will not reach the extension, Alarms and Reminders cannot be
placed on that extension.
To avoid this inconvenience, the SARVAM UCS supports the feature 'OFF-Hook Alert', whereby the system detects
and informs the Operator of the extension phone that remains OFF-Hook accidentally.
How it works
To give the Operator an OFF-Hook Alert,
• SARVAM UCS places a call on the Operator's phone.
• It displays a message on the Operator's phone “<extension number> Stand-By”
• When the Operator answers the call, s/he is played a confirmation tone, the text message “Hangup
<extension number> Properly” is displayed.
• The Operator can send someone to inform the extension user to place the handset of the phone correctly.
• If the extension phone is an SLT, OFF-Hook Alert will be given to the Operator phone only. The Operator
phone can be an EON or a SLT with CLI support.
• If the extension phone that is OFF-Hook is EON, the SARVAM UCS will activate 'OFF-Hook Alert' on the
extension phone, by playing the Error Tone continuously, on speaker to draw the attention of the extension
user.
• While the Error Tone for OFF-Hook Alert is being played on the extension phone, if the user presses the
Speaker Key, the Error Tone will continue to be played on the handset until it is replaced correctly.
How to configure
• the 'OFF-Hook Alert to Operator' flag must be enabled by the System Engineer in the 'System General
Parameters'.
• the Operator phone can be EON or SLT with CLI support, the extension phones may be EON or SLT.
Programming 'OFF-Hook Alert to Operator' using Jeeves


• Go to the parameter Give Off-hook Alert to Operator, select the check box to enable the flag.
• Click Submit to save the change.
• Log out of Jeeves or continue programming as required.
Programming 'OFF-Hook Alert to Operator' using a Telephone

• Dial 5333-1 to enable the flag.
• To disable the flag, dial 5333-0.
• Exit SE mode.

One Touch Transfer
What is this?
SARVAM supports the UC feature, One Touch Transfer. Using this you can transfer an ongoing call from one
extension to another mobile/fixed extension without putting the call on hold or dialing the destination extension
number.
SARVAM UCS will serve the One Touch Transfer request made by a DKP or a SIP Extension (Matrix Extended IP
Phones or Mobile Clients) only. The fixed destination extension number can be a SLT, DKP, SIP Extension or an
ISDN Terminal.
How it works
Consider the scenario, wherein you have the following registered as your extensions:
1. VARTA Mobile UC Client is registered as SIP Extension 1 at location 1

2. Extended IP phone is registered as SIP Extension 1 at location 2.
• You are in speech with A using your Extended IP Phone and you want to move away from your desk.
• In this case, you can transfer the ongoing call on your VARTA Mobile UC Client using One Touch Transfer.
• Press the DSS key assigned to One Touch Transfer (make sure the extension number of SIP Extension 1
is pre-configured as destination number for One Touch Transfer).
• You will receive the call on your VARTA Mobile UC Client.
• Answer the call from the VARTA UC Client. Speech is established and the call on your Extended IP Phone
disconnects.
• You can now move away from your desk but your call can continue.
Similarly, you can transfer a call to another fixed extensions also.
You can access One Touch Transfer only during an ongoing 2-way speech. A held or a waiting call cannot
be transferred using One Touch Transfer.
For using One Touch Transfer between Extended IP Phone/VARTA UC Clients registered on different
locations of the same SIP Extension, make sure the Call Appearance of the SIP Extension is more than 1.
How to configure
To access this feature,
• Make sure the Basic Features are enabled in the Class of Service assigned to you. For instructions, see
“Class of Service (COS)” and “Station Basic Feature Template”.
• Make sure you have configured a DSS Key for One Touch Transfer. For instructions on configuring DSS
Key, see the topic “DSS Keys Programming”.

• Configure the destination extension number for One Touch Transfer. This number can be configured by
extension users themselves from the Phone Menu or for any other extension from the SA mode.
Configuring One Touch Transfer using SA Jeeves

• Click Submit.
• Click One Touch Transfer to expand.
• In Transfer to Extension Number, enter the destination extension number on which you want the call to
be transferred, when the feature One Touch Transfer is accessed.
Configuring One Touch Transfer by Extension Users using Phone Menu

To configure the Destination Extension Number,
• Scroll down the Local Menu of the phone.

• Select One Touch Transfer by pressing the Enter Key.
• Select Set Transfer Number by pressing the Enter Key.
• Enter the destination extension number on which you want the call to be transferred.
• You hear Confirmation tone.

How to use
To use One Touch Transfer:

• During an ongoing call, press the DSS Key assigned to One Touch Transfer.
• The destination extension number configured for One Touch Transfer starts ringing.
• Answer the ringing call.
• You will be in speech with the remote party.

Outgoing Trunk Bundle
What’s this?
The Outgoing Trunk (OG) Trunk Bundle is set of parameters that completely define the grouping of similar
channels. The word channel refers to a speech path. By this definition, a channel can be a CO trunk, an E&M trunk,
a Mobile port. Each speech path of the T1/E1 line is a channel. Bundles of similar trunks are formed. These
Bundles are used to form an OGTBG. The SARVAM UCS supports 128 OG Trunk Bundles.

• Under Configuration, click OG Trunk Bundle.
Configure the following parameters for each Bundle Number:
• Trunk Type: Specific trunk bundle consists of some of these ports. The different types of ports are CO,
BRI, T1E1PRI, E&M, Mobile and SIP.
• Trunk Number: Once the type of port is identified in the bundle, it is required to identify the number of that
port, because there can be more than one port for the given type.
• Start Channel Number: This is applicable to BRI and T1E1 Trunks.
• Total Trunk Count: This is the number of trunks to be kept in the same bundle. For CO and E&M this
could be 128. This value for the BRI and T1E1 Trunks is counted from the start channel. For example, if
the port type is CO, port number is 002 and channel count is 025, then CO channels 002 to 027 would be
grouped together. Consider a second example where channels 15 to 25 of T1E1PRI port are to be
programmed in one channel group, then the port type will be T1E1PRI, port number would be 5, start
channel number will be 15 and channel count will be 11.

• Rotation Type: This parameter shows which channel should be selected when the next call lands on that
port. For example, if ascending order is selected, the system checks 001-128, for first free channel and if
descending order is selected the system checks from 128-001.
• Ascending Order:
• 001 to 128 (for CO/E&M)
• 01 to 30 (for T1E1PRI)
• 01 to 02 (for BRI)
• 01 to 32 (for SIP)
• Descending Order:
• 128-001 (for CO/E&M)
• 30 to 01 (for T1E1PRI)
• 02 to 01 (for BRI)
• 32 to 01 (for SIP)
• Cyclic:
• Always the next channel is picked for a new OG call.
• ANT Apply: Select from enable/disable as required. To use ANT feature, it should be enabled for the trunk
port from which the number is to be dialed out.
• ANT Table No.: This is a Table number in which you have configured the Dialed Numbers and their
corresponding Strip Digits and/or Add Prefixes. This table number is assigned to the specific trunk port
from which the number is to be dialed. Refer chapter “Automatic Number Translation” for more details.

Use following command to program the feature in an OG Trunk Bundle Number:
6702-1-OG Trunk Bundle Number-Feature Number-Code
6702-2-OG Trunk Bundle Number-OG Trunk Bundle Number-Feature Number-Code
6702-*-Feature Number-Code
Where,
Use following command to set default values for OG Trunk Bundle:

6701-1-OG Trunk Bundle Number
6701-2-OG Trunk Bundle Number-OG Trunk Bundle Number
6701-*
Where,

Default values of OG Trunk Bundle
Feature Number
1 2 3 4 5 6
Bundle No. Alternate Number

Trunk Port Start Total
Rotation Translation (ANT)
Channel Trunk
Type
Type Number No. Count Apply ANT Table No.
001 CO 001 01 008 Ascending No 1
002 MOB 001 01 032 Ascending No 1
003 BRI 001 01 002 Ascending No 1
004 T1E1 001 01 032 Ascending No 1
005 SIP 001 01 032 Ascending No 1
006 None 000 01 017 Cyclic No 1
007 None 000 01 017 Cyclic No 1
: None 000 01 017 Cyclic No 1
060 None 000 01 017 Cyclic No 1
061 CO 001 01 008 Ascending No 1
062 BRI 001 01 002 Ascending No 1
063 T1E1 001 01 030 Ascending No 1
064 MOB 001 01 032 Ascending No 1
065 None 000 01 017 Cyclic No 1
: None 000 01 017 Cyclic No 1
128 None 000 01 017 Cyclic No 1
Parameter Values:
Code Port
Type
0 None None 000 No
1 CO 03 001-128 01-30 001-128 Ascending Yes 1-8
2 BRI 04 001-032 Descending
3 T1E1 05 001-008 Cyclic
4 E&M 06 001-128
5 Mobile 25 001-064
6 SIP 26 001-032
Trunks

OG Trunk Bundle Group
What’s this?
OG Trunk Bundle Group provides efficient allocation of trunks to different stations.
All the trunks connected to the system can be bunched in different groups called OG Trunk Bundle Group.
Maximum 8 Trunk Bundles can be put in one OG Trunk Bundle Group and 32 such OG Trunk Bundle Groups can
be formed.These OG Trunk Bundle Groups can be allotted to each individual extension. An Extension can be
allotted different OG Trunk Bundle Group during different timings of the day.
How it works
System uses two methods while selecting a trunk from the OG Trunk Bundle Group: ‘Remember last trunk’ and
‘Don’t Remember last trunk’.
In ‘Remember last trunk’ method, the system remembers the last trunk used and allots next trunk in the group to
the extension. In ‘Don’t remember last trunk’ method, the system searches for a first free trunk from the group.
Following flow chart depicts the chronology of events when an extension grabs a trunk.
Start
Station User dials trunk

group access codes
Which LCR type is

selected
Service
Time based LCR Mixed LCR No LCR Provider Number based LCR
based LCR
System finds the A System allots a free System waits for dialing
cheapest trunk from trunk from the OG B of the entire number
the group depending trunk bundle group
upon the time at depending upon the
which the call is made rotation flag System finds the cheapest
trunk from the group
depending upon the
End tariffs offered according
to the area/distance/
phone number dialed
Is the
cheapest trunk C
free?
No
Is the No
cheapest trunk
Are other Trunks free?
No
available in this Yes
group? Are other Trunks
System dials out
available in this
the number No
System gives busy group?
Yes
tone to the Station End System gives busy
Yes tone to the Station
System waits for Select next cheapest
the next action trunk in the group Select next cheapest
from the user System waits for
trunk in the group
the next action
from the user
End

A B
System waits for dialing System waits for dialing

of the entire number and of the entire number and
stores the number at a stores the number
memory location
System finds the cheapest System ignores the area code,

trunk from the group checks the number in Service-
depending upon the time as provider based LCR Table and
well as the tariffs offered routes the call according to
according to the area/ the trunk assigned for the
distance/phone number dialed respective number
C C

• Under Configuration, click OG Trunk Bundle Groups.
Choose the OG Trunk Bundle Group number (01-08) you want to use. In each group you can program
maximum 8 'members'.
• Now program the following parameters for the selected OG Trunk Bundle Group:
• Rotation: Select the Rotation check box to enable rotation for routing outgoing calls in the group which
has multiple 'member' trunks. When enabled, each outgoing call will be routed through the next
member to the one that routed the previous call. This ensures equal distribution in routing outgoing
calls. The flag has no relevance if the group has only one member. For detailed instructions, see
“Outgoing Trunk Bundle”

• Least Cost Routing: If required select the desired method — Time-based LCR, Number based LCR,
Time and Number based LCR or Service Provider-based LCR — as per your requirement. For details,
see “Configuring LCR”.
• OG Trunk Bundle Member1 to 8: Select the desired OG Trunk Bundle numbers you have already
created. If you have not created the OG Trunk Bundles, for instructions see “Outgoing Trunk Bundle”.
Configure only as many OG Trunk Bundles as members in the group as you want and set the
remaining Members as '000'.
• Repeat the same steps to create another OG Trunk Bundle Group.
Configuring using Telephone

Step 1
Use following command to make default OG Trunk Bundle Group:
1401-1-OGTBG Number
1401-2-OGTBG Number-OGTBG Number
1401-*
Where,
Step 2
Use following command to set OG Trunk Bundle:
1402-1-OGTBG Number-Destination Index-OG Trunk Bundle
1402-2-OGTBG Number-OGTBG Number-Destination Index-OG Trunk Bundle
1402-*-Destination Index-OG Trunk Bundle
Where,
OG Trunk Bundle is from 001 to 128.
For Defaults refer Table1.
Step 3
Use following command to set Rotation Flag. For explanation on how to use, refer important point at the end of
chapter:
1403-1-OGTBG Number-Flag
1403-2-OGTBG Number-OGTBG Number-Flag
1403-*-Flag
Where,

Flag Meaning
0 Rotation OFF
1 Rotation ON
For Defaults refer Table1.
Step 4
For LCR Type-refer chapter “Configuring LCR”.
Step 5
For CPS refer chapter “Least Cost Routing-Carrier Pre-Selection”.
Step 6
Refer the topic “Station Basic Feature Template” for details on assigning OGTBG to extensions.
Step 7
How to assign an access code to OGTBG?
There are maximum 6 trunk access codes. These are Flexible access codes. Trunk access codes are common for
all the users. They cannot be different for different extension.
A default trunk access codes table is given below:
OGTBG Index Default Trunk Access Code
1 0
2 5
3 61
4 62
5 63
6 64
Use following command to program the desirable access code for a trunk access index (TAC):
3112-1-OGTBG Index-Access Code-#*/Press <Hold>
3112-2-OGTBG Index-OGTBG Index-Access Code-#*/Press <Hold>
3112-*-Access Code-#*/Press <Hold>
Where,
OGTBG Index is from 1 to 6.
Access Code is maximum 6 digits (Generally access code for trunk is of two digits).
Use following command to clear the access code for a OGTBG index:
3112-1-OGTBG Index-#*
3112-2-OGTBG Index-OGTBG Index-#*
3112-*-#*
Use following command to assign default access code for a OGTBG index:
3162-1-OGTBG Index

3162-2-OGTBG Index-OGTBG Index
3162-*
How to use
1 Lift the handset. Dial tone
2 Dial 0/5/61 to 64. Dial tone of the trunk.
3 Dial desired external number.
• To grab OG Trunk Bundle Group 3, the user should dial ‘61’.
• To grab OG Trunk Bundle Group 1, the user should dial ‘0’.
How the Rotation flags work?

• Eight OG Trunk Bundles can be programmed as members of each OGTBG.
• OGTBG has “Rotation” flag, and each OG trunk bundle has Rotation type “Cyclic/Descending/Ascending”.
• These two flags don’t have any relation with each other, and so, will work in isolation.
If Rotation is ON in the OGTBG:

• The first call will get routed using the “OG Trunk bundle Member1”. The trunk port from the OG Trunk
Bundle member 1 will get selected using the rotation type programmed for the OG trunk bundle
programmed as member 1.
• When there is a second call, it will get routed using the “OG trunk bundle Member 2”, and the trunk port
from the OG trunk bundle member 2 will get selected according to the rotation type programmed in the OG
Trunk Bundle programmed as member2.
• Accordingly, the member of the OGTBG will be accessible to the station user accessing the OGTBG in
sequence.
If the Rotation is OFF in the OGTBG:

• The calls will always get routed from the “OG trunk Bundle member 1” if any trunk/channel in it is free. If all
the trunks/channels of the “OG trunk bundle member 1” are busy, then the call will get routed using the
“OG trunk bundle member 2”.
• Now the trunk/channel to route the call will get selected as per the rotation type programmed for the OG
trunk bundle used as member 2. When the trunk ports of OG trunk bundle programmed in member 1 and
member 2 all are busy, the OG trunk bundle member 3 will be used to route the call.
• Thus the Rotation flag of OGTBG will be used to select the OG trunk bundle member1 to member 8 as per
call basis while the rotation type flag associated with the OG trunk bundle will decide the rotation
mechanism to select the trunk port from the particular OG trunk bundle.

Default OG Trunk Bundle Group Table (Table-1):
OGTB OGTB OGTB OGTB OGTB OGTB OGTB OGTB

Group
Rotation LCR Member Member Member Member Member Member Member Member
No.
1 2 3 4 5 6 7 8
01  None 01 02 03 04 05 00 00 00
02  None 01 00 00 00 00 00 00 00
03  None 01 00 00 00 00 00 00 00
04  None 01 00 00 00 00 00 00 00
05  None 01 00 00 00 00 00 00 00
::  None 01 00 00 00 00 00 00 00
21  None 01 00 00 00 00 00 00 00
22  None 01 00 00 00 00 00 00 00
23  None 01 00 00 00 00 00 00 00
24  None 01 00 00 00 00 00 00 00
25  None 01 00 00 00 00 00 00 00
26  None 01 00 00 00 00 00 00 00
27  None 01 00 00 00 00 00 00 00
28  None 01 00 00 00 00 00 00 00
29  None 01 00 00 00 00 00 00 00
30 None 61 62 63 00 00 00 00 00
31 None 64 00 00 00 00 00 00 00
32 None 61 62 63 64 00 00 00 00

Paging
What's this?
Paging allows you to make announcements to groups of extension users and to make public announcements over
a public address system. You can deliver a message to a mass of people at once by just lifting the handset of your
phone and dialing a code.
This feature is useful when you want to call several people at once; for example, to inform them about a meeting
you have scheduled. If the persons you want to call have Digital Key Phones (DKP) or the Matrix Extended IP
Phones, Radio devices or Open SIP Phones as their extensions, you can use paging instead of calling them up one
by one.
SARVAM UCS supports paging where announcements are made on DKP/SIP extensions /Radio Ports.
The extensions which are to be paged must be included in 'Page Zones'.
• You can start paging from an SLT, a DKP or any SIP Extension. However, the paged extensions must
be DKPs, Extended IP Phones, Radio device or Open SIP Phones. The Open SIP Phones on which
you are paging must support Call-Info or Alert-Info header for Paging.
• When the Paging call is generated in SIP Extension having multiple call appearance and already a call
is present on the SIP Extension then the SARVAM UCS will place the Paging call as normal call on the
SIP Extension (as headers required in INVITE for paging call and intercom call are same).
• Paging is a one-way communication. As the mic of the paged extensions is muted during Paging, the
users of the paged extensions cannot speak to the paging extension.
How it works
The Pre-requisites
• Page Zones must be created. Each Page Zone accommodates up to 16 DKP/SIP extensions/Radio
devices. You can create 12 different Page Zones of 16 DKP/SIP extensions/Radio devices each.
• Paging must be enabled in the Class of Service allowed to the DKP/SIP/SLT/Radio extension from which
this feature is to be used.
The Process
• A user of a DKP/SLT/SIP Extension having Paging in its Class of Service, dials the Access code for Paging
and the Number of the Page Zone to which the user wants to make the announcement.
• The system activates the speakers of the DKP/SIP Extensions programmed in the Page Zone number.
The system activates the speakers only of those DKP/SIP/Radio extensions in the Page Zone that are
free.
• The calling DKP/SLT/SIP Extension user makes the announcement.

• All DKPs/SIP/Radio Extensions in the Page Zone can hear the announcement. But as the mic of their
phones is muted, their speech will not be heard by the calling DKP/SLT/SIP extension user.
• To answer the Paging call the desired extension user must use Meet Me Paging while the Paging call is
active. For details refer, “Meet Me Paging”.
• If no reply is received via Meet Me Paging, the calling DKP/SLT/SIP extension goes ON-Hook after the
announcement.
• The system deactivates the speakers of the DKP/SIP/Radio Extensions.
How to configure
For this feature to work, you must create Page Zones and enable this feature in the Class of Service of the
extensions which are to be allowed this feature.
Allowing Paging in Class of Service

In the default factory settings, Station Basic Feature Template Number 01 is assigned to all the stations of
SARVAM UCS. Template 01 is assigned CoS group 01 which has Paging enabled. So, all stations of SARVAM
UCS can page.
If you want to allow Paging to all stations, retain CoS group 01 in Template 01.
However, if Paging is to be disallowed to some stations then follow these steps:
a. Define a CoS group with Paging disallowed.
c. Assign this new Template to the stations to which Paging is to be disallowed.
programming.
Creating Page Zones

Decide how many Page Zones you want to create. You can create as many as 12 different Page Zones. You can
include upto 32 DKP/SIP/Radio extensions.
For each Page Zone number, decide and assign the DKP/SIP/Radio extensions.
On a sheet of paper create a three-column table for each page zone, as shown below. Enter the Type of extension,
whether DKP, SIP or Radio, and the Software Port number of the extensions you want to include in the page zone.
Page Zone 1
DKP/SIP
Member Number Type of Extension Extension Port
Number
1 DKP 002

DKP/SIP
Number
2 DKP 003
3 DKP 008
4 SIP 009
32
Page Zone 2
DKP/SIP
Number
1 SIP 007
2 SIP 010
3 SIP 012
4 DKP 013
32
Now, program Page Zones using Jeeves or by dialing SE commands from a Telephone.
Programming Page Zones using Jeeves

• Under Configuration, click Page Zones to open the page.
• The first three page zones appear on this page.
• Refer to the Page Zones you created on paper and program the following parameters in each Page Zone:
• Name of Page Zone: Enter the name you want to assign to the Page Zone.
• Member Type: Select the type of extension you want to include in the Page Zone: DKP, Radio or SIP
extension.

• Port Number: Enter the number of the Software Port of the extension you selected in the Port Type.
• Click Submit at the bottom of the page to save your Page Zone settings.
• To go to other Page Zones, you may click the hyperlinked page zone numbers on the top of the Jeeves
screen.
• If you have completed programming Page Zones, you may log out of Jeeves.
Programming Page Zones using a Telephone

To program a DKP/SIP Extension in a Page Zone, dial:

• 2302-1-Page Zone-Member Index- Port Type-Port Number to program a DKP/SIP extension in a
single Page Zone.
• 2302-2-Page Zone-Member Index- Port Type-Port Number to program the same DKP/SIP Extension
in a range of Page Zones.
• 2302-*-Member Index- Port Type-Port Number to program the same DKP/SIP Extension in all Page
Zones.
Where,
Page Zone is from 01 to 12.
Member Index is from 01 to 32.
Port Type is 02 for DKP, 34 for SIP Extension, 40 for Radio Extension.
Port Number is the software port number of the extension to be included in the Page Zone. Port
Number is
001 to 128 for DKP.
001 to 999 for SIP Extensions.
01 to 16 for Radio Extensions
To include/exclude a member DKP, SIP or Radio Extension in/from a Page Zone, dial:

• 2303-1-Page Zone-Member Index-Flag to include/exclude an extension in/from a single Page Zone.
• 2303-2-Page Zone-Page Zone-Member Index-Flag to include/exclude the same extension in/from a
range of Page Zones.
• 2303-*-Member Index-Flag to include/exclude the same Extension in/from all Page Zones.
Where,
Page Zone is from 01 to 12.
Flag is
0 for Exclude member extension from Page Zone.
1 for Include member extension in Page Zone.
• Exit SE mode.
How to use
It is possible to page from a DKP, a SIP Phone and an SLT.

• Press DSS Key assigned to Paging.
OR
• Dial 1074
• Enter Page Zone Number on the prompt on your phone's display.
• You get the prompt on your phone's display: Start Paging <Page Zone Number>.
• Make your announcement.
• Go ON-Hook at the end of your announcement.
For SLT Users

• Dial 1074-Page Zone Number.
• Start your announcement.
• Replace the handset after completing your announcement.

Peer-to-Peer Calling
What’s this?
Making an IP call without the intervention of a proxy server is called Peer-to-Peer Calling. As Peer-to-Peer calling
does not require a proxy server, voice communication using this application can be done virtually free of cost. The
major cost savings offered by this application makes it a very attractive mode of inter-branch or intra-office voice
communication.
For dialing a number use ‘*’ in-place of ‘.’
How it works
Let us understand how to use Peer-to-Peer Calling with the following illustration:
• Two offices are directly connected over the IP network.
• SARVAM UCS is installed at Location A.
• The System at Location B may also be SARVAM UCS.
• Peer-to-Peer calls can be made between the two locations with suitable configuration of SARVAM UCS
and the System.
• At Location A, you need to do the following configuration in the SARVAM UCS:
• Select a SIP Trunk to be used for this application and enable it. For example, SIP1.
• Set the SIP Trunk Mode of this trunk to Peer-to-Peer.
• Keep the SIP ID field of the SIP trunk blank.
• Set the Treat Incoming call as option on the SIP trunk to Station.
For detailed instruction, see “Configuring SIP Trunks”.

In the Router, you must configure the same SIP and RTP Ports as configured in the SARVAM UCS. In
other words, you must configure Port Forwarding for SIP and RTP on the Router.
• You must also configure the Trusted IP Address/es table for this SIP Trunk to receive incoming calls.
If you do not configure this table, all incoming calls on this SIP Trunk will be rejected. For instructions,
see “Configuring SIP Trunks”.
• Set the Send CLI Option on the SIP trunk as Calling Party Wise. See “Configuring SIP Trunks” for
instructions.
• You may also configure the Closed User Group (CUG) Table to avoid dialing the Trunk Access Code
for the outgoing calls made from this SIP Trunk, i.e. SIP 1.
At Location A, you may configure the CUG table as follows:
• In Route Code, enter the extension numbers of the System at Location B. Instead of entire number
strings, you can configure a single digit, the starting digit of the extension numbers as Route Code.
In this case, you may configure Route Code as ‘3’, as all extensions at Location B start with ‘3’.
• Keep the Strip Digit Count as ‘0’.
• Keep the Self Route check box disabled.
• In Dialed Digit Count, enter the digit length of the extension numbers at Location B. In this case,
‘4’.
• In OG Trunk Bundle Group, select 01. Configure SIP Trunk1 as the only member in this group. The
calls will be routed through this SIP Trunk only.
• When Self Route check box is disabled, system will check Apply Toll Control parameter. By
default, it is enabled. The system will apply toll control to all the outgoing calls.
Disable this check box, if you do not want to apply toll control to the CUG numbers dialed by you.
• By default, Apply Call Cost check box is enabled. Select this check box for Peer to Peer calls if you
require the call cost calculation.

For detailed instructions, see “Closed User Group (CUG)”.
If the System’s at both the offices have the same extension numbers, the users need to dial the Exchange
ID of the System along with the extension numbers. For detailed instructions, see “Closed User Group-
With Exchange ID”.
• Now, configure the Peer-to-Peer Table.
The Peer-to-Peer table stores up to 500 entries. Each entry consists of the parameters Number,
Domain Address, Name and Default Transport for Outgoing Message.
At location A, you would have to configure the Peer-to-Peer table as follows:
• In Number, enter the digit you configured as Route Code in the CUG Table for calling the
extension numbers of the System at Location B. In this case ‘3’.
• In Domain Address, enter the IP Address of the WAN Port of the Router at Location B.
• In Name, enter ‘Location B’ for identification.
• Keep Default Transport for Outgoing Message as ‘UDP’.
• At Location B, you may do a suitable configuration of the System.
• When an extension user 2001 of SARVAM UCS at Location A dials 3001, an extension number of System
B, it checks the CUG table to match the dialed digits with the Route Code and the Dialed Digit Count. As a
match is found, it selects the SIP trunk defined for routing the Route Code, i.e. SIP 1.
• As SIP 1 is set to Peer-to-Peer mode, the system checks the Peer-to-Peer Table configured. It finds a
match for the digit ‘3’ and will route the call to the IP Address configured for this number. In this case, to the
IP Address of the Router at Location B (121.124.130.110).
• Further, the Router will forward the call to System B. With suitable configuration done in System B, the call
will be routed to the desired destination i.e. extension 3001.

Configuring the Peer-to-Peer table using Jeeves
Peer to Peer Table is used, for deciding the destination IP Address for routing calls using non-proxy SIP trunks
(Registrar server address shall be programmed as blank for these trunks). The Peer-to-Peer table stores upto 500
entries. Both IPv4 nd IPv6 addresses are supported. This table is common for all the SIP trunks.
For instructions on configuring the SIP Trunk for the Peer-to-Peer application—SIP Trunk Mode, SIP ID, Treat
incoming calls as Station, Trusted IP Address/es table—see “Configuring SIP Trunks”.
For instructions on configuring the CUG Table, see “Closed User Group (CUG)”.
To configure the Peer-to-Peer table using Jeeves,

.
• Under VoIP Configuration, click Peer-to-Peer Table. The Peer-to-Peer table opens.
The first entry is reserved for No Match Found.
• In Number, enter the peer-to-peer number string—prefix or entire number—that will be dialed. The
number string must not exceed 8 digits. Default: Blank.
• In Domain Address, enter the domain name or IP Address to where the call is to be placed. Both IPv4
nd IPv6 addresses are supported. The Domain Address may consist of up to 48 characters
The Destination Address can also be in the form of Address: Port number.
• In Name, enter a name to identify the number string you configured. It may be the name of your contact
or any name you wish to assign to the number string. The name may consist of 24 characters
The name you configure here will not be used in SIP signaling.
• In the Default for Outgoing Message field, select the option for transporting outgoing SIP messages.
You can select UDP, TCP or TLS.
The parameter Default for Outgoing Message can be configured through Jeeves only.

Configure using Telephone
Enter the SE mode.
For programming parameters using SLT/DKP use ‘#9’ in-place of ‘.’
For dialing a number use ‘*’ in-place of ‘.’
To configure a Number, dial:

• 7801-1-Index-Number String-#* to configure the number for a single index.
• 7801-2-Index-Index-Number String-#* to configure the numbers for a range of indexes.
• 7801-*-Number String-#* to configure the numbers for all the indexes.
Where,
Index is from 001-500.
Number String can be a complete telephone number, truncated telephone number or an area code.
Number string is of maximum 8 digits with (0-9, #. *).
By default, Blank for all indices.
To clear a Number, dial:

• 7801-1-Index-#* to clear the number for a single index.
• 7801-2-Index-Index-#* to clear the numbers for a range of indexes.
• 7801-*-#* to clear the numbers for all the indexes.
To configure the Destination Address, dial:

• 7802-1-Index-Destination Address-#* to configure the Destination Address for a single index.
• 7802-2-Index-Index-Destination Address-#* to configure the Destination Address for a range of indexes.
• 7802-*-Destination Address-#* to configure the Destination Address for all the indexes.
Where,
Destination Address is of maximum 40 char. with ASCII.
IPv6 address can be configured using Jeeves only.
To clear the destination address, dial:

• 7802-1-Index-#* to clear the Destination Address for a single index.
• 7802-2-Index-Index-#* to clear the Destination Address for a range of indexes.
• 7802-*-#* to clear the Destination Address for all the indexes.
To configure the Name, dial:

• 7803-1-Index-Name-#* to configure the Name for a single index.
• 7803-2-Index-Index-Name-#* to configure the Name for a range of indexes.
• 7803-*-Name-#* to configure the Name for all the indexes.
Where,
Name is a string of alphanumeric characters of maximum 24 characters.
Use first 25 keys and the redial key to enter the name. Terminate with #*.
Display Name field is not sent in the SIP message. It is just a tag for the entry.

To clear the Name, dial:
• 7803-1-Index-#* to clear the Name for a single index.
• 7803-2-Index-Index-#* to clear the Name for a range of indexes.
• 7803-*-#* to clear the Name for all the indexes.

PIN Dialing
PIN is a unique four digit code with an associated Class of Service and Toll Control, which can be assigned to the
extension user.
PIN Dialing allows an extension user to make outgoing calls from any extension according to the toll control
assigned to his/her PIN. PIN Dialing must be enabled in the Class of Service of the extension from where the
outgoing call using PIN is to be made. SARVAM UCS supports maximum 500 PINs for each type of toll control.
Calls made using PIN can be logged in the SARVAM UCS and you can print the report online or later as and when
required.
How it works
Let us understand how this feature works with the help of an example.
User A is denied dialing of International Numbers from his/her extension but has PIN Dialing enabled in his/her
CoS. A is assigned a PIN 1234 with toll control level as international call. Now, User A can make international calls
in two ways:
1. User A must dial the feature access code to access PIN Dialing i.e. *2 (programmable, see “Access
Codes”) followed by his/her PIN. As soon as SARVAM UCS detects a valid feature access code and PIN,
it gives trunk dial tone to A. Now user A can dial the desired international number and talk.
2. User A can directly dial an international number from any extension that is assigned a “Station Basic
Feature Template” with PIN Dialing enabled in the Class of Service and LCR enabled in the OGTB group.
SARVAM UCS will prompt A to dial his/her PIN either by playing feature tone or a voice message. Voice
message will be played, if recorded. See “Voice Message Applications” for more details. As soon as s/he
dials a valid PIN, SARVAM UCS will dial out the number and connect A to the called party.
This method of calling is useful while making calls through Open SIP Phones, as it prevents the PIN from
being stored in the call logs of the phone.
• You cannot Redial or set Auto Redial for the numbers dialed out using PIN.
• While dialing the Access Code and PIN from the desired extension, the Extension - Inter Digit Wait
Timer (sec) will be applicable. See “System Timers and Counts” for more details.
How to configure
To be able to use this feature,
• Make sure, PIN Dialing is enabled in the “Class of Service (COS)” of the Extensions from where you want
to make outgoing calls using PIN.
• Enable the parameter Store Outgoing Calls in the “Station Basic Feature Template” assigned to the
extensions from where you want to make outgoing calls using PIN.
• Assign LCR enabled OGTB Group in the “Station Basic Feature Template” of the Extensions from where
you want to make outgoing calls using PIN.

• Configure the PIN table for each type of Toll Control level from the SA Mode. See “Configuring PIN Table”.
• Select the OGTB Group and type of LCR for making outgoing calls using PINs, for each Toll Control Level
from the SE Mode. See “Configuring OGTB Group and LCR”.
Configuring PIN Table

• Make a list of users to whom you want to assign PINs.
• Click PIN Configuration.
• Select the desired Toll Control level. Each Toll Control level includes the Allowed and Denied numbers as
configured by SE. See “Toll Control” for more details.
• Configure the PINs at a desired index’s. The PIN must be of 4 digits. Valid digits are 0 to 9.
• Each PIN must be unique. You cannot assign same PIN to two different index within the same Toll
Control level.
• If you re-assign the same PIN to a different Toll Control Level, it will not be valid for the previously
assigned Toll Control Level.
• To avoid unauthorized access, we recommend you to change the PIN regularly. Make sure it is strong
and is kept confidential.
Configuring OGTB Group and LCR


• Under Configuration, click PIN Configuration.
• In OG Trunk Groups for PIN Dialling, enter the desired OGTB Group number for each Toll Control level.
• To select the desired trunks in the OGTB Group and the type of LCR, see “OG Trunk Bundle Group” for
instructions.
• To configure the LCR Table, see “Configuring LCR” for instructions.
How to use
To use PIN Dialing from any extension:
• Go OFF-Hook.
• Press DSS Key assigned to ‘PIN Dialing’.
OR
• Dial *2
• Dial the PIN.
• You will hear the trunk dial tone.
OR
• Go OFF-Hook.
• Dial TAC.
• You get Feature tone/ Voice message.
• Dial the PIN.
• Speech with the Called Party.

For SLT Users
• Go OFF-Hook.
• Dial *2
• Dial the PIN.
• You will hear the trunk dial tone.
OR
• Go OFF-Hook.
• Dial TAC.
• You get Feature tone/ Voice message.
• Dial the PIN.
• Speech with the Called Party.
Printing Reports of Outgoing Calls made using PIN

You can print call details of users who made outgoing calls using the PIN. For this, you will need to:
• enable Store Outgoing Calls in the Station Basic Feature Template of the extension from where you want
to make outgoing calls using PIN.
• enable Print Calls made using PIN and set the PIN range in Calls made using PIN under Calls filter in
Outgoing Call Report.
• configure the destination port for SMDR-Outgoing Call Report.
Refer “Station Message Detail Recording-Report”, for detailed instructions on printing reports using filters.
You may also print online report of calls made using PIN, when the SARVAM UCS is interfaced with a third party
Call Accounting Software (CAS). For this, you must set the parameter PIN in SMDR - Posting. See “Station
Message Detail Recording-Posting” for more details.

PLCC-An Introduction
What’s this?
SARVAM UCS-PLCC Server is a digital System. It uses a digital switch and hence is a 100% non-blocking system.
In PLCC network, number of PLCC Servers need to be connected.
Refer to diagram showing cluster of exchanges in a PLCC network. Each exchange in PLCC network is assigned
with Exchange Identity (SID) in order to get identified by other exchanges in the network. Thus each exchange is
identified by Exchange Identity (SID). As shown, an exchange is connected to the other exchange through an E&M
tie line. Also, it is possible that an exchange may not be directly connected to all other exchanges in a PLCC
network through E&M tie lines.
For example, SID-52 exchange is connected to SID-51, SID-34, SID-71 and SID-60 exchanges through direct E&M
tie lines. However, SID-52 exchange is not directly connected to SID-61, SID-62 and SID-72 exchanges through
E&M tie lines.
Following example shows how a call is established.
• Consider, an example where subscriber 20 of SID-52 needs to call subscriber 30 of SID-51. After dialing
trunk access code, dial 51 (SID number of the exchange) and then dial 30 (subscriber number of SID-51).
This is how a call is completed, as both exchanges; SID-51 and 52 are connected through a direct E&M tie
line.
• However, consider another example, where subscriber 20 of SID-52 needs to call subscriber 10 of SID-61.
Even though, here both exchanges are not connected through direct E&M tie line, it is possible by dialing
trunk access code with SID number of the exchange (here 61) followed by subscriber number (here 10).
However, the call will proceed through SID-60 and then reach SID-61. This is called Transit Call. Here, the
subscriber will not be able to know that call has proceeded to the required exchange through transit facility.
PLCC Network
E&M Tie Line SID-62
Sub. 10
SID-61 Sub. 11
Sub. 12
SID-34 SID-60
Sub. 20
SID-52 Sub. 21
Sub. 22
SID-51 SID-71
SID-72
Sub. 30 Sub. 31

E1 E2 En
E&M1 E&M2 E&M3
PLCC-System
(SID-70)
E&Mn
PLCC-System
E&M1
(SID-34)
E&M2
E&M1 E1 E1
PLCC-System
(SID-52)
E&M2 E2 E2
E&M3
E&Mn En En
E&Mn
E&M1 E&M2 E&M3 E&Mn
E1 E2 En
PLCC-Priority
All callers do not have same hierarchical position in an organization. ‘Priority’ can be assigned to each caller to
make sure precedence is given to certain extensions over others in being answered.
The SARVAM UCS supports flexible priority assignment for different users. Each port can be assigned a priority
level between “1” to “9”. Higher the priority level, more important the caller is. Accordingly 1 has the least priority
and 9 has the highest priority.
Using this feature the users can free the system resources (Trunk/Extension).
The PLCC functional module requires a license. You must purchase a license to activate CCS Signaling
when End Point and Transit, Express Signaling, and Seizure Pulse and Release Pulse Signaling. Refer the
topic “License Management”.

How it works
Using Priority on a Station:
• Station A and Station B are talking to each other.
• Station C calls Station B and finds it to be busy. If s/he uses Priority, s/he gets connected to Station B.
• Station A gets disconnected and gets error tone. However, for this to happen the priority of Station C
should be higher than that of Station B and Station A.
Using Priority on a Trunk:
• Station A is talking to an external party through trunk 1.
• Station B tries to grab trunk 1. As the trunk is busy, s/he uses Priority. Doing so, s/he gets connected to
trunk 1 and gets PSNT dial tone.
• Station A gets disconnected and gets error tone. However, for this to happen priority of Station B should be
higher than Station A. In this case, priority of the trunk is not considered.
• Similarly, if station A is talking to an external party through trunk 1. Station B calls station A and finds it
busy. S/he uses Priority. Doing so, station B gets dial tone whereas, the trunk 1 gets disconnected.
How to use
• Lift the handset. You hear the Dial tone.
• Dial Station/Trunk Access Code. You get Busy tone.
• Dial #*. The called station/trunk gets disconnected. You get dial tone after the confirmation tone.
• Dial Station/External Number. You get Ring Back tone.
Priority-Within the Exchange

Under this facility any subscriber having priority access can enter in to conversation with two conversing
subscribers within the exchange.
E&M tie line E&M tie line

SID-61 SID-52 SID-34
In conversation Dials. Gets busy tone.

Uses priority by dialling
'#*'
Sub. 20 Sub. 21 Sub. 22

(Without priority access) (Without priority access) (With priority access)

How to use
Please refer above diagram. If any two subscribers, say 20 and 21 of an exchange with SID-52 (without priority
access) are in conversation and a third subscriber, say 22 (of same exchange) has priority access and wants to talk
with subscriber 21. But, when subscriber 22 dials for subscriber 21, it gets busy tone. Subscriber 22 can use its
priority by dialing code ‘#*’ in such case and can come in conference with subscribers 20 and 21.
Priority-In Other Exchange

Under this facility any subscriber having priority access can enter in to conversation with two conversing
subscribers in two different exchange.
E&M tie line E&M tie line

SID-61 SID-52 SID-34
Dials. Gets busy tone.

Uses priority by dialling In conversation
'# *'
Sub. 10 Sub. 20 Sub. 21

(With priority access) (Without priority access) (Without priority access)
How to use
Please refer above diagram. If any two subscribers, say 20 and 21 of an exchange with SID-52 are in conversation
and a subscriber, say 10 of an exchange with SID-61 (from the same network with priority access) want to talk with
subscriber 20 of SID-52. After dialing the required subscriber 20 of SID-52, he gets busy tone. Subscriber 10 of
SID-61 can now press ‘#*’ and can enter into conference with both subscriber 20 and 21 of SID-52.
Priority-Over Busy E&M Tie Line

Under this facility any subscriber having priority access can use priority to free a busy E&M tie line.
E&M tie line

SID-61 SID-52
Dials. Gets busy tone.

Uses priority by dialling
'#*'
In conversation
Sub. 10 Sub. 11 Sub. 20 Sub. 21

(With priority access) (Without priority access) (Without priority access)

How to use
Please refer diagram, if two subscribers, say 20 of SID-52 (without priority access) and 11 of SID-61 (without
priority access) are in conversation on an E&M tie line. Both the exchanges SID-52 and SID-61 are connected with
each other through a single E&M tie line. Now, if subscriber, say 10 (with priority access) of SID-61 wants to talk
with subscriber 21 of SID-52, then he will get busy tone. Subscriber 10 of SID-61 can now use its priority by
pressing ‘#*’ on his telephone and can get into conference with both subscriber 11 of SID-61 and subscriber 20 of
SID-52. After termination of the call by subscribers 11 and 20, subscriber 10 gets dial tone and can make a call
using the E&M tie line to subscriber 21 of SID-52.
PLCC-Routing Table
In PLCC network, number of PLCC-Servers need to be connected. The entire network should behave as a single
unit or one group. It is not feasible to have unique station numbers throughout the network. In such cases, an
Exchange ID is assigned to the Servers and a routing table is programmed in the exchange. In case of newly added
exchange in the network, the routing tables of other exchanges need to be modified.
In the above figure, 3 systems are connected through E&M connectivity.

• S1 to Sn are stations.
• E&M1 to E&M3 are E&M lines between the three Servers.
How it works
In this application, it is possible to have same station number in two or more Servers of the network. Few new
words have been used to explain PLCC routing table, each of these words have been explained below:
OG Call with Trunk Access Code:

In PLCC applications, SLT numbers are duplicated. Internal and External dialing is differentiated. Internal dialing
happens by dialing SLT numbers directly. External dialing starts with dialing the trunk access code (usually ‘0’)
followed by SID and remote SLT number. Suppose the exchange ID of System-A is 21 and that of System-B is 22.
Both Systems have same station numbers, say 2001,2002, etc. Suppose a station of System-A wants to call 2001

of System-B. He has two options. He can dial 222001 or 0222001 i.e one with route code 22 and other with route
code 022. Hence we need to program two entries in the routing table, one with route code 22 and other with route
code 022.
The SARVAM UCS has only one routing table. The same table is used for Closed User Group and Closed
User Group-With Exchange ID. Hence the table has to be programmed keeping the application in mind.
• Routing Table: This table has following parameters viz. Route Index, Route Code, OG Trunk Bundle
Group, Strip Digit Count, Self Route check box, Apply Toll Control check box and Apply Call Cost check
box. The Closed User Group-With Exchange ID programming works according to this table.
Route Strip Digit Dialed Digit Apply Toll Apply Call

Index OGTBG Self Route
Code Count Count Control Cost
001
250
P is Pause, F is Flash, A to D is DTMF Digits. Generally, route code will be a unique number. The route
code should not clash with any of the extension numbers of same System. For example in the figure given
above, route code for System-A can be defined as ‘21’, route code for System-B can be defined as ‘22’
and that for System-C can be defined as ‘23’. This means that no extension in System-A can start with ‘22’
or ‘23’. Similarly, no extension in System-B can start with ‘21’ or ‘23’ and no extension in System-C start
with ‘21’ and ‘22’.
number is dialed on it. The same logic of rotation On/Off for trunk selection from the OGTBG is used. If
group is selected. This helps to select an alternate route. Whereas if rotation is ON then the trunks in the
• Strip Digit Count: This count signifies the number of digits to be stripped off while dialing/decoding a
number. To elaborate: Consider figure 1. The requirement is that if extension 2001 of System-B dials
212002 and if E&M 1 is busy then the call should reach extension 2002 of System-A through alternate
route. In this case the strip digit count of System-A should be programmed as 2 and that of System-B and
System-C should be programmed as 0. Doing so, when extension 2001 of System-B dials 212002 and if
E&M1 is busy then the call is routed through System-C. In this case, System-B dials 212002 on E&M3,
System-C receive this code and dials out the same code, that is, 212002 on E&M2 without striping of any
digit. On receiving 212002, System-A strips of two digits as per the programming and routes the call to
extension 2002.
• Self-Route: This flag signifies that the digits being dialed are for the same System and are not to be dialed
on the E&M trunk.

• Apply Toll Control: This parameter is not relevant as Self Route flag is enabled. This parameter is
relevant when you are configuring “Closed User Group (CUG)”.
• Apply Call Cost: By default, this check box is enabled and the system will calculate the cost of each call.
How to configure
For more details on above steps, please refer topic “Closed User Group (CUG)” and “Closed User Group-With
Exchange ID”.
PLCC-Express Line Network

SARVAM UCS-PLCC Express Line Communication Server is a digital system. It uses a digital switch and hence is
a 100% non-blocking system. Refer to diagram in ‘Introduction to PLCC Network’ topic showing cluster of
substations in a PLCC network. A substation in a PLCC network is connected to any other substation through a
dedicated PLCC express line (that is, 4/6 wire E&M tie line) as shown in the diagram.
How to configure
PLCC Express Line Communication Server is one of the applications of the SARVAM UCS. We recommend the
user to read relevant topics before programming it for PLCC Express Line application.
Step 1
Refer chapter “E&M Feature Template” to program the feature in an E&M Feature Template.
Step 2
Refer chapter “E&M Feature Template” to assign default values to an E&M Feature Template.
Step 3
Refer chapter “E&M Feature Template” to assign an E&M Feature Template to an E&M.
Step 4
Refer chapter “Configuring DKP Extensions” to program a name for E&M trunk.
Step 5
Hardware ID is an attribute of a software port. Hardware ID of a software port decides where the port is physically
located. To derive hardware ID of a software port, we need slot number and port number of the card. Hence, all the
programming is done for the software port and not for the hardware ID. Accordingly, the software port number is
used for all the programming. Please refer “Software Port and Hardware ID” for more details, to assign hardware ID
to an E&M software port:
Step 6
Refer chapter “Station Advanced Feature Template” to assign a Station Advanced Feature Template to an E&M.

Step 7
Refer chapter “Station Advanced Feature Template” to assign a Station Basic Feature Template to an E&M.
Step 8
Please refer “Time Tables” chapter to program the time zone of a trunk.
Step 9
To program a feature in a Trunk Feature Template, please refer “Trunk Feature Template” for more details.
Step 10
To assign a Trunk Feature Template to an E&M, please refer “Configuring the E&M using Jeeves” topic.
Step 11
Refer chapter “DSS Keys Programming” to assign a function to a key of DSS-64 assigned to a DKP. In this
commands function type and function number can be programmed as explained below.
Function:
A function should be assigned to these Keys, which they should perform. There are 2 different types of functions,
which can be assigned to a key for PLCC express line application. Each function that can be assigned to a key is
given unique number. Following table list the function types available:
Function Type Meaning Function Number/Port
00 Null --
06 Access a trunk 001 to 128
Step 12
Refer chapter “DSS Keys Programming” to assign a function to a DKP key. In this command refer above
explanation to program ‘Function Type’.
Step 13
Define dialing property of E&M port. The dialing properties are based on the applications. If the dialing type for a
E&M is programmed as pulse type then Pulse-Dialing ratio should be defined. The E&M support six different Pulse
Dialing Ratio’s.
Value Meaning/Ratio
1 10PPS, 1:2
2 10PPS, 2:3
3 10PPS, 1:1
4 20PPS, 1:2
5 20PPS, 2:3
6 20PPS, 1:1
For assigning Dial Type and Pulse Dial Ratio, please refer “E&M Feature Template” topic.

Presence
What's this?
Presence is an important UC feature as it helps you to know the availability status of other users. Depending on the
status of users you can decide whether to initiate a conversation or find an alternative way to contact the desired
user.
For example, an extension user may want to leave his desk for an indefinite period, but does not want to use Call
Forward or set Do Not Disturb. He wants to indicate to callers about his absence. Similarly, extension users who
are present at their desk may want to hide their presence from other users; or they may want to show their current
activity to the other extension users like they are Busy, or are away from their desks, or on the phone with someone
on another call, etc.
With the Presence feature of SARVAM UCS, extension users, including the Operator, can 'publish' their presence
to callers from other extensions. By doing so, they can indicate to the other extensions about their availability.
In the same way, the Presence feature allows extension users to view the 'Presence' status (availability) of the
extensions that they want to call, before making the call or when their call is not answered.
How it works
Publishing Presence
Any SLT, DKP, ISDN Terminal, and SIP Extension User can 'publish' their presence by setting any of the messages
listed in the following on their phone, by dialing the access code for this feature.
SIP Extension users who want to publish their presence have two options:
• Using the PUBLISH feature supported by the SIP Client.
• Using the feature access code for Publish Presence supported by SARVAM UCS.
The first option requires the parameter 'PUBLISH' to be enabled in the SIP Extension Settings. Refer
“Configuring SIP Extensions”. By default, this parameter is disabled.
Publishing Presence Messages

1. Absent: When an extension user sets 'Absent' as the message, all incoming internal as well as external
calls will be blocked from landing on his/her extension.
When any other DKP/SIP extension user calls this extension, the text message 'User Absent' will appear
on the caller's phone display.
If the extension phone that has set 'Absent' is a DKP/SIP, the letter 'A' appears on the phone's display to
indicate absence.
The letter 'A' disappears when the extension user sets a presence message other than 'Absent'.

• When an SLT extension user calls the extension which has set 'Absent', an error tone will be played.
However, it is possible for the SLT extension user to find out the presence status of the called
extension. Refer “Viewing Presence” later in this topic.
• External callers who call the extension, on which 'Absent' is set, will get an error tone only.
• Outgoing calls can be made from the extension which has set 'Absent'. Only incoming calls are
restricted.
• If more than one extension is configured as "Operator" (routing group), incoming calls will be blocked
only on the Operator extension which has set User Absent.
2. Present:
When an extension user sets 'Present', all incoming calls will be received as normal on this extension.
If previously set as 'Absent', when a DKP/SIP extension user sets 'Present' the letter 'A' will disappear from
the phone's display.
When any other DKP/SIP extension user calls this extension, the name of the extension user will be
displayed on the caller's phone display, when the called extension is ringing.
3. Auto Detect: When an extension user sets 'Auto Detect', the system will detect the state of the phone;
depending on the call state, it will publish the presence message to the other extensions. Three types
Publish Presence messages are possible, with Auto Detect:
a. Idle: When the system detects the extension phone to be ON-Hook, it indicates the status of the phone
to other extensions 'idle'.
b. On the Phone: When the system detects the phone to be OFF-Hook, or in speech with another party
or if it detects an incoming call placed on the phone, it will indicate to the other extensions that this
extension user is 'On the Phone' with another party.
c. DND Text message: When the system detects that the extension phone has Do Not Disturb (DND) set
on it with a DND Text message, it will display to the calling extension, the DND message set by the
called party (this may be the default DND message or the DND Text message set by the called
extension).
3. Away: When an extension user sets 'Away', the system will display this message to the other extensions.
4. On the Phone: When an extension user sets 'On the Phone', the system will display this message to the
other extensions.
5. Do Not Disturb: The extension user can set this message to be published to other extensions, if s/he
wants to work uninterrupted.
Unlike the DND Feature, the extension user who has set this message will continue to receive calls both
internal as well as external calls, as the system considers this extension as 'present'.
6. I am Mobile: The extension user can set this message to be displayed to other extensions, when s/he is
not at the desk.

7. In Meeting: The extension user can set this message to be displayed to the callers, if s/he is busy in a
discussion or meeting.
8. Out for Meal: The extension user can set this message to be displayed to other extensions when going on
a lunch break.
9. Out of Office: The extension user can set this message to be displayed to the callers when s/he leaves
the office temporarily.
When an extension user sets any Publish Presence message other than 'Absent', the system will consider
the user as 'Present'. All incoming and outgoing calls will be allowed on this extension.
It is possible to program another message in place of Publish Presence messages listed from 6 to 9: I am
Mobile, In a Meeting, Out for Meal, Out of Office.
Publish Presence messages can be set or changed for any extension from the System Administrator (SA)
mode.
Viewing Presence
• Extension users can know the status of another extension user before calling or when the extension user
does not answer the call.
• Generally, when DKP extension users call another extension, the name of the called extension is
displayed on the calling DKP extension. Now, if the flag 'Display User Status during Call' is enabled in the
System Parameters, when DKP extension users call another extension, the calling DKP extensions will be
displayed the 'presence' status message published by the called extension269.
• SLT extension users, whose phone is equipped with a CLI display, can see the status of another extension
by dialing a feature access code, then going ON-Hook. The system will ring back the SLT and send the
Presence status of the desired extension as CLI.
• SIP extension users can use the Presence feature of SARVAM UCS to view the presence status of other
extensions. For this, they must dial the feature access code and the number of the desired extension.
• SIP extension users who want to view the status of other extensions using the feature supported by their
SIP Client, must have 'Presence Subscription' enabled in their SIP Extension Settings. Refer “Configuring
SIP Extensions”.
How to configure
This feature involves the programming of the following parameters:
• 'Display User Status during Call' flag: DKP extension users will be able to view the presence status for
the called extension only if this flag is enabled in the System Parameters.
• PUBLISH: SIP extension users who want to publish their presence using the feature supported by their
SIP client will be able to publish their presence status only if this feature is enabled in their SIP Extension
Settings. This parameter is not necessary, if they want to publish presence using the feature of SARVAM
UCS.
269. DKP users can also dial a feature access code and the number of the extension to see the status of that extension on their DKP.
But this would not be required, if the 'Display User Status during Call' flag is enabled in the System Parameters.

• Presence Subscription: SIP extension users who want to view the presence of other extensions using
the feature supported by their SIP client must have this feature enabled in their SIP Extension Settings.
This parameter is not necessary, if they want to view presence using the feature of SARVAM UCS.
• Publish Messages: It is possible to customize the Publish Messages listed above from 6 to 9 viz.: 'I am
Mobile', 'In Meeting', 'Out for Meal', 'Out of Office'.
The above parameters, with the exception of 'Publish Messages', can be programmed using both Jeeves and
a Telephone. You can program 'Publish Messages' using Jeeves only.
Programming Presence using Jeeves

• Go to Display Presence Status during Call on DKP. Click to enable the flag.

• Click Publish Message to expand.
• You can change message number 6 to 9 as desired. The string may consist of a maximum of 16
characters. All ASCII characters except < > and “ (double quote) are allowed.
• Click SIP Extension Settings to open the page.
• Now, go to the desired SIP Extension number for which you want to enable the features PUBLISH and
Presence Subscription. By default both features are disabled. Click the respective check boxes to enable
the features.
Programming Presence using a Telephone

To enable/disable 'Display Presence Status during Call on DKP', dial:

• 5320-Flag
Where,
Flag is
0 for Disable
1 for Enable
By default, flag is disabled.
• Exit SE mode.

How to use
This feature requires you to dial your User Password. The default User Password 1111 is not accepted. Please
change the User Password first.
Publish Presence can be set for an extension also from the System Administrator mode.
Publishing Presence by Extension User

• Press DSS Key assigned to PUBLISH presence.
OR
• Dial 104
• Enter User Password on the prompt.
• Scroll to the desired Publish message from the menu:
• Absent
• Present
• Auto Detect
• Away
• On the Phone
• Do Not Disturb
• I am Mobile
• In Meeting
• Out for Meal
• Out of Office
• Press Enter key to select message.
• You get the confirmatory tone.
Publishing Presence from SA Mode

• Press DSS Key assigned to PUBLISH presence.
OR
• Dial 1072-014
• Enter Destination Number, that is, the number of the extension Publish Presence is to be set.
• Scroll to the desired Publish message from the menu:
• Absent
• Present
• Auto Detect
• Away
• On the Phone
• Do Not Disturb
• I am Mobile
• In Meeting
• Out for Meal
• Out of Office
• Press Enter key to select message.
• You get the confirmatory tone.

To view Presence Status
• Press DSS Key assigned to Display Presence Status.
OR
• Dial 1097.
• Enter Extension number
• The status of the extension number you dialed will be displayed on your phone's LCD.
• Go ON-Hook.
For SLT Users
Publishing Presence by Extension User

• Dial 104-Password-Index Number
Index No. Meaning
0 Absent
1 Present
2 Auto Detect
3 Away
4 On the Phone
5 Do Not Disturb
6 I am Mobile
7 In Meeting
8 Out for Meal
9 Out of Office
Publishing Presence from SA Mode

• Dial 1072-014-Extension Number-Index Number
Index No. Meaning
0 Absent
1 Present
2 Auto Detect
3 Away
4 On the Phone
5 Do Not Disturb
6 I am Mobile

Index No. Meaning
7 In Meeting
8 Out for Meal
9 Out of Office
To view Presence Status

You can view Presence Status of another extension only if your SLT has a CLI display.
• Lift handset.
• Dial 1097-Extension Number.
• Go ON-Hook during confirmation tone.
• Your phone will ring and the status of the extension number you dialed will be displayed on your phone as
CLI.

Preset Call Forward
What's this?
SARVAM UCS supports the Preset Call Forward. This feature is useful when Call Forward is not set by users, as
their calls will automatically be forwarded to the selected destination. This feature is independent of the Class of
Service assigned to the extension users.
Preset Call Forward options can be configured for each time zone by the SE only. The calls will be forwarded to the
selected destination— Voicemail, Extension (SLT, DKP, SIP) or Department Group as per the Preset Call Forward
type selected.
If users set Call Forward from their extensions, it will have a priority over Preset Call Forward. When the users
cancel Call Forward from their extensions, the Preset Call Forward option will be applicable automatically.
The Preset Call Forward feature of SARVAM UCS offers the following forwarding options:
• When Busy - calls are forwarded to the destination phone number only when the called party's phone is
busy.
• When No Reply - calls are forwarded to the destination phone number only when the called party does not
answer the phone. The default time is 30 seconds for all extensions and can be changed by programming
the Call Forward No-Reply Timer.
• When Busy or No Reply - calls are forwarded to the destination phone number when the called party's
phone is either busy or does not reply.
How it works
A has set Preset Call Forward When No Reply to the Voicemail.
• The system waits for the Call Forward No-Reply Timer to expire and forwards all incoming calls to A’s
Voicemail.
A has set Preset Call Forward When Busy to B’s extension.
• The system forwards the call for A to B on detecting Busy signal from A.
B has set Preset Call Forward-No Reply on A and A belongs to a Department Group.
• If the Call Forward-No ReplyTimer set is less than the Ringer Timer of the Department Group, the Preset
Call Forward request will be served and the call will land on A.
• If the Call Forward-No Reply Timer set is greater than the Ringer Timer of the Department Group, the
Preset Call Forward request will not be served.
If the Ignore call forward set by member extension, when call is routed on Routing/Dept. Group option is
enabled in System Parameters, then Preset Call Forward request will not be served. See “System
Parameters” for more information.

A has set Preset Call Forward to Department Group.
• The system forwards the call for A to the Department Group. The free member in the group answers the
call.
A has set Preset Call Forward When Busy or No Reply to the Voicemail.
• Whenever there is a call for A, if the system does not detect a busy signal from A, it waits for the Call
Forward No-Reply timer to expire.
• The system forwards the call to the Voice Mail System.
If A wants to change the Call Forward destination temporarily, then A must set Call Forward from his/her extension.
For detailed instructions, see “Call Forward”. In this case, the Preset option will not be applicable. But as soon as A
cancels Call Forward from his/her extension, the Preset option will be applicable.
• Preset Call Forward cannot be cancelled by the users.
• The system supports only single-point Preset Call Forward, which means, if the destination extension
has also forwarded its calls, the call will not follow the forwarding path. For example: Calls for extension
A are forwarded to extension B. Preset Call Forward is also set on extension B with C as the
destination number. In this case, Calls for A will land on B and calls for B will land on C. Calls for A will
not land on C.
• Only one Preset Call Forward Type can be set for each Time Zone. Every new Preset Call Forward
Type set overrides the previous one.
How to configure
The functioning of this feature is controlled by the following parameters: Preset Call Forward configured in the
'Station Advanced Feature Template' assigned to the extension user and 'Call Forward No-Reply Timer'.
When Preset Call Forward No-Reply is set, if required the Call Forward No-Reply Timer needs to be programmed.
Configuring Preset Call Forward in the Station Advanced Feature Template

The default Station Advanced Feature Template 01 is assigned to all extension users of the SARVAM UCS. In this
template Preset Call Forward is disabled.
Decide which extensions are to be allowed 'Preset Call Forward'. If you want to allow Preset Call Forward to all
extensions, retain the default Station Advanced Feature Template 01 and configure the Preset parameters.
However, if you want to allow Preset Call Forward only to selected extensions, select another Station Advanced
Feature Template and configure the Preset parameters in that template.
Now, to assign the template to selected extensions, follow these steps:
1. Prepare a Station Advanced Feature Template with Preset Call Forward configured for each time zone.
2. Assign this newly prepared template to the desired extensions.
Refer the topics “Station Advanced Feature Template” for detailed programming instructions on how to customize a
Station Advanced Feature Template, configure the Preset Call Forward parameters and how to apply this template
on extensions.

Call Forward No Reply Timer
When using Call Forward -No Reply, each extension can set a different Time after which the incoming call on the
extension should get forwarded when there is no reply from the extension. For this, the Call Forward No-Reply
Timer must be programmed. By default, this timer is set to 30 seconds.
The Call Forward No-Reply Timer is to be programmed in the “Station Advanced Feature Template” applied on the
extensions which are allowed Call Forward in their COS.
If you want to set this timer to the same duration for all extensions, simply set the Call Forward No-Reply Timer in
the default Station Advanced Feature Template 01 which is assigned to all extensions.
If you want to set different Timer duration for different extensions, then prepare separate Station Advanced Feature
Templates with the desired Timer durations and assign different Templates (with different Timer durations) to the
extensions as desired.
Refer the topic “Station Advanced Feature Template” for instructions on customizing the template and applying the
template to extensions.

Priority Calls in E&M MFCR2 Signaling
What's this?
In a network of Systems connected over E&M trunks, callers may be unable to reach the desired station, as the
called station may be busy with another station.
The problem becomes more acute when important decision makers and persons involved in critical tasks in an
organization are unable to get through to the desired extension of the networked System, because it is busy.
SARVAM UCS supports Priority Call in E&M MFCR2 Signaling to overcome this. This feature makes it possible to
land a call on a busy extension and establish speech with the desired extension user. If required, you can also
disconnect the remote party in speech with the called extension.
E&M MFCR2 Signaling is currently supported on ETERNITY GE E&M Card only. This feature is currently
available on ETERNITY GENX only.
How it works
• The extensions which are to be allowed Priority Call feature must be defined as 'Priority Subscriber' in their
Station Advanced Feature Templates.
• The extensions on which Priority Calls should be allowed to land must be 'Non-Priority Subscribers' (that
is, defined as 'Ordinary Subscriber' in the Station Advanced Feature Template).
• In the following illustration, SARVAM UCS is networked with System-A over E&M Lines with MFCR2
Signaling.
Priority Call from SARVAM UCS to Networked System
• SARVAM UCS is the Originating System and System-A is the Terminating System.
• Extension 2001 of SARVAM UCS is a 'Priority Subscriber'. Extension 2002 is an 'Ordinary Subscriber'
(Non-Priority).
• Extension 3001 of System-A is a 'Non-Priority Subscriber'. Extension 3002 of System-A is also a 'Non-
Priority Subscriber'.
• When Extension 2001 calls Extension 3001, depending upon its implementation, System-A understands it
as a priority call.

• If Extension 3001 is busy with another extension, 3002, and since both are non-priority subscribers,
System-A will automatically intrude the conversation between 3001 and 3002 and establish 3-way speech
between Extensions 2001, 3001 and 3002.
• Now, if Extension 2001 (the Priority caller) does not want Extension 3002 to be part of the conversation, s/
he can ask 3002 to go idle OR s/he can use the feature Forced Release Order to disconnect 3002 from the
conversation. 2001 can use Forced Release Order only if it is allowed to it in its “Class of Service (COS)”.
• When Extension 2001 dials the Feature Access Code for Forced Release Order, Extension 3002 will be
disconnected and two-way speech will be established between 2001 and 3001.
Priority Call from Networked System to SARVAM UCS
• System-A is the Originating System and SARVAM UCS is the Terminating System.
• Extension 3002 of System-A is a Priority Subscriber. Extension 3001 of System-A is Non-Priority

Subscriber.
• Extension 2001 and 2002 of SARVAM UCS are Non-Priority Subscribers
• Extension 3002, a Priority Subscriber, calls Extension 2002 of SARVAM UCS.
• SARVAM UCS treats the call as priority call.
• Extension 2002 is busy with Extension 2001. SARVAM UCS checks the priority of both extensions. Since
both are non-priority subscribers, SARVAM UCS gives priority to Extension 3002 and plays beeps to 2002
and 2001 before establishing 3-way speech.
• SARVAM UCS will not treat a call as priority in the following cases:
• If the busy call is itself a priority call, that is, at least one of the two parties in the busy call is a priority
subscriber.
• If any party involved in a matured call is Priority Extension or Trunk, the call will be considered as
priority call and hence no incoming priority call will be allowed to intrude this priority call.
• If the call is routed to data equipment, for example, fax machine.
• SARVAM UCS allows non-priority subscribers to intrude on a busy call using the feature Manual Priority
Intrusion. For this, the intruding extension must have the feature 'Manual Priority Intrusion' in its Class of
Service.

• When a non-priority extension requests Manual Priority Intrusion, SARVAM UCS receives an 80msec
pulse signal on the E wire of the E&M port, on detecting this signal SARVAM UCS will treat it as a Priority
Call and check whether the called extension that is busy can be intruded or not.
• If intrusion is possible SARVAM UCS creates a conference call, after playing beeps to notify both parties of
the 3-way conference call. This beep will be played only if Conference beeps are enabled.
How to configure
For this feature to work, the extensions of the networked Systems which are to be allowed Priority Call feature must
be defined as 'Priority Subscribers'.
Extensions on which Priority Calls must be allowed to land may be defined as 'Non-Priority Subscribers'.
If non-priority subscribers are to be allowed to Priority Calls when necessary, Manual Priority Intrusion feature must
be enabled in their Class of Service.
If the Priority Subscriber extension users are to be allowed to disconnect the second party (other than the desired
party) from the conversation during a Priority Call, the feature Forced Release Order must be enabled in their Class
of Service.
For Stations on which data terminals are connected (like Fax machine), it is recommended that the 'Caller
Category' of these stations be programmed as 'Data Transmission' (in their Station Advanced Feature
Template).
Programming Extensions as Priority Subscribers

To define extensions of SARVAM UCS as 'Priority Subscribers' or 'Non-Priority Subscribers', you must set the
parameter "Caller Category" in the Station Advanced Feature Template assigned to these extensions.
In the default Station Advanced Feature Template 01 assigned to all extensions of the SARVAM UCS, the Caller
Category is set to 'Ordinary Subscriber', which means all extensions are by default, 'Non-Priority Subscribers'.
Since all extensions cannot be defined as 'Priority Subscribers', prepare a new Station Advanced Feature
Template, select the option 'Priority Subscriber' as Caller Category in this template and apply it on the selected
extension that are to be defined as Priority Subscribers.
Refer the topic “Station Advanced Feature Template” for detailed instructions.
Programming Manual Priority Intrusion and Forced Release Order in Class of Service
In the default the default CoS group 01 in Station Basic Feature Template Number 01 assigned by default to all
extensions of SARVAM UCS, 'Manual Priority Intrusion' and 'Forced Release Order' are disabled.
If you want to allow all extensions these features, simply enable them in the default CoS group 01.
However, if either or both these features are to be allowed only to select extensions, follow these steps:
1. Define a CoS group with Manual Priority Intrusion/Forced Release Order enabled.

3. Assign this new Template to the extensions to which Manual Priority Intrusion/Forced Release Order is to
be allowed.
How to use
Manual Priority Intrusion

• On Busy Tone, press Transfer Key.
• Press DSS Key assigned to Manual Priority Intrusion (if programmed)
OR
• Dial *37.
• You are in 3-way speech.
For SLT Users

• On Busy Tone, press Flash to put call on Consultation Hold.
• Dial *37.
• You are in 3-way speech.
Forced Release Order

• When you are in 3-way speech,
• Press DSS Key assigned to Forced Release Order (if programmed)
OR
• Dial *38.
• You are connected the desired party. The second party is disconnected.
For SLT Users

• When you are in 3-way speech,
• Dial *38.
• You are connected the desired party. The second party is disconnected.

Priority
What’s this?
‘Priority’ is the precedence given to certain trunks and extensions over others in being answered by the destination
extension.
When ‘Priority’ is assigned to trunks, whenever there are incoming calls on multiple trunks at the same time, the call
on the trunk with highest priority will be answered by the landing destination extension/Operator first.
When Priority is assigned to Extensions, calls from extensions with highest priority will have precedence in landing
on the destination extension.
You can set priority levels from 1 to 9 as given in the table below.
Priority
Meaning
Level
1 None
2 Lowest
3 Lower
4 Low
5 Normal
6 Medium
7 High
8 Higher
9 Highest
Highest Priority can be assigned to Extensions of important or higher ranking persons in an organization; for
example, calls from senior managers or top executives in an organization can be allowed to be answered first by
the destination extension.
Highest Priority can be assigned to particular Trunks, such as special or private trunk lines, trunk lines dedicated as
help lines or emergency trunks, or trunks designated as hotlines, so that when there are incoming calls on different
trunks at the same time, the call on these trunks gets answered first by the destination extension.
Priority can be assigned to all Trunk types (CO, Mobile, SIP, T1E1PRI, BRI) and Extension types (SLT, DKP, ISDN
terminal, SIP, E&M as Station, Radio, Magneto port).

To understand how this feature works, consider this illustration:
Here,
• There are two incoming calls, one on the Analog Trunk, CO 1 and the Mobile Trunk 1 at the same time.
• CO Trunk 1 has priority, ‘9’, the Mobile Trunk 1 is assigned priority ‘7’.
• Three extensions, 2001, 2002 and 3001 are calling the Operator. Extension 3001 has priority ‘7’, while
extensions 2001 and 2002 have the same priority, ‘5’.
• Now, on the Operator extension, which is the landing destination, the incoming calls from the trunks and
the extensions will land in the following chronological order:
Caller Time of the Call Priority
SLT 2001 10:00:00 5
SLT 2002 10:00:05 5
CO Trunk 1 10:00:10 9
Mobile Trunk 1 10:00:12 7
DKP 3001 10:00:15 7
• These incoming calls, however, will appear on the Display of Operator’s phone (DKP or Extended IP
Phone) in the order of priority:
• CO 1
• Mobile Trunk 1
• DKP 3001
• SLT 2001
• SLT 2002
• Now, when the Operator goes Off-hook (pressing speaker key or picking up the handset), the call on CO 1
will be answered first, as CO Trunk 1 has the highest priority.
• The Operator goes On-hook and then Off-hook, the call on Mobile Trunk 1 will be answered. Though
Mobile Trunk 1 and DKP 3001 have the same priority, ‘7’, Mobile Trunk 1 will be answered first, following
the chronological order.
• When the Operator goes On-hook after answering the call on Mobile Trunk 1, the call from DKP 3001 will
be placed on the Operator phone with a Priority Ring (configurable; default: Triple Ring).

• When the Operator goes Off-hook, the call from DKP 3001 is answered.
• When the Operator goes On-hook and then Off-hook after answering the call from DKP 3001, the call from
SLT 2001 will get answered first, though both 2001 and 2002 have the same priority, ‘5’. In this case,
Priority Ring will not be played.
• Thus, calls from trunks and extensions are answered by the landing destination in the order of priority.
Where priority is the same, calls are answered in chronological order. Calls from extensions with higher
priority are indicated by a Priority Ring on the landing destination.
Priority is relevant only when there is more than one call on the destination.
You can assign Priority to SLT extensions. However, Priority is not relevant when the SLT is a landing
destination, because there cannot be more than one call ringing on an SLT extension at a time.
An “Intercom” call will be placed at the destination only if the caller has a higher priority.
How to configure
To assign Priority to Trunks, you must set the priority in their “Trunk Feature Template”. See “Configuring Trunks”.
To assign Priority to Extensions, see instructions for configuring the respective Extension port type:
• “Configuring E&M Lines”
• “Configuring Radio Interface”
• “Configuring Magneto Interface”
• “Virtual Extension”
If required, you may change the Ring Pattern of the Priority Ring. See “Distinctive Rings” for instructions.

Privacy
What’s this?
Extensions of SARVAM UCS can be protected from the intrusions by other extensions or from trunk calls by
activating Privacy.
How it works
Intrusions can occur on an extension when another extension invokes the following features:
• DND Override
• “Interrupt Request (IR)”
• “Barge-In”
• “Raid”
Intrusions can also occur,

• When an external caller uses “Auto Attendant” to reach an extension.
• When there is a call from another trunk line when you are in speech.
To prevent such intrusions, SARVAM UCS enables you to set the following types of Privacy:
• Privacy from Interrupt Request, Barge-In, DND Override: This type of Privacy protects an extension
from intrusions by other extensions using Interrupt Request, Barge-In or DND Override.
For example: Extension A has Privacy from Interrupt Request, Barge-In and DND Override.
Extension A and B are in speech, Extension C attempts to intrude the conversation Interrupt Request or
Barge-In. Extension C’s call will be blocked and C will get error tone.
Now, Extension A has set DND and Extension B attempts to override it using DND Override. Since A has
Privacy from DND Override, B’s call will be blocked and B will get error tone.
• Privacy from Raid: This type of Privacy protects an extension from intrusions by other extensions using
Raid.
For example: This type of Privacy is set on Extension A. Extension A and B are in speech, Extension C
uses Raid to intrude the conversation. Extension C’s call will be blocked and C will get error tone.
• Privacy from Trunk call intrusion: This type of Privacy prevents the extensions in the Trunk Landing
Groups that are busy from being intruded by another waiting call. For this type of Privacy to work, the
feature “Trunk Call Waiting” must be disabled on the extension.
For example: Extension A is the first trunk landing extension for calls on Trunk 1. Extension A and B are in
speech. A new call lands on Trunk 1. If A has Trunk Call Waiting beeps disabled, A will not hear the
intrusion beeps. The system will land the call on the next extension in the trunk landing group for calls on
Trunk 1.

• Privacy from Built-In Auto Attendant call: This type of Privacy protects the extension from being
accessed by external callers using “Auto Attendant”.
For example: This type of Privacy is set on Extension A. Extension A and B are in speech, external caller
C uses Built-In Auto Attendant to call extension A. C’s call will be blocked and C will get error tone.
How to configure
To provide Interrupt Request to extension users, you must enable this feature in the “Class of Service (COS)”
assigned to them for the time zones in their “Station Basic Feature Template”.
By default, Privacy from Raid is enabled in the Class of Service of all Extension types: SLT, DKP, SIP, ISDN. So,
none of the extensions can raid the other. You may disable this feature in the Class of Service of extensions, which
you want to protect from Raid.
By default, Privacy from Interrupt Request, Barge-In and DND Override are disabled in the Class of Service of all
Extension types. You may enable this feature on extensions which you want to protect from intrusions using any of
these features.
By default, Privacy from Built-In Auto Attendant is disabled on all Extension types. You may enable this feature on
extensions which you do not want external callers to reach.
By default, Trunk Call Waiting is disabled on all Extension types. You may keep this feature disabled on extensions
which you want to provide Privacy from Trunk Call intrusion beeps.
For instructions, see “Class of Service (COS)”, “Station Basic Feature Template”. Also see,

Power Fail Transfer
What's this?
When the power supply to the System fails, SLT 1 and SLT 2 are automatically connected to CO trunks CO1 and
CO2 respectively of the same card. The System switches from normal functioning mode to the Power Failure
Connections. Only the trunks handled by Power Failure Connections can be used during a power failure. Outgoing
calls can be made from such extensions as well as incoming calls on the trunks can land on these extensions.
When power resumes and normal system functioning is restored, calls on the Power Failure Connections remain in
effect until they are terminated.
The following Cards of SARVAM UCS support Power Fail Transfer.
ETERNITY ME CO8+SLT24
Combination card, with 8 CO ports to connect 8 CO analog trunk lines and 24 SLT ports to connect 24 Single Line
Telephones.
L1

Combination Card, with 4 ports to connect 4 Two-wire Trunk lines, 2 ports to connect 2 Digital Key Phones and 12
ports to connect 12 Single Line Telephones.

Combination Card, with 4 ports to connect 4 Two-wire Trunk lines, 2 ports to connect 2 Digital Key Phones and 8
ports to connect 8 Single Line Telephones.
Force Call disconnection: If PFT ports are in conversation during power failure and power supply is restored,
then any other extensions of the system cannot invoke Forced Call Disconnection on PFT SLT or PFT CO port.
ACB (Auto Call back): If PFT ports are in conversation during power failure and power supply is restored, then
other extensions of the system will be able to set ACB on PFT SLT or CO port.
Raid: If PFT ports are in conversation during power failure and power supply is restored, then system shall not
allow other extensions of system to invoke Raid feature on called PFT SLT or CO port.
Barge - In: If PFT ports are in conversation during power failure and power supply is restored, system shall not
allow any other extensions of the system to access Barge in on busy PFT SLT port.
Interrupt Request: If PFT ports are in conversation during power failure and power supply is restored, system
shall not allow any other extension to make an interrupt request on busy PFT SLT port.
Trunk Reservation: If PFT ports are in conversation during power failure and power supply is restored, system
shall allow you to access Trunk reservation on busy PFT CO port.

QSIG
What's this?
Q-Signaling (QSIG) is an ISDN based protocol for signaling between two Systems. QSIG is a protocol based on
internationally agreed Standards for ISDN. Using QSIG you can network two or more SARVAM UCS. This is known
as 'Interoperability'. The SARVAM UCS supports QSIG on the T1E1PRI ports.
The basic call procedure in the QSIG is implemented as per the ECMA-143. The generic functional protocol for the
Supplementary Services is implemented as per ECMA-165.
This provides details about following:

a. QSIG Supplementary Services
b. Configuring for - Basic Call and Supplementary Services
c. Configuring for - IC Call using DDI Routing Over QSIG
To use this feature you must purchase a License. Refer the topic “License Management” to know more.
a. QSIG Supplementary Services

The SARVAM UCS supports following supplementary services in QSIG:
• Advice of Charge (AOC)
• Identification:
• The Calling Line Identification Presentation (CLIP)
• Calling Line Identification Restriction (CLIR)
• The Connected Line Identification Presentation (COLP)
• Connected Line Identification Restriction (COLR)
• Name Identification:
• The Calling Name Identification Presentation (CNIP)
• The Connected Name Identification Presentation (CONP)
• Call Diversion:
• Call Forward Unconditional (CFU)
• Call Forward On Busy (CFB)
• Call Forward on No Reply (CFNR)
• Call Completion:
• Call Completion on Busy Subscriber (CCBS)
• Call Completion on No Reply (CCNR)
• Do Not Disturb (DND)
• Do Not Disturb - Override (DNDO)
• Call Intrusion (CI)

• Call Transfer (CT)
• Recall (RE)
• Message Wait Indication (MWI)
• Call Offer (CO)
The implementation of these features in QSIG is as per specific ECMA standards as described below.
Advice on Charge:
• Advice on Charge (AOC) is implemented as per ECMA-211 and ECMA-212.
• Using this feature, the caller can know the cost of the call made to public network using networked trunk.
The cost of the call will be determined by using Call Cost Calculation. Refer chapter “Call Cost Calculation
(CCC)”
.
• The SARVAM UCS supports 'AOC-E' End of the Call Charge Information. SARVAM UCS supports
charging for calls made to public network. The Cost of the call is calculated by the end System from which
the call is terminated on the public network.
CLIP/CLIR/COLP/COLR/CNIP/CONP
• The Identification features are implemented as per ECMA-163 and ECMA-164.
• Refer chapters “Calling Line Identification and Presentation (CLIP)” and “Calling Line Identification
Restriction (CLIR)”.
Call Diversion
• Call Diversion is implemented as per ECMA-173 and ECMA-174. Refer chapters “Call Forward” and “Call
Forward-Remote”.
• Remote activation/deactivation of Call Forward is not supported.
Call Completion-on Busy Subscriber and on No Reply (CCBS and CCNR)

• Call Completion (CC) features CCBS and CCNR are implemented as per standards ECMA-185 and
ECMA-186.
• The CCBS is implemented as the feature Auto Call Back on Busy and the CCNR is implemented as the
feature Auto Call Back on No Reply. Refer chapter “Auto Call Back (ACB)”.

• DND is implemented as per standards ECMA-193 and ECMA-194. Refer chapters “Do Not Disturb (DND)”.
Do Not Disturb - Override (DNDO)

• DND is implemented as per standards ECMA-193 and ECMA-194. See the topic “Do Not Disturb (DND)”.

Call Intrusion (CI)
• Call Intrusion is implemented as per standards ECMA-202 and ECMA-203.
• The Call Intrusion (CI) is implemented as Raid feature. However the extension on which intrusion is made,
will not get 'beeps'. On successful intrusion, the 'conference 3-party' type call will be established. See the
topic “Raid” and “Conference-3 Party”.
Call Transfer
• Call Transfer is implemented as per standards ECMA-177 and ECMA-178. Refer chapter “Call Transfer”.
• The SARVAM UCS supports Call Transfer by Joining method.
'CLIP-Hold' flag on the Transferring Station:

• If 'CLIP-Hold' flag is enabled on the Transferring Station, the SARVAM UCS will send the held party's
number as calling number while placing call on QSIG.
• If 'CLIP-Hold' flag is disabled on the Transferring Station, the SARVAM UCS will send the calling station's
number as calling number. Refer chapter “Calling Line Identification and Presentation (CLIP)” and “Station
Advanced Feature Template” for more details of CLIP-Hold.
• It is recommended to enable CLIP Hold flag for the operator extension, so that when operator transfers
the call to another System using QSIG, the terminating System (SARVAM UCS) can identify it as call
from Public network and treat it as an incoming call.
• As a Terminating System, the SARVAM UCS will consider the call as internal or from public network
depending upon the length of the digits received as the calling number.
• If Calling number is not received because of any reason, and CLIP- Hold flag is enabled, the SARVAM
UCS will send the 'Trunk Name' configured for the trunk port as calling line identification.
Recall (RE)
• Recall (RE) is implemented as per standards ECMA-213 and ECMA-214 and ECMA-143.
• The SARVAM UCS supports Recall-Busy and Recall-No Answer features like SARVAM UCS features,
Call Transfer-On Busy and Call Transfer-While Ringing.
• Refer chapter “Call Transfer” for more details.
Message Wait Indication (MWI)

• Message Wait (MWI) is supported as per standards ECMA- 241 and ECMA-242.
• Refer chapter “Message Wait” for details about form of indications.
Call Offer
• Call Offer is implemented as per standards ECMA-191 and ECMA-192.
• The call offer feature is implemented as the 'Interrupt Request' feature.

• Refer chapter “Interrupt Request (IR)” for more details.
b. Configuring for - Basic Call and Supplementary Services
For simple application as shown in above figure, make settings as given below for System 1 and System 2.
• Connect T1E1PRI line between System 1 and System 2.
• Now open Web Jeeves and configure the settings in System 1 and System 2 as given below:
Settings for System 1:

Following basic settings are required in SARVAM UCS System 1.
• T1E1 Port Parameters:

• Set Signal Type as "QSIG"
• Set Orientation Type as "Network"
• T1E1 PRI/QSIG Signaling Parameters:

• Set ISDN Switch Variant as "QSIG E1"
• Closed User Groups (CUG)
Strip Digit Self Dialed Digit Apply Toll Apply Call

Index Route Code OGTBG
Count Route Count Control Cost
001 21 01 0 Enable 04
002 31 01 0 Disable 04
003 0 01 0 Disable 04
• OG Trunk Bundle Groups
OGTB OGTB OGTB OGTB

Group No. Rotation Flag LCR Type -------
Member 1 Member 2 Member 3 Member 8
01 Disable None 01 00 00 ------- 00

• OG Trunk Bundle
Automatic Number
Template Rotation Translation (ANT)
Channel Trunk
Number Port Port Type ANT Table
Number Count Apply
Type No. No.
01 T1E1PRI 001 01 030 Cyclic Disable 05
Following settings are required for CLI display.
• In T1E1PRI Port Parameters, under DDI Routing configure Outgoing (OG) Reference ID as "01" .
• DDI Routing-Outgoing Reference Table.
Table ID OG Ref. ID Start Channel No. Channel Count Pilot Number DDI Routing Ref. ID
01 01 01 30 2100 01
• DDI Routing-DDI Routing Table
Table Reference Start DDI Total DDI DDI No. Port Port Start DDI
ID ID No. Numbers Digit Count Type Number Flexible No.
001 01 2100 0100 4 FLEXNUM 000 2100
Settings for System 2:

Following basic settings are required in System 2.
• T1E1 Port Parameters

• Set Signal Type as "E1 QSIG"
• Set Orientation Type as "Terminal"
• T1E1 PRI/QSIG Signaling Parameters:

• Set ISDN PRI Variant as "QSIG E1"
• Closed User Groups (CUG)
Route Strip Digit Self Dialed Apply Toll Apply Toll

Index OGTBG
Code Count Route Digit Count Control Control
001 21 02* 0 Enable 04
002 31 02* 0 Disable 04
*Assign OGTBG of T1E1PRI-1 used for QSIG connectivity with System 1.

• OG Trunk Bundle Groups
Group Rotation LCR OGTB OGTB OGTB OGTB

-------
Number Flag Type Member 1 Member 2 Member 3 Member 8
01 Disable None 01 00 00 ------- 00
02 Disable None 02 00 00 ------- 00
• OG Trunk Bundle
Automatic Number
Channel Trunk
Number Port Port Type ANT Table
no. Count Apply
Type No. No.
01 Mobile 001 01 008 Cyclic Disable 05
02 T1E1PRI 001 01 030 Cyclic Disable 05
Following settings are required for CLI display.
• In T1E1PRI Port Parameters, under DDI Routing configure Outgoing (OG) Reference ID as "01".
• DDI Routing-Outgoing Reference Table.
Table ID OG Ref. ID Start Channel No. Channel Count Pilot Number DDI Routing Ref. ID
01 01 01 30 3100 01
• DDI Routing- DDI Routing Table
Table Reference Start DDI Total DDI DDI No. Port Port Start DDI
ID ID No. Numbers Digit Count Type Number Flexible No.
001 01 3100 0100 4 FLEXNUM 000 3100
Basic Call:
Given below are various examples of calls made/received between the systems.
Example 1:
To make call from station of System 1 to any station of System 2 or vice versa.
• From station 2101 of System 1, dial station number 3101 or from station 3101 of System 2, dial station
number 2101.
Example 2:
To make outgoing call from station of System 1 using Trunk of System 2.
• From station 2101 of System 1, dial 0270 (TAC at System 2 for Trunk) and then dial 02652630555
(External Party number).

Example 3:
Incoming call on Trunk at System 2 using Auto Attendant271.
• Enable Auto Attendant on Mobile Trunk at System 2.
• Now External Party dials the number of the Mobile port. Incoming call is answered by System 2. External
Party gets Auto Attendant Music. Now External Party dials station number.
• If External Party dials station number 3101 then call lands on 3101 of System 2.
• If External Party dials station number 2101 then call lands on 2101 of System 1.
Supplementary Services:
Working of Supplementary Services of QSIG in SARVAM UCS is explained as below.
Identification:
Station 2101 of System 1, dial station number 3101.
• CLIP/CNIP:
Name and Number of station 2101 is displayed as CLI on station 3101.
• COLP/CONP:
When 3101 of System 2 answers the call then Name and Number of station 3101 is displayed on station
2101 using reverse DDI method.
• CLIR:
To restrict CLI at station 3101, configure the following for station 2101.
• Allow "CLI Restriction (CLIR)" in Class of Service for station 2101.

• Enable CLIR by dialing 103-1 from station 2101.
Here, Name and Number of station 2101 is not displayed on station 3101.
If "CLI Restriction (CLIR) Override" is enabled in Class of Service for station 3101, then Name and Number
of station 2101 is displayed on station 3101.
• COLR:
To restrict CLI when incoming call is answered by 3101, configure the following for station 3101.
• Allow "CLI Restriction (CLIR)" in Class of Service for station 3101.

• Enable CLIR by dialing 103-1 from station 3101.
270. Here 2101 of System 1 does not get dial tone after dialing 0 (TAC at System 2 for Trunk) because 0 is programmed in CUG table
for System 1.
271. Refer "“c. Configuring for-IC Call using DDI Routing Over QSIG” topic for how to route Incoming Call on T1E1PRI/SIP over QSIG
using DDI Routing.

Here, Name and Number of station 3101 is not displayed on station 2101 when station 3101 answers the
incoming call.
If "CLI Restriction (CLIR) Override" is enabled in Class of Service for station 2101, then Name and Number
of station 3101 is displayed on station 2101.
Call Diversion:
Station 2101 of System 1 wants to set Call Diversion (Call Forward) to station number 3101 of System 2.
• Call Forward-Unconditional:
From station 2101 of System 1, dial 131-3101-#*. Now all incoming calls on 2101 are forwarded to 3101 of
System 2.
• Call Forward-on Busy:
System 2 when 2101 is busy.
• Call Forward- on No Reply:
System 2 when 2101 does not answer the incoming call for RBT- Transfer on No Reply Timer.
• Call Follow Me does not work in QSIG.

• Call Forward-Remote does not work in QSIG.
• Call Forward-Dual Ring does not work in QSIG.
• Call Forward-External number on QSIG not supported, that is, if '0' is TAC code at System 2 and
station 2101 of System 1 dials 131-0-02652630555 (External Number)-#* then station 2101 gets error
tone.
Call Completion:
Station 2101 of System 1 wants to set CCBS/CCNR (ACB) on station number 3101 of System 2.
• CCBS/CCNR:
From station 2101 of System 1, dial station number 3101 of System 2. Now dial Access code of ACB (Dial
2) to set CCBS/CCNR on station 3101 of System 2.
Call Intrusion:
Station 2101 of System 1 and station 3101 of System 2 are in speech. Now another station can intrude their
conversation using the feature Call Intrusion.
Station 2102 of System 1 wants to intrude the conversation using Call Intrusion on station 2101 of System 1.
• Allow "Raid" feature in Class of Service for station 2102 of System 1.
• Set "Priority" level of station 2102 higher than "Priority" level of station 2101.
• Disable "Privacy from Raid" in Class of Service for station 2101 of System 1.

• Also disable "Privacy from Raid" in Class of Service for station 3101 of System 2.
• Now from station 2102 call 2101. After getting busy tone, dial Access code of Raid (Dial 5). Speech will be
established between all three parties 2101, 2102 and 3101.
If "Privacy from Raid" is enabled in Class of Service for station 2101 of System 1 or 3101 of System 2, then
Raid will not function.
Station 3102 of System 2 wants to intrude the conversation using Call Intrusion on station 2101 of System 1.
• Allow "Raid" feature in Class of Service for station 3102 of System 2.
• Disable "Privacy from Raid" in Class of Service for station 2101 of System 1.
• Also disable "Privacy from Raid" in Class of Service for station 3101 of System 2.
• Now from station 3102 of System 2 call station number 2101 of System 1. After getting busy tone, dial
Access code of Raid (Dial 5). Speech will established between all three parties 2101, 3101 and 3102.
If "Privacy from Raid" is enabled in Class of Service for station 2101 of System 1 or 3101 of System 2, then
Raid will not function.
Call Transfer:
Station 2101 and station 2102 of System 1 are in speech. Now station 2101 of System 1 wants to transfer the call
to any station number or external number of System 2.
• From station 2101, dial Flash and then dial 3101 (Station number of System 2) or dial 0-02652630555
(External Number).
• Station 2101 goes ON hook or presses 'Transfer' key to transfer the call.
• User can transfer the call after speech is established with the second party or when it is 'Ringing' or 'Busy'.
Call Recall:
Station 2101 of System 1 transfers the call to station 3101 of System 2 when station 3101 of System 2 is 'Ringing'
or 'Busy'. Station 3101 does not answer the call.
• Call is returned to station 2101 after expiry of 'Transfer on Busy Timer', in case of transfer- on busy and
'Transfer while Ringing Timer' in case of Transfer- while ringing.
Call Offer:
Station 2101 of System 1 and station 3101 of System 2 are in speech. Now station 2102 of System 1 wants to give
Call Offer to station 3101 of System 2.
• Allow "Interrupt Request" feature in Class of Service for station 2102 of System 1.
• From 2102 of System 1, call station 3101 of System 2. After getting busy tone, dial Access code of
Interrupt Request (Dial 3).

• Station 3101 of System 2 gets beeps.
If "Privacy from Interrupt Request and Barge-In" is enabled in class of Service for station 3101 of System 2,
then Call Offer will not function.
Do Not Disturb (DND):
Station 2101 of System 1 wants to set Do Not Disturb on his/her station.
• Allow "Do Not Disturb" in Class of Service for station 2101 of System 1.
• From station 2101, activate Do Not Disturb by dialing 181.
• Now station 3101’s call will not be placed on 2101 of System 1.
• Do Not Disturb-Override (DNDO):
Station 2101 of System 1 has activated Do Not Disturb. Station 3101 of System 2 calls station 2101 of
System 1.
• Allow "Do Not Disturb- Override" in Class of Service for station 3101 of System 2.
• Station 3101 of System 2 will be able to call station 2101 of System 1.
If "Privacy from DND Override" is enabled in Class of Service for station 2101, then station 3101’s call will
not be placed on station number 2101 of System 1.
Advice of Charge (AOC)

AOC is applicable for all external calls made by station of System 1 using trunk of System 2 (OG Gateway).
AOC-Outgoing Call:
Station 2101 of System 1 makes an external call by dialing 0-02652630555 (External Number).
• After completion of call, System 2 calculates the Cost of the call using AOC.
• Cost of call in AOC is calculated by System 2 as per parameters set in Jeeves for 'Call Cost Calculation’.
• If System 2 does not give any charge in AOC then no charge applies on station 2101 of System 1.
• In System 1 no parameters need to be configured.
AOC-Internal Call:
Station 2101 of System 1 calls station 3101 of System 2.
• All calls between two stations of both System are free of charge.
AOC-Call Transfer
Station 2101 and station 2102 of System 1 are in speech. Now station 2101 transfers the call to External Number
02652630555 using trunk of System 2.

• Cost of the call is calculated as per "Call Splitting" option and "When Call Splitting is OFF, charge calls to"
option set in SMDR-Outgoing Calls at System 2.
• If "Call Splitting" is enabled in System 2, then charge before the call transfer is applied to station 2101 and
charge after the call transfer is applied to station 2102.
• If "Call Splitting" is disabled in System 2, then charge is applied as per "When Call Splitting is OFF, charge
calls to".
• If "When Call Splitting is OFF, charge calls to" is set to 'Originating' then all charges before and after the
call transfer is applied on station 2101 who has transferred the call.
• If "When Call Splitting is OFF, charge calls to" is set to 'Terminating' then all charges before and after the
call transfer is applied on station 2102.
Message Wait Indication (MWI):
Station 2101 of System 1 is an Operator and wants to set Message wait on station 3101 of System 2.
• In the "Station Advanced Feature Template" of System 2 and configure "MW Notification Type" as
“Stuttered Dial Tone".
• From station 2101, dial Access Code "1076 - 3101-1", to set Message wait on station 3101 of System 2.
• Now Message wait is set on station 3101. When station user 3101 goes off hook, s/he will hear stuttered
dial tone.
Similarly, you can cancel message wait set on station 3101 by dialing access code "1076-3101-0".
c. Configuring for-IC Call using DDI Routing Over QSIG

If user wants DDI based routing over QSIG then following setup can be made (Not required for normal
scenario explained in Basic Call and Supplementary Services).
The setup is explained below:
• The T1E1PRI trunk of System 2 has:

• Pilot Number with MSN number as 0265304
• DDI numbers from 2100-2199 and 3100-3199
• Station numbers from 2100 to 2199 are connected at System 1.

• Station numbers from 3100 to 3199 are connected at System 2.
• System 1 and System 2 are connected with each other using QSIG.
• If Calling Party number is received as 0265304-2100 to 0265304-2199 then the call is routed to the
stations of System 1. If Calling Party number is received as 0265304-3100 to 0265304-3199 then the call
is routed to the stations of System 2.
To configure various parameters in System 1 refer Configuration of "Basic Call" and "Supplementary Services".
In System 2 only the following needs to be done:
• Configuring the OG Trunk Bundle
Automatic Number
Channel Trunk
No. Port Port Type
no. Count Apply ANT Table No.
Type No.
02 T1E1PRI 001 01 030 Cyclic Enable 05
• Configuring the External Number List
Location External No. List 05 External No. List 06
001 026530421 21
• Configuring the DDI based routing.
• Program Incoming (IC) Reference ID for all Time Zones as "01" in T1E1 Port Parameters.
• DDI Routing-Incoming Route Reference Table.
Route
Route Trunk
IC Start Total DDI Route Ring on TLG
Table on TLG Feature
Ref. Channel Channel Routing on First Timer when
ID when Template
ID No Count Ref. ID Destination (Sec) No
Busy s
Reply
01 01 01 30 01 No 045 No No 01
02 01 01 30 02 No 045 No No 01
• DDI Routing-DDI Routing Table
DDI No.
Table Reference Start Total DDI Port Port Start DDI
Digit
ID ID DDI No. Numbers Type Number Flexible No.
Count
001 01 3100 0100 4 FLEXNUM 000 3100
002 02 2100 0100 4 ROUTGRP 009

• Routing Group
Routing Group Rotation Flag Member 01 Member 02 Member 03 Member 04-32
09 Enable OTBG-02 None None None
Basic Call:
Below examples explains the calls made/received using DDI Routing.
• External Party call the number 0265304-2101
Call lands directly on station 2101 of System 1 through QSIG.
• External Party calls the number 0265304-3101
Call lands on station 3101 of System 2.
Voice Mail for Extensions over QSIG

SARVAM UCS supports Voice Mail for extension users connected over QSIG. These are extensions of the remote
System (may be SARVAM UCS or any other System) connected with SARVAM UCS over QSIG.
To provide Voice Mail to the remote system extension users you must configure the Voice Mail Settings.
For details, see “Extensions Over Q-SIG”.

Quick Dial
What’s this?
Quick Dial provides DKP and Extended IP phone users the facility of ‘One-touch’ dialing of numbers stored in their
Personal Directory and the Global Directory.
How it works
Quick Dial is based on “Abbreviated Dialing”.
To be able to Quick Dial a number,
• the number must exist in the Personal or Global Directory assigned to the extension.
• Personal and Global Directory dialing must be allowed in the Class of Service of the extension.
• On the DKP and Extended IP Phones, DSS keys must be configured with the Short Codes or Abbreviated
Numbers that are to be dialed out. These short codes are derived from the Index numbers of the Personal
Directory and the Memory Location Index of the Global Directory.
• You can Quick Dial a number simply by pressing the DSS key.
• The system locates the number to be dialed out in the Personal/Global Directory on the basis of the Index
Number/Memory Location Index configured on the DSS Key.
How to configure
See “Abbreviated Dialing” for instructions on configuring and assigning the Personal and Global Directories.
To assign the Short Codes or Abbreviated Numbers to be used for Quick Dial on DSS keys, for each DKP/
Extended IP Phone extension,
• List down the numbers from the Personal Directory and Global Directory to be used for Quick Dial.
• If the number is from the Personal Directory assigned to the extension, note the Index number at which it is
stored in the Personal Directory: 001 to 025.
• If the number is from the Global Directory assigned to the extension, note the Memory Location Index at
which it is stored in the Global Directory: 100 to 999.
• Now, configure the Quick Dial numbers on the DSS keys of the DKP/ Extended IP Phone.
For detailed instructions on configuring DSS Keys on a Digital Key phone, see “DSS Keys Programming”.
For instructions on configuring DSS Keys on Matrix Extended IP Phone, see “Configuring SIP Extension
Settings as per the Extended Phone Type” under “Configuring SIP Extensions”.
• On the desired key, select Quick Dial as Function Type.

• As Offset, select the Index Number against which the number is stored in the Personal/Global Directory.
How to use
For EON and Extended IP Phone Users only

• Press the DSS Key assigned to the Quick Dial numbers.
• Talk when the called party answers.

Raid
What’s this?
Raid allows you to interrupt a telephone conversation between two extension users, turning the conversation into a
three-way call.
You can use Raid to land in a conversation between two extension users, or between an extension user and an
external caller, with a warning beep to the extension user. The extension user will hear a beep when you raid and
you will enter in to three-way speech with both parties.
You may also Raid a conversation without any warning by disabling the beep.
How it works
• A, B and C are extension users.
• A and B are in speech.
• C calls A.
• C dials the feature access code for Raid.
• Beep is played on A. Three-way speech is established between A, B and C.
• If any of these three parties disconnects, two-way speech is established between the remaining parties.
• Raid works only if the dialed extension is busy in two-way speech. The two-way speech may be with
another extension or with an external number on a trunk.
• You cannot Raid on Trunks, that is, if two external numbers are in two-way speech. In this case, C cannot
raid the conversation.
• Raid will not work if Privacy against Raid is enabled in the Class of Service of the extension being raided.
In this case, if Extension A has Privacy against Raid in its Class of Service, C will not be able to Raid the
conversation between A and B. To know more about this feature, see “Privacy”.
• The extension using Raid must have higher Priority assigned to it than the extension being raided. In this
case, C must have higher Priority than A to be able to invoke Raid.
Raid is a sensitive feature. You are advised to restrict access to this feature to select extension users.

How to configure
To be able to use Raid, extension users must have this feature enabled in the “Class of Service (COS)” assigned to
them for the time zones in their “Station Basic Feature Template”.
By default, beep is played as a warning to the extension being raided. If required, you may disable the beep played
during Raid, by clearing the Play Beep when Raid/Conference/Dial-In Conference begins check box in the
System Parameters. For instructions, see “System Parameters”.
How to use

• Press DSS Key assigned to Raid.
OR
For SLT Users


RCOC (Return Call to Original Caller)
What is this?
Generally, extensions users of the System are given a trunk access to make outgoing calls from their phones. It is
also common for a group of extensions to share the same trunks to make outgoing calls.
When an extension user of the System makes an outgoing call and the called party does not answer the call or is
busy on another line, it is possible for the called party to return the call (made by the extension user) on the basis of
the CLI number received.
However, when the called party returns the call, this incoming call is mostly likely to land on the Operator extension,
as incoming calls are usually routed to the Operator.
Now, the Operator has no way of knowing which extension made the call so as to transfer the call to that extension.
Instead, the Operator must either ask the called party whom they wish to speak to and transfer the call or put the
called party on hold and find out the extension that made the call. This is an unwieldy process for all concerned -
the Operator, the called party and the extension user who originally made the call.
This can be overcome if the System is able to route the returned call to the original caller's extension.
SARVAM UCS makes this possible with the Return Call to Original Caller feature.
This feature is supported on Mobile ports, BRI, T1E1PRI (T1E1) and SIP Trunks.
RCOC is not supported when calls are made from analog trunks — CO and E&M — due to the signaling
limitations of these trunks.
How it works
The Prerequisites
• RCOC is enabled on the desired Trunk/s - BRI, T1E1PRI, Mobile and SIP.
• RCOC is enabled in the Class of Service group assigned to the extension.
RCOC Table
RCOC table is maintained internally by the SARVAM UCS and it is non-programmable.
• SARVAM UCS can keep a record of 255272 entries in the RCOC Table.
• Each entry is kept for the duration of the RCOC Record Delete Timer (programmable; default: 999
minutes). Whenever a record is stored in the RCOC database, the Record Delete Timer for that entry is
activated. On the expiry of the Timer, the entry is deleted by the system.
272. ETERNITY LENX/MENX can store 512 entries in the RCOC table.

• If a same external number is dialed using 3 different SIP/Mobile Ports with RCOC enabled on all the
Trunks and Extensions, then if the called party calls back, the call will land on the original callers using
FIFO logic.
• Each record is deleted from the database either after the call is returned or on expiry of the Record Delete
Timer.
• The RCOC database remains unaffected during power outages.
The Process
• When an extension having RCOC feature in its Class of Service makes an out going call, the system
checks if RCOC is enabled on the trunk through which the outgoing call is routed.
• If RCOC is enabled on the trunk, the system stores the record of the outgoing call in the RCOC Table.
• The system sets RCOC for the outgoing call in the following conditions, according to the Destination
Port273:
• If the Destination Port is a Mobile Port, RCOC is set when:

• called party is busy.
• called party is out of coverage/mobile is switched off.
• called party does not reply.
• called party rejects the call.
• caller (extension that made the call) goes ON-Hook before the called party answers the call.
• If the Destination Port is a BRI or T1E1PRI Port or a SIP Trunk, RCOC is set when:
• called party is busy.
• called party does not answer the call.
• caller (extension that made the call) goes ON-Hook before the called party answers the call.
• When SARVAM UCS acts as Gateway,

• If the Originating Port is either BRI-NT or T1E1-NT, the system checks the Class of Service allowed
to the trunk port.
• If the Originating Port is a trunk (T1E1-TE, BRI-TE, CO, MOBILE, SIP), RCOC will be set, if it is
enabled on Destination Port.
• RCOC shall be set only if the Calling Party's Number is available. If calling party number is missing,
then RCOC shall not be set.
While returning call to the original caller, if you want SARVAM UCS to match the Trunk Port type and Trunk
Port number of the incoming call with the Trunk Parameters of the entry stored in the RCOC table, then
you must enable Apply RCOC only if the caller calls back on the same trunk from which the call was
made option in the System Parameters.
• Whenever there is an incoming call on any trunk, the system checks the Apply RCOC only if the caller
calls back on the same trunk from which the call was made flag in the System Parameters.
• If enabled, the system matches the Trunk Port number and the Trunk Port type of the incoming call with
the entry stored in the RCOC Table.
• If disabled, the system matches the CLI of the incoming call with the entry stored in the RCOC Table.
273. The Trunk from which the outgoing call is made.

• If a matching record entry is found, the system routes the call to the original caller and clears the record
entry from the RCOC Table.
• The return call rings on the original caller's extension for the period of the Ring Back Tone Timer
(programmable; default 45 seconds). If the original caller does not answer the call within this Timer, the call
is routed to the Trunk Landing Group programmed for that trunk.
The Ring Back Tone Timer is common to all internal calls; calls made from one extension will ring on the
destination extension till the end of this timer. Change in the Ring Back Tone Timer for RCOC returned
calls on original caller's extension will also be applied on Ring Back Tone Timer for all internal calls. So,
change this Timer taking this into consideration.
• If no match is found in the RCOC Table or the extension or the original caller is busy, the call will be routed
according to the incoming call logic programmed (as programmed in the assigned Trunk Feature
Template) in the system.
• As RCOC is a “Class of Service (COS)” dependent feature, extensions that are not allowed this feature
in their COS cannot have their calls returned; even if this feature is enabled on the Trunk they used to
make the call.
• In case of Call Transfer, RCOC will be set for the extension which made the call on the trunk.
Feature Interaction: RCOC and DISA CLI Authentication
• When DISA CLI Authentication (Multiple Calls or One Call) is enabled on a trunk, whenever there is an
incoming call on the trunk, the system will first check the DISA CLI Authentication Table.
• If a matching entry is found in the DISA CLI Authentication table, the system will give dial tone to the caller.
• The caller can now invoke RCOC feature by dialing ** (pressing Star key twice).
OR
• The caller can make calls to a station or an external number or use a feature as required.
• If the caller invokes RCOC feature by dialing ** (pressing Star key twice), the system will check the RCOC
Table.
• If a matching record entry is found, the system routes the call to the original caller and clears the record
entry from the RCOC Table.
How to configure
For this feature to work, it must be enabled on the Trunk and in the Class of Service of the extensions. See
“Enabling RCOC on Trunk using Jeeves”, “Enabling RCOC on Trunk using SE Command” and “RCOC in Class of
Service” for instructions.
If desired, the related Timers, that is, the RCOC Record Delete Timer and the Ring Back Tone Timer may also be
changed. See “System Timers and Counts” for instructions.
If you want SARVAM UCS to match the Trunk Port type and Trunk Port number of the incoming call with the Trunk
Parameters of the entry stored in the RCOC table, then you must enable Apply RCOC only if the caller calls

back on the same trunk from which the call was made flag in the System Parameters. See “System
Parameters” for instructions.
RCOC on Trunk
Enabling RCOC on Trunk using Jeeves

• Click the trunk parameters of the trunk type on which you want to enable this feature, namely:
• SIP Parameters
• BRI Parameters
• T1E1 Port Parameters (T1E1PRI)
• MOBILE Port Parameters
• The page containing the trunk parameters will open.
• Select the 'Return Call to Original Caller (RCOC)' check box on the page to enable this feature on the
desired trunk port.
• Log out of Jeeves or continue with other programming tasks.
Enabling RCOC on Trunk using SE Command

To enable RCOC on T1E1 Trunk

• Dial 6145-1-T1E1-Code to enable the feature on a single trunk.
• Dial 6145-2-T1E1-T1E1-Code to enable the feature on a range of trunks.
• Dial 6145-*-Code to enable the feature on all trunks.
Where,
Code is
0 for Disable
1 for Enable
Default: Disable
To enable RCOC on BRI Trunk

• Dial 6220-1-BRI-Code to enable the feature on a single BRI trunk.
• Dial 6220-2-BRI-BRI-Code to enable the feature on a range of BRI trunks.
• Dial 6220-*-Code to enable the feature on all BRI trunks.
Where,
BRI is the software port number of the BRI trunk from 01 to 32.
Code is
0 for Disable
1 for Enable
Default: Disabled

To enable RCOC on Mobile Port
• Dial 8030-1-Mobile -Code to enable the feature on a single Mobile port.
• Dial 8030-2-Mobile-Mobile-Code to enable the feature on a range of Mobile ports.
• Dial 8030-*-Code to enable the feature on all Mobile ports.
Where,
Mobile is the software port of the Mobile port from 01 to 32.
Code is
0 for Disable
1 for Enable
Default: Disabled
To change the RCOC Record Delete Timer

• Dial 3521-Minutes
Where,
Minutes are from 001 to 999
Default: 999 minutes
To change the Ring Back Tone Timer

• Dial 3503-Seconds
Where,
• Exit SE mode.
RCOC in Class of Service

The feature 'RCOC' must be enabled in the “Class of Service (COS)” group assigned to the extensions for returned
calls to land on them.
In the default Station Basic Feature Template 01 assigned to all extensions of the SARVAM UCS, the default Class
of Service group 01 has the feature "RCOC" enabled. So, all extensions of SARVAM UCS are by default allowed
this feature.
There is no need to program this feature if all extensions are to be allowed this feature.
However, if you want to deny this COS feature to certain extensions and allow this feature to all other extensions,
follow these steps:
1. Define a CoS group with RCOC disabled.

3. Assign this newly prepared Station Basic Feature Template to the extensions on which 'RCOC' is to be
disabled.
Refer the topic “Class of Service (COS)” and “Station Basic Feature Template” for instructions.

Real Time Clock (RTC)
What’s this?
Various features and facilities supported by SARVAM UCS, such as Alarms, Station Message Detail Records,
System Activity Log, Time Zones, Daylight Savings, certain Voice Mail features need the correct time and date for
their proper functioning.
The SARVAM UCS has a built-in Real Time Clock (RTC) circuit that maintains date and time. When you select
Region, the RTC is automatically set to the current date and time of the country/region where SARVAM UCS is
installed.
Since the RTC circuit may drift over a period, it is recommended that you check and reset RTC values at least once
every month to correct this drift. The RTC of SARVAM UCS takes care of leap years.
How to Configure
Configuring RTC using Jeeves

• Click Configuration, click Date and Time.
• Click Date and Time to expand.
• Click on Click to edit to set the date and time as per your requirement.

• Select the desired Date Format:
• Enter the current Date
• The current Day will appear as per the date you set.
• Set the current Time in HH:MM:SS (hours-minutes-seconds) format.
• Select Time Zone of the country/region where SARVAM UCS is installed.
• Click Submit to save RTC settings.
• You may log out of Jeeves
Configuring RTC using a Telephone

To set the date format, dial:

• 1000-Date Format
Where,
Date format is
1 for DD-MM-YYYY
2 for MM-DD-YYYY
Default: 1.
To set date, dial:

• 1001-Date-Month-Year
Where,
DD is 01 to 31 (leading zero mandatory for single digit date).
MM is 01 to 12 (leading zero mandatory for single digit month).
YYYY is year in four digits from 0000 to 9999.
Default: Current date is set.
First set the date format and then set the date.
To set the current time, dial:

• 1003-Hours-Minutes-Seconds
Where
Hours is 00 to 23 (24-hour format in two digits)
Minutes is 00 to 59 (in two digits)
Seconds is 00 to 59 (in two digits)
Default: Current Time is set.
To select a Time Zone, dial:

• 1002-Time Zone Index Number
Where,
Time Zone Index Number is from 001 to 124, refer the table below.
Default: Index 050 - India.
• Exit SE mode.

Time Zones
Index Time Zone
001 Afghanistan (GMT+04:30)
002 Algeria (GMT+01:00)
003 Antigua and Barbuda (GMT-04:00)
004 Argentina (GMT-03:00)
005 Australia (Perth) (GMT+08:00)
006 Australia (Adelaide) (GMT+09:30)
007 Australia (Brisbane, Canberra, Melbourne, Sydney) (GMT+10:00)
008 Austria (GMT+01:00)
009 Bahamas (GMT-05:00)
010 Bahrain (GMT+03:00)
011 Bangladesh (GMT+06:00)
012 Belarus (GMT+02:00)
013 Belgium (GMT+01:00)
014 Bhutan (GMT+06:00)
015 Bolivia (GMT-04:00)
016 Bosnia and Herzegovina (GMT+01:00)
017 Botswana (GMT+02:00)
018 Brunei (GMT+08:00)
019 Brazil (Fernando De Noronha) (GMT-02:00)
020 Brazil (Brasilia, Rio de Janeiro, Sao Paulo) (GMT-03:00)
021 Brazil (Manaus) (GMT-04:00)
022 Brazil (Acre) (GMT-05:00)
023 Bulgaria (GMT+02:00)
024 Cambodia (GMT+07:00)
025 Cameroon (GMT+01:00)
026 Canada (St. John's) (GMT-03:30)
027 Canada (Halifax) (GMT-04:00)
028 Canada (Montreal, Ottawa, Toronto) (GMT-05:00)
029 Canada (Winnipeg) (GMT-06:00)
030 Canada (Calgary) (GMT-07:00)
031 Canada (Vancouver) (GMT-08:00)
032 Chile (GMT-04:00)

Index Time Zone
033 China (GMT+08:00)
034 Colombia (GMT-05:00)
035 Costa Rica (GMT-06:00)
125 Cote d’Ivoire (GMT+01:00)
036 Croatia (GMT+01:00)
037 Cuba (GMT-05:00)
038 Cyprus (GMT+02:00)
039 Czech Republic (GMT+01:00)
040 Denmark (GMT+01:00)
041 Egypt(GMT+02:00)
042 Fiji(GMT+12:00)
043 Finland(GMT+02:00)
044 France(GMT+01:00)
045 Germany(GMT+01:00)
046 Greece(GMT+02:00)
047 Guyana(GMT-04:00)
048 Hong Kong(GMT+08:00)
049 Hungary(GMT+02:00)
050 India(GMT+05:30)
051 Indonesia(GMT+07:00)
052 Iran(GMT+03:30)
053 Iraq(GMT+03:00)
054 Ireland(GMT)
055 Israel (GMT+02:00)
056 Italy(GMT+01:00)
057 Japan(GMT+09:00)
058 Jordan(GMT+02:00)
059 Kazakhstan(GMT+06:00)
060 Kenya(GMT+03:00)
061 Korea - North(GMT+09:00)
062 Korea - South(GMT+09:00)
063 Kuwait(GMT+03:00)
064 Kyrgyzstan(GMT+06:00)

Index Time Zone
065 Lebanon(GMT+02:00)
066 Libya(GMT+02:00)
067 Malaysia(GMT+08:00)
068 Maldives(GMT+05:00)
069 Mauritius(GMT+04:00)
070 Mexico (Mexico City) (GMT-06:00)
071 Mexico (Chihuahua) (GMT-07:00)
072 Mexico (Tijuana) (GMT-08:00)
073 Mongolia(GMT+08:00)
074 Mozambique(GMT+02:00)
075 Myanmar(GMT+06:30)
076 Namibia(GMT+01:00)
077 Nepal(GMT+05:45)
078 Netherlands(GMT+01:00)
079 New Zealand(GMT+12:00)
080 Nigeria(GMT+01:00)
081 Norway(GMT+01:00)
082 Oman(GMT+04:00)
083 Pakistan(GMT+05:00)
084 Paraguay(GMT-04:00)
085 Peru(GMT-05:00)
086 Philippines(GMT+08:00)
087 Poland(GMT+01:00)
088 Portugal(GMT)
089 Qatar(GMT+03:00)
090 Romania(GMT+02:00)
091 Russia (Moscow, St. Petersburg) (GMT+03:00)
092 Russia (Novosibirsk) (GMT+06:00)
093 Russia (Vladivostok) (GMT+10:00)
094 Singapore(GMT+08:00)
095 Slovakia(GMT+01:00)
096 South Africa(GMT+02:00)
097 Spain(GMT+01:00)

Index Time Zone
098 Sri Lanka GMT+05:30
099 Sudan(GMT+03:00)
100 Sweden(GMT+01:00)
101 Switzerland(GMT+01:00)
102 Syria(GMT+02:00)
103 Taiwan(GMT+08:00)
104 Tajikistan(GMT+05:00)
105 Thailand(GMT+07:00)
106 Turkey(GMT+02:00)
107 Uganda(GMT+03:00)
108 Ukraine(GMT+02:00)
109 United Arab Emirates(GMT+04:00)
110 United Kingdom(GMT)
111 United States (Atlanta, Augusta, Boston, Charlotte, Columbus, Detroit, Indiapolis,
Miami, NY, Philadelphia, Washington) (GMT-05:00)
112 United States (Chicago, Dallas, Des Moines, Memphis, Minneapolis, New Orleans,
Oklahoma, Omaha, St. Louis) (GMT-06:00)
113 United States (Albuquerque, Boise, Cheyenne, Denver, Salt Lake City) (GMT-
07:00)
114 United States (Las Vegas, Los Angeles, Phoenix, San Francisco, Seattle) (GMT-
08:00)
115 United States (Juneau) (GMT-09:00)
116 United States (Hawaii) (GMT-10:00)
117 Uzbekistan(GMT+05:00)
118 Venezuela(GMT-04:30)
119 Vietnam(GMT+07:00)
120 Yemen(GMT+03:00)
121 Yugoslavia(GMT+02:00)
122 Zambia(GMT+02:00)
123 Zimbabwe(GMT+02:00)
124 Saudi Arabia (GMT +3.00)

Reminder
What's this?
Reminders are a variation of the “Alarms” feature, requiring the Date and Time to be set for each Reminder call.
Reminder calls are useful for extension users who wish to be reminded of important tasks or appointments.
For Reminder calls, date and time are set in the following format:
Date is set, according to Date Format you selected in the “Real Time Clock (RTC)” parameters, as:
• Day-Month-Year (DD:MM:YYYY)
Or
• Month-Date-Year (MM:DD:YYYY).
• Reminders can be set and canceled by:

• the Operator from the Operator phone and from Jeeves.
• extension users from their phones.
• Multiple Reminders can be set for an extension by the Operator and/or by the extension user.
• The mechanism for serving Reminders calls can be configured as 'Personalized' or 'Automated'.
• Reminders can be voice-guided, if the SARVAM UCS has a Voice Mail System module installed in it.
• SARVAM UCS can register as many as 999 Reminders set by the Operator and extension users.
How it works
Personalized Reminder
When the Reminder call serving mechanism is configured as 'Personalized',
• The Operator Phone rings first274, displaying the number of the extension to which the reminder call is to
be served.
• When the Operator answers this call, a call is placed on the extension on which the reminder call is set.
• The extension phone rings for the duration of the Alarm Ring Timer.
• When the extension user answers the call, the Operator greets the extension user with the reminder
message.
274. The Operator phone rings for the duration of the Alarm Ring Timer. If the Operator does not answer the call, the SARVAM UCS will
make two more Alarm Attempts at an Alarm Attempt Interval of 5 minutes to call the Operator.

• If the extension user does not answer the call till the Alarm Ring Timer has elapsed, the Operator phone
will display a text message notifying 'No Reply' from the extension. The Reminder is now considered as
served.
• If the extension is busy275, the Operator phone will display a text message notifying that the extension
number is 'Busy'.
• The Operator can now choose to
• inform the extension user about the Reminder in person or send someone to do it.
OR
• try the busy extension again.
OR
• set “Auto Call Back (ACB)”.
Personal Reminders will work even if the extension user has set DND or Call Forward.
Automated Reminder
When the Alarm serving mechanism is configured as 'Automated',
• The extension phone rings at the set time till the end of the Alarm Ring Timer. If the extension phone is a
DKP or the Matrix Extended IP Phone, Reminder message will appear on its display.
• When the extension user answers the call, the user may be played music-on-hold, or a pre-recorded voice
message, or be connected to the Voice Mail, or routed to the Operator, depending upon the Alarm
Notification Type you have configured for the extension.
• If the extension user does not answer the reminder call, the SARVAM UCS makes two more attempts (in
all, 3 attempts) at an interval of 5 minutes between each attempt, to call the extension.
• If all Reminder call attempts go unanswered, the SARVAM UCS places the call on the Operator Phone.
The Operator Phone rings till the end of the Alarm Ring Timer. The Operator Phone displays the number of
the extension with the message 'No Reply'. The Reminder call is now considered as served.
• If the extension phone is busy, the SARVAM UCS will continue to make the Reminder call Attempts at the
Alarm Interval programmed. When all Alarm Attempts go unanswered, SARVAM UCS will place a call on
the Operator phone. The Operator Phone will display the number of the extension phone with the message
'Busy'.
275. An improperly placed receiver may also be the cause for the busy tone on the extension phone. In that case, the system will notify
the Operator Phone with the 'OFF-Hook Alert'.

Snooze
The Snooze function can be added to Automated Reminders to ensure that the extension user answers the call.
Snooze is a system-wide feature; when set, this function will be added to all Automated Reminder calls.
When Snooze is activated,
• The extension phone rings for the Number of Alarm Attempt configured, at the set Alarm Attempt
Interval.
• The extension stops ringing when the user answers the call and dials 0 to acknowledge the Reminder
call. This reminder call Acknowledgement Code 0 is non-configurable.
• Reminder can be set for Operator phones also.
• Reminder settings will be retained in the system during power down and system upgrades.
• When multiple reminder requests have been set by an extension user, the extension user cannot
selectively cancel a particular reminder request. Only the Operator can selectively cancel Reminders
set for an extension user from the System Administrator pages of Jeeves.
• It is not possible to modify—change the date and time—of a reminder request. So, you may cancel the
Reminder request and set a new one.
Reminder Status Report

The Operator can know the details of Reminders that have not been served on the Reminder page of Jeeves, from
the System Administrator (SA) mode or by pressing the DSS key assigned to Wakeup Call Log. To view the log
using DSS key, see “Alarm Status Report”.
The status of Reminders set by Operator as well as extension users appears on this page, with details of time
(hours and minutes), and serving mechanism (personalized, automated).
The Operator can view the Reminder report whenever required and can also print this report.
How to configure
The configuration of Reminders is the same as Alarms.
To configure Reminders feature, do the following:
• Select the Alarm Notification Type for the extensions.
• Configure, as required, the Alarm Call related parameters: Alarm Ring Timer, Number of Attempts,
Alarm Attempt Interval, Configurable Alarm Type and Configurable Alarm Category, and Snooze.
• Configure Macros, if the SLT extension has special function keys, and you want to a function key for the
Reminder feature.
For instructions, see the topic “How to configure” under “Alarms”.

How to use
Reminders can be set by the extension users by themselves. The extension users can also ask the Operator to set
the Reminder for them.
If the Voice Mail System module is installed in the SARVAM UCS, it can offer voice-guided Reminders to extension
users and the Operator.
Voice-guided reminders lead users through a menu, helping them set the alarm in a step-by-step manner.
Voice-Guided Reminders set/canceled by Operator

The Operator can set voice-guided Reminders for extension users

• Press DSS Key assigned to 'Remote Voice-Guided Reminder' function.
OR
• Dial 1072-035
• Follow Voice Mail System Prompts to set/cancel Reminder.
For SLT

• Dial 1072-035
• Follow Voice Mail System Prompts to set/cancel Reminder.
Voice-Guided Reminder set/canceled by Extension Users

• Press the key assigned to Voice-guided 'Reminder' function.
OR
• Dial 164
• Follow Voice Mail System prompts.
For SLT Users

• Dial 164
• Follow Voice Mail System Prompts.
If the SLT of the extension user has a special Reminder function key, the extension user can set the alarm using
this key.
For SLT with 'Reminders' Key
• Press 'Reminders' key. (The label on the SLT key may differ from model to model)
• Follow the Voice Mail System prompts to set/cancel reminders.

• SLTs with special function keys will work only if the corresponding Macros are programmed by the
System Engineer.
• Without the Voice Mail System installed, the extension user having SLT with the special Reminder
function key will not be able to set/cancel Reminders. This extension user can set/cancel Reminders
only by dialing the feature access code for voice-guided Reminders.
Non-Voice Guided Reminders set/canceled by Operator

To set Reminder for the extension user,
• Press the DSS Key assigned the 'Remote Reminder' function.

OR
• Dial 1072-033
• Enter Date and Time in the format:
DD: MM: YYYY: HH: MM
OR
MM: DD: YYYY: HH: MM (users in USA)
• Select 'Personalized' or 'Automated'.

• Press 'Enter' key to set Reminder.
• You get a confirmation tone and message.
To cancel Reminder set for the extension user,
• Press key assigned for ‘Remote Reminder’ function.

OR
• Dial 1072-033
For SLT
To set Reminder for an extension,
• Lift handset.
• Dial 1072-033
• Dial Date and Time in the format:
DDMMYYYYHHMM
OR
• MMDDYYYYHHMM (users in USA)
• Dial 1 for Personalized, Dial 2 for Automated.

To cancel Reminder set for an extension,
• Lift handset.
• Dial 1072-033
• Dial #
To cancel reminder calls selectively, go to 'Reminder Status' page from the System Administrator of
Jeeves.
If the 'Configurable Alarm Category' flag is enabled, only then the system will give the option of selecting
'Automated' or 'Personalized' as the serving mechanism. By default the serving mechanism is Automated.
Non-Voice Guided Reminders set/cancel by Extension Users
To set Reminder:
• Press the 'Reminder' key.

OR
• Dial 162
• Enter Date and Time in the format
DD:MM:YYYY:HH:MM
OR
MM:DD:YYYY:HH:MM (users in USA)
• Go Idle.
To cancel Reminder:
• Press 'Reminder' Key again.
OR
• Dial 162
For SLT Users
To set Reminder,
• Lift handset.
• Dial 162
• Dial Date and Time in the format
DDMMYYYYHHMM
OR
MMDDYYYYHHMM (users in USA)

To cancel Reminder,

• Dial 162
• Dial #
Viewing and Printing Reminder Status

The Operator can view the status of Reminders that are yet to be served from the System Administrator pages of
Jeeves or from an extension of the SARVAM UCS.
To view Reminder Status from the System Administrator pages of Jeeves,
• Open Jeeves.
• Click the Reports link.
• Under Reports, click the Reminder link.
The unserved Reminders appear on the page.
• To cancel any of the unserved Reminders,

• Select the Cancel Reminder check box of the extension number for which you want to cancel the
reminder.
• Click the Cancel Selected Reminders button at the bottom of the page.
• To print this page, click the Print button.
• Click Close to exit the page.
• Click Logout to exit SA mode.
To print Reminder reports from an extension of SARVAM UCS,

• Dial 1072-917.
• The report will be printed on the destination port assigned to Hotel/Motel reports.
Reminder Report AS ON 05-05-2016(Thu) AT 23:50

-------------------------------------------------------------------------------
Room# Phone# Reminder P Room# Phone# Reminder P
-------------------------------------------------------------------------------
3001 04-05-2016 12:25 3001 04-05-2016 12:14
3001 05-05-2016 17:11
-------------------------------------------------------------------------------
+ indicates Personal Reminder Page : 1
---End of Report---
• The ‘+’ sign on the report indicates Personal Reminder.
• If the date format of SARVAM UCS is set as MM-DD-YYYY or the Region 'USA' is selected, then the
reminder report will be printed according to this date format. To know more, see “Real Time Clock
(RTC)”.

Remote Programming
What’s this?
SARVAM UCS offers a Graphic User Interface, Jeeves, for configuration and maintenance. With the Jeeves and IP
connectivity, SARVAM UCS can be configured from across the globe. However, in some cases, where IP
connectivity from public network is not available, remote programming can be done using DISA.
Direct Inward System Access (DISA) facility of the system allows a remote user to login and use most of the
functions of the system. Programming is one such function allowed to the remote user. The remote user can
program the system using the same commands as used by the normal local station to program the system. The
user can login into the SA programming or SE programming mode.
How to use
• Make a DISA call.
• Login as DISA user. Get the DISA login beeps. Enter the SE/SA mode. Get programming/Dial Tone.
• Make programming changes.
• Exit SE/SA mode.
• Terminate DISA call.
• In case none of the trunk lines have DISA facility and it is required to do remote programming, then follow
following steps:
• From the remote location, make a call to the system.

• Ask the person answering the call at the system end to keep you on hold by dialing Flash. When the
person at the system end dials ‘Flash’, you get hold on music.
• Ask the person at the system end to dial 1#91-SE Password. On completion of the command, he gets
confirmation tone. He can disconnect and carry on with the routine job.
• The person at the remote end gets DISA beeps.
• He dials #1 and gets the dial tone.
• He can enter the SE/SA mode from the remote end.
SE Mode
Dial 1#91-SE Password program the system. Dial ‘00’ exit from the programming mode (SE Mode).
When you have logged into SE Mode from DISA, to program #, you need to press # four times.
SA Mode
Dial 1#92-SA Password program the system. Dial 1#92 exit from SA mode.
Once the user is out of SE/SA Mode the user gets DISA beeps.

How it works
Following flow chart depicts the process:
Start
Dial and log-in as a DISA user

remotely. For more details,
please refer section DISA
User is in DISA idle state. Gets DISA login beeps
User dials command for entering

program mode followed by SE password
Is the No
password
correct ?
Error Tone
Yes
System gives programming tone
User dials any of the programming command
Is the
Yes command dialed No
correct and accepted
by the system ?
Confirmation tone Error tone
Exit
Program
Mode Wait for error
Wait for confirmation
tone to get over
tone to get over System gives dial tone
End

Room Monitor
What’s this?
This feature enables the DKP and Extended IP Phone Extension users to listen to the conversations taking place in
another location where a DKP/Extended IP Phone is present.
Room Monitor can be used to monitor activities on the Shop Floors / Manufacturing areas from another location.
• Use this feature in accordance with the local privacy laws.

How it works
• A is a supervisor in a Manufacturing unit.
• A’s room in on the second floor. The manufacturing area is on the ground floor.
• To keep track of the activities in the plant on the ground floor, there must be a DKP or an Extended IP
Phone at the place where the activities are to be monitored, and A’s extension must have higher “Priority”
than the extension at the monitored location.
• If there is a DKP or an Extended IP Phone at the desired location, A can activate Room Monitor.
A can activate Room Monitor only if the DKP/Extended IP Phone at the desired location is idle.
• When A activates Room Monitor, the microphone of the DKP/Extended IP Phone on the ground floor goes
Off-hook. A can now hear all the sounds taking place on the ground floor, without anyone present there
coming to know that they are being monitored.
• To end Room Monitor, A must disconnect.
• Room monitoring will be terminated on the DKP/Extended IP Phone on the ground floor, if someone lifts
the handset of this phone or if there is a call on this phone from another extension.
You can activate Room Monitor from any extension port type, but the extension being monitored must be a
DKP or an Extended IP Phone.
How to configure
To be able to use Room Monitor, extension users must have this feature enabled in the “Class of Service (COS)” in
the “Station Basic Feature Template” assigned to their extensions.

How to use
To enable Room Monitor on an extension,
• Press the DSS Key assigned to Room Monitor.
• Dial Extension Number to be monitored.

OR
• Dial 1073-Extension Number to be monitored.
For SLT Users
• Dial 1073-Extension Number to be monitored.

Response Mapping
What's this?
For calls to be established between a SIP and ISDN networks, the two networks must interoperate. The mapping of
ISDN causes of call failure or completion, to SIP codes and events, as well as mapping of SIP codes and events to
ISDN causes are vital aspects to attain interoperation between the networks. Similarly, mapping of system
disconnect/release events to SIP/ISDN networks is another important aspect.
The SARVAM UCS supports programmable event cause mapping. You can configure the following as per your
requirement:
• Disconnect/error cause for SIP to ISDN calling, that is, when the system receives a SIP disconnect/error
cause from remote SIP peer then you can select the code and event the system must send on ISDN to the
remote ISDN peer.
• Disconnect/error cause for ISDN to SIP calling, that is, when system receives the ISDN disconnect/release
cause then you can select the code and event the system must send as the SIP response to the remote
SIP Peer.
• Disconnects/release cause to be sent by the system to SIP network, that is, when the system disconnect/
release the call with SIP then you can select the code and event the system must send as the SIP
response to remote SIP Peer.
• Disconnects/release cause to be sent by the system to ISDN network, that is, when the system
disconnect/release the call with ISDN then you can select the code and event the system must send as the
ISDN cause to remote ISDN Peer.
Configuring Response Mapping using Jeeves

• Under Configuration, click Response Mapping.
• Click SIP to ISDN cause mapping.

• The SIP column contains the list of the SIP codes and events. For each SIP code and event you can select
the cause that you want the system to send to the remote ISDN peer.
• Click Submit.
• Click the ISDN to SIP cause mapping.
• The ISDN column contains the list of ISDN causes of call failure or completion.
• For each ISDN cause you can select the SIP code and event you want the system to send as the SIP
response to the remote SIP Peer.
• Click Submit.
Similarly, you can also map the system disconnect/release events to SIP/ISDN network. Click the respective link
System to SIP cause mapping or System to ISDN cause mapping to configure the mapping as per your
requirement.

Routing Group
What’s this?
Routing Group is a group of extensions used for landing incoming calls as a Trunk Landing Group, as Alarm
Notification Group, as Floor Service Group and as Department Group.
How it works
SARVAM UCS supports the formation of 96 Routing Groups. In each group you can have upto 32 members.
The member of a Routing Group can be Single Line Telephones (SLT), Digital Key Phones (DKP), SIP Extensions,
ISDN Terminals, Virtual Extensions, Voice Mail Auto Attendant and Outgoing Trunk Bundle Group.
These groups can be used:

• as Trunk Landing Groups to route incoming calls.
• as Alarm Notification Groups to server Alarm Notifications.
• as Floor Service Groups to provide Floor Service.
• as Department Groups to route incoming calls to a particular department.
This is how a Routing Group works,
• There is an incoming call on CO1.
• Routing Group 1 is assigned as the Trunk Landing Group for CO1. The Routing Group has DKP 2001,
2002 and 2003 as landing destinations.
• By default incoming calls will be placed on the members in rotation, that is first call on DKP 2001, second
call on 2002 and so on.
If you want incoming calls to be placed on DKP 2001 always, you must disable Rotation.
• By default, an incoming call will be placed on 2001. 2001 rings for the duration of the Ring Timer, if the call
is unanswered the system re-directs the call to 2002 and so on, till the call is answered.
If you want all the extensions to ring continuously till the call is answered by any member, you must enable
Continuous Ring. 2001 will continue to ring even as the system hunts for other extensions in the routing
group to land the call. If the call still remains unanswered, the system will return the call to 2001 once
again.
• In this way the system places the incoming calls on the member extensions in a Routing Group till the call
is answered.
If you have selected Voice Mail Auto Attendant or OTBG as members in a Routing Group, the parameters
Ring Timer and Continuous Ring are not applicable.

How to configure
Configuring Routing Groups using Jeeves

• Under Configuration, click Routing Group.
• Choose the Routing Group number (01-96) you want to use as routing group. In each routing group you
can program maximum 32 'members'.
• Now program the following parameters for the selected routing group:
• Name: You can assign a Name to the department group to facilitate identification. This name will
appear in the Dial by Name directory along with the department group number. The Name can be a
maximum of 18 characters.
• Rotation: Select the Rotation check box to enable rotation of calls in the routing group which has
multiple 'member' extensions. When enabled, each fresh call will land on the extension which is next to
the one that received the last call. This ensures equal distribution of incoming calls to all the
destinations within the routing group. The flag has no relevance if the routing group has only one
member extension.
• When member rejects the call, place the call again: If any SIP/DKP member extension rejects an
incoming call and the system again checks the routing group sequence, you can allow or restrict
placing the same call on this extension. Select the When member rejects the call, place the call again
check box, if you want the system to place the call again on the extension.
If this check box is cleared (disabled) and you have selected the Continuous check box, the extension
that rejects the call stops ringing while the system hunts for other extensions in the routing group to
land the call.
• Member Type: Select the 'Member Type'. You can select SLT, DKP, SIP, Virtual Extensions, ISDN
Terminal, OTBG or the Voice Mail Auto Attendant.
Configure only as many extensions as you want in the routing group and set the remaining Member
Types to 'None'.
For example: if you want to program only one extension in the routing group, set the Member Type in
the remaining columns (Member 02-Member 32) to 'None.'

• Port Number: Enter the software port number on which the SLT/DKP/SIP/Virtual Extension/ISDN
Terminal is connected.
If you have selected OTBG then enter the OTBG number here.
Port Type, select the VMAA Menu to assign to the respective routing group.
• Ring Timer(s): This timer defines the time for which the extension, on which the call lands, should ring.
By default, the ring timer is set to 015 seconds and can be changed.
• Continuous Ring: Select the Continuous Ring check box, if you want an extension to ring
continuously until the call is answered. The first extension will continue to ring even as the system
hunts for other extensions in the routing group to land the call. If the call still remains unanswered, the
system will return the call to the first extension once again. This parameter is not relevant, if there is
only one member extension in a routing group.
• Repeat the same steps to create another Routing Group.
• To route incoming calls on a trunk, you must assign a Routing Group in the Trunk Landing Group in a
Trunk Feature Template assigned to the trunk.
• To assign a Routing Group as a Trunk Landing Group, under “Configuring Trunks”, see “Trunk Feature
Template”.
• To assign a Routing Group as a Department Group, see “Department Call”.
• To assign a Routing Group as a Alarm Notification Group, under “Alarms”, see “How to configure”.
• To assign a Routing Group as a Floor Service Group, see “Floor Service”.
Configuring Routing Groups using a Telephone

• Enter SE Mode from a DKP/SLT.
To select the destination in the Routing Group, dial:

• 6502-2-Routing Group-Routing Group-Destination Index-Port Type-Port Number
• 6502-*-Number Index-Port Type-Port Number
Where,
Routing Group is from 01 to 96.
Port Type Meaning Port Number
00 Nonea 000
01 SLT 001-512

Port Type Meaning Port Number
02 DKP 001-128
16 OG Trunk Bundle Groupb 001-025
28 ISDN Terminal 01-64
34 SIP Extension 001-999
36 Virtual Extension 01-64
a. Parameters, Ring Timer and Continuous Ring are not applicable if

the SE selects None, OG Trunk Bundle Group as Member Type.
b. Same as per above note.
To program the time for which each station in the group should ring, dial:
• 6503-1-Routing Group-Member Index-Ring Timer
• 6503-2-Routing Group-Routing Group-Member Index-Ring Timer
• 6503-*-Member Index-Ring Timer
Where,
Ring Timer is from 001 to 255.
By default, Ring Timer is 015 seconds.
To program continuous or non-continuous ring for a station in the group, dial:

• 6504-1-Routing Group-Member Index-Flag
• 6504-2-Routing Group-Routing Group-Member Index-Flag
• 6504-*-Member Index-Flag
Where,
Flag Meaning
0 No station rings for the time set (Disable continuous)
1 The station rings till the call matures (Enable continuous)
To program rotation, dial:

• 6505-1-Routing Group-Rotation Method
• 6505-2-Routing Group-Routing Group-Rotation Method
• 6505-*-Rotation Method
Where,
Rotation Method Meaning
0 Fresh call lands on the first station within the group (disable continuous)
1 Fresh call lands on following the rotation method (enable continuous)
To default a routing group, dial:

• 6501-1-Routing Group
• 6501-2-Routing Group-Routing Group
• 6501-*

Where,
On issuing the command all routing groups will be set to default values as follows:
Member Index Port Type Port Number Ring Timer (Sec.) Continuous Ring Rotation
01 DKP 001 015 OFF ON
02 SLT 001 015 OFF ON
03 SLT 002 015 OFF ON
04-32 None 000 015 OFF ON
To clear the member of a Routing Group, dial:

• 6510-1-Routing Group
• 6510-2-Routing Group-Routing Group
• 6510-*
Where,
• Exit SE mode.

Selective Port Access
What’s this?
SARVAM UCS supports different extension and trunk port types. In the Selective Port Access feature, each port
type is assigned a Port Access Code. Extension users can access a particular port by dialing the Port Access Code
assigned to the Port and its Port Number.
How it works
• Extension user A wants to access a particular Mobile port, Mobile Port 1 to make a call. Extension A must
dial the Selective Port Access Feature Code, followed by the Port Type Code for Mobile ports and then dial
the Port Number.
• The following access codes are assigned to each Port Type:
Port Types Port Access Code Port Numbersa
SLT 01 001 to 240
DKP 02 001 to 096
CO 03 001 to 064
BRI 04 01 to 32
T1E1 05 01 to 08
E&M 06 001 to 032
Mobile 25 001 to 040
SIP Trunk 26 01 to 99
ISDN Terminal 28 01 to 64
SIP Extension 34 001 to 999
Magneto 29 001 to 016
Virtual Extension 36 01 to 64
Radio Ports 40 01 to 16
a.The number of ports mentioned here are for ETERNITY GENX. For details regarding
the number of ports supported for ETERNITY LENX/MENX, see “Technical Specifi-
cations - SARVAM UCS”.
Here, Extension A must dial 69-25-01, where 69 is the feature code for Selective Trunk Access, 25 is the
port access code for the Mobile Port, and 01 is the number of the Mobile Port which A wants to access.
Similarly, if Extension A wants to call SIP Extension 10, A can dial 69-34-010.
How to configure
To be able to use Selective Port Access, extension users must have this feature enabled in their “Class of Service
(COS)”.

How to use
To enable Selective Port Access on an extension,

• Press DSS key assigned to Selective Port Access code
• From the menu select the Port Type
• Enter the Port Number of the selected Port Type
OR
• Dial 69 / 89 (for users in USA)
• Dial the Port Type - Port Number
For SLT Users

• Dial 69-Port Type-Port Number
• Dial 89-Port Type-Port Number (for users in USA)

Security Settings
The feature Security Settings enables you to restrict unauthorized access to SARVAM UCS using FTP, Web,
Telnet, Third Party Auto Configuration as well as SIP Extension registration.
For FTP, Web, Telnet, Third Party Auto Configuration

When any user attempts to access FTP Server, Telnet Server, Web Server, Third Party Auto Configuration using
false credentials for 5 times consecutively within 10 minutes, SARVAM UCS blocks such IP Address for 10
minutes.
To allow access to FTP Server, Telnet Server, Web Server and Third Party Auto Configuration, to specific trusted
IP Address/es, you must configure them in the Trusted IP Address/es table. For instructions, see “How to
configure” below.
For SIP Extensions

When any user attempts to register as a SIP Extension using false credentials— Authentication ID or
Authentication Password and the authentication attempt fails for 10 times consecutively, SARVAM UCS blacklists
the IP Address and port used for registration. See “Black List IP Address - SIP Extensions” for more details.
You are recommended to configure the trusted IP Addresses in the Trusted IP Address/es table to avoid
blacklisting. For instructions, see “How to configure” below.
However, if any IP Address is already blacklisted, it will be stored in the Black List IP Address - SIP Extensions
table. To allow access to such blacklisted IP Address, you must remove it from the Black List IP Address - SIP
Extensions table manually.
How it works
• the Trusted IP Address/es table must be configured. You can configure a maximum of 25 addresses in this
table.
• determine the facilities you want to allow to each IP Address — FTP, Telnet, Web, Third Party Auto
Configuration, SIP Extension registration.
• with this table configured,
• access to FTP Server, Telnet Server, Web Server, Third Party Auto Configuration will be allowed only
to the configured Trusted IP Address/es.
• Trusted IP Addresses configured for the registration of the SIP Extension, will not be blacklisted.
The successful attempt to access SARVAM UCS using FTP, Web or Telnet, will be logged in the System Activity
Log.

When you change the jumper position on the CPU Card from BC to AB, the parameters Allow FTP Server
Access From, Allow Telnet Server Access From, Allow Web Server Access From, will be set to default.
The Jumper number depends on the model of SARVAM UCS. For details, see “System Security”.
How to configure
• Under Configuration, click Security Settings.
Trusted IPv4 Address/es

• configure the IP Address/es and their respective Subnet Mask in the Trusted IPv4 Address/es table.
• enable the Allow FTP Server check box in the Trusted IPv4 Address/es table.
• In Allow TELNET Server Access, you can select Don’t Allow, All IP Address/es or Only Trusted IP
Address/es option. By default, Don’t Allow is selected.

• enable the Allow Telnet Server check box in the Trusted IPv4 Address/es table.
es option. By default, Don’t Allow is selected.
If you want to allow access to the Web Server from all IP Addresses, select All IP Address/es option.
• enable the Allow Web Server check box in the Trusted IPv4 Address/es table.
• In Allow Third Party Auto Configuration, you can select Don’t Allow, All IP Address/es or Only Trusted
IP Address/es option. By default, Don’t Allow is selected.
If you want to allow the third party configuration from all IP Addresses, select All IP Address/es option.
If you want to allow Third Party Auto Configuration from specific IP Addresses only:
• enable the Allow Third Party Auto Configuration check box in the Trusted IPv4 Address/es table.
• By default, the Allow SIP Extensions Registration check box is disabled. If you want to allow SIP
extensions registration from the WAN Port, select the check box.
• If you allow SIP Extension Registration, you can Black List SIP Extension IP Address:Port on
multiple Authentication Failure Attempts.
By default, All IP Address/es option is selected. If you do not want to include the Trusted IP
Addresses, select Except Trusted IP Address/es option.
If you select Except Trusted IP Address/es,
• configure the Trusted IP Address/es and their respective Subnet Mask in the Trusted IPv4
Address/es table.
• enable the Black List SIP Extension IP Address:Port except check box for those IP Addresses
which you do not want to blacklist.
Trusted IPv6 Address/es


• configure the IPv6 Address/es and their respective Prefix Length in the Trusted IPv6 Address/es
table. The Prefix Length is a decimal value that indicates how many of the high-order contiguous bits of
the address comprise the prefix (the network portion of the address).
• enable the Allow FTP Server check box in the Trusted IPv6 Address/es table.
• In Allow TELNETServer Access, you can select Don’t Allow, All IP Address/es or Only Trusted IP
Address/es. By default, Don’t Allow is selected.
• enable the Allow Telnet Server check box in the Trusted IPv6 Address/es table.
es. By default, Don’t Allow is selected.
If you want to allow access to the Web Server from all IP Addresses, select All IP Address/es option.
• enable the Allow Web Server check box in the Trusted IPv6 Address/es table.
• In Allow Third Party Auto Configuration, you can select Don’t Allow, All IP Address/es or Only Trusted
IP Address/es. By default, Don’t Allow is selected.
If you want to allow third party auto configuration from all IP Addresses, select All IP Address/es option.
If you want to allow Third Party Auto Configuration from specific IP Addresses only:
• enable the Allow Third Party Auto Configuration check box in the Trusted IPv6 Address/es table.
• By default, the Allow SIP Extensions Registration check box is disabled. If you want to allow SIP
extensions registration from the WAN Port, select the check box.
• If you allow SIP Extension Registration, you can Black List SIP Extension IP Address:Port on
multiple Authentication Failure Attempts.
By default, All IP Address/es option is selected. If you do not want to include the Trusted IP
Addresses, select Except Trusted IP Address/es option.
If you select Except Trusted IP Address/es,

table. The Prefix Length is a decimal value that indicates how many of the high-order contiguous
bits of the address comprise the prefix (the network portion of the address).
• enable the Black List SIP Extension IP Address:Port except check box for those IP Addresses
which you do not want to blacklist.

Self Ring Test
What’s this?
You can use Self Ring Test to check the functioning of your own extension phone. Self Ring Test allows you to call
your own extension. You can check the ringing volume of your extension phone.
How to use

• Go OFF-Hook.
• Press DSS Key assigned to Self Ring Test.
OR
• Dial 1057.
• Go ON-Hook.
• Your phone rings.
• Go OFF-Hook to stop the ring.
• Go ON-Hook.
For SLT Users

• Lift the receiver.
• Dial 1057.
• Replace receiver.
• Your phone rings.
• Lift the receiver to stop the ring.
• Replace the receiver.

Shared Call Appearance
What is this?
Shared Call Appearance (SCA) allows Open SIP Phones that are registered with SARVAM UCS at different
locations with the same address/number, to get notification on call states of the call appearance(s) shared by them.
Whenever a call is made or received from a shared call appearance, SARVAM UCS sends each SIP Phone
sharing the call appearance(s), a notification on the state of the call appearance. Through these notifications, each
user sharing the same address/number can know the current state of the call appearance and act accordingly.
SARVAM UCS supports and displays the following call states for a shared call appearance on the SIP phones:
State Meaning
Idle When the call appearance is free.
When the call appearance is been seized from any User binding using the line-
Seized
seize subscription.
When the User has generated a call using the call appearance and the called
Progressing
destination is ringing.
Ringing When the call is received on the User at the call appearance.
Active When the call of the User at the call appearance is in matured state.
When the call at the call appearance of the User has been put on public hold
Held
from the User binding.
When the call at the call appearance of the User has been put on private hold
Held-private
from the User binding.
• SARVAM UCS supports SCA as per the Broadsoft SCA feature Specifications.
• SARVAM UCS supports this feature only on Standard IP Phones.
• Standard IP Phones may differ in the type of indication (LED color and cadence, text message display)
they provide for the Call States. Refer to the manufacturer’s documentation for SCA indication
supported on the phones.
• Calls put on 'Consultation Hold' from Location -1 of SIP Extension (binding -1), the SIP Extension
registered at Location-2/3 (another binding) can not retrieve that call.
How it works
SARVAM UCS supports up to 10 call appearances on SIP extensions. The number of call appearances that will be
shared by the SIP Phones will depend on the number of call appearances you have configured for the SIP
extension.
To provide SCA to the Standard IP Phones registered with the same ID, the Shared Call Appearance flag must be
enabled in the SIP Extension Settings of SARVAM UCS.
On the Standard IP Phones, make sure you have configured as many call appearances as allowed on the SIP
extension by SARVAM UCS and configure the corresponding number of CA keys.

Here is an example of how Shared Call Appearance works:
• A, B and C are Open SIP phones registered with SARVAM UCS at three different locations with the same
SIP ID, 602.
• Two Call Appearances are configured for A, B and C and Shared Call Appearance is enabled for SIP ID
602.
• Two keys are assigned for the two Call Appearances on A, B and C.
• There is an incoming call for SIP ID 602.
• SARVAM UCS presents the incoming call on a free call appearance, Call Appearance1, as ‘Ringing’.
• A, B and C get the same alert, ‘Ringing’ simultaneously on the same call appearance, Call Appearance 1.
• A answers the call first and gets connected to it on Call Appearance 1.
• A, B and C get indication of the current call state as ‘Active’ on the same call appearance. B and C will not
be able to make or receive any new call from this busy call appearance. However, they can make or
receive a new call from the other free call appearance, Call Appearance 2.
• When A makes an outgoing call using Call Appearance 2, SARVAM UCS presents the state of the same
call appearance on A, B and C as ‘Seized’, then as ‘Progressing’ when the destination number starts
ringing, and then as ‘Active’, when the call is answered.
• B and C will not be able to make or receive a call from Call Appearance 2.
• A can put an ‘Active’ call on public Hold or on private Hold. When A puts an ‘Active’ call on public hold,
SARVAM UCS presents the state of this call as ‘Held’ to A, B and C. Now, B or C can retrieve the call by
pressing the corresponding call appearance key.
• When A puts an ‘Active’ call on private Hold, SARVAM UCS presents the state of this call as ‘private-Held
to A, B and C. Only A can retrieve the call. Thus, if a call is put on private hold (Held-private), it can be
retrieved only from the IP Phone that put it on hold.
How to configure
You can provide this feature only to Standard IP Phones you have registered with SARVAM UCS. To provide this
feature,
• In SARVAM UCS, you must enable the Shared Call Appearance check box on the SIP Extension
Settings.For instructions, see “Configuring SIP Extensions”.
• On the Standard IP phones,

• configure as many call appearances as allowed on the SIP extension.
• For each shared call appearance, configure a corresponding call appearance key on the SIP Phone.
For instructions refer to the manufacturer’s documentation (Installation Guide/User Guide) for the
respective SIP Phones.

SIM Card Balance and Recharging
What's this?
The SARVAM UCS supports Balance Inquiry and Recharging of the SIM Card installed in its Mobile Ports276.
How to use
To be able to use this feature, first collect the following information from your Network Operator:
• Balance Inquiry Number: This is the number provided by the Network Operator to the subscribers to
check Balance. Different Network Operators have different numbers. For example, the Balance Inquiry
number of Vodafone is *141#.
• Recharging Service Number: This is the number provided by the Network Operators to their subscribers
for Recharging Service. Different Network Operators have different numbers for Recharging Service. For
example, the Recharging Service Number of Vodafone is *140*.
This feature can be used from the SE Mode only on Jeeves.
SIM Card Balance Inquiry and Recharging using Jeeves

276. SARVAM UCS supports Unstructured Supplementary Service Data (USSD), the standard for transmitting information over CSM
signaling channels and a commonly used method to query the available balance and other similar information in pre-paid GSM
services.

• Click SIM Balance Recharge to open the page.
• All the mobile ports configured in the system will appear on this page, by their Names you programmed
when configuring the mobile trunk ports.
If you have not programmed any name for a port, the Name field for that port will appear blank.
• Go to the Mobile Port on which you want to request SIM Balance/Recharge.
• Select the Balance Inquiry/Recharge check box to enable this feature.
Balance Inquiry
• To make Balance Inquiry,
• Click the Request option button under Balance Inquiry, for all those Mobile Ports for which you want
to request Balance Inquiry.
• Enter the Balance Inquiry Number provided by the Network Operator whose SIM Card you have
installed in the Mobile Port.
A maximum of 16 digits are allowed. The valid digits for Balance Inquiry number are any digits from 0 to
9 and the characters * and #
• Click Submit.

Recharging the SIM
• To recharge the SIM,
• Click the Request option button under Recharge, for all those Mobile Ports for which you want to make
a recharge request.
• Number: Enter the Recharging Service Number provided by the Network Operator in this field.
A maximum of 16 digits are allowed. The valid digits for Recharging Service number are any digits from
0 to 9 and the characters * and #
• PIN: Enter the PIN number which is printed on the Recharge Voucher/Coupon. Your Recharge PIN
number may consists of a maximum of 20 digits.
The valid digits for PIN number are any digits from 0 to 9 and the characters * and #.
Make sure you enter the digits and characters of the Recharge PIN number exactly as given on the
Recharge Voucher/Coupon.
• Click Submit.
• Click Refresh at the bottom of the page.
• In USSD Reply the response received from the GSM network (including possible error messages) will
be displayed. When the USSD Reply is received from the network, it will appear with the Date and
Time stamp of SARVAM UCS in this field.
• For each port that you send a Balance Inquiry/Recharge Request, you will get this USSD-Reply: "Please
wait, processing the request. Refresh the page to see the current status."
• Click Refresh bottom of the page.
• The response received from the GSM network (including possible error messages) will be displayed under
USSD-Reply. When the USSD Reply is received from the network, it will appear with the Date and Time
stamp of SARVAM UCS in this field.
• For each Mobile Port (SIM Card) at a time you can either request Balance Inquiry or Recharge the SIM
Card.
However, you can send Balance Inquiry/Recharge request for all the Mobile Ports available in the
system.
• During Balance Inquiry/Recharge-Request, the status of the Mobile port will be 'busy'. It will become
idle only after the USSD response is received from the GSM network.
• The SARVAM UCS will clear the USSD Reply after system restart. So each time you open the 'SIM
Balance and Recharge' page after system restart, the USSD Reply box will be blank.

SMS Gateway
What's this?
SARVAM UCS offers the SMS Gateway interface to extend UC functionality.
The SMS Gateway feature of SARVAM UCS enables you to send/receive messages to/from individuals, selective
groups or masses using the Mobile Port of SARVAM UCS.
SARVAM UCS allows you to register multiple SMPP Clients (Software Applications used for sending/receiving
messages) with SARVAM UCS. SARVAM UCS functions as an SMPP Server. These Clients can send/receive
messages using the Mobile port/s of SARVAM UCS.
The messages are sent using the Short Message Peer to Peer Protocol (SMPP Version 3.4). Using this encoding,
it is possible to send up to 160 7-bit characters in one message, in the GSM network.
To use this feature you must purchase the SMS Gateway License. Refer to the topic “License
Management” to know more.
How it works
• you must have the SMS Gateway license.
• you must have the SMPP Clients installed on a computer connected in the same LAN as SARVAM UCS.
• you must configure the required parameters in the SMPP Clients to register itself with SARVAM UCS
• you must define the Mobile port through which the messages are to be sent/received in the SMPP Clients.
• you must configure the SMPP Client parameters in SARVAM UCS.
This is how the SARVAM UCS SMS Gateway works,
• On successful registration of the SMPP Client with SARVAM UCS, a binding is established between the
SMPP Client and SARVAM UCS.
• The SMPP Client can bind itself with SARVAM UCS as a Receiver, Transmitter or Transceiver.
• As a Transmitter, the SMPP Client will only be able to send messages using the Mobile port of
SARVAM UCS.
• As a Receiver, the SMPP Client will only be able to receive messages from the Mobile port of SARVAM
UCS.
• As a Transceiver, the SMPP Client will be able to send and receive messages from the Mobile port of
SARVAM UCS.

• The SMPP Client sends all the information required to send an SMS— the message content, the
destination mobile number and the mobile port through which the message is to be sent to SARVAM
UCS— in the Protocol Data Unit (PDU) format.
• The message will be sent to the destination number through the Mobile port, if it is idle or in speech. The
message will be sent using the SMS Center Number configured for that port.
• If the Mobile Port is any other state the SMS will not be sent.
• If the Mobile port is disabled/unregistered, the messages will not be sent.
• SARVAM UCS will accept a new SMS sent by the SMPP Client for the same Mobile port only after the
preceding SMS is successfully sent.
• All incoming SMS on the Mobile port will be sent to the SMPP Client.
• After the SMS is sent to the SMPP Client the same will be deleted from the SIM card of the Mobile port.
How to Configure
For the SMS Gateway you need to configure the following parameters,
• Under Configuration, click SMS Gateway.
• Click General Parameters. The SMPP Server - Parameters page opens.
• SMPP Server: By default, the SMPP Server is disabled. To use the SMS Gateway feature, select
Enable.

• SMPP Server Port: Enter the SMPP Server’s listening port. The valid range is 1025 to 65535. Default:
SMPP Server Port is 2775.
• Enquiry-Link Timeout (seconds): The SMPP Client sends the Enquiry-Link Requests to the SMPP
Server at regular intervals to refresh its binding with the Server. The system re-loads this timer
(default:120 seconds) every time it receives the request from the Client. If no response is received from
the SMPP Client before the expiry of this timer, the Server considers the Client as disconnected. The
valid range of the timer is 005 to 999. Default: 120 seconds.
• SMPP Server Debug: To monitor the events and processes of the SMPP Server, for trouble shooting
and identifying faults and errors, select the SMPP Server Debug check box. The debug messages are
sent to the remote Syslog Server. For detailed instructions on how to configure the Destination Port,
Syslog Server IP Address and Port, see “System Debug”.
• You can register multiple SMPP Clients with SARVAM UCS. Against each SMPP Client configure the
following parameters:
The maximum number of SMPP Clients which you can register is equal to the maximum number of Mobile
ports supported by SARVAM UCS, that is 64.
• SMPP Client System ID: Enter the ID you want the Server to use to authenticate the Client. A
maximum of 16 characters, including all ASCII characters are allowed.
• SMPP Client Password: Enter the password you want the Server to use to authenticate the Client.
To avoid unauthorized access, make sure the password is strong and is kept confidential. The
Password can be of maximum of 9 characters, including all ASCII characters are allowed.
Make sure you:
• Do not assign the same SMPP Client System ID and Password to multiple SMPP Clients.
• Configure the same SMPP Client System ID and Password in the SMPP Client. This ID is sent to the
Server in the connection request made by the Client.
• Mobile Port: Select the Mobile Port using which the SMS is to be sent from/to the SMPP Client.
The SARVAM UCS supports a maximum of 64 Mobile ports277.
Make sure you do not assign the same Mobile Port to multiple SMPP Clients.
• Debug: If you want to monitor the events and processes of the SMPP Client, for trouble shooting
and identifying faults and errors, select the Debug check box. The debug messages are sent to the
remote Syslog Server. For detailed instructions, for configuring the Destination Port, Syslog Server
IP Address and Port, see “System Debug”.
Viewing SMPP Server Status

You can view the status of SMPP Server on Jeeves only. To do this,
277. Depends on the model you have. Please refer the Appendix for an overview of the system resources and maximum expansion
capacity.

• Under Configuration, click SMS Gateway.
• Click Status.
• The following parameters are displayed for each SMPP Client registered with SARVAM UCS:
• System ID: This is the ID received in the SMPP Client’s connection request.
• Status: This field displays any one of the following:
Status Description
Not Connected When the SMPP Server does not receive any
request from the SMPP Client.
When the SMPP Server receives a request from

Connected the SMPP Client and it accepts it. A connection is
established between the Server and the Client.
• Binding Type: This field displays the type of SMPP Client Binding, that is Transmitter, Receiver or
Transceiver.
• Address: This field displays the SMPP Client’s IP Address and Port.

SMS on No Reply
What's this?
Among the various UC features offered by SARVAM UCS, SMS on No Reply is the most useful business UC
feature.
SMS on No Reply feature of SARVAM UCS is used to send an SMS to the mobile user, when an extension user
calls the mobile user and the mobile user doesn't answer the call. The SMS that is sent to the mobile user contains
the extension user’s contact information and the message to be conveyed in brief, so that the mobile user can call
back if necessary.
How it Works
• you must have installed a GSM Card.

• you must have the SMS Server license to route the SMS through the Mobile ports. For instructions, see
“License Management”.
• you must enable the Send SMS check box in “Mobile Port Parameters”.
• you must enable SMS on No Reply in the Station Advanced Feature Template assigned to the
extension user. For instructions, see “How to Configure”.
• you must configure the SMS on No Reply parameters. For instructions, see “How to Configure”.
This is how the SMS on No Reply works,
• An extension user A calls an external mobile number 9898906336. This call is routed using the Mobile Port
of the SARVAM UCS.
• The call routed through the Mobile Port will be considered as no reply only if ‘No Reply’ event is received
from the service provider.
• The SMS on No Reply will not be applicable for calls routed through the CO Port.
• If the call is routed through the SIP it will be considered as no reply when cause 408 is received from
the network.
• If the call is routed through the BRI/ T1E1 it will be considered as no reply when cause 18 (No User
Response) or cause 19 ( No Answer from User) is received from the network.
• After the No reply event is received from the service provider, the system matches the mobile users
number with the numbers configured in the Number list for sending SMS.
• If a match is found, the system sends the SMS to the mobile user. The SMS contains the Name and/or
Number of the extension user A as well as the message. This message can be customized as per
extension user requirement.

• In this way the mobile user can callback the extension user, if required.
How to Configure
For the SMS on No Reply feature to work, you need to configure the following parameters:
• SMS for OG Call - No Reply in the Station Advanced Feature Template

• Number list for sending SMS for SMS on No Reply.
• Send SMS option for routing the SMS through the Mobile ports.
• SMS Text message to be sent as SMS to mobile users.
Configuring SMS on No Reply in the Station Advanced Feature Template

The default Station Advanced Feature Template 01 is assigned to all extension users of the SARVAM UCS. In this
template SMS on No Reply is disabled.
Decide which extensions are to be allowed 'SMS on No Reply'. If you want to allow SMS on No Reply to all
extensions, retain the default Station Advanced Feature Template 01 and configure the SMS on No Reply
parameters. However, if you want to allow SMS on No Reply only to selected extensions, select another Station
Advanced Feature Template and configure the SMS on No Reply parameters in that template.
Now, to assign the template to selected extensions, follow these steps:
1. Prepare a Station Advanced Feature Template with SMS on No Reply.

2. Assign this newly prepared template to the desired extensions.
Refer the topics “Station Advanced Feature Template” for detailed programming instructions on how to customize a
Station Advanced Feature Template, configure the SMS on No Reply and how to apply this template on
extensions.
Configuring SMS on No Reply Parameters
• Under Configuration, click SMS on No Reply.
• Configure the Number List for sending SMS. In this list enter the number strings that you expect the
callers to dial. By default, Number List 16 is assigned to SMS on No Reply.

Refer to the topic “Number Lists” to know more, and for configuration instructions.
• Select the desired option in Send SMS to route the SMS. You can select Using Fixed Port or Based on
LCR Table. See “Fixed Port Routing (SMS Server)” and “Least Cost Routing” for instructions.
• You can customized the SMS Text as per your requirements. The default text message is Hi, Greetings
from Matrixcomsec Pvt. Ltd.<NAME><EXTENSION No.> tried to call you from Matrixcomsec Pvt.
Ltd., to call back dial "Number for call back".
Here, in the SMS that is sent the NAME will be replaced with the name of the calling extension user and
EXTENSION No with the extension number assigned to the calling user. Make sure that the NAME and
EXTENSION No are written in the same format as displayed in the default message. The callback number
can be mentioned as text, if required or it can be removed.
• Click Submit.

SMTP Settings
The SMTP Settings must be configured if you wish to send mails for VMS applications and/or register the SMS
Server as an SMTP Client.You can configure different account for each application. The systems allows you to
configure a maximum of three accounts.
The Unified Messaging Functionality of SARVAM UCS includes using SMTP to send emails for the following
functions:
• send notification to the extension users about the arrival of new messages (with/without Voice Mail
Attachment).
• send notification to the extension users about the memory usage status of their mailbox.
• memory usage notification to SE.
• send mails from SMTP Clients while using the SMS Server Application.
For email transmission, you must:

• configure the parameter Message Wait Notification via Email278 in the VMS settings of the extension. For
instructions, see “Email Based Notification”.
• configure the parameters VMS E-mail Notification and Use SMTP Account in VMS General Parameters.
For instructions, see “Configuring VMS General Parameters”.
• configure the Mail Settings in SMS Server, see “SMTP Configuration”.
• configure the SMTP Accounts and its parameters.
To configure SMTP Settings using Jeeves,
• Under Configuration, click SMTP Settings to open the page.
• By default no account is configured. You can add a maximum of three accounts. Click Add Account and
then configure the following parameters:
278. You can also have the new voice message mailed as an attachment with the message wait notification.

• Configure the Account Name. This will help to identify the application for which the account is being
used.
• Configure the SMTP Server Address and SMTP Server Port. This is the Server’s IP Address and
Port number that will be used to send outgoing mails. Both IPv4 and IPv6 addresses are supported. If
port is not programmed, the default port value 25, will be used. Valid Port range: 25;465;587;1025 to
65535. The Server Address can be a maximum of 40 characters.
• Configure the Email ID for the account registered with the Email Server, as provided by your network
administrator. This Email ID will appear to the recipient as the originator of the email (that is in the
FROM field). The Email ID you configure may consist of a maximum of 64 characters. Default: Blank.
• If your Email Server uses authentication, select the Require Authentication check box. Default:
Disabled. If you have enabled authentication, you must also configure the User ID and the Password.
• If you have enabled authentication, configure the User ID and the Authentication Password as
provided to you by your network administrator. The User ID may consist of a maximum of 40 characters
and the Password can be a maximum of 24 characters. Default: Blank.
• To transport all data in a secure manner, select Enable Secure Socket Layer (SSL) check box. All the
data to the Email Server will be transported over secure layer. Default: Disabled.
• Configure the Display Name. This name will be displayed to the mail recipient. You can configure a
maximum of 24 characters. Only ASCII characters are allowed. Default: Blank.
• In Connection Timeout Interval configure the time duration for which you want the system to wait for
a response from the SMTP server. You may change the Connection Timeout Interval timer, if required.
The range of Connection Timeout Interval timer is 01 to 99 seconds. By default, it is set to 60 seconds.
• The SMTP Application used by Application displays the name of the application/s — SMS Server,
VMS Notification — which is using the account. It will be blank if the account is not being used by any
application.
• Click Submit to save SMTP settings.
Test SMTP Settings
• Click the 'Test Account’ button to check if the SMTP Parameters have been configured correctly.
When you click this button, the alert message will appear: "Testing SMTP can take up to 99 seconds.
Would you like to continue?" Click 'OK' button.
The Test Result will be displayed in 'Test Status' field.
• Test Status: Any one of the results listed below may appear in this field:
"SMTP Server Connection Not Established" When connection to SMTP server fails due to any
reason.
"Login to SMTP Server Failed" When connection to SMTP server is established

but login to SMTP server fails due to any reason.

"Sending Test Mail Failed" When connection to SMTP has been established
successfully but there is no acknowledgment for
the test mail sent.
"Test Mail Sent Successfully" When acknowledgment for the test mail sent to
SMTP server received successfully.
Already Test is running When the test is already being processed for
another SMTP account.
Timeout When the test is in process and the Connection

Timeout Interval expires.
Email Failure Errors
When message sending fails, the events will be logged into the System Fault Log with a specific code. For more
information, see “System Fault Log”.

Simple Network Management Protocol (SNMP)
What's this?
Simple Network Management Protocol (SNMP) is an application-layer protocol used for exchanging management
information between network devices. Using SNMP, you can manage and monitor network elements, audit network
usage, detect network faults or inappropriate network access.
The SNMP architecture consists of:
• An SNMP Agent is a program that is bundled within the managed device. SNMP agent allows a managed
device to collect the Management Information Base from the device and make it available to the SNMP
Manager on request. It receives SNMP requests and generates SNMP responses or notifications (traps/
informs). The SNMP Agents are SNMP Servers. Here, it is SARVAM UCS.
• SNMP Manager, usually the Network Management Station. The manager communicates with multiple
SNMP Agents implemented in the network. It generates SNMP requests and receives SNMP responses
and notifications (traps/informs). The SNMP Manager is an SNMP Client. Here, it is the SNMP Manager
installed in the PC.
• Managed device or the network element is a part of the network that requires some form of monitoring
and management. For example, switch, routers, servers.
• Management Information Base is the commonly shared database between the Agent and the Manager.
This information is known as managed objects. These managed objects are defined in MIB (Management
Information Base) module.Any sort of status or information that can be accessed by the Manager is
defined in a MIB.
SNMP uses UDP (User Datagram Protocol) as the transport protocol for passing information between Managers
and Agents. By default, the Agent listens on UDP port 161 for requests from Manager and the Manager listens on
UDP port 162 for notification from Agent.
How to Configure
• Click the Maintenance link.

• Click the SNMP link.
MIB Files
If the Manager needs to access any status or information about the client, the Manager must download and install
the MIB files on the local disk.
To download these files,
• Click the Click to download MIB button.
• You will get a prompt with the option to open the file or save the file to a location. Save the file on the local
disk.
SNMP Settings
• Click SNMP Settings to expand.
• Select the Enable SNMP? check box. Default: Disabled.
• Configure the SNMP Listening Port. Valid Range:161, 1025-65535. Default: 161.
• Select the SNMP Version as supported by your SNMP Manager. You can select from:
• SNMPv1

• SNMPv2c
• SNMPv3
For enhanced security, you must select SNMPv3.
• Configure the System Name. When there are multiple devices connected in the same network, the
name configured helps to identify the SARVAM UCS within the network. The System Name can be a
• Configure the System Contact. It is the name and number of the person to be contacted, in case of
notification. The System Contact can be of a maximum of 40 characters. Default: Blank.
• Configure the System Location. This is the physical location of SARVAM UCS. This information is
helpful to the administrator. The System Location may consist of a maximum of 40 characters. Default:
Blank.
• Configure Allow access to SNMP Manager from to allow/restrict accessibility to the SNMP Manager.
You can select Any IP Address or Specific IP Addresses. Both IPV4 and IPv6 addresses are
supported.
If you want to restrict the accessibility of the SNMP Manager, select Specific IP Addresses, configure
IP Addresses 1 to 5. Default: Blank.
• If SNMP version is set as SNMPv1 or SNMPv2c, configure Community Name.
Community Name identifies the SNMP community in which the sender and recipient of the message
are located. It enables communication between SARVAM UCS and the Manager. The Community
Name can be a maximum of 40 characters. Default: Blank. To avoid unauthorized access, we
recommend you to assign a strong Community Name.
• If SNMP version is set as SNMPv3, the System’s Engine ID is displayed. This is a unique
identification of the system. It is a hexadecimal field with length of 22 characters. The ID consists of:
• Enterprise Number (800086df03 which is fixed)
• MAC Address of the system (MAC address of Ethernet (LAN/WAN) Port)
Security Settings
• If SNMP version is set as SNMPv3, click Security Settings to expand and configure the following.
• Enter the User Name. The User Name can be a maximum of 40 characters. User Name will be used
for authentication and privacy in SNMPV3.

• Select the appropriate Security Type as per your requirement. Security Type defines the level of
security.
• When Authentication and Privacy are not required, select No Authentication-No Privacy
• When only Authentication is required, select Authentication without Privacy. Incoming SNMP
Messages will require authentication.
If you select this method, select the Authentication Algorithm as MD5 or SHA. Default: MD5.
In the Authentication Password, enter a password of your choice as Authentication Password for
the User Name you have assigned.To avoid unauthorized access, we recommend you to use a
strong password and make sure it is kept confidential. The Authentication Password must be a
minimum of 8 characters and may have upto 24 characters. Default: Blank.
• When both Authentication and Privacy are required, select Authentication with Privacy. Incoming
SNMP Message will require authentication and these messages will be encrypted, which will be
decrypted at the receivers end only.
If you select this method, select the Authentication Type as MD5 or SHA. Default: MD5.
Enter Authentication Password for the User Name you have assigned. To avoid unauthorized
access, we recommend you to use a strong password and make sure it is kept confidential. The
Authentication Password must be a minimum of 8 characters and may have upto 24 characters.
Default: Blank.
Select the Privacy Algorithm as DES or AES. Default: DES.
Enter the Privacy Password of your choice.We recommend you to use a strong password and
make sure it is kept confidential. The Privacy Password must be a minimum of 8 characters and
may have upto 24 characters. Default: Blank.

Notification Settings
• Click Notification Settings to expand.
If SNMP version is set as SNMPv1, configure the following parameters.
• Select Enable Notification check box, if you want SARVAM UCS to generate Trap message for an
error.
• You must configure the Notification Destination IP Address and Notification Destination Port
of the Manager or of any other device where you want to receive the trap messages.
SARVAM UCS will send the notification (error message) to the destination configured.
Valid range of the Notification Destination Port is 162 or 0-65535. Default port is 162.
• Click Submit button to save the settings.
If SNMP version is set as SNMPv2c or SNMPv3, configure the following parameters.
• Select Enable Notification check box, if you want SARVAM UCS to generate Trap or Inform message
for an error.
• Select the Notification Type. You may select Trap or Inform.
If you want the system to send notification message without acknowledgment, select Trap.
If you want the system to send notification message with acknowledgment, select Inform.
• If you select Inform as the Notification Type, you must configure Retry Attempts and Retry Interval.
If acknowledgment is not received from the Manager for the notification sent, the system will keep
retransmitting the message for the number of attempts you have configured as the Retry Attempts.
Default: 3.
The system will retransmit the messages at regular time intervals you have configured as Retry
Interval. Default: 10 seconds.
• You must configure the Notification Destination IP Address and Notification Destination Port
of the Manager or of any other device where you want to receive the trap messages.
SARVAM UCS will send the notification (error message) to the destination configured.
Valid range of the Notification Destination Port is 162 or 0-65535. Default port is 162.

• Click Submit button to save the settings.
• Click the Send Test Notification button, to send a test notification message.
Notification Filters
By default, you get notifications of errors, information and warnings for events related to the Application, Network,
Extension, Trunk Ports —CO, Mobile, SIP Trunks, T1E1and BRI, SMS Server/Gateway, Voice Mail System and
VoIP. Refer to the table at the end of this topic for the event list. You can choose the type of notification you want by
setting the notification filters.
To set the filters,
• Click Notification Filters to expand.
• By default, all the filters are enabled. To disable any filter, clear the respective check box.
The List of Events for which you will receive notifications is presented in the following table.
Application
Errors Warning Information
System Soft Reboot - User has

Card absent requested to restart the system System boots or starts
using SMS
Unlicensed Slot, Card Inactive:

Slot=xxa

Loading default configuration -

User requested to default the
HPI Queue Fail Card present
configuration of the system from
Web Jeeves
Loading default configuration -

User requested to default the SE Login from Web JEEVES -
Slave DSP Alive Error
configuration of the system using Logged in successfully
command
SA Login from Web JEEVES -

DSP Frame Sync Error USB Memory 100% Full
Logged in successfully
Call drop between CO and SE Login from Extension - Logged

DSP Drop Error
Extension due to fake call in successfully
SA Login from Extension - Logged

DSP Time Out Error Demo period expired
in successfully
Black List HTTP Auto Provisioning

DSP DMS Address Error Emergency Number dialing
IP Address
Temperature threshold level Emergency call reporting

Black List SE/SA/FDU Login
triggers Acknowledge
Temperature threshold level

Temperature warning level triggers Call Budget is exhausted on trunk
triggers
Temperature warning level triggers SMDR All IC Calls deleted
SMDR All OG Calls deleted
SMDR All Internal Calls deleted
USB Memory 80% Full
Test Notification
a. xx = Slot Number
Network
Ethernet - LAN Port Link Down Ethernet - LAN Port Link Up
Ethernet - WAN Port Link Down Ethernet - WAN Port Link Up
LAN Port and WAN Port IP Address IP Address of the Master Ethernet
are in same subnet/IP Conflict - LAN Port
IP Address of the Master Ethernet

- WAN Port
Extension
DKP Absent DKP Present
SIP Extension Unregistered SIP Extension Registered

CO
CO Port Down CO Port Up
BRI and T1E1
BRI Port Layer-1 Down BRI Port Layer-1 Up
BRI Port Layer-2 Down BRI Port Layer-2 Up
T1E1 Port Layer-1 Down T1E1 Port Layer-1 Up
T1E1 Port Layer-2 Down T1E1 Port Layer-2 Up
Mobile
SIM Absent SIM PUK Required SIM Present
Mobile Port Unregistered Mobile Port Registered
SIM PIN Wrong
SIP Trunk
SIP Trunk Registration Fail -

SIP Trunk - Active
Invalid SIP Configuration

Invalid Authentication Password

Expiry of Registration Timer
SIP Trunk - Unregistered and go to

registering state for Registrar or
Outbound Proxy
SIP Trunk - Disabled
VMS
Personal Mailbox full for extension
VoIP
SIP Stack Construction Fail - VoIP Application Restart - Due to

Authentication Fail change in Network Parameters

SIP Stack Construction Fail - VoIP Application Restart - Due to

Invalid IP Address change in SIP Parameters
Black List IP Address for SIP

VOPP Program Download Fail
Extension
VOPP SYNC Fail
VOPP failed to open Kernel file
VOPP failed to boot

Simple Network Time Protocol - SNTP
What’s this?
SARVAM UCS has its own Real Time Clock (RTC) to store date and time. When you select the Region, the RTC
parameters are set automatically.
However, the RTC can drift over a long period. So, you may check and reset the RTC values at regular intervals to
correct this drift. You can set the RTC manually (see “Real Time Clock (RTC)”) or synchronize it with any SNTP
Server in the Public Network. To synchronize time with the SNTP Server, SARVAM UCS supports SNTP Client.
How to Configure
To use SNTP for synchronizing with the Real Time Clock,
• Under Configuration click Date and Time.
• Click SNTP to expand.
• Configure the following parameters under SNTP Client:
• Enable SNTP Client: Select this check box to enable. Default: Disabled.
• SNTP Server Address: Enter the Time Server Address. SNTP Server address can be of maximum 64
characters. Both IPv4 and IPv6 addresses are supported. Default: time.nist.gov
• Synchronize Date and Time: You can Synchronize Date and Time with the SNTP Server on daily or
weekly basis. If you select weekly, the system will sync the date and time with the SNTP Server on
every Monday. Select the desired option as per your requirement. Default: Daily.
• Last Synchronized On: This field displays the time when the system last synchronized with the SNTP
Server.
• Sync Date and Time Server with SNTP now: Click this button to synchronize date and time of
SARVAM UCS with the SNTP server, whenever required.


Software Port and Hardware ID
What’s this?
The SARVAM UCS supports different types of ports as illustrated below:
Port
Station Port Virtual Port
SLT
Trunk Port
DKP
ISDN Terminal
SIP Extension (VoIP)

E&M Port
Magneto
BRI Port
T1/ E1 Port
SIP Port (VoIP)
Mobile Port

The SARVAM UCS treats a port as an entity and processes it on the basis of port type and its programmed
attributes.
Type of Port
SLT
DKP
Trunk
Attributes of Port
Access Code
Software Port COS Group
Toll Control Group
DID, etc.
Hardware Port
Slot Number
Port Number
Software Port
The SARVAM UCS takes a software port as fundamental entity. It processes the software port. Hardware ID and
the access code are just two attributes of a software port and hence they are not used anywhere in processing and
programming.
Software ports are always numbered from 1 to the maximum number of ports supported for each port type. The
following table lists this range for each port type.
Port Type Max. Ports Supported Port No. Rangea
SLT 240 001-240
DKP 96 001-096
CO Trunk 64 001-064
E&M 32 001-032
BRI 32 01-32
T1E1PRI 8 01-08
Mobile (GSM) 40 01-40
SIP Trunks 99 01-99
Magneto 16 001-016
a.The number of ports mentioned here are for ETERNITY GENX.

For details regarding the number of ports supported for ETERNITY
LENX/MENX, see “Technical Specifications - SARVAM UCS”.
Each type of software port has different attributes. The System Engineer (SE) programs these attributes using the
corresponding template at the time of installing the SARVAM UCS.

Hardware ID is an attribute of a software port. Hence, all the programming is done for the software port and not for
the hardware ID. Accordingly, the software port number is used for all programming tasks.
An access code is just a Flexible number assigned to the software port. Programming is for the software port and
not for the access code (Flexible number). Accordingly, the software port number is used for all the programming.
The System Engineer allocates software port numbers to different users. This allocation is Flexible and any
software port number can be assumed for any user. Hardware ID is not relevant at this stage. Hardware ID can be
programmed for a software port any time. Further, it can be changed any time in case of hardware failure of a port.
Following example elaborates this point.
Name Position Port Type S/w Port Used
Anil Sharma Managing Director DKP 001
Nikhil Rao VP (Marketing) DKP 002
Revathi Thyagarajan VP (Finance) DKP 003
Anand Chakraborty Manager (MD) SLT 001
Pankaj Shah Accountant SLT 003
Ravi Tandon Sales Executive SLT 002
Conference Room -- DKP 005
Canteen -- SLT 004
Leased Line -- Trunk 003
STD Line -- Trunk 001
Please take a note of following points:
• Software port numbers start from 001 for all different port types.
• Order is not important while allocating software port numbers.
Hardware ID
The hardware ID of a software port denotes where the port is physically located. To derive hardware ID of a
software port, we need:
• Slot number of the card.
• Port number on that card (Port offset).
Slot Assignment for SARVAM UCS
Model Port Name Connector
ETERNITY LENX 01-27 Universal Slots
ETERNITY MENX 01-16 Universal Slots
ETERNITY GENX 01-12 Universal Slots

Use following command to assign hardware ID to a SLT software port:
1101-SLT-Slot-Port offset on the card

Where,
SLT = Software port number of SLT from 001 to 512.
Slot=Slot number from, 01 to 16, in which the card is inserted.
Port offset = Port number on that card, from 01 to 99.
By default, it is auto assigned.
To clear the hardware ID assigned to a SLT software port use the following command:
1101-SLT-00-00
Use following command to assign hardware ID to a DKP software port:

1102-DKP-Slot-Port offset on the card
Where,
DKP = Software port number of DKP from 001 to 128.
Slot=Slot number from 01 to 16 in which the card is inserted.
Use the following command to clear the hardware ID assigned to DKP software port:
1102-DKP-00-00
Use following command to assign hardware ID to DSS software port:

1103-DKP-DSS-Slot-Port offset on the card
Where,
DKP = Software port number of DKP from 001 to 128.
DSS is 0 or 1.
Use the following command to clear the hardware ID assigned to the DSS software port:
1103-DKP-DSS-00-00
Use following command to assign hardware ID to a CO software port:
1104-CO-Slot-Port offset on the card
Where,
CO = Software port number of CO from 001 to 128.
Use the following command to clear the hardware ID assigned to CO software port:
1104-CO-00-00
Use following command to assign hardware ID to an E&M software port:

1105-E&M-Slot-Port offset on the card
Where,
E&M = Software port number of E&M from 001 to 128.
Use the following command to clear the hardware ID assigned to E&M software port:
1105-E&M-00-00
Use following command to assign hardware ID to a BRI software port:

1106-BRI-Slot-Port offset on the card
Where,

Slot is from 01 to 16, in which the card is inserted.
Port Offset on the card is from 01 to 99. (port number 01 to 08 are practically used values).
By default, the Hardware Slot and Port Offset are auto assigned.
Use following command to de-assign the hardware slot and the hardware port assigned to the BRI software port.
1106-BRI-00-00
Use following command to assign hardware ID to a T1E1PRI software port:

1107-T1E1-Slot-Port offset on the card
Where,
Slot = Slot number from 01 to 16, in which the card is inserted.
Port offset = Port number on that card, from 01 to 99 (port number 01 and 02 are practically used values).
Use the following command to clear the hardware ID assigned to T1E1PRI software port:
1107-T1E1-00-00
Use following command to assign hardware ID to a Mobile port:

1108-Mobile Trunk Number-Slot-Port offset on the card
Where,
Mobile Trunk Number is from 001 to 064.
Slot is from 01 to 16.
Port Offset on the card is from 01 to 99. (port number 01 to 04 are practically used values).
Use following command to de-assign the hardware slot and the hardware port assigned to the Mobile port.
1108-Mobile Trunk Number-00-00
Use following command to assign hardware ID to a Magneto port:

1110-Magneto-Slot-Port offset on the card
Where,
Magneto is from 001 to 128.
Slot is from 01 to 16
Port Offset on the card is from 1 to 8.
By default, the Hardware Slot and Port Offset is auto assigned.
Use following command to de-assign the hardware slot and the hardware port assigned to the Magneto port.
1110-Magneto port-00-00

Static Routing Table
What's this?
Static Routing Table is required when you have more than one router (gateway) in your network and you want
SARVAM UCS to send packets to multiple routers/gateways for different types of calls.
Static Routing Table helps route calls between point to point sites (connected through Multi Protocol Label
Switching-MPLS, Frame Relay, etc.) and to public internet at the same time.
How it works
For example, two Local Area Networks, Network A and Network B, are connected through Frame Relay/ Multi
Protocol Label Switching (MPLS) network to give access to local resources and also to make Peer-to-Peer calls.
59.162.252.82
SIP Proxy
A B
192.168.1.0/24 192.168.2.0/24
Public
IP
Frame Relay/MPLS
192.168.1.1
SARVAM
UCS
At both sites SARVAM UCS is connected behind a router.
These sites are also connected to public IP network to:

• give internet access to local hosts.
• access DID service provided by ITSPs to make PSTN/ GSM calls over IP network.
Network A and Network B are in different subnets.
The Static Routing Table makes it possible to route different types of outgoing calls—Peer to Peer or Proxy—made
to different subnets through different Gateways.
You can also access a PMS system installed in a different network using the Static Routing Table.

The Static Routing Table defines the appropriate Gateway Address (or Router’s LAN Address) where the IP
packets are to be sent.
In the Static Routing Table, you must configure:

• The address of the final Destination where the packets are to be sent.
• The Subnet Mask to be applied on the final destination address.
• The Gateway Address where the IP packets are to be sent.
When SARVAM UCS sends packets, if the final destination IP Address and SARVAM UCS are not in the same
Subnet, the system will check the Static Routing Table.
If a perfect match is found, SARVAM UCS will start sending the IP packets to the corresponding Gateway Address
configured in the table.
If no match is found, SARVAM UCS will send the IP Packets to the Default Gateway Address (Network
Connection Type) you configured in the Network Parameters. For detailed instructions, see “Configuring Network
Parameters”.
How to configure
The Static Routing Table must be configured at each location where SARVAM UCS is installed. You may configure
the Static Routing Table using Jeeves.
Configuring Static Routing Table using Jeeves
• Click Static Routing Table.
• The Static Routing Table page opens.

The Static routing table allows you to configure upto 8 entries in each table, IPv4 Addresses table and IPv6
Addresses. Each entry is stored against an Index number.
IPv4 Addresses table

For each entry, you must configure the following fields:
• Destination Address: This is the address of the final destination where the call is to be made. This
can be a device IP Address or Network Address.
• Subnet Mask: This is the mask to be applied on destination address.
• Gateway Address: This is the IP address of the node where the IP packets are to be sent. Generally,
it is the IP address of the LAN interface of the Router.
The Gateway Address must be in the same subnet as SARVAM UCS.
Click Submit to save your entries.
IPv6 Addresses table
For each entry, you must configure the following fields:
• Destination Address: This is the address of the final destination where the call is to be made. This
can be a device IP Address or Network Address.
Valid Range of the IPv6 Address is A to F, a to f, 0 to 9,:(colon). It can be a maximum of 39 characters.

Default: Blank.
• Prefix Length: The Prefix Length is a decimal value that indicates how many of the high-order
contiguous bits of the destination address comprise the prefix (the network portion of the address).
• Gateway Address: This is the IP address of the node where the IP packets are to be sent. Generally,
it is the IP address of the LAN interface of the Router.
The Gateway Address must be in the same subnet as SARVAM UCS.
Click Submit to save your entries.
The CPU Card will restart after the Static Routing parameters have been configured.
To take the above example further, the Static Routing Table of SARVAM UCS at Location A should be
configured as:
Index Destination Address Subnet Mask Gateway Address
1 192.168.2.0 255.255.255.0 192.168.1.1

Index Destination Address Subnet Mask Gateway Address
• The Destination Address 192.168.2.0 specifies the network address of Location B.
• The Subnet Mask is the mask to be applied on the Destination address.
• The Gateway Address 192.168.1.1 specifies the LAN address of the Router A which connects location
A and location B.
The IP address of the LAN interface of the router which connects Location A to the public internet
should be configured as Default Gateway in the Network Parameters of SARVAM UCS in location A.
With the Static Routing Table configured thus, all calls made by SARVAM UCS to 192.168.2.0/ 24 will
be routed through the router which connects Location A to Location B. Whereas, all calls made by
SARVAM UCS to addresses other than 192.168.2.0/ 24 will be routed through the Default Gateway.
Similarly, configure the Static Routing Table in SARVAM UCS at location B to enable calling from
Location B to Location A.

Station Message Detail Recording (SMDR)
What’s this?
The SARVAM UCS can record the details of Internal, Incoming and Outgoing calls made from/to all the extensions.
This feature is called Station Message Detail Recording (SMDR).
You can store SMDR; obtain SMDR as a Report whenever you want or obtain it Online, immediately after the call
has been made or received. You can also use SMDR to calculate the cost of the calls.
To be able to use SMDR, you must configure:
• SMDR Storage: These parameters are programmed to enable the storing of the IC, OG and Internal calls.
To know more, see “Station Message Detail Recording-Storage”.
• SMDR Report: These parameters are programmed to assign destination port for getting report of IC, OG
and Internal calls and to get offline report. To know more, see “Station Message Detail Recording-Report”.
• SMDR Online: These parameters enable you to obtain Online report of Incoming, Outgoing and Internal
calls. With Online SMDR you can obtain details of each call immediately after the call has been made or
received. You can also set the call record format you want for Incoming calls when SARVAM UCS is
interfaced with a third party call accounting software (CAS). To know more see, “Station Message Detail
Recording-Online”.
• SMDR Posting: These parameters enable you to interface third-party call accounting software (CAS) with
SARVAM UCS for call cost calculation. You can select the protocol supported by the call accounting
software and further customize the handshaking parameters and call record formats. To know more see,
“Station Message Detail Recording-Posting”.
Refer to the Hospitality Manual for more details about PMS.

Station Message Detail Recording-Online
What's this?
The SARVAM UCS can generate report for the calls as and when the call is made and send the report to the
computer.
SARVAM UCS also supports a Syslog Client for SMDR. The Syslog Client enables the system to send call records
in syslog format to the remote ‘Syslog Server’. You can view the call records on the remote server. SARVAM UCS
supports SMDR only on TCP.
SMDR generated as and when calls are made or received is called SMDR Online report. In the SMDR Online
report,
• Each Internal call is stored with following fields:
• Extension who made the call.

• Extension to which the call was made.
• Date and time when the call was made.
• Duration of the call in seconds.
• Each Outgoing call is stored with following fields:
• Extension who made the call.

• Trunk line port used for the call.
• Number dialed.
• Date and time when the call matured.
• Duration of the call in seconds.
• Call Units.
• Call Maturity Type.
• Call Type (Normal, DISA, ECF etc.).
• Each Incoming call is stored with the following fields:
• The Trunk on which the call is received.

• Extension number which answered the call.
• Date and time when the call was received.
• Calling Number.
• Hold, speech, ring duration of the call in seconds.
How to configure
To get the Online report you must do the following:
• Enable SMDR Storage in the SMDR buffer. See “Station Message Detail Recording-Storage”.
• Select the Destination port for Incoming, Outgoing and Internal calls. If you select Ethernet as the
destination port, you must configure the Destination IP Address. The Online report is sent to this address
as soon as the incoming call is completed.

• You may also change the default format for the SMDR Online report for incoming calls, like column
position and field length for calling number, speech duration, type of call etc., as required. For this you
need to configure the settings of SMDR Incoming Online Record Format.
Configuring SMDR-Online using Jeeves
• Under Configuration, click Station Message Detail Recording.
• Click SMDR Online to open the page.
• Click SMDR Online to expand.
For SMDR - Outgoing Call Online,
• Select the Destination Port.
If you select Ethernet, in Destination IP Address, enter the IP Address of the remote Syslog Server.
Both IPv4 and IPv6 addresses are supported.
In Port, enter the port of the remote Syslog Server. Valid port range is: 514; 1025 to 65535.
For SMDR - Incoming Call Online,

For SMDR - Internal Call Online,
• To configure Call Record Format for Incoming Calls, click SMDR Incoming Online Call Record Format
to expand.
• Configure the following Call Record Format parameters as required.
For each parameter explained briefly below, you can define the column position, field length (i.e. the
number of digits), the alignment (whether left aligned or right), and the filler characters, wherever required.
• Serial Number: This is the serial number generated for each call record. Serial numbers are generated
from 001 to 999. When serial number '999' is reached, the numbers roll over to 001.
When this field rolls over, it increments the increment counter.
• Increment Counter: It increments when the serial number counter rolls over. The Increment counter
starts from A, ending at Z, and then roll over back to A.
• Property Code: This is the property code, if required. This may be an abbreviation of the property
name.
• Extension Number: This is the extension number that answered the call. You can define the column
position and the field length for the extension number.
• Trunk Number: This is the number of the trunk on which the call was received.

• The Matrix Format occupies 5 character spaces.
• Check-Inn Format occupies 4 character spaces.
• The First Character in the Check-Inn Format is X (Fixed). The remaining three characters show the
software port number.
• Date: The date on which the call was received. The date fill flag is to be enabled.
• Filler Character field is applicable for Date, Month and Year, i.e. whether the single digit date is to be
printed as space-X or 0-X. For example, date = 1 is to be displayed as '1' or '01'.
• Where leading zeroes are not required, the date, month and year sub-fields are right aligned and the
spaces are filled with character 'space'.
• The Date field is not linked to the global flag of Date Format. The global Flag of Date format is used,
while using features or in configuration reports but not for SMDR Online. This is because the date
format used by the CAS is not the same as used by the users of the system.
• Time: The time when the call was received. The format of the time field and the time fill flag are to be
programmed.
• Filler Character field is applicable for Hours, Minutes and Seconds i.e. whether the single digit hour is
to be printed as space-X or 0-X. For example, hour = 1 is to be displayed as '1' or '01'.
• In case leading zeroes are not required, Date, Month and Year sub-fields are right aligned and the
• Answer Duration: The time after which the call was answered. Program the duration unit and the
duration fill flag.
• Hold Duration: The time for which the call was put on hold.
• Speech Duration: The time for which the call was in speech with the extension.
When Duration Unit = Minutes, the rounding off to the nearest whole number is done. For seconds <= 30,
Minute is not incremented. For seconds > 30, minute is incremented.
• Called Number: This is applicable only for calls received on SIP trunks. The number dialed by the
caller is referred to as Called Number.
• Calling Number: This is the number of the Caller.
• Digits dialed in Built-In Auto Attendant: This is the number dialed by the caller using Built-In Auto
Attendant.
• Remarks: You may use this for indicating the Type of Call, for example, Built-In Auto Attendant.
• Reset Serial Number to 001: The Serial number counter can be reset to 001 after 24 hours (from
00:00 HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial
number counter will not be automatically reset.

• Reset Increment Counter: The Increment Counter can be reset to 001 after 24 hours (from 00:00
HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial
number counter will not be automatically reset.
• Property Code: This may be an abbreviation of the property name.
Configuring SMDR-Online using a Telephone

Internal Calls
To assign a destination port for Online SMDR-Internal call record, dial:

• 2830-Code
Where,
Code Meaning
0 None
1 COM Port
2 Ethernet Port
3 USB to COM Port
By default, the no port is assigned, that is Online printing is disabled.
To assign an IP Address, if you select Ethernet Port as the destination port, dial:
• 2832-IP Address
By default, IP Address is 192.168.1.104
To assign a Port, if you select Ethernet Port as the destination port, dial:
• 2833- Port
Where,
IP Port is from 514 and 1025-65535
By default, IP Port is 514.
Outgoing Calls
To assign destination port for Online SMDR-OG Call Record, dial:

• 2730-Code
Where
Code Meaning
0 None
1 COM Port

Code Meaning
2 Ethernet Port
3 USB to COM Port
By default, the port assigned is None. This means the On-line printing is disabled.
To assign an IP Address, if you select Ethernet Port as the destination port, dial:
• 2732-IP Address
To assign the Port, dial:

• 2733-IP Port
Where,
Port is from 514 and 1025-65535
To Start/Abort report generation, dial:

• 1072-101-Flag
Where,
Flag Meaning
0 Abort
1 Start
The SARVAM UCS provides a facility to abort the report generation in midway (1072-101-0). Once the
report generation is aborted, then it has to be explicitly started with command (1072-101-1). This
command is issued from the SA mode.
Incoming Calls
To assign destination port for Online SMDR-IC Call Record, dial:

• 2930-Code
Where,
Code Meaning
0 None
1 COM Port
2 Ethernet Port
3 USB to COM Port
By default, the port assigned is None. This means the Online printing is disabled.

To assign an IP Address, if you select Ethernet Port as the destination port, dial
• 2932-IP Address
By default, IP Address is 192.168.1.104.
To assign the Port:

• 2933-Port
Where,
To start/abort report generation:

• 1072-151-Flag
Where,
Flag Meaning
0 Abort
1 Start
The SARVAM UCS provides a facility to abort the report generation midway (1072-151-0). Once the
report generation is aborted, then it has to be explicitly started with (1072-151-1). This command is
issued from the SA mode.
SMDR record format for IC Call Printing-Online

You can customize the format of the incoming call records as per your requirement. This can be done with the
commands given below:
Serial Number
To program column position for serial number, dial:

• 8200-Column Position
Where,
Column Position is from 00 to 78.
By default, Column Position is 01.
To program field length for serial number, dial:

• 8201-Field Length
Where,
Field Length is from 00 to 78.
By default, Field Length is 04.
To program alignment for serial number, dial:

• 8202-Alignment
Where,
Alignment Meaning
1 Left Alignment

Alignment Meaning
2 Right Alignment
By default, Alignment is 2.
To program fill character for serial number, dial:

• 8203-Fill Character
Where,
Fill Character is 3 digit ASCII value.
By default, Fill Character is ‘Space’.
To program the reset for serial number, dial:

• 8204-Reset
Where,
Reset Meaning
1 No Compulsory Reset
2 Reset to 001 every 24 hours (at 00:00 Hrs.)
By default, Reset is ‘1’.
Serial Number starts from 1 and not 0.

Increment Counter
To program column position for increment counter, dial:

Where,
By default, Column Position is 00 (This field is not available by default).
By default, Field Length is 1, which is fixed.
To program the reset for increment counter, dial:

• 8206-Reset
Where,
Reset Meaning
By default, Reset is ‘1’.
Increment Counter starts from A to Z and then rolls over back to A.

Increment Counter increments when Serial Number Counter rolls over.

Property Code
To program column position for property code, dial:

Where,
By default, Column Position is 00 (This field is not available by default).
To program field length for property code, dial:

Where,
To program property code string for property code, dial:

• 8209-Property Code String
• This is a fixed code that appears in each call.
• The Enterprise can have its own code.
• This code is required by the Property Management System (PMS) when it is catering to more than one
PMS interfaces.
• Refer separate Manual for Hotel/Motel Applications for more details about PMS and Hotel applications
for this feature.
Extension Number
To program column position for extension number, dial:

Where,
To program field length for extension number, dial:

Where,
To program alignment for extension number, dial:

• 8212-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for extension number, dial:

Where,

Trunk Number
To program column position for trunk number, dial:

Where,
To program format type for trunk number, dial:

• 8215-Format Type
Where,
Format Type Meaning
1 Matrix Format
2 Check-In Format
By default, Format Type is ‘1’.
• Matrix Format occupies 5 character space.
• Check In Format occupies 4 character space.
• First Character in Check In Format is X (Fixed). Remaining three characters show the software port
number. However, this will not specify whether the call is made through CO 125 or E&M 125. Also the
channel number will not be specified in case of call made through T1E1PRI port or BRI port.
Date
To program column position for date field, dial:

Where,
To program field length for date field, dial:

Where,
To program alignment for date field, dial:

• 8218-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment

To program fill character for date field, dial:

Where,
By default, Fill Character is ‘Zero’.
To program date format for date field, dial:

Where,
Date Format Meaning
01 DD-MM-YY
02 DD/MM/YY
03 DD.MM.YY
04 DD MM YY
05 DDMMYY
06 DD-MM-YYYY
07 DD/MM/YYYY
08 DD.MM.YYYY
09 DD MM YYYY
10 DDMMYYYY
11 MM-DD-YY
12 MM/DD/YY
13 MM.DD.YY
14 MM DD YY
15 MMDDYY
16 YY-MM-DD
17 YY/MM/DD
18 YY.MM.DD
19 YY MM DD
20 YYMMDD
21 YYYY-MM-DD
22 YYYY/MM/DD
23 YYYY.MM.DD
24 YYYY MM DD
25 YYYYMMDD
26 MM-DD

Date Format Meaning
27 MM/DD
28 MM.DD
29 MM DD
30 MMDD
31 DD-MM
32 DD/MM
33 DD.MM
34 DD MM
35 DDMM
By default, Date Format is 1.
To program date fill flag for date field, dial:

• 8257-Date Fill Flag
Where,
Date Fill Flag Meaning
0 Disable
1 Enable
By default, Date Fill Flag is ‘1’ (that is, single digit in Date, Month and year is printed with prefix ‘0’).
• Leading Zeros field is applicable for Date, Month and Year, that is, whether the single digit date is to be
printed as space-X or 0-X. For example: date = 1 is to be displayed as ‘1’ or ‘01’. In case when leading
zeroes are not required, the date, month and year sub-fields are right aligned and the spaces are filled
with character ‘space’.
• This Date field is not linked to the global flag of Date Format. The global Flag of Date format is used
while using features or in configuration reports but not in PMS. This is because the date format used by
the PMS is not the same as used by the users of the system.
Time
To program column position for time field, dial:

Where,
To program field length for time field, dial:

Where,
To program alignment for time field, dial:
• 8224-Alignment

Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for time field, dial:

Where,
By default, Fill Character is ‘Zero’.
To program time format for time field, dial:

• 8226-Time Format
Where,
Time Format Meaning
0 Disable
1 Enable
By default, Time Format is 1.
To program time fill flag for time field, dial:

• 8258-Time Fill Flag
Where,
Time Fill Flag Meaning
0 Disable
1 Enable
By default, Time Fill Flag is ‘1’.
Leading Zeros field is applicable for Hours, Minutes and Seconds, that is, whether the single digit hour is to
be printed as space-X or 0-X. For example: hour = 1 is to be displayed as ‘1’ or ‘01’. In case when leading
zeroes are not required, Date, Month and Year sub-fields are right aligned and the spaces are filled with
character ‘space’.
Answer Duration
To program column position for answer duration field, dial:

Where,
To program field length for answer duration field, dial:

Where,
To program alignment for answer duration field, dial:

• 8229-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for answer duration field, dial:

Where,
To enable/disable the Filler character flag for Answer Duration, dial:

• 8259-Filler Character Flag for Answer Duration
Where,
Filler Character Flag for

Meaning
Answer Duration
0 Disable
1 Enable
Default = Enable, (that is, the Filler Character will used as programmed)
To program duration unit for answer duration field:

• 8231-Duration Unit
Where,
Duration Unit Meaning
1 HH:MM:SS
2 HHMMSS
3 Minutes
4 Seconds
By default, Duration Unit is ‘4’.
When Duration Unit = Minutes, rounding to nearest whole number is done. For seconds <= 30, Minute is
not incremented and for seconds > 30, minute is incremented.

Hold Duration
To program column position for hold duration field, dial:

Where,
To program field length for hold duration field, dial:

Where,
To program alignment for hold duration field, dial:

• 8234-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment for hold duration field is 2.
To program fill character for hold duration field, dial:

Where,
To enable/disable the Filler character flag for Hold Duration, dial:

• 8260-Filler Character Flag for Hold Duration
Where,
Filler Character Flag

Meaning
for Hold Duration
0 Disable
1 Enable
Default = Enable, (that is, Filler Character will used as programmed)
Speech Duration
To program column position for speech duration field, dial:

Where,
To program field length for speech duration field, dial:


Where,
To program alignment for speech duration field, dial:

• 8239-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for speech duration field, dial:

Where,
To enable/disable the Filler character flag for Speech Duration, dial:

• 8261-Filler Character Flag for Speech Duration
Where,
Filler Character for

Meaning
Speech Duration
0 Disable
1 Enable
Default = Enable, (that is, Filler Character will used as programmed).
Called Number
To program column position for called number field, dial:

Where,
By default, Column Position is 00. (This field is not available by default)
To program field length for called number field, dial:

Where,
To program alignment for called number field, dial:

• 8244-Alignment

Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program number format for called number field, dial:

• 8245-Number Format
Where,
Number Format Meaning
1 Continuous
2 Separated
By default, Number Format is ‘1’.
When separated is selected, put ‘-‘ in the called party number of 10 digits. First after three digits and
another after six digits.
Calling Number
To program column position for calling number field, dial:

Where,
To program field length for calling number field, dial:

Where,
To program alignment for calling number field, dial:
• 8248-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program number format for calling number field, dial:


Where,
1 Continuous
2 Separated
By default, Number Format is 1.
When separated is selected, put ‘-‘ in the called party number of 10 digits. First after three digits and
another after six digits.
Digits dialed in Built-in Auto Attendant
To program column position for the digits field, dial:

Where,
By default, Column Position is 00, that is, this field is not available by default.
To program field length for the digits field, dial:

Where,
To program alignment for the digits field, dial:

• 8252-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
Remarks
To program column position for remarks field, dial:

Where,
To program field length for remarks field, dial:

Where,
To program alignment for remarks field, dial:

• 8255-Alignment

Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To assign default IC SMDR format, dial:

• 8256
How to use
You can start and stop SMDR Online report from the System Administrator mode using Jeeves or dialing SA
Commands from an extension phone.
To start/stop Online report using Jeeves,
• Open Jeeves.
• Under SMDR Management, click SMDR-Online to open the page.
• To start SMDR Online for Outgoing Calls, Incoming Calls and Internal Calls, click the Start button.
• To stop SMDR Online for any of these call types, click Abort button.

To generate SMDR Online report from an extension phone,
To start/stop Online report for Internal Calls:
• Dial 1072-136-1 to start

• Dial 1072-136-0 to stop
To start/stop Online report for Outgoing Calls:

To start/stop Online report for Incoming Calls:

• Exit SA mode.
IC calls-Online report will look like as shown below:
1 5563@[2001::192: V001 I010 07-01-16 12:26:06 0 0 33 N

2 5563@[2001::192: V001 I006 07-01-16 12:26:23 0 0 17 N
3 5563@[2001::192: V001 I006 07-01-16 12:28:02 0 0 29 N
4 5563@[2001::192: V001 I011 07-01-16 12:28:33 0 0 5 T
5 5563@[2001::192: V001 I006 07-01-16 12:28:41 0 0 10 N
Internal calls-Online report will look like as shown below:
IN I006 I011 07-01-2016 12:28:46 5

IN I011 I006 07-01-2016 15:21:52 5
IN I048 2005 07-01-2016 15:29:19 8
IN 2005 2017 07-01-2016 15:28:59 37
IN I011 I006 07-01-2016 15:37:31 6
OG calls-Online report will look like as shown below:
3001 000 M003 9429834769 05-02-16 19:15:12 7 1 3.10 I

3001 000 M003 9429834769 05-02-16 19:16:08 3 1 3.10 I
3001 000 M003 8460961699 05-02-16 19:16:50 6 1 3.10 I
3001 000 M003 7878545045 05-02-16 19:17:57 4 1 3.10 I
3001 000 M003 9429834769 05-02-16 19:18:30 4 1 3.10 I

Station Message Detail Recording-Posting
The Station Message Detail Record (SMDR)-Posting feature of SARVAM UCS is used for interfacing the system
with a third party Call Accounting Software (CAS).
When SARVAM UCS is interfaced with a third party Call Accounting Software (CAS) to determine the cost of the
calls made by the extension users, the system uses SMDR-Posting to send to CAS call record details, like number
to which the call was made by the extension user, number of the extension from which the call was made, the date
and time when the call was made, the duration of the call, metering pulses incurred for the call, etc. On receipt of
this information, the CAS calculates the cost of the call for billing.
As different CAS interfaces support different protocols, the SARVAM UCS offers the flexibility to send call detail
records using the protocol supported by CAS. SARVAM UCS supports as many as 16 different widely-used CAS
protocols such as, Holidex, Hobic, Micros A, Micros B, Comm One, Call-Inn, Bell-HOBIC, XIOX, RSI and others.
Each posting protocol has its own handshaking protocol and call record format. You may configure any one of
these depending upon the protocol supported by CAS you have interfaced with SARVAM UCS. It is also possible to
customize the posting protocol to match the settings required by the CAS you have interfaced.
SARVAM UCS supports SMDR-Posting on Serial RS232 Communication Port as well as on TCP/IP Ethernet(LAN/
WAN) Port. Thus, the CAS can be interfaced either on the COM port or the Ethernet(LAN/WAN) of the SARVAM
UCS.
SMDR-Posting sends outgoing call records only.
SMDR-Posting Protocols
The SARVAM UCS supports different SMDR posting protocols from the system to CAS. The flow of messages
between the SARVAM UCS and the protocols of CAS Interface (Matrix and Blind Send) are described below:
Matrix
• Positive Response from the CAS
SARVAM UCS to CAS CAS to SARVAM UCS
<STX> -Call Record-<ETX> <BCC>
ACK
• Negative Response from the CAS
<STX> -Call Record-<ETX> <BCC> and wait for

Response to Data Timeout (sec), default 3 sec.
NAK
Retransmit <STX> -Call Record-<ETX> <BCC>

and wait for Data Transfer Retry Timer (sec) - on
Negative Response, default 3 seconds.
NAK


NAK

NAK

NAK
The SARVAM UCS will make 5 attempts (default value of Data Transfer Retry Count - on Negative Response)
to send the message after a regular interval of 3 seconds (default value of Data Transfer Retry Timer - on
Negative Response). If the ACK is still not received from the CAS, the SARVAM UCS will proceed to the next
message.
• Busy Response from the CAS
<STX> -Call Record-<ETX> <BCC> and wait for

Response to Data Timeout (sec), default 3 sec.
NAK (CAS responds but cannot accept

at this time)

NAK

NAK

NAK

NAK

The SARVAM UCS will make 5 attempts (default value of Data Transfer Retry Count - on Negative Response)
to send the message after a regular interval of 3 seconds (default value of Data Transfer Retry Timer - on
Negative Response). If the ACK is still not received from the CAS, the SARVAM UCS will proceed to the next
message.
• No Response from the CAS
<STX> -Call Record-<ETX> <BCC> and wait for Response

to Data Timeout (sec), default 3 sec.
(no response)
Retransmit <STX> -Call Record-<ETX> <BCC> and wait for

Data Transfer Retry Timer (sec) - on No Response, default
3 seconds.

3 seconds.

3 seconds.

3 seconds.
The SARVAM UCS will make 5 attempts (default value of Data Transfer Retry Count - on No Response) to
send the message after a regular interval of 3 seconds (default value of Data Transfer Retry Timer - on No
Response). If the ACK is still not received from the CAS, the SARVAM UCS send the new message to CAS.
Blind Send
If you select this protocol as the SMDR-OG Posting Protocol, SARVAM UCS sends the call details without waiting
for any response from the CAS. Each record is sent with the End of Packet Character.
Customized
If you select this protocol as the SMDR-OG Posting Protocol, SARVAM UCS provides you the flexibility to set the
values for the OG Handshaking Protocol and the OG Online Call Record Format as per your requirement.
Call Detail Record Formats

The default Call Detail Record formats for Blind Send and Matrix are given below.

Matrix
Filler Char.
Start
Field Filler Char. Decimal
Parameter Column Format Alignment Remarks
Length Required? Value
Number
Serial Number 01 04 Fixed Right Yes 032 Every 6 hours it is

cleared to 001.
(Starting from
mid-night
00:00:00)
Increment 00 01 Fixed Left NA NA Every 6 hours it is

Counter cleared to A.
(Starting from
mid-night
00:00:00)
Property 00 04 Fixed Left Yes 032 As per the

Code Programmed
String
Extension 06 05 Fixed Right Yes 032

Number
Authority 00 03 Fixed Left NA NA

Code
Trunk Number 12 05 Matrix Left Yes 032

Format
Date 37 10 DD- Right Yes 032

MM-
YYYY
Time 48 08 HH:MM: Right Yes 032

SS
Duration 057 005 Second Right Yes 032

s
Units 063 004 Fixed Right Yes 032
Amount 068 007 Currenc Right Yes 032 Format is

y with DDD.CC
Decimal
Point
Currency 000 001 Fixed Right Space 032 Country Specific

Filler Char.
Start
Number
Call Type 000 001 Fixed Right NA NA As per the Call

Indicator Type Indicator
table
programmed by
the SE. The SE
should program L
= local,
F=International
and Space shall
be used for long
distance.
Location 000 005 Fixed Right NA NA
Called 18 19 Continu Left Space NA

Number ous
PIN 00 04 Fixed Right Yes 032 --
Account Code 00 04 Fixed Right Yes 032
Remarks 76 02 Fixed Left Space NA
Reset Serial
Number to Do not Reset
001
Staring
Character - A
Increment
Counter
Reset
Increment Do not Reset
Counter
Prefix String No
Required
Property 000
Code
Currency
Symbol (Enter 003 000 000 000 000 000 000 000
Decimal
Value)

AST
Filler Char.
Start
Number
Serial Number 01 04 Fixed Right Yes 032 Every 6 hours it is

cleared to 001.
(Starting from
mid-night
00:00:00)
Increment 00 01 Fixed Left NA NA Every 6 hours it is

Counter cleared to A.
(Starting from
mid-night
00:00:00)

Code Programmed
String

Number

Code
Trunk Number 12 05 Matrix Left Yes 032

Format
Date 37 10 DD-MM- Right Yes 032 MM/DD

YYYY
Time 48 08 HH:MM:SS Right Yes 032 HH:MM
Duration 057 005 Seconds Right Yes 032 Duration is in

Minutes.
Amount 068 007 Currency Right Yes 032 Format is

with DDD.CC
Decimal
Point
Currency 000 001 Fixed Right Space 032 $

Filler Char.
Start
Number

table
programmed by
the SE. The SE
should program L
= local,
F=International
and Space shall
be used for long
distance.
Called 18 19 Continuous Left NA NA Area code,

Number Exchange code
and Subscriber
Number
separated by
dash. Space is
sent in place of
Area Code and
first dash if area
code is not
present.
Account Code 00 04 Fixed Right Yes 032 --
Remarks 76 02 Fixed Left NA NA --
Reset Serial Do not Reset

Number to
001
Staring A
Character -
Increment
Counter
Reset Do not Reset

Increment
Counter
Prefix String No
Required
Property 000
Code

Filler Char.
Start
Number
Currency 000 000 000 000 000 000 000 000

Symbol (Enter
Decimal
Value)
Blind Send
Filler Char.
Start
Number
Serial 01 04 Fixed Right Yes 032

Number
Increment 00 01 Fixed Left NA NA

Counter

Code Programmed
String

Number

Code
Trunk 12 05 Matrix Left Yes 032

Number Format
Date 37 10 DD-MM- Right Yes 032

YYYY
Time 48 08 HH:MM:SS Right Yes 032
Duration 057 005 Seconds Right Yes 032
Amount 068 007 Currency Right Yes 032 Format is

with Decimal DDD.CC
Point
Currency 000 001 Fixed Right Space 032 Country Specific

Filler Char.
Start
Number

table
programmed by
the SE. The SE
should program L
= local,
F=International
and Space shall
be used for long
distance.
Called 18 19 Continuous Left Space NA

Number
Account 00 04 Fixed Right Yes 032

Code
Remarks 76 02 Fixed Left Space NA
Reset
Serial Do not Reset
Number to
001
Staring
Character A
-
Increment
Counter
Reset
Increment Do not Reset
Counter
Prefix No
String
Required
Property 000
Code
Currency
Symbol 013 010 000 000 000 000 000 000
(Enter
Decimal
Value)

Customized SMDR-Posting Protocol
You can use the Customized SMDR Posting Protocol to match the settings required by the CAS you have
interfaced. When you use Customized SMDR-Posting Protocol, you can customize the Call Detail Record format to
match your requirement.
When the Call Detail Record format is customized, if there is a gap between two fields, these fields will be 'space'
(ASCII-32).
It is also possible to customize the posting protocol to match the settings required by the CAS you have interfaced.
Setting up CAS Interface

You can setup the CAS Interface on Communication Port or the Ethernet (LAN/WAN) Port:
• To setup the CAS Interface on COM Port279, the following functional components are required to make the
interface work:
• A PC with a spare serial/COM port (not supplied by Matrix).

• The CAS Software (not supplied by Matrix).
• The SARVAM UCS (supplied by Matrix).
Now, connect the COM port of the PC with the COM port the SARVAM UCS using the communication
cable supplied by Matrix280.
• To setup the CAS Interface on the Ethernet (LAN/WAN) Port, the following functional components are
required to make the interface work:
• A PC with a spare LAN/WAN Port (not supplied by Matrix) Or any free LAN Port of the LAN Switch on
which the CAS server application software is running.
• The CAS Software (not supplied by Matrix).
• The SARVAM UCS (supplied by Matrix).
Now, connect the Ethernet (LAN/WAN) Port of the CPU Card of the SARVAM UCS with the Ethernet Port
of the PC (on which CAS server application is running) or to one of the Ethernet port of the LAN Switch, if
the CAS server is in the same LAN.
How to configure
Configuring the SMDR-Posting feature involves the following steps:
• Enabling storage of Outgoing (OG) SMDR. By default, OG SMDR storage is enabled. Refer “Station
Message Detail Recording-Storage”.
• Selecting the appropriate SMDR-OG Posting Protocol to be used.
• Selecting the Destination Port for SMDR-Posting.
279. This is applicable only for ETERNITY GENX.

280. This cable is supplied by Matrix as an optional item.

• If SMDR-Posting is through RS232 (that is, the CAS Interface is to be set up on the COM Port),
program the attributes of the COM port. Refer the chapter “Communication Ports” to set attributes of
the COM port.
• If SMDR-Posting is through TCP/IP (that is, the CAS Interface is to be set up on the Ethernet (LAN/
WAN) Port, program the destination IP address and Port.
• Refining the Handshake parameters, if required.
• Refining Call Detail Record format, if required.
• Starting SMDR-Posting process.
Configuring SMDR Posting using Jeeves,

Selecting SMDR-Posting Protocol for CAS

• Click the SMDR Posting. The SMDR-Posting parameters page opens.
• Click SMDR-Posting to expand.
• In the SMDR-OG Posting Protocol (Handshaking and OG Call Record Format) drop down list, select the
appropriate protocol to be used. Default: Matrix.
• Select the Destination Port on which the SMDR Posting is set up. You can select COM Port or Ethernet.
Default: None.

If you select Ethernet,
• In Destination IP Address, enter the IP Address of the PC on which the CAS server application
software is running, that is, where SARVAM UCS should post SMDR. Both IPv4 and IPv6
• In Port, enter the port of the PC on which the CAS server application software is running, that is,
where SARVAM UCS should post SMDR. Valid port range is: 5000; 514; 1025 to 65535.
• To start SMDR posting, select Process as Start.
Refining Handshake Parameters

You may need to refine some of the Handshake parameters of the selected SMDR-Posting protocol, that is,
change the factory default values of the protocol, to match the software requirements of the CAS being used in the
organization. Refer the below table for default values of each protocol supported by SARVAM UCS.
To refine Handshake Parameters,
• Click SMDR-Posting OG Handshaking Protocol.
Configure the following parameters as required:

• Response to ENQ Timeout (sec): The time for which the sender waits for a response to ENQ from the
receiver.

• ENQ Retry Count - on No Response: The number of times the sender should send ENQ before
dropping the process, in case response is not received for the last message sent.
• ENQ Retry Timer (sec) - on No Response: The time after which the sender should sent the ENQ
again, in case the response is not received for the last message sent.
• ENQ Retry Count - on Negative Response: The number of times the sender should send ENQ
before dropping the process, in case of a negative response received for the last message sent.
• ENQ Retry Timer (sec) - on Negative Response: The time after which the sender should sent the
ENQ again.
• Response to Data Timeout (sec): The time for which the sender waits for a response to data from the
receiver.
• Data Transfer Retry Count - on No Response: The number of times the sender should send ENQ
before dropping the process. This parameter is used when ACK is received against ENQ and there is
some problem while sending the data.
• Data Transfer Retry Timer (sec) - on No Response: The time after which the sender should send the
ENQ again before dropping the process. This parameter is used when ACK is received against ENQ
and there is some problem in sending the data.
• Data Transfer Retry Count - on Negative Response: The number of times the sender should send
ENQ before dropping the process. This parameter is used when ACK is received against ENQ and
there is some problem in sending the data.
• Data Transfer Retry Time (sec) - on Negative Response: The time after which the sender should
sent the ENQ again before dropping the process. This parameter is used when ACK is received
against ENQ and there is some problem in sending the data.
• Use ENQ Character: This is enabled if the protocol uses ENQUIRE (ENQ) Signal.
• ENQ Character: This parameter signifies the ASCII character (Single Character) used to send
ENQUIRE (ENQ) signal to the receiver.
• Acknowledgement Character: This parameter signifies the ASCII character (Single Character) used
by the receiver to acknowledge the receipt of the Link Control Character/Message Data.
• No Acknowledgement Character: This parameter signifies the ASCII character (Single Character)
used by the receiver to dis-acknowledge the receipt of the Link Control Character/Message Data.
• Start of Packet Character: A string of four ASCII characters used by the receiver to indicate Start of
Packet. Each ASCII character is from 000 to 252. Start of Packet may be of one character only, in this
case the string should be completed by programming remaining three characters with ASCII Null
Character (000).
• End of Packet Character: A string of four ASCII characters used by the receiver to indicate End of
Packet. Each ASCII character is from 000 to 252. End of Packet may be of one character only, in this
case, the string should be completed by programming the remaining three characters should be
programmed as ASCII Null (000).
• Use Byte Code Check (BCC): This flag is to be enabled when the protocol uses BCC Signal.

Refining Call Detail Record Format Parameters

The Call Detail Record (CDR) format for the selected SMDR-Posting protocol can also be refined to match the
software requirements of the CAS being used by the organization.
This may be required if you have selected a 'customized' protocol. To refine Call Record Format,
• Click SMDR-Posting OG Online Call Record Format to expand.
• Serial Number: This is the serial number generated for each call record. Serial numbers are generated
from 000 to 999. When serial number '999' is reached, the numbers roll over to 000.
Serial Number starts from 1 and not 0.

• Increment Counter: It increments when the serial number counter rolls over. The Increment counter starts
from A, ending at Z, and then roll over back to A.
• Property Code: This is the property code required by the CAS used in the organization. It is a string of
alphanumeric characters and is to be terminated with #*. This field has a maximum of 128 alphanumeric
characters.
• You must program this string keeping in mind the field length used by the selected/customized posting
protocol.

• The default value of the default Property Code String has been set as '000', as at least two known
protocols use this field. You can set a different value here and the new value will appear in the CDR
record, irrespective of the protocol type selected.
• If XIOX protocol has been selected as SMDR-OG Posting Protocol (Handshaking and OG Call Record
Format), you should program Property Code as HTL.
• Extension Number: This is the extension number from which the call was made. You can define the
column position and the field length of the Extension number in the Call Detail Record.
• Trunk Number: This is the number of the trunk from which the call was made.
• The Matrix Format occupies 5 character spaces.
• Check-Inn Format occupies 4 character spaces.
• The First Character in the Check-Inn Format is X (Fixed). The remaining three characters show the
software port number. However, this does not specify whether the call is made through CO 125 or E&M
125. Also, the channel number is not specified in case of call made through T1E1PRI port or BRI port.
• Date: The date on which the call was made. The date fill flag is to be enabled.
• Filler Character field is applicable for Date, Month and Year, that is, whether the single digit date is to
be printed as space-X or 0-X. For example, date = 1 is to be displayed as '1' or '01'.
• Where leading zeroes are not required, the date, month and year sub-fields are right aligned and the
• The Date field is not linked to the global flag of Date Format. The global Flag of Date format is used,
while using features or in configuration reports but not in CAS. This is because the date format used by
the CAS is not the same as used by the users of the system.
• Time: The time when the call was made. The format of the time field and the time fill flag are to be
programmed.
• Filler Character field is applicable for Hours, Minutes and Seconds, that is, whether the single digit hour
is to be printed as space-X or 0-X. For example, hour = 1 is to be displayed as '1' or '01'.
• In case when leading zeroes are not required, Date, Month and Year sub-fields are right aligned and
the spaces are filled with character 'space'.
• Duration: The duration of each call. Program the duration unit and the duration fill flag.
When Duration Unit = Minutes, the rounding off to the nearest whole number is done. For seconds <= 30,
Minute is not incremented. For seconds > 30, minute is incremented.
• Units: The duration of the call interpreted in terms of units. The number of units depends on the Pulse
Rate. The number of units is derived from the Call Unit = Call duration in seconds/Pulse rate in seconds.
• Amount: This is the Amount of the call. Program the amount format and the fill flag.
• Filler Character field is applicable for both the sub fields of Amount viz. Rupees/Paisa, that is, whether
the single digit Rupee is to be printed as space-X or 0-X. For example, Rupee = 1 is to be displayed as

'1' or '01'. Where leading zeroes are not required, the Rupee and Paisa are right aligned and the
• When Amount Format = Higher Currency, rounding to nearest whole number is done. For Lower
Currency <= 50, Higher Currency is not incremented and for Lower currency > 50, Higher Currency is
incremented.
• Currency: This is the symbol of the currency in which the Amount is charged. A maximum of 8 ASCII
Characters are allowed.
• Generally, Currency Symbol field prefixes to Amount field. Hence, to comply with various CDR formats,
it is recommended that the column position of Currency Symbol and Amount field should be
programmed properly.
• You can change the Currency Symbol used in the OG-SMDR Format. However, this change will not be
reflected in the Front Desk User Wizard.
• Call Type Indicator: This indicates the type of call made, that is, whether local, international, information,
etc.
Your must program the Number String, the Text String and its Meaning as explained in following table:
Text
Number Index Number String Meaning
String
01 0 LD Long Distance
02 95 IC Inter Circle
03 197 INFO Information
04 0 INTL International
: : : :
36 2 L Local
The Text String is a string of Alphanumeric characters. Number String is of a maximum 4-digits.
The Number Index is kept as '36' as one of the SMDR-OG Posting protocols, INN-FORM XL supports 24
different types of calls.
By default, all the entries in this table are blank.
You are advised to program the first 10 entries of this table as below if the selected posting protocol is Bell
Hobic or XIOX.
Text
String
01 1 A
02 2 A
03 3 A
04 4 A

Text
String
05 5 A
06 6 A
07 7 A
08 8 A
09 9 A
10 0 A
: : :
36 Blank Blank Blank
You are advised to program the first 11 entries of this table as below, if the selected posting protocol is
Holidex or Hobic.
Text
String
01 1 L
02 2 L
03 3 L
04 4 L
05 5 L
06 6 L
07 7 L
08 8 L
09 9 L
10 0 L
11 0 F International
: : :
36 Blank Blank Blank
• You are advised to use default (that is, Blank) table, if the selected protocol is Hilton, as Hilton uses
blank entries in this field which is 12 bytes long.
• The Text String should preferably be same as Field Length. If not, the remaining spaces will be filled
with character 'Space'. If the Field length is less than the Text string characters, then the number of text
characters equal to the Field length will be printed.
• Location: This column indicates the location of the external number to which the call was made.
• The system detects the location from the called location programmed in the Area and Country Code
Tables.

• Called Location is programmed as one of the parameters of the Area Code Table and Country Code
Table. Depending upon the prefix dialed, the Location string is picked up from either Country Code
table or Area Code table.
• Called Location is not displayed for Local Calls.
• The Called Location parameter in the Country Code table and Area Code table is of 8 Characters.
• If the number of characters in the field Called Location is more than Field length then the remaining
characters will not be printed (overlapped by next field).
• If the number of characters in the field Called Location is less than Field length then the remaining
characters in the field Called Location will be filled by spaces.
• Called Number: This is the external number to which the call was made.
• One way to separate the called party number is by Area Code, Exchange code and Subscriber
Number. This is difficult in an Open numbering system, in which the field size of area code, exchange
code are not standard but vary from two digits to four digits (for example, the Area code for 'Mumbai' is
of 2 digits, whereas that of 'Vadodara' is 3 digits).
• In the Closed numbering system, the Area Code, Exchange Code and the Subscriber number are of
fixed length. In such case, including '-' in the called party number is not difficult. Hence, '-' is put in the
called party number. The called party number is assumed to be of 10 digits. The first '-' is placed after
four digits, counting from the right. The second '-'is placed after seven digits, counting from the right. If
the dialed number is a local number of 7 digits then the second '-'is not placed. Also, the remaining
three digits are not placed, but filled with character 'space'.
• In this case, even if the call is made to a geographical area where open numbering system is followed,
'-' is placed in the same way.
• Account Code: This is the Account Code (Refer Note4) using which the call was made.
• Remarks: This column indicates the details of the call; whether it was a DISA call, DOSA call, Auto Redial
Call, type of call maturity.
Fixed Characters are used to indicate the type of call, call details, etc. The notations for the Remarks field
are:
D DISA Call
A Auto Redial Call
C CPD
K 12KHz/16KHz
R Reversal
D Delay
I Connect
• Reset Serial Number to 001: The Serial number counter can be reset to 001 after 24 hours (from 00:00
HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial number
counter will not be automatically reset.

• Starting Character - Increment Counter: Specify the starting character of the increment counter as the
serial number rolls over, in this field.
• Reset Increment Counter: The Increment Counter can be reset to 001 after 24 hours (from 00:00
HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial number
counter will not be automatically reset.
• Prefix String Required: This flag is to be programmed if the prefix string 0ac1 is to be sent when
interfacing with OG-SMDR Posting Protocol.
• Property Code: Enter the property code required by the CAS.
• Currency Symbol (Enter Decimal Value): Enter currency symbol to be used.
Call Type Indicator

In the Call Record, you can also include the Type of Call: local, national, international, by configuring the Call Type
Indicator Table.
• Click Call Type Indicator to expand.
• In the Dialed Number String column, enter the number strings for each Call Type. You can enter the
prefix, e.g. 0 for long distance calls, 2 for local numbers, etc.
• For each Dialed Number String, define a Call Type Indicator, this is an abbreviation of the Call Type,
e.g.: LD for long distance, INTL for International, etc.

Number String is of a maximum 4-digits. The Text String is a string of 4 alphanumeric characters. Your
entries may look like these:
Dialed Call Type Indicator

Number Index Meaning
Number String (Text String)
01 0 LD Long Distance
02 95 IC Inter Circle
03 197 INFO Information
04 00 INTL International
: : : :
36 2 L Local
You may enter as many Call Types as supported by the Posting Protocol you have selected.
Configuring SMDR Posting using a Telephone
Selecting SMDR -Posting Protocol for CAS

To enable/disable Storage of SMDR of Outgoing Calls, dial:

• 2701-Flag
Where,
Flag is
1 for Enable
0 for Disable
By default, storage of outgoing calls is enabled.
To assign Destination Port for SMDR-OG Posting, dial:

• 8330-Code
Where,
Code Meaning
0 None
1 COM Port
2 Ethernet Port
3 USB to COM Port
By default, the port assigned is Ethernet.
For example, to select COM Port as destination port for SMDR Posting in SARVAM UCS, dial: 8330-1.
To program CAS Server's IP Address (destination IP Address), dial:

• 8331-Address-Address-Address-Address
Where,

Address is from 000 to 255 for the first 3 Octets and 001 to 254 for the fourth Octet.
By default, Destination Server IP Address is 192.168.1.103.
To program CAS Server IP Port (Destination Port), dial:

• 8332-Destination IP Port
Where,
Destination IP Port number is from 1025 to 65535.
By default, destination IP Port is 5000.
To select appropriate SMDR-OG Posting Protocol:

• 8301-SMDR-OG Posting Protocol
Where,
SMDR-OG Posting protocols is from 01 to 16.
Posting Protocol Protocol Name
01 Blind Send
02 Matrix
03 Holidex
04 HOBIS A
05 HOBIS B
06 HOBIC
07 BELL HOBIC
08 MICROS A
09 MICROS B
10 Hilton
11 Xiox
12 Comm One
13 Call-Inn
14 RSI-CMS
15 Customized
16 AST
By default, Posting Protocol is 02, that is, Matrix.
When the above command is issued, the system will set the default values of the OG Handshaking
Protocol and OG Online Call Record Format parameters as per the selected Posting Protocol.
To start/stop SMDR Posting Process, dial:
• 8333-Code

Where,
Code Meaning
0 Abort
1 Start
By default, SMDR Posting is Stop.

To restore the default OG SMDR-Posting Parameters, dial:
• 8300
• Exit SE mode.
Handshaking Parameters
• Enter SE mode.
To set ENQ no response timer, dial:

• 8302-ENQ No Response Timer (Response to ENQ Timeout)
Where,
ENQ No Response Timer is from 01-99 Seconds.
To set ENQ no response retry count, dial:

• 8303-ENQ Retry Count
Where,
ENQ Retry Count is from 01-99.
To set ENQ no response Retry Timer, dial:

• 8304-ENQ No Response Retry Timer
Where,
ENQ No Response Retry Timer is from 01-99 Seconds.
To set ENQ Retry Count, dial:

• 8305-ENQ Retry Count
Where,
ENQ Retry Count is from 01-99.
To set ENQ Retry Time, dial:

• 8306-ENQ Retry Timer
Where,
ENQ Retry Time is from 01-99 Seconds.
To Response to Data Timeout, dial:

• 8307-Response to Data Timeout
Where,
Response to Data Timeout is from 01-99 Seconds.
To set Data Transfer Retry Count, dial:

• 8308-Data Transfer Retry Count
Where,
Data Transfer Retry Count is from 01-99.
To set Data Transfer Retry Time, dial:

• 8309-Data Transfer Retry Time
Where,
Data Transfer Retry Time is from 01-99 Seconds.
To set Data Transfer Retry Count, dial:

• 8310-Data Transfer Retry Count
Where,
Data Transfer Retry Count is from 01-99.
To set Data Transfer Retry Time, dial:

• 8311-Data Transfer Retry Time
Where,
Data Transfer Retry Time is from 01-99 Seconds.
To enable/disable ENQUIRE Signal, dial:

• 8312-ENQUIRE Signal
Where
ENQUIRE Signal Meaning
0 Disable
1 Enable
To set the ENQUIRE character, dial:

• 8313-ENQUIRE
Where,
ENQUIRE is an ASCII Character from 000 to 252.
To set the ACK Character, dial:

• 8314-Set ACK Character
Where,
Set ACK Character is an ASCII Character from 000 to 252.
To program the NAK Character, dial:

• 8315-Set NAK Character
Where,
Set NAK Character is an ASCII Character from 000 to 252.
To set the Start of Packet string, dial:

• 8316-Character 1-Character 2-Character 3-Character 4
Where,
Start of Packet is a string of four ASCII Characters. Each ASCII character is from 000 to 252. If the
Start of packet contains only one ASCII character then the string should be completed by programming
remaining three characters with ASCII Null Character (000).
If STX is to be programmed as 'Start of Packet', dial: 8316-002-000-000-000
To program the End of Packet string, dial:

• 8317-Character 1-Character 2-Character 3-Character 4
Where,
End of Packet is a string of four ASCII Characters. Each ASCII character is from 000 to 252. If the End
of packet contains only one ASCII character then the string should be completed by programming
remaining three characters with ASCII Null Character (000).

If ETX is to be programmed as 'End of Packet', dial: 8317-003-000-000-000
To enable/disable BCC Flag, dial:

• 8318-BCC Flag
Where,
BCC Flag Meaning
0 Disable
1 Enable
• Exit SE mode.
Call Detail Record format parameters

• Enter SE mode.
To program column position for serial number, dial:

Where,
To program field length for serial number, dial:

Where,
To program alignment for serial number, dial:

• 8102-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for serial number, dial:

Where,
Fill Character is 3 digit ASCII value ranging from 032 to 254.
By default, Fill Character is 'Zero'.
To program reset for serial number, dial:

• 8104-Reset

Where,
Reset Meaning
By default, Reset is '1'.
To program column position for increment counter, dial:

Where,
By default, Field Length is 1, which is fixed.
To program reset for increment counter, dial:

• 8106-Reset
Where,
Reset Meaning
By default, Reset is '1'.
To program starting character for increment counter, dial:

• 8174-Starting Character
Where,
Starting Character is from A to Z.
By default, Starting Character is 'A'.
To program column position for property code, dial:

Where,
By default, Column Position is 000 (This field is not available by default)
To program field length for property code, dial:

Where,
To program property code string for property code, dial:

• 8109-Property Code String
Where,
Property Code String is maximum of 16 characters.

By default, it is '000'.
To program column position for extension number, dial:

Where,
To program field length for extension number, dial:

Where,
To program alignment for extension number, dial:

• 8112-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for extension number, dial:

Where,
Fill Character is 3 digit ASCII value and ranging from 032 to 254.
By default, Fill Character is 'Space'.
To program column position for trunk number, dial:

Where,
To program format type for trunk number, dial:

• 8115-Format Type
Where,
Format Type Meaning
1 Matrix Format
2 Check-Inn Format
By default, Format Type is '1'.
To program column position for date field, dial:

Where,

To program field length for date field, dial:

Where,
To program alignment for date field, dial:

• 8118-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for date field, dial:

Where,
Fill Character is 3 digit ASCII value from 032 to 254.
To program date format for date field, dial:

Where,
Date Format Meaning
01 DD-MM-YY
02 DD/MM/YY
03 DD.MM.YY
04 DD MM YY
05 DDMMYY
06 DD-MM-YYYY
07 DD/MM/YYYY
08 DD.MM.YYYY
09 DD MM YYYY
10 DDMMYYYY
11 MM-DD-YY
12 MM/DD/YY
13 MM.DD.YY
14 MM DD YY
15 MMDDYY

Date Format Meaning
16 YY-MM-DD
17 YY/MM/DD
18 YY.MM.DD
19 YY MM DD
20 YYMMDD
21 YYYY-MM-DD
22 YYYY/MM/DD
23 YYYY.MM.DD
24 YYYY MM DD
25 YYYYMMDD
26 MM-DD
27 MM/DD
28 MM.DD
29 MM DD
30 MMDD
31 DD-MM
32 DD/MM
33 DD.MM
34 DD MM
35 DDMM
By default, the date format depends upon the Posting Protocol selected.
To program date fill flag for date field, dial:

• 8170-Date Fill Flag
Where,
Date Fill Flag Meaning
0 Disable
1 Enable
By default, Date Fill Flag is '1'.
To program column position for time field, dial:

Where,
To program field length for time field, dial:

Where,
To program alignment for time field, dial:

• 8124-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for time field, dial:

Where,
To program time format for time field, dial:

• 8126-Time Format
Where,
Time Format Meaning
1 HH:MM:SS
2 HH:MM
By default, Time format is 1.
To program time fill flag for time field, dial:

• 8171-Time Fill Flag
Where,
Time Fill Flag Meaning
0 Disable
1 Enable
By default, Time Fill Flag is '1'.
To program column position for duration field, dial:

Where,
To program field length for duration field, dial:


Where,
To program alignment for duration field, dial:

• 8129-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for duration field, dial:

Where,
To program duration unit for duration field, dial:

• 8131-Duration Unit
Where,
Duration Unit Meaning
1 HH:MM:SS
2 HHMMSS
3 Minutes
4 Seconds
By default, Duration Unit is '4'.
To program duration fill flag for duration field, dial:

• 8172-Duration Fill Flag
Where,
Duration Fill Flag Meaning
0 Disable
1 Enable
By default, Duration Fill Flag is '1'.
To program column position for units field, dial:

Where,

To program field length for units field, dial:
Where,
To program alignment for units field, dial:

• 8134-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for units field, dial:

Where,
To program column position for amount field, dial:

Where,
To program field length for amount field, dial:

Where,
To program alignment for amounts field, dial:

• 8138-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program fill character for amounts field, dial:

Where,

To program amount format for amount field, dial:
• 8140-Amount Format
Where,
Amount Format Meaning
1 Higher Currency
2 Lower Currency
3 Spoken Currency with decimal point
4 Spoken Currency without decimal point
By default, Amount Format is '3'.
To program the amount fill flag for amount field, dial:

• 8173-Amount Fill Flag
Where,
Amount Fill Flag Meaning
0 Disable
1 Enable
By default, Amount Fill Flag '1'.
To program column position for currency symbol field, dial:

Where,
To program field length for currency symbol field, dial:

Where,
To program alignment for currency symbol field, dial:

• 8143-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
By default, Alignment is '2'.
To program fill character for currency symbol field, dial:

Where,

To program symbol for currency symbol field, dial:

• 8145-Character 1-Character 2-Character 3-Character 4-Character 5-Character 6-Character 7-
Character 8
Where,
Character 1 to Character 8 shall be in 3 digit Decimal values.
Decimal values 000 and 032 to 255 are allowed.
If currency string/symbol to be used is fewer than 8 characters, terminate the command with #*.
Refer following table to know Decimal value of corresponding currency character:
Dec ASCII Char. Meaning Dec ASCII Char. Meaning
0 NUL Null 32 Space

1 SOH Start of heading 33 ! !
2 STX Start of text 34 " "
3 ETX Break/end of text 35 # #
4 EOT End of transmission 36 $ $
5 ENQ Enquiry 37 % %
6 ACK Positive acknowledgment 38 & &
7 BEL Bell 39 ' '
8 BS Backspace 40 ( (
9 HT Horizontal tab 41 ) )
10 LF Line feed 42 * *
11 VT Vertical tab 43 + +
12 FF Form feed 44 , ,
13 CR Carriage return 45 - -
14 SO Shift out 46 . .
15 SI Shift in/XON (resume output) 47 / /
16 DLE Data link escape 48 0 Zero
17 DC1 XON - Device control character 1 49 1 One
18 DC2 Device control character 2 50 2 Two
19 DC3 XOFF - Device control character 3 51 3 Three
20 DC4 Device control character 4 52 4 Four
21 NAK Negative Acknowledgment 53 5 Five
22 SYN Synchronous idle 54 6 Six
23 ETB End of transmission block 55 7 Seven
24 CAN Cancel 56 8 Eight
25 EM End of medium 57 9 Nine
26 SUB substitute/end of file 58 : :
27 ESC Escape 59 ; ;
28 FS File separator 60 < <
29 GS Group separator 61 = =
30 RS Record separator 62 > >
31 US Unit separator 63 ? ?

Dec ASCII Char. Meaning Dec ASCII Char. Meaning
64 @ @ 96 ` `
65 A A 97 a a
66 B B 98 b b
67 C C 99 c c
68 D D 100 d d
69 E E 101 e e
70 F F 102 f f
71 G G 103 g g
72 H H 104 h h
73 I I 105 i i
74 J J 106 j j
75 K K 107 k k
76 L L 108 l l
77 M M 109 m m
78 N N 110 n n
79 O O 111 o o
80 P P 112 p p
81 Q Q 113 q q
82 R R 114 r r
83 S S 115 s s
84 T T 116 t t
85 U U 117 u u
86 V V 118 v v
87 W W 119 w w
88 X X 120 x x
89 Y Y 121 y y
90 Z Z 122 z z
91 [ [ 123 { {
92 \ \ 124 | |
93 ] ] 125 } }
94 ^ ^ 126 ~ Tilde
95 _ _ 127 DEL Delete
To program column position for call type indicator field, dial:

Where,
To program field length for call type indicator field, dial:

Where,

To program alignment for call type indicator field, dial:
• 8148-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program number string for call type indicator field, dial:

• 8149-Number Index-1-Number String
Where,
Number String is of four digits. Terminate with #* if the string is less than four digits.
To program text string for call type indicator field, dial:

• 8149-Number Index-2-Text String
Where,
Text String is a string of Alphanumeric characters. Terminate with #* if the string is less than four digits.
Keep the Text String same as the Field Length.
By default, all the entries in this table are blank.
To program column position for called location field, dial:

Where,
To program field length for called location field, dial:

Where,
To program alignment for called location field, dial:

• 8152-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program column position for called number field, dial:

Where,

To program field length for called number field, dial:

Where,
To program alignment for called number field, dial:

• 8156-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To program number format for called number field, dial:

Where,
1 Continuous
2 Separated
By default, Number Format is '1'.
To program column position for account code field, dial:

Where,
To program field length for account code field, dial:

Where,
To program alignment for account code field, dial:

• 8160-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment

To program fill character for account code field, dial:
Where,
To program prefix string (ac01), dial:

• 8165-Code
Where,
Code Meaning
0 No
1 Yes
By default, Code is '0'.
To program column position for remarks field, dial:

Where,
To program field length for remarks field, dial:

Where,
To program alignment for remarks field, dial:

• 8168-Alignment
Where,
Alignment Meaning
1 Left Alignment
2 Right Alignment
To assign default CDR format, dial:

• 8169
Configuring Area Code and Country Code

Country code and Area Code must be programmed so that a call can be placed to a particular location in a
particular country. Country and Area Codes need to be programmed also for the purpose of calculating cost of
calls. Refer chapter “Call Cost Calculation (CCC)”.
To program a country code, dial:

• 8321-Index-Country Code-#*
Where,

Country Code is a number string of maximum of 4 digits, terminate with #* if less than four digits.
Index Country Code
001 041
002 051
: :
200 096
To program a location in country code, dial:

• 8322-Index-Country Name-#*
Where,
Country Name is of 8 characters; terminate the command string with #*, if the name is less than 8
characters.
Index Country Name
001 India
002 Kenya
: :
200 UAE

Station Message Detail Recording-Report
What's this?
The SARVAM UCS can generate SMDR reports in two modes:
• Online: as and when a call is made or received (see “Station Message Detail Recording-Online”)
Or
• Offline: whenever required, the records of calls stored in the buffer can be printed.
Generation of call record reports offline, is called SMDR - Report.
You can generate SMDR Report, either
• Manually: The report is generated whenever you want.

Or
• As per Schedule: The report is generated on a preset Day, Date and Time.
SARVAM UCS allows you to set a variety of filters for printing SMDR Reports.
SARVAM UCS supports SMDR-Reports on Serial RS232 Communication Port as well as on TCP/IP Ethernet Port.
How to configure
To be able to generate SMDR -Report, you must do the following:
• Enable SMDR Storage in the SMDR buffer. See “Station Message Detail Recording-Storage”.
• Assign the Destination port for Incoming, Outgoing and Internal calls.
Configuring SMDR-Online using Jeeves

• Click SMDR Reports to open the page.
• For SMDR - Outgoing Call Report,
• Select the Destination Port. Default: None.
• If you select Ethernet as the Destination Port,
• In Port, enter the port of the remote Syslog Server. Valid port range is: 514; 1025 to 65535.
• For SMDR - Incoming Call Report,
• For SMDR - Internal Call Report,

Configuring SMDR-Online using a Telephone
Incoming calls report

Destination Port:
To assign a destination port for SMDR-IC Report, dial:

• 2931-Code
Where,
Code Meaning
0 None
1 COM Port
2 Ethernet Port
3 USB to COM Port
If you assigned Ethernet Port as destination port, to assign the IP Address to the Ethernet Port, dial:
• 2934-IP Address
To assign the Port, dial:

• 2935- Port
Where,
Internal Calls Report

Destination Port
To assign destination port for Online SMDR-Internal Report, dial:

• 2831-Code
Where,
Code Meaning
0 None
1 COM Port
2 Ethernet Port
3 USB to COM Port
• 2834-IP Address

To assign the IP Port, dial:

• 2835-IP Port
Where,
OG Calls Report
Destination Port
To assign a destination port for SMDR-OG Report, dial:

• 2731-Code
Where,
Code Meaning
0 None
1 COM Port
2 Ethernet Port
3 USB to COM Port
• 2734-IP Address
To assign the IP Port, dial:

• 2735-IP Port
Where,
How to use
You can print SMDR Report whenever you want or schedule printing of the report from the System Administrator
mode using Jeeves or dialing SA Commands from an extension phone.
Print SMDR-Report using Jeeves

• Open Jeeves.
• Click SMDR Management to expand.

Outgoing Calls
• To print Outgoing Calls with filters, click OG Call Report.
Setting Print Filters
To print calls originating and termination on specific trunks/extensions,
• Click Calls to expand.

• Set the following filters as desired. You can print records of outgoing calls made by all extensions or by
specific extensions: Calls made by All Extensions, Calls made by Extensions. Default: Calls made by
All Extensions
To print records of specific extensions, clear the Calls made by All Extensions check box and enter the
desired extension range in From and To in Calls made by Extension.
You can also print records of outgoing calls Originating and outgoing calls Terminating on specific
Extension numbers, and trunks: CO, E&M, T1E1, BRI, Mobile and SIP.
You can also print outgoing calls made using Account Code, Authority Codes, Calls for Department
Billing Groups, PIN and Unanswered Outgoing Calls.
If you have extension numbers beginning with # or *, make sure the range you assign in From and To
either have # or *. A mix of both will not work.
To print outgoing calls made to certain numbers,
• Click Number List to expand.
• Enter the desired numbers in the Number List of your choice and select the same number list in Calls
made on called numbers matching with Number List.
To filter calls according to specific dates, time and duration,
• Click Filter Calls to expand,
• To print outgoing calls made on a certain date or between a certain time period, set the filter Calls made
between. To print calls made on a particular date, select the same Date, Month and Year in both fields.
• To print outgoing calls made at a particular time, set the Hours and Minutes in 24-hour format calI in the
filter Calls made between 00: 00 and 23:59.

• If you want outgoing calls that exceed as certain duration to be printed, set the filter Calls with duration
more than (sec) to the desired duration. All outgoing calls with duration greater than this value, will be
printed.
• If you want outgoing calls exceeding certain metering units to be printed, set the filter Calls with units
more than (units) to the desired value. All outgoing calls that have metering units greater than this value
will be stored.
• Click Scheduled Report Generation to expand.
• To generate SMDR Report of outgoing calls on a particular day, day of the week, or day of the month, set
Scheduled Report Generation, as required.
• You can print the SMDR Report of outgoing calls any time you want. You can print the report on a local
printer or the Destination Port of SARVAM UCS.

To view/print the report on the local printer,
• A detailed report is displayed on the screen.
• Click Print to print the report.
To print the report on the Destination Port,
• Click the Send to Destination Port button.
• To stop printing, click Abort button.

Incoming Calls
• To print Incoming Calls with filters, click IC Call Report.
To print calls received on specific trunks/extensions,
• Click Extension and Trunks to expand.
• Set the filters as desired. You can print records of incoming calls received on a specific extension or a
range of extensions and trunk ports: CO, E&M, T1E1, BRI, Mobile and SIP.
To print records of specific extensions, clear the Calls received on All Extensions check box and enter
the desired extension range in From and To.

• To print calls received only on extensions that have not been assigned Access Codes, select the Print
Calls received on Extensions having blank Access Code check box. Make sure in Calls received
from Extensions, you have assigned 0 in the From and To fields.
To print incoming calls according to Call Type,
• Click Calls to expand.
• You can also print incoming calls of different Call Types: Normal calls, calls received using Built-In Auto
Attendant, calls that remained Unanswered, Unanswered calls on Built-In Auto Attendant, and calls
made using DISA.
To print incoming calls received from certain numbers,
• Click Number List to expand.
• To print outgoing calls received from certain numbers, enter the CLI of these numbers in the Number
List of your choice and select the same number list in Calls received from calling numbers
matching with Number List.
• To print calls received without CLI, select the Print Calls received from caller without CLI check box.
To filter calls between specific dates, time or with a specific duration,
• Click Filter Calls to expand.
• To print incoming calls received on a certain date or between a certain time period, set the filter Calls
received between. To print calls made on a particular date, select the same Date, Month and Year in
both fields.

• To print incoming calls received at a particular time, set the Hours and Minutes in 24-hour format calI in
the filter Calls received between 00: 00 and 23:59.
• To print incoming calls that remained unanswered for more than a certain duration, set the filter Calls
remain unanswered for duration more than (sec).
• To print calls that were kept on hold for more than a certain duration, set the filter Calls kept on hold
with duration more than (seconds) to the desired value.
• To print calls with speech duration of a certain duration, set the filter Calls with speech duration more
than (seconds)
• To generate SMDR Report of incoming calls on a particular day, day of the week, or day of the month, set
• You can print the SMDR Report of incoming calls any time you want. You can print the report on a local


Internal Calls
• To print Internal call Report with filters, click Internal Call Report link.
To print calls made/received by particular extensions,

• Click Extension to expand.
• By default, Calls made by All Extensions check box is selected. Hence calls made and calls received by
all the extensions will be printed.
You can also print calls made and calls received by all extensions by selecting the Call Type.
• Select Both as Call Type, to print calls made and received by the extensions.
• Select Caller as Call Type, to print only those calls that were made by the extension.
• Select Receiver as Call Type, to print only those calls that were received by the extension.
• Select None, if you do not want to use the Call Type filter.
To print calls made by a particular extension or a range of extensions, clear the Calls made by All
Extension and set the filter Calls made by Extensions, by entering the extension numbers in the From
and To fields.
If you want to print calls made by a particular extension only, enter the same extension number in both
From and To fields.
You can also print calls made and calls received by these extensions by selecting the Call Type.
• Select Both as Call Type, to print calls made and received by the extensions.

• Select Caller as Call Type, to print only those calls that were made by the extension.
• Select Receiver as Call Type, to print only those calls that were received by the extension.
• Select None, if you do not want to use the Call Type filter.
To print calls with a particular speech duration,
• To print calls with speech duration of a certain duration, set the filter Calls with speech duration more
than (seconds)
• To generate SMDR Report of internal calls on a particular day, day of the week, or day of the month, set
• You can print the SMDR Report of internal calls any time you want. You can print the report on a local


Print SMDR-Report using a Telephone
Print Filter Settings - Incoming Calls

The SA can program the filters as per user's need to generate a customized report. These commands enable the
user to select the type of call reports generated viz. Normal calls, Built-In Auto Attendant calls, Unanswered calls,
Long Speech Duration calls. It is also possible to generate call reports for a range of extensions (SLT, DKP),
different type of trunks (CO, BRI, T1E1PRI etc.).
Filter Commands Values
To set filter to print all normal calls 1072-152-Flag Flag:

0 = Disable
1 = Enable
To set filter to print all Built-In Auto 1072-153-Flag Flag:

Attendant calls 0 = Disable
1 = Enable
To set filter to print all Unanswered 1072-154-Flag Flag:

calls 0 = Disable
1 = Enable
To set filter to print all Built-In Auto 1072-155-Flag Flag:

Attendant Unanswered calls 0 = Disable
1 = Enable
To set filter to print all DISA calls 1072-156-Flag Flag:

0 = Disable
1 = Enable
To set filter to print all calls with 1072-157-Seconds Seconds:000-255

specific call duration
To set filter to print all call with specific 1072-158-Seconds Seconds:000-255

Unanswered duration
To set filter to print all calls with 1072-159-Seconds Seconds:000-255

specific hold duration
To set filter to print all IC calls received 1072-160-Flexible No.- Flexible Number: 000000 to
by the extension Flexible No. 999999
To set filter to print all IC calls received 1072-161-CO-CO CO: 000-128

on the CO
To set filter to print all IC calls received 1072-162-BRI-BRI BRI: 00-32

on the BRI
To set filter to print all IC calls received 1072-163-T1E1PRI-T1E1PRI T1E1PRI: 01-08

on the T1E1PRI
To set filter to print all IC calls received 1072-164-E&M-E&M E&M: 000-128

on the E&M
To set filter to print all IC calls received 1072-165-Mobile-Mobile Mobile: 00-64

on Mobile

To set filter to print all IC calls received 1072-166-SIP-SIP SIP: 00-32

on SIP
To set filter to print all IC calls on or 1072-167-DD-MM-YYYY-DD- Date: 01-31

from date MM-YYYY Month: 01-12
Year: 0000-9999
To set filter to print all IC calls received 1072-168-HH-MM-HH-MM HH: 00-23

At/from-to Time MM: 00-59
To set filter to print all IC calls from 1072-169-Number List Number List: 00-16
numbers matching in a Number list
Note: If you do not want to print calls for a particular type, set the start and end range as 0.
Default the Filters
To default the report generation filters, dial:

• 1072-170
Calls made by all extensions, all CO, All BRI, All E&M and All T1E1PRI are printed.
The Date range is 01-05-2005 to 31-12-2100.
The Time range is 00:00 to 23:59.
The Number List is 02.
By default, All calls are printed as per the set filters.

SA should issue an explicit start command, to initiate the report generation.
To start/abort report generation, dial:

• 1072-171-Flag
Where,
Flag Meaning
0 Abort
1 Start
By default, the Flag is 0.
Once the Start command is issued, the report generation stops only after the complete report based on the
filters set is generated. SARVAM UCS provides a facility to abort the report generation in midway (1072-
171-0). Once the report generation is aborted, then it has to be explicitly started (1072-171-1) if the report
is required again.
The Scheduled Report Generation

SA can issue the command to start report, automatically at a particular time (Daily) or at a particular time on a
particular day of the week (Weekly) or at a particular time on a particular date of a month (Monthly).
Once these parameters are programmed and the report generation is enabled, there is no need of any initiation of
the command.

Enable/disable Scheduled Report
To enable/disable the generation of daily/weekly/monthly reports, dial:

• 1072-172-Flag
Where,
Flag Meaning
0 Disable Schedule Report Generation
1 Enable Daily Scheduled Report
2 Enable Weekly Scheduled Report
3 Enable Monthly Scheduled Report
By default, the generation report is disabled.
Daily Reports
To program the time for daily scheduled reports, dial:

• 1072-173-HH-MM
Where,
HH and MM is in 24 hour format.
By default, HH:MM is 18:00.
Weekly Reports
To program the day and time for weekly scheduled reports, dial:
• 1072-174-Day-HH-MM
Where,
Day is from 1 to 7 (1 is Sunday, 2 is Monday ……… and 7 is Saturday).
By default, HH:MM is 10:00 and Day is 2.
Monthly Reports
To program the date and time for monthly scheduled reports, dial:
• 1072-175-Date-HH-MM
Where,
By default, HH:MM is 10:00 and Date is 01.
SARVAM UCS provides a facility to Abort the scheduled report generation midway (1072-171-0). This
aborts the current report generation but does not affect any other scheduled report generation. The
abortion of a report does not affect its consecutive schedule. That is if a daily report is aborted on Monday,
it does affect the report generation schedule of Tuesday.

Print Filter Settings - Internal Calls
Filter Command Value
To set filter to print all internal calls 1072-137-Flexible No.- Flexible Numbers: 000000-999999
from an extension. Flexible No.-Type Type:
0 - Do not print the calls made to or from
the extension.
1 - Print calls made to this extension
2 - Print calls made from this extension
3 - Print all calls made to/from this
extension
By default, Type is 3 and seconds is 000.
To set filter to print all internal calls 1072-138-Seconds Seconds: 000-999

with duration greater than specified.

SA should issue an explicit start command to initiate the report generation.
To start/abort Report generation, dial:

• 1072-141-Flag
Where,
Flag Meaning
0 Abort
1 Start
By default, flag is 0.
Once the start command is issued, the report generation stops only after the complete report based on
the filters set is generated. SARVAM UCS provides a facility to abort the report generation in midway
(1072-141-0). Once the report generation is aborted, then it has to be explicitly started (1072-141-1) if
the report is required again.

SA should issue the command to start the report, automatically at a particular time (Daily) or at a particular time on
a particular day of the week (Weekly) or at a particular time on a particular date of a month (Monthly).
Once these parameters are programmed and the report generation is enabled, there is no need of any initiation
command.
• To enable/disable the generation of daily/weekly/monthly scheduled reports, dial:

1072-142-Flag

Where,
Flag Meaning
Daily Reports

• 1072-143-HH-MM
Where,
By default, HH:MM is 18:00
Weekly Reports:
• 1072-144-Day-HH-MM
Where,
Monthly Reports:
• 1072-145-Date-HH-MM
Where,
By Default, HH:MM is 10:00 and Date is 01.
SARVAM UCS provides a facility to abort the scheduled report generation in midway (1072-141-0). This
Print Filter Settings - Outgoing Calls

SA can configure various filters to generate a report as per the user's requirement. It is possible to program the
following filters:
To set filter to print all calls 1072-103-CO-CO CO: 001-128

terminated on CO
To set filter to print all calls 1072-104-BRI-BRI BRI: 01-32

terminated on BRI

To set filter to print all calls 1072-105- T1E1PRI - T1E1PRI T1E1PRI : 01-08
terminated on T1E1PRI
To set filter to print all calls 1072-106-E&M-E&M E&M: 001-128

terminated on E&M
To set filter to print all calls 1072-107-Mobile-Mobile Mobile: 00-64

terminated on Mobile
To set filter to print outgoing 1072-108-SIP-SIP SIP: 00-32

calls terminated on SIP
To set filter to print outgoing 1072-109-Group No.-Group No. Dept Group: 00 to 99

calls for department billing
group
To set filter to print all calls 1072-110-DD-MM-YYYY-DD-MM- Date: 01-31.

made on or from date YYYY Month: 01-12.
Year: 0000-9999.
To set filter to print all calls 1072-111-HH-MM-HH-MM HH: 00-23

made at or in between a time MM: 00-59
To set filter to print all calls of 1072-112-External Number List Number List = 01-16.
numbers matching in a Number
list
To set filter to print all calls 1072-113-Seconds Seconds = 000-999.

duration more than specified
To set filter to print all calls with 1072-114-Units Unit = 0000-9999

units more than specified
To set filter to print all calls 1072-115-Account Code-Account Account Code = 0000-9999
made using Account Code Code
To set filter to print all calls 1072-102-Flexible No.-Flexible Flexible Numbers: 000000 to
originated on extensions No. 999999
To set filter to print calls 1072 - 183 - CO - CO CO = 000-128

originated on CO
To set filter to print calls 1072 - 184- BRI- BRI BRI = 00-32
originated on BRI
To set filter to print calls 1072 - 185 - T1E1PRI - T1E1PRI T1E1PRI = 01-08
originated on T1E1PRI
To set filter to print calls 1072 - 186 - E&M- E&M E&M = 000-128
originated on E&M
To set filter to print calls 1072 - 187 - Mobile- Mobile Mobile = 00-64
originated on Mobile
To set filter to print calls 1072 - 188 - SIP- SIP SIP = 00-32
originated on SIP
To change the CPU from Active 1072-189

to Standby mode

To default the report generation filters, dial:
• 1072-120
By default, All calls are printed as per the set filters.
Calls made by all extensions, all CO, ALL BRI, ALL E&M and All T1E1PRI are printed.
The Date range = 01-05-2005 to 31-12-2100.
The Time range = 00:00-23:59.
The External Number list = 02.
Units = 0000.
Seconds = 000.
Delete calls made to/from an extension
The SARVAM UCS supports Deletion of SMDR Calls (From SA Mode) made from a particular extension or a range
of extensions.
To delete the calls made by an extension or a range of extensions, dial:

• 1072-131-Flexible Number-Flexible Number
To delete the calls made on a Date or from a Date, dial:

• 1072-132-DD-MM-YYYY-DD-MM-YYYY (The format of the date depends on the date format of the
system)
Manual Report Generation:

SA should issue an explicit start command to initiate the report generation.
To start/abort Report generation, dial:

• 1072-121-Flag
Where,
Flag Meaning
0 Abort
1 Start
Once the start command is issued, the report generation stops only after the complete report based on
the filters set is generated. SARVAM UCS provides a facility to abort the report generation in midway
(1072-121-0). Once the report generation is aborted, then it has to be explicitly started (1072-121-1) if
the report is required again.

SA should issue the command to start the report, automatically at a particular time (Daily) or at a particular time on
a particular day of the week (Weekly) or at a particular time on a particular date of a month (Monthly).
Once these parameters are programmed and the report generation is enabled, there is no need of any initiation
command.
To enable/disable the generation of daily/weekly/monthly scheduled reports, dial:

• 1072-122-Flag

Where,
Flag Meaning
Daily Reports

• 1072-123-HH-MM
Where,
By default, HH:MM is 18:00.
Weekly Reports
• 1072-124-Day-HH-MM
Where,
Monthly Reports
• 1072-125-Date-HH-MM
Where,
By Default, HH:MM is 10:00 and Date is 01.
SARVAM UCS provides a facility to abort the scheduled report generation in midway (1072-121-0). This
The Offline report for Outgoing calls looks like shown below:
SMDR Outgoing Calls Report As On 27-02-2017(Mon) At 13:21

-------------------------------------------------------------------------------
Originated on Terminated on
Extension : ALL CO :001 To 064 CO :001 To 064
Date : 01-05-2005 To 31-12-2037 BRI :001 To 032 BRI :001 To 032
Time : 00:00 To 23:59 T1E1 :001 To 008 T1E1 :001 To 008
Number List: 02 E&M :001 To 032 E&M :001 To 032
Department : 00 To 00 MOB :001 To 040 MOB :001 To 040
Dur - Sec : 000 SIP :001 To 099 SIP :001 To 099

Units : 0000
Accout No : 000 To 000
Authrty Cod: 000 To 000
-------------------------------------------------------------------------------
SrNo Extn AuC Trnk Called No Date Time Dur Unit Amount R
-------------------------------------------------------------------------------
-------------------------------------------------------------------------------
Total Calls : 0
Total Dur-Sec : 0
Total Units : 0
Total Amount : 0.00
-------------------------------------------------------------------------------
Trunk : C=CO, B=BRI, P=T1E1, E=E&M, M=MOB, V=SIP
Extension: S=SLT, D=DKP, G=MAG, I=SIP Extn, R=Virtual Extn, N=ISDN Extn
Additional Ports: X=MODEM,
R(CALL TYPE):O=DISA,A=Auto Redial,E=External Forwarded,G=Gateway
MATURITY: C = CPD, U = Unanswered, R = Reversal, D = Delay, I = Connect
AuC = Authority Code, T = Transfer
-------------------------------------------------------------------------------
SARVAM UCS V1R4 Page : 1

-------------------------------------------------------------------------------
Time : 00:00 To 23:59 T1E1 :001 To 008 T1E1 :001 To 008
Units : 0000
-------------------------------------------------------------------------------
-------------------------------------------------------------------------------
1 3303 000 V004 9999 16-04-16 10:30:08 13 1 1.10 I
-------------------------------------------------------------------------------
Total Calls : 1
Total Dur-Sec : 13
Total Units : 1
Total Amount : 1.10
-------------------------------------------------------------------------------
-------------------------------------------------------------------------------

-------------------------------------------------------------------------------
Time : 00:00 To 23:59 T1E1 :001 To 008 T1E1 :001 To 008
Units : 0000
-------------------------------------------------------------------------------
-------------------------------------------------------------------------------
1 3303 000 V004 9999 16-04-16 10:30:08 13 1 1.10 I
2 3303 000 V004 7777 16-04-16 10:32:03 10 1 1.10 I
-------------------------------------------------------------------------------
Total Calls : 2
Total Dur-Sec : 23
Total Units : 2
Total Amount : 2.20
-------------------------------------------------------------------------------
-------------------------------------------------------------------------------

Station Message Detail Recording-Storage
What's this?
SARVAM UCS stores SMDR of Incoming calls, Outgoing Calls and Internal Calls. The call records are stored in the
SMDR buffer. For this, SMDR storage for these types of calls must be enable, and if required further filters can be
set.
SARVAM UCS can store 6000 outgoing calls, 4999 incoming calls, and 999281 internal calls in the SMDR buffer.
Once the SMDR buffer is full, the next call is stored in place of the oldest call in the SMDR buffer, using the First In
First Out (FIFO) logic.
The buffer can be cleared at any time from the System Administrator mode.
The SMDR buffer data is maintained even during power failures. However it is advisable to take frequent printouts
of the calls to avoid accidental loss of the data.
How to configure
To enable storage of SMDR of Outgoing, Incoming and Internal Calls, you must enable this feature in the system
and set the storage filters as per your requirement.
Configuring SMDR-Storage using Jeeves

• Click Storage Filters to open the page.
281. ETERNITY LENX/MENX can store 9999 Outgoing Calls, 9999 Incoming Calls and 2000 Internal Calls.

• To configure storage filters for outgoing calls, click SMDR Storage - Outgoing Calls to expand,
• Select the Store Outgoing Calls check box to enable storage of outgoing calls as per the filters you
set. If outgoing call storage is disabled, no outgoing call will be stored.
• You can limit the storage of calls to certain numbers. Select a number list and enter the desired called
party numbers in the selected list. In Store Calls of Called Number matching with Number List,
select the same list number.
• If you want outgoing calls that exceed as certain duration to be stored, set the filter Store Calls with
speech duration more than (sec) to the desired duration. All outgoing calls with duration greater than
this value, will be stored.
• If you want outgoing calls exceeding certain metering units to be stored, set the filter Store Calls with
metering units more than (units) to the desired value. All outgoing calls that have metering units
greater than this value will be stored.
• Outgoing calls made by an extension user can be transferred to another extension. In such cases, you
may enable Call Splitting if you want to charge the amount to each extension according to the duration
of speech that each extension was involved in the call.
• If Call Splitting is disabled, you have the option of charging the call amount either to the extension that
originally made the call, i.e. the Originating Extension, or to the extension that was last in speech on the
call, i.e. the Terminating Extension.
In the When Call Splitting is OFF, charge calls to field, select the desired extension you want to
charge the call to as Originating Extension or Terminating Extension.
• Select the Store Unanswered Outgoing calls check box to enable storage of unanswered outgoing
calls as per the filters you set. If this check box is disabled, unanswered outgoing calls will not be
stored.
• Click Submit to save your Outgoing call storage filter settings.

• To configure storage filters for incoming calls, click SMDR Storage - Incoming Calls to expand,
• Select the Store Incoming Calls check box to enable storage of incoming calls as per the filters you
set. If incoming call storage is disabled, no incoming call will be stored.
• If you want incoming calls that exceed as certain duration to be stored, set the filter Store Calls with
speech duration more than (sec) to the desired duration. All calls with duration greater than this
value, will be stored.
• If you want incoming calls that remain unanswered for certain duration to be stored, set the filter Store
Calls remaining un-answered for more than (sec) to the desired duration. All calls with duration
greater than this value, will be stored.
• If you want incoming calls that were kept on hold for a certain duration to be stored, set the filter Store
Calls kept on hold for more than (sec) to the desired duration.
• If you want all calls, except calls received using Auto Attendant to be stored, select the Store Normal
Calls check box.
• If you want calls received on Auto Attendant to be stored, select the Store calls received on Built-In
Auto Attendant check box.
• If you want all calls, that remained unanswered to be stored, select the Store Unanswered Calls
check box.
• If you want all calls received using Auto Attendant that remained unanswered to be stored, select the
Store Unanswered calls from Built-In Auto Attendant check box.
• If you want calls made using DISA, select the Store DISA Calls check box.
• Click Submit to save Incoming Call storage filters.

• To configure storage filters for internal calls, click SMDR Storage - Internal Calls to expand,
• Select the Store Internal Calls check box to enable storage of internal calls as per the filters you set.
• To store internal calls that exceed as certain duration, set the filter Store Calls with speech duration
more than (sec) to the desired duration. All internal calls with duration greater than this value, will be
stored.
• Click Submit to save Incoming Call storage filters.
Configuring SMDR-Storage using a Telephone
SMDR Storage-Incoming Calls

Enable SMDR-IC Storage
To set SMDR storage mode, dial:

• 2901-Storage Flag
Where,
Storage
Meaning
Flag
0 Don't store the incoming calls
1 Store the incoming calls
By default, all the calls are stored as per the filters set.
Filter Settings - Incoming Call Storage
These commands enable the user to select the type of calls to be stored, namely, All calls, Trunk wise calls, Un-
answered calls, Built-In Auto Attendant calls, etc. Calls will be stored only if duration, exceeds the set value.
Flag:
Store Normal Calls 2902-Flag 0 = Don't Store
1 = Store
Flag:
Store Built-In Auto Attendant Calls 2903-Flag 0 = Don’t Store
1 = Store
Flag:
Store Unanswered Calls 2904-Flag 0 = Don't Store
1 = Store

Flag:
Store Unanswered Built-In Auto Attendant Calls 2905-Flag 0 = Don't Store
1 = Store
Flag:
Store DISA Calls 2906-Flag 0 = Don't Store
1 = Store
Store Calls-Speech Duration More than 2907-Seconds Seconds:000-999
Store Calls-Unanswered Duration for more than 2908-Seconds Seconds:000-999
Store Calls-Hold Duration for more than 2909-Seconds Seconds:000-999
Default IC Storage Filters
To default the incoming call storage filters, dial:

• 2915
By default,
All Normal, Built-In Auto Attendant, Unanswered, Unanswered Built-In Auto Attendant, DISA calls will be
stored.
The speech duration is set to 000.
The unanswered duration is set to 000.
The Hold Duration is set to 000.
SMDR Storage-Internal Calls

Enable SMDR-Internal Storage:
To set SMDR storage mode, dial:

Where,
Storage
Meaning
Flag
0 Don't store the internal calls
1 Store the internal calls
By default, all the calls are stored as per the filters set.
Filter Settings - Internal Call Storage
Filter Command Value
Store Calls-Speech Seconds: 000-

2802-Seconds
duration 999
Default Internal Storage Filters
To default the internal call storage filters, dial:

• 2815

SMDR Storage-Outgoing Calls
Enable SMDR-Outgoing Storage
To set SMDR storage flag, dial:

Where,
Storage
Meaning
Flag
0 Don't store the outgoing calls
1 Store the outgoing calls
By default, the Storage Flag is 1.
Filter Settings - Outgoing Call Storage
These commands enable the user to select the type of calls to be stored, namely, destination number wise,
duration wise or cost wise.
Destination Wise
It is possible to store outgoing calls selectively depending on the destination numbers. The SARVAM UCS supports
this feature in association with Number Lists. An outgoing call will be stored only if the number matches with an
entry in the Number List assigned.
To assign a Number list containing numbers for call storage, dial:

• 2702-Number List
Where,
Number List is 01-16.
By default, Number List assigned is 02.
Duration wise
Sometimes it is required to filter out the calls of small durations. System will not store the calls with duration less
than duration, programmed.
To set the filter of call duration, dial:

• 2703-Seconds
Where,
By default, Seconds is 000.
Unit wise
Sometimes it is required to filter out the calls based on Call units. System will not store the calls with units less than
the units programmed:
To set the filter for call Units, dial:

• 2704-Units
Where,
Units are from 0000 to 9999.
By default, Units is 0000.

Default OG Storage Filters
To default the outgoing calls storage filters, dial:

• 2715
By default, Number List is 02, Seconds is 000 and Units is 000.
Call Toggle
To set the Call Toggle flag, dial:

• 2716-Toggle Flag
Where,
Toggle Flag Meaning
0 Toggle OFF
1 Toggle ON
By default, Toggle is ON.
To set the originating flag, dial:

• 2717-Originating Flag
Where,
Originating Flag Meaning
0 OFF
1 ON
By default, the flag is originating extension.
When Call Toggle Flag is set OFF, then only above command (2717) can be effective. Refer chapter “Call
Toggle” for more details.
How to use
The SMDR stored in the buffer can be cleared at any time from the System Administrator mode, using Jeeves or by
dialing SA commands from an extension phone.
To delete SMDR records using Jeeves,

• Under SMDR Management, click SMDR - Delete Call Record to open the page.
• To delete all records in the internal SMDR buffer, select Delete All Internal Calls check box.
• To delete all records in the Incoming SMDR buffer, select the Delete All Incoming Calls check box.
• If you want to delete all records in the Outgoing SMDR buffer, select the Delete All OG Calls radio button.
• You can also delete records of outgoing calls selectively, i.e. delete only records of outgoing calls made by
a particular extension or a range of extensions, or calls made between a certain period.
• To delete records of outgoing calls selectively, select the Delete Selective OG Calls radio button.
• To delete calls made by a particular extension or a range of extensions, select the Delete OG Calls
made by Stations button.
• In the first edit box, enter the number of the first extension in the range. In the second box, enter the
number of the last extension in the range. If you want to delete the records of a particular extension,
enter the same extension number in both fields.
• To delete the records of outgoing calls made on a particular date or during a certain period, select the
Delete calls made between radio button. Select the start and end Date, Month and Year for this
period. If you want to delete the records of a particular date, enter the same date as start and end.
• The SMDR buffer will be cleared according to the settings you enabled on this page.
To delete SMDR records from an extension phone,
To delete all Incoming calls:
• Dial 1072-180-Reverse SA Password

To delete all Internal calls:
To delete all Outgoing calls:
To delete calls made by a particular extension or range of extensions:
• Dial 1072-131-Extension Number-Extension Number
To delete calls made by on a particular date or a between a certain time period:
• 1072-132-DD-MM-YYYY-DD-MM-YYYY (The format of the date depends on the Date Format of the
system)
• Exit SA mode.

System Activity Log
What’s this?
The SARVAM UCS monitors all its activities and maintains records of these activities in the System Activity Log.
The System Activity Log has a buffer capacity of 500 records. The Activity Log stores records using the FIFO
method.
This log can be printed on a local printer or downloaded on a computer in form of a report. The activity log can be
printed or downloaded in two modes:
• Online: The activity report is printed/downloaded as and when an activity occurs.
• Report (Offline): The activity report is printed/downloaded whenever desired. In the Offline mode, the last
500 activities recorded by the system are printed/downloaded.
The System Administrator can print/download System Activity Log, online or offline using any serial device
connected to the COM Port of SARVAM UCS.
SARVAM UCS also supports Syslog Client for System Activity Logs. The Syslog Client enables the system to send
activity logs in syslog format to the remote ‘Syslog Server’. You can view the logs on the remote server.
You may use Syslog for System Activity Log, if you have no spare COM Port on your SARVAM UCS.
How it works
• A destination port, serial or ethernet, must be assigned for activity logs. The system will send the activity
log to this port.
• If the System Administrator extension is a DKP or an Extended IP Phone, a DSS Key can be assigned for
System Activity Log.
• Each activity is stored in the Activity Log in this format:

<DD-MM-YYYY> < HH:MM:SS> <Activity Text>
• Whenever an activity is recorded by the system, the DSS key, if assigned for this feature on the System
Administrator’s DKP/Extended IP Phone extension, is turned ON.
• The System Administrator can view the activity log by pressing the DSS key (if assigned). The DKP/
Extended IP Phone of the System Administrator will display the activity in this format:
DD-MM HH:MM <Activity Index>
The format of the Date will be DD-MM or MM-DD as per Date Format selected in the Real Time Clock
settings of the system.

Index of the Type of Activities recorded in the System Activity Log:
Event Index Activity Description
xx = 2 digit Version number

01 System VxxRyy.zz Started yy = 2 digit Revision number
zz = Either 1 digit or 2 digits (not always 2 digits)
02 Default Configuration Loaded
xx = Slot number
Card Present: Slot=xx, Type: CARD TYPE
03 CARD TYPE = Card Type with version revision
nnnnnn = max. 6 digit flexible number assigned to SLT

04 SLT Normal: nnnnnn, Slot=xx, Port=yy xx = Slot number
yy = Port number
nnnnnn = max. 6 digit flexible number assigned to DKP

05 DKP Normal: nnnnnn, Slot=xx, Port=yy xx = Slot number
yy = Port number
xx = Slot number
BRI Layer-1 Up : Slot=xx, Port=yy
06 yy = Port number
xx = Slot number
T1E1 Layer-1 Up : Slot=xx, Port=yy
07 yy = Port number
08 SMDR-IC buffer deletion: nnnnnn nnnnnn = max. 6 digit flexible number of the station
09 SMDR-OG buffer deletion: nnnnnn nnnnnn = max. 6 digit flexible number of the station
10 SMDR-Internal buffer deletion: nnnnnn nnnnnn = max. 6 digit flexible number of the station
11 SE Access From: nnnnnn nnnnnn = max. 6 digit flexible number of the station
12 SA Access From: nnnnnn nnnnnn = max. 6 digit flexible number of the station
13 Emergency Number Dialed: nnnnnn nnnnnn = max. 6 digit flexible number of the station
14 Log In nnnnnn nnnnnn = max. 6 digit flexible number of the station
15 Log Out nnnnnn nnnnnn = max. 6 digit flexible number of the station
Reg Fail, Authentication PW Invalid,

xx = SIP Trunk number
16 SIPTrk=xx
Reg Fail, Config Parameters Invalid,

xx = SIP Trunk number
17 SIPTrk=xx
Stack Construct Authenticate Fail, Slot=xx, xx = Slot number

18 Port=yy yy = Port number
Stack Construct IP Addr Invalid, Slot=xx, xx = Slot number

19 Port=yy yy = Port number
xx = Port Type
20 Call Budget exhausted, xx - yyy, NAME yy = Port offset
NAME = Name of the Trunk
21 Reserved Reserved for future use
23 Personal Mailbox is full for Extension

nnnnnn = max. 6 digit flexible number of the station
nnnnnn

xx = Slot number
VMS USB Memory is Full
24 yy = Port number
Restart due to change Network Para xx = Slot number

25
Slot=xx, Port=yy yy = Port number
Restart due to change SIP Para Slot=xx, xx = Slot number

26
Port=yy yy = Port number
Restart due to Master request Slot=xx, xx = Slot number

27
28 CPU Card- is in Active Mode.
29 CPU Card- is in Standby Mode.
30 Power Supply Card- O.K.
31 CPU- Ethernet link up.
32 CPU- Ethernet link down.
33 Data Port link up, Port=xx
34 Data Port link down, Port=xx
35 Usb is full!!
36 VMS USB Memory 80% Full
37 VMS USB Memory Usage in Limit
38 DSP Frame Sync error, Slot=xx xx = Slot number
39 DSP Drop error, Slot=xx xx = Slot number
40 DSP Time Out error, Slot=xx xx = Slot number
41 DSP DMA Address error, Slot=xx xx = Slot number
Message Notification Retry Count over for

42 nnnnnn = max. 6 digit flexible number of the station
nnnnnn
Default Password restored for Authority

43 nnn = 3 digit Authority Code
Code:nnn
Call between nnnnnn and COxxx dropped

44 nnnnnn = max. 6 digit flexible number of the station
as prone to fake
xxxxxx = max. 6 digit flexible number of the station

from which the Emergency Call was made
Emergency Call of xxxxxx is Acknowledged
45
by yyyyyy.
yyyyyy = max. 6 digit flexible number of the station
which had acknowledged the Emergency call.
System fan has failed. VoIP module will

47
stop working
System fan has failed. System will go in

48
Thermal Sleep Mode

xx = Slot number
50 T1E1 Layer-2 Up : Slot=xx, Port=yy
yy = Port number
xx = Slot number
51 BRI Layer-2 Up : Slot=xx, Port=yy
yy = Port number
52 System Restart Using SMS
DD = Date , MMM = Month, HH: Hour, MM: Minute at

53 Forward DST Applied at: DD-MMM HH:MM
when Forward DST is applied
Backward DST Applied at: DD-MMM DD = Date , MMM = Month, HH: Hour, MM: Minute at
54
HH:MM when Backward DST is applied
When DST mode is set to either Manual or Scheduled

from Disabled or vice versa then first message is
"DST Config Change: Enabled/Disabled
55 displayed. If any change is made in DST -Manual/
DST Config Change:"
Scheduled Parameters the second message is
displayed
xx = 2 digit Version number

System VxxRyy.zz Started with System
56 yy = 2 digit Revision number
Command App
zz = Either 1 digit or 2 digits (not always 2 digits)
Command will execute using System At every power-on when the System is up using
57
Function System Commands
"IP_ADDRESS" should be the IP Address from which

the last failure attempt was made.
LOGIN_NAME Login blocked for
58
IP=IP_ADDRESS
LOGIN_NAME should be actual Login ID i.e. SE or SA
or FDU.
"IP_ADDRESS" should be the IP Address from which

IP= IP_ADDRESS blocked for Auto Prov. of the last failure attempt was made.
59
User SIP_ID
SIP_ID should be SIP ID of user.
IP_ADDRESS:PORT should be the IP Address and

60 "IP= IP_ADDRESS:PORT blocked
Port from which the last failure attempt was made.
Sendto Network Error for Connection ID XXX is the connection id for which the call is released
61
XXX by the VoIP.
62 Shadow+ file removed
63 General Mailbox is Full
64 Reserved for future use
65 Reserved for future use
When anyone logs into the SE, SA or FDU mode.

66 XYZ Login IP Address
XYZ is the login mode - SE/SA or FDU.
When the system is restarted from the Web GUI or

Extension number using commands.
67 System Restart by YY
YY is the IP Address or the Extension Number.

When the CPU card is restarted from the Web GUI

68 CPU-X Restart by IP Address
XX is CPU 1 or 2
When any card is restarted from the Web GUI or

Extension Number using commands.
68 Card-XX Restart by YY
XX is slot number from 01 to 27 and YY is the IP
Address or Extension number.
When installed in the Hotel mode, the SARVAM UCS captures Hotel-Motel Activity Log. To know more, see the
SARVAM UCS Hospitality System Manual.
How to configure
The two functional parts of system activity log are: Storage and Report Generation in the Online or Report modes.
To be able to use this feature, you must enable storage of Activity Logs, and assign the Syslog Server address as
Destination Port for the logs.
The destination port may be a COM Port or Ethernet Port.
If you want to use Syslog Server, you must assign the IP Address of the remote Syslog server as the Destination
Port for the System Activity Log.
If the System Administrator phone is a DKP or an Extended IP Phone, you may assign a DSS key for System
Activity Log.
For instructions on configuring a DSS key on a DKP, see “DSS Keys Programming”.
For instructions on configuring a DSS key on an Extended IP Phone, see “DSS Key Settings” in “Configuring Matrix
SPARSH VP330”, “DSS Key Settings” in “Configuring Matrix SPARSH VP248”, “DSS Key Settings” in “Configuring
Matrix SPARSH VP310” and “DSS Key Settings” in “Configuring Matrix SPARSH VP510”.
Configuring System Activity Log using Jeeves

• Open Jeeves
• Under Configuration, click System Log link.

• Click System Activity Log.
• By default, System Activity Log Storage is Enabled. Select Disable, if you do not want the system to
keep a record of all the system activities.
• To generate System Activity Log - Online, that is, as and when the activity occurs, select Destination
Port for Online SAL from the following options:
• COM Port: Select a communication port if you want to use a serial device to capture the logs. Make
sure the device is connected to the COM port.
• Ethernet: Select Ethernet port if you want to use the remote syslog server for the logs.
• USB to COM Port: Select USB to COM port if you want to use the External USB as a COM port for
capturing the logs. Make sure you connect the USB to COM Converter to the USB Port.
• In Port, enter the port of the remote Syslog Server. Valid port range: 1025 to 65535;514.
By default, no destination port is assigned for Online SAL.
• To generate System Activity Log - Report, that is, offline, whenever desired, select Destination Port for
SAL Report from the options COM Port, Ethernet Port or USB to COM Port. Default: None.
If you selected Ethernet as the Destination port,

Configuring System Activity Log using a Telephone

To set enable/disable SAL storage, dial:

Where,
Storage Flag is
0 for Disable
1 for Enable
Default: Enable
To select the Destination Port for SAL Online, dial:

• 6402-Port
Where,
Port is
0 for None
1 for COM Port
2 for Ethernet Port
3 for USB to COM Port
Default: None
If you selected Ethernet Port as Destination port, to assign the IP Address for the Ethernet Port, dial:
• 6404-IP Address
To assign Port for the IP Address, dial:

• 6405-Port
Where,
Default: 514.
To select the Destination Port for SAL Report, dial:

• 6403-Port
Where,
Port is
0 for None
1 for COM Port
2 for Ethernet Port
Default: None
If you selected Ethernet Port as Destination port, to assign the IP Address for the Ethernet Port, dial:
• 6406-IP Address

• 6407-Port
Where,
Default: 514.
To restore default values of the system activity log parameters, dial:

• 6410
• Exit SE Mode.
How to use
You can start and stop System Activity Log - Online and Report from the System Administrator mode using Jeeves
or dialing SA Commands from an extension phone.
To start/stop report generation using Jeeves,
• Open Jeeves.
• Click the System Activity Log link.
• To start System Activity Log - Online, click the Start button.

• To stop System Activity Log - Online, click the Abort button.
• To start System Activity Log - Report, click the Start button.
• To stop System Activity Log - Report, click the Abort button.
• To clear System Activity Logs from the buffer, click the Clear SAL button.
By default, the online and offline reports are printed on the destination port assigned by you.
• To view the System Activity Log on your computer screen, click the View button.
System Activity Log appears on your screen.

• To print this report on the local printer connected to your computer, click the Print button. An alert
message will appear.
• Click the OK button. The System Activity Log - Report will be printed on the local printer connected to
your computer.
Make sure you have selected ‘NONE’ as the destination port to print the report on the local printer
connected to your computer.
•You may log out of the System Administrator mode.
To generate Activity logs from an extension phone,
To start/stop Online log generation:

To start/stop offline Report generation:

To clear Activity Logs,

• Exit SA mode.
You may print the logs captured on the Syslog Server after suitable modification of the format.
The Online System Activity Log report would look like this:
30-01-2017 17:11:11 Card Present: Slot=07, Type: DKP16 V3R0

The Offline System Activity Log report would look like this:
SYSTEM ACTIVITY LOG AS ON 27-02-2017 Mon AT 12:00

-------------------------------------------------------------------------------
30-01-2017 17:10:57 System V01R02.0 Started with System Command App
30-01-2017 17:11:11 Card Present: Slot=29, Type: VoIP NX CPU V1R4
30-01-2017 17:11:11 Card Present: Slot=07, Type: DKP16 V3R0
30-01-2017 17:11:11 Card Present: Slot=33, Type: SWITCH V0R0
30-01-2017 17:11:29 DKP Normal : 3001 , Slot=07, Port=01
30-01-2017 17:58:02 Restart due to change Network Para Slot=29, Port=01
30-01-2017 17:58:03 Stack Construct IP Addr Invalid Slot=29, Port=01
31-01-2017 09:17:03 Stack Construct IP Addr Invalid Slot=29, Port=01
-------------------------------------------------------------------------------
SARVAM UCS V1R4 Page: 1

System Activity Log Display
What’s this?
The SARVAM UCS provides a facility to display the last activity monitored by the system on the System
Administrator’s extension phone. The system also provides you the facility to view all the activities through Jeeves.
To know more, see “How to use” in “System Activity Log”.
How to use
To be able to use this feature optimally, the System Administrator extension phone must be a DKP or an Extended
IP Phone, and a DSS Key must be assigned on the phone to System Activity Log Display.
For instructions on configuring DSS Keys on a Digital Key phone, see “DSS Keys Programming”.
For instructions on configuring DSS Keys on Matrix Extended IP Phone, see “DSS Key Settings” in “Configuring
“Configuring Matrix SPARSH VP310” and “DSS Key Settings” in “Configuring Matrix SPARSH VP510”.
To view the System Activity Log from System Administrator Mode,
• Go Off-hook.
• Press the DSS key assigned to System Activity Log Display.
OR
• Dial 1072-009
• The last recorded Activity log appears on your phone’s display in this following format: Date-Time-Activity
Index
The Date and Time are in <DD-MM-YYYY HH:MM:> format
The Activity Index is a two digit number from 01 to 23.
See System Activity Log Activity Index table in “System Activity Log”.
The date and month format will be DD-MM or MM-DD as per date format set in the system. See “Real Time
Clock (RTC)” for setting the date format.

System Fault Log
What’s this?
The SARVAM UCS maintains a log of all system faults. The system Fault Log has a buffer capacity of 500 records.
The Fault Log stores records using the FIFO method.
The System Fault log can be printed on a local printer or downloaded on a computer in form of a report. The report
can be printed or downloaded by the System Administrator in two modes:
• Online: The fault report is printed/downloaded as and when a fault occurs.
• Report (Offline): The faulty report is printed/downloaded whenever desired. In the Report (Offline) mode,
the last 100 faults recorded by the system are printed/downloaded.
The System Administrator can print/download System Fault Log, Online or Report using any serial device
connected to the COM Port of SARVAM UCS.
Matrix SARVAM UCS also supports Syslog Client for System Fault Logs. The Syslog Client enables the system to
send fault logs in syslog format to the remote ‘Syslog Server’. You can view the logs on the remote server.
You may use Syslog for System Fault Log, if you have no spare COM Port on your SARVAM UCS.
How it works
• A destination port for sending the report must be selected to which the system can send the log.
• If the System Administrator extension is a DKP or an Extended IP Phone, a DSS Key can be assigned for
System Fault Log.
• Whenever a fault is detected, the LED of the Fault Log DSS key, if assigned, is turned ON.
• If more than one DKP/Extended IP extension is assigned Fault Log DSS Key, the LED of all keys will be
turned ON.
• The System Administrator must acknowledge the Fault indication by pressing the Fault Log key or by
dialing the Fault Log access code. The LED of the Fault Log key is turned OFF.

The different fault events that are logged are summarized in this table:
Card Absent: Slot=xx, Type: CARD xx = Slot number

01
TYPE CARD TYPE = Card Type with version revision
nnnnnn = max. 6 digit flexible number assigned to DKP

02 DKP Absent: nnnnnn, Slot=xx, Port=yy xx = Slot number
yy = Port number

03 SLT Short : nnnnnn, Slot=xx, Port=yy xx = Slot number
yy = Port number

SLT Standby-Open: nnnnnn, Slot=xx, xx = Slot number
04
xx = Slot number
05 BRI Layer-1 Down : Slot=xx ,Port=yy
yy = Port number
xx = Slot number
06 T1E1 Layer-1 Down : Slot=xx ,Port=yy
yy = Port number
07 Reserved
08 Com Port Link Break Down
09 RTC Failure Reserved for future use
VOPP Program Download Fail, Slot = xx, xx = Slot number

10
Mod = y y = VoIP module number
xx = Slot number
11 VOPP - SYNC Failed, Slot = xx, Mod = y
y = VoIP module number
12 Registration Timer Fail, SIP Trunk=xx xx = SIP Trunk number
13 CPU Card - gets Absent
14 Power Supply Card- gets Absent
15 HPI Queue Fail, Slot=xx xx = Slot number
16 Slave DSP Alive Error, Slot=xx xx = Slot number
17 SMTP Error Codea - xx xx = Error Code Number
VOPP - Failed to Open Kernel File, Slot =

18 xx = Slot number
xx
19 VOPP - Failed to Boot, Slot = xx xx = Slot number

STRING: One of the following will be displayed:

3.5 VDC Module 1 Fault
3.5 VDC Module 2 Fault
5 VDC Module Fault
-27 VDC Module 1 Fault
Power Supply Module Fail (LENX):
20 -87 VDC Module 1 Fault
STRING, Power Supply VER. Card x
Over Temperature Alarm
VER.: Version with Version Revision
x = Card Number (1 or 2)
System fan has failed. VoIP module has

21
stopped working
xx = Slot number
22 T1E1 Layer-2 Down: Slot=xx ,Port=yy
yy = Port number
xx = Slot number
23 BRI Layer-2 Down : Slot=xx ,Port=yy
yy = Port number
Scheduled backup file of SMS Server is

24
deleted
xx = Slot number
25 SLT Thermal Shutdown: Slot=xx, Port=yy
yy = Port number
System fan has failed. System is in

26
Thermal Sleep Mode
LAN and WAN Port IP Addresses are in

27
the same subnet
Power Supply Card Fail (MENX): VER.: Version with Version Revision
28
Power Supply VER. Card x Fail/OFF x = Card Number (1 or 2)
Unlicensed Slot, Card Inactive: Slot=xx, xx = Slot number

29
Type: CARD TYPE CARD TYPE = Card Type with version revision
X = CPU 1 or 2
30 CommMngr Health Failed: CPU-X
DSP of CPU card stops responding.
a. For SMTP Error Code details see “SMTP Errors” at the end of this topic.
VOPP Fail: If VoPP fail message is logged, the SARVAM UCS CPU Card will not be functional.
Registration Timer Fail: The system may fail to load either the Re-registration Timer or the Registration
Retry Timer. In such a case the Proxy SIP trunk will remain un-registered and will not be functional.
The system will decode the registration status message received from the VoIP module and, if it is found to
be a problem caused by Registration Timer Failure, this will be logged in the System Fault Log.
This can happen to one or more SIP trunks, while the other SIP Trunks are functioning normally. You need
to restart the system to resolve the problem.

How to configure
To be able to use this feature, you must enable storage of Fault Logs, and assign a Destination Port for the Fault
Logs. The destination port may be a COM Port, Ethernet Port or USB to COM Port.
If the System Administrator phone is a DKP or an Extended IP Phone, you may assign a DSS key for System Fault
Log.
For instructions on configuring DSS Keys on a Digital Key phone, see “DSS Keys Programming”.
For instructions on configuring DSS Keys on Matrix Extended IP Phone, see “DSS Key Settings” in “Configuring
“Configuring Matrix SPARSH VP310” and “DSS Key Settings” in “Configuring Matrix SPARSH VP510”.
You may configure the System Fault Log settings using Jeeves and by dialing system commands from a telephone
connected to the SARVAM UCS.
Configuring System Fault Log using Jeeves

• Open Jeeves
• Under Configuration, click System Log link.
• Click System Fault Log.
• By default, System Fault Log Storage is Enabled. Select Disable, if you do not want the system to keep
a record of all the system faults.
• To generate System Fault Log - Online, that is, as and when the fault occurs, select the Destination
Port for Online SFL from the following options:

• COM Port: Select a communication port if you want to use a serial device to capture the logs. Make
sure the device is connected to the COM port.
• Ethernet: Select Ethernet port if you want to use the remote syslog server for the logs.
• USB to COM Port: Select USB to COM port if you want to use the External USB as a COM port for
capturing the logs. Make sure you connect the USB to COM Converter to the USB Port.
By default, no destination port is assigned for Online SFL.
• To generate System Fault Log - Report, that is, offline, whenever desired, select Destination Port for
SFL Report from the options: COM Port or Ethernet. Default: None.
• If you selected Ethernet as Destination Port,
Configuring System Fault Log using a Telephone

To set enable/disable SFL storage, dial:

Where,
Storage Flag is
0 for Disable
1 for Enable
Default: Enable
To select the Destination Port for SFL Online, dial:

• 6452-Port
Where,
Port is
0 for None
1 for COM Port
2 for Ethernet Port
Default: None

If you selected Ethernet Port as Destination port, to assign the IP Address for the Ethernet Port (Online),
dial:
• 6454-IP Address
To assign Port for the IP Address (Online), dial:

• 6455-Port
Where,
Default: 514.
To select the Destination Port for SFL Offline (Report), dial:

• 6453-Port
Where,
Port is
0 for None
1 for COM Port
2 for Ethernet Port
Default: None
If you selected Ethernet Port as Destination port, to assign the IP Address for the Ethernet Port (Report),
dial:
• 6456-IP Address

• 6457-Port
Where,
Default: 514.
• Exit SE Mode.
How to use
You can start and stop System Fault Log - Online and Report from the System Administrator mode using Jeeves or
dialing SA Commands from an extension phone.
To start/stop fault report generation using Jeeves,
• Open Jeeves.
• Click the System Fault Log link.

• To start System Fault Log - Online, click the Start button.
• To stop System Fault Log - Online, click the Abort button.
• To start System Fault Log - Report, click the Start button.
• To stop System Fault Log - Report, click the Abort button.
• To clear System Fault Logs from the buffer, click the Clear SFL button.
By default, the online and offline reports are printed on the destination port assigned by you.

• To view System Fault Log on your computer screen, click the View button.
System Fault Log appears on your screen.
• To print this report on the local printer connected to your computer, click the Print button. An alert
message will appear.
• Click the OK button. The System Fault Log - Report will be printed on the local printer connected to
your computer.

Make sure you have selected ‘NONE’ as the destination port to print the report on the local printer
connected to your computer.
• You may log out of the System Administrator mode.
To generate Fault logs from an extension phone,
To start/stop Online fault log generation:

To start/stop offline Report generation:

• Exit SA mode.
You may print the logs captured on the Syslog Server after suitable modification of the format.
The format of the standard system fault report looks as below:
Field Column Position
Blank 00
Date (As per Date format selected) 01
Time (HH:MM:SS) 12
Even Index No. (two characters) 21
Event Description 24
The Online report would look like this:
06-02-2017 15:59:54 DKP Absent : 3017 , Slot=06, Port=01
The Offline report would look like this:
SYSTEM FAULT LOG AS ON 27-02-2017 Mon AT 12:13

-------------------------------------------------------------------------------
30-01-2017 17:58:03 Card Absent: Slot=29, Type: VoIP NX CPU V1R4

11-02-2017 20:37:20 VOPP Program Download Fail, Slot=29
11-02-2017 20:38:52 Card Absent: Slot=01, Type: CO2 DKP2 SLT16 V3R0
-------------------------------------------------------------------------------
SARVAM UCS V1R4 Page: 1
SMTP Errors
The VMS may fail to send emails. These email failures are logged into the System Fault Log with a specific code.
The table below describes the meaning of the codes.
Error Code Error Description

SMTP Client Mail Failure Errors
01 SMTP Client Failed to Receive Mail from IPCQ
02 File Attachment Failed
03 Invalid "To" Field
04 Invalid "From" Field
05 Invalid SMTP Server address or IP Specified
06 Invalid "HELO" domain
07 Failed to Connect to SMTP Server (Check SMTP Server Address or Port)
08 "HELO" or "EHLO" Command Failed
09 "AUTH CRAM-MD5" Command Failed
10 "CRAM-MD5" Login Failed
11 "AUTH LOGIN" Command Failed
12 Invalid Username for "LOGIN"
13 Invalid Password for "LOGIN"
14 "AUTH PLAIN" Login Failed
15 Can't Open Attachment File
16 "MAIL FROM" Command Failed
17 "RCPT TO" Command Failed
18 "DATA" Command Failed
19 SMTP Mail Failed (During final process to Prepare Mail)
20 Mail Failed (Wait For Server Response)
VMS Application Mail Failure Errors
The calling process does not have write permission on the message
51
queue, and does not have the CAP_IPC_OWNER capability
The message can't be sent due to the msg_qbytes limit for the queue and
52
IPC_NOWAIT was specified in msgflg
53 The address pointed to by msgp isn't accessible
54 The message queue was removed
55 Sleeping on a full message queue condition, the process caught a signal
Invalid msqid value, or non-positive mtype value, or invalid msgsz value
56
(less than 0 or greater than the system value MSGMAX)
The system does not have enough memory to make a copy of the
57
message pointed to by msgp

System Fault Log Display
What’s this?
The SARVAM UCS provides a facility to display the last fault monitored on the system on the System
Administrator’s extension phone. The system also provides you the facility to view all the faults through Jeeves. To
know more, see “How to use” in “System Fault Log” .
How it works
To be able to use this feature optimally, the System Administrator extension phone must be a DKP or an Extended
IP Phone, and a DSS Key must be assigned on the phone to System Fault Log.
• When a fault occurs, the LED of the DSS Key assigned for the System Fault Log, glows.
• The System Administrator may press the DSS key or dial the System Fault Log feature access code to
acknowledge it.
• On pressing the DSS Key or dialing of the acknowledgment command, the LED of the Fault Log key is
turned OFF.
How to use
To view the System Fault Log from System Administrator Mode,
• Go Off-hook.
• Press the DSS key assigned to System Fault Log Display.

OR
• Dial 1072-010
• The Fault log appears on your phone’s display in this format: Date-Time-Fault Index
The Date and Time are in <DD-MM-YYYY HH:MM:> format
The Activity Index is a two digit number from 01 to 12.
See System Fault Log Activity Index table in “System Fault Log”.

System Log Notification
What’s this?
Whenever a fault occurs in the system or an activity takes place, the details of the fault/ activity can be sent to the
concerned person to notify him/ her of the same.
The notification can be sent,

1. as a message to the specific mobile number.
or/ and
2. as an email to the specific email id.
The SARVAM UCS maintains a log of all the system activities and faults. These logs can be printed on a local
printer or downloaded on a computer in form of a report. To know more, see “System Activity Log” and “System
Fault Log”.
How to configure
To be able to use this feature, you must configure the following parameters.
• Open Jeeves.
• Under Configuration, click the System Log link.
• Click System Log Notification.
• By default, the System Activity Log Notification via SMS check box is enabled. Clear this check box, if
you do not want the system to send notifications for the system activities via SMS.
• By default, the System Fault Log Notification via SMS check box is enabled. Clear this check box, if you
do not want the system to send notifications for the system faults via SMS.

• In Send SMS to Mobile Number-1 and Send SMS to Mobile Number-2, configure the mobile numbers to
which you want the system to send notifications as SMS.
• You can Send SMS through a fixed Mobile Port or through different Mobile Ports according to specific
numbers and time.
• To send messages through fixed Mobile Ports, select Using Fixed Port. Click the link to configure
the port link, the SMS Routing-Fixed Port table opens. Configure the SMS Routing - Fixed Port
Routing table. For detailed information, see “Fixed Port Routing (SMS Server)”.
• To send messages through certain preferred Mobile Port/s during a defined time interval, select Based
on LCR Table. Click the link to configure the port link, the SMS Routing-LCR table opens.
Configure the SMS Routing-LCR table. For detailed information, see “Least Cost Routing”.
• By default, the System Activity Log Notification via Email check box is enabled. Clear this check box, if
you do not want the system to send notification for the system activities via Email.
• By default, the System Fault Log Notification via Email check box is enabled. Clear this check box, if
you do not want the system to send notification for the system faults via Email.
• In Send Email to Email ID-1 and Send Email to Email ID-2, configure the email ids to which you want the
system to send notifications as an email.
• In Subject to be Send in Email, configure the text that is to be displayed as the subject to the receiver of
the Email.
• In Footer to be attached in SMS/Email, configure the text you want to be displayed as the footer to the
receiver of the SMS/Email.

System Parameters
System Parameters are general parameters, related to features and facilities that are applied system-wide, such as
customer name, Day-Night mode, storage of call logs, alarms, Built-In Auto Attendant call disconnect options,
Presence, and DND messages. Each of these is described briefly along with the instructions for configuring them
using the Jeeves and from a telephone.
How to configure
Configuring System Parameters using the Jeeves

• Log in to the Jeeves as System Engineer.
• Under Configuration, click System Parameters. The System Parameters page opens.

System Parameters
• Click System Parameters to expand. To view more parameters, use the vertical scroll bar on your right.
• Under System Parameters, do the following:
• Customer Name: You can assign the name of the enterprise/organization that is using SARVAM UCS
as the Customer Name. The Customer Name may contain up to 80 characters. You may enter the
address of organization/enterprise along with the name.
The Customer Name you assign will appear on the various System Reports generated and printed by
the SARVAM UCS. Default: Blank.
You can assign Customer Name also on the “Configuring System Pre-requisites” page. If you have
entered the Customer Name on this page, the same Name will appear on the System Parameters page.
• SARVAM UCS two major applications: Enterprise application to meet requirements of businesses, and
Hospitality application to meet the specific requirements of Hotels and Hospitals.
You must select Customer Profile as Enterprise or Hotel according to the application you are using.
When you select the Customer Profile, all the features and facilities specific to the application
Enterprise/Hotel along with their default settings are loaded. By default, the Customer Profile of
SARVAM UCS is defined as 'Enterprise'.
• If you want the system to present for configuration only those trunks and extension ports that are
detected by it at Power-On for configuration, select the On-site Configuration check box. Default:
Disabled.
When you enable On-site Configuration, the Jeeves, will show the pages for only those trunks and
extension port types that are on board the system, that is, detected by the system at Power-On.

• Station Name Pattern: The Station Name Pattern is the format in which the names of extensions will
be stored on the extension phones and displayed to other extensions. You can store names by First
Names only, First names and Last Names. You can also add Titles indicating gender, designation,
rank, social standing, like Mr. Mrs. Ms., Dr., Prof. Cmdr., Rev., to Names of extensions.
SARVAM UCS supports the following Station Name patterns.
Option Meaning
1 Title<space>First Name<space>Name
2 First Name only
3 Name only
4 First Name<space>Name
5 Title<space> First Name
6 Title<space>Name
Station Name Pattern must be configured for the Guest Name and Title feature of the SARVAM UCS
Hospitality module. To know more, refer the feature description in the SARVAM UCS Hospitality
System Manual.
By default, Name Only is selected as the Station Name Pattern when SARVAM UCS is operated in the
Enterprise mode, and Title<space>Name is selected as the Station Name Pattern when SARVAM
UCS is operated in the Hotel Mode.
• Default Call Hold Type: This parameter is related to the “Call Hold” feature of SARVAM UCS. You can
select Global Hold or Exclusive Hold. Default: Exclusive Hold.
• To have SARVAM UCS store also internal calls in the Missed Call Log, select the Store Internal Calls
in Missed Call Log check box. To know more about this feature, see “Call Logs”. Default: Enabled.
• You may have SARVAM UCS store internal calls in the Dialed Call Log by selecting the Store Internal
Calls in Dialed Call Log check box. To know more about this feature, see “Call Logs”. Default:
Enabled.
• To have the system store internal calls in the Answered Call Log, select the Store Internal Calls in
Answered Call Log check box. To know more about this feature, see “Call Logs”. Default: Enabled.
• To have the system store internal calls in the Redial Call Log, select the Store Internal Calls in Redial
Call Log check box. To know more about this feature, see “Last Number Redial”. Default: Disabled.
• In the MoH Source when Station kept on Hold box, keep the option Internal (VM-01). To know more,
read the feature description for “Music on Hold (MOH)”. SARVAM UCS will play the Music-On-Hold
recorded in the Voice Module Number 01 to the extension that is put on hold. Default: Internal (VM-01).
• In the MoH Source when Trunk kept on Hold box, keep the option Internal (VM-01). To know more,
read the feature description for “Music on Hold (MOH)”. SARVAM UCS will play the Music-On-Hold
recorded in the Voice Module Number 01 to the external callers who are put on hold. Default: Internal
(VM-01).
• The DKP/SIP Extension users can set multiple call appearances on their phones. If you want the
system to play MOH to all the queued internal calls when the user extension is busy, select the Play

MOH to Queued Internal Calls on DKP/Extended IP Phone check box. To know more, read the
feature description for “Music on Hold (MOH)”.
• If you select the Give Off-hook Alert to Operator check box, the system will detect extensions that are
off-hook and ring on the Operator extension to alert the Operator about the state of the phone. This
alert is useful for detecting whether the handset of extension phones are placed correctly. Read the
feature description for “OFF-Hook Alert” to know more.
• You can set the Time Zone of the system as Working-Hours or Break Hours or Non-Working hours any
time you want by setting the Day/Night Mode. You can set the system in the Day Mode or the Break
Hours Mode or the Night Mode, or let the system Operate as per the Time Table assignment282.
For more details see “Day Night Mode” and “Time Tables”. Default: Operate System as per Time Table
assignment.
• To have the system detect the extension that has made an emergency call, select the Emergency
Dialing Reporting check box.
When this flag is enabled, the system detects the extension that has made the emergency call and
reports it to the Operator extension. Thus the Operator can know which extension has made an
emergency call. To know more about this feature, see “Emergency Detection and Reporting”.
• If you want to remove the ‘+’ prefix in the CLI of the calling party (presented by the Mobile Network) and
replace it with another string, select the Replace '+' from CLI check box. Default: Disabled.
If you want to program the number string with which the ‘+’ prefix is to be replaced, in the Replace '+'
from CLI with the number string field, enter the desired number string.
If you keep the number string field blank, SARVAM UCS will remove '+' sign from the CLI of calling
party and present the remaining digits on the CLI of the Called Party.
For example:
The number string +919327237228 is received as CLI.
If ‘00’ is configured as the replace string, the CLI number would become 00919327237228
If no replacement string is configured (that is, left blank), the CLI number would be presented as
919327237228.
• To disconnect Built-In Auto Attendant Calls when the landing extensions are busy, select the
Disconnect Built-In Auto Attendant Call, when Dialed Number is Busy check box. When you
enable this flag, the call will not be routed to the Operator. Instead, it will be disconnected. Default:
Disabled.
• To disconnect Built-In Auto Attendant Calls when there is no reply from the landing destination
extensions, select the Disconnect Built-In Auto Attendant Call, when Dialed Number is not
Responding check box.
When this flag is enabled, the system disconnects the call if there is no reply from the landing
destination extensions. The call will not be routed to the Operator. Default: Disabled.
282. Certain features of the SARVAM UCS require extensions and trunks to behave differently according to the working hours, break
hours and non-working hours, which are referred to as Time Zones. The Time Zones, are defined for the entire week in a Time
Table. Time Table is assigned to trunks, extensions and other time-zone dependent features.

• To disconnect Built-In Auto Attendant Calls if the caller fails to dial a digit, select the Disconnect Built-
In Auto Attendant Call, when Caller Does not Dial any Digit check box.
When this flag is enabled, the system will disconnect the call if the caller fails to dial a digit within the
First Digit Wait Timer. The call will not be routed to the Operator. Default: Disabled.
• If the Extension creating 3 party conference, disconnects during Conference, you can select
either to Transfer the call or Disconnect other parties.
• If you select to Transfer the call, the 3-party conference is converted into a two-way speech
between the other two parties.
• If you select to Disconnect other parties, all the parties involved in the 3-party conference are
disconnected.
See “Conference-3 Party” for more details.
• To play a beep to participants of a conference to indicate inclusion of a new participant, select the Play
Beep when Conference/Dial-in Conference begins check box selected. Default: Enabled.
This flag is common for the features “Conference-Multiparty”, “Conference Dial-In”, “Emergency
Conference” and “Raid”. When this flag is enabled,
• the system plays a warning beep to the extension which is being raided by another extension,
before establishing three-way speech.
• the system plays beeps to the other participants in a Dial-In Conference when a new participant
joins in (dials into an on-going Dial-In Conference)
• the system plays beeps to the other participants connected in a Multi-Party Conference and an
Emergency Conference, when a new participant is included.
If you disable this flag, no warning Beep will be played in Raid, the existing participants in a Dial-In or
Multi-party conference will not hear any beep tone indicating the addition of a new participant.
• Play Beep when Raid/Call Taping/Conversation Recording Starts is a common flag for the features
“Call Taping” and “Conversation Recording”. When this flag is enabled, the system plays Beeps to the
extensions/calling party and extension before it starts taping the call in the common mailbox or
recording the conversation in the extension’s mailbox.
When this flag is disabled, no indication will be given to the opposite party when the call is being taped/
conversation is being recorded. Default: Enabled.
• In Play Feature Tone in place of Dial Tone when Call Forward is Set, you can select whether you
want the system to play Feature Tone instead of Dial Tone to the extensions when Call Forward is set
on these extensions. When this flag is disabled, the system will play dial tone to the extension on which
Call Forward is set, whenever the extension goes Off-hook. Default: Enabled.
• Select the Ignore call forward set by member extension, when call is routed on Routing/Dept.
Group flag, if you want the system to place calls on member extensions in a Routing Group even if
they have set Call Forward.
• Select Call Proceeding Tone for Multistage Dialing. Default: Network Tone.

This parameter is used in Multistage Dialing where you need to configure Pause and Wait for Answer in
the Dialed Number column for the number string dialed by the extension users.
When an extension user makes a call using a Calling Card, and the system dials out the number in
stages, the extension user will get Ring Back Tone twice; first after the system has dialed the Calling
Card Number, and again after the system has dialed out the destination number (called party number).
Thus the extension user will get Ring Back Tone, twice. To avoid this, you may configure the 'Call
Proceeding Tone' to be played by the system when using Multi-Stage Dialing.
You must configure the type of 'Call Proceeding Tone', according to your requirement; whether the
extension user should be connected to the speech path when the Calling Card number is out dialed or
when the called party number is out dialed. You can select any of the following Call Proceeding Tone
options, as per your requirement:
• Network Tone: If this option is selected, the extension user will get Ring Back Tone after dialing the
Calling Card number and again, after the system has dialed the called party number (when the
system is dialing out the number with Pause and Wait for Answer configured in Dialed number
string).
• Pseudo Tone: If this option is selected, the extension user will get Feature Tone when the user has
completed dialing all the digits. At the end of the tone, the extension user gets connected to the
called party (destination number).
• Silent: If this option is selected, the extension user will get Silence (no tone), after the extension
user has completed dialing all digits. After dialing out the called party number in DTMF, the system
will connect the caller to the called party number (destination number).
• You may select the Companding Algorithm according to the Regulatory Requirement of the country
where SARVAM UCS is installed. Default: A-law
The companding Algorithm —A law or law—is automatically selected when you select Region for
SARVAM UCS. However, if necessary, you may change the default companding Algorithm that
appears in this field.
If you change the Companding Algorithm, the terminals - EON48D, EON310 and EON74 (Turret) will
restart.
• You can select the Language of SE, SA and Front Desk User Web Interface as per your
requirement. The GUI of SARVAM UCS supports the languages English, Italian, Spanish, French,
German, and Portuguese. When you select ‘Region’ for SARVAM UCS, one of these languages will be
applied as appropriate for the region you selected. For instance, if you selected India, English will be
applied. If you selected Spain, Spanish will be applied. If you selected a country where none of these
languages are the local language, English will be applied.
The language set by the system on Region selection will be applied on the pages of the GUI for every
login session. You can change the default language set on Region selection, by configuring this
parameter.
• To print each system report on a separate page, keep the Form Feed in Report Printing check box
enabled. Default: Enabled.

• To be able to distinguish between incoming calls from the Public Network and those from the Private
Network on the basis of the number digits received in the CLI, enter the desired number of digits in the
Minimum No. of digits received in CLI to consider the call is from Public N/w box.
This parameter is applicable to calls originating on the E1-PRI ports (Tie-Line) configured for the Q-
Signaling. By default, CLI number with 8 or more digits will be considered as call from Public Network.
SARVAM UCS will check this parameter, whenever the incoming call is to be analyzed as call from
PISN (Private Integrated Subscriber Network) or non-PISN (Public Network Number).
• If you want to enable the Operator Console to view the presence status of the extension they are
calling, select the Display Presence Status during Call on DKP/Extended IP Phone check box.
Default: Disabled.
• To use the Watch dog function supported by SARVAM UCS, select the Enable Watch Dog check box.
SARVAM UCS supports the Watchdog function to detect and restart the system, whenever the system
hangs. Default: Disabled.
When Watch Dog function is disabled, you must manually restart the system when it hangs.
• Enable Apply RCOC only if the caller calls back on the same trunk from which the call was made
check box, if you want the SARVAM UCS to match the Trunk Port Number and Trunk Port Type of the
incoming call with the Trunk Parameters of the entry stored in RCOC table before routing the call to the
original caller. Default: Disabled.
• Enable the Stuttered Dial Tone When DND is set check box, if you want SARVAM UCS to play a
Stuttered Dial Tone on the extension when DND is set.
• If you have installed two CPU cards and want that each card must switch to the Stand-by mode after a
fixed number of hours automatically, then select the desired option in Set Active CPU Card to Stand-
by after (Hrs.). This is applicable only for ETERNITY LENX/MENX.
• When there is an incoming call on the CO Trunk and if total 5 digits are dialed out, before expiry of
Trunk Inter Digit Wait Timer, then the system will treat this call as an outgoing fraudulent call. To drop
this call, select the Detect Possible toll bypass attempt by Extn. during IC Call from CO Line &
Drop the Call check box.
• Select the Power On Self Test check box, if at every hard/soft system restart, you want the system to
generate a system Health call. This call will be placed on a DKP and its LCD will display the slot
number with the name of the respective card type. If the slot is empty, Absent will be displayed against
the slot number.
• If you have enabled Power on Self Test, configure the DKP Port on which call for Power on Self
Test is to be placed.
• By default the SMS/Email Notification Service is disabled. Select this check box if you want to send/
receive SMS/Email notifications using the features System Log Notification, or send/receive SMS using
SMS over IP or SMS on No-Reply.
• Select the Play beeps when Assistant present in 3-Party Conference check box, if you want the
system to play beeps during the conference to indicate the presence of the Operator. For details, see
“Conference-3 Party”.

• Select the Play beeps when Assistant present in Multiparty Conference check box, if you want the
system to play beeps during the conference to indicate the presence of the Operator. For details, see
“Conference-Multiparty” and “Conference Dial-In”.
• Select the Play beeps when Assistant leaves the Conference check box, if you want the system to
play beeps to the conference participants when the Operator leaves the conference.
Magneto
• Click Magneto to expand.
• If you want calls from Magneto ports to be disconnected automatically, when silence is detected, select the
Enable Silence Detection on Magneto check box. Default: Enabled.
• You may change the duration for the Magneto Silence Detection Timer, to the desired duration. The
range of this timer is from 01 to 32 seconds. Default: 30 seconds.
• Set the Magneto VAD Threshold Level to the desired level. Default: -71(dBm).
Radio
• Click Radio to expand.
• DTMF Detection - Minimum Level (dB): This parameter signifies the minimum level (dB) of the DTMF
digit to be considered as valid. By default, Minimum levels set to -10.5dB.
• DTMF Detection - Minimum ON Time (msec): This parameter signifies the minimum time period for
which the DTMF signal should be present in order to be detected. The valid range of this time is 17 to
204 milliseconds. By default, Minimum ON Time is set to 17 milliseconds.
• DTMF Detection - Minimum OFF Time (msec): This parameter signifies the minimum time period
between successive DTMF digits. The valid range of this time is 17 to 204 milliseconds. By default,
Minimum OFF Time is set to 17 milliseconds.

• DTMF Generation - DTMF ON Time (msec): It is the width of DTMF digit to be dialed out by the Radio
port. By default the ON Time is set to 102 milliseconds.
• DTMF Generation - Inter Digit Pause Time (msec): When the Radio port dials out the DTMF digits, it
waits for the Inter Digit Pause Timer, while dialing the DTMF digits. By default the timer is set to 102
milliseconds.
The DTMF Detection and Generation parameters are applicable only when there are Dial Pads connected
to the Radio devices.
Alarm
• Click Alarm to expand.
• Configure the following Alarm and Reminder parameters:
• Use Alarm with snooze: Enable this flag if you want to use the Snooze function for the Alarm Call.
• Alarm Ring Timer (Sec.): You may change the time for which the Alarm Call will ring on the extension
phone and the time for which the Operator phone will ring to notify an unanswered Alarm Call.
• Number of Alarm Attempts: You may increase or decrease the number of attempts the system
should make to serve an Alarm call.
• Alarm Attempt Interval: You may increase or decrease the time gap between each attempt the
system makes to serve an Alarm call.
• Configurable Alarm Type flag: Disable this flag, if you do not what the system to provide the Operator
and the extension users the option of setting 'Once Only' or 'Daily' Alarms. When this flag is disabled,
the system will allow only 'Once Only' alarms to be set.
• Configurable Alarm Category: Disable this flag, if you do not want the system to provide the Operator
the option of setting 'Personalized' or 'Automated' Alarm calls. When this flag is disabled, the system
will follow the 'Automated' Alarm call serving mechanism. The Operator will not be prompted to choose
between 'Automated' and 'Personalized' Alarm calls when setting Alarm calls for an extension phone.
• Voice Guided Alarm Verification: Enable this flag if you want to enable extension users to confirm
the Time they have set for an alarm or the Date and Time they have set for a reminder. Default:
Disabled.

Distinctive Rings
• Click Distinctive Rings to expand.
Distinctive Rings are ringing patterns used for distinguishing between different types of call events, like
Internal Calls, Trunk Calls, Auto Call Back, Auto Redial, Alarm, Emergency call, Priority, etc. If you want to
customize the Ringing pattern, for a call event, select the desired Ring Type. For more details see
“Distinctive Rings”.

• Click Incoming CLI Modification to expand.
Incoming CLI Modification is useful in countries where the Calling Line Identification (CLI) received by the
System extension users must be suitably modified before it can be used to dial out the number. To know
more, see “Incoming CLI Modification”.
If you receive CLI in dialable format, there is no need to use this feature. In such case, keep the flag
disabled. You do not need to program any of the CLI Modification parameters.
• To apply Incoming CLI Modification, select the Enable Incoming CLI Modification check box.
• Configure the following options for CLI modification:

• Country Code: This is the Country Code of the country where SARVAM UCS is installed. The Country
Code helps SARVAM UCS detect whether the Incoming CLI received is a national or an international
number. Do not enter any prefix such as ‘+’ or ‘00’ for the Country Code. Default: ‘91’ (India).
• Area Code: This is the Area Code of the place where the SARVAM UCS is installed. The Area Code
helps SARVAM UCS detect whether the Incoming CLI received is a local number. Do not enter any
prefix for the Area Code. For example, if you want to enter Area Code for Mumbai, enter only ‘22’. Do
not enter the prefix ‘0’ to the area code. Default: ‘265’ (Vadodara city).
• International Prefix: These are digits required as Prefix for dialing International Numbers. The prefix
may be up to 5 digits, with numbers from 00000 to 99999. Default: ‘00’.
• National Prefix: These are digits required as Prefix for dialing long distance, National (within the
country) numbers. The prefix may be up to 5 digits, with numbers from 00000 to 99999. Default: ‘0’.
• Area Code required to make local calls?: Depending on the dialing pattern of your local public
telephone network, you may choose:
• No (Area Code not required), if your public telephone network does not require the dialing of Area
Code for local numbers.
• Yes (Area Code is required), if your public telephone network requires you to dial the Area Code for
local numbers.
• Yes, with Prefix Digit, if your public telephone network requires you to dial Area Code with a
particular Prefix for local numbers. If you select this option, you must also enter the prefix digits for
the area code for local calls in the Prefix Area Code field.
• Click Clock Synchronization to expand.
The SARVAM UCS supports four clock sources for Clock Synchronization for the E1-PRI and BRI ports.
To know more, see “Clock Synchronization”.
• You must select the Clock Source in the order of Priority from 1 to 4. By default, the Clock Source priority
is selected as follows:
• Clock Source - Priority 1 - T1E1 - 001

• Also set the Clock Synchronization Frequency as: 8KHz Derived, 8KHz, 2.048MHz, 1.54 MHz Default:
8 KHz
• Set the PLL Locking Mode as Slow or Fast. Default: Fast.
• Set the PLL TIE Control as Enable or Disable. Default: Enable.
• Select the PLL Operating Mode as Normal, Hold Over or Free Run. Default: Normal.
You can program PLL TIE Control and PLL Operating Mode using Jeeves only.
DND Text Message

• Click DND Text Message to expand.
SARVAM UCS allows you to program DND different Text Messages, which extension users can select
when setting DND on their extensions. The DND text message they select is displayed to the calling
extensions. See “Do Not Disturb (DND)” to know more.
The default DND Text messages appear on your screen. You may customize messages 2 to 9 according
to your requirement. Text Message can be of maximum 16 alphanumeric characters. All ASCII characters
except < > and “ (double quote) are allowed.
Message No. Message Text
1 Do Not Disturb
2 Unavailable
3 In Meeting
4 In Conference
5 Try on Mobile
6 On Vacation

7 On Business Trip
8 Out of Office
9 With a Guest
Publish Message
• Click Publish Message to expand.
SARVAM UCS offers 10 different text Messages to Publish Message, as listed in the table below. You can
customize message 6 to 9 to match your requirement. Text Message can be of maximum 16 alphanumeric
characters. All ASCII characters except < > and “ (double quote) are allowed.
0 Absent
1 Present
2 Auto Detect
3 Away
4 On the Phone
5 Do Not Disturb
6 I am Mobile
7 In Meeting
8 Out for Meal
9 Out of Office
To know more about this feature, see “Presence”.

Greeting Message Time
• Click Greeting Message Time to expand.
• When Auto Attendant is enabled on trunks, the greeting messages are played to the callers according to
the time of the day, morning, afternoon, evening.
• If Built-In Auto Attendant is enabled on trunks, the system answers the call and plays the greeting
message as per the voice modules.
• If Voice Mail Auto Attendant is enabled on trunks, the Voice Mail System answers the call and plays
the greeting message.
To know more about this feature see “Auto Attendant”.
• You can set the desired Start Time for Morning, Afternoon and Evening greetings.
In Start Morning Greeting at set the start time for the Morning Greeting Message. Similarly, in Start
Afternoon Greeting at and Start Evening Greeting at set the start time for the Afternoon and Evening
Greeting Message respectively.
The time must be in HH:MM format. The valid range for Hours (HH) is 00 to 23 and for Minutes (MM) is 00
to 59. By default the time in Start Morning Greeting at is set to 00:00, Start Afternoon Greeting at is set
to 12:00 and Start Evening Greeting at is set to 16:00.
The system plays the Morning Greeting Message between the Morning and Afternoon Greeting Start time,
the Afternoon Greeting Message between the Afternoon and Evening Greeting Start time and the Evening
Greeting Message between the Evening and Morning Greeting Start time.
As the system plays the Evening Greeting Message between the Evening and Morning Greeting Start
time, to prevent the Evening Greeting Message from being played after midnight, you are recommended to
set the Morning Greeting Start time to 00:00 hrs.

Custom Logo
• Click Custom Logo to expand.
For VARTA Mobile UC Clients — Android, iPhone — SARVAM UCS supports uploading of any customized
logo. This logo shall appear on the Home screen of the VARTA ADR100 and VARTA iOD100.
Make sure the image you want to upload fulfils the following:
• The image must be a JPEG, JPG or PNG file
• Maximum Dimension of the image : 200 x 200px
• Maximum File Size of the image : 20KB
To upload the file,
• Click the Choose File button to Select the image file to be uploaded from the location on the local
disk.
A preview of the image you uploaded appears.
• If you wish to remove the image, click the Remove button.
If you upload another image directly without removing the previous image, the system will automatically
overwrite the image.
Configuring System Parameters using a Telephone

Customer Name
• To enter Customer Name, dial:
• 5401-Customer Name-#*
• To clear the Customer Name, dial:
• 5401-#*

System Parameters
• To select Customer Profile, dial:
• 5315-Code
Where, Code is
1 for Enterprise Mode
2 for Hotel Mode
Default: Enterprise Mode.
• To enable/disable On-site Configuration flag, dial:

• 2108-Flag
Where, flag is
1 for Enable
0 for Disable
Default: Disable.
• To change Station Name Pattern, dial:

• 3615-Station Name Pattern Code
Where,
Station Name Pattern Code is
1 for Title <space> First Name <space> Name
2 for First Name only
3 for Name only
4 for First Name <space> Name
5 for Title<space>First Name
6 for Title<space>Name
Default: Name only
• To select Default Call Hold Type, dial:

• 5318-Hold Type
Where,
Hold Type is
1 for Exclusive Hold
2 for Global Hold
Default: Exclusive Hold.
• To enable/disable Log Internal Calls in Missed Calls, dial:

• 5361-Code
Where,
Code is
0 for Disable (Do not store internal calls in "Missed Calls" log)
1 for Enable (Store internal calls in "Missed Calls" log)
Default: Enable.
To enable/disable Log Internal Calls in Answered Calls, dial:

• 5362-Code
Where,
Code is
0 for Disable (Do not store internal calls in "Answered" log)
1 for Enable (Store internal calls in "Answered Calls" log)
Default: Enabled.
To enable/disable Log Internal Calls in Dialed Calls, dial:

• 5363-Code

Where,
Code is
0 for Disable (Do not store internal calls in "Dialed Calls" log)
1 for Enable (Store internal calls in "Dialed Calls" log)
Default: Enabled.
• To select the type of music to be played when stations are kept on hold, dial:
• 3552-Code
Where,
Code is
1 for Voice Module 01
Default: 1.
• To select the type of music to be played when trunks are kept on hold, dial:
• 3553-Code
Where,
Code is
1 for Voice Module 01
Default: 1.
If all the extensions of the Routing Group you selected for Alarm Notification type are busy, the extension
user will be played MoH (MoH can be Voice Module 01).
• To enable/disable OFF Hook Alert flag, dial:

• 5333-Code
Where, Code is
1 for Enable
2 for Disable
Default: Enable
• To set the system in Day/Night mode, dial:

• 4801-Code
Where,
1 is for Day Mode.
2 is for Night Mode.
4 is for Break Hours Mode.
• To enable/disable emergency reporting, dial:

• 5110-Code
Where,
Code is 0 for Disable and 1 for Enable.
Default: Enable
• To enable/disable Replace '+' from CLI, dial:

• 5334-Code
Where,
Code is
0 for Disable
1 for Enable.
Default: Disable

• To configure number string to be used as Replacement for ‘+’ in CLI, dial:
• 5335-Replacement string
Where,
Replacement string is any digit string of up to a maximum of 6 digits, from 0 to 9. Default: blank.
• To enable/disable the Disconnect when Caller Doesn't Dial a Digit' flag, dial:
• 5338-Code
Where, Code is
0 for Disable.
1 for Enable
Default: Disable.
• To enable/disable the Disconnect Built-In Auto Attendant call, when Dialed Number Busy, dial:
• 5336-Code
Where, Code is
0 for Disable.
1 for Enable
Default: Disable.
• To enable/disable Disconnect Built-In Auto Attendant Call, when Dialed Number does Not Reply, dial:
• 5337-Code
Where, Code is
0 for Disable.
1 for Enable
Default: Disable
To enable/disable Beep when Conference/Dial-In Conference, Raid starts, dial:

• 5331-Flag
Where,
Flag is
0 for Disable
1 for Enable
Default: Enable
• To enable/disable Beep when Call Taping and Conversation Recording starts, dial:
• 5332-Flag
Where,
Flag is
0 for Disable
1 for Enable
Default: Enable
• To disable/enable Feature Tone in place of Dial Tone when Call Forward is set, dial:
• 5312-Feature Tone Flag
Where,
Feature Tone Flag is
0 for Disable
1 for Enable
Default: Enable
• To select Call Proceeding Tone Type - Multistage Dialing, dial:

• 5311-Call Proceeding Tone Type
Where,

Call Proceeding Tone Type is
1 for Network Tone
2 for Pseudo Tone
3 for Silent
Default: Network Tone
• To select the Companding Algorithm, dial:

• 5322-Companding type
Where
Companding type is
1 is for A-Law
2 is for µ-Law:
Default: A-Law.
• To select a Language for SE, SA and Front Desk User Web Interface, dial:
• 5319-Language
Where,
Language is
1 for English
2 for French
3 for German
4 for Spanish
5 for Portuguese
6 for Italian
Default: English (country specific)
• To enable/disable Form Feed:

• 5321-Code
Where,
Code is
0 for Disable Form Feed.
1 for Enable Form Feed.
Default: Enable Form feed.
• To program minimum number of digits received in CLI to consider as call from the Public Network, dial:
• 5314-Minimum Caller ID Digits
Where,
Minimum Caller ID Digits is from 01 to 16.
Default:8
• To enable/disable 'Display Presence Status during Call on DKP', dial:

• 5320-Flag
Where,
Flag is
0 for Disable
1 for Enable
Default: Disable.
• To enable/disable Watch Dog function, dial:

• 5309-Flag
Where,
Flag is
1 for Enable

0 for Disable
• To enable/disable VMS Alarm Verification, dial:

• 2211-Code-#*
Where,
Where,
Code is
1 for Enabled
0 for Disabled.
Magneto
To enable or disable the "Enable Silence Detection on Magneto?" flag, dial:
• 5357-Flag
Where,
Flag is
1 for Enable
0 for Disable
Default: Enabled.
To set Magneto-Silence Detection Timer, dial:

• 5356-Silence Detection Timer
Where,
Silence detection timer value range is from 001 to 255 seconds.
To set the Magneto Threshold Level, dial:

• 5358 - Magneto VAD Threshold Level
Where,
The value of the Level is from 0 to -96.
Default: -25 dBm

To configure Alarm Ring Timer, dial:
• 2201-Seconds
Where,
Seconds is from 001 to 255. Default: 45 seconds.
To configure Number of Alarm Attempts, dial:

• 2202-Number of Alarm Attempts
Where,
Number of Alarm Attempts is from 1 to 9. Default: 3.
To configure Alarm Attempt Interval, dial:

• 2203-Alarm Attempt Interval
Where,
Alarm Attempt Interval is from 1 to 9 minutes. Default: 5 minutes.
To configure Snooze function, dial:

• 2204-Snooze
Where,
Snooze is 0 or 1.
Select '0' to disable Snooze and '1' to enable snooze.

To disable/enable Configurable Alarm Type, dial:
• 2208-Flag
Where,
Flag is 0 for Disable, 1 for Enable1. Default: Enabled.
To disable/enable Configurable Alarm Category, dial:

• 2209-Flag
Where,
Flag is 0 for Disable, 1 for Enable. Default: Enabled
To configure Alarm Notification Type, dial:

• 5602-1-Template Number-12-Alarm Notification Type
Where,
Template Number is Station Advanced Feature Template from 01 to 50.
Default: 01.
Alarm Notification Type is:
1 for Music on Hold
2 for Voice Message (Voice Modules)
3 for Routing Group
4 for Voice Mail

• To enable/disable Incoming CLI Modification, dial:
• 5367-Flag
Where,
Flag is
0 for Disable
1 for Enable
Default: Disabled
• To program the Country Code, dial:

• 5368-Country Code
Where,
Country Code is a number string up to 5 digits. The number string may consist of numbers from 00000
and 99999.
Default: 91 (India)
Program the Country Code without any prefix. For example, if you want to program USA as Country
Code, enter ‘1’ only (without the prefix ‘00’ or ‘+’)
• To program the Area Code, dial:

• 5369-Area Code
Where,
Area Code is a number string up to 5 digits. The number string may consist of numbers from 00000 and
99999.
Default: 265 (Evacuator)
Program the Area Code without any prefix. For example, if you want to program Mumbo as Area Code,
enter ‘22’ only (without the prefix ‘0’)
• To program International Call Prefix, dial:

• 5370-International Call Prefix
Where,

International Call Prefix is a number string up to 5 digits. The number string may consist of numbers
from 00000 and 99999.
Default: 00
• To program National Call Prefix, dial:

• 5371-National Call Prefix
Where,
National Call Prefix is a number string up to 5 digits. The number string may consist of numbers from
00000 and 99999.
Default: 0
• To enable/disable ‘Area Code required to make local calls?’, dial:

• 5372-Code
Where,
Code is
1 for No (Area Code is not required)
2 for Yes (Area Code is required)
3 for Yes (Area Code with Prefix required)
Default: 1
• To program Prefix Digits to Area Code for Local Calls, dial:

• 5373-Prefix Digits
Where,
Prefix Digits to Area Code is a number string up to 5 digits. The number string may consist of numbers
from 00000 and 99999.
Default: Blank
• To select clock source, dial:
• 5341-Clock Source Index-Port Type-Port Offset
Where,
Clock Source Index is 1 to 4.’
Port Offset is 01 to 32.
Clock Source Index is from 1 to 4.
Port Type Meaning Port Offset
05 T1E1 01-08
04 BRI 01-32
00 Null 000
0 is a valid port to follow internal clock.
Default:
1 T1E1-1
2 T1E1-2

3 T1E1-3
4 T1E1-4
• To select System Clock Synchronization, dial:

• 5342-System Clock Synchronization
Where,
System Clock Synchronization from 1 to 4.
System Clock Synchronization Meaning
1 8 KHz Derived
2 8 KHz
3 2.048 MHz
4 1.54 MHz
Default: 8 MHz
• To select PLL Locking Mode, dial:

• 5343-PLL Locking Mode
Where,
PLL Locking Mode is
1 for Fast
2 for Slow
Default: Fast.
Publish Message
You can customize Publish Messages using Jeeves only.

System Timers and Counts
What’s this?
Several features of the SARVAM UCS are based on Timers and Counts. For example, how long and how many
times an extension should ring when Message Wait is set, or how long the Busy Tone, the Ring Back Tone, or the
Error Tone should be played to an extension. SARVAM UCS allows you to configure most of these Timers and
Counts to suit your requirement. Listed below are the Timers and Counts related to the various features and
facilities, along with a brief description and default value of each.
Auto Redial
Name Description Range Default
Auto Redial - Ring Back The time for which system waits to sense the 0 to 255 60 seconds
Tone Wait Timer (sec.) Ring Back Tone from the PSTN/CO Network
after dialing the requested number.
Auto Redial - Ring The time for which the extension that has 0 to 255 45 seconds
Timer (sec.) requested Auto Redial should ring.
Auto Redial - Normal The time interval between auto redial attempts 0 to 255 45 seconds
Timer (sec.) when Auto Redial ‘Normal’ is set.
Auto Redial - Normal The number of auto redial attempts the system will 0 to 255 5 tries
Count make when Auto Redial ‘Normal’ is set.
Auto Redial - Priority The time interval between auto redial attempts 0 to 255 10 seconds
Timer (sec.) when Auto Redial ‘Priority’ is set.
Auto Redial - Priority The number of auto redial attempts the system will 0 to 255 20 attempts
Count make when an extension having the feature Auto
Redial Priority in its Class of Service uses Auto
Redial ‘Priority’.
Built-In Auto Attendant The time after which the system releases the 0 to 255 60 seconds
Inactivity Timer (sec.) trunk, if the caller has not dialed any digit, or
when a Built-In Auto Attendant or Trunk Auto
Answer call is not answered by the landing
destination.
Built-In Auto Attendant 0 to 255 5 seconds

The time for which the system waits before
Answer Wait Timer
answering a Built-In Auto Attendant call.
(sec.)
Built-In Auto Attendant The time for which the system plays music after 0 to 255 5seconds
Music Timer (sec.) answering the Built-In Auto Attendant call.
Built-In Auto Attendant The time for which the system plays beeps to the 0 to 255 10 seconds
Beeps Timer (sec.) caller to prompt the caller to dial the desired
extension number when the call is answered by
the Built-In Auto Attendant.

Built-In Auto Attendant The time for which the system rings on the 0 to 255 30 seconds
Ring Timer (sec.) landing destination extension in a Built-In Auto
Attendant call.
Built-In Auto Attendant In a Built-In Auto Attendant call, the time for 0 to 255 15 seconds
Busy Tone Timer (sec.) which the system plays Busy Tone, if the dialed
extension is busy.
Built-In Auto Attendant In a Built-In Auto Attendant call, the time for 0 to 255 5 seconds
Error Tone Timer (sec.) which the system plays Error Tone to the caller, if
the caller has dialed an invalid code.
Call Progress Tones
Dial Tone Timer (sec.) The time for which the system plays the Dial tone. 2 to 255 7 seconds
Ring Back Tone Timer The time for which the system plays the Ring Back 1 to 255 45 seconds
(sec.) Tone.
Busy Tone Timer (sec.) The time for which the system plays the Busy Tone. 1 to 255 7 seconds
Error Tone Timer (sec.) The time for which the system plays the Error Tone. 1 to 255 30 seconds
Feature Confirmation The time for which the system plays the 1 to 255 7 seconds
Tone Timer (sec.) Confirmation Tone when a feature is set or
canceled.
Programming Error The time for which the system plays the Error Tone 1 to 255 3 seconds
Tone Timer (sec.) when you have entered an invalid command string
while configuring a feature from a phone.
Programming The time for which the system plays the 1 to 255 3 seconds
Confirmation Tone Confirmation Tone when a system command is
Timer (sec.) successfully executed when configuring the system
from a phone.
Tone Demo Timer (sec.) The time for which the system plays Call Progress 1 to 255 30 seconds
Tone when you are demonstrating the tone.
The time for which the system will wait for an 1 to 255 30 seconds
Call Forward - No Reply
extension (Department Group member) to answer
Timer for Department
an incoming call, before forwarding the call to the
Group (sec)
programmed destination.
DISA Idle State Timer In a DISA PIN Authentication call, the time for which 0 to 255 20 seconds
(sec.) the system waits for the caller to go Off-hook after
entering DISA. If the caller does not go Off-hook
within this timer, the system releases the call.

DISA Inactivity Timer In a DISA call, the time for which the system waits 0 to 255 2 minutes
(min.) for the caller to dial digits. If the caller does not dial
any digit within this timer, the system disconnects
the call. This timer is applicable only for Analog
Trunks.
Message Notification
Message Notification This count defines the number of times the 0 to 15 3

Retry Count system must make Message Wait Notification
Calls to the destination number.
Message Notification The time for which the extension on which the 0 to 255 45 seconds
Ring Timer (sec) Message Wait Notification Call is made will ring.
Message Notification This defines time interval between two retries. It is 1 to 255 5 minutes
Interval (min) the time after which the system must make another
attempt to place the notification call on the
destination number.
Other Features
Auto Call Back Ring The time for which the extension requesting the 1 to 255 30 seconds
Timer (sec.) Auto Call Back and the destination extension will
ring.
Interrupt Request Timer The time for which the extension on which the 1 to 255 45 seconds
(sec.) Interrupt Request is made will get the beeps.
Barge-In Timer (sec.) The time after which the extension that has 1 to 255 10 seconds
activated Barge-In gets connected to the
extension which is barged in.
Trunk Reservation The time for which a trunk remains reserved for 1 to 255 10 minutes
Timer (min.) the extension that has reserved the trunk.
Transfer while Ringing When an extension transfers a call to another 1 to 255 30 seconds
Timer (sec.) extension after it starts ringing, this is the time for
which the system will wait for the transfer target
extension to answer the call. If the transfer target
does not answer the call within this timer, the call
is returned to the transferror.
Transfer on Busy Timer When a call is transferred to a Busy extension, 1 to 255 30 seconds
(sec.) this is the time for which beeps are played on the
transfer target extension.
Trunk to Trunk Inactivity In a Trunk-to-Trunk call, this is the time for which 1 to 255 2 minutes
Timer (min.) the system waits after call maturity for any digit to
be dialed. If no digit is dialed within this timer, the
system drops the call.

Call Park Timer (min.) The time after which the call comes back to the 2 to 255 2 minutes
extension that has parked the call.
Call Park Release Timer The time after which the parked call gets 1 to 255 3 minutes
(min.) disconnected.
Live Call Screening The time for which the speaker of the DKP/ 1 to 255 10 seconds
(sec.) Extended IP Phone extension remains ON while
the message from the caller is being recorded.
Message Wait Ring It is the Number of times the extension should 0 to 255 10 attempts
Count ring after the Message Wait is set on an
extension. This count is applicable only when
‘Ring’ is selected as the Message Wait
Notification type for the extension.
Message Wait Ring The time for which the extension rings to indicate 1 to 255 30 seconds
Timer (sec.) that Message Wait is set for the extension. This
timer is applicable only when ‘Ring’ is selected as
the Message Wait Notification type for the
extension.
Message Wait Ring The time after which the extension should ring 1 to 255 30 minutes
Interval Timer (min.) again to indicate Message Wait is set. This timer
is applicable only when ‘Ring’ is selected as the
Message Wait Notification type for the extension.
Conflict Dialing Timer The time for which the system waits for the 1 to 255 2 seconds
(sec.) extension user to dial the next digit to resolve
conflicting access codes dialed by the extension
user.
Extension - Inter Digit The time for which the system waits for the 2 to 255 7 seconds
Wait Timer (sec.) extension user to dial the next digit. On the expiry of
this timer, the system considers it as the end of
number dialing.
SA Command - Inter When the user dials SA Commands, the system 2 to 255 15 seconds
Digit Wait Timer (sec.) waits for the SA Command - Inter Digit wait timer for
the user to dial the next digit. On the expiry of this
timer, the system considers it as end of number
dialing and proceeds further with executing the
command.
Trunk - First Digit Wait the time for which the system waits for the 1 to 255 25 seconds
Timer (sec.) extension user to dial the first digit, after grabbing
the trunk.
Trunk - Inter Digit Wait When an extension user has grabbed the trunk and 1 to 255 3 seconds
Timer (sec.) is dialing a number, the system waits for the Trunk-
Inter-Digit wait timer for the extension user to dial
the next digit. On the expiry of this timer, the system
considers it as end of number dialing and proceeds
with the call.

Global Hold Retrieval This is the time for which a call put on Global 1 to 999 60 seconds
Timer (sec.) Hold remains connected in the system. If the call
put on Global Hold is not retrieved within this
timer, the call is returned to the DKP/Extended IP
Phone which put it on hold.
Exclusive Hold Retrieval This is the time for which a call put on Exclusive 1 to 255 2 minutes
Timer (min.) Hold remains connected in the DKP/Extended IP
Phone. If the call put on Exclusive Hold is not
retrieved within this timer, the call is returned to
the DKP/Extended IP Phone which put it on hold.
RCOC Record Delete This is the time for which the record of the 1 to 999 999 seconds
Timer (min.) outgoing call is stored in the RCOC Table. The
timer is activated whenever a record is stored in
the RCOC table. At the end of this timer, the
system deletes this record from the table.
Release Conference if This is the time for which the system will wait for 1 to 255 2 minutes
Idle for more than (min.) participants of a Dial-In Conference to withdraw or
release themselves from the conference, one-by-
one. On the expiry of this timer, the system releases
the Dial-In Conference and frees the resource
occupied by this conference in the conferencing
circuit.
Watchdog Refresh Wait This is the time within which the system should 1 to 255 60 seconds
Timer (sec.) signal to the Watchdog. If the system does not send
signal to the Watchdog within this Timer, the
watchdog device will restart SARVAM UCS.
Line Echo Cancellation The time after which the SARVAM UCS sends 1 to 255 45 seconds
Start Timer (sec.) the Line Echo Cancellation start request to the
DKP port.
Retry Counts for When the user enters a wrong Authority 0-9 3 attempts
Authority Code Password, this count defines the number of times
the system allows the user to re-enter the
password.
Emergency Reporting The time for which the Operators extension rings, 1 to 255 10 minutes
Call - Ring Timer (min) to inform the operator the number of the
extension that dialed out an emergency number.
Held Call Disconnection The time after which a held call will be 1 to 255 5 minutes
Timer (min)* disconnected, if not retrieved.
Conference - Assistant The time after which you want the system to play 1 to 255 5 seconds
Present Beep Interval beeps to indicate the presence of the Operator in
(sec) the conference.
*Applicable for Matrix SPARSH VP330 and Matrix VARTA Mobile UC Clients.

How to configure
Configuring Timers and Counts using Jeeves
The Timers and Counts on this page are arranged by the name of the feature or function these are related
to.
• Change the value of the Timer or Count by entering the desired duration or count in the respective fields.
• You may log out.
Configuring Timers and Counts using a Telephone

Auto Redial
To set Auto Redial Normal - Timer, dial:

• 1704-Seconds
Where,
To set Auto Redial Normal - Count, dial:

• 1705-Count
Where,
To set Auto Redial - Priority, dial:

• 1706-Seconds
Where,
To set Auto Redial - Count, dial:

• 1707-Count
Where,
To change Auto Redial RBT Wait Timer, dial:

• 1702-Seconds
Where,
To change Redial Ring Timer, dial:

• 1703-Seconds
Where,
Call Progress Tones

To set Dial Tone Timer, dial:
• 3502-Seconds
Where,
Default: 007 seconds
To set Ring Back Tone Timer, dial:

• 3503-Seconds
Where,
To set Busy Tone Timer, dial:

• 3504-Seconds
Where,

To set Error Tone Timer, dial:
• 3505-Seconds
Where,
To set Confirmation Tone Timer, dial:

• 3506-Seconds to program the Confirmation Tone Timer.
Where,
To set Programming Confirmation Tone Timer, dial:

• 3509-Seconds
Where,
To set Programming Error Tone Timer, dial:

• 3508-Seconds
Where,
To set Tone Demo Timer283, dial:

• 3542-Seconds.
Where,

To set Built-In Auto Attendant Inactivity Timer, dial:
• 2411-Seconds
Where,
To set Built-In Auto Attendant Answer Wait Timer, dial:

• 2412-Seconds
Where,
To set Built-In Auto Attendant Music Timer, dial:

• 2413-Seconds
Where,
283. Time for which the system demonstrates the tone/ring to the user.

To set Built-In Auto Attendant Beeps Timer, dial:
• 2414-Seconds
Where,
To set Built-In Auto Attendant Ring Timer, dial:

• 2415-Seconds
Where,
To set Built-In Auto Attendant Busy Tone Timer, dial:

• 2416-Seconds
Where,
To set Built-In Auto Attendant Error Tone Timer, dial:

• 2417-Seconds
Where,
DISA
To set DISA Idle State Timer, dial:
• 2420-Seconds
Where,
To set DISA Inactivity Timer, dial:

• 2421-Minutes
Where,
Minutes is from 001 to 255 minutes.
Default: 002 minutes.
Other Features
To set Auto Call Back Ring Timer, dial:
• 3801-Seconds
Where,
To set Interrupt Request Timer, dial:

• 3802-Seconds
Where,
Seconds is 001 to 255 seconds.
To set Barge-In Timer, dial:

• 3803-Seconds
Where,
Seconds is 001 to 255 seconds.
To set Trunk Reservation Timer, dial:

• 3804-Minutes
Where,
Minutes is 001 to 255 seconds.
To set Transfer While Ringing Timer, dial:

• 3806-Seconds
Where,
Default: 030
To set Transfer on Busy Timer, dial:

• 3807-Seconds
Where,
Default: 030
To set the Trunk-to-Trunk Inactivity Timer, dial:

• 3808-Minutes
Where,
Default: 2 minutes.
To set Call Park Timer, dial:

• 3809-Minutes
Where,
Default: 2 minutes.
To set Call Park Release Timer, dial:

• 3810-Minutes
Where,
To set Live Call Screening Timer, dial:

• 3811-Seconds
Where,
To set Message Wait Ring Count, dial:

• 4403-Seconds
Where,
Default: 010.
To set Message Wait Ring Timer, dial:

• 4404-Seconds
Where,
To set Message Wait Ring Interval, dial:

• 4405-Minutes
Where,
To set Conflict Dialing Timer, dial:

• 5351-Seconds
Where,
To set Station - Inter Digit Wait Timer, dial:

• 5352-Seconds
Where,
To set Station - Inter Digit Wait Timer, dial:
• 5352-Seconds
Where,
To set Trunk - First Digit Wait Timer, dial:

• 5353-Seconds
Where,
To set Trunk - Inter Digit Wait Timer, dial:

• 5354-Seconds
Where,
To set Global Hold Retrieval Timer, dial:
• 3805-Seconds
Where,
Seconds is from 001-999 Seconds.
To set Exclusive Hold Retrieval Timer, dial:

• 3812-Minutes
Where,
Minutes is from 001- 255 Minutes.
Default: 2 minutes.
To set the RCOC Record Delete Timer, dial:

• Dial 3521-Minutes

Where,
Minutes are from 001 to 999
To set Release Conference if idle for more than Timer, dial:

• 5355-Minutes
Where,
To set Watch Dog Refresh Wait Timer, dial:

• 5316-Seconds
Where,
• Exit SE mode.

System Security
What’s this?
Access to the SARVAM UCS is secured at three levels by way of a password:
• at the System Engineer Level with the System Engineer (SE) password.
• at System Administrator Level with the System Administrator (SA) password.
• at the User Level with the User Password.
The System Engineer and the System Administrator passwords secure the system settings from access and
alteration by unauthorized persons (anyone other than the System Engineer and the System Administrator), thus
preventing possible misuse of the features and facilities.
System Engineer (SE) Password
For Accessing Jeeves

The SE password is a code used to prevent unauthorized access and alterations or misuse of the features and
facilities. As this password is meant for restricting access to the SE mode, we strongly recommend you to:
• Change the password regularly
The default SE Password is 1234. The password can be changed using Jeeves only and it must be as per the
specifications given below:
• It must be a minimum of 6 characters and a maximum of 12 characters.
• It must include atleast one upper-case, one lower-case, one number and one special character.
• All ASCII characters (except Percentage %, Hash #, Equal to =, Plus +, And &, Backslash \, Less than <,
Greater than >, Apostrophe ’, Double Quote " and Space) and digits 0 to 9 are allowed.
To provide additional security,

• the password will be valid for 90 days and you will not be able to login with the existing password. You will
be prompted to change the password.
• if you enter a wrong password five times consecutively within 10 minutes, the system will block the source
IP Address for 10 minutes. This activity will be logged in the “System Activity Log” as well as “Simple
Network Management Protocol (SNMP)”.
• The SE password is stored in the CPU Card. If you forget the SE password, the only way to restore the
default SE password is to change the Jumper settings of the CPU Card.
• You are advised to record and store the SE password at a safe place, where it can be accessed by you
(the System Engineer) to avoid the inconvenience of restoring the default SE password.
Changing the SE Password using Jeeves

• Under Configuration, click the Change SE P/w.
• Under For Web Interface and FTP,
• Enter Current Password.
• Enter New Password. The new password must be:

• all ASCII characters (except Percentage %, Hash #, Equal to =, Plus +, And &, Backslash \, Less
than <, Greater than >, Apostrophe ’, Double Quote " and Space) and digits 0 to 9 are allowed.
• Click Submit button to save your new password.
For programming from extensions

The SE password is a code used to restrict unauthorized access to the SE Mode. The password can be a minimum
of 4 digits to a maximum of 12 digits. The valid digits are from 0 to 9. The default SE Password is 1234. To avoid
unauthorized access, we recommend you to change the password. Make sure it is strong and is kept confidential.
The SE Password can be changed using Jeeves as well using commands.
Changing the SE Password using Jeeves

• Under Configuration, click the Change SE P/w.
• Under For Programming from Extensions,
• Enter New Password. The new password can be a minimum of 4 digits to a maximum of 12 digits. The
valid digits are from 0 to 9.
Changing SE Password using a Telephone

• Dial 5306-New SE Password
• Exit SE mode.
If the new password is less than 12 digits, you must dial 5306-New SE Password #*
Forgot the SE Password?

If you have already changed the default SE-Password for accessing Jeeves and are unable to recall or locate it,
you must restore the default SE password.

Restoring Default SE Password
Restoring the Default SE Password requires you to change the Jumper/Switch Settings in the CPU Card. Follow
the instructions given below for each model of SARVAM UCS.
ETERNITY LENX/MENX55 Video calls
• Remove the CPU Card from the slot.

• Locate the Switch SW1 (PASSWORD_IP_DEFAULT) on the PCB of the CPU Card.
• Change the position of the Switch from 1 to ON (that is, from 'Normal' to 'Reset SE Password')
• Reinsert the CPU Card into the slot.
• Switch ON the system and wait for 15 seconds.
• Switch OFF the system and remove the CPU Card from the slot.
• Change the Switch position from ON to the original position 1.
• Insert the CPU Card back into the slot.
• Switch ON the system.
The following parameters are also set to default when Switch SW1 is turned ON:
• All the Network parameters except TCP NAT Keep Alive and UDP NAT Keep Alive.
• Allow FTP Server Access, Allow TELNET Server Access and Allow Web Server Access parameters under
Security Settings on WAN.
• For Web Interface and FTP parameter under Change SE Password.
ETERNITY GENX
• Remove the CPU Card from the slot.

• Locate Jumper J1 on the PCB of the CPU Card.
• Change the position of the Jumper from BC to AB (that is, from 'Normal' to 'Reset SE Password')
• Reinsert the CPU Card into the slot.
• Switch ON the system and wait for 15 seconds.
• Switch OFF the system and remove the CPU Card from the slot.
• Change the Jumper position from AB to the original position BC.
• Insert the CPU Card back into the slot.
• Switch ON the system.
The default SE password will be restored to 1234. You can now enter the programming mode by dialing 1#91-
1234 (the default password). You can also change the SE password again using Jeeves or by dialing a
command as described above.
If you do not restore the position of the Jumpers from AB back to BC, the new password will not be saved.
Make sure you change the Jumper position to BC before you change the default password.
System Administrator (SA) Password
For Accessing Jeeves

The SA password is a code for preventing unauthorized access to the SA mode. As this password is meant for
restricting access to the SA mode, we strongly recommend you to:
• Change the password regularly

The default, SA Password is 1111. The password can be changed using Jeeves only and it must be as per the
specifications given below:
• It must be a minimum of 6 characters and a maximum of 12 characters.
• It must include atleast one upper-case, one lower-case, one number and one special character.
• All ASCII characters (except Percentage %, Hash #, Equal to =, Plus +, And &, Backslash \, Less than <,
Greater than >, Apostrophe ’, Double Quote " and Space) and digits 0 to 9 are allowed.
To provide additional security,

• the password will be valid for 90 days and you will not be able to login with the existing password. You will
be prompted to change the password.
• if you enter a wrong password five times consecutively within 10 minutes, the system will block the source
IP Address for 10 minutes. This activity will be logged in the “System Activity Log” Log as well as “Simple
Network Management Protocol (SNMP)”.
Changing SA Password using Jeeves

• Click SA Password.
• Under For Web Interface of SA and Front Desk User.
• Enter New Password.The new password must be:

• all ASCII characters (except Percentage %, Hash #, Equal to =, Plus +, And &, Backslash \, Less
than <, Greater than >, Apostrophe ’, Double Quote " and Space) and digits 0 to 9 are allowed.

For programming from extensions
Changing SA Password using Jeeves

• Click SA Password.
• Under For Programming from Extensions.
• Enter New Password. The new password can be a minimum of 4 digits to a maximum of 12 digits. The
valid digits are from 0 to 9.
• Click Submit button to save your new SA password.
Changing SA Password using a Telephone

• Dial 5310-New SA Password.
• Exit SE mode.

If the new password is less than 12 digits, you must dial 5310-New SA Password #*
Forgot the SA Password?

In case the System Administrator has forgotten the password, a new password can be issued by the System
Engineer only.
The System Engineer can assign a new SA password from Jeeves only.
To issue a new SA Password from Jeeves,
• Under Configuration, click Change SA P/w.
• Under For Web Interface of SA and Front Desk User.
• Enter New Password. The new password must be:

• Click Submit to save new SA password.

User Password
Extension Users can secure their respective stations/extensions from unauthorized use with a password unique to
each extension. The User password too is a combination of any four digits, from 0 to 9. The default User Password
is 1111, which each user can change from their respective extensions.
To avoid unauthorized access,

• the extension users must change the default password.
• make sure the new password is strong and is kept confidential.
• change the password regularly.
Refer the topic “User Password” to know more.
Forgot the User Password?

If an extension user forgets the User Password, a new password can be issued to the extension user by the
System Administrator using Jeeves or by dialing a command.
To issue a new User password using Jeeves,

• Click Submit.
• The page of the desired extension will appear.
• Enter the new User Password in the field Change User Password to. The User Password may be a
combination of 4 digits. Valid digits: 0 to 9.
• Click Submit button to save your new SA password.
To issue a new User password using a telephone,
• Enter SA mode from a DKP/Extended IP Phone/SLT.

• Dial 1072-012-Extension Number-New User Password
• Exit SA mode.
Additional Security to Extension Users

It is also possible to secure unattended extensions. This may be necessary when extension users forget to lock
their extensions and are away from their desks. In such situations, the System Administrator can protect the
extension from unauthorized access and use by:
a. locking the Keypad the extension phone; possible only on DKP/Extended IP Phone extensions.
b. setting the User Status for the extension as "Absent"; possible on both DKP/Extended IP Phones and SLT
extensions. Read the feature description “User Absent/Present” to know more.
The System Administration can do this, lock the keypad of DKP/Extended IP extensions and set DKP/Extended IP
Phones and SLT extension users as ‘Absent’ using Jeeves and by dialing commands from a telephone.
To secure an extension from Jeeves,


• Click Submit.
• The page of the desired extension will appear.
• Change user status by selecting the option Absent for the parameter Presence. When you want to
change user status again to present, select Present.
• Select the option Lock for the parameter Keypad. When you want to remove keypad lock, select Unlock.
When the keypad is locked, the features Call Log, Contact, Call Forward, Dynamic Lock, User Status,
DND, Call Cost Display, Hotline, Alarm, Change User Password will not be accessible by the extension
user.
• Click Submit button.

To secure an extension using a telephone,
• Enter SA mode from a DKP/Extended IP Phone/SLT.

• Dial 1072-013-Extension Number-1 to lock keypad of an extension.
• Dial 1072-013-Extension Number-0 to unlock keypad of an extension.
• Dial 1072-014-Extension Number-0 to change user status to Absent.
• Dial 1072-014-Extension Number-1 to change user status to Present.
• Exit SA mode.
• Extension users can also set their status as 'Absent' or 'Present' from their respective extension
phones. Refer “User Absent/Present”).
• DKP/Extended IP Phone extension users can also lock the keypad of their phones from the Phone
Menu. Refer “Digital Key Phone-Operation” and “Extended IP Phone/VARTA UC Client - Operation” for
instructions on navigating the phone menu.
• It is also possible to lock/unlock the keypad and set the user extension status as 'Absent'/'Present' from
a remote location using “Direct Inward System Access (DISA)”.
FTP Access for Extended IP Phones

The access to the FTP Server of the Extended IP Phones is secured by a password. The FTP password is a code
for preventing unauthorized access. The default FTP Password is 1234. As this password is meant for restricting
access, we strongly recommend you to:
• change the default password.
The password can be changed using Jeeves only. The password can be a minimum of 6 characters to a maximum
of 12 characters. All ASCII characters (except Percentage %, Hash #, Equal to =, Plus +, And &, Backslash \, Less
than <, Greater than >, Apostrophe ’, Double Quote " and Space) and digits 0 to 9 are allowed. It must include
atleast one upper-case, one lower-case, one number and one special character.
When you set the System to default, the FTP Password will not be set to default.
If you use jumpers to restore the default SE Password, the FTP Password will not be set to default.

Changing the FTP Password using Jeeves
• Under Configuration, click the Change FTP Password for Extended IP-Phones.
• Enter New Password.
• Click Submit to save your new password.

Time Tables
What's this?
Certain features of the SARVAM UCS like Operator, Class of Service, Toll Control, Outgoing Trunk Access, among
others, require stations and trunks to behave differently according to the time of the day, which is referred to as
Time Zone.
For example, incoming calls are to be routed to the security personnel extension, instead of the Operator when the
office is closed, or certain features in the Class of Service are to be allowed only during working hours, or access to
outgoing long distance calls are to be denied during non-working hours, or the station must play a different greeting
message to the callers during break hours and holidays.
Time Tables can be assigned to stations and trunks to define their behavior according to the time of the day, that is,
Time Zone.
Time Zones
A day can be divided into three time zones: Working hours, Break hours and Non-working hours. The default Time
Zones defined for each day are:
• Working hours: 09:00 to 18:00
• Break hours: 13:00 to 14:00
• Non-working hours: 18:00 to 09:00
Working, Break and Non-Working hours are set to 00:00 for Sunday.
You can define a different Time Zone for your organization. Further, you can program each day of a week with
different time zones. For example, you may define the Working hours from Monday to Friday as 09:30 to 18:30, and
for Saturday, from 09:30 to 15:00. If you have a 24x7 business, you may set Working Hours also for Sunday.
Time Tables
A Time Table is a schedule of the three Time Zones, namely: Working Hours, Break Hours, Non-Working hours, for
the entire week.
A Time Table is assigned to stations defining the Time Zones for the entire week, so that the system can execute
the Time Zone-dependent features and facilities according to the Time Table.

There are 8 different Time Table templates to select from.
Time Table 8
Time Table 7
Time Table 6
Time Table 5
Time Table 4
Time Table 3
Time Table 2
Time Table 1
By default, the Time Table 1 is assigned to all stations and trunks in their Station Basic Feature Template and
Trunk Feature Template respectively. In Time Table 1, six days of the week - Monday to Saturday -have working
hours from 9:00-18:00, break hours from 13:00-14:00 hours and non-working hours from 18:00 to 09:00. Sunday is
a holiday, with all three Time Zones set to 00:00 hours.
You may also customize the default Time Table 1 OR customize and assign a different Time Table to the stations
and trunks.
SARVAM UCS offers the facility to switch the system manually into "Day/Night mode", at any point in time,
by issuing a command. When you set the system in Day/Night Mode, the system overrides the Time
Tables assigned to Trunks, Stations and Operator. According to the mode you selected, it applies Working
Hours/Non-Working Hours to run all the Time-Zone dependent features of the system.
Refer the topic “Day Night Mode” to know more.
How to configure
A Trunk port can be assigned Time Table in the “Trunk Feature Template” assigned to it. A Station port can be
assigned a time table in the “Station Basic Feature Template” assigned to it.
The default Time Table 1 is assigned to both stations and trunks of SARVAM UCS. Check if this time table matches
the working hours of the organization, and the Time Zone requirements of the individual stations and trunks.
The following station parameters can be programmed differently for different Time Zones:
• Class of Service.
• Toll Control.
• OG Trunk Bundle Group.
The following trunk parameters can be programmed differently for different Time Zones:
• Auto Attendant
• DISA

• Trunk Landing Group
You may retain the default Time Table 1 or customize it to suit your requirements. Or you may customize different
Time Tables and assign them to different stations and trunks.
You can customize a Time Table using Jeeves or by dialing commands from a Telephone.
Programming Time Table using Jeeves

• Under Configuration, click Time Table.
• Select the desired Time Table number and define the Time Zones, that is, working hours, break hours and
non-working hours.
• Now, assign the Time Table you program to the desired Station/Trunk.
• To assign Time Table to Stations, go to Station Basic Feature Template. Refer the topic “Customizing
Station Basic Feature Template using a Telephone” for instructions.
• To assign Time Table to Trunks, go to Trunk Feature Template. Refer the topic “Customizing Trunk
Feature Template using a Telephone” for details a how to assign timetable to trunks.
• Log out of Jeeves or continue programming, as required.
Programming Time Table using a Telephone


To program a timetable, dial:
• 1052-1-Time Table-Day-Time Zone-Start Time-End Time to program a single time Table.
• 1052-2-Time Table-Time Table-Day-Time Zone-Start Time-End Time to program the same time
zones for a range of time tables.
• 1052-*-Day-Time Zone-Start Time-End Time to program the same time zones for all time tables.
Where,
Time Table is from 1 to 8.
Day is the Day of the week

1 for Sunday
2 for Monday
3 for Tuesday
4 for Wednesday
5 for Thursday
6 for Friday
7 for Saturday
Start Time is Time Zone start time in 24-hours, HH:MM format. Where Hours is 00 to 23 and Minutes is
00 to 59.
End time is Time Zone end time in 24 hours, HH:MM format. Where Minutes is 00 to 59.
By default, Time Zone 1 is 0900 to 1800 and Time Zone 2 is 1300 to 1400. The left over time
automatically is treated as Non-Working Hours.
To default a time table, dial:

• 1051-1-Time Table to default a single time table.
• 1051-2-Time Table-Time Table to default a range of time tables.
• 1051-* to default all time tables.
Where,
Time Table is 1 to 8.
To assign Time Table to Stations and Trunk ports using SE commands, refer the topics “Customizing
Station Basic Feature Template using a Telephone” and “Customizing Trunk Feature Template using a
Telephone”.
• Exit SE mode.

Time Zone Display
What’s this?
The current time zone—Working Hours, Break Hours and Non-working Hours—is displayed on the LCD of the DKP
and the Extended IP Phone.
During Non-working hours the letter ‘N’ is displayed on the LCD of the DKP and the Extended IP Phone in the idle
state, and during Break hours, the letter ‘B’ is displayed.
During working hours, in the idle state, the phone display will look like this:
M on 20 D EC 16:58
3003 R eception 2
During Non-working hours, in the idle state, the phone display will look like this:
M on 20 D EC 16:58 N
3003 R eception 2
During Non-working hours, in the idle state, if the extension user has set User Absent and activated Keypad Lock
on the phone, the phone display will look like this:
Mon 20 DEC 16:58 N AL

3003 Recepti on 2

Toll Control
What's this?
Toll Control (or Toll Restriction) is an expense control feature of SARVAM UCS. It enables you to program the
system so that each extension has a designated calling permission referred to as 'Call Privilege'.
Each type Call Privilege allows the extension to call certain areas and restricts it from calling others. The extension
can also be restricted from the dialing of specific telephone numbers.
The SARVAM UCS supports five types of Call Privileges, these are:
• No Calls: Dialing of all external numbers is restricted. Only internal (extension-to-extension) calls are
allowed.
By default, only the numbers programmed in Global Directory Part 1 will be allowed to be dialed out as it is
enabled in the Class of Service of the extension.
If you have enabled Global Directory Part 2 and 3 in the Class of Service of the extension, the numbers
programmed in these also will be allowed to be dialed out.
• Local Calls: Dialing of outgoing calls to Local area numbers, in addition to internal calls, is allowed. It is
possible to restrict calls to certain local numbers. To apply this Call Privilege, you must configure the 'Local
Numbers' list.
• Regional Calls: Dialing of outgoing calls to regional numbers is allowed, in addition to internal and local
calls. It is possible to restrict calls to certain regions. To apply this Call Privilege type, you must configure
the 'Regional Numbers' list.
• National Calls: Dialing of domestic, long-distance numbers within the country is allowed, in addition to
internal and regional calls. You can also restrict calls to certain parts of the country. To apply this Call
Privilege type, you must configure the 'National Numbers' list.
• International Calls: Dialing of international numbers is allowed, in addition to local area, long distance
and internal numbers. You can also restrict calls to certain countries. To apply this Call Privilege type, you
must configure the ‘International Numbers’ list.
• All Calls: Dialing of all types of numbers - local, regional, national, international- is allowed, without any
restriction.
• Limited Calls: Dialing of only specific Telephone numbers (local, regional, national or international) is
allowed. By applying this Call Privilege type, you can allow and restrict dialing of telephone numbers
starting with a particular digit, or a particular area code, or certain telephone numbers only. To apply this
Call Privilege type, you must program a list of the ‘Limited Numbers’ that are to be allowed and numbers
that are to be restricted. You can configure three such ‘Limited Numbers’ lists.
Toll Control forms the basis of the features Dynamic Lock, Call Budget on Extension and Call Budget on Trunk.
Using “Dynamic Lock”, extension users can change the Toll Control (Call Privilege) of their extensions on their own.
The Operator or System Administrator can also change the Toll Control of the extension using Dynamic Lock. To
support this feature, SARVAM UCS offers fours levels of Toll Control, from 0 to 3.

Call Budget is a cost control feature that allows you to keep a control on the total cost of phone calls made by the
extension users. You may also define the calling permission for extensions and trunks can be allotted a budget. For
more details, see “Call Budget on Extension” and “Call Budget on Trunk”.
Toll Control Levels

For each Toll Control Level from 0 to 3, a 'Call Privilege' is defined. The system applies the Toll Control Level (that
is, the Call Privilege) set by the extension users themselves or set by the System Administrator/Operator for the
extensions using Dynamic Lock.
• Toll Control - Level 0 is Time Zone based, wherein you must define the Call Privilege Type for each Time
Zone, that is, Working Hours, Break Hours and Non-Working Hours. For instance, you may define 'All
Calls' as Call Privilege for Working Hours, 'Local Calls' as Call Privilege for Break Hours and 'No Calls' as
Call Privilege for 'Non-Working' Hours.
By default, Call Privilege 'All Calls' is selected for all three Time Zones.
• Toll Control - Level 1 is not based on Time Zones. By default, the Call Privilege Type for this level is
'Local Calls'.
• Toll Control - Level 2 is not based on Time Zones. By default, the Call Privilege type set for this level is
'National Calls'.
• Toll Control - Level 3 is not based on Time Zones. By default, Call Privilege 'No Calls' is selected for this
level.
• Toll Control - Call Budget Consumed is applied only if the “Call Budget on Extension” feature is enabled
on the extension.
• If the feature Call Budget on Trunks is enabled and the budget is exhausted, no calls will be routed through
the trunk. For details, see “Call Budget on Trunk”.
SARVAM UCS offers you the flexibility to redefine the Call Privilege for each of the above Toll Control Levels
according to user requirements.
How it works
• When a call is made, the SARVAM UCS checks the Toll Control Level assigned to the extension making
the call.
• The system checks the 'Call Privilege' programmed in the Toll Control Level of the extension.
• For each call privilege type detected, the system will check the following to determine if call is to be
allowed or denied, as summarized in the table below:
Type Call Privilege detected Logic will check
Local calls Local Number List programmed.
Regional calls Regional Number List programmed
National calls National Number List programmed
Internationals calls International Number List programmed

Type Call Privilege detected Logic will check
Limited Calls Limited Number list (1,2 or 3) assigned to the extension
• The Local, Regional, National, International and Limited Calls Number Lists consist of Allowed Numbers
and Denied Numbers.
• The system compares the each digit of the dialed number string with the number strings programmed in
the Allowed and Denied Number Lists of the Local/Regional/National/International/Limited Number Lists,
using the following logic:
Allowed Number List Denied Number List Result
match found match found Call allowed
match found no match found Call allowed
no match found no match found Call allowed
no match found match found Call denied
• The call is allowed to be made, if the dialed number:
• matches with Allowed Number list and the Denied Number list.
• matches with Allowed Number list, but not with the Denied Number list.
• matches with neither the Allowed List nor the Denied List.
• The call is restricted, if the dialed number matches with the Denied Number list, but not with the Allowed
Number list.
How to configure
Decide the type of Call Privilege you wish to assign to each extension port type: SLT, DKP, SIP, ISDN Terminals.
For Toll Control to work, you must first program the lists of Local Numbers, Regional Numbers, National Numbers,
International and Limited Numbers, according to the type of Call Privilege you want to assign to the extensions. To
do this,
Make a two-column tables each for Local, Regional, National, International and Limited Call Numbers on paper or
using a computer.
In one column of each list, write down the numbers you want to permit as Allowed Numbers. In the other column
write down the numbers you want to restrict as Denied Numbers. Your Table may look like these:
List of Local Numbers for Call Privilege - Local Calls
Sr. No. Allowed List Denied List

999
List of Regional Numbers for Call Privilege - Regional Calls
999
List of Regional Numbers for Call Privilege - International Calls
999
List of Limited Numbers for Call Privilege - Limited Calls 1
999
Configuring Toll Control using Jeeves


Local Numbers
• Click Local Numbers to open the page.
• Enter the local area numbers that are permitted to be dialed in the Allowed Numbers list and the numbers
that are to be restricted in the Denied Numbers list. You may enter as many as 999 numbers in each list.
• Click Submit at the bottom of the page to save the entries.
Regional Numbers
• Click Regional Numbers to open the page.
• Enter the regional area numbers that are permitted to be dialed in the Allowed Numbers list and the
numbers that are to be restricted in the Denied Numbers list.
• Repeat the entries you made in the Local Numbers list also in the Regional Numbers list.

National Numbers
• Click National Numbers to open the page.
• Enter the long distance numbers within the country that are to be permitted in the Allowed Numbers list
and the numbers that are to be restricted in the Denied Numbers list.
• Repeat the entries you made in the Local Numbers and Regional Numbers lists in this list.
International Numbers
• Click International Numbers to open the page.
• Enter the overseas numbers that are to be permitted in the in the Allowed Numbers list and the numbers
that are to be restricted in the Denied Numbers list.
• Repeat the entries you made in the Local Numbers, Regional Numbers and National Numbers lists in
this list.

Limited Numbers
• Click Limited Numbers 1 to open the page.
• Enter the specific numbers or digits that are to be allowed to be dialed in the Allowed Numbers list.
• Enter the specific numbers or digits that are to be restricted from being dialed in the Denied Numbers this
list.
It is not mandatory to assign the same Limited-Calls Allowed-List and Denied-List for all Time Zones of Toll
Control Level 0 or to other Toll Control Levels. You can prepare different Allowed and Denied Lists for
each Toll Control Level.
Toll Control Levels
• Click Station Basic Feature Template to open the page.
• By default all stations of SARVAM UCS are assigned Template 01. You may customize this template or
select another template.

• Go to Toll Control Level 0.
• Select the desired Call Privilege Type for each Time Zone - Working Hours, Break Hours, Non-Working
hours.
• Similarly, select the Call Privilege type for other Toll Control Levels 1, 2 and 3.
• For the type of call privilege you select, the respective number list - Local, Regional, National, International
or Limited- you configured will be automatically assigned.
Programming Toll Control using a Telephone

Local Numbers
To program Local Numbers Allowed List, dial:

• 4303-Index-Number String-#*
Where,
Index is the location at which the number should be stored in the list, from 001 to 999.
Number String can be maximum 16 digits and should be terminated with #* if it has fewer than 16
digits.
Refer the following table for codes for dialing special digits: 0-9, #, *, A, B, C, D, Flash (F), Pause (P), +,
Dot (.).
Special Digit Code
Flash (F) #2
Pause (P) #3

Special Digit Code
A #4
B #5
C #6
D #7
+ #8
Dot (.) #9
# ##
* **
To program Local Numbers Denied List, dial:

To default the Local Numbers List, dial:

Regional Numbers
To program Regional Numbers Allowed List, dial:

Where,
digits.
For keying in special digits, refer the above table.
To program Regional Numbers Denied List, dial:

To default the Regional Numbers List, dial:

National Numbers
To program National Numbers Allowed List, dial:

Where,
digits.
For keying in special digits, refer the above table.
To program National Numbers Denied List, dial:

To default the National Numbers List, dial:

To set the number list to default, dial:

• 4301-1-List Number
Where,
List Number is the number of the List you customized as Allowed/Denied List, from 01 to 16.
To program a number in a list, dial:

• 4302-List Number-Location Index-Number-#*
Where,
List is the number of the List you customized as Allowed/Denied List, from 01 to 16.
Location Index is the index number at which the number should be stored in the list, from 001 to 999.
Number is the number string you want to store at this location index on the list. The number can be
maximum 16 digits and should be terminated with #* if it has fewer than 16 digits. Refer the following
table for codes for dialing special digits: 0-9, #, *, A, B, C, D, Flash (F), Pause (P), +, Dot (.).
Special Digit Code
Flash (F) #2
Pause (P) #3
A #4
B #5
C #6
D #7
+ #8
Dot (.) #9
# ##
* **
To clear a number programmed in a List, dial:

• 4302-List Number-Location Index-#*
Where,
List Number is the number of the list from which the number is to be cleared, from 01 to 16.
Location Index is the number on this list at which the number to be cleared is stored, from 001 to 999.
Toll Control Levels
To program Toll Control - Level 0 (WH), dial:

• 5502-1-Template-07-Call Privilege Type
Where,
Template is number of the Station Basic Feature Template, from 01 to 50
07 is the feature number for Toll Control Level 0 (WH) on the template.
Call Privilege Type is
1 for No Calls
2 for Local Calls
3 for Regional Calls
4 for National Calls
5 for International Calls

6 for All Calls
7 for Limited Calls 1
To program Toll Control - Level 0 (WH) Allowed List, dial:

• 5502-1-Template-08-Number List
Where,
08 is the feature number on the template for Toll Control Level 0 (WH) Allowed List for Limited Calls.
Number List is the number of the list prepared as Allowed List for Limited Calls, from 01 to 16.
To program Toll Control - Level 0 (WH) Denied List, dial:

Where,
09 is the feature number on the template for Toll Control Level 0 (WH) Denied List for Limited Calls.
Number List is the number of the list prepared as Denied List for Limited Calls, from 01 to 16.
To program Toll Control - Level 0 (BH), dial:

Where,
10 is the feature number for Toll Control Level 0 (BH) on the template.
1 for No Calls
2 for Local Calls
6 for All Calls
To program Toll Control - Level 0 (BH) Allowed List, dial:

Where,
11 is the feature number on the template for Toll Control Level 0 (BH) Allowed List for Limited Calls.
To program Toll Control - Level 0 (BH) Denied List, dial:

Where,
09 is the feature number on the template for Toll Control Level 0 (BH) Denied List for Limited Calls.
To program Toll Control - Level 0 (NH), dial:

Where,

10 is the feature number for Toll Control Level 0 (NH) on the template.
Call Privilege Type is:
1 for No Calls
2 for Local Calls
6 for All Calls
To program Toll Control - Level 0 (NH) Allowed List, dial:

Where,
11 is the feature number on the template for Toll Control Level 0 (NH) Allowed List for Limited Calls.
To program Toll Control - Level 0 (NH) Denied List, dial:

Where,
09 is the feature number on the template for Toll Control Level 0 (NH) Denied List for Limited Calls.
To program Toll Control - Level 1, dial:

Where,
16 is the feature number for Toll Control Level 1 on the template.
Call Privilege Type is:
1 for No Calls
2 for Local Calls
6 for All Calls

Where,
1 for No Calls
2 for Local Calls

6 for All Calls

Where,
1 for No Calls
2 for Local Calls
6 for All Calls
To program Toll Control - Call Budget Consumed, dial:

Where,
19 is the number for this feature on the template.
1 for No Calls
2 for Local Calls
6 for All Calls
• Exit SE mode.
Also refer the topics “Configuring Extensions”, “Station Basic Feature Template”, “Number Lists”.

Trunk Auto Answer
What’s this?
Trunk Auto Answer enables calls landing on a trunk to be answered automatically by greeting the caller with a voice
message before the call is actually handled.
Trunk Auto Answer is useful when you want callers to remain connected until one of the landing destinations
selected for incoming trunk calls becomes free to attend to the caller.
Trunk Auto Answer is useful in call centres, railway enquiry, banks, where callers need to be notified that they
would be attended shortly, so that they do not disconnect the call.
SARVAM UCS offers two types of Trunk Auto Answer:
• For all Calls: the system answers all incoming calls landing on the trunk line.
• When Busy: the system answers incoming calls on the trunk, only if the landing destinations are busy.
• Delayed: the system first routes the calls to the Trunk Landing Group. If not answered by any extension,
the call is answered by the system.
How it works
SARVAM UCS handles incoming calls on the trunk according to the type of Trunk Auto Answer selected for the
trunk: For all Calls or When Busy or Delayed.
When Trunk Auto Answer–For all Calls is enabled on a Trunk, for each incoming call on the trunk,
• The System answers the call with a Greeting message, known as the Trunk Auto Answer Greeting, and
rings the landing destination selected for the time of the day.
The system starts the Built-In Auto Attendant Inactivity Timer (default: 60 seconds). The Trunk Auto
Greeting message is played once. You may assign a Trunk Auto Answer Greeting of your preference.
• If the landing destination does not answer before the Trunk Auto Answer Greeting message ends, the
system plays Trunk Auto Answer Ring Back Tone message to the caller.
The Ring Back Tone message is played repeatedly for the duration of the Built-In Auto Attendant Inactivity
Timer.
However, if no Trunk Auto Answer Ring Back Tone message is assigned, the system will plays Ring Back
Tone to the caller for the duration of this timer.
• If any of the landing destinations answers the call before the expiry of the Built-In Auto Attendant Inactivity
Timer, the system stops the Built-In Auto Attendant Inactivity Timer and the Ring Back Tone message, and
connects the caller to the extension that answered the call.
• If none of the landing extensions answers the call before the expiry of the Built-In Auto Attendant Inactivity
Timer, the system plays the Trunk Auto Answer Busy Bye message and releases the trunk port.

If no Trunk Auto Answer Busy Bye message is assigned, the system plays the Busy Tone for the duration
of the Busy Tone Timer and releases the trunk port.
When Trunk Auto Answer–When Busy is enabled on a Trunk, for each incoming call on the trunk,
• The System answers the with the Trunk Auto Answer Greeting message and loads the Built-In Auto
Attendant Inactivity Timer.
The Greeting message is played once.
• The System waits for any of the landing destinations selected for the time of the day to be free.
• If no landing destination is free at the end of the Trunk Auto Answer Greeting message, the system plays
Ring Back Tone or Trunk Auto Answer Ring Back Tone message, if assigned, to the caller for the duration
of the Built-In Auto Attendant Inactivity Timer.
• If any of the landing destinations is free before the expiry of the Built-In Auto Attendant Inactivity Timer, the
system places the call on that destination.
• If none of the landing destinations is free at the end of the Built-In Auto Attendant Inactivity Timer, the
system plays the Trunk Auto Answer Busy Bye message, if assigned, and releases the trunk port.
If the Busy Bye message is not assigned, the system will play the Busy Tone to the caller for the duration
of the Busy Tone Timer.
When Trunk Auto Answer–Delayed is enabled on a Trunk, for each incoming call on the trunk,
• The system first routes the incoming calls to the Trunk Landing Group. It waits for the duration of the
Delayed Trunk Auto Answer Timer (programmable; default:10 seconds) for any of the extensions in the
Trunk Landing Group to answer the call.
• If the call is not answered by any of the extensions, the System answers the call with a Greeting message,
known as the Trunk Auto Answer Greeting, and rings the landing destination selected for the time of the
day.
The system starts the Built-In Auto Attendant Inactivity Timer (default: 60 seconds). The Trunk Auto
Greeting message is played once. You may assign a Trunk Auto Answer Greeting of your preference.
• If the landing destination does not answer before the Trunk Auto Answer Greeting message ends, the
system plays Trunk Auto Answer Ring Back Tone message to the caller.
The Ring Back Tone message is played repeatedly for the duration of the Built-In Auto Attendant Inactivity
Timer.
However, if no Trunk Auto Answer Ring Back Tone message is assigned, the system will plays Ring Back
Tone to the caller for the duration of this timer.
• If any of the landing destinations answers the call before the expiry of the Built-In Auto Attendant Inactivity
Timer, the system stops the Built-In Auto Attendant Inactivity Timer and the Ring Back Tone message, and
connects the caller to the extension that answered the call.
• If none of the landing extensions answers the call before the expiry of the Built-In Auto Attendant Inactivity
Timer, the system plays the Trunk Auto Answer Busy Bye message and releases the trunk port.

If no Trunk Auto Answer Busy Bye message is assigned, the system plays the Busy Tone for the duration
of the Busy Tone Timer and releases the trunk port.
How to configure
• Enable Trunk Auto Answer on the Trunk Feature Template of the desired trunks. To know more and for
instructions, see “Trunk Feature Template”.
• Select the Trunk Auto Answer Greeting message, the Trunk Auto Answer Ring Back Tone Message,
and the Trunk Auto Answer Busy Bye Message for the Working Hours, Break Hours and Non-Working
Hours.
You may select different Greeting, Ring Back Tone and Busy Bye Message for each time zone. You can
also select Music-On-Hold instead of Ring Back Tone Message for the time zones.
Configure the Delayed Trunk Auto Answer timer, if required.
• Configure the Trunk Auto Answer related Timers, if required. The following Timers are of relevance to the
Trunk Auto Answer Feature:
• The Built-In Auto Attendant Inactivity Timer (default: 60 seconds)
• The Ring Back Tone Timer (default: 45 seconds)
• The Busy Tone Timer (default: 7 seconds)
• You may change the duration of these timers from the “System Timers and Counts” page.
The Ring Back Timer and the Busy Tone Timer are also applicable for the Ring Back Tone and the Busy
Tone played for internal calls.
• Record and assign Voice Modules for the following Voice Messages related to this feature:
• Trunk Auto Answer Greeting Message
• Trunk Auto Answer Ring Back Tone Message
• Trunk Auto Answer Busy Bye Message
For each of these messages, you can record four different messages.
See the topic “Voice Message Applications” for instructions on recording and assigning voice modules to
greeting messages.

Trunk Call Waiting
What's this?
The Trunk Call Waiting feature gives indication to the user of a busy extension about the waiting call on a Trunk.
This is a “Class of Service (COS)” dependent feature. Only those extensions which have this feature enabled in the
COS allowed to them, will be given indication of the incoming call waiting on the trunk.
How it works
This feature is not supported on SIP Extensions.
• A and B are extensions.

• Trunk Call Waiting feature is enabled in the Class of Service of B but not on A.
• There is an incoming call on a trunk for B.
• B is busy on a call with A.
• SARVAM UCS plays Beeps to B to indicate the call waiting.
• To answer the waiting call, B may dial Flash or press the HOLD key.
• B will be connected to the caller.
• A will be put on hold.
• When there is an incoming call on a trunk for A, but A is busy on another call, A will not be provided any
indication of the waiting call.
How to configure
For Trunk Call Waiting to work on an extension, it must be enabled in the Class of Service allowed to that
extension. This can be done using Jeeves as well as a Telephone.
SARVAM UCS. Template 01 is assigned CoS group 01. Trunk Call Waiting is disabled in the CoS. So, none of the
extensions of SARVAM UCS are provided call waiting indication for incoming trunk calls.
If you want to allow Trunk Call Waiting uniformly to all extensions of SARVAM UCS, simply enable this feature in
the default CoS group (01) in the default Template (01).
However, if you want to allow this feature to only selected extensions,
1. Define a CoS group with Trunk Call Waiting enabled.
3. Assign this new Template to the extensions to which Trunk Call Waiting is to be allowed.
Refer the topics “Class of Service (COS)” and “Station Basic Feature Template” for programming instructions.

Trunk Landing Group (TLG)
What’s this?
A Trunk Landing Group is a group of extensions on which incoming calls on a particular trunk are landed.
Trunk Landing Groups are formed for efficient call management. Generally, incoming calls on a trunk are landed on
the Operator extensions. However, when several trunks are interfaced with the system, it becomes difficult for the
operator to answer all calls efficiently. Trunk Landing Groups relieve the Operator to a great extent, as the incoming
calls get distributed among several extensions.
How it works
• A Trunk Landing Group (TLG) is a “Routing Group”.
• You can configure as many as 95 TLGs. Each group is numbered from 01 to 95.
• A maximum of 32 stations — SLT, DKP, SIP, ISDN, OGTB or Virtual Extensions — can included in each
Trunk Landing Group.
To use the Gateway Application of SARVAM UCS, select OGTB as the station.
• For each group that you create, you can do the following:
• set the Sequence in which the stations in the group should ring, by selecting the member stations in a
sequence from 1 to 32.
• set the Time for which each station in the group should ring, by setting the Ring Timer (default: 15
seconds).
• set each station to ring continuously till the call matures by enabling Continuous Ring (default:
disabled).
When Continuous Ring is enabled, once a station receives a ring, it rings continuously till the call
matures. The station continues to ring even as other stations in the group are hunted.
If the call is not answered even after the last station in the group has been hunted, the system will loop
back and start hunting from the first station, all over again.
• have a number of stations in the group ring simultaneously by enabling Continuous Ring on these
stations and setting the Ring Timer for these stations to ‘00’ seconds.
• set equal distribution of incoming calls on all stations in the group, by enabling Rotation for the entire
group (default: disabled).
When Rotation is enabled on a TLG, for each new call on a trunk, the system will land the call on the
extension next to the one that received the last call.
When Rotation is disabled in a TLG, for each new call on a trunk, the system will land the call on the
first free station of the TLG.

• To each Trunk, you must assign a TLG for the Time Zones, working hours, break hours and non-working
hours. You may assign the same TLG for all three Time Zones, or a different TLG for each Time Zone.
How to configure
• According to the number of trunks interfaced with your SARVAM UCS and the number of extensions you
have, identify the trunks to which you want to assign TLG for each Time Zone. This will help you decide the
number of TLGs to be formed, the type and number of extensions in each group, and their sequence.
• Decide the Trunk number to which each TLG is to be assigned.
• Configure each TLG as a Routing Group. See “Routing Group” for instructions.
• Assign the TLGs you formed for each trunk for the three Time Zones in its Trunk Feature Template. for
instructions, see “Trunk Feature Template”.
Example:
Two CO Lines (configured on software ports 001 and 002) are interfaced with SARVAM UCS.
• Incoming calls on CO 1 during working hours should land on SLT stations 2001, 2003, 2005 (configured on
software ports 008, 010, 012 respectively).
Incoming calls on CO 1during break hours and non-working hours should land on SLT stations 2002, 2004
(configured on software ports 009, 011 respectively).
• Incoming calls on CO 2 should land always on DKP stations 3001, 3002, 3003, 3009, 3010 (configured on
software ports 013, 014, 015, 016, 017).
• Incoming call on CO 1 should ring for 10 seconds on each station in the TLG.
• Incoming call on CO 2 should ring for 20 seconds on each station in the TLG.
• The stations of the TLGs of CO 1 and CO2 should ring for the set time only.
• “Rotation Method” to be followed in the TLG of CO1 and CO2.
In this example, for CO 1, you would need to form 2 TLGs; one TLG for working hours and one for break hours and
non-working hours. So, form two Routing Groups. For example, Routing Group 10 for working hours and Routing
Group 11 for break hours and non-working hours.
In Routing Group 10, select the member SLTs in this sequence: 2001, 2003 and 2005. Set the Ring Timer for each
member SLT to 10 seconds. Disable Continuous Ring for the SLT, as each station in the group must ring for the set
time.
In Routing Group 11, select the member SLT in this sequence: 2002 and 2004, and set the Ring Timer for each SLT
to 10 seconds. Disable Continuous Ring for the SLT, as each station in the group is required to ring for the set time
only.
Enable Rotation for Routing Group 10 and Routing Group 11.
For CO 2 you would need to form a common TLG for working hours, break hours and non-working hours. So,
configure as single routing group, for example, Routing Group 13 for CO 2. In Routing Group 13, select the

member stations in this sequence: 3001, 3002, 3003, 3009, 3010. Set the Ring Timer for each DKP member station
to 20 seconds. Disable Continuous ring for each DKP in the group. Enable Rotation for Routing Group 13.
Select a Trunk Feature Template number for CO 1 and CO 2. For example, feature template 04 for CO 1 and
template 05 for CO 2.
In the Feature Template of CO 1, assign TLG. For working hours, assign Routing Group 10, for break and non-
working hours, assign Routing Group 11.
In the feature Template of CO 2, assign Routing group 13 as TLG for all three time zones.

Trunk Reservation
What’s this?
This feature enables any extension user to reserve a trunk for exclusive use, for a specific time period.
Trunk Reservation can be requested from an SLT extension, a DKP extension and from an Extended IP Phone
extension.
This feature is not supported on SIP Trunks.
How it works
Let us understand this feature with the help of an example:
In an organization there are four CO trunk lines, CO 1, 2, 3 and 4, but all of these have full traffic throughout the
day.
Extension user A is a sales executive. To complete the sales target, A needs to make long-distance calls to
customers. Since there is full traffic on all the four trunks throughout the day, and these trunks are constantly busy,
A would need a dedicated trunk line to save time and complete the target.
So, A can reserve one of the four trunk lines for the desired duration. To do this,
• Extension A grabs a trunk by dialing the Trunk Access Code.
• The Trunk is busy.
• Extension A dials the feature access code for Trunk Reservation for the busy trunk.
• As soon as the trunk is free, A’s extension rings.
• A answers the call and gets connected to the trunk, and gets dial tone.
• A can now make as many calls as required.
• The trunk remains reserved for the duration of the Trunk Reservation Timer. This timer is configurable, and
by default is set to 10 minutes. A can have this Timer configured to the desired duration.
• All other extension users who try to access this trunk get error tone, even if this trunk is free.
• If A is finished with the calls before the expiry of the Trunk Reservation Timer, A has two options:
a) release the trunk manually, by cancelling Trunk Reservation.
or
b) wait for the expiry of Trunk Reservation Timer.
• Only when the trunk is released (by A or at the end of the Timer) will other users be able to access it.

How to configure
To be able to use Trunk Reservation, extension users must have this feature enabled in their Class of Service for
the time zone. For configuration instructions, see “Class of Service (COS)”, and “Station Basic Feature Template”.
You may increase or decrease the duration of the Trunk Reservation Timer. See “System Timers and Counts” for
instructions.
How to use
To set Trunk Reservation, when Trunk you access is busy,
• Press DSS Key assigned to Trunk Reservation
OR
• Dial 6 on Busy Tone
To release a reserved Trunk, wait for the Timer to expire, or cancel Trunk Reservation, manually.
To cancel Trunk Reservation Manually,
• Press DSS Key assigned to Trunk Reservation.
OR
• Dial 102
For SLT Users
To set Trunk Reservation, when Trunk you access is busy,
• Lift handset.
To cancel Trunk Reservation Manually,

• Lift handset.
• Dial 102

User Absent/Present
What's this?
Extension users may sometimes want to leave their desks, and expecting to return soon, they may not have
forwarded their calls or set Do Not Disturb on their extensions. In such cases, incoming calls will continue to land on
the extension and go unanswered. The callers have no way of knowing that the extension user is not present at the
extension and may try the extension number repeatedly.
With the User Absent/Present feature of SARVAM UCS, extension users, including the Operator, can set 'User
Absent' when they leave their desks. By doing so, they can block all incoming external as well as internal calls from
landing on their extension. When they return to their desks, they can set 'User Present' and receive incoming calls
again.
There are more options for indicating availability to other extensions. Refer the topic “Presence” to know
more.
How it works
When an extension user of EON/Extended IP Phone user sets 'User Absent', the letter 'A' appears on the phone's
display:
EON48P
Fri 29 JAN 16:58 A

617 Himanshu
The letter 'A' disappears when the extension user sets 'User Present'.
When an extension user of DKP/Extended IP Phone calls the extension which has set 'User Absent', the text
message 'User Absent' will appear on the caller's phone display.
When an SLT extension user calls the extension which has set 'User Absent', callers who dial this extension will get
an error tone.
External callers who call the extension, on which 'User Absent' is set, will get an error tone only.
• Outgoing calls can be made from the extension which has set 'User Absent'. Only incoming calls are
restricted.
• User Absent/Present can be set on an extension from the SA mode.
• If more than one extension is configured as "Operator' (routing group), incoming calls will be blocked
only on the Operator extension which has set User Absent.

• User Password is required for this feature. The default User Password, 1111, will not work. Change the
User Password first.
How to use
For Extension Users

To set User Absent on your extension:
• Dial 104-User Password-0
To set User Present on your extension:

• Dial 104-User Password-1
For System Administrator

To set User Absent on an extension:
To set User Present on an extension:

The System Administrator can also set an extension as Absent/Present using Jeeves. For instructions, read
“Additional Security to Extension Users” under the topic System Security.

User Password
What’s this?
The User Password is a 4-digit code for extension users to protect their extension phones from unauthorized use.
The default User Password is 1111. It must be changed by the extension users from their phones to any desired
value, not exceeding 4 digits.
To avoid unauthorized access,

• the extension users must change the default password.
In case the extension user forgets the password, it can be cleared and restored to the default value 1111 by the
System Engineer (SE) or the System Administrator (SA). Refer the topic “System Security” for instructions.
The User Password is also required to access and use certain features of SARVAM UCS, which are listed below.
• Call Follow Me
• Dynamic Lock
• Direct Inward System Access (DISA)
• Presence
• Hot Desk
• Phone Settings of the DKP
• Mailbox of Voice Mail
The extension user must change the default password for all the above listed features except: Phone Settings,
Mailbox of Voice Mail. Both these features allow the extension user to use the default User Password, whereas in
the case of others, the system will not allow feature access without changing the User Password.
In the case of Hot Desking, the default password will work only for one extension involved.
The User Password for an extension can be changed only from that extension phone only.
Since the Mailbox can be accessed using the default User Password, extension users who are assigned a
mailbox are recommended to change their User Password to prevent unauthorized access to their mailbox.
How to use

• Press ‘Enter’ key to enter DKP menu.
• Scroll to Change User Password.
• Press ‘Enter’ key.
• You get the prompt: Enter Old User Password.
• Dial you current user password.
If the default password 1111 has not be changed, enter default password.
• You get the prompt: Enter New User Password.
• Dial your new user password, not exceeding 4-digits.

You get confirmation tone and the confirmatory message ‘User Password Changed’ on your phone
display.
For SLT Users

• Dial 114
• Dial Current User Password (default password 1111 if not changed already).
• Dial New User Password.

Video Call
What's this?
Video calling has become an increasingly important tool in today’s business world. It offers people the power of
face-to-face communication, at reasonable cost without incurring the expense of traveling. SARVAM UCS allows
you to make and receive video calls by connecting video capable user terminals.
Among the various UC features offered by SARVAM UCS, Video calling is the most important.
SARVAM UCS supports video calling over:

• T1E1 PRI Trunks and ISDN Terminals
• SIP Trunks and SIP Extensions
How it Works
Video Calling over T1E1 PRI Trunks and ISDN Terminals

For video calling over T1E1 PRI Trunks, you require the following:
• Connect the T1E1 PRI Trunks to SARVAM UCS.
• Make sure you have a BRI Card installed in the system and set the Orientation of the BRI Port as Network.
• Connect ISDN Terminals with video capability to the BRI ports.
The minimum bandwidth required for a video stream to flow is 128 kbps. Hence, 2 BRI channels will be consumed.
When this video stream flows through the PRI line it will consume 2 PRI channels too. However, for higher
resolution you can connect multiple BRI ports, if supported by the ISDN Terminal.
To conduct a video call,
• From the ISDN Terminal dial the number of the desired party (other ISDN Terminal). This is a one to one
call.
• You can also make a video call from one ISDN Terminal of SARVAM UCS to another ISDN Terminal of
SARVAM UCS. In this case 4 BRI channels will be consumed.
How to configure
To conduct a video call you must configure the following:
• Assign Access Codes to the ISDN Terminals.

• Assign the BRI ports to the ISDN Terminals.
• For outgoing calls, assign a Station Basic Feature Template to the ISDN Terminal. Make sure the
Outgoing Trunk Bundle Group in the Station Basic Feature Template has the desired PRI lines as
members to route the outgoing calls.
• For incoming calls, configure the Outgoing Reference ID, the Incoming Reference ID and the DDI Routing
Table. Configure the same so that the incoming calls on the T1E1 ports are routed directly to the ISDN
Terminals.

For detailed information, see “Configuring BRI Trunks”, “Configuring ISDN Terminals”, “Configuring E1
Trunks”, “Configuring T1 Trunks” and “Direct Dialing-In (DDI)”.
To reduce call processing time, it is recommended that you enable Number based LCR on the OG Trunk
Bundle Group assigned in the Station Basic Feature Template. You must also configure the Number Based
LCR Table.
Video Calling over SIP Trunks and SIP Extensions

SARVAM UCS supports video calling for SIP to SIP calls only. SARVAM UCS only acts as a relay agent to place
video calls between two video capable user terminals with no additional supplementary features.
Video capable user terminals which are communicating through SARVAM UCS must confirm to the same type of
video codecs284 as per industry standards. If video codec negotiation is not successful between the end-to-end
video terminals, either the call will be converted into an audio call or the call will be dropped.
For enhanced video quality during the call, some of the advanced video attributes including profile/level (H.263+,
MPEG4, H.264), bandwidth, standard annexes, framerate, image size must be confirmed between the
communicating video terminals. SARVAM UCS will not modify or alter any of these attributes which can effect the
characteristics of the video stream during communication.
For a video call, two VOCODER channels285 of SARVAM UCS are consumed. During the entire video call, those
VOCODER channels are reserved and used till either of the parties disconnects the call.
Since SARVAM UCS support a maximum of 128 VOCODER channels. A maximum of 55 simultaneous video calls
are supported.
You can convert a normal SIP to SIP audio call into a video call only if both the SIP extensions support video
calling.
• Video calling is supported in VARTA UC Clients.

• None of the proprietary Extended SIP Phones of Matrix support video calling.
• SRTP is not supported for video calling.
When the below mentioned features are accessed during an ongoing Video call, the call will be established into an
Audio call:
• Features where-in the system establishes a conference such as 3-Party Conference, Multiparty
Conference, Conversation Recording, Call Taping, Raid etc.
• When users access Barge-In or Forced Answer.
• When users unhold, unpark or Blind Transfer the call.
• When system answers the call on a trunk, such as DID, DISA, Trunk Auto Answer, Callback on Trunk etc.
When the below mentioned features are accessed the call will always be an Audio call:
• When users make a Paging or Meet Me Paging call, access Voicemail , initiate an Emergency Conference
or make an Intercom request.
• Features where-in the system initiates the call such as ACB Return call, Auto Redial Return call, Alarm
call, Handover call etc.
284. Current industry standard video codecs include H.261, H.263, H.263p, H.264, MPEG4 etc. For more details, refer the documenta-
tion of the corresponding video terminals you are using for video calling.
285. The Channels will be reserved only for Transcoding Mode.

• To dial Flash, make sure your IP Phone supports Flash dialing.
• You can use #2, if your IP Phone does not support Flash dialing.

Virtual Extension
What’s this?
The Virtual Extension feature of SARVAM UCS enables multiple users to share one telephone instrument as their
extension, yet be considered as individual extensions by the system, with distinct extension properties and class of
service.
Such shared extensions are called Virtual Extensions, as their users do not have individual phones for their use.
Virtual Extensions are useful in laboratories, common rooms, dormitories, shop floors, and wherever it is not
feasible to provide dedicated telephone instruments to individual extension users. Virtual extensions allow you
make optimum use of the existing phones without investing in new ones.
How it works
The shared telephone instrument is called the Master Extension. A Master Extension can be an SLT, a DKP or the
Matrix Extended IP Phone.
• Virtual Extensions are assigned to the Master Extension. A Master Extension can have multiple Virtual
Extensions, but a Virtual Extension can have only one Master Extension.
• The Virtual Extension functions as any other extension of SARVAM UCS. It can be assigned all features
and facilities, like Class of Service, Toll Control, Call Forward, just like any other physical extension of
SARVAM UCS. You can assign Station Basic Feature Template and Advanced Feature Template to the
Virtual Extension.
• Incoming calls to a Virtual Extension will ring on the Master Extension.
• All incoming, outgoing, internal and external calls of the Virtual Extensions are recorded in the Station
Message Detail Records.
• To make outgoing calls, the Virtual Extension user must use the feature “Walk-In Class of Service”.
• The Virtual Extension user is logged out of the Master Extension according to the Walk Out mode
assigned to it: Walk out single call or Walk out multiple calls.
How to configure
• Make a list of the number of Virtual Extensions required by you along with their names and numbers.
• Decide the Master Extension (landing destination) that is, the Port Type and Port Number. The Port Type
may be SLT, DKP, ISDN Terminal, or SIP Extension. Port Number is the number of the software port to
which the landing destination extension is connected.

• If required, you can customize the Station Basic Feature Template and Station Advanced Feature
Template you want to assign to the Virtual Extensions. For making outgoing calls users of Virtual
Extensions must have “Walk-In Class of Service” enabled in their COS.
• Decide the Priority you want to assign to the Virtual Extensions. When an outgoing call is made from any
Virtual Extension, the ring type played to the called party will be as per the set priority.
Configuring Virtual Extensions using Jeeves

• Under Configuration, click Virtual Extensions.
Configure the following parameters for each Virtual Extension:
• Access Code: Assign Station Access Codes to the Virtual Extensions. Station Access Codes are
commonly referred to as Extension Numbers. These may be a combination of 1, 2, 3, 4, 5 and 6 digits,
which are dialed to call the Virtual Extension to which they are assigned.
To assign Station Access Codes according to your preference and requirement to a range of Virtual
Extensions, see “Assigning Access Codes to a Range of Extensions”.
more.
• Name: Assign a 'Name' to the Virtual Extension. The name may be of the person who will use the
extension. This name will be displayed on the LCD of the remote user's phone, if it is equipped with
Caller ID.

• Login Destination: Configure the Port Type and Port Number you want the Virtual Extension user to
log into to make outgoing calls.
• Landing Destination: Configure the Port Type and Port Number on which you want the Virtual
Extension user to receive incoming calls.
• Priority: Select a Priority Level for the Virtual Extension.
Each extension of the SARVAM UCS is assigned a Priority Level starting from 1, 2, 3... to 9, with '1'
being lowest Priority and '9' being highest Priority. Whenever an extension phone with higher priority
calls an extension with lower priority, a triple ring is placed on the called station. To know more, read
the feature description “Priority”.
By default, the Priority of all Virtual Extensions is set to '1-None'. So, decide what Priority Level you will
assign to each of the Virtual Extensions and set the desired level for each extension.

• Click the Advanced button at the bottom of the page and configure the following parameters:
• Mobile Number: Enter the Mobile Number of the extension user you wish to store. The Number can be
a maximum of 16 digits.
• Email ID: Enter the Email ID of the extension user you wish to store. The Email ID can be a maximum
of 64 characters.
• Group: You can assign the extension user to a Group. The system clubs together extension users
assigned the same Group. The Group can be a maximum of 16 characters. Default: Blank.
• If you have completed configuring the parameters, click Submit at the bottom of the page to save your
settings.
It is possible to default all the parameters by clicking the Default button. You can also restore default
values of the parameters of a single Virtual Extension by clicking the Default One button and specifying
the Virtual Extension Number you want to set to default.
Configuring Virtual Extension Parameters using a Telephone

To assign Access Code to a Virtual Extension, dial:
• 3109-1-Virtual Extension-Access Code-#* to assign the Access Codes to a single Virtual Extension
port.
• 3109-2-Virtual Extension-Virtual Extension-Access Code to assign the Access Codes for a range of
Virtual Extension ports.
• 3109-1-Virtual Extension-Access Code-#* to assign the Access Codes to all the Virtual Extension
ports.
Where,
Access Code is a number string of any combination of 1, 2, 3 or 4 digits. Terminate the command with
#* if the number string has fewer than 4 digits.
To clear the access codes for all the Virtual Extension, dial:
• 3109-*

To assign a Name to an Virtual Extension, dial:
• 5414-1-Virtual Extension-Name-#* to assign a Name to a single Virtual Extension port.
• 5414-2-Virtual Extension-Virtual Extension-Name-#* to assign the same Name to a range of Virtual
Extension ports.
• 5414-*-Name-#* to assign the same Name to all Virtual Extension ports.
Where,
Virtual Extension is the Software Port number of the Virtual Extension port from 001 to 512.
Name is a string of a maximum 18 alphanumeric characters. Terminate the commands with #* if the
To clear the name of the Virtual Extension, dial:

• 5414-1-Virtual Extension-#* to clear the Name of a single Virtual Extension port.
• 5414-2-Virtual Extension-Virtual Extension-#* to clear the Names of a range of Virtual Extension
ports.
• 5414-*-#* to clear the Names of all Virtual Extension ports.
To assign a Station Basic Feature Template to a Virtual Extension, dial:

• 5613-1-Virtual Extension-Template Number to assign a Template to a single Virtual Extension.
• 5613-2-Virtual Extension-Virtual Extension-Template Number to assign the same Template to a
range of Virtual Extension.
• 5613-*-Template Number to assign the same Template to all Virtual Extension.
Where,
Template is the Station Basic Feature Template from 01 to 50. Default: 01.
To assign a Station Advanced Feature Template to a Virtual Extension, dial:

• 5611-1-Virtual Extension-Station Advanced Feature Template to assign a Template to a single
Virtual Extension.
• 5611-2-Virtual Extension-Virtual Extension-Station Advanced Feature Template to assign the
same Template to a range of Virtual Extensions.
• 5611-*-Station Advanced Feature Template to assign the same Template to all Virtual Extensions.
Where,
Station Advanced Feature Template is the Station Advanced Feature Template from 01 to 50. Default:
01.
To define the Priority for a Virtual Extension, dial:

• 3919-1-Virtual Extension-Priority to define Priority for a single Virtual Extension.
• 3919-2-Virtual Extension-Virtual Extension-Priority to define the same Priority for a range of Virtual
Extensions.
• 3919-*-Priority to define the same Priority for all Virtual Extensions.
Where,
Priority is from 1 to 9. Default: 1-None.
To program Landing Destination for Virtual Extension, dial:

• 3001-1-Virtual Extension-Port Type-Port Number to program landing destination for a single Virtual
Extension.
• 3001-2-Virtual Extension- Virtual Extension -Port Access Code - Port Number to program landing
destination for a range of Virtual Extensions.
• 3001-*-Virtual Extension-Port Access Code-Port Number to program landing destination for all the
Virtual Extensions.
Where,

Port Type and Port Number is:
Port Type Port Access Code Port Number
SLT 01 001-512
DKP 02 001-128
ISDN Terminal 28 01-64
SIP Extension 34 001-500

Voice Help
What’s this?
The Voice Help feature of SARVAM UCS allows you to record and play short (16 second duration) voice messages
to provide quick help to extension users. Voice Help messages may contain important instructions, or frequently
accessed feature codes, or important phone numbers, etc.
For example, the Access Codes of the frequently used features can be recorded in the Voice Help message.
Extension users may simply dial the Voice Help feature code and listen to the voice message.
How to configure
To be able to use Voice Help, you must first record a voice module with the contents you wish to provide as help.
Record the voice module considering the maximum duration of the voice module, so that the message is not
truncated.
The voice module must be assigned to the Voice Help application. Refer topic “Voice Message Applications” for
instructions on recording the voice module and assigning it to Voice Help.
How to use

• Press DSS Key assigned to Voice Help.
OR
• Dial 1090
You will hear the Voice Help message.
• Go idle after the message ends.
For SLT Users

• Dial 1090.
• You will hear the Voice Help message.
• Replace handset after the message ends.

Voice Message Applications
What’s this?
SARVAM UCS allows you to record different voice messages which can be played to callers/extension users
according to the situation. For example, if you have activated “Auto Attendant” on some of the trunk lines, you can
use an appropriately recorded Voice Message to guide callers to reach the desired party/destination extension.
Similarly, an extension user who wants to set an alarm can record a personal voice message on his/her own, which
will be played to him/her when the alarm request is served.
If RTP mode is set as RTP Relay or Direct RTP and when calls are made to/from SIP, voice messages will not be
played for the following tones — Dial Tone, Ringback Tone, Busy Tone, Error Tone and Confirmation Tone. See
RTP Mode under “Configuring VoIP Parameters” for more details.
How it works
The voice messages are recorded in 'Voice Modules' and the voice modules are assigned to the features/
applications with which they are to be used.
The SARVAM UCS supports 15 Voice Modules of a maximum duration of 16 seconds each as well as a
customized option wherein you can customize the module size as per your requirements. By default, the option is
16 Voice Modules of 16 seconds. You can record up to 16 short messages of a maximum 16 seconds each or less.
A voice message can be of two types:
• Once-Only: the message is played only once from its start to its end.
• Continuous: the message is played repeatedly from the start to the end.
The Voice Messages you. recorded can also be Uploaded and Downloaded through Jeeves. For instructions,
see “How to configure”.
When the recorded voice modules are assigned to the features/applications, they are played to the callers/
extension users whenever the feature/application is activated.
As many as nine286 different voice messages can be played simultaneously to a caller/extension user.
Voice messages can be used for different applications or situations as described in the following.
Built-In Auto Attendant Greeting Messages

On Trunk lines that have Built-In Auto Attendant enabled, you can use a recorded Voice Message to guide callers
to reach the desired party/destination extension. It is possible to play Built-In Auto Attendant Welcome Greeting
Messages according to the time zone, that is, working hours, break hours, and non-working hours.
For example:
286. In ETERNITY LENX/MENX, 11 different voice messages can be played simultaneously to the caller/extension user.

• Built-In Auto Attendant-Time-based Greeting message in the morning: Good Morning.
• Built-In Auto Attendant-Time-based Greeting message in the afternoon: Good Afternoon.
• Built-In Auto Attendant-Time-based Greeting message in the evening: Good Evening.
• Built-In Auto Attendant - Welcome Greeting message for working hours: "Welcome to Cotton
Software".
• Built-In Auto Attendant - Welcome Greeting message for lunch hours: "Welcome to Cotton Software.
This is lunchtime. Please call after 2.00 pm".
• Built-In Auto Attendant - Welcome Greeting Message for non-working hours: "Welcome to Cotton
Software. We are closed for the day. Please call later".
• Built-In Auto Attendant - Dial Message: "Please dial the desired station number".
Built-In Auto Attendant Guidance Messages

You can use voice messages to guide Auto Attendant callers at various stages of the call. For instance, prompting
them to dial a number, or giving an alert when they dial a wrong or invalid number, or informing them when the
desired destination extension is busy or when there is no response from the dialed extension, etc.
For example,
• Built-In Auto Attendant - Dial Message: "Please dial the desired Extension Number".
• Built-In Auto Attendant - Wrong Dial Message: "Sorry you have dialed an invalid number”.
• Built-In Auto Attendant - Ring Back Tone (RBT) Message: "The number you dialed is ringing.”
• Built-In Auto Attendant - Busy Message: "The extension you have dialed is busy".
• Built-In Auto Attendant - No-Reply Message: "The Extension you have dialed is not responding".
• Built-In Auto Attendant - No-Dial Message: "Sorry you have not dialed any Extension Number".
• Built-In Auto Attendant - Conference Number287 Message: "Please dial the Conference Number".
• Built-In Auto Attendant - Conference Password Message: "Please dial the Conference Password".
• Built-In Auto Attendant - Call Transfer Message: "Transferring the call to the Operator". This message
is played to the caller, when he does not dial any number and the call is transferred to the Operator.
Voice Message for Features

• Alarms: A pre-recorded voice message is played to the extension on which Alarm is set when the wake-
up call is served. This feature is very useful in hotels where wake-up alarms are to be set for guests at the
oddest hours. With the Voice Message for alarms, guests can be greeted when they pick up the handset to
answer their phone.
287. Refer the description of the feature “Conference Dial-In”

For example, "Good Morning. This is a wake up call. You may please call room service for any assistance.
Thank you. Have a nice day".
You are recommended to record the message "Please press 0 to acknowledge the Alarm and Reminder"
as a voice module for the Alarm/Reminder message, so that extension users can acknowledge Snooze
calls. Refer the topics “Alarms” and “Reminder” to know more.
• Security/Emergency Message: A voice message is played to the external number and to the operator
extension which answers the Emergency call.
• Voice Help: You can record and play voice message to provide quick help to extension users. The Voice
Help message may contain important instructions, or frequently accessed feature codes, or important
phone numbers, etc.
For example, the Access Codes of the frequently used features can be recorded in the Voice Help
message. Extension users may simply dial the Voice Help feature code and listen to the voice message.
Help message must be recorded considering the maximum duration of the Voice Module (16 seconds), so
that voice messages are not truncated.
• Music-on-Hold: Callers who are put on hold are usually played music from an internal/external source as
they wait. You can play a voice message instead of music to the callers. The message may contain any
promotional information about your company or services provided by your organization, etc.
For example, "Welcome to Progressive Bearings. We are glad to announce that we are now an ISO 9001
company."
• Message Waiting: Whenever there is a new message in the mailbox of the extension user and if the VMS
informs the SARVAM UCS about the new message, the SARVAM UCS changes the dial tone of the
extension to a stuttered dial tone. The SARVAM UCS also offers the facility to playback a message instead
of the stuttered dial tone to indicate the waiting message. An appropriate voice message can be played
back to the extension user when he lifts the handset.
For example, "You have a new message in your Mailbox. Please access your mailbox".
• DND Notification: A voice message can be played to extension users who try to call an extension that has
set DND.
Voice Message for Dial Tone

Normally, when a caller goes OFF-Hook to dial an extension, he gets the dial tone for a fixed time interval (Dial
Tone Timer). If the caller does not respond within this time, he gets an error tone. The caller however, may not
know about the Timer. So, a message may be played to the caller.
For example, "Please dial the number immediately after the beep".
Voice Message for Ring Back Tone

When a caller dials an extension, if the station is free, the caller gets a Ring Back Tone. Instead of the ring Back
Tone, the caller can be played a message like: "The station you have dialed is free. Please wait till the station
responds".

Voice message for Busy Tone
When a caller dials an extension, if the extension is busy, the caller gets a busy tone. The SARVAM UCS can be
made to play back message instead of the busy tone, like: "The station you have dialed is currently busy. Please
dial after some time".
Voice message for Error Tone

When a caller performs a wrong operation or uses a feature without access, it is possible to play a voice message
to the caller instead of the Error Tone, such as: "Please check the number you have dialed".
Voice message for Confirmation Tone

The SARVAM UCS confirms the successful usage of features with a confirmation tone which could be replaced
with a voice message like: "The requested operation is performed successfully".
Voice Message for Toll Control/CoS Violation

When an extension dials out a number, the system checks the Toll Control assigned to that extension. If the dialed
number does not match with the number strings programmed for the Call Privilege type set for the Toll Control
assigned to the extension, the following message can be played: "Please check the number you have dialed. This
facility is not available on your Telephone".
Similarly, a voice message can be played to the extension user who attempts to invoke a feature that is not allowed
to his/her extension phone in its Class of Service (COS).
Voice Message for Trunk Auto Answer

Trunk Auto Answer enables calls landing on a trunk to be answered automatically by greeting the caller with a voice
message before the call is actually handled. For example, the caller can be played the message: "Please Wait!
Your call will be attended shortly".
SARVAM UCS supports 4 Trunk Auto Answer messages.
Direct Dialing-In Guidance Messages

You can use voice messages to inform callers that the dialed extension is busy or there is no response from the
dialed extension or to inform the caller that the call is being transferred to a Trunk Landing Group.
Voice Messages for Station User Greetings

SARVAM UCS supports two greeting messages. These messages are played to the extension users when they lift
the handset. These Messages may be festive greetings like 'Happy New Year", or an announcement like "Meeting
in the Conference Room at 4 O'clock", or any other voice message to be played to the extension users when they
lift the handset. Once the greeting message is in effect, on lifting the handset Station User Greeting1 and Greeting2
are played followed by dial tone.

Voice Guidance-Time Based Greetings
SARVAM UCS supports 3 time-based greetings: Morning, Afternoon, Evening greetings. Messages like "Good
Morning", "Good Afternoon", "Good Evening" can be recorded.
These time based greetings are played to callers before the Auto Attendant -Greeting Message is played on a Built-
In Auto Attendant enabled trunk. You can set the desired Start Time for Morning, Afternoon and Evening greetings,
see “Greeting Message Time” under System Parameters.
OG Call - Called Party Busy

SARVAM UCS allows extension users to make outgoing calls that can be routed through different networks. When
outgoing calls are made using the Mobile, SIP or ISDN networks, if the called party is busy, the network provides a
busy signal along with a busy message or tone.
In certain cases, when SARVAM UCS receives only busy signal without any message or tone from the network,
you can use a voice message (Outgoing Call-Busy message), that can be played to the callers.
Radio Extension-Dial Messages

You can use voice messages to guide Radio extension users callers at various stages of the call. For instance,
prompting them to dial a number, or giving an alert when they dial a wrong or invalid number, or informing them
when the desired destination extension is busy or when there is no response from the dialed extension, etc.
PIN Dialing
You can record and play a voice message instead of beeps to prompt the user to dial PIN while using PIN Dialing
feature. See “PIN Dialing” for more details.
How to configure
To be able to play Voice Messages, you must first record them in voice modules. Once you have recorded the
voice messages, you must assign the voice module to the appropriate Voice Message Application.
Pre-recorded voice messages are in-built in the SARVAM UCS. You may either use them when recording the voice
modules or you may record messages of your choice.

You can record Voice Messages on voice modules from a DKP/SLT connected to the SARVAM UCS.
When you record voice messages, make sure that the audio files are recorded in .wav file format, have the
attributes listed below:
• Chunk ID : "RIFF"
• ChunkSize : Filesize-8 (check file size using right click property)
• SubChunk1ID : "frnt"
• Channel :1 (mono)
• Samples per seconds per channels : 8000
• Wav Byte Rate : 8000
• Bit Per Sample: 8

Recording Voice Messages from a Phone
To record a Voice Message from a phone,

• Enter SE mode from the DKP/SLT.
Record the Voice Message, dial:

• 2502-Voice Module Number
Where,
Voice Module is from 01 to 16.
• If using a DKP, press Enter key.

• You will get a beep.
• At the end of the beep, start recording the message, by speaking into your telephone instrument or if using
an external music source (PC or Music System), play the voice message.
• Limit your message to 16 seconds.
• Go ON-Hook on completion of the message.
To check/verify the Voice Message you recorded, dial:

• 2503-Voice Module
Where,
Voice Module is from 01 to 16.
In this case, dial the number of the Voice Module you just recorded.
If the audibility of the recorded message is not satisfactory, you may repeat this procedure again.
Voice Module 01 is reserved for Music-on-Hold by default. You are advised not to assign this module to
any other Voice Message Application.
Configuring Voice Messages Application parameters using Jeeves

• Under Configuration, click Voice Message Applications to open the page.

• Click Recording Parameters to expand.
• Recording Source: This parameter displays the default source of recording as the Telephone
Instrument.
• Click Voice Module Partitions to expand.
• By default, SARVAM UCS supports 16 Voice Modules of 16 seconds each. If you want to change the
Voice Module size, select the Customize the Voice Module partitions as per my requirement
option.
• If you select Customize the Voice Module partitions as per my requirement, for each Module
Number you can customize the Module Size.
• Total Voice Module Size available for the Voice Message Applications is 256 seconds—16 Modules of
16 seconds each. Therefore, when any Module Size is increased, the system accordingly reduces the
total number of voice modules.
• The Voice Module Partitions will not be set to default values even when the system is set to default.
• Click Upload the Voice Message to expand.
• To upload recorded voice messages,
• Select the desired Voice Module Number.
• Click Browse to Select the .WAV file to be uploaded.
• Click Upload.

• Click Download the Voice Message to expand.
• To download the voice messages,
• Select the desired Voice Module Number.
• Click Download.
• Click Voice Module Assignment to expand.
• Voice Message Application-Voice Module-Duration (sec): Go to the desired Voice Message

Application to assign the voice module number you recorded for this application. For example, you have
recorded Voice Module 08 as DND Notification Message. Enter this number against "DND Notification".
• By default, the following Voice Messages Applications have been assigned to Voice Modules 01 to 13
(see table below). If the default Message recorded in the module suits your purpose, simply assign the
Voice Module number to the relevant Voice Application. The system will automatically set the duration
of the Voice Message Application.

If you want to change the duration of a recorded Voice Module, you can do so, by changing the
duration manually for the particular recorded message.
• For example, if you want to use Morning, Afternoon and Evening Greetings. You may simply assign the
default Voice Modules 02, 03, and 04.
• Refer the following table for Voice Message Applications assigned by default to the Voice Modules:
Default Voice Message Applications assigned to Voice Modules in the Enterprise Mode
Voice
Number
01 Music-On-Hold
02 Built-In Auto Attendant -Morning Good Morning!

Greeting
03 Built-In Auto Attendant -Afternoon Good Afternoon!

Greeting
04 Built-In Auto Attendant -Evening Good Evening!

Greeting

Greeting for Day Time
(Working Hours)

Greeting for Night time
(Non-working and Break hours)
07 Built-In Auto Attendant - Dial prompt / Please dial the desired number.
Radio Extension-Dial Message
08 Built-In Auto Attendant - No Dial Sorry! You have not dialed any number.
message
message
10 Built-In Auto Attendant - Destination The person you dialed is busy.

Busy message
Ringing message (Ring Back Tone)
Reply message
13 Built-In Auto Attendant - Call Transfer Please hold, transferring your call to the Operator.
to Operator message
Default Voice Message Applications assigned to Voice Modules in the Hotel Mode
Voice
Number
01 Music-On-Hold

Voice
Number
02 Built-In Auto Attendant -Morning Good Morning!

Greeting
03 Built-In Auto Attendant -Afternoon Good Afternoon!

Greeting
04 Built-In Auto Attendant -Evening Good Evening!

Greeting

Greeting for Day Time
(Working Hours)

Greeting for Night time
(Non-working and Break hours)
07 Built-In Auto Attendant - Dial prompt Please dial desired number.
08 Built-In Auto Attendant - No Dial Sorry! You have not dialed any number.
message
message
10 Built-In Auto Attendant - Destination The person you dialed is busy.

Busy message
Ringing message
(Ring Back Tone)
Reply message
13 Built-In Auto Attendant - Call Transfer Please hold, transferring your call to the Operator.
to Operator message
14 Alarm This is your wake up call.
15 Built-In Auto Attendant - Dial prompt Please dial Room Number.
16 Blank
• It is possible to assign the same Voice Module to more than one Voice Message Application.
• If you have already recorded Voice Module 08, SARVAM UCS will automatically detect and display the
duration of the Voice Module you recorded. So you need not define the duration of the Voice Module.
• You may define the duration of the Voice Module, only if you want the recorded voice message to be
played for a specific duration. For example, the message you recorded in the voice module is 15
seconds long, but you want to play only the message contents of the first 8 seconds, you can define the
duration of the message as 8 seconds.
• Voice Module 01 is reserved for Music-on-Hold by default. You are advised not to assign this module to
any other Voice Message Application.

Assigning Voice Modules to Voice Message Applications using a Telephone
• Enter SE mode.
To assign a voice message application to a voice module:

• Dial 2505-Voice Message Application Number-Voice Module
Where,
Voice Message Application Number is from 01 to 45. See table below.
Voice Module is the voice module number from 1 to 16.
By default, Voice Module assigned is 00.
Voice Message Once/Continuous

Meaning
Application Number (Not Programmable)
01 Built-In Auto Attendant -Morning Greeting Once
02 Built-In Auto Attendant - Afternoon Greeting Once
03 Built-In Auto Attendant - Evening Greeting Once
04 Built-In Auto Attendant - Welcome Greeting Once

for Working Hours

for Break Hours

for Non-working Hours
07 Built-In Auto Attendant - Dial Prompt Once
08 Built-In Auto Attendant - Destination Ringing Continuous

Message
09 Built-In Auto Attendant - Wrong Dial Message Once
10 Built-In Auto Attendant - Destination Busy Once

Message
11 Built-In Auto Attendant - Destination No Reply Once

Message
12 Built-In Auto Attendant - No Dial Message Once
13 Built-In Auto Attendant - Conference Number Once
14 Built-In Auto Attendant - Conference Once

Password
15 Built-In Auto Attendant - Call Transfer to Once

Operator
16 Toll - Control & COS Violation Once
17 Trunk Auto Answer Greeting 1 Once

Voice Message Once/Continuous
Meaning
Application Number (Not Programmable)
21 Trunk Auto Answer RBT Message 1 Continuous
25 Trunk Auto Answer Bye Message 1 Once
29 DDI - Destination No Reply Message Once
30 DDI - Destination Busy Message Once
31 DDI - Call Transfer to TLG Message Once
32 Alarm Continuous
33 Station User Greeting 1 Once
34 Station User Greeting 2 Once
35 Message Wait Notification Continuous
36 Message Wait - No message Once
37 DND Notification Once
38 Security/Emergency Message Continuous
39 Dial Tone Once
40 Ring Back Tone Continuous
41 Busy Tone Once
42 Error Tone Once
43 Confirmation Tone Once
44 Voice Help Continuous
45 OG Call - Called Party is Busy Once
46 Radio Extension-Dial Message Once
To use the default Voice Modules, refer the table “Default Voice Message Applications assigned to Voice
Modules in the Enterprise Mode”, and assign the desired Voice Module to a Voice Message Application.
To define voice message duration for voice modules288:

• Dial 2504-Voice Module-Duration
Where,
Voice Module is 01 to 16.
Duration is from 00 to 16 seconds.
288. Necessary only if the duration is less than 16 seconds.

By default, Duration of Voice Modules is 16 seconds.
To clear the voice module assigned to a voice message application:

• Dial 2505-Voice Message Application Number-00
Where,
VM Application Number is from 01 to 46.
• Exit SE mode.
Assigning Station User Greeting Message to Extensions
This is done from SA mode only, using a DSS Key (if the phone is EON and a key is programmed for this function)
or by dialing SA command strings.
To set/cancel a User Greeting message on an extension phone, from SA mode:

• Dial 1072-008-Extension Number-Greeting Number-Code
Where,
Greeting Number is
0 for Greeting Message 1
1 for Greeting Message 2
Code is
0 for Cancel
1 for Set

Walk-In Class of Service
What’s this?
Every extension of SARVAM UCS is assigned a Class of Service and Toll Control defining its access to features
and its calling permission.
With Walk-In Class of Service, extension users of SARVAM UCS can make calls or access features from any other
extension of the system as per the Class of Service, Toll Control and other features / facilities assigned to their own
extension.
This feature is useful to extension users who frequently move away from their desk, as it allows them the same
level of feature access and calling permission as their own extension, from another extension.
Extension users can 'Walk-In' from any extension port: DKP, SLT, ISDN Terminal, E&M (with Station as Orientation
Type), and SIP.
SARVAM UCS offers two types of Walk-In:
• One call: The extension user is automatically logged out from the extension into which the user has
walked-in, after one call.
• Multiple Calls: The extension user can make as many calls as desired, and remains 'walked-in' until the
user dials the feature code to 'Walk-Out', or until another extension user walks into the same extension.
To allow ‘One call’ or ‘Multiple Calls’ to an extension, you need to set the 'Walk-Out Mode’ in the “Station Advanced
Feature Template” of the extensions.
Walk-In Class of Service is a password-protected facility and the default User Password 1111 will not be accepted.
To be able to walk into another extension, extension users must first change their User Password to another value.
How it works
With the help of this illustration, let us understand how Walk-in Class of Service works.
In this illustration, Extension user A has a DKP with the number 3001, with long distance calling facility (toll control:
All calls). Extension user B has an SLT with the number 2001, without long distance calling (toll control: local calls).
Here,
• 3001 is the Source Extension, whose CoS, Toll Control and other features / facilities (assigned in the
Station Basic Feature template) are used from another extension (2001) by performing Walk-In.
• 2001 is the Destination Extension on which Walk-In is performed.
Walking into another extension to make calls

• Now, extension user A is at B's desk and needs to make a long distance call. B’s extension does not have
long distance calling.

• Extension user A can 'Walk-In' into B's extension (2001) by dialing
• the feature code for 'Walk-In Class of Service'.
• A’s extension number, 3001.
• A’s User Password (the default password 1111 will not be accepted, it must be changed first).
• On successful Walk-In, SARVAM UCS applies the Class of Service, Toll Control and other features /
facilities as per the Station Basic Feature Template of the Source Extension 3001 on the Destination
Extension 2001.
• Extension user A can make external and internal call from extension B.
• If Extension 2001 has 'One Call' selected as the 'Walk-Out Mode' for the extension, A will be 'Walked-Out'
when the current call ends or if A goes ON-Hook at any time after walking into extension 2001.
• If Extension 2001 has 'Multiple Calls' selected as the 'Walk-Out Mode' for the extension, extension user A
must manually walk out by dialing the feature code for 'Walk-Out'.
• If Extension user A does not 'Walk-Out', the system will perform a walk out for A only when another
extension user walks into extension 2001.
At a time, only one extension user can walk in to another extension.
Calls made after walk-in will be charged to the Source Extension. Here, calls made by Extension user A
from extension 2001 using Walk-In will be calculated and charged to extension 3001 only.
Call record details of calls made after walk-in will be recorded for the Source Extension. Here, call record
details of calls made by Extension user A from extension 2001, using Walk-In, will be recorded in the
Station Message Detail Record of extension 3001 only.
Walking into another extension to access a feature
• Extension user A with number 3001 has Call Forward in the Class of Service.
• Extension user B with number 2001 does not have Call Forward in the Class of Service.
• Extension user A is currently at B’s desk. A needs to forward calls of own extension to an external number.
To do this,
• A walks into B’s extension by dialing
• the feature code for 'Walk-In Class of Service'.
• A’s extension number (3001)
• A’s User Password289
• On successful Walk-In, the system applies the Class of Service, Toll Control and other features / facilities
as per the Station Basic Feature Template of the Source Extension 3001 on the Destination Extension
2001.
• From extension 2001, extension user A sets Call Forward for 3001, to an external number.
289. The default password 1111 will not be accepted, it must be changed first.

• Call Forward can be canceled from the Source Extension or from the Destination Extension. To cancel
Call Forward, extension user A can go back to 3001 and cancel Call Forward from 3001, or can cancel Call
Forward from 2001, if user A is still walked-in on 2001.
CAUTION! The Destination Extension user can access their Class of Service or Toll Control only after the
Source Extension user has walked out from their extension. For example, user B cannot set or cancel Call
forward on extension 2001, until user A has walked out from 2001.
Incoming calls on the Destination Extension (2001) will work according to the setting of the Destination
Extension only, whereas outgoing calls on the Destination Extension will work according the settings of the
Source Extension (3001).
The following set of features of the Destination Extension will remain unaffected:
• Key map
• Language
• Priority
• Call Pick-Up group
• Personal Directory
CAUTION! There is a risk of fraudulent calls being made from your extension, if a third party comes to
know the User Password of your extension. The cost of such fraudulent calls will have to be borne by the
owner of SARVAM UCS.
So, protect your system from unauthorized access and misuse by putting strong authentication
mechanisms in place.
• Keep Passwords strictly confidential.
• Change Passwords regularly.
• Choose Passwords that are complex and difficult to guess.
How to configure
This feature is available to all extensions of SARVAM UCS. All you need to do is, select of the 'Walk-Out Mode' for
the extensions in their “Station Advanced Feature Template”.
By default, ‘One call’ is selected as the ‘Walk -Out’ mode in the default Station Advanced Feature Template 01
assigned to all extensions.
If you want to allow different walk-out modes to different extensions, use different Templates, but make sure other
features on these templates are also configured according to requirement.
For detailed instructions refer the topics “Customizing Station Advanced Feature Template using Jeeves” and
“Customizing Station Advanced Feature Template using a Telephone”.
How to use

To perform a Walk-In, on the Destination Extension,
• Press DSS Key assigned to Walk-In Class of Service (if programmed).

OR

• Dial 111
• Select ‘Walk-in’ and press Enter key.
• You get the prompt 'Walk in from which Station?'
• You get the prompt 'Enter your User Password'.
• Dial your User Password (default password will not be accepted).
• You get the confirmation message 'Walked In Successfully' on the phone’s display and a confirmation
tone.
• You can now make your call(s) or access a feature.
To perform a Walk-Out, on the Destination Extension or on the Source Extension,
• Press DSS Key assigned to Walk-In Class of Service.

OR
• Dial 111.
• Select 'Walk-Out’ on your phone's display.
• You get the confirmation message 'Walked Out' on your phone's display and confirmation tone.
You need to perform a Walk-Out only if the Source Extension as Multiple Calls set as Walk-Out mode.
If the Source Extension has 'One Call’ set as the Walk-Out mode, you will be walked out of the Destination
Extension when you go ON-Hook after making a call or accessing a feature.
If the extension you are walking in has 'One Call’ as the Walk-Out mode, and you go ON-Hook before you
make the call, you will be 'Walked Out'. You must Walk-In again.
For SLT Users

To perform a Walk-In:
• Lift the handset of the Destination Extension.
• Dial 111-1
• Dial your User Password.
• You get a confirmation tone.
• You can make your calls or access features now, during the confirmation tone or after you get the dial
tone.
To perform a Walk-Out:
• Lift the handset of the Destination Extension or the Source Extension.
• Dial 111-0
• You get a confirmation tone.

CHAPTER 15 Voice Mail Features
If you have CDMA Mobile Card installed in your system, it is recommended to avoid using Voice Mail
features as the DTMF Detection might not work efficiently.
Accessing your Mailbox

You will be able to access your mailbox only if there is a free VMS channel. For more details, see “Configuring VMS
To access your Mailbox from your own extension:
• Dial 3931.
• System prompts you with:
“You have no new messages”, if there are no new messages in your mailbox.
“You have n new messages”, if there are new messages in your mailbox (n = No. Of messages).
• Enter your Mailbox password.
• You will enter the “Mailbox Access” menu.
To access your Mailbox from another extension:
• Dial 3941/3942/ 3943.
• System prompts you with, “Welcome! Please dial the extension number or To dial by name press ‘6’,To
leave a message press ‘7’, To access your Personal Mailbox press ‘8’, For further assistance press ‘9’, To
disconnect the call press ‘#’.
• Dial 8. System prompts you to dial extension number.
• Dial your extension number to access your mailbox.
“You have n new messages”, if there are new messages in your mailbox (n = No. of messages).

To access your Mailbox through an external number using DISA Login:
• Log into the SARVAM UCS using DISA.
• If CLI Based Authentication is enabled, the system prompts you to dial desired extension number.
• If PIN Authentication is enabled, the system prompts you to dial your Extension Number and Password.
• Dial 3931 to access your Mailbox.
To access your Mailbox using VMS Auto Attendant:
• Call the trunk on which Voice Mail Auto Attendant is enabled.
• The incoming call on the trunk is answered by the VMS Auto Attendant.
• The VMS greets the caller with the Greeting message followed by the Welcome Message: “Welcome!
Please dial the extension number or To dial by name press ‘6’,To leave a message press ‘7’, To access
your Personal Mailbox press ‘8’, For further assistance press ‘9’, To disconnect the call press ‘#’.
• Dial 8. System prompts you to dial extension number.
• Dial your extension number to access your mailbox.

Alarms and Reminders
What’s this?
The Voice Mail System of SARVAM UCS offers voice-guided Alarms and Reminders, which can be set by the
Operator as well as extension users.
How it works
Voice-guided Alarm and Reminder requests are served as per the date and time set by extension users. The
different ways in which Alarm or Reminder requests will be served are described in the following:
• Alarm with Snooze Off (Once Only)
• The VMS plays system greeting for the current time zone and the Extension Name followed by the
message "This is your Wake up Call. Music of 5 seconds."
• Alarm with Snooze On (Once Only)
• The VMS plays system greeting as per time zone and the Extension Name followed by the message
"This is your Wake up Call. For Acknowledge, Please Press 0. Music of 5 seconds."
• When user press '0', VMS prompts: "Your Alarm is Acknowledged."
• Daily Alarm with Snooze Off
• The VMS plays system greeting as per the time zone and the Extension Name followed by the
message "This is your Daily Wake up Call. Music of 5 seconds."
• Daily Alarm with Snooze On
• VMS plays system greeting as per time zone and the Extension Name followed by the message "This
is your Daily Wake up Call. For Acknowledge, Please Press 0. Music of 5 seconds."
• When user press '0', VMS prompts: "Your Alarm is Acknowledged."
• Reminder with Snooze Off
is your Reminder call. Music of 5 seconds."
• Reminder with Snooze On
is your Reminder Call. For Acknowledge, Please Press 0. Music of 5 seconds."
• When user press '0', VMS prompts: "Your Reminder is Acknowledged."
How to configure
The VMS allows you to enable/disable the Alarm Verification for alarms and reminders, allowing extension users
who want to use alarms and reminders to confirm

• Time set for an alarm
• Date and time set as a reminder.
When Alarm Verification is disabled, the VMS will not confirm the alarm and reminder set by the extension user.
If you want the VMS prompts to be played when the extension users answer an Alarm Call/Reminder Call and
acknowledge it, make sure you have selected Voice Mail as Alarm Notification Type.
To configure Alarm Verification,
• On the System Parameters page, click Alarm to expand.
• Select the Voice Guided Alarm Verification check box to enable extension users to confirm the Time
they have set for an alarm or the Date and Time they have set for a reminder. Default: Enabled.
To configure Alarm Notification Type,
• Scroll to Alarm Notiifcation Type and select Voice Mail.
• Assign this Station Advanced Feature Template to the desired extensions.

How to use
Alarm set by Extension Users
• Pick up the handset of your telephone and dial 163  VMS prompts: "Enter the time, HH MM in twenty four
hour format290. To cancel all alarms, press # (pound/hash)."
• If no time is entered, VMS prompts: "You have not entered any input"
• If invalid time is entered, VMS prompts: "You have entered invalid input."
• To set alarm, dial valid time  VMS prompts: "To set once, Press 1, To set Daily Press 2."
Once Only
• Dial 1  VMS responds with: "You have set Wake up Alarm at …." (WakeupVeri.wav) followed by the
prompt: To Confirm291, Press 1, To Re-enter, Press 2."
• Dial 1 to confirm the time set for alarm  VMS responds with: "Your Wake up Alarm is set." followed by the
prompt: "Thanks for using this Service."
• If alarm is not set, the VMS responds with: "Sorry! Your Wake Up Alarm cannot be set. Please call
Operator for further assistance." VMS further responds with: "Thanks for using this Service."
Daily Alarm
• Dial 2  VMS responds with: "You have set Daily Wake up Alarm at …." followed by the prompt: "To
Confirm, Press 1, To Re-enter, Press 2."
• Dial 1 to confirm the time set for alarm  VMS responds with: "Your Daily Wake up Alarm is set." followed
by the prompt: "Thanks for using this Service."
• If no alarm is set, the VMS responds with: "Sorry! Your Wake Up Alarm cannot be set. Please call
Operator for further assistance." The VMS further responds with: "Thanks for using this Service."
• Dial # (pound/hash) to cancel all alarms  VMS responds with: "Your all Wake up Alarm are canceled.”
followed by the prompt: "Thanks for using this Service."
• If no alarms are set, the VMS responds with: "Sorry! There is no Alarm to cancel." followed by the
Alarm set by Operator

• Enter System Administrator Mode, dialing 1072.
• Dial 034  the VMS prompts: "Enter the Extension number for which you have to set or cancel Wake Up
Alarm."
• Dial 1 to select the extension user for which the Alarm is to be set. VMS responds with: "Enter the time, HH
MM in twenty-four hour format. To cancel all alarms, press '#’ (pound/hash)'."
290. The Date and time format depends on the Region/Country selected for the system.
291. This option will not be played if Alarm Verification is disabled in the System Parameters.

• If no time is entered, the VMS prompts: "You have not entered any input"
• If invalid time is entered, the VMS prompts: "You have entered invalid input."
• To set alarm, dial valid time  VMS prompts: "To set once, Press '1', To set Daily Press '2'.”
Once Only
• Dial 1  the VMS responds with: "To set it as Personal, Press 1. To set it as Automated, Press 2."
• Dial 1  VMS responds with: "You have set Personal Wake up alarm at…." followed by the prompt: "To
• Dial 1  VMS responds with: "Your Personal Wake up Alarm is set." followed by the prompt: "Thanks
for using this Service."
• If alarm is not set, the VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please call
OR
• Dial 2  the VMS responds with: "You have set Automated Wake up alarm at…." followed by the
prompt: "To Confirm, Press 1, To Re-enter, Press 2."
• Dial 1  the VMS responds with: "Your Automated Wake up Alarm is set." followed by the prompt:
"Thanks for using this Service."
• If alarm is not set, VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please call
Daily Alarm
• Dial 2  VMS responds with: "To set it as Personal, Press 1. To set it as Automated, Press 2."
• Dial 1  VMS responds with: "You have set Daily Personal Wake up alarm at…." followed by the
• Dial 1  VMS responds with: "Your Daily Personal Wake up Alarm is set." followed by the prompt:
• If the alarm is not set, the VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please
call Operator for further assistance." VMS further responds with: "Thanks for using this Service."
OR
• Dial 2  the VMS responds with: "You have set Daily Automated Wake up alarm at…." followed by the
• Dial 1  VMS responds with: "Your Daily Automated Wake up Alarm is set." followed by the prompt:
• If alarm is not set, VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please call Operator
for further assistance." VMS further responds with: "Thanks for using this Service."

Reminders set by Extension Users
• Pick up handset of your telephone and dial 164  VMS prompts: "Enter the Date in DD MM YYYY
format292. To Cancel all Reminders, Press '#’ (pound/hash)'. For example, To enter Date 17th March
2008, Dial One Seven Zero Three Two Zero Zero Eight."
• If no date is entered then VMS prompts: "You have not entered any input"
• If invalid date is entered then VMS prompts: "You have entered invalid input."
• Dial valid Date  the VMS prompts: "Enter the time, HH MM in twenty four hour format."
• If no time is entered, the VMS prompts: "You have not entered any input"
• If invalid time is entered, the VMS prompts: "You have entered invalid input."
• Dial valid time  VMS prompts: "You have set Reminder for..." followed by the prompt: "To Confirm, Press
1, To Re-enter, Press 2."
• Dial 1 to confirm the date and time set for Reminder  the VMS responds with: "Your Reminder is set."
followed by the prompt: "Thanks for using this Service."
• If Reminder is not set, the VMS responds with: "Sorry! Your Reminder cannot be set. Please call
• Dial # (pound/hash) to cancel Reminder  the VMS responds with: "Your Reminder is canceled.” followed
by the prompt: "Thanks for using this Service."
• If no reminder is set, the VMS responds with: "Sorry! There is no Reminder to cancel." followed by the
Reminders set by Operator

• Enter System Administrator Mode, dialing 1072.
• Dial 1072 to enter SA mode followed by 035  VMS prompts: "Enter the Extension number for which you
have to set or cancel Reminder."
• Dial 1 to select the station user for which the Reminder is to be set. VMS responds with: "Enter the Date in
DD MM YYYY format. To Cancel all Reminders, Press '# (pound/hash)'. For example, To enter Date 17th
March 2008, Dial One Seven Zero Three Two Zero Zero Eight."
• If no date is entered, the VMS prompts: "You have not entered any input"
• If invalid date is entered then VMS prompts: "You have entered invalid input."
• Dial valid date  VMS prompts: "Enter the time, HH MM in twenty-four hour format."
• If no time is entered then VMS prompts: "You have not entered any input"
• If invalid time is entered then VMS prompts: "You have entered invalid input."
292. The date format in the prompt will be MM DD YYYY, if you selected USA as the Region/Country for your system.

• Dial valid time  VMS prompts: "To set it as Personal, Press 1. To set it as Automated, Press 2."
• Dial 1  VMS responds with: "You have set Personal Reminder for…." followed by the prompt: "To
Confirm293, Press 1, To Re-enter, Press 2."
• Dial 1  VMS responds with: "Your Personal Reminder is set." followed by the prompt: Thanks for
using this Service."
• If Reminder is not set, the VMS responds with: "Sorry! Your Reminder cannot be set. Please call
OR
• Dial 2  VMS responds with: "You have set Automated Reminder for…." followed by the prompt: "To
• Dial 1  VMS responds with: "Your Automated Reminder is set." followed by the prompt: "Thanks for
using this Service."
• If Reminder is not set, VMS responds with: "Sorry! Your Reminder cannot be set. Please call Operator
for further assistance." VMS further responds with: "Thanks for using this Service."
• Dial # to cancel Reminder  VMS responds with: "Your Reminder is canceled.” followed by the prompt:
• If no reminder is set, VMS responds with: "Sorry! There is no Reminder to cancel." followed by the
293. This option will not be played, if Alarm Verification is disabled in the System parameters.

VMS DISA Login
What’s this?
This feature allows the remote users to log into the DISA mode using the trunks on which the Voice Mail Auto
Attendant is enabled. The remote user can access and use the system's features and facilities using the DISA
enabled trunks.
The VMS supports DISA-PIN Authentication-Multiple Calls only. For detailed information on the types of DISA
Variants, see “Direct Inward System Access (DISA)”.
Using VMS DISA login, remote users can:

• call any extension.
• make external calls.
• use features and facilities of the system.
• configure features and facilities of the system and administer the system.
All these can be done as if being done from a local extension of the SARVAM UCS.
How it works
• a VMS module must be installed in the system.
• you must enable DISA-PIN Authentication-Multiple Calls on the desired trunk: CO, Mobile, SIP,
T1E1PRI, BRI.
• you must select Voice Mail Auto Attendant as the Auto Attendant option on the desired trunks: CO,
Mobile, SIP, T1E1PRI, BRI.
• you must enable DISA in the “Class of Service (COS)” of the extension the caller is allowed to log into
(using PIN Authentication).
• you must change the default User Password (1111) of the extension the caller is allowed to log into.
This is how VMS DISA Login works:
• A call lands on a DISA enabled Trunk.
• The incoming call on the trunk is answered by the VMS Auto Attendant. By default, the VMS greets the
caller with the Greeting message followed by the Welcome Message: “Welcome! Please dial the extension
number or To dial by name press ‘6’,To leave a message press ‘7’, To access your Personal Mailbox press
‘8’, For further assistance press ‘9’, To disconnect the call press ‘#’.
• The caller must dial the DISA Login Code, 1079. The VMS checks if DISA-PIN Authentication-Multiple
Calls is enabled on the trunk for the current time zone, that is, working hours, break-hours and non-
working hours.

• The VMS finds DISA is enabled on the trunk and prompts the caller: “Please enter the Extension Number.”
and starts the First Digit Wait Timer (programmable; default: 25 seconds).
• The caller must dial the DISA Extension Number before the expiry of this timer.
• The VMS checks the CoS of the dialed Extension Number.
• DISA is enabled in the CoS of the dialed Extension Number, the VMS prompts: “Please enter your
password.” and waits to receive digits till the expiry of the First Digit Wait Timer.
• The Password must be dialed by the caller before the expiry of this timer.
• The VMS checks if the Password is valid.
• The Password is valid and the DISA Login is successful. The caller is logged into the dialed Extension
Number.
• The VMS hands over the DISA call to the SARVAM UCS.
• When the caller goes Off-hook by dialing the Off-hook code #1, the system plays the internal dial tone and
waits for the caller to dial digits.
(configurable; default: 2 minutes)294.
• The system waits for the caller to dial digits within the DISA Inactivity Timer.
• The system reloads this timer each time it receives digits from the caller. If the caller fails to dial any digit
within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed; 15
seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA session.
If digits are received before the end of the Warning Beeps, the system reloads the DISA Inactivity Timer.
• The caller can make as many trunk calls and internal calls as the caller wants.
• The caller can terminate the DISA login session either by disconnecting from the remote end or by dialing
the Termination Code #9.
The VMS plays the default Greeting Message, Welcome Greeting and DISA prompts to the callers. You can
customize them as per your requirement, if required.
How to configure
• For instructions to enable DISA-PIN Authentication-Multiple Calls and Voice Mail Auto Attendant on the
desired trunks, see the topic “Trunk Feature Template” in Configuring Trunks.
• For instructions to enable DISA in the CoS of the extensions which you want to allow callers to access
using DISA, see “Class of Service (COS)”.
• To change the default User Password (1111) of the extensions which you want to allow callers to access
using DISA, see “User Password” and “System Security”.
294. DISA Inactivity Timer is not applicable for T1E1PRI lines, BRI lines, SIP and Mobile trunks.

• To set the DISA Timers as per your requirement, see “System Timers and Counts”.
• To customize the DISA prompts as per your requirement, see “Recording Voice Messages”.
• To upload the customized Voice Prompts, see “Prompts Management”.

Sending Messages
What’s this?
The VMS enables extension users to send messages to other extensions that have a mailbox. An extension user
can send a message to as many as 10 destinations at a time. The extension user can send the message either to a
specific mailbox or to a Distribution List.
VMS also gives facility to the Sender of the message to request a read receipt of the message sent. When the
Recipient has read the message, the VMS generates a file containing the first 5 seconds of the message that was
sent and delivers it to the Sender's mailbox in the form of a new message with the Date and Time stamp and the
prompt: "This message was read by <Extension Name> <5 seconds of message sent>". If the Sender does not
request ‘read receipt’, no such message is delivered to the Sender.
How to use
• Call the VMS by dialing 3931,
• The VMS takes you to your mailbox.
• VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password".
• Enter your mailbox password  VMS prompts: "You have n new/no new messages."
• Dial 3  VMS prompts: "Enter the destination/s and dial hash to end."
• Dial valid extension numbers/distribution list number  VMS prompts: "Record your message after the
beep and press any digit to end".
• Speak to record the message and press any digit to end.
The system provides an option to the caller to verify the message before storing the same. Enable the
Message Verification check box and configure the digits for the Message Leave Options. The message will
be played as per the digits selected for the options. For detailed instructions, see “Message Send/Forward
Settings”
• VMS prompts "To request read receipt press '1', to ignore read receipt press '2'."
• If Message Verification flag is disabled then VMS will not playback the recorded message.
• Dial 1 to request read receipt of your message, else dial 2.
• VMS responds: "Message sent as normal."

Extension users must be careful in dialing destination numbers. If invalid destination is entered then the
VMS will clear all the entries and will ask the mailbox owner to re-enter all the destinations again.
Once a valid destination number is entered and no more extensions are selected, the VMS
understands it to be the end of list and sends the message.

Redirecting Messages
What’s this?
The VMS offers extension users to re-direct the messages in their mailbox to another mailbox. The feature can be
used by employees who are out of office or unable to access their mailbox. Using Redirect Messages, they can
ensure that important messages are attended to by their colleagues in their absence.
To be able to use this feature, make sure a digit has been assigned to Mailbox Management. For
instructions, see “Mailbox Access”.
How to use
password".
• Enter your mailbox password  VMS prompts: "You have n new/no new messages."
• Dial 4  The VMS responds with: “For Mailbox Name, press '1', For message redirection, press '2', To
delete all old messages of your mailbox, press '3', To delete all messages of your mailbox, press '4', For
Mailbox Greetings, press ‘5’, To go to previous menu press '#'.”
• Dial 2  VMS prompts: "To set message redirection press '1', to cancel message redirection press '2', to
go to previous menu press '#."
• Dial 1 to set message redirection  VMS prompts: "Enter the destination extension."
• Dial valid destination extension  VMS responds: "Message redirection set."
• Dial 2 to cancel message redirection  VMS responds: "Message redirection cancelled."
Message Redirection using SA Jeeves
For Extension Users


• Click Submit.
• Click Redirect VMS Messages to expand.
• In Redirect Messages to Extension, enter the extension number on which you want the messages to be
redirected.
• Click the Apply Message Redirection button.
For Department Group


• Click Department Group Properties.
• Click the desired Department Group Number tab for which you want to set message redirection.
• Click Redirect VMS Messages to expand.
• In Redirect Messages to Extension, enter the extension number on which you want the messages to be
redirected.
• Click the Apply Message Redirection button.
Message Redirection using SA Commands

• Enter SA mode from a DKP/SLT.
To redirect Messages, dial:

• 1072-314-Source Station-Code-Destination Station
Where,
Source Station is the Access Code of any SLT, DKP, SIP Extension, ISDN, Department Group.
Destination Station is the Access Code of any SLT, DKP, SIP Extension, ISDN, Department Group or
General Mailbox.
Code is
0 to Cancel Message Redirection
1 is to Set Message Redirection
• Exit SA mode.

Recording Personal Greetings
What’s this?
The VMS offers extension users the facility to customize the greeting messages for their mailbox. Extension users
can record a different message for each time zone, namely working hours, break hours and non-working hours.
How to use
password".
• Enter your mailbox password and the VMS prompts: "You have n new/no new messages.”
• Dial 5  The VMS prompts: For Personal Greetings, press ‘1’, For Conditional Greetings, press ‘2’, To go
to previous menu press '#'.
• Dial 1  The VMS prompts: "For working hours greeting, press '1', for break hours greetings, press '2', for
non-working hours greeting, press ‘3, to go to the previous menu, press ‘#’’.
• Dial the digit of the desired time zone. The VMS prompts: "To record, press ‘1’, to play, press ‘2’, to erase,
press ’3’, to go to the previous menu, press ‘#’ ”.
• To record the message, press 1. You can record a greeting message of 120 seconds. If the message
duration is less than 60 seconds, press # after you have completed the message.
• The VMS prompts: "To record, press ‘1’, to play, press ‘2’, to erase, press ’3’, to go to the previous menu,
press ‘#’ ”.
• To listen to the greeting message you recorded, press ‘2’.
• On completion of play back of the greeting message, the VMS prompts: "To record, press ‘1’, to play, press
‘2’, to erase, press ’3’, to go to the previous menu, press ‘#’ ”.

Recording Conditional Greetings
What’s this?
The VMS offers extension users the facility to customize the greetings messages played to callers for certain
conditions—busy, no reply or unconditional/unregistered Call Forward. Extension users can record a different
message for each condition.
How to use
password".
• Enter your mailbox password and the VMS prompts: "You have n new/no new messages."
• Dial 5  The VMS prompts: For Personal Greetings, press ‘1’, For Conditional Greetings, press ‘2’, To go
to previous menu press '#'.
• Dial 2  The VMS prompts: "For Busy press '1', for No Reply press '2', for Unconditional press ‘3’, To go to
previous menu press '#'.
• Dial the desired digit. The VMS prompts: "To record press ‘1’, to play press ‘2’, to erase press ’3’, to go to
the previous menu press ‘#’ ”.
• To record the message, press 1. You can record a greeting message of 120 seconds. If the message
duration is less than 60 seconds, press # after you have completed the message.
• The VMS prompts: "To record press ‘1’, to play press ‘2’, to erase press ’3’, to go to the previous menu
press ‘#’ ”.
• To listen to the greeting message you recorded, press ‘2’.
• On completion of play back of the greeting message, the VMS prompts: "To record press ‘1’, to play press
‘2’, to erase press ’3’, to go to the previous menu press ‘#’ ”.

Message Verification
What’s this?
Message Verification enables
• extension users to check the message they have recorded before sending it to someone.
• extension user to check the message they have recorded as a reply to any message.
• external callers to check the message they have recorded in the mailbox of an extension user.
Thus Message Verification is used in the VMS features — Messages left by External Callers, “Sending Messages”,
and “Broadcast Message”.
How it works
• For Message Verification to work, it must be enabled in the VMS.
• With Message Verification enabled, each time a caller or an extension user records a message, the VMS
offers to the caller/extension user the option to verify the recorded message and re-record the message, if
they want.
• When the caller/extension user uses the option to verify and re-record the message, the VMS sends the
message to the mailbox of the receiver.
How to configure
By default, Message Verification is disabled in the VMS. So, callers and all extension users with voice mail facility
cannot verify the message they have recorded. If you want to enable this feature,
• Open Jeeves.
• Click Configuration, click VMS Configuration.
• Click Message Profile.
• To allow external callers to check the message they have recorded in the mailbox of an extension user or
extension user to check the message they have recorded as a reply to any message.

• Click Message Leave Settings to expand.
• Select the Message Verification check box, to enable.
• To allow extension users to check the message they have recorded before sending it to someone,
• Click Message Send/Forward Settings to expand.
• Select the Message Verification check box, to enable.

Message Notification
What’s this?
The VMS sets Message Wait on the extension, whenever a new message arrives in its personal Mailbox of the
extension. The VMS indicates the new message to the extension as per the Type of Message Wait Indication set
for the extension. This may be in the form of a Stuttered Dial Tone, a Voice Message, Ring or LED Lamp. See
“Message Wait” to know more.
The Message Notification feature of the VMS is an extension of the Message Wait feature. When Message Wait
Indication for an extension is set as Ring, the VMS makes a Message Notification call to the extension.
How it works
• Extension A has Message Wait Indication type set as Ring.
• Whenever there is a new message in A’s mailbox, the system will play the Message Wait Ring (Short,
Fast) on extension A. See “Distinctive Rings”.
• Extension A will ring for the duration of the Message Wait Ring Timer (configurable; default: 30 seconds).
• When Extension A answers the call within this timer, Extension A gets connected to the VMS.
• The VMS answers the call and allows access to Extension A’s mailbox.
• If Extension A does not answer the Message Notification call within the Message Wait Timer, the system
will ring on the extension again for as many times as the Message Wait Ring Count (configurable; default:
10 times), and at the interval set as the Message Wait Ring Timer Interval (configurable; default: 30
minutes).
How to configure
To provide Message Notification Call to extensions, you must configure Ring as Message Wait Indication for the
extension. For detailed instructions, refer “Message Wait Settings” in “Extension Voice Mail Settings”.
You may also configure the related Message Wait Timer, the Message Wait Ring Count and the Message Wait Ring
Interval, if required. See “System Timers and Counts” for instructions.
How to use
• Go Off-hook, when your extension rings to indicate Message Wait (short, fast ring),
• The VMS greets with the message: “This is your message notification call”.
• The VMS plays the message: “You have <n> new messages” followed by the prompt: “Enter your mailbox
password”.
• Dial your mailbox password, and follow VMS prompts.

Mailbox Settings
What’s this?
The VMS allows extension users to change the settings of the following facilities of their mailbox:
• Record the Extension Name for their mailbox.
• Redirect Messages from their mailbox.
• Delete all old messages.
• Delete all messages.
• Record Personal and Conditional Greetings for their mailbox.
• Configure the Personal (Mobile/Alternate) Number and the Assistance Number.
How to use
password".
• Dial 4 to go to Mailbox Settings.
• The VMS responds with: “For Mailbox Name press '1', For message redirection press '2', To delete all old
messages of your mailbox press '3', To delete all messages of your mailbox press '4', For Mailbox
Greetings press ‘5’, To go to previous menu press '#'.”
By default no digits are assigned for the options - Assistance Number and Personal Number, hence these
options will not be played to the caller. For details, see “Mailbox Management”.
To assign Name to your Mailbox:
• Dial 1  VMS prompts: "To record press '1'. To play press '2'. To erase press '3'. To go to previous menu,
press '#'."
• Dial 1 to record name for mailbox  VMS prompts: "Record your name after the beep and press any digit
to end."
Names for extensions can also be recorded from the System Administrator mode. See “Record Greetings/
Name”.

To delete old Messages in your Mailbox:
• Dial 3  VMS prompts: “You are about to delete all old messages of your mailbox. To proceed press 1, to
cancel press press any digit.”
• Dial 1 to delete all old messages in your mailbox  VMS responds with: “Your old messages have been
deleted”.
To delete all Messages in your Mailbox:
• Dial 4  VMS prompts: “You are about to delete all messages of your mailbox. To proceed press 1, to
cancel press press any digit.”
• Dial 1 to delete all messages in your mailbox  VMS responds with: “Your messages have been deleted”.
To programme Assistance Number for your Mailbox:
• Dial 6  VMS prompts: “To enter number press ‘1’, to play number press ‘2’, to clear number press ‘3’, to
go to previous menu press ‘#’.”
Make sure the Assistance Number is an extension number.
To programme Personal (Mobile/Alternative) Number for your Mailbox:
• Dial 7  VMS prompts: “To enter number press ‘1’, to play number press ‘2’, to clear number press ‘3’, to
go to previous menu press ‘#’.”
Any external number can be configured as the Personal (Mobile/Alternative) Number.
To redirect Messages in your Mailbox, see “Redirecting Messages”.
To record personal greetings for your mailbox, see “Recording Personal Greetings”.
To record conditional greetings for your mailbox, see “Recording Conditional Greetings”.

Listening to Messages
What’s this?
The callers/extension user leave messages in the mailbox of extension users, when they are inaccessible or the
user has forwarded his calls to the mailbox. Messages may also be received by the extensions users as
notifications for certain events. User should access their mailboxes to listen to the messages.
VMS offers two options:

• To listen to old messages.
• To listen to new messages.
Once the message is heard by the mailbox owner, VMS treats it as an old message and places it in the old
message list. The VMS also offers you the option of saving the message you have heard, as a new message.
How to use
password".
• Enter your mailbox password.
• The VMS checks the utilized mailbox memory,
• if 80% of the mailbox memory has been consumed, the VMS prompts the caller: “Your Mailbox is 80%
Full. Please Delete old messages of your mailbox.”
• if 100% of the mailbox memory has been consumed, the VMS prompts the caller: ”Your Mailbox is Full.
Please Delete old messages of your mailbox.”
press '3', to change your mailbox settings press '4’."
• Dial 1 to listen to new messages Or dial 2 to listen to old messages.
• On the completion of a message, the VMS plays the message: ‘To replay the message press '1', for Date
and time stamp press '2', to reply to message press ‘3’, to delete the message press '4', to play the next
message press '5', to forward the message press '6', to save the message as new press '7', to go to
previous menu press '#'.

Leaving a Message
What’s this?
The VMS allows,
• External callers to leave messages in the extension user’s mailbox.

• Extension users to leave message in the mailbox of other extensions.
To leave a message, the called extension must have a mailbox. The length of message recorded by the callers/
extension users must not exceed the message length set for the called extension’s mailbox. If the message
recorded by the callers/extension users exceeds the message length set for the called extension’s mailbox, the
VMS will stop recording the message after the time set and save the partially recorded message.
How to use
• The incoming call on the trunk is answered by the VMS Auto Attendant.The VMS greets the caller with the
Greeting message followed by the Welcome Message: “Welcome! Please dial the extension number or To
dial by name press ‘6’,To leave a message press ‘7’, To access your Personal Mailbox press ‘8’, For
further assistance press ‘9’, To disconnect the call press ‘#’.
The System Greeting may vary depending on the timezone — Working Hours, Break Hours, Non-working
Hours. In the above prompt the greeting is as per working hours.
• The caller dials 6 to leave message  VMS prompts: "Enter the Extension number for which you wish to
leave message."
• The caller dials the desired extension number  VMS prompts: "Record your message after the beep and
press any digit to end.”
• VMS prompts: “ Thank you for your call.”
Settings”
It is mandatory for the caller/extension user to terminate the recording by dialing # (or the digit configured).
If recording of the message is terminated simply by going on-hook, the VMS will not terminate the recording
and the call will be disconnected only after time-out, that is, the Maximum Message Length configured for
the extension.
If extension users wish to leave a message for another extension, refer “Sending Messages”.

Accessing the General Mailbox
What’s this?
A General mailbox is a shared mailbox between extension users. The General Mailbox is used for recording
messages when the mailbox of an extension is full.
How it works
The VMS offers the following options to extensions when their mailbox is full:
• Not offer the caller to record a message.
• Overwrite the existing messages in the mail box with the new message.
• Deliver the new message to the General Mailbox.
If you configure Delivery of new messages to General Mailbox on an extension, whenever the mailbox of the
extension is full, the VMS will offer the caller to record a message. This message will be recorded in the General
Mailbox.
The extension user, whose mail box is full, can listen to the new message by accessing the General Mailbox.
The extension user can access the General Mailbox, if this feature is enabled in the Class of Service of the
extension and the Password assigned to the General Mailbox (if configured) is known to the extension user. For
details see “General Mailbox Settings”.
How to configure
To offer extension users the facility of the General Mailbox when their mailbox is full, you must do the following:
• Configure the option New Message Delivery Option in Mailbox Full Condition in the Extension Voice
Mail Settings of the extension.
For the option New Message Delivery Option in Mailbox Full Condition, select Deliver to General
Mailbox.
For instructions on configuring the parameter, see “Extension Voice Mail Settings”.
• Enable the feature General Mailbox in the “Class of Service (COS)” of the extension.
• If required, assign a Password for accessing the General Mailbox, see “General Mailbox Settings”.
How to use

• Press DSS Key assigned to General Mailbox.
OR
• Dial 1176.
• Follow VMS prompts.

For SLT Users
• Lift handset.
• Dial 1176.
• Follow VMS prompts.

Forwarding Messages
What’s this?
The VMS enables extension users to forward messages of their mailbox to other mailboxes.
How it works
The Forwarding Messages feature of the VMS offers to extension users the following options:
• forward messages after adding a comment.
• forward messages without adding comment.
• forwarding messages with Message Read Receipt request.
Before forwarding a message, the VMS asks the Sender, if the Sender needs a confirmation that the message has
been read by the Recipient.
If the Sender requests for 'Message Read Receipt', the VMS stores this request. When the Recipient reads the
message, the VMS generates a file containing the first 5 seconds of the message that was sent by the Sender and
delivers it to the Sender's mailbox in the form of a new message with a prompt: "This message was read by
<Extension Name><5 seconds of the message sent>" with the Date and Time at which the message was read.
How to use
• Call the VMS by dialing 3931.
• VMS responds with: "You have <n> new messages" followed by "Enter your mailbox password".
• Enter your mailbox password  VMS responds with: "You have no/n new messages".
• The VMS prompts: "To listen to new messages press 1, to listen to old message press 2, to send a
message press 3, to change your mailbox settings press 4”.
• Dial 1 or Dial 2 and listen to the messages  VMS prompts: "To replay the message press 1' for Date and
Time stamp press 2, to reply to message press 3, to delete the message press 4, to play the next message
press 5, to forward the message press 6, to save the message as new press 7, to go to previous menu
press #."
• Dial 6  VMS prompts: "To forward the message with comment before the message press 1, to forward
the message with comment after the message press 2, to forward the message without comment press 3,
to go to previous menu press #."
• To forward message with comment before the message dial '1' or To forward message with comment after
the message dial ‘2’. The VMS prompts: "Enter the Destinations".
• Dial Destinations/Distribution list number to forward the message.
• The VMS prompts: "Record your message after the beep and press any digit to end."

• Speak to record your comment and press # to end recording.
• The VMS prompts: "To request read receipt press 1, to ignore read receipt press 2."
• To forward message without comment dial '3'. The VMS prompts: "Enter the Destinations".
• Dial Destination number/Distribution list number to forward the message.
Settings”
Extension users must be careful in dialing destination numbers. If invalid destination is entered then the
VMS will clear all the entries and will ask the mailbox owner to re-enter all the destinations again.
A message can be forwarded to maximum 10 destinations. A destination can be an extension or a a

distribution list.

Email Based Notification
What’s this?
VMS supports E-mail Based Notification, the Unified Messaging feature. This is used to inform the extension users,
the arrival of new messages in their mailbox and the memory usage status of their mailbox.
Extension users can also receive new messages as attachments to the email.
Extension users will receive the notification for the mailbox memory usage for the following:
• when 80% of their mailbox memory has been consumed.
• when 100% of their mailbox memory has been consumed.
For Email Notification to function, you must configure the “SMTP Settings” .
How to configure
To be able to use this feature, you must configure the parameters for Message Wait Notification via Email under
Message Wait Settings in Extension Voice Mail Settings.
• You can select the desired option as Notification: Without attachment , With attachment or With
attachment and mark voicemail as read.
• Specify the E-mail Address of the extension user to which the notifications are to be sent.
For the General Mailbox you only need to specify the E-mail Address to which the notifications are to be
sent.
For details see “Message Wait Notification via E-Mail” in “Extension Voice Mail Settings”.
The Notification sent for arrival of new messages in the mailbox and the memory usage status users mailbox will be
sent to the Email ID configured under Message Wait Notification via Email.
The messages sent can be customized as per your requirement. For detailed instructions, refer “VMS E-Mail
Notification”.

Message Wait Notification via Call
What’s this?
The VMS supports Notification via Call to inform the extension users about the arrival of new messages in their
mailbox.
Extension users can receive new message notification calls on a phone number of their choice. This number may
be another extension number or an external number. You can set the Type of notification calls as
• Immediate: Users will receive notifications as soon as a new message arrives in their mail box.
Or
• Scheduled: Extension users will receive notification at specified time intervals.
You can set the preferred time slots in a day during which notification calls should be made to extension users. In
addition to the time slot preference, you can also choose to receive notification calls on a Holiday.
Message Notification via Call for Department Group will not work if the destination number is an external
number.
How it works
For this feature to work, you must do the following configuration for the extension:
• Select the type of Notification call.
• Define the preferred time slots by configuring Time Zones. You can configure four different Time Zones,
defining the Start Time and End Time for each Time Zone.
• Configure the phone number to which the notification call is to be made. If the number is an external
number, configure the Trunk Access Code to be used for making the calls.
When Immediate is selected as the Type of notification,
• The system checks the preferred start and end time of the time zones configured for the extension. If the
message has arrived within the preferred time slot (Start and End Time) it immediately makes the
notification call on the number configured for the user.
If the number is an external number, the system dials out the number using the Trunk Access Code (TAC)
assigned for making notification calls.
• When the call is answered, the extension user gets connected to the VMS and can listen to the message.
• If the notification call is not answered, by default, the system makes three attempts (Message Notification
Retry Count; programmable) at an interval of 5 minutes (Message Notification Interval; programmable)
between each attempt.
• If the notification call remains unanswered after the third attempt, the system will not make any more
attempts to place this notification call. The next notification call will be made only when another new
message arrives in the mailbox of the user between the start and end time of the configured time zone.

When Schedule is selected as the Type of notification,

• The system checks the start time of the time zone(s) configured and the notification call will be made on
the number at the subsequent start time.
If the number is an external number the system dials out the number through the Trunk Access Code
(TAC) assigned for making notification calls.
• When the call is answered the extension user gets connected to the VMS and can listen to the message.
• If the notification call is not answered, the system makes three attempts (Message Notification Retry
Count; programmable) at an interval of 5 minutes (Message Notification Interval; programmable) for each
time zone. The system will continue to make attempts to place the notification call till the call is answered.
Thus, when Notification type is Immediate, notification call is made for each message that is received within the
start and end time configured in the time zone.
When Notification type is Scheduled, notification call is made for all messages received before the start time
configured in the time zone. Where multiple time zones are configured, notification call will be made at the start
time of the next time zone.
How to configure
For Message Wait Notification via Call, you need to configure:
• the parameters for Message Wait Notification via Call under Message Wait Settings in “Extension Voice
Mail Settings”.
• select the desired profile in Schedule Profile. For instructions to configure the profile parameters, see
“Configuring Notification via Call - Profile”.
• Make Message Notification call using TAC for calls to be made to external numbers. See “Configuring
VMS General Parameters”.
• if required, the Message Notification Retry Count, Message Notification Interval and Message Notification
Ring. See “System Timers and Counts”.

Dial By Name
What’s this?
The VMS Auto Attendant allows external callers and extension users to reach the desired person in an organization
by dialing the name of that person. This feature is useful when caller/extension user cannot recall the extension
number of the person they want to speak to.
How to configure
For this feature to work, each extension user’s name must be abbreviated and configured on the extension. It is
recommended that the extension users’ names be abbreviated to the first three letters of the name. As far as
possible, abbreviate names such that no two names are the same.
You must configure the Abbreviated Name in the Extension Voice Mail Settings. For detailed instructions, refer
“Extension Voice Mail Settings”.
Make sure the names for the desired extension users are recorded. For details, see “Mailbox Settings”.
How to use
• The incoming call on the trunk is answered by the VMS Auto Attendant. The VMS greets the caller with the
• The caller dial 7  VMS prompts: "Please enter first three letters of the name."
• If multiple matches are found the VMS prompts: “More than on match found. Matching Names will be
played one by one. To Select the name press ‘1’, to Skip the name press ‘2’, To Repeat the last name
press ‘3’.”
• Dial 1  VMS prompts: “To confirm press ‘1’, to Re-enter press ‘2’.”
• The caller dials 1  VMS transfers the call as per the transfer type assigned to the selected station. Talk.
• If the caller dials invalid digits, the VMS prompts: "Sorry no match found." followed by the prompt:
"Please enter first three letters of the name."
Extension users can use “Dial By Name” to reach another extension.

Dial by Extension Number
What’s this?
The VMS Auto Attendant allows external callers and extension users to reach directly the desired person in an
organization by dialing the extension number of that person.
How to use
• The incoming call on the trunk is answered by the VMS Auto Attendant. The VMS greets the caller with the
• The caller dials valid extension number  VMS transfers the call as per the Call Transfer Type selected for
the extension.
• If the extension number dialed by the caller is invalid, the VMS prompts: "The number is not valid."
followed by the prompt: "Please dial the extension number or To dial by name press ‘6’,To leave a
message press ‘7’, To access your Personal Mailbox press ‘8’, For further assistance press ‘9’, To
disconnect the call press ‘#’.
• Dial valid extension number and talk.

Call Transfer Types
What’s this?
The VMS Auto Attendant answers calls of external callers and extension users (referred to here as ‘callers’) and
transfers the call to the extensions according to the Call Transfer type set for the extension.
You must configure the Call Transfer Type in the Call Transfer Profile assigned to the extension. Different Call
Transfer Profiles can be assigned for Working, Break and Non-working hours.
The VMS Auto Attendant offers the following types of Call Transfers:
• Blind: When the caller dials the extension number, the VMS Auto Attendant transfers the call on the
extension without checking whether it is busy or free.
• Wait for Ring: When the caller dials the extension number, the VMS Auto Attendant waits for the
extension to start ringing and then transfers the call.
• Wait for Answer: When the caller dials the extension number, the VMS Auto Attendant transfers the call
when the extension answers (goes OFF-Hook).
• Screened: The VMS Auto Attendant prompts the caller to record his/her name. It puts the caller on hold
and places the call on the desired extension. If the extension is free and answers the call, the VMS
announces the caller’s name to the extension user and prompts the extension user to choose whether or
not to speak to the caller. If the extension user chooses to talk, the VMS transfers the call.
• None: When the caller dials the extension number, the VMS Auto Attendant transfers the call to the
desired extension users mailbox directly.
How to configure
Call Transfer Type must be configured in the Call Transfer Profile assigned to each extension. For instructions,
refer “Call Transfer Settings” in “Extension Voice Mail Settings” and “Call Transfer Profile”.
For calls received on trunks the VMS Auto Attendant transfer the call as per the configuration in “Auto-Attendant
Settings” in “Voice Mail Auto-Attendant Menu”.

Broadcast Message
What’s this?
Broadcasting Message allows you to send the same message to all extension users having voice mail, at the same
time. You can use Broadcast Message to make general announcements like hosting of an event, an unplanned day
off, and other such activities or events.
How to use
To Broadcast Message,
• Enter SA Mode.
• Dial 1072-301
• VMS prompts: "Record your message after the beep and press any digit to end". (Recmsg.wav)
• Speak to record your message after the beep, and press # (hash/pound) to end the message.
Settings”
The length of the message you want to broadcast must be equal to or less than the minimum of message
length programmed for the mailboxes, or your message will be truncated. For instance, if the Maximum
message length for a mailbox is configured as 15 seconds, maximum length of the message to be
broadcast must be less than or equal to 15 seconds. If the broadcast message exceeds this limit, the
system will play the first 15 seconds and truncate the remaining part of the message.

CHAPTER 16 System Maintenance
Configuration Backup/Restore
SARVAM UCS provides you the facility to Backup the configuration files from the system to the hard drive and
Restore the configuration files from the hard drive to the system at the click of a button.
Restore Configuration
• To restore configuration files from any hard drive to the system, Restore Configuration option is
provided.
• Click on the Browse button to reach the location on the local disk where the configuration files are stored
in PC. Make sure that the file is a zip file.
• After selecting the required configuration zip file from PC, click on the Restore button.
The system initiates the restoring of configuration zip file in flash storage. After successfully restoring and
validating the file, the system restarts with the new configuration.
If you select a file other than zip file, an error message is displayed when you click on the Restore button.

Backup Configuration
• To save the existing configuration files as backup, click the Backup Configuration button.
The SARVAM_UCS_SME_Config_ddmmyyyy.zip window will open; where ddmmyyyy signifies the
current date.
• You can either open the zip file or save the file to a location.
OR

Save the back up configuration files by tagging the file name with the Version-Revision of the Firmware
and tag the name of the backup folder on your computer with the date. This will help you at the time of
restoring the back up configuration files.
• Open the SARVAM_UCS_SME_Config_15032016 folder to view the configuration files.

• Keep this folder as a backup. In case there is a problem with the system configuration files these
backup files can be restored back in the system.

Firmware Management
What's this?
SARVAM UCS provides you a system to manage the upgradation of the system software with a click of a button.
The new ETERNITY LENX/MENX/GENX system with the firmware V1R3 and later cannot be downgraded
to lower firmware versions. In case you downgrade, the system will reboot and stop functioning. Consult
Matrix Technical Support Team for further assistance.
From a Personal Computer
Using Firmware Upgrade from a Personal Computer, you can manually upgrade SARVAM UCS firmware whenever
you want.
For this, you need to upload the Firmware zip file from your PC onto the SARVAM UCS system.
You can upgrade SARVAM UCS firmware manually from PC by:
• Browsing and selecting a ZIP file and then clicking on the Upgrade button.
The table below describes a few possible cases and the corresponding action taken by SARVAM UCS.
Version-Revision of Manual Selection for

Action Taken by SARVAM UCS
your SARVAM UCS Firmware Upgrade
V1R5.1.0 V1R6.1.0 SARVAM UCS will upgrade the firmware.
V1R4.1.0 SARVAM UCS will downgrade the firmware.
V1R5.1.0 SARVAM UCS will neither upgrade nor downgrade and

an error message will be displayed for the same.
V2R2.1.0 SARVAM UCS will not upgrade the Firmware and dis-
and benchmark version is play error message“Please update the firmware
V2R1.1.0 first” for least benchmark version.
V2R1.1.0
V2R5.1.0 SARVAM UCS will not upgrade the Firmware and dis-
and benchmark version play error message“Please update the firmware
sequence is V2R3.1.0 first” for least benchmark version.
V2R3.1.0 << V2R2.1.0 <<
V2R1.1.0 To upgrade the Firmware with V2R5.1.0 you must
upgrade Firmware in the following sequence:
1. V2R1.1.0
2. V2R2.1.0
3. V2R3.1.0
4. V2R5.1.0

How to Configure
• Click the Firmware Management link.
Manual Firmware Upgrade

To manually upgrade firmware of SARVAM UCS from server,
• In Upgrade Firmware from PC, click the Browse button to reach the location on the local disk where the
firmware files are stored in PC. Make sure that the file is a zip file with extension — zip, Rar or 7z.
• After selecting the required firmware zip file from PC, click on the Upgrade button.
The system initiates the uploading of firmware zip file in flash storage. After successfully storing and
validating the file, the system restarts with the upgraded firmware.
If you select a file other than zip file, an error message is displayed when you click on the Upgrade button.

Auto Update
SARVAM UCS provides you the facility to selectively update the firmware of the cards installed in the system.
The system displays the respective card firmware uploaded using “Firmware Management”.
This feature is supported in ETERNITY LENX and ETERNITY MENX only.
To upgrade Data Card Firmware, refer “DATA Card Firmware Upgrade”.
If the CO Card is being updated, you will not be able to conduct the AC Impedance Test.
To auto update the firmware of cards,
• Under Maintenance, click Auto Update.
• Slot No.: These are the number of universal slots in the system. These will differ according the variant of
SARVAM UCS you have.
• Card Name: The type of card installed in the slots. For example, SLT16, SLT8, Switch.
• Firmware: The current firmware version and revision of the card, for example V05R10.
• Firmware Available on the FTP: The firmware version and revision of the card available on the
embedded FTP server of the SARVAM UCS.
• To update the current firmware of any of the cards that appear on this page, with the firmware available on
the FTP for the card, select the Firmware Update check box.
.
For example, in slot 8 the T1E1 Dual Card is installed. The Firmware field displays the current version and
revision of the card, V05R01 and the Firmware available on the FTP field displays V05R02. If you select
the Firmware Update check box of this card, SARVAM UCS will update the firmware of the T1E1 Dual
Card to V05R02.
Matrix ETERNITY System Manual 2427

• Click Submit. The system starts the updating process.
The status of the updating process is displayed in the Firmware column. After the updating process is
completed, the Firmware Update check box is unavailable.
If you have selected the Firmware Update check box for multiple cards, SARVAM UCS will update only
two cards simultaneously.
2428 Matrix ETERNITY System Manual

Uploading Custom MoH
If the RTP Mode is set as RTP Relay or Direct RTP, the default MoH is played to the users from the CPU Card.
No special programming is required for using the default MoH. However, if you want to use the customized MoH,
you must first:
• record the message
• upload the new recorded message file in the CPU Card.

When you record MoH of your choice, consider these important points:
• The MoH file is in WAV format, so the customized MoH must also be in the same format and can be of
maximum 120 seconds.
• The MoH file has a unique name MoH.wav, make sure the audio file of the custom MoH you have recorded
must have the same unique file name as the existing default audio file.
• You can record custom MoH from any external source, and upload the audio file using the embedded FTP
server of the CPU Card.
• When you record the MoH from any external source and upload it, make sure that the audio file is recorded
in.wav file format, with the attributes listed below:
• Bit Rate: 128 kbps
• Channels: 1 (mono)
• Audio Sample Rate: 8 KHz
• Audio Format: PCM
Uploading Custom MoH

You can record a piece of music or a message of maximum 16 seconds duration and assign it to Voice Module 01,
which is reserved for Music-on-Hold.
Refer the topic “Voice Message Applications” for instructions on recording and assigning voice modules.

Default Settings
What's this?
SARVAM UCS is supplied with preset values for system and feature settings, as which may be altered and
customized by users to match their requirements and preferences. The factory-set values for system and feature
settings that are automatically assigned by the system are referred to as Default Settings or standard settings.
Every configurable parameter in the system has factory-set default values, which may be changed or customized to
match user requirements and preferences.
How it works
The default settings are to be loaded or restored in the following situations:
1. Installing the SARVAM UCS in a country other than India.
SARVAM UCS provides default settings to match country/region-specific requirements of users worldwide.
The default settings are set to match user requirements of India.
So, you must select the appropriate “Region” for the country/region in which the system is installed.
The system will load the default settings for the country/geographical region where the system is installed.
The system is designed to work efficiently with the default settings. So, if the country/region-specific
default settings match their requirements, you may not even need to alter or customize the values of
various parameters.
They may work with default settings for the most part, customizing only some of the parameters to match
their specific requirements.
The country-specific default settings of various parameters that will be loaded on changing the 'Region' are
presented in the table below. For default values of Trunk Access Codes, Configuring Emergency Number
Dialing, Distinctive Rings, for various countries refer the respective topics.
Default
Default Default Default Distinctive
Country Country DST Abbr.
Time DST CPTGa DKP Opr TAC
Code Name
Zone Mode
Schedule
Language Ringb Dialing
Type
001 Afghanistan GMT+04:30 English
002 Algeria GMT+01:00 English
003 Antigua and GMT-04:00 English
Barbuda
004 Argentina GMT-03:00 04 Spanish
005 Australia GMT+08:00 Scheduled 2 05 English 9 0
(Perth)
006 Australia GMT+09:30 Scheduled 2 05 English 9 0
(Note2)
(Adelaide)

Default
Code Name
Zone Mode
Schedule
Type
007 Australia GMT+10:00 05 English 9 0
(Brisbane,
Canberra,
Melbourne,
Sydney)
008 Austria GMT+01:00 Scheduled 1 German 9 0
009 Bahamas GMT-05:00 English 9 0
010 Bahrain GMT+03:00 Scheduled 3 English 9 0
011 Bangladesh GMT+06:00 English
012 Belarus GMT+02:00 English
013 Belgium GMT+01:00 Scheduled 2 French 0 0
014 Bhutan GMT+06:00 English
015 Bolivia GMT-04:00 Spanish
016 Bosnia and GMT+01:00 English
Herzegovina
017 Botswana GMT+02:00 English
018 Brunei GMT+08:00 English
019 Brazil GMT-02:00 06 Portuguese 0 9
(Fernando De
Noronha)
020 Brazil GMT-03:00 Scheduled 4 06 Portuguese 0 9
(Brasilia, Rio
de Janeiro,
Sao Paulo)
021 Brazil GMT-04:00 06 Portuguese 0 9
(Manaus)
022 Brazil (Acre) GMT-05:00 06 Portuguese 0 9
023 Bulgaria GMT+02:00 English
024 Cambodia GMT+07:00 English
025 Cameroon GMT+01:00 English
026 Canada (St. GMT-03:30 Scheduled 5 03 English T3 0 9 6
John's)
027 Canada GMT-04:00 Scheduled 5 03 English T3 0 9 6
(Halifax)
(Montreal,
Ottawa,
Toronto)
(Winnipeg)
(Calgary)
(Vancouver)
032 Chile GMT-04:00 Scheduled 6 Spanish 0 9
033 China GMT+08:00 08 English 0 9
034 Colombia GMT-05:00 Spanish
035 Costa Rica GMT-06:00 Spanish
036 Croatia GMT+01:00 English
037 Cuba GMT-05:00 Scheduled 18 Spanish 0 9
038 Cyprus GMT+02:00 English
039 Czech GMT+01:00 English
Republic
040 Denmark GMT+01:00 Scheduled 7 English 0 9
041 Egypt GMT+02:00 Scheduled 11 09 English 9 0
042 Fiji GMT+12:00 English
043 Finland GMT+02:00 Scheduled 8 English 9 0

Default
Code Name
Zone Mode
Schedule
Type
044 France GMT+01:00 Scheduled 2 10 French 9 0
045 Germany GMT+01:00 Scheduled 2 11 German 9 0
046 Greece GMT+02:00 Scheduled 2 12 English 9 0
047 Guyana GMT-04:00 English
048 Hong Kong GMT+08:00 English 9 0
049 Hungary GMT+02:00 Scheduled 2 English 9 0
050 India GMT+05:30 01 English T1 9 0 8
051 Indonesia GMT+07:00 14 English 0 0
052 Iran GMT+03:30 15 English 9 0
053 Iraq GMT+03:00 Scheduled 9 16 English 9 0
054 Ireland GMT Scheduled 7 English 0 9
055 Israel GMT+02:00 17 English
056 Italy GMT+01:00 Scheduled 2 18 Italian 9 0
057 Japan GMT+09:00 19 English
058 Jordan GMT+02:00 English T1 0 9
059 Kazakhstan GMT+06:00 English
060 Kenya GMT+03:00 20 English
061 Korea - North GMT+09:00 21 English
062 Korea - South GMT+09:00 English
063 Kuwait GMT+03:00 English
064 Kyrgyzstan GMT+06:00 Scheduled 10 English 9 0
065 Lebanon GMT+02:00 Scheduled 12 English 9 0
066 Libya GMT+02:00 English
067 Malaysia GMT+08:00 22 English 0 9 6
(Note1)
068 Maldives GMT+05:00 English
069 Mauritius GMT+04:00 English
070 Mexico GMT-06:00 Scheduled 3 03 Spanish T3 0 9 6
(Mexico City)
(Chihuahua)
(Tijuana)
073 Mongolia GMT+08:00 English
074 Mozambique GMT+02:00 Portuguese
075 Myanmar GMT+06:30 English
076 Namibia GMT+01:00 Scheduled 13 03 English T3 9 0 6
077 Nepal GMT+05:45 English
078 Netherlands GMT+01:00 English
079 New Zealand GMT+12:00 Scheduled 14 24 English 0 1
080 Nigeria GMT+01:00 English
081 Norway GMT+01:00 Scheduled 15 English 9 0
082 Oman GMT+04:00 English
083 Pakistan GMT+05:00 English
084 Paraguay GMT-04:00 Scheduled 16 Spanish 9 0
085 Peru GMT-05:00 Spanish
086 Philippines GMT+08:00 25 English
087 Poland GMT+01:00 Scheduled 1 26 English 9 0
088 Portugal GMT Scheduled 7 27 Portuguese 9 0
089 Qatar GMT+03:00 English
090 Romania GMT+02:00 English
091 Russia GMT+03:00 Scheduled 1 28 English 9 0
(Moscow, St.
Petersburg)

Default
Code Name
Zone Mode
Schedule
Type
(Novosibirsk)
(Vladivostok)
094 Singapore GMT+08:00 30 English 9 0
095 Slovakia GMT+01:00 English
096 South Africa GMT+02:00 31 English
097 Spain GMT+01:00 Scheduled 1 32 Spanish 9 0
098 Sri Lanka GMT+05:30 English
099 Sudan GMT+03:00 English
100 Sweden GMT+01:00 Scheduled 2 English 9 0
101 Switzerland GMT+01:00 Scheduled 2 German 9 0
102 Syria GMT+02:00 Scheduled 17 English 9 0
103 Taiwan GMT+08:00 English 0 0
104 Tajikistan GMT+05:00 English
105 Thailand GMT+07:00 33 English
106 Turkey GMT+02:00 34 English 9 0
107 Uganda GMT+03:00 English
108 Ukraine GMT+02:00 English
109 United Arab GMT+04:00 35 English 9 0
Emirates
110 United GMT Scheduled 7 02 English T2 0 9 8
Kingdom
111 United States GMT-05:00 Scheduled 3 03 English T3 0 9 6
(Atlanta,
Augusta,
Boston,
Charlotte,
Columbus,
Detroit,
Indiapolis,
Miami, NY,
Philadelphia,
Washington)
(Chicago,
Dallas, Des
Moines,
Memphis,
Minneapolis,
New Orleans,
Oklahoma,
Omaha, St.
Louis)
(Albuquerque,
Boise,
Cheyenne,
Denver, Salt
Lake City)
(Las Vegas,
Los Angeles,
Phoenix, San
Francisco,
Seattle)
(Juneau)
116 United States GMT-10:00 03 English T3 0 9 6
(Hawaii)

Default
Code Name
Zone Mode
Schedule
Type
117 Uzbekistan GMT+05:00 English
118 Venezuela GMT-04:30 Spanish
119 Vietnam GMT+07:00 English
120 Yemen GMT+03:00 English
121 Yugoslavia GMT+02:00 English
122 Zambia GMT+02:00 English
123 Zimbabwe GMT+02:00 English
124 Saudi Arabia GMT +3.00 English T1 9 0
125 Cote d'Ivoire GMT+01:00 Scheduled 2 10 French 9 0
a. See “Call Progress Tones” for more information on CPTG.

b. See “Distinctive Rings” for more information on Ring Type and Cadence.
2. Installing the SARVAM UCS in a Hospitality Application.
The two main applications of the SARVAM UCS are:
• Enterprise application to meet the communication requirements of businesses.

• Hospitality application to meet the specific requirements of Hotels and Hospitals.
In addition to the common set of System features, there is a distinct set of in-built features for each of
these applications. When SARVAM UCS is to be installed in any of the two application scenarios, the
'Customer Profile' - whether the user is an Enterprise or a Hotel - is to be defined at the time of installation.
When the Customer Profile is defined, all features specific to the application Enterprise/Hotel, along with
their default settings are loaded. By default the Customer Profile of SARVAM UCS is defined as
'Enterprise'.
3. Malfunctioning of the System.
When there is a system malfunction, possibly caused by a programming error that you are unable to
diagnose, you may restore default settings.
Restoring Default Settings

Default settings can be loaded or restored from the programming mode (software default).
To be able to do this, you must have the Programming Password, also referred to as the "SE password".
Whenever you restore the default settings in the system, all the programmable parameters except Network
Port Parameters295 and the “Region” will be set back to their default values.
295. The IP Address, Subnet Mask, Primary DNS, Secondary DNS, Host Name, Domain Name, DHCP Server Address.

All parameters will be assigned default values except the following, when the system is set to default:
• SE Password
• SA Password
• Date Format
• CPTG Type
• Companding Algorithm - A-Law / u-Law
• Voice Module
• Clock Synchronization Parameters
• Source Port
• Clock Type
• PLL TIE Control
• PLL Operating Mode
• Locking Mode
• SMDR
• SIM PIN of Mobile
• Network Parameters
• Wizard Parameters
Restoring Default Settings using Jeeves

• Under Maintenance, click Default the System.
• You will be prompted whether you want to default the system.
• Click OK.

• Enter Reverse SE Password on the prompt.
• The SE password you enter must be the current password. For Example: if it is 1234, enter 4321 and click
OK.
• The system will restart and load the default settings. It takes 50 seconds to load default settings. The
system count down in the seconds will appear on your screen as “Loading Default Settings...Please Wait
for ____ seconds.
• You are returned to the Login page of Jeeves.
• You may now enter programming mode again.
Restoring Default Settings using a Telephone

• Dial command 5302-Reverse SE Password.
For Example: if your current SE password is 1234, you must dial 5302-4321 to restore default settings.
• The names programmed for the DKP, SLT, SIP Extensions will disappear and the default flexible numbers,
that is, extension numbers assigned to the DKP, SLT, SIP Extension stations will appear.
Without the SE Password, you cannot restore default values via the software. If you forget the SE
Password, you must resort to hardware default of the SE-Programming Password first. Refer the topic
“System Security” for instructions on restoring the default SE Password.

PCAP Trace
What's this?
PCAP or packet capture consists of intercepting and logging the traffic passing over a digital network or a part of a
network. PCAP intercepts each packet in the data streams that flow across the network, and can decode and
analyze its contents.
PCAP can be used, among others, to monitor the network, detect and analyze network problems, debug client/
server communications, debug network protocol implementations.
SARVAM UCS supports PCAP Trace for the LAN/WAN Port of the system and the Matrix Extended IP Phones.
PCAP Trace is supported for both IPv4 and IPv6 addresses.
The PCAP Trace of the Extended IP Phones can be accessed using the FTP of the phone. The access to the FTP
is secured by a password and it can be changed. For detailed instructions, see “FTP Access for Extended IP
Phones”.
Packets traveling over a network are captured and saved in the system. You can save these trace files (packets
captured by the system) on a PC and open these trace files using a graphical packet capture and protocol analysis
tool such as Wireshark or Ethereal.
SARVAM UCS also supports Filters and 'Promiscuous' mode for capturing packets, which you can use to specify
the types of data packets to be captured.
How to use
Using PCAP Trace of SARVAM UCS

PCAP Trace for the LAN/WAN Port allows a maximum of 10 MB of packets can be captured and stored in the
SARVAM UCS.
To use PCAP Trace of SARVAM UCS,

• Under Maintenance, click PCAP Trace link to open the page.
• Decide the type of packets to be captured and set the Filter accordingly. The Filter Settings parameter
should be maximum 60 characters in length; all ASCII characters are allowed. By default, this field is
blank. So all packets will be captured.
Refer the following examples to know how to set the Filters.
Examples of Filter settings:
• To capture only SNMP traces:

• Filter Settings = port 161
Where, 161 is the SNMP Port number for which the traces are to be captured.
• To capture packets which are transmitted from the system, from IP address 192.168.1.191:
• Filter Settings = src 192.168.1.191
• To capture packets which are received for the system, to IP address 92.168.1.191:
• Filter Settings = dst 192.168.1.191
• To capture only packets which are transmitted from the system and received to the system, IP address
192.168.1.191:
• Filter Settings = src 192.168.1.191 or dst 192.168.1.191
• To capture packets which are transmitted from the system for particular port number only, from IP
address 192.168.1.191 and port number 161
• Filter Settings = src 192.168.1.191 and port 161
If you do not enter a valid filter, you will get the message: 'Invalid filter! Please enter valid filter'.

It is not mandatory to set Filters. When the Filter Settings field is left blank, the system will capture all
packets.
• You can set the Enable Promiscuous Mode? to Yes, if you want.
When you enable Promiscuous mode, the SARVAM UCS will capture all network traffic. However, this will
work only in a non-switched environment.
When Promiscuous Mode flag is disabled, the system will capture only traffic that is directly related to it.
Only traffic to, from or routed through the SARVAM UCS will be picked up by the PCAP Trace.
'Filter Settings' and 'Promiscuous Mode' (enabled) will not be cleared during power down.
• Click the Start button to begin the capturing of the packets.
• Click the Stop button to stop packet capture.
OR
• Wait for the system to stop packet capturing. The system stops packet capturing once the maximum
allotted memory of 10 MB (RAM) is utilized.
• Number of Packets and bytes captured as per the filter setting will be displayed as Packets Captured and
Total Bytes respectively.
Capturing of packets will not stop if you open any other page of Jeeves. So, you may continue using
Jeeves for any other purpose while PCAP Trace is being used.
• When the packet capturing is stopped (by you or the system), click the Save Trace File button to save the
files on the PC.
The current packets captured will not be deleted after you have saved the trace file. The current packets
will be deleted when you start the PCAP capture again.
• Now, you can open the downloaded trace file using Wireshark or Ethereal or any other similar software
which supports opening of trace files.
Using PCAP Trace for Matrix SPARSH VP248 Extended IP Phone

PCAP Trace supported by the Matrix Extended IP Phone can capture up to 1 MB of packets and store them in the
embedded FTP server of the phone. The access to the FTP is secured by a password, for details see “FTP Access
for Extended IP Phones”.
You are not required to set any filters; the phone captures all packets that it sends and receives.
To be able to use PCAP Trace for a Matrix Extended IP Phone, there must be a DSS Key for accessing the Local
Menu on the phone. If you have not already configured the DSS Key for Local Menu, you may do so now. For
instructions on configuring DSS keys of the Extended IP Phone, see “Configuring Matrix SPARSH VP248”.
To use PCAP Trace from an Extended IP Phone extension,
• Press the DSS Key assigned to ‘Local Menu’.

You can access the Local Menu only when your phone is in idle state.
M on 10 M AY 1 5: 40
Local Menu
1 2 abc 3 def
4 ghi 5 jkl 6 mno
7 pqrs 8 tuv 9 wxyz

CA04
0 #
CA03 *
CA02
CA01
• The Local Menu appears on your phone display.
LOCAL ME NU
N et wo r k P a r a m e t er s
N et wo r k S t a t u s
• The cursor appears under ‘Network Parameters’.
• Press Enter key to select ‘Network Parameters’.
M A C :00 :1 b:09 :00 :9a :a 7
I P A d d r e ss
Sub net Ma sk
G at e w a y A d d r e s s

• Scroll with the DOWN navigation key to PCAP.
P P P o E S e rv i c e N a m e
S e r v er A d d re ss
S e r v er P o r t
VLAN S etti ng
PCAP
• ‘Start PCAP’ appears on your phone display.
PCAP T RACE
Sta rt PCAP
• Press Enter key. You get a confirmation message ‘Started!’
PCAP TRACE
Sta rted !
To stop PCAP,
• Enter the Local Menu of the phone again by pressing the DSS key.
• Enter the Network Parameters submenu.
• Select PCAP and press Enter.
• The message Stop PCAP appears on your phone display.
PCAP T RACE
Stop PCAP

• Press Enter key. You get the confirmation message: ‘Stopped!’
PCAP T RACE
Sto ppe d!
• Go idle.
Using PCAP Trace for Matrix SPARSH VP310 Matrix Extended IP Phone
PCAP Trace supported by the Matrix Extended IP Phone can capture up to 1 MB of packets and store them in the
embedded FTP server of the phone.
You are not required to set any filters; the phone captures all packets that it sends and receives.
To use PCAP Trace,
• When the phone is in idle state, press the DOWN key to access the Network Settings.
• Scroll with the DOWN navigation key to PCAP.

• ‘Start PCAP’ appears on your phone display.
• Press Enter key. You get a confirmation message ‘Started!’
To stop PCAP,
• Press the DOWN key to access the Network Settings.
• Scroll with the DOWN navigation key to PCAP and press Enter key.
• The message Stop PCAP appears on your phone display.
• Press Enter key. You get the confirmation message: ‘Stopped!’
• Go idle.

You can download the Trace file from the embedded FTP server of the Extended IP Phone. To access the FTP
server using Windows FTP, do the following:
• Go to My Computer.
• Type the current IP Address of the Extended IP Phone in the Address bar. For example: ftp://
192.168.201.134
• Click Go or Press Enter key on your keyboard.
• The Log on as window of the FTP server opens.
• In User Name, type se (lower case).
• In Password, enter the FTP Password for the phone.
• Click Log on.
• On successful login, the FTP window will open. You will see the different Configuration folders in this
window.
• Click the folder ramdisk.
• In the ramdisk folder, right click the file trace.pcap and copy it on to your local disk.
• Open the trace.pcap file using Wireshark or Ethereal or any other similar software which supports
opening of trace files.
You may also use FireFTP, if you are using Mozilla Fire Fox. Make sure your browser has the FireFTP
Add-on installed.
To download the Trace file from the FTP server for SPARSH VP330 and SPARSH VP510, refer to their
respective User Guides.

Network Diagnosis
What's this?
SARVAM UCS provides you an option to check the Internet/WAN connectivity using Ping and Traceroute as the
diagnostic tools.
How to Configure
• Click the Network Diagnosis link.
• In Diagnostic Utility, select the diagnostic tool — Ping or Traceroute — to check the Internet/WAN
connectivity.
• In IP Address/Domain Name, enter the IPV4 or IPV6 Address or the Domain Name of the system whose
connectivity you wish to test. Default: Blank
If you have selected Ping as the Diagnostic Utility option, configure the following parameters:
• In Ping Packet Size, enter the number of bytes you want the system to send for Ping test. Valid
Range: 4 to 1024. Default: 32 bytes.

• In Ping Count, enter the number of times you want system to send the request message for Ping test.
Valid Range: 1 to 50. Default: 4 times.
• In Ping Timeout (sec), enter the time for which you want the system to wait to get the response for
each request message sent. Valid Range: 1 to 9. Default: 3 sec.
If you have selected Traceroute as the Diagnostic Utility option, configure the following parameters:
• In Traceroute Max TTL, enter the maximum number of hops (Time-To-Live value) you want the
system to take in the path to find the IP Address configured. Valid Range: 1 to 255. Default: 30.
• In Traceroute Protocol, select the protocol — ICMP or UDP — which you want the system to use for
traceroute functionality.
• To start the Network Diagnosis, click Start button.
The Diagnostic result will appear on the screen.
• To clear the Diagnostic result, click Clear button.

Network Drive Settings
The Network Drive Settings allows you to test the connectivity between the system and the Network Drive where
you wish to save the mailbox backup files. The authenticity to access the shared folder is also verified by the
Network drive for security purpose.
Backup on Network Drive provides you the facility to retrieve valuable information in case internal USB is full.
For information regarding the voicemail backup, see “Voicemail Backup”.
Make sure the System and the Network Drive are in the same Local Network.
How to Configure
• Under Maintenance, click Network Drive.
• Select the Network Drive check box to enable the Network Drive.
Clear the check box to disable the Network Drive.
By default, it is disabled.
• In IP Address, enter the IP Address of the Network Drive where you wish to store the mailbox backup
files.
• Select the Authentication Required check box to enable the authentication process. Enable this
check box if the Network Drive folder requires authentication for accessibility.
• In User Name, enter the authentication ID set for the Shared Folder of the Network Drive.
• In Password, enter the authentication password set for the Shared Folder of the Network Drive.
• In Shared Folder Name, enter the name of the Shared Folder where you wish to store the mailbox
backup files.
• Click Submit.

You may now test the connection between the system and the Network Drive. To do so,
• Click Test.
A message will be displayed showing whether the connection is successful or not. In case of an error, the
system will display the error message.

Redundancy Debug
Redundancy Debug is applicable only for ETERNITY LENX/MENX system.
You can monitor the state of — Active and Stand by Card information, Feature Information, Call Information, the
states of the various software ports, Department Group etc. — being transfered from the Active Card to the Stand
by Card using Redundancy Debug.
SARVAM UCS supports Syslog Client for debugging. Debug messages are sent to the remote Syslog Server.
Configuring Redundancy Debug using Jeeves

• Under Maintenance, click Redundancy Debug.
• Select the respective check box of the events/processes you want to debug:
• Active Card Information
• Stand By Card Information
• Configuration File Transfer
• Communication Process
• Communication Protocol
• Call Information
• Card Information
• Feature Information
• Phone Information
• RTC Information
• System Information
• SLT Port

• DKP Port
• CO Port
• BRI Port
• DS1 Port
• E&M Port
• VMS Port
• DISA Port
• Mobile Port
• VoIP Port
• SIP Port
• ISDN Terminal Port
• Magneto Port
• QSIG Port
• Room Port
• Guest Port
• SIP Extension
• Virtual Station
• Q-SIG Extension Port
• Radio Interface Port
• Department Group
• Click Submit.

Redundancy Process
Redundancy Process is applicable only for ETERNITY LENX/MENX system.
Viewing Redundancy Process Status

You can view the status of the Redundancy Process in Jeeves. To do so,
• Under Status, click Redundancy Process.
• Redundant Status displays the status of the CPU Card — Active or Stand by.
If the CPU Card is Active, the following details of the same will be displayed:
• The Connection Status will display the connectivity status of the CPU Card — Connected or Not
Connected.
• Transmit Queue Status displays the status of the data that is pending to be transmitted. The
Redundancy is executed properly if there is no pending data left to be transmitted.
• Received Queue Status displays the status of the received data that is still remaining to be read and
executed. The Redundancy is executed properly if all the data is received.
• CPU Switch over Timer displays the time left (in seconds) for the CPU Card to completely switch from
the Stand by mode to the Active mode and vice-versa. The time left will be calculated as per the time
you configured for the Set Active CPU Card to Stand-by after (Hrs.) in the “System Parameters”.
• Active Card Status displays the status of the data (Configuration and Call Information) being
transfered from the Active Card to the Stand by Card.

Make sure that both the active and stand-by CPU Cards have the same firmware installed. Only then all
the details on this page will be displayed.

Restart the Card
Applicable only for ETERNITY LENX and MENX.
You can restart any of the cards installed in the system.
To do this,
• Under Maintenance, click Restart the Card to open the page.
• The page displays the Slot Numbers and the card installed in the slot.
• To restart a card, select the Restart check box of the respective card.
• Click Submit.
• The system will restart the card.

Restart the System
To restart the System,
• Under Maintenance, click Restart the System to open the page.
• You will get the message, “This will restart the system. Do you want to continue?”.
• Click OK.
• Enter Reverse SE Password on the prompt.
• The SE password you enter must be the current password. For Example: if it is Matrix@1234, enter
4321@xirtaM and click OK.
• The system will restart.

SIP Extension
What's this?
SARVAM UCS provides you an option to synchronize configuration changes of all SIP Extensions and reconfigure
all SIP Extensions with all configuration files with a click of a button.
How to Configure
• Click the SIP Extension link.
• In Synchronize configuration changes of all SIP Extensions, click the Sync Now button to
synchronize the configuration of Extended Clients, VP110 and Third Party IP-Phones.
Only those configuration files in which there are changes will be synchronized.
• In Reconfigure all SIP Extensions with all configuration files, click the Reconfigure Now button to
reconfigure all the configuration files of Extended Clients, VP110 and Third Party IP-Phones.

System Debug
You can monitor the state of software ports and IO operations for trouble shooting and identifying faults and errors
using System Debug. SARVAM UCS supports Syslog Client for debugging. Debug messages are sent to the
remote Syslog Server.
Configuring System Debug using Jeeves

• Under Maintenance, click System Debug.
Debug Settings
• In Debug port, select the destination port for sending debug messages as either COM Port, Ethernet
Port or USB to COM Port.
If you are using the Syslog Server for debugging, select Ethernet, and configure Syslog Server IP
Address and Port. Both IPV4 and IPv6 addresses are supported. Valid port range is: 1025 to 65535;514.
• Now, select the respective check box of the events/processes you want to debug:
• Port State
• Card IO
• PMS
• Parameter Initialization on Power-ON
• Error
• Communication Manager
• File Open/Read/Write
• VMS
• ACB
• Maturity/AOC

• Auto-upgradation / RCOC
• Web Jeeves
• T1E1 Alarm Status
• SMS Server Main Application
ARM Debug
• In Syslog Server IP Address, enter the server address where the debug log is to be sent and in Port
enter the server port number where the debug log is to be sent. Both IPV4 and IPv6 addresses are
supported. Default port number is 514. The valid range of this port is 1025 to 65535; 514.
• Enter the Debug levels in the Debug Parameter 1, 2 and 3 fields.
DSP Debug
• In Syslog Server IP Address, enter the server address where the debug log is to be sent and in Port
enter the server port number where the debug log is to be sent. Both IPV4 and IPv6 addresses are
supported. Default port number is 514. The valid range of this port is 1025 to 65535; 514.
• Enter the Debug levels in the Debug Parameter 1, 2 and 3 fields.

SMS Server Debug
• You can enable the Debug for the SMS Server Parameters. Select the check box of the respective
parameter you want to debug:
• SMTP Client
• POP3 Client
• SMS Record
• SMS Sender
• SMS Receiver
• For the SMS Server debug, configure the Syslog Server IP Address and Port. Both IPV4 and IPv6
addresses are supported. Valid port range is: 1025 to 65535;514.
Port Debug
• You can also debug a particular Port, by configuring the Port Debug settings.
• You can also debug a card installed in a particular slot, by configuring the Slot Debug settings.
• Click Submit.

Configuring System Debug using Telephone
• Enter SE mode from an extension phone.
To program the IP Address of the Syslog Server (the computer on which the Syslog Server is running), dial:
• 2178-Syslog Server IP Address
Default: 192.168.1.104.
To program the Port of the Syslog Server, dial:

• 2179-Syslog Server's Listening Port
Listening port can be from 1025 - 65535;514.
Default: 514
To start/stop debug for required process, dial:

• 2104-Value
Where, Value is a 4-digits string for each process type, as in this table:
Process Type Value
Port State 0001
Card IO 0002
PMS 0004
Parameter Initialization on Power-ON 0008
Error 0016
Communication Manager 0032
File Open/Read/Write 0064
VMS 0128
ACB 0256
Maturity/AOC 0512
Auto-upgradation/RCOC 1024
• If you need to enable a combination of the above process types, use the sum of their values in the
command.
• This command will be saved in the configuration and not changed even in the case of power failure.
Example:
• To enable debug for 'VMS' and 'PMS' use values = 0004 + 0128 = 0132 and give command 2104 - 0132.
• To disable debug for all process use command: 2104-0000.
To start/stop state debug, dial:

• 2105-Port Type-Start Port Number-End Port Number-Flag

Where,
Port Type Meaning
01 SLT
02 DKP
03 CO
04 BRI
05 T1E1
06 E&M
25 MOBILE
26 SIP TRUNK
29 MAGNETO
Flag is 0 for Enable, 1 for Disable
This command will not be saved in the configuration. If the system restarts, you need to dial this command
again.
To enable SARVAM UCS HOST Debug, dial:

• 2181-1-Code
Where,
Code Meaning
001 Enables HOST Debug on Ethernet
002 Enables SARVAM UCS Master Cmd/Event Debug
004 Enable ISDN Relate Host to DSP Cmd/Event Debug
007 Enable all three (above given) debugs
000 Enables DSP Serial Real Time Debug
Default is 000.
To program IP Address for ARM Debug, dial:

• 2182-IP Address
Default: Blank.
To program ARM Syslog Port, dial:

• 2183- Port
Where,
ARM Syslog Port range is from 514, 1025 to 65535.
Default: 514.
To enable SARVAM UCS DSP Para 1 Debug, dial:

• 2184-1-Code
Where,

Code Meaning
000 Enables DSP Serial Real Time Debug
001 Enables DSP Serial Slow Debug
002 Enables DSP Ethernet Debug
Default is 000.
It is not possible to enable all three debugs at a time, only one of the three can be enabled at a time.
To enable SARVAM UCS DSP Para 2 Debug, dial:

• 2184-2-Code
Where,
Code Meaning
001 Enable SLT Debug
002 Enable CO Debug
004 Enable DKP Debug
08 Enable ENM Debug
016 Enable ISDN Debug
032 Enable GSM Debug
064 Enable VOIP Debug
128 Enable VMS Debug
255 Enable debug for all the ports listed above
000 Disable debug for all the ports listed above
Default is 000.
To program IP Address for DSP Debug, dial:

• 2185-IP Address
Default: Blank.
To program IP Port for DSP Debug, dial:

• 2186- Port
Where,
IP Port range for DSP Debug is from 514, 1025 to 65535.
Default: 514.
To enable SARVAM UCS PCM Capture Debug, dial:

• 2172-Slot Number-Hardware Port Offset296- Code
Where,
Code is
296. For Slot Assignment in ETERNITY LENX, refer “Software Port and Hardware ID”.

Slot Number is 01 to 24
0 for Rx and Tx PCM Capture Disable
1 for Rx PCM Capture Enable
2 for Tx PCM Capture Enable
3 for Rx and Tx PCM Capture Enable
To initiate the debug of IO operations, dial:

• 2199-Slot Number-1-Port Number297-Code
• 2199-Slot Number-2-Port Number-Port Number-Code
• 2199-Slot Number-*-Code
Where,
Slot Number is 01 to 24
Port Number is the Port offset on the card in 2 digit
Code is 1 for Enable, 0 for Disable
If the Slot No. and the Port Number are programmed as 99 the debug of all slots and ports is generated.
• Exit SE mode.
You are advised to backup System Configuration before you upgrade Firmware.
297. For Slot Assignment in ETERNITY LENX, refer “Software Port and Hardware ID”.

VoIP Debug
Matrix SARVAM UCS supports a feature by which SE can view the VoIP debugs on the server over IP network.
This is done by 'Syslog Client' in the SARVAM UCS which supports multiple debug levels.
Configuring VoIP Debug using Jeeves

• Under Maintenance, click VoIP Debug.
• Configure the following:
• Select the Debug check box to enable.
• Configure Syslog Server IP Address and Port.

Both IPv4 and IPv6 addresses are supported. By default, the Port is 514. Valid range of the port is from
1025 to 65535 and 514.
• Now, select the respective Debug Level check box to enable:

• System
• Serial
• SIP
• Call
• Reg.User (Registered User)
• Reg. Trunk (Registered Trunk)
• BLF/MWI
• Media
• VoPP

• Call Adv. (Advance)
• Reg. User Adv. (Registered User Advance)
• Reg. Trunk Adv. (Registered Trunk Advance)
• BLF/MWI Adv. (BLF/MWI Advance)
• Media Adv. (Media Advance)
• Presence
• IM
• RTCP
As per the level selected, debug log will be generated. For example: if debug log of Call is required,
enable 'CALL' level and disable all other debug levels.
• Click Submit.
We recommend you to consult the Matrix support team before you enable Advance debugs. These debugs
lead to debug data streaming in huge quantities which may affect the system performance.

System Details
The System Details displays the details of the system along with the slot details of the cards which are installed in
the system with the Slot Number they are installed in. It also displays their respective Firmware Version, CPLD
Version and Kernel Date.
To view the System Details,
• Under Status, click System Details.
If you have installed ETERNITY LENX, this page will also display the details of the Power Supply card.
See “The Power Supply Card” for details.
Switch Application
If you want to run some other application on the ETERNITY GENX platform, click on the Switch Application
button.
By clicking on the Switch Application, you will be diverted to the ETERNITY GENX Login page.
You must login with the same password you kept for the Application.
Select the Application Selection link and then the desired application you wish to run on the ETERNITY GENX
Platform. Click on Next button. See “Application Selection”.
The system will backup the existing configuration you did on this application. If you wish to take the backup of
configuration in your PC, click on the Download Configuration link.

• Configuration Backup/Restore page opens.
You can restore or backup configuration. For instructions, see “Configuration Backup/Restore”.

System Usage
To view the active channels and their activities,
• Under Status, click System Usage.
The page displays the status of the calls along with the ports and trunks involved.

Appendix
Technical Specifications - ETERNITY LENX

System Resource Specifications
Processor Type 32 bit RISC processor
Processor speed 900 MHz dual core
Universal Slots 27
Number of USB Ports 1 Internal USB 2.0

1 External USB 3.0 (for future use)
USB Storage Capacity Internal USB = 64 GB
Ethernet Ports 1 Gbps Eth for WAN

1 Gbps Eth for LAN
Vocoder Modules for VoIPa 64 Channels supported by each module
VMS Module for VMSb 64 Channels
Mounting Options 1. 19” Rack Mountable

2. Table Top
3. Wall Mountable
Dimensions (LxBxH) 715 X 474.25 X 449 mm (W/o caster wheel, Side clamp
& Wall mounting clamp)
749 X 474.25 X 449 mm (With caster wheel & w/o Side
clamp & Wall mounting clamp)
749 X 495.5 X 482.50 mm (With caster wheel, Side
clamp & Wall mounting clamp)
Shipping Weight (Kg) 52 kg approx. (with packing)
Shipping Material Type Corrugated Box
Power Supply Options DC PS
Power Supply Input Range DC PS : 48 VDC (+20% & -15%)
Power Consumptionc For DC PS :688.19 W
Operating Temperature 0o C to 45o C
Operating Humidity 5 - 95 RH, Non-Condensing
Storage Temperature - 20 Degree to +70 Degree
Storage Humidity 0 - 95% RH, Non-Condensing
Matrix SARVAM IP-PBX System Manual

a. Max number of VoIP modules that can be placed on the board are 4.
b. Max number of VMS module that can be placed on the board is 1.
c. Considering 30% SLT off-hook when 1296 SLTs are connected.
Technical Specifications - ETERNITY MENX

Universal Slots 16


1 Gbps Eth for LAN

2. Table Top
3. Wall Mountable
Dimensions (WxHxD) 612 x 405.5 x 325.55 mm
Shipping Weight (Kg) 18 Kg approx. (with packing)
Power Supply Options AC PS

DC PS
Power Supply Input Range AC PS : 100 - 240 VAC

DC PS : 48 VDC (+20% & -15%)
Power Consumptionc For AC PS : 402.17 W

For DC PS : 325.90 W

Technical Specifications - ETERNITY GENX
Universal Slots 12


1 Gbps Eth for LAN
Communication Ports 1

2. Table Top
3. Wall Mountable
Dimensions (WxHxD) 436.5 x 226 x 173.5 mm
Unit Weight (Kg) 5.00 Kg
Shipping Weight (Kg) 8.20 Kg
Power Supply Options AC PS

DC PS
Power Supply Input Range AC PS : 100 - 240 VAC

DC PS : 48 VDC (+20% & -15%)
Power Consumptionc For AC PS : 227 W

For DC PS :183 W

Technical Specifications - SARVAM UCS
ETERNITY LENX
SIP Extensions 2000 Max
SIP Trunks 99 Max
DKP Ports 128 Max
SLT Ports 1296 Max
T1E1 Ports 24 Max
BRI Ports 32 Max
GSM Ports 64 Max
CO Ports 128 Max
E&M Ports 64 Max
Radio Ports 16 Max
Magneto Ports 16 Max
Vocoder Channelsa 248 Max
Voice Mail Channelsb 64 Max
IP to IP audio Calls with DRTP/RTP Relay 550 Max
IP to IP audio Calls with Vocoder channels 64 Maxc
IP to non-IP audio Calls 128 Maxd
IP to IP Video Calls 55 Max
Maximum 3-Party Conference 15
Maximum Parties in a Single Multi-Party 21

Conference
Maximum Multi-Party Conference 2 x 21 Parties + 1 x 3 Parties
Maximum Call Taping 20
Number of Voice Modulese 16
Number of voice modules that can be simultaneously 11

played
a. Four optional Vocoder Modules (NX DBM VOCODER64) can be installed.

b. Optional VMS Module (NX DBM VMS64) required.
c. When two NX DBM VOCODER64 Modules are installed.
d. When two NX DBM VOCODER64 Modules are installed.
e. 16 seconds each.

Supported Cards
Cards Remarks
ETERNITY LE Card SLT48
ETERNITY ME CARD DKP16
ETERNITY LE Card ILC48
ETERNITY ME DKP32
ETERNITY ME DKP16
ETERNITY ME DKP8
ETERNITY ME Card BRI8
ETERNITY ME Card T1E1PRI Dual
ETERNITY ME Card T1E1PRI Single
ETERNITY ME Card E1FO Dual
ETERNITY ME Card E1FO Single
ETERNITY ME Card E&M8
ETERNITY ME Card GSM8 2G/3G
ETERNITY ME Card Magneto8
ETERNITY MENX
SIP Trunks 99 Max
DKP Ports 128 Max

SLT Ports 512 Max
T1E1 Ports 8 Max
BRI Ports 32 Max
GSM Ports 64 Max
CO Ports 128 Max
E&M Ports 64 Max
Radio Ports 16 Max

Conference

played
a. Four optional Vocoder Modules (NX DBM VOCODER64) can be installed.

e. 16 seconds each.
Supported Cards
Cards Remarks
ETERNITY ME CARD DKP16
ETERNITY ME DKP32

Cards Remarks
ETERNITY ME DKP16
ETERNITY ME DKP8
ETERNITY ME Card E1FO Dual
ETERNITY ME Card E1FO Single
ETERNITY ME CARD CDMA4
ETERNITY ME Card Magneto8
ETERNITY GENX
SIP Trunks 99 Max
DKP Ports 96 Max
SLT Ports 240 Max
T1E1 Ports 8 Max
BRI Ports 32 Max
GSM Ports 40 Max
CO Ports 64 Max
E&M Ports 32 Max
Radio Ports 16 Max


Conference

played
a. Two optional Vocoder Modules (NX DBM VOCODER64) can be installed.

e. 16 seconds each.
Supported Cards
Cards Remarks
ETERNITY GE CARD CO4+DKP2+SLT12 With inbuilt PFT
ETERNITY GE CARD CO4+DKP2+SLT8 With inbuilt PFT
ETERNITY GE CARD GSM4 2G/3G
ETERNITY GE CARD CDMA4
ETERNITY GE CARD T1E1PRI SINGLE

Cards Remarks
ETERNITY GE CARD E1FOPRI SINGLE
ETERNITY GE CARD RIC4
ETERNITY GE CARD MAGNETO4
ETERNITY GE CARD E&M4
ETERNITY GE CARD BRI4
Supported Terminals
UC Clients
VARTA ADR100 Unified Communication Client for Android
VARTA AMP100 Unified Communication Client for iOS
VARTA WIN200 Unified Communication Client for Windows
Desk Phones
EON48 The Digital Key Phone
EON310 Executive Digital Key Phone
EON510 Premium Digital Key Phone
SPARSH VP248 The High-Definition Edge to your IP Communication
SPARSH VP310 The Executive IP Phone
SPARSH VP330 Intuitive Touch-Screen IP Phone
SPARSH VP510 Premium IP Phone
SPARSH VP110 The Business IP Phone
Analog SLT ports supported for Short Loop with Loop Current
programmed
Loop Current Programmed 25mA 30mA 35mA 40mA
Number of ports supported in talk mode (OFF-Hook 200 175 150 128
short loop) according to loop current programmed
SLT (Analog Station)

Signaling Loop Start
Dialing DTMF and Pulse (10/20PPS)
Off Hook AC Impedance 600/900/Complex
Off Hook Current 39mA max

Loop Limit 1800 max (excluding Telephone)
On-Hook Voltage (Tip/Ring) -48V nominal
DTMF Detection ITU-T Q.24
Return Loss >18dB
Longitudinal Balance >50dB
Transmission Level Adjust Tx Gain: -3dB to +6dB, Rx Gain: -3dB to 6dB
Ringing Trapezoidal 60VRMS/25Hz and Sinusoidal 52VRMS/25Hz
REN 3
CLI Reception DTMF, FSK ITU-T V.23 and FSK Bellcore 202
Protection Over Voltage Secondary Protection
Physical Connector RJ45
DKP (Digital Station)

Signaling Proprietary Digital (2B+D)
Interface Single Pair for Speech, Signaling and Power
Loop Limit 100
Speech Level Adjustable Tx and Rx Gain for Handset and Hands-free
Protection Over Voltage Secondary Protection
EON48
EON48S EON48P
LCD 2 lines x 24 characters 6 lines x 24 characters
Dimensions 241x203x77.50mm (L x B x H) 241x203x77.50mm (LxBxH)
Weight 1.174 kg 1.302 kg
Environmental
Operating Temperature 0° C to 45°C 0° C to 45°C
Operating Humidity 5 to 95% RH, Non-Condensing 5 to 95% RH, Non-Condensing
Storage Temperature -20°C to +70°C -20°C to +70°C
Storage Humidity 5 to 95% RH, Non-Condensing 5 to 95% RH, Non-Condensing
EON310
LCD 2*24 Character display

Dimensions 207x335x78 mm (LxBxH)
Weight 820gms
Environmental
Operating Temperature 0° C to 45°C
Operating Humidity 5 to 95% RH, Non-Condensing
Storage Temperature -20°C to +70°C
Storage Humidity 5 to 95% RH, Non-Condensing
EON510
LCD 240*64 Pixel Graphical LCD display
Dimensions 247x183x43 mm (LxBxH)
Weight 805gms
Environmental
EON74 (Turret) Specifications

Compatible PBX Matrix ETERNITY LENX and ETERNITY MENX
Display 4x40 Characters LCD
Handset Dual Handsets with Push-to-Mute Switch
Handset Interface RJ12
Headset Interface 3.5mm Mono Jack
Dimension 15.01 x 13.01 x 1.43 inch
Mounting Angle 35°
Enclosure 1.50mm thickness Aluminium
Power Supply 12VDC, 1.25A
Weight 3.1 Kg (Without Handsets)
SPARSH VP248
VoIP
VoIP Protocols SIP v2, SDP, RTP, RFC 2833
Network Protocol IPv4, TCP, UDP, DHCP, STUN, PPPoE
NAT STUN and NAT Keep Alive

Voice CODECS G.722 Wideband, G.711 A/µ-Law, G.723-5.3, G.723-6.3,
G.726-16, G.726-24, G.726-32, G.726-40, G.729AB
Call Progress Tones Dial Tone, Ring Back Tone, Busy Tone, Error Tone,
Waiting Tone
Voice Dynamic Jitter Buffer (Adaptive), Comfort Noise

Generation and Voice Activity Detection
Quality of Service Layer 3 DIFFServ and TOS
Data Network LAN Port (RJ45), 10/100 Base T (PoE Optional)

PC Port (RJ45), 10/100 Base T
LCD Display 2 Lines and 6 Lines Display
Security Password Protected Configuration
Power Supply
Input 5VDC @2A through External Adaptor (90 - 265 VAC, 47

- 63Hz) and Power-over-Ethernet (PoE)
Power Consumption 5W (Typical)
Mechanical
Dimensions (WxHxD) 20.7 x 23.2 x 4.5 cm
Material ABS Plastic
Installation Mounting Wall Mount and Table - Top
Environmental
Operating Temperature 0°C to 45°C
Unit Weight 1.18 Kgs (2.6 lbs) Approx.
SPARSH VP310
VoIP
Network Protocol IPv4, TCP, UDP, DHCP, PPPoE
Voice CODECS G.722 Wideband, G.711 A/µ-Law, G.723, G.729, iLBC -

20/30 msec
Waiting Tone
Voice Dynamic Jitter Buffer (Adaptive)
Quality of Service Layer 2 CoS, Layer 3 DIFFServ and TOS

LCD Display 2 x 24 Character Display

Security TLS, SRTP
Power Supply
Input 5VDC @2A through External Adaptor (100-240 VAC, 50

- 60 Hz)
Mechanical
Environmental
Weight (Without Foot Stand) 830 gms Approx.
SPARSH VP330
LCD Display 4.3" Colour TFT Touch Screen Display
VoIP
Voice CODECS G.722 Wideband, G.711 A/µ-Law, G.723, G.729
Waiting Tone

LCD Display 2 x 24 Character Display
Security TLS, SRTP
Power Supply

- 60 Hz)
Power Consumption 5.50 W (Typical)
Mechanical

Environmental
Weight (Without Foot Stand) 840 gms Approx.
SPARSH VP510
VoIP
Voice CODECS G.722 Wideband, G.711 A/µ-Law, G.723, G.729

Security TLS, SRTP
LCD 240*64 Pixel Graphic LCD Display
Power Supply

- 60 Hz)
Power Consumption 2.25W (stand alone)
Mechanical
Weight 805 gm
Dimension in mm [L*B*H] 247*183*43
Environmental Conditions
Operating Temperature Range 0°C to 45°C
Operating and Storage Humidity 5 to 95%, RH, Non-Condensing
SPARSH VP248 (Open SIP Phone)

VoIP

Network Protocol IPv4, TCP, UDP, DHCP, STUN, PPPoE
SIP 9 Multiple SIP Accounts

Out Bound Proxy Support
Main and Secondary DNS Server Support
NAT STUN and NAT Keep Alive
Voice CODECS G.722 Wideband, G.711 A/µ-Law, G.723-5.3, G.723-6.3, G.726-

16, G.726-24, G.726-32, G.726-40, G.729AB
Call Progress Tones Dial Tone, Ring Back Tone, Busy Tone, Error Tone, Waiting Tone
Voice Dynamic Jitter Buffer (Adaptive), Comfort Noise Generation and

Voice Activity Detection
Quality of Service Layer 3 DIFFServ and TOS

Security Password Protected Administration
Power Supply
Input 5VDC @2A through External Adaptor

(90-265VAC, 47-63Hz)
Power-over-Ethernet (PoE)
Mechanical
Dimensions (WxHxD) 24.0 x 20.0 x 9.9 cm (9.4”x7.9”x3.9”)
Installation Mounting Wall Mount and Table-Top
Environmental
DSS64
Mechanical
Weight (Product + Stand) 639gm
Product 487gm
Stand 152gm
Dimension in mm [L*B*H] 185 x 200 x 46.50
Operating and Storage Humidity 5 to 95%, RH, Non-Condensing
Storage Humidity 0 to 95%, RH, Non-Condensing

DSS532
Terminals Supported EON510, SPARSH VP510
Source of Power Powered from the Host Phone
Programmable Keys 32
Stackable upto 4 modules
LED Indicator Dual Colour, Red/Blue
Signaling Proprietary Digital (2B+D)
Interface Single Pair for Speech, Signaling and Power
Physical Port IN Port: Connects with the AUX port of the phone or
OUT Port of the preceding DSS532 attached.
OUT Port: Connects with IN Port of the succeeding

DSS532 attached.
Installation Option Table Mount
Mechanical
Weight (Product + Stand) 235g
Product Weight 218g
Stand Weight 17g
Dimension [L*B*H] 178.7mm * 100mm * 40.5mm
Operating Humidity 5 to 95%, RH, Non-Condensing
Storage Humidity 0 to 95%, RH, Non-Condensing
CO (Central Office) / Two-Wire Trunk (TWT)

Signaling Loop Start
Loop Limit 1200
Off Hook AC Impedance 600/900/Complex
Pulse Dialing 10/20 PPS
DTMF Dialing and Reception ITU-T Q.23 and Q.24
Return Loss >18dB
Longitudinal Balance >50dB
Transmission Level Adjust Tx Gain: -15dB to +10dB, Rx Gain: -15dB to 10dB
CLI Reception DTMF, FSK ITU-T V.23 and FSK Bellcore 202
Call Maturity Delay and Polarity Reversal

Protection Over Voltage and Over Current Secondary Protection
ISDN BRI
Channels 2B+D
Personality Network (NT) and Terminal (TE)
AT&T 4ESS, DMS-100, ETSI NET3, ITU-T Q.921, ITU-T Q.931, NTT INS64, US NI1
Switch Variant
(National ISDN1), France VNx
Protection Solid state (Over Voltage and Over Current)

Built-in Secondary Protection
Physical
RJ45 (120)
Connector
ISDN PRI
Channels 23B+D and 30B+D
Personality Network (NT) and Terminal (TE)
Line Coding AMI/B8ZS for T1 and HDB3 for E1
Framing ESF for T1 and CEPT1 (with/without CRC) for E1
AT&T 4ESS, AT&T 5ESS, DMS-100, ETSI NET5, ITU-T Q.921, ITU-
Switch Variant T Q.931, NTT INS64, US NI2 (National ISDN 2), QSIC ECMA,
France VN
Protection Solid State (Over Voltage and Over Current)

Physical Connector RJ45 (Impedance Selectable) and fiber optic
Supplementary
QSIG ECMA
Services
E1 CAS
Bit Rate 2048 kbps +/-50 ppm
Line Coding HDB3
Framing CEPT1 (with/without CRC) with CAS MF
Line Signaling ITU-T Q.400 - Q. 490
Register Signaling MFC-R2
Alarms I.431. G.732, ETSI 300-233
Protection Solid State (Over Voltage and Over Current)

Physical Connector RJ45 (Impedance Selectable) and fiber optic

T1 RBS
Bit Rate 1544 kbps +/- 50 ppm
Line Coding AMI and B8ZS
FXS Loop Start, FXO Loop Start, FXS Ground Start, FXO Ground Start,
Line Signaling
E&M (Immediate, Wink Start, Wink Start FGD)
Framing D4, ESF
Digit Dialing DTMF
Alarms ANSI T1.231
Performance ANSI T1.403, ANSI T1.231, AT&T TR54016
Protection Solid state (Over Voltage and Over Current)

Physical
RJ45 (Impedance Selectable)
Connector
GSM Trunks
GSM Band (MHz) Quadband: GSM850, EGSM900, DSC1800, PCS1900
Compliant ETSI GSM Phase2/2+
SIM Card One SIM per GSM Port
SIM Interface 1.8V, 3V
Transmission Class 4 (2W) at GSM850MHz and EGSM900MHz band

Power Class 1 (1W) at DCS1800MHz and PCS1900MHz band
RF Sensitivity Better than -102dBm
Protocol AT Command Interface
External Antenna One Antenna per 4 GSM Ports,1.8/3.0*dBi, 50, SMA (Male)
Connector, Omni-Directional with cable of 3 meters length
* Depends on GSM Frequency Band.
3G
GSM Band Quadband: GSM850, EGSM900, DSC1800, PCS1900
(MHz) Triband: WCDMA 850/1900/2100 and WCDMA 900/1900/2100
Compliant ETSI GSM Phase2/2+
SIM Card One SIM per GSM Port
SIM Interface 1.8V, 3V
Transmission Output Power

Power Class 4 (2W) at GSM850/GSM900
Class 1 (1W) at DCS1800/PCS1900
Class 3 (0.25W) at UMTS 850/900/1900/2100

RF Sensitivity < -106dBm at GSM850, EGSM900, DCS1800, PCS1900
< -108dBm at WCDMA2100, WCDMA1900
< -106dBm at WCDMA850
Protocol AT Command Interface
External Antenna One Antenna per 4 GSM Ports,1.8/3.0*dBi, 50, SMA (Male)
Connector, Omni-Directional with cable of 3 meters length
* Depends on GSM Frequency Band.
CDMA
CDMA 800MHz CDMA cellular; 1900MHz PCS
Frequency Band
Compliant CDMA 1x; IS-95A
RUIM Card One RUIM per CDMA Port
RUIM Interface 1.8V, 3V
Max. Output +25dBm

Power
RF Sensitivity Typical -107dBm for 800MHz cellular and 1900MHz PCS
VoIP
VoIP Protocols SIP v2, SIP over TCP, Symmetric RTP, RTCP, 100rel/PRACK
Network Protocol IPv4, TCP, UDP, SNTP, STUN, ARP, ICMP, PPPoE, DHCP, DNS,
SMTP
SIP Maximum 99 SIP Accounts per system, Outbound Proxy Support,

Display Name, User Name, Password, URL, Proxy URL, Register
URL, Register Interval
NAT/Firewall
STUN and NAT Keep Alive
Support
Voice Codecs G.711 (A-Law, µ-Law), G.723, G.729AB, GSM-FR, iLBC
Line Echo
G.168 with 64 /128ms Tail Length
Cancellation
Voice Dynamic Jitter Buffer (Adaptive), Comfort Noise Generation and Voice
Activity Detection
Call Progress Tones Dial Tone, Ring Back Tone, Busy Tone, Error Tone
Fax T.38 Relay and Pass Through
Quality of Service SIP QoS and RTP QoS
Security MD5 Authentication for SIP, Password Protected Configuration by

Admin and User

E&M Trunks
Signaling Type IV and Type V
Speech Interface 2W/4W
Trunk Seizure Type Immediate, Immediate+Wink, Seizure Pulse,

Seizure Pulse+Wink, Express, Compander
Control Signal (CCS)
Address Signaling Pulse Dial - Pulse 10PPS, Pulse 20PPS

Tone Dial - DTMF
Return Loss 20dB
Transhybrid Loss 20dB against configurable balance of 600 ohm

or AT&T complex impedance
Transmit Gain +/- 1.0 dB
Receive Gain +/- 1.0 dB

Packing List
Verify contents of the package shipped to you with the contents listed below. If any of the items is missing or
damaged, contact your Dealer/Reseller.
ETERNITY LENX
Sr. No Item Name Qty
ETERNITY LENXa
1 1
2 Caster Wheels 4
3 Rack Mounting Clamp 2
4 Wall Mounting Clamp 2
5 Power Cable for 48VDC Power Supply Card 1
Ethernet Cable for Ethernet Port (Cable with RJ45

6 1
connector on both ends)
7 Cable with RJ45 connector on one end 1
8 M4/8 CSK PH Screw for Rack/ Wall Mounting 6
9 M16/50 Screws and screw grips for Wall Mounting 4
10 Warranty Cards 1
11 Quick Start (Printed) 1
12 Mounting Templates 1
a. Factory fitted with the Power Supply Card and CPU Card.
ETERNITY LENX Cards
ETERNITY MENX-LENX CPU Card
Sr. No. Item Quantity
1 ETERNITY MENX-LENX CARD CPU 1
2 Cable with RJ45 connector on one end - Length 3 meters 1
3 Cable with RJ45 connectors on both ends 1
ETERNITY LE Power Supply Card
1 ETERNITY LE CARD PS48VDC 1
2 Cable 1

ETERNITY LE ILC48 Card
1 ETERNITY LE CARD ILC48 1
2 Cable with 50 Pin Centronics Connector (Male) 2
ETERNITY LE SLT48 Card
1 ETERNITY LE CARD SLT48 1
2 Cable with 50 Pin Centronics Connector (Male) 2
ETERNITY LE Emerson PS Tray
1 ETERNITY LE Emerson PC Tray 1
2 M4/10 CSK PH Screws (to affix the PS onto the Tray) 3
3 M3/8 CSK PH Screws (to fix the Tray onto the LE rack)) 2
4 Tray Mounting Template 1
ETERNITY MENX
1 ETERNITY MENX16SAC IN a 1
2 3-pin Power Cord, MC-4 Black 1
5 Screw 50/16 for Wall Mounting for Wall Mounting 4
6 Screw grip 50x16 for Wall Mounting for Wall Mounting 4
7 Warranty Cards 1
9 Side Clamp Black 2
10 Screw M4X12 CSK for the Side Clamp 4
11 Antistatic Wrist Belt 1
12. Mounting Templates 1

ETERNITY MENX16SDC IN a
1 1
2 3-pin Power Cord, MC-4 Black 1
5 Screw 50/16 for Wall Mounting for Wall Mounting 4
6 Screw grip 50x16 for Wall Mounting for Wall Mounting 4
7 Warranty Cards 1
9 Side Clamp Black 2
11 Antistatic Wrist Belt 1
12. Mounting Templates 1
ETERNITY MENX Cards
ETERNITY MENX-LENX CPU Card
1 ETERNITY MENX-LENX CARD CPU 1
2 Cable with RJ45 connector on one end - Length 3 meters 1
ETERNITY ME Card PS48VDC 500W (Power Supply)
1. ETERNITY ME CARD PS48VDC-500 Watts 1
2. 3-Way Battery Cable 1
ETERNITY ME Card PSUNI 500W
1. ETERNITY ME Card PSUNI 500W 1
2. 3-pin Power Cord, MC-4 Black 1

1. ETERNITY ME Card SLT32 1
2. Cable with RJ45 connector on one end 8
1. ETERNITY ME Card DKP32 1
1 ETERNITY ME CARD DKP16 1
1 ETERNITY ME CARD DKP8 1
1 ETERNITY ME CARD CO32 1

1 ETERNITY ME CARD CO8+SLT24 1
1 ETERNITY ME CARD BRI8 1
3 Mini jumper 16
1 ETERNITY ME CARD BRI4 1
3 Mini jumper 8
1. ETERNITY ME Card T1E1PRI Dual 1
2. Cable with RJ45 connectors on both ends 2
1. ETERNITY ME Card T1E1PRI Single 1

1. ETERNITY ME Card GSM8 1
2. GSM Antenna External SMA 2
1. ETERNITY ME Card GSM4 1
1. ETERNITY ME Card E&M8 1
1. ETERNITY ME Card E&M4 1
ETERNITY GENX
Sr.No. Item Name Qty
1 ETERNITY GENX12Sa 1
3 3-pin Power Cord, MC-4 Blackb 1
4 3-pin DC Input Cablec 1

Sr.No. Item Name Qty
5 Screws M 7/30 for Wall Mounting 2
6 Screw Grips for Wall Mounting 2
7 Side Clamp 2
9 Warranty Cards 1
11 Mounting Templates 1
a. ETERNITY GENX12S AC with factory fitted AC Power Supply and CPU Cards.
ETERNITY GENX12S DC with factory fitted DC Power Supply and CPU Cards.
b. Supplied with ETERNITY GENX12S AC.
c. Supplied with ETERNITY GENX12S DC.
ETERNITY GENX Cards
ETERNITY GE CARD PS48VDC 250W
1. ETERNITY GE CARD PS48VDC 1
2. 3-pin DC Input Cable 1
ETERNITY GE CARD PSUNI 250W
1. ETERNITY GE CARD PSUNI 1
2. 3-pin Power Cord, MC-4 Black 1
ETERNITY GENX CARD CPU
1. ETERNITY GENX CARD CPU 1
1. ETERNITY GE CARD SLT20 1

1. ETERNITY GE CARD DKP16 1
1. ETERNITY GE CARD DKP8 1
1. ETERNITY GE CARD DKP4+SLT16 1
1. ETERNITY GE CARD DKP4+SLT12 1
1. ETERNITY GE CARD CO2+DKP2+SLT16 1

1. ETERNITY GE CARD CO4+SLT16 1
1. ETERNITY GE CARD CO16 1
1. ETERNITY GE CARD CO8 1
ETERNITY GE CARD BRI4
1. ETERNITY GE CARD BRI4 1
ETERNITY GE CARD T1E1PRI Single
1. ETERNITY GE CARD T1E1PRI Single 1

ETERNITY GE CARD E1FOPRI Single
1. ETERNITY GE CARD E1FOPRI Single 1
ETERNITY GE CARD GSM4
1. ETERNITY GE CARD GSM4 1
ETERNITY GE CARD E&M4
1. ETERNITY GE CARD E&M4 1
1. ETERNITY GE CARD ILC20 1
EON48
1. Phone, Handset and Spring Cord 1
2. Line Cord 1
3. Foot Stand 1
4. Wall Mounting Template 1
5. Warranty Card Set 1
6. Quick Installation Guide (printed copy) 1
EON74
1. EON74 Unit 1
2. Handsets 2
3. Foot Stand 1

4. Spring Cords 2
5. Power Adapter 12V, 1.25 Amp 1
EON310
2. Line Cord 1
3. Foot Stand 1
EON510
2. Line Cord 1
3. Foot Stand 1
SPARSH VP248 (Extended/Open)
2. Ethernet Cable 1
3. Power Adapter (5V, 2.0 Amp) 1
4. Foot Stand 1

SPARSH VP310 (Extended)
2. Ethernet Cable 1
4. Foot Stand 1
SPARSH VP330 (Extended)
2. Ethernet Cable 1
4. Foot Stand 1
DSS64
1. DSS64 Console 1
2. RJ45 Cable 1
DSS532
1. DSS532 Console Unit 1
2. DSS Extender 1
3. Foot Stand 1
4. RJ11 Cable 1
Clamps (2 DSS-Phone Clamps and 2 DSS-DSS

5 4
Clamps)

5. DSS532 Quick Installation Guide (printed copy) 1
EONSOFT
1. EONSOFT Dongle 1
2. Handset with Spring Cord 1
3. Communication Cable 1
4. Screw m7/30 2
5. Screw Grip 2
6. Mounting Template 1
PPM4
1. Matrix PPM4 1
2. Wall Mounting Screws 2
3. Wall Mounting Screw Grips 2
4. Mounting Template 1

How to Make the Telecom Earth
The Earth (Ground) is the most important safety procedure to prevent electrical shocks and fires. It protects from
lightning strikes, electrical transients, static discharges, electromagnetic interference and electrical hazards.
A proper earth must be in place to protect people and the system. The following explanation shows how a perfect
electrical earth can save lives.
P
Vs Vc
A.C. Input Electrical Parts of the Gadget
N Z1
Z2
Enclosure of the Gadget
Earth
In the above diagram, Vc = VS *Z2 / Z1 + Z2

Where,
• Z1 is the stray impedance between the electrical parts of the Gadget and the Chassis.
• Z2 is the stray impedance between the Chassis and the Earth.
• If Z2 = 0 then VC = 0
This formula implies that if the impedance between the Chassis and the Earth is reduced to 0 then the Voltage on
the Chassis, that is, VC, would be Zero and hence any person touching the enclosure will not get an electric shock.
Hence Z2 should be made Zero.
This can be done by providing a perfect earth to the electrical equipment.
It is recommended that you provide a dedicated earth for the ETERNITY GENX/any other telecom equipment. This
dedicated earth is called the Telecom Earth (Ground).
Providing a separate Telecom Earth to the telecom equipment eliminates the possibility of any back-voltage on the
earth.
CO Line CO Line
System
Lightning
Protective Earth
Protectors
Terminal
Telecom Earth

How to make a perfect Earth
Exchange Room
G.1 PIPE 4 Inch

Pit of area 2x2x6
BUS BAR
Copper strip 1 inch wide
x 3 mm thick x 6 mm long
Soil
1 layer of salt and charcoal 1:3 ratio
Copper plate 1.5x1.5 x 1/4 thick
• Dig a pit of area 2 feet x 2 feet x 6 feet (L x B x D).
• Get a copper plate of size 1.5 feet x 1.5 feet x 0.25 feet.
• Connect a copper strip of size 1-inch wide, 3 mm thick and 6 feet length at the center of the copper plate
by welding or nuts and bolts.
• Insert a G.I. pipe onto the copper strip till it reaches the copper plate.
• Place this set up into the pit. Make sure that at least 4 inches of the G.I. pipe is above the ground level.
• Fill the bottom of the pit with a 1-inch layer of charcoal and salt in the proportion of 3:1 (3 parts charcoal, 1
part salt) and then cover with the soil.
• Connect a bare 14 SWG copper wire (double) on the top of the copper strip and run it to the exchange
room and connect it on the bus bar.
• The Bus bar is a copper strip, 4 inches long with 6 screws and nuts mounted on it. It has to be fixed on the
wall in the exchange room.
• The earth wire of the Primary Protection Modules (PPM) should be connected to this Bus bar.
• Water the earth at regular intervals.

Connecting ETERNITY GENX to the Earth
Follow the instructions given below to connect the ETERNITY GENX to the Earth. The instructions are given with
reference to the above diagram:
1. Loosen the screw.

2. Insert an earthing wire into the lug.
3. Crimp the earth wire with an appropriate tool.
4. Tighten the screw.
5. Connect the earthing wire to the earth.
Make sure you use an earthing wire that has a conductor with a cross-sectional area greater than 1.0
mm2 or AWG less than or equal to 16 AWG with Green and Yellow insulation.
• Make sure you comply with all applicable laws, regulations and guidelines.
• Proper earthing is very important to protect the ETERNITY GENX from external noise and to reduce
the risk of electrocution in the event of a lightning strike.
• The AC cable's earthing pin may not be enough to protect the ETERNITY GENX from external noise
and lightning strikes. A permanent connection must be made between earth and the earth terminal of
the main unit.

VMS Prompts
Given below are the prompts which are already recorded in English language and are provided to you by default.
Along with the prompts, the Prompt Names and File Names are also listed below. You must use these prompts as
a reference while recording and uploading the prompts for other languages. For further information, refer “Prompts
Management”.
Greetings
Prompt Name File Name Prompts
Greeting 01 Morning.wav Good Morning!
Greeting 02 Afternoon.wav Good Afternoon!
Greeting 03 Evening.wav Good Evening!
Auto Attendant Prompts

Auto Attendant 01 Working.wav Welcome! Please dial the extension number
or
To dial by name press 6.
To leave a message press 7.
To access your Personal Mailbox press 8.
For further assistance press 9.
To disconnect the call press # (HASH).
Auto Attendant 02 Break.wav Thank you for calling! We are closed at this time. To leave a
message press 7.
Auto Attendant 03 Nonworking.wav Thank you for calling! We are closed at this time. To leave a
message press 7.
Auto Attendant 04 WH_Hotel.wav Welcome!
Please dial the room number or to dial by name press 6.
Auto Attendant 05 BH_Hotel.wav Welcome!

Auto Attendant 06 NWH_Hotel.wav Welcome!
To disconnect the call press #. (HASH)
Auto Attendant 07 WH_USA.wav Welcome! Please dial the extension number or to dial by name
press 6.
To connect to the operator press 0.
To disconnect the call press £. (POUND)
Auto Attendant 08 BH_USA.wav Thank you for calling! We are closed at this time. To leave a
message press 7.
Auto Attendant 09 NWH_USA.wav Thank you for calling! We are closed at this time. To leave a
message press 7.
Auto Attendant 10 WH_Hotel_USA.wav Welcome!
Auto Attendant 11 BH_Hotel_USA.wav Welcome!
Auto Attendant 12 NWH_Hotel_USA.wav Welcome!
Auto Attendant 13 Holiday.wav Thank you for calling! We are closed due to holiday. To leave a
message press 7.
To disconnect the Call press #. (HASH)

Auto Attendant 14 Holiday_Hotel.wav Welcome!
To disconnect the Call press #. (HASH)
Auto Attendant 15 Holiday_USA.wav Thank you for calling! We are closed due to holiday. To leave a
message press 7.
To disconnect the Call press £. (POUND)
Auto Attendant 16 Holiday_Hotel_USA.wavWelcome!
To disconnect the Call press £. (POUND)
Number Dialing
Number Dialing 01Entextn.wav Please enter the extension number.
Number Dialing 02DialName.wav Please enter the first three letters of the name.
Number Dialing 03MBAccess.wav Please enter your extension number.
Number Dialing 04Leavemsg.wav Please enter the extension number for which you wish to leave
message.
Number Dialing 05LeavemsgH.wav Please enter the room number or the extension number for
which you wish to leave message.
Number Dialing 06MsgdestI.wav Please enter the destination and press hash [#] to end.
Number Dialing 07MsgdestU.wav Please enter the destination and press Pound [£] to end.
No Digit Dialed
No Digit Dialed 01 NoDigitDialed.wav Sorry, You have not dialed any digit.
No Digit Dialed 02 NoOptionSelected.wav Sorry, You have not selected any option.
No Digit Dialed 03 NoInput.wav Sorry, You have not entered any input.
No Digit Dialed 04 NoDestination.wav Sorry, this is an invalid input.
Invalid Digit Dialed

Invalid Digit Dialed 01 InvalidInput.wav Sorry, this is an invalid input.
Invalid Digit Dialed 02 InvalidDigit.wav Sorry, this is an Invalid digit.

Invalid Digit Dialed 03 Invalidno.wav Sorry, this is an Invalid Number.
Invalid Digit Dialed 04 NoMatchFound.wav No match found.
Invalid Digit Dialed 05 InvalidDest.wav Invalid recipient. Please enter all recipients again.
Expiry of Count
Expiry Of Count 01 RetryCountOver.wav Sorry! Maximum attempts exceeded.
Call Transfer Type

Call Transfer Type 01 PlsHold.wav Please hold.
Call Transfer Type 02 XfrCall.wav Please hold, transferring the call.
Call Transfer Type 03 Xfrcallto.wav Please hold, transferring the call to
Call Transfer Type 04 XfrcalltoOpr.wav Please hold, transferring the call to Operator.
Call Transfer Type 05 XferMailbox.wav Please hold, transferring the call to mailbox.
Call Transfer Type 06 XferMailboxof.wav Please hold, transferring the call to mailbox of
Call Transfer Unsuccessful

Call Transfer Unsuccessful 01 Noreply.wav Sorry! The person is unavailable to take your call right
now.
Call Transfer Unsuccessful 02 Extbusy.wav Sorry! The person you are trying to call is busy.
Call Transfer Unsuccessful 03 Noattnd.wav Sorry! The call could not be attended.
MoH
MoH 1 Holdmusic.wav <Music>
Disconnect
Disconnect 01 Thankyou.wav Thank you for your call.
Language
Language 01 english.wav For English,

Dial by Name
Dial by Name 01 PreName.wav The following matches are found
Dial by Name 02 selectname.wav To select the name,
Dial by Name 03 nextname.wav To skip to the next name,
Dial by Name 04 repeatlastname.wav To repeat the previous name,
Dial by Name 05 repeatnames.wav To repeat all the names,
Dial by Name 06 enternameagain.wav To re-enter the name,
Dial by Name 07 previousmenu.wav To return to the previous menu,
Dial by Name 08 disconnect.wav To disconnect,
Dial by Name 09 Selected.wav Name is selected,
Dial by Name 10 NameSelected.wav Selected name is,
Dial by Name 11 ConfirmName.wav You have selected,
Dial by Name 12 confirm.wav To confirm,
Dial by Name 13 reenter.wav To re-enter,
Call Transfer
Call Transfer 01 RecName.wav Please Record your name after the beep and press any digit to
end
Call Transfer 02 Callfrom.wav Hello! There is a call from,
Call Transfer 03 acceptcall.wav To accept the call,
Call Transfer 04 reject.wav To reject the call,
Call Transfer 05 rejectbusy.wav To reject the call as busy,
Call Transfer 06 rejectnoreply.wav To reject the call as no reply,
Call Transfer 07 Leavemsg1.wav To leave a message,
Call Transfer 08 XfertoOperator.wav To transfer the call to operator,
Call Transfer 09 XfertoAssitance.wav To transfer the call to assistant,
Call Transfer 10 XfertoAlternate.wav To transfer the call to alternate number,
Call Transfer 11 DialExtn.wav To dial an extension,
Call Transfer 12 XferMainMenu.wav To return to Main Menu,
Call Transfer 13 XferPreviousMenu.wav To return to Previous Menu,
Call Transfer 14 StayonHold.wav To stay on hold,
Call Transfer 15 disconnect.wav To disconnect,
Call Transfer 16 nonumber.wav Number not programmed.
Message Record
Message Record 01 Recmsg.wav Please Record your Message after the beep and press any
digit to end.
Message Record 02 RecMsgStopCode.wav Please Record your message after the beep and press

Message Record 03 RecMsgEnd.wav To end,
Message Record 04 MsgLeaveConfirm.wav Message sent as,
Message Record 05 Rerecord.wav To re-record the message,
Message Record 06 Confirm.wav To confirm the message as normal,
Message Record 07 urgent.wav To confirm the message as urgent,
Message Record 08 private.wav To confirm the message as private,
Message Record 09 listenmsg.wav To play the recorded message,
Message Record 10 appendmsg.wav To append to the recorded message,
Message Record 11 disconnect.wav To disconnect,
Message Record 12 normalset.wav Normal
Message Record 13 urgentset.wav Urgent
Message Record 14 privateset.wav Private
Message Record 15 securedset.wav Secured
Message Record 16 NoMailbox.wav Mailbox not assigned.
Message Record 17 Mailboxfull.wav Sorry! your message cannot be delivered as Mailbox is full.
Message Record 18 Memoryfull.wav Sorry! your message cannot be delivered as System Storage is
full.
Message Record 19 appendfail.wav Sorry! message cannot be appended as System Storage is full.
Message Record 20 urgentandprivate.wav To confirm the message as urgent and private,
Message Send Forward

Message Send Forward 01 Recmsg.wav Please Record your Message after the beep and
press any digit to end
Message Send Forward 02 RecMsgStopCode.wav Please Record your message after the beep and
press
Message Send Forward 03 RecMsgEnd.wav to end
Message Send Forward 04 MsgSendConfirm.wav Message sent as
Message Send Forward 05 reenter.wav To re-enter
Message Send Forward 06 Confirm.wav To confirm
Message Send Forward 07 commentatstart.wav To forward the message with comment before the
message
Message Send Forward 08 commentatend.wav To forward the message with comment after the
message
Message Send Forward 09 nocomment.wav To forward the message without comment
Message Send Forward 10 Rerecord.wav To re-record the message
Message Send Forward 11 Confirm.wav To confirm the message as normal
Message Send Forward 12 urgent.wav To confirm the message as urgent
Message Send Forward 13 private.wav To confirm the message as private
Message Send Forward 14 listenmsg.wav To play the recorded message
Message Send Forward 15 appendmsg.wav To append to the recorded message
Message Send Forward 16 Requestreceipt.wav To request read receipt
Message Send Forward 17 ignorereceipt.wav To ignore read receipt

Message Send Forward 18 previousmenu.wav To return to previous menu
Message Send Forward 19 normalset.wav Normal
Message Send Forward 20 urgentset.wav Urgent
Message Send Forward 21 privateset.wav Private
Message Send Forward 22 securedset.wav Secured
Message Send Forward 23 NumberCollected.wav Recipient number entered is
Message Send Forward 24 FwdMsgFail.wav Sorry! Message cannot be forwarded as message
length has exceeded
Message Send Forward 25 Pending.wav Sorry! the message cannot be sent
Message Send Forward 26 NoDigitDialed.wav You have not dialed any digit
Message Send Forward 27 nextnumber.wav Enter the next recipient number
Message Send Forward 28 NoMailbox.wav Sorry! Mailbox is not assigned
Message Send Forward 29 appendfail.wav Sorry, message cannot be appended as System
Storage is full
Message Send Forward 30 Mailboxfull.wav Sorry, your message cannot be delivered as Mailbox
is full
Message Send Forward 31 enterDDMMYY24.wav Enter Delivery Date in DD MM YYYY format and time
in Twenty four hour format
Message Send Forward 32 enterDDMMYY12.wav Enter Delivery Date in DD MM YYYY format and time
in Twelve hour format. For AM, press 1, for PM,
press 2.
Message Send Forward 33 enterMMDDYY24.wav Enter Delivery Date in MM DD YYYY format and time
in Twenty four hour format
Message Send Forward 34 enterMMDDYY12.wav Enter Delivery Date in MM DD YYYY format and time
in Twelve hour format. For AM, press 1, for PM,
press 2.
Message Send Forward 35 checkfuturedelivery.wav To check the last entered Future Delivery
Message Send Forward 36 cancelfuturedelivery.wav To cancel the last entered Future Delivery
Message Send Forward 37 futuredelconf.wav To confirm press 1. To cancel press any digit
Message Send Forward 38 DiscardMsg.wav To discard the message and return to previous
menu
Mailbox Access
Mailbox Access 01 Enterpwd.wav Please Enter your mailbox password.
Mailbox Access 02 Youhave.wav You have,
Mailbox Access 03 Newmsg.wav New message.
Mailbox Access 04 Newmsgs.wav New messages.
Mailbox Access 05 Nonewmsg.wav You have no new messages.
Mailbox Access 06 Urgentmsg.wav Urgent message
Mailbox Access 07 Urgentmsgs.wav Urgent messages
Mailbox Access 08 Nooldmsg.wav You have no old messages.
Mailbox Access 09 PerMB.wav In your personal mailbox
Mailbox Access 10 DeptMB.wav In your department group mailbox

Mailbox Access 11 MessageCount.wav Message
Mailbox Access 12 Msgrecon.wav This message was recorded on,
Mailbox Access 13 ConvBetn.wav This conversation was between,
Mailbox Access 14 Callnum.wav And calling number was,
Mailbox Access 15 MsgreadBy.wav This message was read by,
Mailbox Access 16 DeptOrStn.wav Press '1' to access personal mailbox, Press '2' to access
department group mailbox
Mailbox Access 17 RecdCncl.wav Sorry! Your message cannot be delivered.
Mailbox Access 18 MB80Full.wav Your Mailbox is 80% Full. Please Delete old messages of your
mailbox.
Mailbox Access 19 MBFull.wav Your Mailbox is Full. Please Delete old messages of your
Mailbox.
Mailbox Access 20 Invalpwd.wav Sorry! This is an invalid password
Mailbox Access 21 MBblocked.wav Sorry! Mailbox is currently in use. Please try again later.
Mailbox Access 22 NoMailbox.wav Sorry! Mailbox is not assigned
Mailbox Access 23 SystemMemFull.wav System Storage is Full. Please Delete some stored messages.
Mailbox Access 24 privatemsg.wav Private Message
Mailbox Access 25 urgentprivatemsg.wav Urgent and Private message
Mailbox Access 26 Disconnect.wav Thank you for your call.
Mailbox Access 27 Messagefor.wav Message for extension,
Mailbox Access 28 MsgDelete.wav Message deleted.
Mailbox Access 29 MsgSaveCnf.wav Your message has been saved as new.
Mailbox Access 30 recorddel.wav Recording erased.
Mailbox Access 31 Numdel.wav Number erased.
Mailbox Access Menu

Mailbox Access Menu 01 listennewmsg.wav To listen to a new message,
Mailbox Access Menu 02 listenoldmsg.wav To listen to an old message,
Mailbox Access Menu 03 sendmsg.wav To send a message,
Mailbox Access Menu 04 Mbmgnt.wav For mailbox management,
Mailbox Access Menu 05 replaymsg.wav To re-play the message,
Mailbox Access Menu 06 playmsgdetails.wav For Date and Time of the message,
Mailbox Access Menu 07 replymsg.wav To reply the message,
Mailbox Access Menu 08 deletemsg.wav To delete the message,
Mailbox Access Menu 09 listennextmsg.wav To Play the next message,
Mailbox Access Menu 10 forwardmsg.wav To forward the message,
Mailbox Access Menu 11 msgasnew.wav To mark the message as unread,
Mailbox Access Menu 12 Mbname.wav For mailbox name,
Mailbox Access Menu 13 MsgRedirection.wav For message redirection,
Mailbox Access Menu 14 DeleteAllOldmsgs.wav To delete all old messages of your mailbox,
Mailbox Access Menu 15 RecMBgrt.wav For mailbox greetings,
Mailbox Access Menu 16 Assistance.wav For assistant number,

Mailbox Access Menu 17 Personal.wav For personal number,
Mailbox Access Menu 18 SetMsgRedirection.wav To set message redirection,
Mailbox Access Menu 19 CancelMsgRedirection.wav To cancel message redirection,
Mailbox Access Menu 20 MsgRedirectNo.wav Enter the recipient number,
Mailbox Access Menu 21 PersonalGrt.wav For personal greetings,
Mailbox Access Menu 22 WHGrt.wav For working hours greeting,
Mailbox Access Menu 23 BHGrt.wav For break hours greeting,
Mailbox Access Menu 24 NWHGrt.wav For non-working hours greeting,
Mailbox Access Menu 25 ConditionalGrt.wav For conditional greetings,
Mailbox Access Menu 26 BusyGrt.wav For busy,
Mailbox Access Menu 27 NoReplyGrt.wav For no-reply,
Mailbox Access Menu 28 UnconditionalGrt.wav For unconditional,
Mailbox Access Menu 29 Record.wav To record,
Mailbox Access Menu 30 Play.wav To play,
Mailbox Access Menu 31 Erase.wav To erase,
Mailbox Access Menu 32 EnterNum.wav To enter number,
Mailbox Access Menu 33 PlayNum.wav To play number,
Mailbox Access Menu 34 ClearNum.wav To clear number,
Mailbox Access Menu 35 EnterAssistanceNum.wavEnter the assistant extension number.
Mailbox Access Menu 36 EnterPersonalNum.wav Enter the personal number and Press Hash [#] to end
Mailbox Access Menu 37 previousmenu.wav To return to Previous Menu
Mailbox Access Menu 38 Recname.wav Please Record your name after the beep and press
any digit to end.
Mailbox Access Menu 39 Recgreeting.wav Please Record the greeting after the beep and press
any digit to end.
Mailbox Access Menu 40 DelAllOldCnf.wav You are about to delete all read messages of your
mailbox. To proceed, Press 1, to Cancel, Press any
digit.
Mailbox Access Menu 41 DelAllOldDone.wav All old messages are deleted.
Mailbox Access Menu 42 MsgRedirectSet.wav Message redirection set.
Mailbox Access Menu 43 MsgRedirectCancel.wav Message redirection canceled.
Mailbox Access Menu 44 Numbersaved.wav Number saved.
Mailbox Access Menu 45 Numberdelete.wav Number deleted.
Mailbox Access Menu 46 Invalidno.wav Sorry, this is an invalid number.
Mailbox Access Menu 47 NoDigitDialed.wav Sorry, you have not dialed any digit.
Mailbox Access Menu 48 DelAllMsgsCnf.wav You are about to delete all messages of your
mailbox. To proceed press 1. To cancel, press any
digit.
Mailbox Access Menu 49 DelAllMsgsDone.wav All messages are deleted.
Mailbox Access Menu 50 DelAllMsgs.wav To delete all messages of your mailbox,
Mailbox Access Menu 51 Recordingdelete.wav Recorded file deleted.
Number and Month


Number and Month 01 Num0.wav Zero
Number and Month 02 Num1.wav One
Number and Month 03 Num2.wav Two
Number and Month 04 Num3.wav Three
Number and Month 05 Num4.wav Four
Number and Month 06 Num5.wav Five
Number and Month 07 Num6.wav Six
Number and Month 08 Num7.wav Seven
Number and Month 09 Num8.wav Eight
Number and Month 10 Num9.wav Nine
Number and Month 11 Num10.wav Ten
Number and Month 12 Num11.wav Eleven
Number and Month 13 Num12.wav Twelve
Number and Month 14 Num13.wav Thirteen
Number and Month 15 Num14.wav Fourteen
Number and Month 16 Num15.wav Fifteen
Number and Month 17 Num16.wav Sixteen
Number and Month 18 Num17.wav Seventeen
Number and Month 19 Num18.wav Eighteen
Number and Month 20 Num19.wav Nineteen
Number and Month 21 Num20.wav Twenty
Number and Month 22 Num30.wav Thirty

Number and Month 23 Num40.wav Forty
Number and Month 24 Num50.wav Fifty
Number and Month 25 Num60.wav Sixty
Number and Month 26 Num70.wav Seventy
Number and Month 27 Num80.wav Eighty
Number and Month 28 Num90.wav Ninety
Number and Month 29 Num100.wav Hundred
Number and Month 30 Num1000.wav Thousand
Number and Month 31 Month1.wav January
Number and Month 32 Month2.wav February
Number and Month 33 Month3.wav March
Number and Month 34 Month4.wav April
Number and Month 35 Month5.wav May
Number and Month 36 Month6.wav June

Number and Month 37 Month7.wav July
Number and Month 38 Month8.wav August
Number and Month 39 Month9.wav September
Number and Month 40 Month10.wav October
Number and Month 41 Month11.wav November
Number and Month 42 Month12.wav December
Number and Month 43 Hash.wav Hash
Number and Month 44 Pound.wav Pound
Number and Month 45 Star.wav Star
Number and Month 46 asterisk.wav Asterisk
Alarm
Alarm 01 EntertimeI.wav Enter the time, HH:MM in Twenty four hour format. To cancel
all alarms, press Pound [£]
Alarm 02 EntertimeU.wav Enter the time, HH:MM in Twelve hour format. For AM, press
1. For PM, press 2 To cancel all alarms, press Pound [£]
Alarm 03 WakeupCancel.wav Your all wake up alarms are canceled.
Alarm 04 SetOnceDaily.wav To set once press '1'. To set daily press '2'
Alarm 05 WakeupVeri.wav You have set wake up alarm for
Alarm 06 DailyWakeupVeri.wav You have set daily wake up alarm for
Alarm 07 WakeupSet.wav Your wake up alarm is set.
Alarm 08 DailyWakeupSet.wav Your daily wake up alarm is set.
Alarm 09 Am.wav A.M.
Alarm 10 Pm.wav P.M.
Alarm 11 AlarmConf.wav To confirm press 1, To re-enter press 2
Alarm 12 AlarmDateI.wav Enter the Date in DD MM YYYY format. To cancel all
reminders, press Pound [£]
Alarm 13 AlarmDateU.wav Enter the Date in MM DD YYYY format. To cancel all
reminders, press Pound [£]
Alarm 14 ReminderCancel.wav Your all reminders are canceled.
Alarm 15 ReminderVeri.wav You have set reminder for
Alarm 16 ReminderSet.wav Your reminder is set
Alarm 17 Alarmnoset.wav Sorry! Your wake up alarm cannot be set. Please call operator
for further assistance.
Alarm 18 Remindernoset.wav Sorry! Your reminder cannot be set. Please call operator for
further assistance.
Alarm 19 RemoteExt.wav Please Enter the extension number for which you wish to set
or cancel wake up alarm.
Alarm 20 RemoteExtH.wav Please Enter the room number for which you wish to set or
cancel wake up alarm.
Alarm 21 PerWakeupVeri.wav You have set personal wake up alarm for,
Alarm 22 AutoWakeupVeri.wav You have set automated wake up alarm for,
Alarm 23 PerWakeupSet.wav Your personal wake up alarm is set.
Alarm 24 AutoWakeupSet.wav Your automated wake up alarm is set.

Alarm 25 DailyPerWakeupSet.wav Your daily personal wake up alarm is set.
Alarm 26 DailyAutoWakeupVeri.wav You have set daily automated wake up alarm for,
Alarm 27 DailyPerWakeupVeri.wav You have set daily personal wake up alarm for,
Alarm 28 DailyAutoWakeupSet.wav Your daily automated wake up alarm is set.
Alarm 29 PerReminderVeri.wav You have set personal reminder for,
Alarm 30 AutoReminderVeri.wav You have set automated reminder for,
Alarm 31 PerReminderSet.wav Your personal reminder is set.
Alarm 32 AutoReminderSet.wav Your automated reminder is set.
Alarm 33 Alarmmode.wav To set it as Personal, Press 1. To set it as Automated, Press 2.
Alarm 34 Alarmnocancel.wav Sorry! There is no alarm set to cancel.
Alarm 35 Remindernocancel.wav Sorry! There is no reminder set to cancel.
Alarm 36 WakeUpgreeting.wav This is your wake up call.
Alarm 37 DailyWakeUpgreeting.wav This is your daily wake up call.
Alarm 38 Remindergreeting.wav This is your reminder call.
Alarm 39 SWakeUpgreeting.wav This is your wake up call. To acknowledge press '0'.
Alarm 40 SDailyWakeUpgreeting.wav This is your daily wake up call. To acknowledge press '0'.
Alarm 41 SRemindergreeting.wav This is your reminder call. To acknowledge press '0'.
Alarm 42 Acknowledge.wav Your alarm is acknowledged.
Alarm 43 Entertime2I.wav Enter the time, HH MM in Twenty four hour format
Alarm 44 Entertime2U.wav Enter the time, HH MM in Twelve hour format. For AM, press
1. For PM, press 2.
Alarm 45 RemoteExtRem.wav Please Enter the extension number for which you wish to set
or cancel reminder.
Alarm 46 RemoteExtHRem.wav Please Enter the room number for which you wish to set or
cancel reminder.
Alarm 47 RemAcknowledge.wav Your reminder is acknowledged.
Alarm 48 Thankservice.wav Thank You for using this service.
Alarm 49 Morning.wav Good Morning!
Alarm 50 Afternoon.wav Good Afternoon!
Alarm 51 Evening.wav Good Evening!
Alarm 52 NoInput.wav Sorry, You have not entered any input.
Alarm 53 InvalidInput.wav Sorry, this is an invalid input.
Miscellaneous
Miscellaneous 01 Press.wav Press,
Miscellaneous 02 Dial.wav Dial,
Miscellaneous 03 And.wav and,
Miscellaneous 04 At.wav At,
Miscellaneous 05 Beep.wav <Beep>
Miscellaneous 06 NoDISA.wav Sorry! DISA feature is not allowed.
Miscellaneous 07 NoDISAStn.wav Sorry! DISA feature is not allowed for dialed extension.
Miscellaneous 08 MsgNtfyFor.wav Message notification for extension number,
Miscellaneous 09 DialDigit.wav Press any digit to proceed.
Miscellaneous 10 Chkinwel.wav Welcome! It is our pleasure to serve you. We will do our best to
make your stay comfortable.
Miscellaneous 11 Checkout.wav Sorry! The Guest has checked out.

Miscellaneous 12 to.wav to,
Miscellaneous 13 Holdmusic.wav <Music>
Miscellaneous 14 Entextn.wav Please enter the extension number.
Miscellaneous 15 EntPwd.wav Please enter your password.
Miscellaneous 16 Invalpwd.wav Sorry, this is an invalid password.
Miscellaneous 17 enterfoldernum.wav Enter folder number.
Miscellaneous 18 enterfilenum.wav Enter file number.
Miscellaneous 19 RecPrompt.wav Please Record your prompt after the beep and press any digit to
end.
Miscellaneous 20 InvalidInput.wav Sorry, this is an invalid input
Miscellaneous 21 NoInput.wav Sorry, you have not entered any digit
Miscellaneous 22 Record.wav To record
Miscellaneous 23 Play.wav To play
Miscellaneous 24 Erase.wav To erase

SARVAM UCS Features Supported in Terminals

Features at a Glance
Abbreviated Dialing
Personal/Global Abbreviated Dialing 8-Location Code
Program Personal memory 1071-Location Code-Number-#*-TAC
Account Code
Account Code by Number 1058-Account Code
Account Code by Name 1059-Account Name
Alarms
Once Only Alarm 161-Hours-Minutes-1
Daily Alarms 161-Hours-Minutes-2
Cancel Once Only/Daily Alarm 161-#
Set/Cancel Voice Guided Alarm 163-Follow VMS Prompts
Auto Call Back
Auto Call Back-On Busy 2
Auto Call Back-On No Reply 2
Cancel Auto Call Back 102
Auto Redial
Auto Redial 17
Cancel Auto Redial 1070
Barge-In
Barge-In 4
Call Cost Display
Last Ten dialed numbers Cost display 1075
Call Chaining
Call Chaining 1050

Call Forward
Call Forward-All Calls to Another Station 131-Station/Department Group/VMS
Call Forward-All Calls to External Station 131-Trunk Access Code-Dest. Number-#*
Call Forward-If Busy 132-Station/Department Group/VMS
Call Forward-If Busy-All Calls to External Number 132-Trunk Access Code-Dest. Number-#*
Call Forward-If No Reply 133-Station/Department Group/VMS
Call Forward-If No Reply-All Calls to External Number 133-Trunk Access Code-Dest. Number-#*
Call Forward-If Busy or No Reply 134-Station/Department Group/VMS
Call Forward-If Busy or No Reply-All to External

134-Trunk Access Code-Dest. Number-#*
Number
Call Forward-Dual Ring 1361
Disable Call Forward-Dual Ring 1360
Cancel Call Forward 130
Call Forward - Scheduled
For Working Hours
Call Forward-Scheduled - Unconditional 1175-1-1-Destination Number
Call Forward - Scheduled - Busy 1175-1-2-Destination Number
Call Forward - Scheduled - No Reply 1175-1-3-Destination Number
Call Forward - Scheduled - Busy/No Reply 1175-1-4-Destination Number
Call Forward - Scheduled - Set Dual Ring 1175-1-5-1
Call Forward - Scheduled - Cancel Dual Ring 1175-1-5-0
Call Forward - Scheduled - Cancel for Working Hours 1175-1-0
For Break Hours
Call Forward - Scheduled - Cancel for Break Hours 1175-2-0
For Non-Working Hours

Call Forward - Scheduled - Cancel for Non-Working

1175-3-0
Hours
Cancel Call Forward - Scheduled for all Time Zones 1175-0
Call Forward - Department Group
Call Forward-Department Group - Unconditional 1179-Department Group Number (Access Code)-1-

Destination Number
Call Forward-Department Group - Busy 1179-Department Group Number (Access Code)-2-

Destination Number
Call Forward-Department Group - No Reply 1179-Department Group Number (Access Code)-3-

Destination Number
Call Forward-Department Group - Busy/No Reply 1179-Department Group Number (Access Code)-4-
Destination Number
Call Forward-Department Group - Cancel 1179-Department Group Number (Access Code)-0
Call Hold
Put the caller on Hold Flash
Retrieve the caller Flash
Call Park
To Park a Call 115-Orbit Number
To Retrieve the Parked Call 116-Orbit Number
Call Pick Up
Call Pick Up-General 4
Call Pick Up-Selective 12-Station
Call Toggle
Flash-1 (for SLT), Transfer-1 (for DKP/Extended IP

Call Toggle (Toggle)
phone)
Call Transfer
Call Transfer to Station Speech with Station-Flash-Station (Transfer

Target)-OnHook

Call Transfer to Station (External Number) Speech with External Number-Flash-Station
(Transfer Target)-OnHook
Call Transfer to Trunk (External) Speech with External Number -Flash-TAC-External

Number (Transfer Target)-OnHook
Call Transfer to Trunk Speech with Station -Flash-TAC-External Number-

Go OnHook
Blind Transfer to Mail Box (VMS) Flash-1078-Station
Calling Line Identification Restriction
Enable/disable CLIR 103
Cancel All Station Features
Cancel all station features 1051
Conference 3-Party
Conference 3-Party Flash-*3
Conference Dial-In
Schedule a Conference *19-1-Conference Number-Conference Password
Initiate a Conference *19-2-Conference Number-Conference Password
Cancel a Conference *19-0
Conference Multiparty
To Temporarily Leave / Rejoin a Conference 191
Terminate Conference 190
Conversation Recording Flash-1095
Department Call
Department Call Department Number (3901-3916)
Direct Inward System Access
Enter DISA Mode 1079-Station Number-User Password

Do Not Disturb
Set Do Not Disturb with Text Message 18-DND Message Number
Cancel Do Not Disturb 18-0
DND Override 4
Dynamic Lock
Set Dynamic Timer 142-User Password-Minutes
Change Toll Control Level 141-User Password-Level
Make Emergency Conference 1177-Department Group Number
While in speech with any one party, Go Off-Hook

Cancel Emergency Conference
and dial 190
Emergency Dialing
Go Off-Hook and dial Emergency Number

Dial Emergency Number or
Dial TAC-Emergency Number
Flashing on Trunks
Flashing on Trunks Flash-*
Floor Service
To access Floor Service 38
Follow Me
Set Call Follow Me 135-Station-User Password
Cancel Call Follow Me 130
Forced Answer
Forced Answer 5
Forced Call Disconnection
Forced Call Disconnection #*

Hot Desking
Set Hot Desk 1091-Your Station Number-User Password
Hotline
Set Hotline 151-Station
Set Hotline Timer 154-Seconds
Hot Outward Dialing 152-Trunk Access Code
HOD with Number 153-Trunk Access Code-Number-#*
Cancel Hotline/HOD 150
Interrupt Request
Interrupt Request 3
Last Caller Recall
Last Caller Recall 1092
Last Number Redial
Last Number Redial 7
Live Call Screening
To Activate LCS 1094-1
To Deactivate LCS 1094-0
Live Call Supervision
Set Live Call Supervision 1098-Destination Station
Maid-In
Maid Status from a Room 1054-Code
Meet Me Paging
Meet Me Paging-Caller 1093-Page Zone Number
Meet Me Paging-Called Party 1093-Station Number of the Caller
Message Wait
Message Wait Set/Cancel 1076-Station-Code

Message Wait Retrieval 1077
Mini Bar
To check the utilities of the room 1056-Item Number-Quantity
Mute
Mute 1052
To grab OG Trunk Bundle Group 0/5/6-OGTBG Index
Operator
Call to Operator 9
Paging
Paging 1074-Page Zone Number
Presence
Publishing Presence by extension user 104-Password-Index No.
To view Presence Status 1097-Extension Number
Priority Calls in E&M MFCR2 Signaling
Manual Priority Intrusion *37
Forced Released Order *38
Raid
Raid 5
RCOC
RCOC in DISA Mode ** on dial tone
Reminder
Set Reminder 162-DD-MM-YYYY-HH-MM
Cancel Reminder 162-#
Set/Cancel Voice Guided Reminder 164-Follow VMS Prompts

Room Monitor
Room Monitor 1073-Station
Selective Port Access
Selective Port Access 69-Port Type-Port Offset
Self Ring Test
Self Ring Test 1057
System Administrator Mode
Enter System Administrator Mode 1#92-SA Password
Exit SA Mode 1#92
SA Command 1072
System Engineer Mode
Enter System Engineer Mode 1#91-SE Password
Exit SE Mode 00
Trunk Reservation
Reserve a Trunk 6
Cancel a Reserved Trunk 102
User Absent/Present
User Absent 104-User Password-0
User Present 104-User Password-1
User Password
Change User Password 114-Old User Password-New User Password
Voice Help
Voice Help 1090
Walk-In Class Of Service
To Walk-In 111-1-Your Extension Number-Your User Password
To Walk-Out 111-0

SA Commands
Call Phases
Feature Access
Feature Name Matured Matured
2-Way 3-way
Check-In 001 1072-901 Y Y
Check-Out 002 1072-902 Y Y
Guest Name 003 1072-903 Y Y
Guest Group 004 1072-904 Y Y
Guest-In/Out 005 1072-905 Y Y
Guest Title 006 1072-906 Y Y
Change Check-In Profile of Room 007 1072-907 Y Y
Change Occupancy Status of 008 1072-908 Y Y

Room / Extension
Change Clean status of Room/ 009 1072-909 Y Y

Extension
Room Shift 010 1072-910 Y Y
Reprint Check Out Report 011 1072-911 Y Y
Print Room Status Report 012 1072-912 Y Y
Print Alarm Status Report 013 1072-913 Y Y
Delete Checked Out calls 014 1072-914 Y Y
Set DND-Remote 015 1072-001 Y Y
Set Dynamic Lock settings – 016 1072-002 Y Y

Remote
Set Wakeup Alarms –Remote 017 1072-003 Y Y
Set Call Budget for Remote 018 1072-004 Y Y

Station
Assign/De-assign Mailbox to a 019 1072-005 Y Y

Station/Department Group -
Remote
Set Call Forward – Remote 020 1072-006 Y Y
Set Call forward for all stations- 021 1072-007 Y Y

Remote
Assign Station User Greeting 022 1072-008 Y Y

Message
Display & Acknowledge System 023 1072-009 Y Y

Activity
Display & Acknowledge System 024 1072-010 Y Y

Fault
Station Budget Display 025 1072-011 Y Y

Call Phases
Feature Access
2-Way 3-way
Change User password of a 026 1072-012 Y Y

Station
Lock/Unlock DKP’s Keypad 027 1072-013 Y Y
User Absent / Present 028 1072-014 Y Y
Change SA password 029 1072-015 Y Y
Change SA mode timer 030 1072-016 Y Y
Display Registered Mobile 031 1072-017 Y

Operator ID
Set Day/Night mode 032 1072-018 Y
Clear System Activity Log 036 1072-022 Y Y
Start/Abort SAL in Offline mode 037 1072-023 Y Y
Start/Abort SAL in Online mode 038 1072-024 Y Y
Clear Mailbox for a range of 039 - Y Y

extensions
Cancel Dial in Conference 040 1072-026 Y Y
Start/Abort SFL in Offline mode 041 1072-027 Y Y
Start/Abort SFL in Online mode 042 1072-028 Y Y
Display Port Parameters 043 1072-029 Y Y
Start/Abort Online OG Report 044 1072-101 Y Y
OG Print Filter: Print calls 045 1072-102 Y Y

originated from station/s
OG Print Filter: To Print calls 046 1072-103 Y Y

terminated from CO

terminated from BRI

terminated from T1E1

terminated from E&M

terminated from Mobile

terminated from SIP

Department Bill Group wise
OG Print Filter: To print calls made 053 1072-110 Y Y

on dates

Call Phases
Feature Access
2-Way 3-way
OG Print Filter: Print calls made 054 1072-111 Y Y

between time
OG Print Filter: To Print calls made 055 1072-112 Y Y

to numbers matching with the
numbers programmed in the
Number List
OG Print Filter: To Print calls of 056 1072-113 Y Y

Duration more than this time
OG Print Filter: To Print calls of 057 1072-114 Y Y

Units more than the units
programmed
OG Print Filter: To Print calls made 058 1072-115 Y Y

to account code
Assign default OG Print filters 059 1072-120 Y Y
Start/Abort offline report 060 1072-121 Y Y
Enable/ Disable OG Schedule 061 1072-122 Y Y

Reports
Program Time for Daily OG 062 1072-123 Y Y

Scheduled Reports
Program Day and Time for OG 063 1072-124 Y Y

Weekly Scheduled Reports
Program Date and Time for OG 064 1072-125 Y Y

Monthly Scheduled Reports
Delete calls made by stations 065 1072-131 Y Y
Delete calls made on/from date 066 1072-132 Y Y
To clear SMDR OG buffer 067 1072-133 Y Y
Start/Abort Internal calls Report 068 1072-136 Y Y
Set filter to print Internal calls 069 1072-137 Y Y

Report Station wise
Set filter to print internal calls with 070 1072-138 Y Y

duration greater than that given
here
Start/Abort Offline Internal Call 071 1072-141 Y Y

Report
Enable/ Disable Internal 072 1072-142 Y Y

Scheduled Reports
Program Time for Internal Daily 073 1072-143 Y Y

Scheduled Reports
Program Day and Time for Internal 074 1072-144 Y Y


Call Phases
Feature Access
2-Way 3-way
Program Date and Time for 075 1072-145 Y Y

Internal Monthly Scheduled
Reports
To Clear SMDR Internal Buffer 076 1072-150 Y Y
Start/Abort Online – IC Report 077 1072-151 Y Y
Set filter to print all Normal calls 078 1072-152 Y Y
Set filter to print all Built-In Auto 079 1072-153 Y Y

Attendant calls
Set filter to print all Unanswered 080 1072-154 Y Y

calls
Set filter to print all Built-In Auto 081 1072-155 Y Y

Attendant Unanswered calls
Set filter to print all DISA calls 082 1072-156 Y Y
Set filter to print all calls with 083 1072-157 Y Y

speech duration More than timer
Set filter to print all calls 084 1072-158 Y Y

unanswered for duration More
than timer
Set filter to print all calls kept on 085 1072-159 Y Y

hold for duration more than timer
Set filter to print all IC calls 086 1072-160 Y Y

received by the station
Set filter to print all IC calls recd. 087 1072-161 Y Y

On the CO

On the BRI

On the T1E1

On the E&M
Set filter to print all IC calls from 091 1072-165 Y Y

Mobile
Set filter to print calls received 092 1072-166 Y Y

from SIP

On/from date

At/from-to Time

From nos. matching the External
Number List

Call Phases
Feature Access
2-Way 3-way
Default IC Print filters 096 1072-170 Y Y
Abort/Start IC Offline Report 097 1072-171 Y Y
Enable/ Disable IC Scheduled 098 1072-172 Y Y

Report
Program Time for IC Daily 099 1072-173 Y Y

Scheduled Reports
Program Day and Time for IC 100 1072-174 Y Y

Program Date and Time for IC 101 1072-175 Y Y

Monthly Scheduled Reports
Clear SMDR-IC buffer 102 1072-180 Y Y
Start/Abort Printing of Online T1E1 103 1072-030 Y Y

Performance Report
Start/Abort Offline T1E1 104 1072-031 Y Y

Performance Report
Signal Strength of Mobile Port 105 1072-032 Y Y
Enable/Disable Call Cost Display 106 1072-181 Y Y

for a Station
Start/Abort Hotel/Motel Activity log 107 1072-176 Y Y

in Offline mode
Start/Abort Hotel/Motel Activity log 108 1072-177 Y Y

in Online mode
Display and Acknowledge Hotel/ 109 1072-178 Y Y

Motel Activity
Change Guest VIP Status of 110 1072-915 Y Y

Station
Change Phone Ringing Pattern of 111 1072-916 Y Y

Room
Print Reminder Status Report 112 1072-917 Y Y
Remote Reminder 113 1072-033 Y Y
Remote Voice Guided Alarm 114 1072-034 Y Y
Remote Voice Guided Reminder 115 1072-035 Y Y
Redirect Messages of a Station 116 1072-314 Y Y
To record Holiday message 117 Y Y
Enable/Disable Scheduled Alarm 118 1072-036 Y Y

Report
Program Time for Scheduled 119 1072-037 Y Y

Alarm Report

Call Phases
Feature Access
2-Way 3-way
Enable/Disable Scheduled 120 1072-038 Y Y

Reminder Report
Program Time for Scheduled 121 1072-039 Y Y

Reminder Report
Request Database 122 1072-040 Y Y

Synchronisation to PMS
Enable/Disable Scheduled Room 123 1072-041 Y Y

Status Report
Program Time for Schedule Room 124 1072-042 Y Y

Status Report
Enable/Disable Scheduled 125 1072-043 Y Y

Change of Room Clean Status
Program Time for Schedule 126 1072-044 Y Y

Change of Room Clean Status
To playback Holiday message 127 Y Y
Enable/Disable Internal Call Block 128 1072-045 Y Y

For Guest Phones
Software Version Revision Display 129 1072-191 Y Y

of CPU Card
User Definable Fields 130 1072-920 Y Y
OG Print Filter: To print calls 131 1072-183 Y Y

originated from CO

originated from BRI

originated from T1E1

originated from E&M

originated from Mobile

originated from SIP

originated from Magneto
Set Budget Type for CO Port 138 1072-192 Y Y
Program Budget Amount for CO 139 1072-193 Y Y
Program Free Minutes on CO 140 1072-194 Y Y
Set Budget Type for T1E1 Port 141 1072-195 Y Y
Program Budget Amount for T1E1 142 1072-196 Y Y

Call Phases
Feature Access
2-Way 3-way
Program Free Minutes on T1E1 143 1072-197 Y Y
Set Budget Type for BRI Port 144 1072-198 Y Y
Program Budget Amount for BRI 145 1072-199 Y Y
Program Free Minutes on BRI 146 1072-200 Y Y
Set Budget Type for SIP 147 1072-201 Y Y
Program Budget Amount for SIP 148 1072-202 Y Y
Program Free Minutes on SIP 149 1072-203 Y Y
Set Budget Type for MOBILE 150 1072-204 Y Y
Program Budget Amount for 151 1072-205 Y Y

MOBILE
Program Free Minutes on MOBILE 152 1072-206 Y Y
Call Budget Reset Mode for CO 153 1072-207 Y Y
Scheduled Date to Rest Call 154 1072-208 Y Y

Budget Statistics on CO
Reset consumed Budget Amount / 155 1072-209 Y Y

minutes on CO manually
Call Budget Reset Mode for T1E1 156 1072-210 Y Y

Budget Statistics on T1E1

minutes on T1E1 manually
Call Budget Reset Mode for BRI 159 1072-213 Y Y

Budget Statistics on BRI

minutes on BRI manually
Call Budget Reset Mode for SIP 162 1072-216 Y Y

Budget Statistics on SIP

minutes on SIP manually
Call Budget Reset Mode for 165 1072-219 Y Y

MOBILE

Budget Statistics on MOBILE

minutes on MOBILE manually

Call Phases
Feature Access
2-Way 3-way
Swap Ports for CO8/SLT8 - MAG8 168 1072-046 Y Y

Card
Reset ASR and ACD for Mobile 169 1072-222 Y Y

port
Set/Cancel Scheduled Call 170 1072-223 Y Y

Forward
To Broadcast Message 175 1072-301 Y Y

System Commands
Abbreviated Dialing
Program telephone number in a personal directory 1902-Personal Directory-Location Code-Number
Clear telephone number from a location in a PM 1902-Personal Directory-Location Code-#*

Group
Program a name in PM Group 1903-Personal Directory-Location Code-Name
Clear a name from location in a PM Group 1903-Personal Directory-Location Code-#*
To program a TAC Index to PM Group 1904-Personal Directory-Location Code-TAC Index
To assign PM Group to a SLT 1905-1-SLT-Personal Directory
To clear PM Group assigned to a SLT 1905-1-SLT-00
To assign PM Group to a DKP 1906-1-DKP-Personal Directory
To clear PM Group assigned to a DKP 1906-1-DKP-00
To assign PM Group to an ISDN Terminal 1907-1-ISDN-Personal Directory
To clear PM Group assigned to an ISDN Terminal 1907-1-ISDN-00
To clear a PM Group 1901-1-PM Group
To program a telephone number in global directory 1801-Location Code-Number
Clear telephone number from a location in global 1801-Location Code-#*

directory
Program a name in global directory 1802-Location Code-Name
Clear a name from a location in the global directory 1802-Location Code-#*
Program the OGTBG for global directory 1803-1-Location Code-OGTBG
Clear a location code in the global directory 1800-1-Location Code
Access Codes
Program the access code for features 3111-1-Feature Number-Access Code-#*
Assign default access code 3161-1-Feature Number
Account Codes
Program account name for the account code 4851-1-Account Code-Account Name
To enable Forced Account Code flag in Station 5602-1-Template Number-Feature Number-Flag

Advanced Feature Template Code
To enable Forced Account Code flag on a Trunk 5802-1-Template Number-Feature Number-Flag

Code

AC Impedance Test
To run the AC Impedance Test using the Accurate 3361-1-CO Port Number
Mode
To view the report 3362
Alarms
Program alarm/reminder ring timer 2201-Seconds
Program number of alarm attempts 2202-Number of Alarm Attempts
Program alarm attempt interval 2203-Alarm Attempt Interval
Enable/disable snooze alarm/reminder 2204-Code
To disable/enable Configurable Alarm Type 2208-Flag
To disable/enable Configurable Alarm Category 2209-Flag
5602-1-Template Number-12-Alarm Notification

To configure Alarm Notification Type
Type
To program Macros 1810-Macro Index-Number String
To clear a Macro 1810-Macro Index-#*
To program Access Code for Macro 3115-1-MAcro Index-Access Code
To clear the Access Code for the Macro 3115-1-Macro Index
To assign Destination Port for Hotel Reports 3701-Destination Port Code
Alternate Number Dialing
To assign an alternate group number to a location 1804-1-Memory Location Code-Alternate Number

code Group
To clear an alternated group number to a location 1804-1-Memory Location Code-000

code
Auto Answer
To set auto answer on DKP 1214-1-DKP-Auto Call Answer Mode
To set Auto Answer Timer 1215-1-DKP-Auto Call Answer Timer
To enable/disable Headset Connectivity flag 1213-1-DKP-Headset Connectivity Flag
Auto Call Back
To program Auto Call Back when Busy/No Reply in a 1302-1-COS Group-Feature Number-Code
CoS group
To assign the CoS group with Auto Call Back Busy/No 5502-1-Template Number-Feature Number-Code
Reply to a Station Basic Feature Template
To apply the Station Basic Feature Template with Auto 5503-1-SLT-Template Number
Call Back when Busy and No Reply, to an SLT

To apply the Station Basic Feature Template with Auto 5504-1-DKP-Template Number
Call Back when Busy and No Reply, to a DKP
To program the Auto Call Back Ring Timer 3801-Seconds
Auto Redial
Program time duration between two trials for low 1704-Seconds

priority
Program the number of trials for low priority 1705-Count
Program time duration between two trials for high 1706-Seconds

priority
Program the number of trials for high priority 1707-Count
Program auto redial RBT wait timer 1702-Seconds
Program auto redial ring timer 1703-Seconds
Automatic Number Translation
To enable Automatic Number Translation Flag on 6702-1-OG Trunk Bundle Number-Feature Number-
OGTB Code
To assign an Automatic Number Translation Table 6702-1-OG Trunk Bundle Number-Feature Number-
Code
To default the ANT Table 4750-1-ANT Table No.
Barge-In
To program Barge-In Timer 3803-Seconds
To enable / disable Call Back on CO Port 3310-1-CO-Code
To program Call Back Timer on CO Port 3311-1-CO-Call Back Timer
To select Call Back Mode for CO Port 3312 -1-CO-Call Back Mode
To program Call Back On for CO Port 3313-1-CO-Call Back on
To assign Call Back - Incoming Number List to a CO

3314-1 -CO-Incoming Number List
Port
To assign Call Back - Outgoing Number List to a CO

3315-1 -CO-Outgoing Number List
Port
To select Call Back From port for a CO Trunk port 3316-1-CO-Call Back From
To assign Call Back - OGTB Group to a CO port 3317-1-CO-OGTB Group
To enable/disable Call Back on Mobile Port 8010-1-Mobile-Code
To change Call Back Timer 8011-1-Mobile-Call Back Timer
To select the Call Back Mode 8012-1-Mobile-Call Back Mode

To program Call Back on for the Mobile Port 8013-1-Mobile-Call Back on
To assign a Call Back - Incoming Number List 8034-1-Mobile-Incoming Number List
To assign a Call Back - Outgoing Number List to a 8035-1-Mobile-Outgoing Number List

Mobile Port
To select Call Back From for a Mobile Port 8036-1-Mobile-Call Back From
To assign a Call Back - OGTB Group for a Mobile Port 8037-1-Mobile-OGTB Group
To enable/disable Call Back on BRI port 6242-1-BRI- Call Back Flag
To program Call Back Timer for BRI port 6243-1-BRI-Call Back Timer
To program Call Back Mode on BRI port 6244-1-BRI-Call Back Mode
To program Call Back On method for BRI port 6245-1-BRI-Call Back on selection
To assign Call Back - Incoming Number List to a BRI 6246-1-BRI-Incoming Number List
port
To assign a Call Back - Outgoing Number List to a BRI 6247-1-BRI-Outgoing Number List
port
To define Call Back From for a BRI port 6248-1-BRI-Call Back From
To assign a Call Back - OGTB Group for a BRI port 6249-1-BRI-OGTB Group
To enable/disable Call Back on T1E1 port 6176-1-T1E1- Call Back Flag
To program Call Back Timer for T1E1 port 6177-1-T1E1-Call Back Timer
To program Call Back Mode on T1E1 port 6178-1-T1E1-Call Back Mode
To program Call Back On method for T1E1 port 6179-1-T1E1-Call Back on selection
To assign Call Back - Incoming Number List to a T1E1 6180-1-T1E1-Incoming Number List
port
To assign a Call Back - Outgoing Number List to a 6147-1-T1E1-Outgoing Number List

T1E1 port
To define Call Back From for a T1E1 port 6148-1-T1E1-Call Back From
To assign a Call Back - OGTB Group for a T1E1 port 6149-1-T1E1-OGTB Group
Call Budget
To program default Call Budget Amount 3710-Preset Call Budget Amount
Call Budget on Trunk
To program Call Budget Type on CO 3301-1-CO-Budget Type
To program Call Budget Amount on CO 3302-1-CO-Budget Amount
To program Free minutes on CO 3303-1-CO-Minutes
To program Call Budget Reset Mode for CO 3304-1-CO-Call Budget Reset Mode
To program the Date for Scheduled Reset mode 3305-1-CO-Date
To program Call Budget - Number of Calls on CO 3309-1-CO-Number of Calls

To program Call Budget Type on Mobile trunk port 8019-1-Mobile-Budget Type
To program Call Budget Amount on Mobile trunk port 8020-1-Mobile-Budget Amount
To program Call Budget Minutes on Mobile trunk port 8021-1-Mobile-Minutes
To program Call Budget Number of Calls on Mobile 8033-1-Mobile-Number of Calls

trunk port
To program Call Budget Reset Mode for Mobile trunk 8022-1-Mobile-Call Budget Reset Mode
port
To program the Date for Scheduled Reset mode for 8023-1-Mobile-Date

Mobile trunk port
To program Call Budget Type on BRI 6214-1-BRI-Budget Type
To program Call Budget Amount on BRI 6215-1-BRI-Budget Amount
To program Call Budget Minutes on BRI 6216-1-BRI-Minutes
To program Call Budget Number of Calls on BRI 6205-1-BRI-Number of Calls
To program Call Budget Reset Mode for BRI 6217-1-BRI-Reset Mode
To program the Date for Scheduled Reset mode for 6218-1-BRI-Date

BRI
To program Call Budget Type on T1E1 Port 6122-1-T1E1-Budget Type
To program Call Budget Amount on T1E1 Port 6123-1-T1E1-Budget Amount
To program Call Budget Minutes on T1E1 Port 6124-1-T1E1-Minutes
To program Call Budget Number of Calls on T1E1 Port 6125-1-T1E1-Number of Calls
To program Call Budget Reset Mode for T1E1 6138-1-T1E1-Call Budget Reset Mode
To program the Date for Scheduled Reset mode for 6139-1-T1E1-Date

T1E1
To manually reset consumed Call Budget of CO trunks 3306-1-CO
To manually reset consumed Call Budget of Mobile 8024-1-Mobile

trunks
To manually reset consumed Call Budget of BRI 6219-1-BRI

trunks
To manually reset consumed Call Budget of T1E1 6140-1-T1E1

trunks
Call Cost Calculation (CCC)
Program the unit charge for first unit when 16 KHz 2600-Unit Charge for First Unit
metering is used
Program the unit charge for additional unit when 16 2601-Unit Charge for Additional Unit
KHz metering is used
To select service charge type 2602-Service Charge Type
To program service charge 2603-Service Charge
To set the service charge percentage when is selected 2604-Percentage

based on the percentage

Program duration of first unit for a pulse rate type on 2607-Pulse Rate Type-Time Zone-Duration of First
normal days Unit
Program duration of additional unit for a pulse rate 2608-Pulse Rate Type-Time Zone-Duration of
type on normal days Additional Unit
Load default normal pulse rate type 2606
Program cost of first unit for a pulse rate type on 2609-Pulse Rate Type-Time Zone-Cost of First Unit
normal days
Program cost of additional unit for a pulse rate type on 2610-Pulse Rate Type-Time Zone-Cost of
normal days Additional Unit
Program duration of first unit for a pulse rate type on 2612-Pulse Rate Type-Time Zone-Duration of First
holidays Unit
Program duration of additional unit for a pulse rate 2613-Pulse Rate Type-Time Zone-Duration of
type on holidays Additional Unit
Program cost of first unit for a pulse rate type on 2614-Pulse Rate Type-Time Zone-Cost of First Unit
holidays
Program cost of additional unit for a pulse rate type on 2615-Pulse Rate Type-Time Zone-Cost of
holidays Additional Unit
Load default Discounted pulse rate type 2611
To program an area code 2620-Area Code Index-Area Code-#*
To clear an area code for an Index 2620-Area Code Index-#*
To program Pulse Rate Type for Pulse Rate Option of 2621-Area Code Index-Pulse Rate Option-Pulse
area code index Rate Type
To program pulse rate for an area code 2621-Area Code Index-Pulse Rate Type
To delete the complete area code table 2622-Reverse SE Password
To program ignore digit count when SP_SP LCR is 2623-Area Code Index-Ignore Digit Count
used
Program weekly off day 2630-Day-Code
Program a Special date 2631-Special Date Index-Date-Month
To clear a Special date index 2632-Special Date Index
To program area code name 2633-Area Code Index-Name
To clear an area code name 2633-Area Code Index-#*
Call Duration Control (CDC)
To enable CDC for outgoing call 4202-1-CDC Table-Code
To enable CDC for incoming call 4203-1-CDC Table-Code
To enable CDC for internal call 4204-1-CDC Table-Code
To assign an external number to a table (allowed) 4205-1-CDC Table-Number List
To assign an external number to a table (denied) 4206-1-CDC Table-Number List
To assign CDC time to CDC table 4207-1-CDC Table-CDC Timer

To assign disconnection flag to a CDC table 4208-1-CDC Table-Disconnection Flag
To default a CDC Table 4201-1-CDC Table
Call Hold
To select Default Call Hold Type 5318-Hold Type
To change Global Hold Retrieval Timer 3805-Seconds
To change Exclusive Hold Retrieval Timer 3812-Minutes
Call Logs
To enable/disable Log Internal Calls in Missed Calls 5361-Code
To enable/disable Log Internal Calls in Answered

5362-Code
Calls
To enable/disable Log Internal Calls in Dialed Calls 5363-Code
Call Park
Set call park timer 3809-Minutes
Set call park release timer 3810-Minutes
Call Pick Up
To assign call pickup group for SLT 3901-1-SLT-Call Pickup Group
To de-assign call pickup group for SLT 3901-1-SLT-00
To assign call pickup group for DKP 3902-1-DKP-Call Pickup Group
To de-assign call pickup group for DKP 3902-1-DKP-00
To assign call pickup group for ISDN Terminal 3903-1-ISDN Terminal-Call Pickup Group
To de-assign call pickup group for ISDN Terminal 3903-1-ISDN Terminal-00
Call Progress Tones
To select the region 3501-Region Code
To change CPT-related Timers 3502-Seconds
To program the ring back tone timer 3503-Seconds
To program busy tone timer 3504-Seconds
To program error tone timer 3505-Seconds
To program confirmation tone timer 3506-Seconds
To program the programming confirmation tone timer 3509-Seconds
To program the programming error tone timer 3508-Seconds
To select a dial tone 5307-Flag

To demonstrate call progress tones 3541-Code
To set demonstration timer 3542-Seconds
Call Taping
To enable/disable beeps during conversation

5332-Code
recording
To program a number in a List 4302-List Number-Location Index-Number-#*
To clear a number from a Location Index 4302-List Number-Location Index-#*
To program Number List - Incoming in Station 5602-1-Template Number-Feature Number-Code

Advanced Feature Template
To enable Call Taping Internal Flag in a Station 5602-1-Template Number-Feature Number-Code

Advanced Feature Template
To enable Tape calls coming without CLI Flag in a 5602-1-Template Number-Feature Number-Code
Call Transfer
To program transfer while ringing timer 3806-Seconds
To program transfer-on busy timer 3807-Seconds
To program trunk-to-trunk inactivity timer 3808-Minutes
Call Line Identification and Presentation
To program parameter in a SLT hardware Template 5702-1-Template Number-Parameter Number-Code
To program the feature in a Station Advanced Feature 5602-1-Template Number-Feature Number-Code

Template
To program replacement ‘+’ string in CLI 5334-Code
To program replacement ‘+’ string in CLI 5335-Replacement String-#*
To removed replacement ‘+’ string in CLI 5335-#*
Class of Service (COS)
To enable a feature in a CoS group 1302-1-CoS Group-Feature Number-Code
To default a CoS group 1301-1-CoS Group
To assign CoS group to a template 5502-1-Template Number-Feature Number-Code
CLI Based Routing
To program the incoming number in a CLI table 4101-Index-Telephone Number-#*
To clear a number from the CLI table 4101-Index-#*
To program name of calling party 4102-Index-Name-#*

To clear a name of calling party 4102-Index-#*
To assign landing destination for the incoming number 4103-Index-Port Type-Port Number
To clear a location index in the CLI table 4104-1-Index
To program the clock sources 5341-Clock Source Index-Port Type-Port Offset
To select 'System Clock Synchronization' 5342-System Clock Synchronization
To select the 'PLL Locking Mode' 5343-PLL Locking Mode
Closed User Group (CUG)
To program route code 4502-1-Route Index-Route Code-#*
To clear a particular route code 4502-1-Route Index-#*
To assign OGTBG to a router code 4503-1-Route Index-OGTBG
To program strip digit count for a route 4504-1-Route Index-Strip Digit Count
To program self route flag for a route 4505-1-Route Index-Code
To program maximum dialed digits to select router for 4506-1-Route Index-Dialed Digit Count
a route code
To clear an entry in a routing table 4501-1-Route Index
Communication Ports
To set data transfer rate of a COM port 3201-Port-Speed
To set data bit of a COM port 3202-Port-Data Bits
To set parity of a COM port 3203-Port-Parity
To set stop bit of a COM port 3204-Port-Stop Bits
To set default parameters of a COM port 3210-Port
Changing Login Session Time Out of Jeeves 2118-Time
Logging Out users from Jeeves 2188
Configuring WAN Port
To select Connection Type 2116-Connection Type
To program PPPoE User ID 2117 - PPPoE User ID
To program PPPoE Password 2123- PPPoE Password
To program PPPoE Service Name 2124-PPPoE Service Name
To assign IP Address to WAN Port 2110-IP Address

To assign Subnet Mask to the WAN Port 2111-Subnet Mask
To program the Gateway IP Address 2112-Gateway IP Address
To select DNS Assignment Type 2115-DNS Address Assignment Type
To program Primary DNS Server Address 2113-Primary DNS Server Address
To program Secondary DNS Server Address 2114-Secondary DNS Server Address
To program Web Server Port (Listening port of

2121-Port
Jeeves)
To enable/disable Dynamic DNS 2125-Code
To program Dynamic DNS User ID 2126-DDNS User ID
To program Dynamic DNS User Password 2127-DDNS User Password
To program Dynamic DNS Host Name 2128 - DDNS Host Name
To program Dynamic DNS - Retry Trial Count 2129 - DDNS Retry Count
To Update Dynamic DNS IP Address binding (‘Update

2130
IP Address Now?’ flag)
To program Router's Public IP Address 2132-IP Address
To program STUN Server Address 2133-STUN Server Address
To program STUN Server Port 2134- Port
To set STUN Query Interval 2135 - Interval
To view IP Address of the Master Ethernet Port 2150
To view Subnet Mask of the Master Ethernet Port 2151
To view Gateway Address of the Master Ethernet Port 2152
To view current STUN query status 2159
To view Router’s Public IP Address fetched by STUN 2160
To view Dynamic DNS Status 2161
To view Ethernet link Status 2162
Configuring Extensions
To change the default value of a SLT Hardware 5702-1-Template Number-Parameter Number-Code

Template
To default SLT Hardware Template 5701-1-Template Number
To apply the Customized SLT Hardware Template to 5703-1-SLT-Template Number

SLT port
To program a feature in a Station Basic Feature 5502-1-Template Number-Feature Number-Code

Template
To default a Station Basic Feature Template 5501-1-Template Number
To assign a Station Basic Feature Template to SLT 5503-1-SLT-Template Number
To assign a Station Basic Feature Template to a DKP 5504-1-DKP-Template Number

To assign a Station Basic Feature Template to an 5507-1-ISDN-Template Number
ISDN Terminal port
To assign a Template to an E&M (Station) port 5505-1-E&M-Template Number
To assign a Station Basic Feature Template to a 5506-1-T1E1PRI-Template Number

T1E1PRI port
To assign a Station Basic Feature Template to a BRI 5509-1-BRI-Template Number

port
To assign a Station Basic Feature Template to a 5511-1-Magneto Port-Template Number

Magneto Port
To assign a Station Basic Feature Template to a 5513-1-Virtual Extension-Template Number

Virtual Extension.
To program a feature in a Station Advanced Feature 5602-1-Template Number-Feature Number-Code

Template
To default a Station Advanced Feature Template 5601-1-Template Number
To assign a Station Advanced Feature Template to 5603-1-SLT-Template Number

SLT
To assign a Station Advanced Feature Template to a 5604-1-DKP-Template Number

DKP
To assign a Station Advanced Feature Template to an 5607-1-ISDN-Template Number

ISDN Terminal port
To assign a Station Advanced Feature Template to an 5605-1-E&M-Template Number

E&M (Station) port
To assign a Station Advanced Feature Template to a 5606-1-T1E1PRI-Template Number

T1E1PRI port
To assign a Station Advanced Feature Template to a 5609-1-BRI-Template Number

BRI port
To assign a Station Advanced Feature Template to a 5613-1-Virtual Extension-Station Advance Feature

Virtual Extension. Template
To assign a Station Advanced Feature Template to a 5611-1-Magneto-Station Advanced Feature

Magneto Port. Template Number
Configuring SLT Extensions
To assign Hardware Slot-Port to an SLT 1101-SLT-Port Offset on the Card
To de-assign the hardware slot and the hardware port 1101-SLT-00-00

of an SLT port
To assign Access Code to an SLT Port 3101-1-SLT-Access Code-#*
To clear the Access Code to assigned to the SLT Port 3101-1-SLT-#*
To assign default Access Codes assigned to SLT 3151-1-SLT
To assign a Name to an SLT 5402-1-SLT-Name
To clear the name of the SLT 5402-1-SLT-#*
To assign an SLT Hardware Template to an SLT port 5703-1-SLT-Template Number

To assign a Station Basic Feature Template to an SLT 5503-1-SLT-Template Number
Port
To assign a Station Advanced Feature Template to an 5603-1-SLT-Template Number

SLT Port
To assign a Call Pick-Up Group to an SLT Port 3901-1-SLT-Call Pickup Group
To remove an SLT from a Call Pick Up group 3901-1-SLT-00
To assign a Personal Directory to an SLT Port 1905-1-SLT-Personal Directory
To clear the Personal Directory assigned to the SLT 1905-1-SLT-00
To define the Priority for an SLT Port 3911-1-SLT-Priority
Configuring DKP Extensions
To assign Hardware Slot-Port to a DKP 1102-DKP-Port Offset on the Card
To de-assign the hardware slot and the hardware port 1102-DKP-00-00

of a DKP port
To assign Access Code to a DKP Port 3102-1-DKP-Access Code-#*
To clear the Access Code to assigned to the DKP Port 3102-1-DKP-#*
To assign default Access Codes assigned to DKP 3152-1-DKP
To assign a Name to a DKP 5403-1-DKP-Name
To clear the name of the DKP 5403-1-DKP-#*
To assign a Station Basic Feature Template to a DKP 5504-1-DKP-Template Number

Port
To assign a Station Advanced Feature Template to a 5604-1-DKP-Template Number

DKP Port
To define the Call Capacity of a DKP Port 1201-1-DKP-Call Capacity
To assign a Key Map for a DKP Port 1221-1-DKP-DKP Key Template Number
To assign the DKP Port to a Call Pick-Up Group 3902-1-DKP-Call Pickup Group
To remove a DKP from a Call Pick Up group 3902-1-DKP-00
To assign a Personal Directory to a DKP Port 1906-1-DKP-Personal Directory
To clear the Personal Directory assigned to the DKP 1906-1-DKP-00
To define the Priority for a DKP Port 3912-1-DKP-Priority
To select CO CLIP Pattern for a DKP Port 1243-1-DKP-CO CLIP Pattern
To select Language for a DKP port 1224-1-DKP-Language
To select Ringer Mode for a DKP Port 1204-1-DKP-Ringer Mode
To select Ring Delay Timer for a DKP Port 1205-1-DKP-Ring Delay Timer
To set the Acknowledgement Mode 1206-1-DKP-Ringer Auto Acknowledge Mode
To set the Ringer Auto Acknowledge Timer 1207-1-DKP-Ringer Auto Acknowledge Timer
To select Destination for 'Play Ring ON' for a DKP Port 1220-1-DKP-Ring Destination

To select Ring Tune for a DKP Port 1202-1-DKP-Ring Tune
To set Ringer Volume for a DKP Port 1203-1-DKP-Ringer Volume
To set Handset Transmit (Tx) Volume Level for a DKP 1208-1-DKP-Handset MIC Volume Level
To set Handset Receive (Rx) Volume Level for a DKP 1209-1-DKP-Handset Speaker Volume Level
To set Right Handset Transmit (Tx) Volume Level for a

1225-1-DKP-Handset MIC Volume Level
DKP
To set Right Handset Receive (Rx) Volume Level for a

1226-1-DKP-Handset Speaker Volume Level
DKP
To set Headset Transmit (Tx) Transmit Volume Level 1222-1-DKP-Headset MIC Volume Level
for a DKP
To set Headset Receive (Rx) Volume Level for a DKP 1223-1-DKP-Headset Speaker Volume Level
To set Hands-free Transmit (Tx) Volume Level for a 1210-1-DKP-Speaker Phone MIC Volume Level
DKP
To set Hands-free Receive (Rx) Volume Level for a 1211-1-DKP-Speaker Phone Speaker Volume Level
DKP
To set Key Click Volume Level for a DKP 1212-1-DKP-Key Click Volume
To enable/disable DTMF Generation Flag for a DKP 1241-1-DKP-DTMF Generation
To set DTMF Transmit Level for a DKP 1218-1-DKP-DTMF Transmit Level
To enable/disable Headset Connectivity for a DKP 1213-1-DKP-Headset Connectivity Flag
To enable/disable Auto Answer for a DKP 1214-1-DKP-Auto Call Answer Mode
To set Auto Answer Timer (sec) for a DKP 1215-1-DKP-Auto Call Answer Timer
To set LCD Back Light Level of a DKP 1216-1-DKP-LCD Backlight Level
To change LCD Backlight OFF Timer of a DKP 1219-1-DKP-LCD Backlight OFF Timer
To change LCD Contrast Level of a DKP 1217-1-DKP-LCD Contrast Level
To assign Hardware Slot-Port to DSS connected to a 1103-DKP-DSS-Slot-Port Offset on the Card

DKP
To clear the hardware Slot-Port assigned to the DSS 1103-DKP-DSS-00-00

software port
Configuring ISDN Terminal
To assign BRI Software Port to an ISDN Terminal 7301-1-ISDN Terminal-BRI
To de-assign ISDN Terminal from a BRI Software Port 7301-1-ISDN Terminal-00
To assign Access Code to an ISDN Terminal 3103-1-ISDN Terminal-Access Code-#*
To clear the Access codes assigned to the ISDN 3103-1-ISDN Terminal-#*

Terminal
To assign default Station Access Codes to ISDN 3153-1-ISDN Terminal

Terminal
To assign a Name to an ISDN Terminal 5409-1-ISDN Terminal-Name-#*
To clear the Name of the ISDN Terminal 5409-1-ISDN Terminal-#*

To assign a Station Basic Feature Template to an 5507-1-ISDN Terminal-Template Number
ISDN Terminal
To assign a Station Advanced Feature Template to an 5607-1-ISDN Terminal-Template Number

ISDN Terminal
To assign a Personal Directory to an ISDN Terminal 1907-1-ISDN Terminal-Personal Directory
To clear the Personal Directory assigned to an ISDN 1907-1-ISDN Terminal-00

Terminal
To define the Priority for an ISDN Terminal 3913-1-ISDN Terminal-Priority
To assign an ISDN Terminal to a Call Pick-Up Group 3903-1-ISDN Terminal-Call Pick-Up Group
To remove an ISDN Terminal from a Call Pick-Up 3903-1-ISDN Terminal-00

Group
Configuring Region
To select Region 5301-Region Code
Configuring Operator
To assign a Time Table to an Operator 1602-1-Operator-Time Table
To define an Operator for Working Hours 1611-1-Operator-Routing Group
To define an Operator for Break Hours 1612-1-Operator-Routing Group
To define an Operator for Non-working Hours 1613-1-Operator-Routing Group
To default an Operator 1601-1-Operator
Configuring Trunks
To change the default value of a CO Hardware 5902-1-CO Hardware Template Number-Feature

Parameter in a Template Number-Code
To default CO Hardware Template 5901-1-CO Hardware Template
To assign a CO Hardware Template to a CO port 5903-1-CO-CO Hardware Template
To change the default value of a Trunk Feature 5802-1-Trunk Feature Template Number-Feature
Parameter in a Template Number-Code
To default Trunk Feature Templates 5801-1-Trunk Feature Template Number
To assign a Trunk Feature Template to a CO Trunk 5803-1-CO-Trunk Feature Template Number
To assign a Trunk Feature Template to a BRI Trunk 5804-1-BRI-Trunk Feature Template Number
To assign a Trunk Feature Template to an E&M Trunk 5805-1-E&M-Trunk Feature Template Number
To assign a Trunk Feature Template to a T1E1 Trunk 5806-1-T1E1-Trunk Feature Template Number
To assign a Trunk Feature Template to a Mobile Trunk 5807-1-Mobile-Trunk Feature Template Number
To change the default values of a Parameter in an 6002-1-Template Number-Parameter Number-Code

To restore default values to the Parameters of an E&M 6001-1-Template Number

Feature Template

To assign an E&M Feature Template to an E&M port 6003-1-E&M-Template Number
To assign E&M Feature Template to a T1E1 Port 6004-1-T1E1-Template Number
Configuring CO Trunks
To assign Hardware Slot and Port to the CO Port 1104-CO-Slot-Port offset on the Card
To clear the Hardware ID assigned to a CO Port 1104-CO-00-00
To program a Name for a CO Port 5404-1-CO-Name
To clear the Name of a CO trunk 5404-1-CO-#*
To enable/disable the CO Trunk 3307-1-CO-Flag
To assign a Trunk Feature Template to a CO Port 5803-1-CO-Trunk Feature Template Number
To assign a CO Hardware Template to a CO Port 5903-1-CO-Hardware Template Number
To assign a Cost Factor to a CO Port 3308-1-CO-Cost Factor
To enable / disable Call Back on CO Port 3310-1-CO-Code
To program Call Back Timer on CO Port 3311-1-CO-Call Back Timer
To select Call Back Mode for CO Port 3312 -1-CO-Call Back Mode
To program Call Back On for CO Port 3313-1-CO-Call Back on
To assign Call Back - Incoming Number List to a CO

3314-1 -CO-Incoming Number List
Port
To assign Call Back - Outgoing Number List to a CO

3315-1 -CO-Outgoing Number List
Port
To select Call Back From port for a CO Trunk port 3316-1-CO-Call Back From
To assign Call Back - OGTB Group to a CO port 3317-1-CO-OGTB Group
Configuring Mobile Trunks
To assign Hardware Slot and Port 1108-Mobile-Slot-Port Offset on the Card
To de-assign Hardware Slot and Port 1108-Mobile-00-00
To enable/disable the Mobile Port 8000-1-Mobile-Flag
To assign a Name to the Mobile Port 5408-1-Mobile-Name-#*
To clear the Name of the Mobile Port 5408-1-Mobile-#*
To select the frequency Band for the Mobile Port 8009-1-Mobile Port Number-Mobile Frequency
Band Code
To select the Preferred Network Mode for the Mobile 8038-1-Mobile Port Number-Preferred Network
Port Mode
To assign SIM PIN to the Mobile Port 8006-1-Mobile Port Number-SIM PIN-#*
To select Incoming Call mode on the Mobile port 8005-1-Mobile-Mode
To assign a Trunk Feature Template to the Mobile

5807-1-Mobile Port Number-Template Number
Port

To enable/disable CLIR on the Mobile Port 8031-1-Mobile Port Number-CLIR
To enable RCOC on Mobile Port 8030-1-Mobile-Code
To assign Cost Factor to a Mobile Port 8001-1-Mobile-Cost Factor
To set Network Registration Retry Count 8004-1-Mobile-N/w Registration Retry Count
To enable/disable Call Back on Mobile Port 8010-1-Mobile-Code
To change Call Back Timer 8011-1-Mobile-Call Back Timer
To select the Call Back Mode 8012-1-Mobile-Call Back Mode
To program Call Back on for the Mobile Port 8013-1-Mobile-Call Back on
To assign a Call Back - Incoming Number List 8034-1-Mobile-Incoming Number List
To assign a Call Back - Outgoing Number List to a 8035-1-Mobile-Outgoing Number List

Mobile Port
To select Call Back From for a Mobile Port 8036-1-Mobile-Call Back From
To assign a Call Back - OGTB Group for a Mobile Port 8037-1-Mobile-OGTB Group
To assign a Trusted Caller List Number to the Mobile 8013-1-Mobile-Number List

Port
To enable/disable Accept Anonymous Calls on Mobile 8029-1-Mobile-Code

Port
To program the Pause Timer on Mobile Port 8014-1-Mobile-Pause Timer
To program DTMF ON Time on Mobile Port 8015-1-Mobile-DTMF ON Time
To set DTMF Detection Mode on Mobile Port 8051-1-Mobile-Mode
To program DTMF Detection Duration on Mobile Port 8052-1-Mobile-Duration
To assign the Mobile Port to a 'Category' for Logical 8018-1-Mobile-Category

Partition
To enable/disable Gateway Application on the Mobile 8016-1-Mobile-Gateway Application Flag

Port
To program the DTMF String for the Gateway 8017-1-Mobile-DTMF String

Application on the Mobile Port
To enable/disable Debug on a Mobile Port 8028-1-Mobile Port Number-Debug Code
To program SIP Rx Gain on the Mobile Port 8041-1-Mobile-SIP Rx Gain
To program SIP Tx Gain on the Mobile Port 8042-1-Mobile-SIP Tx Gain
To select Network Selection mode 8007-1-Mobile Port Number-Code
To program the network operator codes in order of 8008-1-Mobile-Priority-Network Operator Code-#*

priority for Mobile port
To reset ASR and ACD calculation for Mobile Port 8032-1-Mobile-1
Configuring E&M Lines
To assign Hardware ID to an E&M Software Port 1105-E&M-Port offset on the Card
To clear the Hardware ID assigned to an E&M 1105-E&M-00-00

Software Port

To enable/disable an E&M Port 3321-1-E&M-Code
To program a Name for an E&M Port 5406-1-E&M-Name
To clear a name assigned to an E&M port 5406-1-E&M-#*
To assign an E&M Feature Template to an E&M Port 6003-1-E&M-Template Number
To assign a Trunk Feature Template to an E&M Port 5805-1-E&M Template Number
To assign a Station Basic Feature Template to an 5505-1-E&M-Template Number

E&M Port
To assign a Station Advanced Feature Template to an 5605-1-E&M-Template Number

E&M Port
To set a Priority Level for an E&M Port 3915-1-E&M-Priority
To assign a Cost Factor to an E&M port 3322-1-E&M-Cost Factor
Configuring Magneto Interface
To assign Slot-Port Assignment to a Magneto Port 1110-Magneto Trunk-Slot-Port offset on the Card
To enable/disable Magneto Port 6801-1-Magneto-Code
To program Access Code for Magneto Port 3107-1-Magneto-Access Code-#*
To clear the Access Code assigned to the Magneto 3107-1-Magneto-#*

port
To assign default Access Codes assigned to Magneto 3156-1-Magneto

Port
To assign a Name to the Magneto Port 5411-1-Magneto-Name-#*
To clear the Name of the Magneto Port 5411-1-Magneto-#*
To assign Priority level of a Magneto Port 3919-1-Magneto-Priority
To assign a Station Basic Feature Template to 5511-1-Magneto-Station Basic Feature Template

Magneto Port Number
To assign a Station Advanced Feature Template for 5611-1-Magneto-Station Advanced Feature

Magneto Port Template Number
To assign MRE Key to a DKP 1261-1-DKP Key Template-EON Terminal Type-Key

Number-Function Type-Function Number-Channel
To enable or disable the "Enable Silence Detection on 5357-Flag

Magneto?" flag
To set Magneto-Silence Detection Timer 5356-Silence Detection Timer
To set the Magneto Threshold Level 5358 - Magneto VAD Threshold Level
Configuring Virtual Extensions
To assign Access Code to a Virtual Extension 3109-1-Virtual Extension-Access Code-#*
To clear the access codes for all the Virtual Extension 3109-*
To assign a Name to a Virtual Extension 5414-1-Virtual Extension-#*
To clear the name of the Virtual Extension 2152-1-Virtual Extension-#*

To assign a Station Basic Feature Template to a 5613-1-Virtual Extension-Template Number
Virtual Extension
To assign a Station Advanced Feature Template to a 5613-1-Virtual Extension-Station Advanced

Virtual Extension Feature Template
To define the Priority for a Virtual Extension 3919-1-Virtual Extension-Priority
To program Landing Destination for Virtual Extension 3001-1-Virtual Extension-Port Type-Port Number
Configuring LCR
To program Time Zone at a Time Zone index 3402-Time Zone Index-Start Time-End Time
To program the Cost Factor (Service Provider 3403-Time Zone Index-CF1-CF2-CF3-CF4

preference) for the Time Zone
To default the Time Zone based LCR table 3401
To program a number at number Index in the Time- 3411-Number Index-Number String-#*

based LCR table
To clear a number string programmed at a number 3411-Number Index-#*

index
To program Cost Factor (Service Provider preference) 3412-Number Index-CF1-CF2-CF3-CF4

for the each Number
To default the Number-based LCR table 3410
To define Time Zone for Time+Number-based LCR 3421-Time Zone Index-Start Time-End Time
To program the number string for Time+Number- 3422-Number Index-Number String-#*

based LCR
To clear a number index from the Time+Number- 3422-Number Index-#*

based LCR Table
To program Cost Factor (Service Provider preference) 3423-Number Index-Time Zone Index-CF1-CF2-
for the each Number and Time Zone CF3-CF4
To default the Time and Number-based LCR table 3420
To program Area Code in the Area Code Table 2620-Area Code Index-Area Code-#*
To clear an Area Code in the Area Code Table 2620-Area Code Index-#*
To program Ignore Digit Count for an Area Code 2623-Area Code Index-Ignore Digit Count
To program a number at number Index in the Service 3441-Number Index-Number String-#*

Provider-based LCR table
To clear a number string programmed at a number 3441-Number Index-#*

index
To program Cost Factor (Service Provider preference) 3442-Number Index-CF1-CF2-CF3-CF4

for the each Number
To default the Number-based LCR table 3440
To select LCR type in OG Trunk Bundle Group 1404-1-OGTBG-LCR Type

Configuring Emergency Number Dialing
To program Emergency Numbers in the table 3116-Index-Emergency Number-#*
To program OG Trunk Bundle Group (OTBG) for an 3117-Index-OG Trunk Bundle Group
emergency number
Conflict Dialing
To program conflict dialing timer 5351-Seconds
To enable/disable conversation recording beeps 5332-Code
Customer Name
To program the customer name 5401-Customer Name-#*
To clear the customer name 5401-#*
Day Night Mode
To set the system in Day/Night mode 4801-Code
Daylight Saving Time (DST)
To program the DST Mode 1010-DST Mode
To program the start time of DST when manual 1011-Date-Month-Current Time-Advance Time
selected
To program the end time of DST when manual 1012-Date-Month-Current Time-Delay Time
selected
To program the DST Type 1013-DST Type
DDI Routing Table
To program the feature in a DDI Routing Table 6322-1-DDI Routing Table ID-Parameter Number-
Value
To default a DDI table 6321-1-DDI Routing Table ID
Default Settings
To load the default parameters 5302-Reverse SE Password
Department Call
To program the destination in the routing group 6502-1-Routing Group-Destination Index-Port

Type-Port Number

To program the time for which each station in the 6503-1-Routing Group-Destination Index-Ring
group should ring Timer
To program continuous or non-continuous ring for a 6504-1-Routing Group-Destination Index-Flag

station in the group
To program rotation method of a department group 6505-1-Routing Group-Routing Flag
To assign routing group to the department group 2001-1-Department Group Index-Routing Group
To clear the routing group assigned to the department 2001-1-Department Group Index-00
group
To program the access code for a department number 3113-1-Department Group Index-Access Code
To default the access code for a department group 3163-1-Department Group Index
To clear the access code for a department number 3113-1-Department Group Index-#*
Digest Authentication
To program User Password in the Digest 4119-Index-User Password

Authentication Table
Digital Key Phone-Soft Keys Programming
To assign a feature to a key in a Template 1261-1-DKP Key Template-EON Terminal Type-Key

Number-Function Type-Function Number
To assign a DKP Key template to a DKP 1221-1-DKP-DKP Key Template
To assign a Personalized Key map to a DKP 1221-1-DKP-0
To assign a function to a DKP key 1252-1-DKP-Key Number-Function Type-Function

Number
To set Built-In Auto Attendant inactivity timer 2411-Seconds
To set Built-In Auto Attendant answer wait timer 2412-Seconds
To set music timer 2413-Seconds
To set beeps timer 2414-Seconds
To set ring timer 2415-Seconds
To set busy tone timer 2416-Seconds
To set error tone timer 2417-Seconds
To enable/disable the Disconnect when Caller Doesn't 5338-Code

Dial a Digit' flag
To enable/disable the Disconnect Built-In Auto 5336-Code

Attendant call when Dialed Number Busy:
To enable/disable Disconnect Built-In Auto Attendant 5337-Code

Call, when Dialed Number does Not Reply

To program DISA idle state timer 2420-Seconds
To program DISA inactivity timer 2421-Minutes
To program Calling Number on DISA Application 4111- Index-Calling Number-#*
To program Port Type and Port Number for DISA 4112- Index-Port Type-Port Number
Application
Distinctive Rings
To demonstrate the ring 4003-Ring Pattern
To change demonstration timer 3542-Seconds
To make default all ring type situation 4001
To assign ring type to a situation 4002-Event-Ring Pattern
To default all DND Text Messages 1501
Emergency Detection and Reporting
To enable/disable emergency reporting 5110-Code
Emergency Dialing
To program emergency numbers 3116-Index-Emergency Number-#*
To program an OGTBG for emergency number 3117-Index-OG Trunk Bundle Group
Flexible Numbers
To program the access code for a SLT 3101-1-SLT-Access Code-#*
To clear the access code for a SLT 3101-1-SLT-#*
To assign the default flexible number of a SLT 3151-1-SLT
To program the access code for a DKP 3102-1-DKP-Access Code-#*
To clear the access code for a DKP 3102-1-DKP-#*
To assign the default flexible number of a DKP 3152-1-DKP
Floor Service
To program a routing group with member extensions 6502-1-Routing Group-Destination Index-Port

Type-Port Number
To program the Ring Timer for the routing group 6503-1-Routing Group-Destination Index-Ring
Timer

To program the Continuous Ring Flag for the routing 6504-1-Routing Group-Destination Index-Flag
group
To program the routing group in a Station Advanced 5602-1-Template Number-11-Routing Group

Feature Template
IC Reference Table
To default the IC Reference Table 6301-1-IC Reference Table Index
6302-1-IC Reference Table Index-feature Number-

To program IC Reference Table Index
Value
Interrupt Request
To set interrupt request timer 3802-Seconds
ISDN BRI
To assign a name to the BRI port 5405-1-BRI-Name
To enable/disable BRI port 6201-1-BRI-Port Status
To assigning a service provider to a BRI port 6202-1-BRI-SP
To program the BRI ISDN switch variant of the BRI 6203-1-BRI-BRI ISDN Switch Variant
port
To program orientation type for a BRI port 6204-1-BRI-Orientation Type
To program the type of interface companding for the 6205-1-BRI-Companding

BRI
To program Power Feed 6227-1- BRI Port- Feed Power to Port
To program the idle code for the BRI 6207-1-BRI-Idle Code
To set overlap receiving timer for BRI 6208-1-BRI-Timer
To program Pause Timer 6209-1-BRI- Pause Timer
To program DTMF ON Time 6210-1-BRI-DTMF ON Time
To program DTMF Inter digit Pause Timer 6211-1-BRI- DTMF Inter digit Pause Time
To get appropriate debug information 6291-1-BRI-Level-Code
To program a caller TON for the BRI 6221-1-BRI-Caller TON
To program a caller NPI for the BRI 6222-1-BRI-Caller NPI
To program a called party TON for the BRI 6223-1-BRI-Called Party TON
To program a called party NPI for the BRI 6224-1-BRI-Called Party NPI
To enable/disable Call Back on BRI port 6242-1-BRI- Call Back Flag
To program Call Back Timer for BRI port 6243-1-BRI-Call Back Timer
To program Call Back Mode on BRI port 6244-1-BRI-Call Back Mode
To program Call Back On method for BRI port 6245-1-BRI-Call Back on selection

To assign Call Back - Incoming Number List to a BRI 6246-1-BRI-Incoming Number List
port
To assign a Call Back - Outgoing Number List to a BRI 6247-1-BRI-Outgoing Number List
port
To define Call Back From for a BRI port 6248-1-BRI-Call Back From
To assign a Call Back - OGTB Group for a BRI port 6249-1-BRI-OGTB Group
To program an OG reference ID-WH for the BRI 6231-1-BRI-OG Reference ID
To program an OG reference ID-BH for the BRI 6241-1-BRI-OG Reference ID
To program an OG reference ID-NH for the BRI 6242-1-BRI-OG Reference ID
To program an IC reference ID-WH on the BRI 6232-1-BRI-IC Reference ID
To program an IC reference ID-BH on the BRI 6233-1-BRI-IC Reference ID
To program an IC reference ID-NH on the BRI 6234-1-BRI-IC Reference ID
To program number of channels reserved for data 6235-1-BRI-Channel Count

transmission
To reserve the number of channels for OG calls on 6236-1-BRI-Channel Count

BRI
To reserve the number of channels for IC calls on a 6237-1-BRI-Channel Count

BRI
To select TEI Negotiation on a BRI port 6238-1-BRI-TEI Negotiation Mode
To program TEI Negotiation value when programmed 6239-1-BRI-TEI Value

as Fixed
To program access code for ISDN terminal 3103-1-ISDN Terminal-Access Code-#*
To default access code for ISDN terminal 3153-1-ISDN Terminal
To assign a BRI software port to ISDN terminal 7301-1-ISDN Terminal-BRI
To de-assign a BRI software port to ISDN terminal 7301-1-ISDN Terminal-00
To assign personal directory to an ISDN terminal 1907-1-ISDN Terminal-Personal Directory
To program Gateway Application-Answer Signaling 6206-1-BRI-Gateway Application-Answer Signaling

Flag Flag
To program "Gateway Application-Answer Signaling 6212-1-BRI-Gateway Application-Answer Signaling

DTMF String". DTMF String
To program Layer 1 Mode for the BRI Port 6225-1-BRI-Layer 1 Mode
Key Board Macro
To create a macro 1810-Macro Index-Number String
To clear a macro 1810-Macro Index-#*
To program access codes for the macro 3115-1-Macro Index-Access Code
To clear access code 3115-1-Macro Index
To program to default access code for the macro 3165-1-Macro Index

Logical Partition
To define call permission across and between 5317-Category-Category-Flag

Categories
Live Call Screening
To program LCS timer 3811-Seconds
Message Wait
To program Message Wait Ring Count 4403-Message Wait Ring Count
To program the message wait ring timer 4404-Message Wait Ring Timer
To program the message wait ring interval timer 4405-Message Wait Ring Interval Timer
Number List
To program a number in a Number List 4302-List Number-Location Index-Number-#*
To clear the number programmed in a Number List 4302-List Number-Location Index-#*
To default a Number List 4301-1-List Number
OFF-Hook Alert
To enable/disable OFF-Hook Alert to Operator 5333-Code
OG Reference Table
To program an OG reference table 6312-1-OG Reference Table Index-Parameter

Number-Value
To default an OG reference table 6311-1-OG Reference Table Index
OG Trunk Bundle
To program the feature in OG trunk bundle 6702-1-OG Trunk Bundle Number-Port Type-Port
Number-Code
To set default values for OG trunk bundle 6701-1-OG Trunk Bundle Number
To make default OGTBG 1401-1-OGTBG Number
To set OG trunk bundle 1402-1-OGTBG No-Destination Index-OG Trunk

Bundle
To set rotation flag 1403-1-OGTBG Number-Flag
To program the desirable access code for a trunk 3112-1-OGTBG Index-Access Code-#*
access index

To clear the access codes for an OGTBG index 3112-1-OGTBG Index-#*
To assign default access code for a OGTBG index 3162-1-OGTBG Index
Paging
To program a DKP in a Page Zone 2302-1-Page Zone-Member-DKP
To include/exclude a member DKP in/from a Page 2303-1-Page Zone-Member-Flag

Zone
Presence
To enable/disable 'Display Presence Status during 5320-Flag

Call on DKP'
Peer-to-Peer Calling
To program the number string in peer-to-peer table 7801-1-Index-Number String-#*
To clear the number string for an index 7801-1-Index-#*
To program the destination address in peer-to-peer 7802-1-Index-Destination Address-#*

table
To clear the destination address for an Index 7802-1-Index-#*
To program the name in peer-to-peer table 7803-1-Index-Name-#*
To clear the name for an index 7803-1-Index-#*
Priority
To assign priority to SLT 3911-1-SLT-Priority
To assign priority to DKP 3912-1-DKP-Priority
To assign priority to ISDN Terminal 3913-1-ISDN Terminal -Priority
To assign priority to T1E1 3914-1-T1E1-Priority
To assign priority to E&M 3915-1-E&M-Priority
To assign priority to BRI 3916-1-BRI-Priority
To assign priority to Magneto 3919-1-Magneto-Priority
RCOC
To enable RCOC on T1E1 Trunk 6145-1-T1E1-Code
To enable RCOC on BRI Trunk 6220-1-BRI-Code
To enable RCOC on Mobile Port 8030-1-Mobile-Code
To change the RCOC Record Delete Timer 3521- Minutes
To change the Ring Back Tone Timer 3503-Seconds

Real Time Clock
To program the date format 1000-Date Format
To set the date 1001-Date-Month-Year
To select the Country for the time zone 1002-Time Zone
To set time 1003-Hours-Minutes-Seconds
Reminder
Program alarm/reminder ring timer 2201-Seconds
Program number of alarm attempts 2202-Number of Alarm Attempts
Program alarm attempt interval 2203-Alarm Attempt Interval
Assign destination port to the Hotel/Motel report 3701-Flag
Enable/disable snooze alarm/reminder 2204-Code
Routing Group
To program the destination in the routing group 6502-1-Routing Group-Destination Index-Port

Type-Port Number
To program the time for which each station in the 6503-1-Routing Group-Member Index-Ring Timer
group should ring
To program continuous or non-continuous ring for a 6504-1-Routing Group-Member Index-Flag

station in the group
To program rotation method of a department group 6505-1-Routing Group-Rotation Method
To default a routing group 6501-1-Routing Group
To clear the destination in a routing group 6510-1-Routing Group
Software Port and Hardware ID
To assign hardware ID to a SLT software port 1101-SLT-Slot-Port offset on the Card
To clear the hardware ID assigned to a SLT software

1101-SLT-00-00
port
To assign hardware ID to a DKP software port 1102-DKP-Slot-Port offset on the Card
To clear the hardware ID assigned to a DKP software

1102-DKP-00-00
port
To assign hardware ID to DSS software port 1103-DKP-DSS-Slot-Port offset on the Card
To clear the hardware ID assigned to DSS software

1103-DKP-DSS-00-00
port
To assign hardware ID to a CO software port 1104-CO-Slot-Port offset on the Card
To clear the hardware ID assigned to a CO software

1104-CO-00-00
port
To assign hardware ID to an E&M software port 1105-E&M-Slot-Port offset on the Card

To clear the hardware ID assigned to an E&M
1105-E&M-00-00
software port
To assign hardware ID to a BRI software port 1106-BRI-Slot-Port offset on the Card
To clear the hardware ID assigned to a BRI software

1106-BRI-00-00
port
To assign hardware ID to a T1E1 software port 1107-T1E1-Slot-Port offset on the Card
To clear the hardware ID assigned to a T1E1 software

1107-T1E1-00-00
port
To assign hardware ID to a mobile software port 1108-Mobile-Slot-Port offset on the Card
To clear the hardware ID assigned to mobile software

1108-Mobile-00-00
port
To assign hardware ID to a Magneto software port 1110-Magneto-Slot-Port offset of the card
To de-assign hardware ID to a Magneto software port 1110-Magneto-00-00
Static Routing Table
To program Destination Address in Static Routing

7811-1-Index-Destination Address
Table
To program Subnet Mask in the Static Routing Table 7812-1-Index-Subnet Mask
To program Gateway Address in the Static Routing

7813-1-Index-Gateway Address
Table
To clear specific entries of the index of Static Routing

7814-1-Index
Table
Station Message Detail Recording-Online
To assign destination port for Online SMDR-IC Call 2930-Code

Record
To assign the IP Address to the Ethernet Port 2932-IP Address
To assign the IP Port 2933-IP Port
To assign a destination port for Online SMDR-Internal 2830-Code

call record
To assign destination port for Online SMDR-OG Call 2730-Code

Record
To program column position for serial number 8200-Column Position
To program field length for serial number 8201-Field Length
To program alignment for serial number 8202-Alignment

To program fill character for serial number 8203-Fill Character
To program reset for serial number 8204-Reset
To program column position for increment counter 8205-Column Position
To program reset for increment counter 8206-Reset
To program column position for property code 8207-Column Position
To program field length for property code 8208-Field Length
To program property code string for property code 8209-Property Code String
To program column position for station number 8210-Column Position
To program field length for station number 8211-Field Length
To program alignment for station number 8212-Alignment
To program fill character for station number 8213-Fill Character
To program column position for trunk number 8214-Column Position
To program format type for trunk number 8215-Format Type
To program column position for date field 8216-Column Position
To program field length for date field 8217-Field Length
To program alignment for date field 8218-Alignment
To program fill character for date field 8219-Fill Character
To program date format for date field 8220-Date Format
To program date fill flag for date field 8257-Date Fill Flag
To program column position for time field 8222-Column Position
To program field length for time field 8223-Field Length
To program alignment for time field 8224-Alignment
To program fill character for time field 8225-Fill Character
To program time format for time field 8226-Time Format
To program time fill flag for time field 8258-Time Fill Flag
To program column position for answer duration field 8227-Column Position
To program field length for answer duration field 8228-Field Length
To program alignment for answer duration field 8229-Alignment
To program fill character for answer duration field 8230-Fill Character
To enable/disable the Filler char. flag for Answer 8259-Filler Character Flag for Answer Duration
Duration
To program duration unit for answer duration field 8231-Duration Unit
To program column position for hold duration field 8232-Column Position
To program field length for hold duration field 8233-Field Length
To program alignment for hold duration field 8234-Alignment
To program fill character for hold duration field 8235-Fill Character

To enable/disable the Filler character flag for Hold 8260-Filler Character Flag for Hold Duration
Duration
To program column position for speech duration field 8237-Column Position
To program field length for speech duration field 8238-Field Length
To program alignment for speech duration field 8239-Alignment
To program fill character for speech duration field 8240-Fill Character
To enable/disable the Filler char. flag for Speech 8261-Filler Character Flag for Speech Duration
Duration
To program column position for called number field 8242-Column Position
To program field length for called number field 8243-Field Length
To program alignment for called number field 8244-Alignment
To program number format for called number field 8245-Number Format
To program column position for calling number field 8246-Column Position
To program field length for calling number field 8247-Field Length
To program alignment for calling number field 8248-Alignment
To program number format for calling number field 8249-Number Format
To program column position for Digits dialed in Built-in 8250-Column Position

Auto Attendant
To program field length for Digits dialed in Built-in Auto 8251-Field Length
Attendant
To program alignment for Digits dialed in Built-in Auto 8252-Alignment

Attendant
To program column position for remarks field 8253-Column Position
To program field length for remarks field 8254-Field Length
To program alignment for remarks field 8255-Alignment
To assign default IC SMDR format 8256
Station Message Detail Recording-Posting
To enable/disable storage of SMDR OG Calls 2701-Code
To assign Destination Port for SMDR-OG Posting 8330-Code
To program SMDR OG Posting Protocol 8301-SMDR OG Posting Protocol
To set ENQ no response timer 8302-ENQ No Response Timer
To set ENQ no response retry count 8303-ENQ Retry Count
To set ENQ no response Retry Timer 8304-ENQ No Response Retry Timer
To set ENQ Retry Count 8305-ENQ Retry Count
To set ENQ Retry Time 8306-ENQ Retry Time
To set SMDR-OG posting Data response to Data 8307-Response to Data Timeout

Timeout

To set Data Transfer Retry Count (No Response) 8308-Data Transfer Retry Count
To set Data Transfer Retry Time (No Response) 8309-Data Transfer Retry Time
To set Data Transfer Retry Count (Negative 8310-Data Transfer Retry Count
Response)
To set Data Transfer Retry Time (Negative Response) 8311-Data Transfer Retry Time
To enable/disable ENQUIRE Signal 8312-ENQUIRE Signal
To set the ENQUIRE character 8313-ENQUIRE
To set the ACK Character 8314-Set ACK Character
To program the NAK Character 8315-Set NAK Character
To set the Start of Packet string 8316-Start of Packet
To program the End of Packet string 8317-End of Packet
To enable/disable BCC Flag 8318-BCC Flag
To set SMDR-OG Posting Parameters to Default 8300

Value
To program column position for serial number 8100-Column Position
To program field length for serial number 8101-Field Length
To program alignment for serial number 8102-Alignment
To program fill character for serial number 8103-Fill Character
To program reset for serial number 8104-Reset
To program column position for increment counter 8105-Column Position
To program reset for increment counter 8106-Reset
To program starting character for increment counter 8174-Starting Character
To program column position for property code 8107-Column Position
To program field length for property code 8108-Field Length
To program property code string for property code 8109-Property Code String
To program column position for station number 8110-Column Position
To program field length for station number 8111-Field Length
To program alignment for station number 8112-Alignment
To program fill character for station number 8113-Fill Character
To program column position for trunk number 8114-Column Position
To program format type for trunk number 8115-Format Type
To program column position for date field 8116-Column Position
To program field length for date field 8117-Field Length
To program alignment for date field 8118-Alignment
To program fill character for date field 8119-Fill Character
To program date format for date field 8120-Date Format

To program date fill flag for date field 8170-Date Fill Flag
To program column position for time field 8122-Column Position
To program field length for time field 8123-Field Length
To program alignment for time field 8124-Alignment
To program fill character for time field 8125-Fill Character
To program time format for time field 8126-Time Format
To program time fill flag for time field 8171-Time Fill Flag
To program column position for duration field 8127-Column Position
To program field length for duration field 8128-Field Length
To program alignment for duration field 8129-Alignment
To program fill character for duration field 8130-Fill Character
To program duration unit for duration field 8131-Duration Unit
To program duration fill flag for duration field 8172-Duration Fill Flag
To program column position for units field 8132-Column Position
To program field length for units field 8133-Field Length
To program alignment for units field 8134-Alignment
To program fill character for units field 8135-Fill Character
To program column position for amount field 8136-Column Position
To program field length for amount field 8137-Field Length
To program alignment for amounts field 8138-Alignment
To program fill character for amounts field 8139-Fill Character
To program amount format for amount field 8140-Amount Format
To program amount fill flag for amount field 8173-Amount Fill Flag
To program column position for currency symbol field 8141-Column Position
To program field length for currency symbol field 8142-Field Length
To program alignment for currency symbol field 8143-Alignment
To program fill character for currency symbol field 8144-Fill Character
To program symbol for currency symbol field 8145-Character1.......Character8
To program column position for call type indicator field 8146-Column Position
To program field length for call type indicator field 8147-Field Length
To program alignment for call type indicator field 8148-Alignment
To program number string for call type indicator field 8149-Number Index-1-Number String
To program text string for call type indicator field 8149-Number Index-2-Text String
To program column position for called location field 8150-Column Position
To program field length for called location field 8151-Field Length

To program alignment for called location field 8152-Alignment
To program column position for called number field 8154-Column Position
To program field length for called number field 8155-Field Length
To program alignment for called number field 8156-Alignment
To program number format for called number field 8157-Number Format
To program column position for account code field 8158-Column Position
To program field length for account code field 8159-Field Length
To program alignment for account code field 8160-Alignment
To program fill character for account code field 8161-Fill Character
To program prefix string (ac01) 8165-Code
To program column position for remarks field 8166-Column Position
To program field length for remarks field 8167-Field Length
To program alignment for remarks field 8168-Alignment
To assign default CDR format 8169
To program a country code 8321-Index-Country Code-#*
To program a location in country code 8322-Index-Country Code-#*
To start/stop SMDR Posting Process 8333-Code
To program the destination IP Address for Posting 8331-Address-Address-Address-Address

SMDR-OG Call Record
To program the destination IP Port for Posting SMDR- 8332-Destination IP Port

OG call Record
To program Listening Port of ETERNITY GENX 8334-Listening Port
Station Message Detail Recording-Report
To assign a destination port for SMDR-IC Report 2931-Code
To assign destination port for Online SMDR-Internal 2831-Code

Report
To assign a destination port for SMDR-OG Report 2731-Code
Station Message Detail Recording-Storage
To set SMDR storage mode (IC) 2901-Storage Flag

Store Normal Calls 2902-Flag
Store Built-in Auto Attendant Calls 2903-Flag
Store Unanswered Calls 2904-Flag
Store Unanswered Built-in Auto Attendant Calls 2905-Flag
Store DISA Calls 2906-Flag
Store Calls-Speech Duration More than 2907-Seconds
Store Calls-Unanswered Duration for more than 2908-Seconds
Store Calls-Hold Duration for more than 2909-Seconds
To default the incoming call storage filters 2915
To set SMDR storage mode (Internal) 2801-Storage Flag
Store Calls-Speech duration 2802-Seconds
To default the internal call storage filters 2815
To set SMDR storage flag (OG) 2701-Storage Flag
To assign a Number List containing numbers for call 2702-Number List

storage
To set the filter of call duration 2703-Seconds
To set the filter for call Units 2704-Unit
To default the outgoing call storage filters 2715
To set the Call Toggle flag 2716-Toggle Flag
To set the originating flag 2717-Originating Flag
System Activity Log
To program the storage flag to enable/disable SAL 6401-Storage Flag
To assign a port for online printing of SAL 6402-Port
To assign a port for offline printing of SAL 6403-Port
To default system activity log parameters for SAL 6410
System Debug
To program a port for debug 2103-Port
To start/stop debug for required process 2104-Value
2105-Port Type-Port Number Start-Port Number

To start/stop state debug
End-Flag

To enable ETERNITY GENX Host debug 2181-1-Code
To set IP Address for ARM Debug 2182-IP Address
To set Port for ARM Debug 2183-Port
To enable ETERNITY GENX DSP Para 1 Debug 2184-1-Code
To enable ETERNITY GENX DSP Para 2 Debug 2184-2-Code
To set IP Address for DSP Debug 2185-IP Address
To set IP Port for DSP Debug 2186-Port
To enable or disable PCM capture - Debug 2172-Slot Number-Hardware Port Offset- Code
To program the IP Address of the Syslog Server 2178-Syslog Server IP Address
To program Port number on which ETERNITY GENX

2179-Syslog Server's Listening Port
shall send debug to Syslog Server
To initiate the debug of IO operations 2199-Slot Number-1-Port Number-Code
System Fault Log
To enable/disable storage of faults 6451-Flag
To assign the port to system fault log-online printing 6452-Port
To assign a port to system fault log-offline printing 6453-Port
System Parameters
To assign Station Type to SLT 3921-1-SLT-Station Type
To assign Station Type to DKP 3922-1-DKP-Station Type
To assign Station Type to ISDN Terminal 3923-1-ISDN Terminal-Station Type
To restart the Card installed in Universal Slot of the 2187-Slot No

ETERNITY GENX
To restart the system 5305-Reverse SE Password
To load default values of all system timers 5303
To know the software version/revision of the system 5304
To enable/disable form feed 5321-Code
To select a Language for SE, SA and Front Desk User 5319-Language

Web Interface
To set the A-law/M-law 5322-Type
To monitor a port 7902-Slot-LED Number-Port
To enable/disable the watch dog 5309-Flag

To enable/disable feature tone 5312-Feature Tone Flag
To program the end of dialing digit 5313-End of Dialing Digit
To program Caller ID digits to be considered as call 5314-Minimum Caller ID Digits

from Public Network
To program Listening Port of Web Server 2121-Port
To select Hotel/Enterprise Mode 5315-Code
To program default Call Budget Amount 3710-Preset Call Budget Amount
To display 'MAC Address' on DKP 2122
To program Call Proceeding Tone Type for multi-stage 5311-Call Proceeding Tone Type
dialing
System Security
To change SE password 5306-New SE Password
To assign a new password to the SA 5310-New SA Password
T1 Maintenance
To enable/disable T1 FDL on a T1E1 port 6164-1-T1E1-T1 FDL
To program the T1 FDL protocol for a T1E1 port 6165-1-T1E1-T1 FDL Protocol
T1 RBS Parameters
To program the T1 line signaling variants for the T1E1 6181-1-T1E1-T1 Line Signaling Variants
port
To program T1 wink timer for T1E1 6182-1-T1E1-Wink Timer
To program the T1 wink wait timer for T1E1 6183-1-T1E1-Wink Wait Timer
To program the T1 wait wink timer for T1E1 6184-1-T1E1-Wait Wink Timer
To program the T1 delay duration for T1E1 6185-1-T1E1-Delay Duration
To program the T1 Start Delay timer for T1E1 6186-1-T1E1-Start Delay timer
To program T1 register signaling variant for the T1E1 6161-1-T1E1-T1 Register Signaling Variant
port
To select the Inbound ANI/DNIS Format 6166-1-T1E1-Code
To assign the Inbound Delimiter (?) character 6167-1-T1E1-Character
To select the Outbound ANI/DNIS Format 6168-1-T1E1-Code
To assign the Outbound Delimiter (?) character 6169-1-T1E1-Character
To program digital pulse dial ratio for the T1E1 port 6163-1-T1E1-Code
T1E1 Trunks
To enable/disable the port 6101-1-T1E1-Port Status

To assign a name to the T1E1 port 5407-1-T1E1-Name
To select the line Carrier 6108-1-T1E1-Carrier
To program the line coding mechanism for the E1 port 6103-1-T1E1-Line Coding
To program the line coding mechanism for the T1 port 6195-1-T1E1-Line Coding
To program the framing mode for the E1 port 6104-1-T1E1-Framing
To program the framing mode for the T1 port 6196-1-T1E1-Framing
To program Signaling type/line type for E1 6105-1-T1E1-Line Type
To program Signaling type/line type for T1 6151-1-T1E1-Line Type
To program interface companding of a T1E1 6108-1-T1E1-Interface Companding
To program auto receive equalization mode 6110-1-T1E1-Mode
To program the receive equalization parameters of a 6111-1-T1E1-Receive Equalization Parameters

T1E1
To program glare option for the T1E1 port 6112-1-T1E1-Glare Option
To program the Category for the T1E1 port 6121-1-T1E1-Category
To program the idle code of a T1E1 6113-1-T1E1-Idle Code
To program overlap receiving timer 6114-1-T1E1-Timer
To program Pause Timer 6109-1-T1E1-Pause Timer
To program DTMF ON Time 6117-1-T1E1-DTMF ON Time
To program DTMF Inter Digit Pause Timer 6118-1-T1E1-DTMF Inter Digit Pause Timer
To program orientation type for the T1E1 port 6106-1-T1E1-Orientation Type
To program source TON for a T1E1 6126-1-T1E1-Source TON
To program source NPI for T1E1 6127-1-T1E1-Source NPI
To program OG destination party TON or a T1E1 6128-1-T1E1-Destination TON
To program OG destination party NPI for T1E1 6129-1-T1E1-Destination NPI
To program to select whether the inband tones should 6130-1-T1E1-Flag

be feed on T1E1-NT before sending DISCONNECT
message?
To program the dial tone flag for T1E1 6115-1-T1E1-Flag
To program the routing tone flag for T1E1 6116-1-T1E1-Flag
To reserved channels for data transmission on T1E1 6135-1-T1E1-Channel Count (Data)
To program number of channels reserved for OG 6136-1-T1E1-Channel Count (OG)
To program number of channels reserved for IC 6137-1-T1E1-Channel Count (IC)
To assign OG reference ID to T1E1 6131-1-T1E1-OG Reference ID
To assign IC reference ID for working hour 6132-1-T1E1-IC Reference ID
To assign IC reference ID for break hour 6133-1-T1E1-IC Reference ID
To assign IC reference ID for non-working hour 6134-1-T1E1-IC Reference ID
To assign a Cost Factor to the T1E1 Trunk 6102-1-T1E1-Cost Factor

To active/deactivate near end loopback for T1E1 6141-1-T1E1-Loopback
To start/stop far end loopback test for T1E1 6142-1-T1E1-Code
To assign the port to performance report for online 6143-Port

printing
To assign the port to performance report for offline 6144-Port

printing
To program E1 line signaling variant for the E1 6152-1-T1E1-E1 Line Signaling Variant
To program E1 register signaling variant for the E1 6153-1-T1E1-E1 Register Signaling Variant
To enable/disable E1 auto alarm for the T1E1 E1 6154-1-T1E1-Flag

signaling
To program the line build out parameters of a T1E1 6162-1-T1E1-Code
To enable/disable customer pulse width flag for T1E1 6171-1-T1E1-Flag

T1 signaling
To program customer pulse width word1 for T1E1 T1 6172-1-T1E1-Customer Pulse Width Word 1
signaling
signaling
signaling
signaling
To enable/disable customer pulse width flag for T1E1 6155-1-T1E1-Flag

E1 signaling
To program customer pulse width word1 for T1E1 E1 6156-1-T1E1-Customer Pulse Width Word 1
signaling
signaling
signaling
signaling
To program the T1 DTMF digit timer for T1E1 6187-1-T1E1-DTMF Digit Timer
To program the T1 DTMF inter digit timer for T1E1 6188-1-T1E1-DTMF Inter Digit Timer
To program the ISDN PRI switch variant 6107-1-T1E1-ISDN PRI Switch Variant
To program Send Called Party Number Using for 6146-1-T1E1-Send Called Party Number Using
T1E1 port
To program forward tone maximum ON timer 7101-1-T1E1-Forward Tone Maximum ON Timer
To program forward tone maximum OFF timer 7102-1-T1E1-Forward Tone Maximum OFF Timer
To program the maximum compelled cycle time 7103-1-T1E1-Maximum Compelled Cycle Time
To program pulse duration for pulsed signals 7104-1-T1E1-Pulse Duration for Pulsed Signals
To program the pulsed

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SARVAM UCS

Know Your SARVAM UCS............................................................................................................................ 11

Before You Start ........................................................................................................................................... 35

Installing ETERNITY LENX........................................................................................................................... 47

Installing ETERNITY MENX........................................................................................................................ 211

Installing ETERNITY GENX ........................................................................................................................ 361

Configuring ETERNITY GENX ................................................................................................................... 501

Configuring SARVAM UCS ........................................................................................................................ 507

Configuring Voice Mail System ............................................................................................................... 1183

SMS Server................................................................................................................................................ 1301

Computer Telephony Integration (CTI) ................................................................................................... 1341

Incoming and Outgoing Call Routing ..................................................................................................... 1345

Features and Facilities ............................................................................................................................. 1359

Table of Contents iii

Voice Mail Features .................................................................................................................................. 2385

System Maintenance ................................................................................................................................ 2421

Appendix ................................................................................................................................................... 2469

Table of Contents vii

viii Table of Contents

About the Product

ETERNITY GENX Platform

UCS Universal Media Gateway

SARVAM UCS SME SARVAM UMG

ETERNITY GENX Platform

Matrix SARVAM UCS System Manual 1

SARVAM UCS SME

ETERNITY GENX Platform

ETERNITY LENX/MENX Platform

2 Matrix SARVAM UCS System Manual

Matrix SARVAM UCS System Manual 3

About this System Manual

The configuration commands are written with respect to ETERNITY MENX.

This System Manual is aimed at:

Matrix SARVAM UCS System Manual 5

Organization of this Document

6 Matrix SARVAM UCS System Manual

• Call another extension, Department Group,

Terminology used in this System Manual

Matrix SARVAM UCS System Manual 7

• Station: same as extension.

• CO Network: the public telephone exchange.

8 Matrix SARVAM UCS System Manual

• Calling party/Caller: the person who makes a call.

• Internal numbers: same as extension numbers.

Matrix SARVAM UCS System Manual 9

This Unified Communication Server creates a collaborative environment by:

• Integrating UC Features like unified messaging, video conferencing etc.

Features and Benefits

Matrix SARVAM UCS System Manual 11

Flexibility and Scalability

Cost Saving and Productivity Enhancing Features

12 Matrix SARVAM UCS System Manual

Other Key Features

Matrix SARVAM UCS System Manual 13

14 Matrix SARVAM UCS System Manual

Matrix SARVAM UCS System Manual 15

16 Matrix SARVAM UCS System Manual

Matrix SARVAM UCS System Manual 17

PMS Interface PMS

18 Matrix SARVAM UCS System Manual

Power Line Carrier Communication Application

Matrix SARVAM UCS System Manual 19

20 Matrix SARVAM UCS System Manual

Matrix SARVAM UCS System Manual 21