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SIP Application Note
SIP Application Note
SIP Application Note
Abstract
This document contains instructions for installing and configuring SIP functionality
on your IC Server.
3 Glossary of Terms................................................................................. 16
4 Introduction ......................................................................................... 16
4.1 Available SIP-Related Application Notes ......................................... 16
4.2 Standards........................................................................................ 17
4.2.1 Other Companies..........................................................................17
4.2.2 What is an RFC.............................................................................17
4.2.3 SIP Standards ..............................................................................18
4.2.4 Why has RFC 2543 been replaced with RFC 3261?.............................18
4.2.5 IP Address and Ports .....................................................................19
4.2.6 Security Alert ...............................................................................19
4.3 SIP Q&A .......................................................................................... 20
4.4 Implementation Overview Diagrams............................................... 27
4.4.1 Picture: SIP Hardware Approach Overview .......................................27
13 Outbound Logic.................................................................................. 57
14 Platforms ........................................................................................... 59
14.1 Platform Combinations and Supported Status .............................. 59
14.2 Platform Comparison.................................................................... 59
Authors: If you are making a change to this document, update the cover page
date to match the date of your latest changes.
Change Date
Updated “Specifying your firmware” section with new table. This now appears on 1/14/2002
page 6.
Updated firmware specifications table. Added procedure for changing firmware 1/17/2002
values. Updated DCM Network configuration settings with examples and corrected
values.
Added Things to watch for… section with a note about not using Terminal Services 1/23/2002
or Citrix Metaframe to run Dialogic Configuration Manager.
Added related documents to introduction section and added Troubleshooting section 1/24/2002
at the end of the document. Fixed typo is hexidecimal Subnet Mask field description.
Added section at end on monitoring SIP line details through Interaction Client.
Added Configuring Your System For Mu-law section, Notes About User and Station 2/11/2002
Extensions section, Notes About Quality of Service Section, describe the new station
parameters (persistent, call appearances, use proxy), AudioCodes Specific Section,
Sample Configuration section, DID section
Vendor specific portions, removed terminal services section, add MWI and message 3/18/2002
button configurations, add sip Q&A section, VAD, when changes of stations and line
take affect, add pictures of topologies
AudioCodes update, new AudioCodes boards, new Firewall/NAT section, new 4/14/2002
Identification section in station configuration
Added RTP Sender Report section, better incoming logic description, N+1 and 5/14/2002
redundancy, unique station and user extensions, better info on dial plans
Update on AudioCodes board model numbers (ver P03), SIP channel bank Q&A. 6/3/2002
On CIC 2.2
GA CD
More version numbers for Aculab, audiocodes firmware path is mandatory, Cisco SIP 7/1/2002
products Q&A, remove retired version P02 AudioCodes boards and ScBus IPM-260A-
120-TIP-CI board, more info on routing (section “SIP Message Routing”), details on
configuring voicemail for unmanaged phones, RFC2833 configuration, configuration
examples in “Sample Configuration” section, better remote site pictures.
Added EIC release directly by CIC release, new table for hardware platforms, 4.2 SIP 7/30/2002
standards section, misc tools in trouble shoot section, 7.2 VALN info, updated
Dialogic model numbers, more info on trouble shooting DTMF
Added H.100 termination to AudioCodes Setup server parameter, better dial plan for 8/19/2002
gateways
More info on setting 601 Dialogic boards to mu-law (15.3), more info on SIP
standards (4.2), removed ipvs_evr_isdn_net5_311.pcd and 8/30/2002
ipvs_evr_isdn_qsige1_311.pcd from 301 (15.2), fix typo in 15.5 (0x0A should be
0xA0).
More info on makecall button in the troubleshooting section, add more info on
security, attribute 3 for MWI, /NoDataprobe flag for switchover, Bus termination and 9/26/2002
VAD for audio quality problems, updated dates that CIC SR-B fixes are in EIC 2.2
GA, decision tree for “when do I need a proxy”, added known issues section On EIC 2.2
GA CD
More known issues, support for Audiocodes 30 and 60 port boards, Dialogic HMP,
more updates on when a proxy server is needed. 10/18/2002
Updated managed short cut info, large packet size info, reworked known issues
section, firewall config, HMP issues, better diversion documentation, HMP link, better 11/18/2002
doc for switchover and station configuration
Multiple NIC explanation, more work on known issues, tel scheme, more on vad,
HMP fixes, identification for stations. 12/17/2002
More on no audio and hold in trouble shooting section, no IVR trouble shooting,
documented audiocodes switch issue in known issues, better known issues section, 2/28/2003
dial plan config for only displaying user portion of SIP address for inbound and
outbound calls, AudioCodes plug and play PCI drivers
Delayed media, HF 1372 (for CIC SR-C) and 1384 (for EIC GA), repair screen shots
in Phone servers 3/18/2003
New server parameters (AudioCodes Network Gain and AudioCodes Bus Gain) for
Audiocodes (CIC 2.2 SR-C HF 1462, EIC 2.2 GA HF 1163), new hot fix doc for 1462 4/3/2003
and 1463.
In section 14.1 “Platform Combinations and Supported Status”, added the following
qualification to the Intel/Dialogic PCI Hardware and AudioCodes IP Boards 4/4/2003
combination: “Please note that Interactive Intelligence assumes no liability with (PL)
respect to performance under load of the Intel/Dialogic and AudioCodes
combination.”
Ethereal tool (section 33.2.5.5), trouble shooting echo with server parameters
AudioCodes Network Gain and AudioCodes Bus Gain, audiocodes 4.0 firmware, new 5/9/2003
audiocode board part numbers, remove IPLink configuration.
Remote Survivability and redo chapters on connectivity, Disable Delayed Media
config, 2.1 and 2.2 information sections 6/3/2003
Tell me about Cisco’s skinny protocol in the Q&A section, Tie Line and Multi-site
Configuration chapter, section in Audiocodes chapter about switch configuration, 6/16/2003
bandwidth usage, Cisco SIP SRST routers, QoS bytes, multiple gateway selection
Gateway selection (section 22 “Gateway Selection”), new hot fixes 1577 and 1578,
UseOffHookEventForSIPDialing server parameter (section 33 “Server Parameters”), 6/30/2003
new hot fix 1599 and 1601.
New server parameters for Aculab gain control and agc, typos, dsedit parameter
sections, 1633 and 1637 hot fixes 7/28/2003
Registry setting for HMP, global station dsedit parameters, more screen shots for
gateway selection. 7/31/2003
Echo in trouble shooting section, more info on Network and Bus gain, more fax info,
disabling secondary clock master 10/13/2003
Dialogic/AudioCodes combo is certified, new features for early media and connection
call warmdown time, always run wdreg_gui install, Eic_OutboundSetupParams 12/11/2003
attribute, modem configuration chapter
New 8.4 section for 2.3 external audio path, Broken RTP Disconnect Time warning
1/20/2004
Microsoft
Messenger
Some versions
Microsoft Messenger
will try to use an odd
port number for audio.
This is not valid with
HMP or AudioCodes.
Cisco VPN
Software
Microsoft Messenger Use Microsoft’s PPTP Cisco’s 4.0 VPN.
does not work with VPN software rather
Cisco VPN software. than Cisco’s VPN
software.
The Cisco VPN does
not expose its
interface, thus
Messenger passes
internal IP addresses
in its SIP messages
(SDP and 200 OK).
ActionTec
Actiontec phones only: None. CIC 2.2 SR-C*
A buzz is heard by
remote caller when an EIC 2.2 SR-A*
actiontec phone goes * Actiontec support
offhook to answer a is new in these
call. releases.
4 Introduction
With SIP (session initiation protocol) being the emerging standard now used for
call routing, state functions and control within IP Networks, Interactive
Intelligence now offers interoperability with SIP-based solutions.
As an open software solution, the Interactive Intelligence product line was
designed as a flexible and affordable alternative to traditional telecom solutions.
With a new SIP interface, Interactive Intelligence is excited to leverage it’s proven
Interaction Center Platform to contact centers, enterprises, e-businesses and
service providers that wish to utilize a SIP-based infrastructure.
Although SIP-based Soft switches provide an excellent answer for next generation
call transport over packet networks, they still lack the compelling applications that
will drive the level of acceptance that their unique offerings strive to achieve. For
example, capabilities as simple as voice mail are not available. Interaction Center
Platform answers this shortcoming by not only adding voice mail, but also a
number of applications.
SIP standards are evolving quickly, and the Interaction Center continues to
adhere to the specs for this emerging open standard. Below are the specifications
used. These will continue to changes as the new RFC standards/drafts:
4.2.4 Why has RFC 2543 been replaced with RFC 3261?
The status of RFC 2543 is that it has been obsoleted by RFCs 3261-3266.
These documents mainly clarify and resolve issues and mistakes in RFC
There are security alerts for VoIP protocols, H.323 (which is not used by
Interactive Intelligence) and SIP (which is used by Interactive Intelligence).
For H.323: Several critical flaws have been discovered in VoIP products
based on the widely used H.323 protocol:
http://www.cert.org/advisories/CA-2004-01.html
Note that Interactive Intelligence does not use H.323, it has chosen SIP
exclusively as its VoIP protocol.
Does the same Interaction Client work with all these new devices?
Yes. The same Client works with analog phones, SIP hard phones (such as the
Cisco 7960 and the Pingtel Expressa), and SIP soft phones (such as Microsoft
Messenger). In fact, some PBX digital phones are now supported with the Intel
NetStructure PBX-IP Media Gateway.
Does Cisco back SIP? I’ve heard different stories, depending on the
account and salesperson.
From the Cisco web site: “Cisco is enabling the advance of new
communications services with a complete SIP-enabled portfolio including IP
phones and analog telephone adaptors, packet voice gateways, proxy servers,
call control and signaling, and firewalls. These products are available today.”
See http://www.cisco.com/warp/public/cc/techno/tyvdve/sip/ for more info.
Also, Cisco Phone Data Sheets can be found at
http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_shee
ts_list.html. Note that all Cisco phones do not support SIP (yet).
Tell me about Cisco’s skinny protocol (called SCCP), H.323, and SIP.
How do their interrelate?
Here is a little Q&A:
• Why aren't Cisco’s skinny protocol (i.e. SCCP) and other proprietary call
control protocols a standard?
A: In 1997 and 1998, vendors were clamoring for VOIP call control protocols.
Unfortunately, there was not a lightweight call control protocol
available for vendors to standardize upon. As such, vendors such as 3COM,
Cisco, and Avaya each modified the H.323 call control protocols (H.225 and
Q.931 for example) to provide features and functions which would allow them
to compete with existing analog/digital PBX equipment. Because each of
these call control protocols were developed internally at competing
organizations there was never support for standardizing on any one protocol.
• Why would vendors protect their proprietary call control protocols when
the SIP standard is available?
A: Each of the vendors which created a proprietary call control protocol has a
large investment to recoup in order to justify the development efforts of the
protocol. As such, these vendors will be the last to adopt the SIP standard
and open nature that SIP brings to the deployment of VOIP networks. By
continuing to market their non-standard protocols, customers are put into a
very bad position where they must purchase the entire solution from the
vendor.
With SIP's ability to seamlessly integrate network and application components
from multiple parties, the cost justification for deploying a non-SIP based
network is rapidly eroding. There is ample evidence of this in the rapidly
falling cost of phones and network components for a VOIP deployment. Due
to this influence we are seeing the quiet adoption of SIP by the vast majority
of vendors who originally created a proprietary protocol.
Show me a stack with Cisco and non Cisco devices and what works
with and without SIP.
The chart below shows that Cisco apps can only use Cisco equipment, require
a Cisco CallManager, and can not take advantage of Cisco SIP phones and SIP
gateways. The Interaction Center with SIP can use Cisco SIP phones and SIP
gateways, PLUS other vendors SIP equipment, and does not need a Cisco
CallManager.
Cisco H.323 gateways and proprietary Cisco SIP gateways and Other vendors SIP
skinny (SCCP) phones SIP phones gateways and phones,
such as Microsoft
Messenger
Hardware – does all the Yes. We have a single No. Multi vendor solutions
gateways, routers, and phones vendor dependency on are used. The first
have to be from Cisco? Cisco. This typically leads gateways and phones we
to higher cost equipment. certified are from Cisco.
We have now certified
phones from Microsoft
and Pingtel. Our system
can work with any
certified SIP compliant
gateway or phone.
Application Yes.
Proxy Yes – or we can work with any SIP compliant SIP Proxy.
SIP Gateway Yes – because you can add SIP to an existing Interaction
Center with all its working telephony boards. You can
also use the Interaction Center in SIP only mode, without
any analog phone or trunking.
Flash is used to hold or bring Some IP phones have buttons to do hold, transfer,
up a voice menu to do features conference,…
like conference and transfer.
What features do you lose using a SIP Channel bank (SIP Phone
Gateway) over a T1 or E1 channel bank?
None. SIP channel banks (SIP Phone Gateways) could take a lot less
hardware, depending on your usage numbers. Why? Because a SIP phone only
uses a resource when there is an active call to that phone (rather than tying
up a dedicated T1/E1 channel for each phone like traditional channel banks).
Take a site with 400 business users and assume only 25% of the phones are
in use at any given time. With T1 Channel banks this would take 17 T1s. With
E1 Channel banks this would take 13 E1s. With HDSI this would take 4 HDSI
cards. With SIP channel banks this would use 100 IP resources. Note that
besides figuring out “trunking”, you need to consider “stationing” usage. The
closer the phones have to a 100% usage number (like in a call center), the
less gain you get from SIP channel banks.
I want the Operator for our company to be able to receive more calls
than the physical IP phone is capable of handling. For instance, I want
the Operator to be able to handle 20 simultaneous calls. Can I do that?
Yes, if you want to handle more calls than the IP phone is capable, check the
“Persistent” checkbox in the Station configuration within Interaction
Administrator. The Interaction Client can be used to manipulate a large
number of calls while the phone will be the audio device for the calls. The
phone will show one call while the Interaction Client will be used to manipulate
the calls. See section 18 “Defining Global Configurations SIP Stations” about
configuring Persistent connections.
Vo IP Audio (RTP)
IP Cards
Gateway/Routers
Telepho ny Bus
WAN
LAN
SIP Proxies can be
used to route SIP Interaction
Messages. The SIP M
Phones, Gateways, e s s ag Centers
SIP Phones and Gateways e P at
and Interaction h
Centers can use the
Proxy for all the
routing decisions.
SIP Proxy Server
Some Gateways
can route SIP PSTN / WAN LAN
Messages. If the
specific gateway SIP Gateway Interaction
can not route SIP SIP
M essa Centers
messages, then a ge P
a th
SIP Proxy must be
used.
See the SIP 3rd Party Component Feature Matrix spreadsheet for the values in the
Decision Tree below.
First, check if the SIP phones require a proxy. Check the “Backup Proxy”
capability in the SIP 3rd Party Component Feature Matrix spreadsheet.
If “Backup Proxy” is “Yes” or “N/A”, then the phones don’t require a proxy.
If “Backup Proxy” is “No”, then:
• If using the Interaction Client to make calls, no SIP proxy is
needed. Why? Because when the Client makes a call, it sends a
makecall request to the Interaction Center server, which will place
a call to the phone associated with the Interaction Client.
Check the “Dial Plan Routing” capability in the SIP 3rd Party Component
Feature Matrix spreadsheet.
If “Dial Plan Routing” is “Yes”, then the answer is:
Yes, the phone can route calls to a local gateway based on what is
dialed. No proxy is needed to do this routing.
If dialing using the Interaction Client, no proxy is needed. The
Interaction Center will have to be configured to send these calls from
that user to that specific gateway.
If dialing using the phone, no SIP proxy is needed. The phone’s
dialplan will do the routing.
If “Dial Plan Routing” is “No”, then the answer is:
The phone can not route calls to a local gateway based on what is
dialed.
If dialing using the Interaction Client, no proxy is needed. The
Interaction Center will have to be configured to send these calls from
that user to that specific gateway.
If dialing using the phone, a SIP proxy is required (since the phone
does not support a dialing plan).
First, check if the SIP phones require a proxy. Check the “Backup Proxy”
capability in the SIP 3rd Party Component Feature Matrix spreadsheet.
If “Backup Proxy” is “Yes”, then the phones don’t require a proxy.
If “Backup Proxy” is “No”, then the phones require a proxy server
(remember, dialing from the Interaction Client is not possible if the
Interaction Center server is unreachable). Since the phone has no backup
Depends.
If dialing using the Interaction Client, “No”. Since both the primary
and backup Interaction Center are not reachable, the Interaction Client
can not complete a call.
If dialing using the phone, “Yes”, but a SIP proxy is required. The
current phones do not have the ability to have multiple backup proxy
servers (the primary Interaction Center is the main proxy for the
phone, the backup Interaction Center is the backup proxy for the
phone, and the gateway would need to be the second backup proxy for
the phone).
SIP
PSTN / WAN LAN
3
SIP
4 PSTN / WAN LAN
1 2 3 4
IC Servers IC Servers IC servers with IC Servers with no
with no with ISDN SIP gateway, using SIP
gateways, connections to connections to connections to
using ISDN gateways gateways PSTN/WAN
connections to
the PSTN
Gateway No gateway. Connect to the Connect to the No gateways necessary.
Features PSTN PSTN and PSTN and PSTN and WAN
connectivity is WAN via WAN via connectivity is done via
done via the tradition traditional SIP. This is not available
telephony connections connections yet, but is coming soon
boards. (ISDN, Frame (ISDN, Frame by large carriers.
Relay) and then Relay) and then
connect to the convert all
IC server via traffic to SIP.
traditional
connections
(ISDN,…).
Are Telephony Tradition ISDN Tradition ISDN Optional. With Optional. With the
boards (or T1, E1, (or T1, E1, the hardware hardware platform
needed? Analog) Analog) platform (telephony boards), IP
telephony telephony (telephony boards are used to do
boards are boards are boards), IP the do the RTP and
used to connect used to connect boards are transcoding. With the
to the PSTN. to the gateway. used to do the software platform (Intel
do the RTP and HMP),
transcoding.
With the
software
platform (Intel
HMP),
N+1 The calls are The calls are The calls are The calls are distributed,
Configuration distributed, by distributed, by distributed, by by the PSTN, across the
(multiple IC the PSTN, the gateways, the gateways, IC servers, simply by
servers) across the IC across the IC across the IC sending the SIP
servers, by servers, by servers, simply messages to different IP
sending the call sending the call by sending the addresses.
to different to different SIP messages
ISDN trunks. ISDN trunks. to different IP
addresses.
1
2 SIP Co mpliant Soft
Analog Phones
Phones with or
without Interaction
IP WAN
Client
1 2 3
IP Phones SIP Phone Media Gateways Analog Phones
Is SIP used to Yes. Yes. The IC server communicates with No. Tradition T1/E1 boards
communicate to the Phone Media Gateway with SIP. The for channel banks, or analog
the phones Phone Media Gateway then station boards are used to
communicates with the phone the same connect to analog stations.
way a traditional channel bank does.
Are resources No. IP resources are No. IP resources are only used when Yes. The phone uses a
used when phone only used when there is there is a voice connection. physical resource even when
is idle? a voice connection. it is idle.
Does the phone No. The SIP hard or No. The Phone Media Gateway is simply Yes. The phone has a
have to be directly SIP soft phones are an IP device anywhere on the network physical connection to the IC
connected to IC simply IP devices (LAN or WAN). server.
server? anywhere on the
network (LAN or WAN).
Phone Types Many vendors make Standard analog phones (2500 sets) and Standard analog phones
supported SIP hard and SIP soft PBX digital phones can be connected to a (2500 sets).
phones. wide variety of Phone Media Gateways.
Not shown: Every remote site requires backup central site connectivity.
Outbound: The phone at the remote site can not reach the Cisco CallManagers at the central
site. It will then send the outbound call request to SRST capable router running at its remote
site. The SRST capable router will route the call according to its configuration, typically
using the router’s own connection to the PSTN.
Inbound: An inbound call is received by the a gateway at the remote site and the gateway can
not reach the Cisco CallManagers at the central site. It will then send the call to a SRST
capable router running at its remote site. The SRST capable router will route the call
according to its configuration, typically to a phone at the remote site.
Central Site
with Remote Site
Interactive WAN
Intelligence’s LAN
Interaction
Centers
PSTN
SIP capable SRST SIP Proxy (optional)
Cisco Router
Not shown: Every remote site requires backup central site connectivity.
Outbound: The phone at the remote site can not reach the Interaction Center Server at the
central site. It will then send the outbound call request to SRST capable router running at its
remote site. The SRST capable router will route the call according to its configuration,
typically using the router’s own connection to the PSTN.
Inbound: An inbound call is received by the a gateway at the remote site and the gateway can
not reach the Interaction Center Server at the central site. The gateway (a SRST capable
router) will route the call according to its configuration, typically to a phone at the remote site.
PSTN
SIP Router SIP Proxy (optional)
Not shown: Every remote site requires backup central site connectivity.
Outbound: The phone at the remote site can not reach the Interaction Centers at the central
site. It will then send the SIP outbound call request to a SIP capable router running at its
remote site. The SIP capable router will route the call according to its configuration,
typically using the router’s own connection to the PSTN. Note that if the phone is not
capable of making routing decisions based on unreachable systems, then either a router (which
could do all the SIP routing decisions with its dial plan) or a SIP proxy is needed at the remote
site.
Inbound: An inbound call is received by the gateway at the remote site and the gateway can
not reach the Interaction Centers at the central site. It will then send the call to a SIP capable
router running at its remote site. The SIP capable router will route the call according to its
configuration, typically to a phone at the remote site. Note that if the router is not capable of
routing decisions based off of unreachable systems, then a SIP proxy is needed at the remote
site.
Central Site
with Remote Site
Interactive
Intelligence’s WAN
Interaction LAN
Centers
PSTN
SIP Router SIP Proxy (optional)
Not shown: Every remote site requires backup central site connectivity.
The phone at the remote site dials 911. It will then send the SIP outbound call request to a SIP
capable router running at its remote site, rather than to the Interaction Centers at Central
Site. The SIP capable router will route the call directly to the PSTN. Note that if the phone is
not capable of making routing decisions based on unreachable systems, then either a router
(which could do all the SIP routing decisions with its dial plan) or a SIP proxy is needed at the
remote site.
Options if the gateway is at the remote site AND that gateway is to be used for
inbound and outbound dialing:
1. IC 2.2 will have the audio take two trips across the network, one from the
phone to the Interaction Center at the remote site, and the second from the
Interaction Center to the remote gateway.
Advantages: Features, such as recording, monitoring, and conferencing are
all available.
Disadvantages: The audio will be taking two trips across the network, which
will use bandwidth and add delay.
2. IC 2.3 will redirect the audio so the audio stays at the remote site (the audio
is not sent to the central site unless necessary for recording, monitoring, or
conferencing). Also, with a future release, multiple Interaction Centers will
be able to work as one, so an Interaction Center could be added to the
remote site so the audio never leaves the site, even when advanced features
such as recording, monitoring or conferencing are used.
Advantages: The call audio does not take a round trip to the central site.
However, the Interaction Center Server is fully aware of the call. Dialing
can be done from either the phone or the Interaction Client. The audio
can be sent to the central site dynamically if needed (if recording or
monitoring are requested).
Disadvantages: None.
3. Some calls originated from the remote phones can be sent directly (via the
phones’ dial plan or a remote proxy) to the remote gateway for emergency
dialing (911), for local dialing, or if the central site is not reachable.
Advantages: The call audio does not take a round trip to the central site.
Disadvantages: The Interaction Client can not be used for this type of
dialing (the dialing must be done from the phone). Also, the central site
Interaction Center server is not aware that the call was made (no recording or
no monitoring capabilities, call does not show on the Interaction Clients).
8.1 IP Resources
Each IP session will use an IP resource. An IP session is either:
• An active SIP connection from a gateway (typically an external call).
• An active SIP connection from the Interaction Center to a managed phone.
Examples
• A idle IP phone will not use an IP resource.
• An idle SIP gateway will not use an IP resource.
• a call into an ISDN telephony board to an agent using a SIP phone will use
one IP resource.
• A call from a SIP gateway to an agent using a SIP phone will use 2 IP
resources.
Scenario
• Inbound call from A to Interaction Center (IVR, dial by name, fax detection
…).
• Call transferred to Device B
Configuration
• Both A and B are configured in IA as with an AudioPath of Dynamic.
• A and B could have codecs configure or configured to determine their own
codecs with the AudioPath is dynamic.
Device A to IC
Now one IP resource on the IC server is being used. Note that the call might be
destined for device B, it will first come to the Interaction Center. The Interaction
Center will use an IP resource (this allows IVR, dial by name, fax detection, …).
IC to Device B
Once the call’s destination is discovered (ACD agent becomes available,
extension dialed, user’s name dialed,…), IC will send the call to Device B.
Direction AudioPath SIP Details
Message
IC to B Internal INVITE The INVITE contains the codec list
configured for Device A in IA and IC’s IP
address and port number.
IC to B External INVITE Contains either:
• For external audio and the checkbox
“Let external devices determine codecs
is selected”, the INVITE contains A’s
advertised codecs and A’s IP address
and port number.
• For external audio and the above
check box is not selected, the INVITE
contains the intersection of A’s
advertised codecs, A’s configured
codecs in IA, and B’s configured
codecs in IA – and A’s IP address and
port number.
B to IC Internal or OK The OK contains B’s advertised codecs.
External
IC to B Internal or ACK Audio can now start for Internal.
External
IC to A External Re-INVITE • For external audio and the checkbox
“Let external devices determine codecs
is selected”, the re-INVITE contains B’s
advertised codecs and B’s IP address
and port number.
• For external audio and the above
check box is not selected, the re-
INVITE contains the original negioated
codec and B’s IP address and port
number.
For internal audio path, two IP resources on the IC server is being used.
For external audio path, zero IP resources on the IC server are being used.
Rules:
• Devices must be SIP (i.e a SIP gateway and an IP phone; or two IP phones).
A ISDN trunck coming into the IC server will always have Must configure
codecs if 2 devices are to talk to each other.
• To insure G.729 is used by remote phone, then must make that the only
codec configured. Otherwise, another codec could be used.
Each voice or data datagram has a byte in the IP header that provides the quality of
service treatment desired. Routers, switches, and gateways must be configured to
observe this byte. This byte, depending on the protocol, can be used differently.
Vendor specific settings of the Type of Service byte:
• AudioCodes IP boards set this byte, by default, to 0xA0 (=1010 0000), and
can be changed in the line configuration section in Interaction Administrator.
• Intel HMP set this byte, by default, to 0x00, and can be changed in the line
configuration section in Interaction Administrator.
• Cisco IP phones set this byte, by default, to 0xA0 (=1010 0000).
• Messenger 4.6 sets this byte to 0x00.
• Messenger 4.7.0041 sets this byte to 0xA0 (=1010 0000).
9.2 Echo
For a Cisco Echo overview, see
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper091
86a00800d6b68.shtml
See the trouble shooting section for echo (section 35.3 “Echo”).
Transmit Info
tx packets The total number of RTP data packets transmitted by the sender since starting
transmission.
tx octets The total number of payload octets (i.e., not including header or padding)
transmitted in RTP data packets by the sender since starting transmission
Receive Info
rx lost packets The total number of RTP data packets from source that have been lost since
the beginning of reception. This number is defined to be the number of packets
expected less the number of packets actually received, where the number of
packets received includes any which are late or duplicates. Thus packets that
arrive late are not counted as lost, and the loss may be negative if there are
duplicates.
jitter (avg) An estimate (in milliseconds) of the statistical variance of the RTP data packet
inter-arrival time, measured in timestamp units and expressed as an unsigned
integer. The inter-arrival jitter J is defined to be the mean deviation (smoothed
absolute value) of the difference D in packet spacing at the receiver compared
to the sender for a pair of packets.
N/A implies this data is not available (not available with Intel/Dialogic
products).
10.1 Security
Interactive Intelligence recommends using SIP access over a WAN by utilizing a VPN;
opening port 5060 (the default port used for SIP) in corporate firewalls is NOT
User and station extensions must unique extensions (i.e. user extensions are
different than phone extensions). This allows users to “roam”, which means a user
can be associated with any phone (by logging in or starting a client) and his calls will
follow him.
The Interaction Center will process the call with the following logic:
1. Check if the call was made from a managed station. If so, make the call on
behalf of the station.
2. Set the following attributes for handlers:
Eic_LocalTn Set from the SIP message “To” header address.
For sip address scheme (addresses that start with “sip:”), type and port
number are added if not present in the header (sip:user@host:port).
Eic_LocalName Set from the SIP message “To” header display name
Eic_RemoteTn Set from the SIP message “From” header address.
For sip address scheme (addresses that start with “sip:”), type and port
number are added if not present in the header (sip:user@host:port).
In CIC 2.2 SR-B/EIC 2.2 GA, tel address schemes were changed to sip
schemes and would never appear. In CIC 2.2 SR-C/EIC 2.2 SR-A, and
2.2 CIC SR-B/2.2 EIC GA hot fixes 1206, 1203, 1212, 1213: For tel
address scheme (addresses that start with “tel:”), the address will look
like tel:number, with no port number.
In CIC 2.2 SR-B/EIC 2.2 GA, tel address schemes were changed to sip
schemes and would never appear. In CIC 2.2 SR-C/EIC 2.2 SR-A, and
2.2 CIC SR-B/2.2 EIC GA hot fixes 1206, 1203, 1212, 1213: For tel
address scheme (addresses that start with “tel:”), the address will look
like tel:number, with no port number.
In CIC 2.2 SR-B/EIC 2.2 GA, this attribute is only set if the URI
matches the IP VoiceMail Direct server parameter.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1178 and 1146 and in CIC 2.2
SR-C/EIC 2.2 SR-A, this attribute is set if the Diversion header is
present.
In CIC 2.2 SR-A/EIC 2.2 GA, this is the exact address in the Diversion
header. Type and port number are NOT added as they are to the
Eic_LocalTn, Eic_RemoteTn, and Eic_RedirectionTn.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1178 and 1146 and in CIC 2.2
SR-C/EIC 2.2 SR-A, for sip address scheme (addresses that start with
“sip:”), type and port number are added if not present in the header
(sip:user@host:port).
In CIC 2.2 SR-B/EIC 2.2 GA, tel address schemes were changed to sip
schemes and would never appear. In CIC 2.2 SR-C/EIC 2.2 SR-A, and
2.2 CIC SR-B/2.2 EIC GA hot fixes 1206, 1203, 1212, 1213: For tel
address scheme (addresses that start with “tel:”), the address will look
like tel:number, with no port number.
In CIC 2.2 SR-A, Eic_RedirectingTn is the top most header (i.e. the last
diverted user).
In CIC 2.2 SR-B/EIC 2.2 GA, Eic_RedirectingTn is the bottom most
header (i.e. the first diverted user).
In CIC 2.2 SR-B/EIC 2.2 GA, this attribute is only set if the URI
matches the IP VoiceMail Direct server parameter.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1178 and 1146 and in CIC 2.2
SR-C/EIC 2.2 SR-A, this attribute is set if the Diversion header is
In CIC 2.2 SR-B/EIC 2.2 GA, this attribute is only set if the URI
matches the IP VoiceMail Direct server parameter.
In a hot fix and in CIC 2.2 SR-C/EIC 2.2 SR-A, this attribute is set if the
Diversion header is present.
In CIC 2.2 SR-B/EIC 2.2 GA, this attribute is only set if the URI
matches the IP VoiceMail Direct server parameter.
In a hot fix and in CIC 2.2 SR-C/EIC 2.2 SR-A, this attribute is set if the
Diversion header is present.
Eic_OutboundSetupParams Requires Hot Fix 2.2 CIC SR-D 1877, 2.2 EIC SR-A HF 1878, 2.2 CIC
SR-E, or 2.2 EIC SR-B.
The value in the call attribute Eic_OutboundSetupParams will be put in
the ININAttr header in the SIP message during outbound call logic and
blind transfer logic.
Note: Eic_OutboundSetupParams can not contain any double quotes.
Eic_OutboundSetupParams syntax:
[name=value[;name=value]*]
Example:
If Eic_OutboundSetupParams is set to:
Agent=Frank Smith;Number In Queue=12
Then the outbound SIP message will include this header:
ININAttr: “Agent=Frank Smith;Number In Queue=12”
And the inbound call will have the follow attributes set before the
incoming call handler is invoked:
Agent will be set to: Frank Smith
Number In Queue will be set to: 12
13 Outbound Logic
When a call is made from using the Interaction Client, an audio path must be made
between the SIP phone and the Interaction Center server. The Interaction Center
server will make a call to the SIP managed station, and then once the connection is
made, will complete the requested call. Note that if persistent connections are used
(see section 18 “Defining Global Configurations SIP Stations”), an audio connection
might already be established, which means that the request call will start
immediately.
Example:
If Eic=OutboundSetupParams is set to
Aculab AudioCodes Yes (this is the preferred hardware configuration). See section
Hardware IP boards 15 for AudioCodes IP board configuration details.
Intel/Dialogic AudioCodes Yes, with caveats. Dialogic plus AudioCodes configuration has been
PCI IP boards validated for IC 2.2 for the addition of one (1) AudioCodes card in
Hardware selected Dialogic configurations. Two or more AudioCodes cards in a
Dialogic system have not been tested and is not supported at this
time.
Please refer to the Validated Server Matrix spreadsheet for existing
installs
(http://www.inin.com/support/cic/22/hardware/serverlist.asp?q=670&
) and for new installs
(http://www.inin.com/support/cic/22/hardware/download/New_Install
_Server_Matrix.xls)
While no significant issues have been found with the combination of
Dialogic and AudioCodes cards, we are unable to give blanket approval
to older existing servers due to the higher CPU loads required for
AudioCodes SIP processing. There may be older PCI servers in the
customer base that will not perform with AudioCodes cards.
Price Check with your hardware vendor. Check with Interactive Intelligence.
Density 30, 60, and 120 simultaneous RTP sessions. HMP 1.1: See the HMP chapter for the
latest densities (section 16 “Installing and
Benefit: The actual number of usable resources Configuring Intel HMP Software Solution”)
ports provided may exceed the rated capacity of
the Audiocodes boards. Currently, the Audiocodes
30 port board reports in as a 40 port board. All 40
sessions are usable on the 30 port board, but this
is not guaranteed in future AudioCodes firmware.
Number of New numbers will be coming out shortly with the NA. This is a total software solution
boards per new worst case scenario numbers. These without boards.
Server numbers are what should be used when installing a
system.
Note that 600 ports was tested with a light call rate
(3 calls/second, 180 calls/minute). Also, all calls
were not being recorded and tracing was set to
default.
Aculab Servers: 600 AudioCodes IP ports can be
in a single server (five 120 port boards). Also,
there is an Aculab limitations of 300 simultaneous
audio operations (plays and records) on a single
Aculab system. Note, since the Audiocodes
boards are non-universal, there are only a few
servers that can accept many non-universal
boards.
Multiprocessor Yes. HMP 1.0: No.
Capable
HMP 1.1: Yes, dual processors.
HMP 2.0: Yes, quad processor and
processor affinity.
Play and No. Must use an additional resource board. I3 does Yes.
Record not use the AudioCodes voice resources and the
board is priced accordingly. Voice resources from
Aculab or Intel/Dialogic cards are used for audio.
• Note: Aculab Prosody boards (each with up to
4 DSPs and 4 T1s) are supported. Each of
the 4 DSPs can be used for either 60 voice
resources, 8 faxes, 24 conference resources
with echo cancellation, or 64 conference
resources without echo cancellation.
• Note: Intel/Dialogic 240 voice resource board
(DMV2400A-PCI) is a high density voice
board. It can be configured for 240 voice
resources, or 120 conference resources, or
60 of each (60 is not a typo – when
Check for support status and vendor combination in section 14.1 ”Platform
Combinations and Supported Status” before using AudioCodes IP Boards.
15.1.1 Servers
The certified servers for AudioCodes boards can be found at
http://www.inin.com/support/cic/22/hardware/serverlist.asp?q=670
15.2 Prerequisites
• Load the Interaction Center software first. The Interaction Center install will
copy all the AudioCodes files that are needed for the AudioCodes
configuration.
• Note the MAC addresses of the AudioCodes boards in your system. It is
printed on a label on the board. The MAC address will be needed for
configuration later in this chapter.
The negotiated speed and duplex are logged in the event viewer.
Interaction Center Product Release/Hotfix for AudioCodes Plug and Play drivers
CIC 2.2 SR-D Install HF 1713 to install the PnP driver (5.0.5.1).
OLD: PnP driver 5.0.5.0 is installed with SR-D.
Must activate PnP drivers below to prevent a resource conflict.
CIC 2.2 SR-C Install HF 1713 to install the PnP driver (5.0.5.1).
OLD: Install HF 1549 to install the PnP driver (5.0.5.0).
Must activate PnP drivers below to prevent a resource conflict.
CIC 2.2 SR-B Install HF 1713 to install the PnP driver (5.0.5.1).
OLD: Install HF 1549 to install the PnP driver (5.0.5.0).
Must activate PnP drivers below to prevent a resource conflict.
EIC 2.2 SR-A Install HF 1714 to install the PnP driver (5.0.5.1).
OLD: Install HF 1550 to install the PnP driver (5.0.5.0).
Must activate PnP drivers below to prevent a resource conflict.
If the PnP drivers are not activated, you could have a TsServer startup problem
caused by a resource conflict. The TsServer log’s last line is an attempt to
open the AudioCodes board:
The resource conflict can be solved by activating the PnP drivers with the
hardware wizard.
1. Press the Next button when the Found new Hardware Wizard dialog
appears.
3. Clear all of the Optional search locations: and then press the Next
button.
4. After a few seconds the wizard should report that it was able to locate a
driver for this device at C:\WINNT\inf\ipm260.inf. Select Next to select
1. Verify the driver was installed with the Interaction Center install (this
occurs automatically). The driver file should be located at
C:\winnt\system32\drivers\windrvr.sys
2. To insure the driver is previously added, you must remove the driver.
This is done by clicking Start -> Run… and entering “cmd” to bring up a
DOS prompt. Then run the program
“D:\I3\IC\Server\Diagnostics\AudioCodes\wdreg_gui remove”. If the
default path for the Interaction Center install was not used then replace
the “D:\I3” with the install path that was used.
Ignore any error message wdreg_gui remove displays since the drivers
might (or might not) were already added.
3. Add the driver to the list of devices that the operating system loads on
boot. This is done by clicking Start -> Run… and entering “cmd” to
bring up a DOS prompt. Then run the program
“D:\I3\IC\Server\Diagnostics\AudioCodes\wdreg_gui install”. If the
default path for the Interaction Center install was not used then replace
the “D:\I3” with the install path that was used. The following dialog will
appear if the drivers are successfully installed.
Note that if you run “wdreg_gui install” twice, you will get a failure
message. To fix, simply run “wdreg_gui remove” and then “wdreg_gui
install” again.
New in CIC 2.2 SR-C HF 1462 and EIC 2.2 GA HF 1463: There are two
AudioCodes specific server parameters for audio adjustment.
These server parameters been replaced with the gain settings in the line,
station, and global station in Interaction Administrator and will only work if
those values in IA have not been changed; If the values in IA have been
changed, the values in IA will be used. The IA parameters were added in CIC
2.2 SR-D.
These server parameters are retired in 2.3.
Note that these configurations are dynamic, and will take affect when the next
SIP call starts.
o AudioCodes Network Gain (optional, default is 0, valid range: –31 to 31
(dB)). This value controls the gain applied to the audio signal received
from the IP network. For example, this would be applied to the signal
coming from an agent’s phone.
o AudioCodes Bus Gain (optional, default is 0, valid range: –31 to 31 (dB)).
This value controls the gain applied to the audio signal received from the
TDM bus (and going to the IP network). For example, this would be
applied to the signal going to an agent’s phone.
Example 1: 00908f12ab89;0;0;10.1.3.50;255.255.255.0;10.1.3.1
Board with MAC address 00908f12ab89 is configured as a
bus slave and does not terminate the H.100 bus. It is
assigned IP address 10.1.3.50, subnet mask 255.255.255.0,
gateway 10.1.3.1.
Example 2: 0;0;0;172.16.128.76;255.255.0.0;172.16.1.1
First board discovered is configured as a bus slave and does
not terminate the H.100 bus. It is assigned IP address
172.16.128.76, subnet mask 255.255.0.0, gateway
172.16.1.1.
Example 3: 00908f12ab89;0;1;172.16.128.76;255.255.0.0;172.16.1.1
Board with MAC address 00908f12ab89 is configured as a
bus slave and terminates the H.100 bus. It is assigned IP
address 10.1.3.50, subnet mask 255.255.255.0, gateway
10.1.3.1.
MACAddress 12-digit MAC address of the board or 0 (zero). The MAC
address should be entered as shown on the sticker attached
to the physical card. Note: If the board is not required to
have a specific IP address then 0 (zero) can be entered in
this field and the system will assign the IP address to the
next discovered board that does not have a specific IP
assigned to it.
MasterOrSlave 1 or 0, represents whether the board is the clock master for
the bus or clock slave, respectively. If your system contains
16.1.1 Servers
The certified servers for HMP can be found at
http://www.inin.com/support/cic/22/hardware/serverlist.asp?q=670
16.1.2 Densities
These limits apply to HMP 1.1.
16.3.2 IP addresses
Currently, the IP address is configured when you install HMP. If you change
your IP address, you must reinstall HMP or go into the registry and change
it. It can be found at HKEY_LOCAL_MACHINE/Software/SBLabs/dm3ssp.
16.3.3 Timers
HMP requires a high resolution timers for real time processing of 10, 20, and
30 millisecond frames. There are two timers that can be used:
• Microsoft Windows Multimedia Timer (mmtimer). This is a software
timer. This timer is on higher speed machines (1GHz and beyond).
• Advanced Programmable Interrupt Controller (APIC). This is a
hardware timer (via an on-chip controller on the Pentium family
processors). This is more accurate than the mmtimer.
By default, the APIC timer is used.
Problem 1: The local APIC is disabled and unavailable for use.
Solution: The HMP system will detect the presence and state of the local APIC timer by
performing software checks at initialization. If an operational APIC is detected, it will be used. If
an APIC is not detected, the HMP system will default to using the mmtimer.
Problem 2: The local APIC’s operation may not be reliable when used in conjunction with some
chipsets if Advanced Power Management (ACPI) is installed.
Major Issues
Voice Activate Checked (On) means use VAD on any connection that is
Detection (VAD) NOT to a station. If the connection is to a station, the
VAD configured in the station is used.
Default: Off
Vendor Specific
Intel/Dialogic software (HMP) does not support VAD.
Default: On
Vendor Specific
Intel/Dialogic software (HMP) support RFC2833.
AudioCodes hardware boards support RFC2833.
Disable T.38 Faxing Unchecked (Off) means that T.38 will be used for faxes
over SIP.
Default: Off.
Vendor Specific
Intel/Dialogic software (HMP) does not support T.38
(planned for HMP 1.1).
AudioCodes hardware boards support T.38.
Default: On.
Vendor Specific
Intel/Dialogic software (HMP) does not support echo
cancellation.
AudioCodes hardware boards support 30ms of echo
cancellation on the voice going from the TDM bus to the IP
network.
Network Gain -31 to 31 dB, Default is 0
Vendor Specific
Intel/Dialogic software (HMP) does not support network
gain.
AudioCodes hardware boards support network gain.
Bus Gain -31 to 31 dB, Default is 0
Vendor Specific
Intel/Dialogic software (HMP) does not support bus gain.
AudioCodes hardware boards support bus gain.
Maximum Number of
Calls
Combined If the Combined radio prompt is selected, the Combined value
Inbound/Outbound prompt is shown.
Protocol
Transport Protocol Can be UDP or TCP. If UDP is selected, the following additional
fields must be defined. Most of the following fields are grayed
out if TCP is selected.
Default: UDP
Maximum Invite Retry UDP: Maximum packet retry for INVITE and ACK requests
Valid: 0 to 6
Default: 6
Authentication
New in CIC 2.2 SR-D, CIC 2.2 SR-C Hot Fix 1372 and EIC 2.2 SR-A:
Delay media negotiation is simply delaying the advertising of supported codecs in the
SIP codec negotiation process. Delayed media on outbound calls gives the
Interaction Center server more control over the coder negotiation process.
If one codec is used, normal media negotiation is used for outbound calls. If more
than one coder is selected, delayed media negotiation is used for outbound calls.
The calling system always controls what time of media timing occurs.
Delayed media negotiation can also be disabled by setting “Disable Delayed Media”
setting “Disable Delayed Media” (see section 17.1 “Line Configurations not exposed
through Interaction Administrator”) – “Disable Delayed Media” requires HF CIC 2.2
SR-C Hot Fix 1562, EIC 2.2 SR-A Hot Fix 1564, CIC 2.2 SR-D, or EIC 2.2 SR-B.
Codecs An ordered list of codecs. The Interactive Center will negotiate the connection
to use the first codec on the supported list. You can select multiple codecs and
then prioritize them by moving them up or down in the list.
IA will only store an ordered list of those Codecs that are checked. The
Up/Down buttons are available to order this list. Also only the G.711 codecs
allow the frame size to be modified.
Notes:
• Note 1: Only configure the codecs the platform supports.
• Note 2: These codecs are not supported by Intel/Dialogic Software
HMP 1.1. Only configure the coders the platforms support.
• Note 3: HMP does not support 4 frames/packet with G.723 and does
not support 1 frame/packet with G.729.
• Note 4: Data Rate: The data rates shown below do not include
packet header overhead. For example, G.711 actually uses 80K-
100Kbps. The data rates below are all for half duplex (which is what
most conversation are). However, if VAD is not used, silence is
transmitted, thus using double the bandwidth indicated.
• Note 5: Packet and Frame size: nice summary on the topic of packet
size and frequency from the www.erlang.com website: "The
frequency at which the voice packets are transmitted have a
significant bearing on the bandwidth required. The selection of the
packet duration (and therefore the packet frequency) is a compromise
between bandwidth and quality. Lower durations require more
bandwidth. However, if the duration is increased, the delay of the
system increases, and it becomes more susceptible to packet loss;
20ms is a typical figure." So, the more of the voice you put in a
single packet (say 60ms versus 20ms) then the more of the voice you
lose if that packet is lost.
Proxy
List of Proxy Addresses Note: A SIP proxy server is not required, but does provide some
features that might be needed in certain network topologies. A
SIP proxy can do network and also do gateway selection.
Do not put the Interaction Center Server in this field, since it will
cause all SIP calls to be looped back to the Interaction Center.
Registrar
External List List of external telephone numbers that are not configured in our
system but need to be directed to our server when encountered.
Therefore, we must register them with the registrar. Typically,
these are numbers that are provisioned on the PSTN interface
but not provisioned in our system, like a 1-800 number.
IP Addresses Priority list of registrars available for contact registration by the
IC. If a registrar is configured then all IC contacts are sent to it
in a SIP REGISTER message. All messages will be sent to the
first registrar in the list. The remaining registrar entries will only
be used if the first entry is deemed not operational.
IC supports a default configuration for all the SIP station (note that each station has
the ability to overwrite one or all of these configurations). To get to this
configuration, go to Interaction Administrator, select (on the left pane) the container
that is your server name. “Configuration” will be displayed on the right pane. Double
click on “Configuration” and then select the “SIP Station” tab. The following dialog
will be shown.
Vendor Specific
Intel/Dialogic software (HMP) does not support VAD.
Station Connections are Checked indicates that connections to the station are persistent,
Persistent and will not be disconnected until the station initiates the
disconnection. Unchecked indicates that when the Interaction
Center determines that the audio path to the station is no longer
needed, the Interaction Center will initiate the disconnection.
Note that if Persistent is used, the number of call appearances
will be 1. The connection will be established by the SIP phone
(when it makes a call) or by the Interaction Center server (when
it calls the SIP phone because a connection is requested via the
Interaction Client (pickup, makecall, listen,…).
Recommended setting:
Operators: If you want to handle more calls than the phone is
capable (for instance an operator want to handle up to 20
simultaneous calls), check the Persistent checkbox. The
Interaction Client can be used to manipulate a large number of
calls while the phone will be the audio device for the calls. The
phone will show one call while the Interaction Client will be used
to manipulate the calls.
Call Center Agents: If call center agents are using an IP phone
with a headset and using the Interaction Client, Persistent
should be used.
Example1: The Interaction Center makes it’s PSTN call via SIP
calls through a SIP/ISDN gateway. This particular SIP/ISDN
gateway only sends a SIP connect message back to the
Interaction Center after the remote party answers the call. If
call analysis is used, you would want to keep checked Terminate
Analyses On Connect, so that call analysis will terminate when
the SIP connect message is received.
Example2: The Interaction Center makes it’s PSTN call via SIP
calls through a SIP/analog gateway. This particular SIP/Analog
gateway always sends a SIP connect message back to the
Interaction Center prematurely, before the remote party
answers the call. If call analysis is used, you would want to
uncheck Terminate Analyses On Connect, so that call analysis
will continue after the SIP connect message is received.
Default: On
Number of Call Appearances per Enter the number of call appearances the phone can handle. The
Station Interaction Center will send up to the configured number of calls
to the phone.
Note that if Persistent is used, the number of call appearances
will be 1.
Recommended Setting:
General: This value should be over 1 for only experienced
phone users
Vendor Specific
Cisco: The Cisco IP phone 7960 can have up to 6 line
appearances (each line appearance is equivalent to a station).
Each line appearance has a unique SIP address. Don’t confuse
line appearances with call appearances. Each line appearance
handles 2 call appearances. Configure the phone to one line
appearance and then this station configuration to 1 or 2 call
appearances.
Pingtel: Pingtel Expressa IP phone has one line appearance that
handles 4 call appearances. Configure station configuration to 1,
2, 3, or 4 call appearances.
Vendor Specific
Intel/Dialogic software (HMP) support RFC2833.
AudioCodes hardware boards support RFC2833.
Echo Cancellation Checked (On) means that echo cancellation will be used.
Default: On.
Vendor Specific
Intel/Dialogic software (HMP) does not support echo
cancellation.
AudioCodes hardware boards support 30ms of echo cancellation
on the voice going from the TDM bus to the IP network.
Network Gain -31 to 31 dB, Default is 0
This value controls the gain applied to the audio signal received
from the IP network. For example, this would be applied to the
signal coming from an agent’s phone.
Vendor Specific
Intel/Dialogic software (HMP) does not support network gain.
AudioCodes hardware boards support network gain.
Bus Gain -31 to 31 dB, Default is 0
This value controls the gain applied to the audio signal received
from the TDM bus (and going to the IP network). For example,
this would be applied to the signal going to an agent’s phone.
Vendor Specific
Intel/Dialogic software (HMP) does not support bus gain.
AudioCodes hardware boards support bus gain.
Notes:
• When a station is created, the changes take affect immediately.
• When a station is modified, the changes take affect when the station is idle
(there are no more calls on this station’s queue).
Use Default If checked, the values configured in the Global SIP Station
Configuration (see section 18 “Defining Global Configurations
SIP Stations”).
• Use Proxy for Station If the “Use Default” check box is checked, the values
Connections configured in the “Global SIP Station Configuration” will be
• Voice Activate Detection (VAD) used. If the check box is not checked, then these values will
• Station Connections are be able to be set independently. See the Global SIP Station
Persistent Configuration (see section 18 “Defining Global Configurations
• Terminate Analysis On Connect SIP Stations”) for complete description.
• Number of Call Appearances
per Station
• DTMF Type
• Echo Cancelation
• Network Gain
• Bus Gain
Connection This is the SIP address of the SIP device. This address is
used by the Interaction Center to connect to this SIP station.
See below for details.
Identification This is the SIP address that identifies the SIP device. This
address is used by the Interaction Center to identify this SIP
station. See below for details.
When you press the “Connection” button, the following dialog appears:
Option 2 (RECOMMENDED)
Recommended
Option 3
Note: The address of the station when a call comes into the IC server
(identification address) can be different than the address the IC server needs to
use when calling the station (contact address). This is because an inbound call
from a station could be coming through a proxy server. The header of the SIP
message could contain the address of the Interaction Center and not the Station
address itself.
In the above dialog options 1 and 3, the Interaction Center Server will use the
address 7111@2.2.2.2 to identify the phone. For option 2, the Interaction Center
Server will use the address 7111.
Option 2 (RECOMMENDED) (new 2.2 CIC SR-B/2.2 EIC GA, “Use a alternate
format”) must be used if for the phone if all the following is true:
• The phone’s connection address is NOT the same address it sends in the
“From” header of the SIP messages
• Switchover is used
• The extension value configured on the phones is unique amongst all your IP
phones.
Option 3 (“Use a predefined format”) should be used rarely. All the following
must be true for Option 3 to be used.
• The phone connection address is not the same address it sends in the “From”
header of the SIP messages. For Cisco phones, change the host portion to
match the value configured for proxy1_address in the Cisco phone
configuration, which is the addresses of the Interaction Center.
• Switchover is not used.
The Identification User and Host value when using “proxy_backup” configuration
ro IC switchover should be (using Option 3)
line1_name (no “sip:”, no “@”, no host name, no port number).
This value can only be set using the alternate format of the Identification User and
Host value.
Pingtel: For Pingtel IP phones, the user field is the same value configured for
“PHONESET_EXTENSION” and the host field is the phone’s IP address. The
Identification User and Host value when using Pingtel phones should be sip:[value
configured for PHONESET_EXTENSION]@[phone’s IP address]:5060. See the “SIP
3rd Party Component Application Note” for details (Option 1).
In the above example, the PC’s IP address “1.1.1.1”, and the user@host
Messenger “Sign-in name” would be “7111@2.2.2.2.
port The port value is, by default 5060. It is the same value configured in the line
configuration (see section 17.2.1 “SIP Configuration Page”).
Authentication
Both input conversions above convert the number to “sip:something”. On the Dial
Plan page, you can specify a dial group for handling outbound SIP calls (calls in
the format of “sip:something”). See the Phone Numbers in IC whitepaper (located
in the \Documentation directory) for more information on working with phone
numbers and dial plans in IC.
Dial Group The line group with the sip lines to be used for the call.
Dial String The number to be dialed for the specified input pattern. In the above
dialog, “sip:something” for the input pattern “sip:something”.
Important: The trailing “Z” is present to allow “/” dialing and account
code dialing (someone might dial 201-555-1111/123).
Dial The line group with the sip line to be used for the call. This line group (“SIP Lines” in the above
Group dialog) will typically contain one SIP line.
Multiple Gateway Note when using a single SIP Line: Add a second dial group entry when using
multiple gateways. This second entry will be used if the first entry fails. The second dial group
entry will have a dial string equal to the 2nd gateway’s name or IP address. The same SIP line
group can be specified on both entrys. When using a single SIP Line in the two entries, you are
using the SIP message responses from the gateway to inform the Interaction Center that the
gateway is congested.
More complicated and rarely used - Multiple Gateway Note when using multiple SIP Lines: Add a
second dial group entry when using multiple gateways. This second entry will be used if the first
entry fails. The second dial group entry will have a dial string equal to the 2nd gateway’s name
or IP address. When using a multiple SIP Lines in the two entries, you are using both the SIP
message responses from the gateway and the number of calls configured in the line
configuration to restrict the number of calls sent to a particular gateway.
Dial
String
Dial The number to be dialed for the specified input pattern. In the above dialog, the number to be
String dialed is 1201Nxxxxxx@gateway1 for the input pattern +1201NxxxxxxZ.
Important: Ordinals are used (i.e. “{7}”) rather than the wildcard syntax (NXYZ?) since the
wildcard syntax (NXYZ?) can NOT be used with alpha characters, such as “gateway1”. A
wildcard syntax is shown below.
Important: Specify the dial string with the Dial Group rather than in the “Default Dial String”
field. The default dial string field is only used when no dial groups are specified.
Important: The trailing “{13}” is present to allow “/” dialing and account code dialing (someone
might dial 201-555-1111/123).
Dial The line group with the sip line to be used for the call. This line group (“SIP Lines” in the above
Group dialog) will typically contain one SIP line.
Multiple Gateway Note when using a single SIP Line: Add a second dial group entry when using
multiple gateways. This second entry will be used if the first entry fails. The second dial
group entry will have a dial string equal to the 2nd gateway’s name or IP address. The same
SIP line group can be specified on both entries. When using a single SIP Line in the two
entries, you are using the SIP message responses from the gateway to inform the Interaction
Center that the gateway is congested.
More complicated and rarely used - Multiple Gateway Note when using multiple SIP Lines: Add
a second dial group entry when using multiple gateways. This second entry will be used if the
first entry fails. The second dial group entry will have a dial string equal to the 2nd gateway’s
name or IP address. When using a multiple SIP Lines in the two entries, you are using both
the SIP message responses from the gateway and the number of calls configured in the line
configuration to restrict the number of calls sent to a particular gateway.
Dial The number to be dialed for the specified input pattern. In the above dialog, the number to be
String dialed is 1202Nxxxxxx@172.16.128.4 for the input pattern +1202NxxxxxxZ.
Important: Ordinals (i.e. “{7}”) are not used in this example. The wildcard syntax (NXYZ?)
could be used since there are no alpha characters, such as “gateway1”.
Important: Specify the dial string with the Dial Group rather than in the “Default Dial String”
field. The default dial string field is only used when no dial groups are specified.
Important: The trailing “Z” is present to allow “/” dialing and account code dialing (someone
might dial 201-555-1111/123).
21.2.1 Example 1
The example below, a new Dial Plan object “sip:NxxNxxxxxx@Z” was created so
only the user portion of the SIP address (({5}{6}{7} {8}{9}{10}-
{11}{12}{13}{14}) is the ordinal of the user portion) is displayed.
A sip inbound call from sip:3178723000@sip.inin.com will be displayed as (317)
872-3000.
22.1 Availability
T.38 is available with AudioCodes in CIC 2.2 SR-D and EIC 2.2 SR-A (via HF 1674).
It is also available in all 2.3 releases.
T.38 is available with HMP in CIC 2.2 SR-E. It is NOT available in EIC 2.2 releases.
It is also available in all 2.3 releases.
22.3 Scenarios
Benefits: Tie-Lines
WAN
Interaction Center C
LAN LAN
9 Instant connectivity, each SIP IC can connect to other SIP IC’s - virtual tie-lines
9 Great for IC’s running in a Multi-site configuration
9 Direct dialing, 101@A can be dialed, with no configuration changes on either
system, from B or C.
9 ‘Transparent’ dialing between ssstems with Milti-site or with dial plan
configuration
©2003 Interactive Intelligence Inc.
By simply adding SIP, connectivity to all other SIP devices on your LAN and WAN
becomes available. This is true of connectivity between SIP IC servers. All SIP IC
servers can communicate with each other.
There are three basic techniques:
• Manual dialing between systems can be accomplished with SIP addressing
(the user will dial a user extension, followed by an “@” sign, followed by the
IC server name). For example, a user with extension 101 on IC server A can
dialed by users on server B or C by simply dialing 101@A.
• Tie lines can be configured between systems. A SIP line is no different
than a T1 or ISDN line and can be added to a line group in the very same
manner. The dial plan can be configured to use a line group when dialing a
specific number or a specific set of numbers. For example, when dialing
715-xxxx, the dial plan can be configured to modify the dial string to
715xxxx@B and chose the line group with the SIP line can be used.
• Multi-site can be configured with SIP lines, just like any other T1 or ISDN
lines. Again, in Multi-site, each system is configured with a set of numbers
indicating how to reach each other system in the collective. So, on IC server
A, you would configure xxx@B as the number to reach IC server B. When
someone on server A dials a user extension and that user extension is on B,
Multi-site will dial xxx@B to get to that server. xxx@B will be configured in
the dial plan to use the line group containing the SIP line.
Read the Switchover white paper for the most up-to-date switchover information.
See the Switchover application note on the Interactive Intelligence web site. In
an all SIP environment, no dataprobe equipment is required. In a mixed trunk
line and SIP environment, dataprobe equipment is still required to switch the
trunk lines.
How:
Use regedt32 to go to the following key:
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Interaction
Center\ProcessTree\Level1\SwitchoverService.
With the SwitchoverService key selected in the left pane, double click on the
CommandLineArguments entry in the right pane. When the String Editor
dialog box appears, type /NoDataProbe, then click OK.
Since each SIP line can have multiple active calls, the Line Details page in
Interaction Client is the best place to monitor SIP line activity.
To add the Line Details page in Interaction Client:
1. In Interaction Client, right-click the area to the right of the telephone pages
(shown in the oval in the following figure.) Select Insert page… from the menu
that appears.
2. In the Pages dialog, select Line Details form the Available list, as shown in the
following figure.
Instructions:
c. Click Next
Example:
Assume there are 240 IP resources in your system.
Assume there are 500 SIP stations (i.e. managed phones) and in the worst case,
1 our of 4 phones will be in used at any given time. Therefore, the 500 SIP
phones will only use up to 125 IP resources at any given time.
The other 115 IP resources (240 minus 125) can be used for inbound and
outbound calls to a SIP gateway. Out of the 115, we want to reserve 100 for
inbound calls.
The IP Message Button value should be either the whole SIP address with type and
port number (sip:user@host:port) or just the user portion (user).
For example, setting IP Message Button to “9999” will allow IP phones to configure
their message buttons to this number as a convenience for users to retrieve their
voicemail. Users can also directly dial this number if a message button is not
available on the IP phone.
The Interaction Center will not ask for a user name and password if the user’s client
is active and set to an available status. You can change this behavior with the
Interaction Administrator Server parameter “Force Message Button Login”. This
defaults to “No” and if set to “Yes” will force users to enter their user id and
password.
Vendor Specific
Cisco: The parameter for configuring Cisco phones’ message button is
messages_uri. See the “SIP 3rd Party Component Application Note” for details. An
example would be 9999@172.16.132.16 where 172.16.132.16 is the Interaction
Center’s IP address and 9999 is the value set in the server parameter IP Message
Button.
Pingtel: The parameter for configuring Pingtel phones’ message button is
PHONESET_VOICEMAIL_RETRIEVE. See the “SIP 3rd Party Component Application
Note” for details. An example would be 9999@172.16.132.16 where 172.16.132.16
is the Interaction Center’s IP address and 9999 is the value set in the server
parameter IP Message Button.
Voicemail is handled automatically for managed phones. If you configure all your
phones as stations in Interaction Administrator, voicemail configuration is already
complete. Skip this section.
This section is for phones that are unmanaged phones (phones unknown to the
Interaction Center). These phones (or their proxies) will divert calls to the
Interaction Center for voicemail gathering.
30.1 Logic
The diverted SIP message will have:
• URI: sip:voicemail@204.180.46.185
• diversion header(s) original destination, and diversion header(s) divert
reason: CC-Diversion: <sip:5858652@siptest.wcom.com>;reason=no-answer
Notes:
• CC_Diversion header is equivalent to the Diversion header
• If multiple diversion headers are received (or multiple entries in a single
diversion header), the top most header (or first entry) is the last diverted
user.
In CIC 2.2 SR-B/EIC 2.2 GA, tel address schemes were changed
to sip schemes and would never appear. In CIC 2.2 SR-C/EIC
2.2 SR-A, and 2.2 CIC SR-B/2.2 EIC GA hot fixes 1206, 1203,
1212, 1213: For tel address scheme (addresses that start with
“tel:”), the address will look like tel:number, with no port
number.
In CIC 2.2 SR-B/EIC 2.2 GA, this attribute is only set if the URI
matches the IP VoiceMail Direct server parameter.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1178 and 1146 and in CIC
2.2 SR-C/EIC 2.2 SR-A, this attribute is set if the Diversion
header is present.
In CIC 2.2 SR-A/EIC 2.2 GA, this is the exact address in the
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1178 and 1146 and in CIC
2.2 SR-C/EIC 2.2 SR-A, for sip address scheme (addresses that
start with “sip:”), type and port number are added if not present
in the header (sip:user@host:port).
In CIC 2.2 SR-B/EIC 2.2 GA, tel address schemes were changed
to sip schemes and would never appear. In CIC 2.2 SR-C/EIC
2.2 SR-A, and 2.2 CIC SR-B/2.2 EIC GA hot fixes 1206, 1203,
1212, 1213: For tel address scheme (addresses that start with
“tel:”), the address will look like tel:number, with no port
number.
In CIC 2.2 SR-B/EIC 2.2 GA, this attribute is only set if the URI
matches the IP VoiceMail Direct server parameter.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1178 and 1146 and in CIC
2.2 SR-C/EIC 2.2 SR-A, this attribute is set if the Diversion
header is present.
In CIC 2.2 SR-B/EIC 2.2 GA, this attribute is only set if the URI
matches the IP VoiceMail Direct server parameter.
In a hot fix and in CIC 2.2 SR-C/EIC 2.2 SR-A, this attribute is
set if the Diversion header is present.
In CIC 2.2 SR-B/EIC 2.2 GA, this attribute is only set if the URI
matches the IP VoiceMail Direct server parameter.
In a hot fix and in CIC 2.2 SR-C/EIC 2.2 SR-A, this attribute is
set if the Diversion header is present.
Notes:
• Eic_RedirectionTn contains the whole SIP address with type and port
(sip:user@host:port) of the SIP message URI.
• The handlers check if the IP VoiceMail Direct server parameter equals the
whole SIP address (sip:user@host:port) in Eic_RedirectionTn OR just the SIP
address user portion (user) in Eic_RedirectionTn.
• If there is a match, the Interaction Center will route the call to the user’s
mailbox that has the Eic_RedirectingTn (or the user portion) configured as the
user’s extension (in Hot fix for CIC 2.2 SR-C, in EIC 2.2 SR-A and CIC 2.2 SR-
D) or Attribute 2 in the user configuration.
The IP VoiceMail Direct value should be either the whole SIP address with type and
port number (sip:user@host:port) or just the user portion (user).
Your phones or proxies must be configured to send the calls to the number
configured as the IP VoiceMail Direct number.
Configure, in Attribute 2 in the user configuration in Interaction Administrator, the
address in the diversion header.
Before CIC 2.2 SR-B/EIC 2.2 GA, the attribute 2 value must match exactly the SIP
address received in the diversion header.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1178 and 1146, the attribute 2 value should
EITHER be the user portion (user) OR the whole SIP address with type and port
number (sip:user@host:port) . The address received in the diversion header will get
“sip:” and port number appended (if necessary) to it before the compare occurs.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1206, 1203, 1212, 1213 and CIC 2.2 SR-C/EIC
2.2 SR-A, the attribute 2 value should be:
For sip scheme (addresses that start with “sip:”), attribute 2 should be EITHER the
user portion (user) OR the whole SIP address with type and port number
(sip:user@host:port).
For tel scheme (addresses that start with “tel:”), attribute 2 should be EITHER the
number portion (number) OR the whole SIP address with type (tel:number). There
is no user or port number with tel addresses.
Make sure you publish the new handlers that are hot fixes in 2.2 SR releases.
Insure the voice form on each workstation has “View” | “Control Message Waiting
Indicator” selected. This option is selected from the voice form when a voicemail is
opened. This is needed to turn the MWI off.
Currently, the phone that the client is logged on will light. If no Interaction Client is
active, then the default workstation of the user will light. If no default workstation,
then attribute 3 will be used. This logic is in System_MessageLight.ihd and can be
changed on a site by site basis.
The IP Managed Phone Shortcut value should be either the whole SIP address with
type and port number (sip:user@host:port) or just the user portion (user).
For example, setting IP Managed Phone Shortcut to “*” or “123” will allow IP phones
to dial this number as a convenience to get to the main IVR for managed phones.
Note that the phones must be able to dial this number (some IP phones do not
consider a “*” as a dialed number).
33.1 Central Site Only, Primary Interaction Center Only, Cisco IP Phones
Specifications
Interaction Center
• Primary Interaction Center’s IP Address: 1.1.1.1
In Interaction Administrator, create a station (see section 0, “Creating and
Configuring SIP stations in Interaction Administrator”)
• IP Address: 2.2.2.2
• line1_name: 7101
• outbound_proxy:
If using a proxy, set to the IP address of the proxy.
33.2 Central Site Only, Primary and Backup Interaction Centers, Cisco IP
Phones
Specifications
Is Switchover being utilized (primary and backup Interaction Center servers)? Yes
Interaction Center
• Primary Interaction Center’s IP Address: 1.1.1.1
• Backup Interaction Center’s IP Address: 9.9.9.9
In Interaction Administrator, create a station (see section 0, “Creating and
Configuring SIP stations in Interaction Administrator”)
• IP Address: 2.2.2.2
• line1_name: 7101
• outbound_proxy:
If using a proxy, set to the IP address of the proxy.
If not using a proxy, leave empty.
• proxy1_address: Set to the IP address of the Primary Interaction Center
(1.1.1.1).
• proxy_backup:
If using a single proxy, leave empty.
If using two proxies, set to the backup.
If not using a proxy, set to the IP address of the Backup Interaction
Center (9.9.9.9).
Interaction Client
The computer that is running the client should be named the same as the
station (“Station7101) OR use the /w=Station7101 (Station Name) flag (see
section 26.1, “Associating the Interaction Client with a Station)
Proxy Server
• Optional in this configuration if phones are capable of routing
• Would be configured to send SIP calls to the Interaction Center backup if
the Interaction Center primary is not reachable.
• Could be used to route calls to the remote site’s gateways for long
distance savings.
33.3 Central and Remote Site (no remote gateways), Primary Interaction
Center Only, Cisco IP Phones
Specifications
Interaction Center
• Primary Interaction Center’s IP Address: 1.1.1.1
In Interaction Administrator, create a station (see section 0, “Creating and
Configuring SIP stations in Interaction Administrator”). In this configuration,
the stations are the same at the Central and Remote sites.
• IP Address: 2.2.2.2
• line1_name: 7101
• outbound_proxy:
If using a proxy, set to the IP address of the proxy.
If not using a proxy, leave empty.
• proxy1_address: Set to the IP address of the Primary Interaction Center
(1.1.1.1).
• proxy_backup:
If using a single proxy, leave empty.
If using two proxies, set to the backup.
If not using a proxy, leave empty.
Interaction Client
33.4 Central and Remote Site (with remote gateways), Primary Interaction
Center Only, Cisco IP Phones
Specifications
Interaction Center
• Primary Interaction Center’s IP Address: 1.1.1.1
In Interaction Administrator, create a station (see section 0, “Creating and
Configuring SIP stations in Interaction Administrator”). In this configuration,
the stations are the same at the Central and Remote sites.
• IP Address: 2.2.2.2
• line1_name: 7101
• outbound_proxy:
If using a proxy, set to the IP address of the proxy.
If not using a proxy, leave empty.
• proxy1_address: Set to the IP address of the Primary Interaction Center
(1.1.1.1).
• proxy_backup:
If using a single proxy, leave empty.
If using two proxies, set to the backup.
If not using a proxy, set to the gateway at the remote site.
• Dial plan
If number dialed is local, route call to gateway at remote site.
If number dial is emergency (911), route call to gateway at remote
site.
Interaction Client
The computer that is running the client should be named the same as the
station (“Station7101) OR use the /w=Station7101 (Station Name) flag (see
section 26.1, “Associating the Interaction Client with a Station)
Proxy Server
• Optional in this configuration if phones are capable of routing
• Could be used to route calls to the remote site’s gateways for long
distance savings.
33.5 Central and Remote Site (with remote gateways), Primary and
Backup Interaction Center Only, Cisco IP Phones
Specifications
Is Switchover being utilized (primary and backup Interaction Center servers)? Yes
Interaction Center
• Primary Interaction Center’s IP Address: 1.1.1.1
• Backup Interaction Center’s IP Address: 9.9.9.9
In Interaction Administrator, create a station (see section 0, “Creating and
Configuring SIP stations in Interaction Administrator”). In this configuration,
the stations are the same at the Central and Remote sites.
• IP Address: 2.2.2.2
• line1_name: 7101
• outbound_proxy:
Set to the IP address of the proxy.
• proxy1_address: Set to the IP address of the Primary Interaction Center
(1.1.1.1).
• proxy_backup:
If using a single proxy, leave empty.
If using two proxies, set to the backup.
• Dial plan
If number dialed is local, route call to gateway at remote site.
If number dial is emergency (911), route call to gateway at remote
site.
• IP Address: 2.2.2.2
• line1_name: 7101
• outbound_proxy:
If using a proxy, set to the IP address of the proxy.
If not using a proxy, leave empty.
• proxy1_address: Set to the IP address of the Interaction Center (1.1.1.1).
Vendor Specific
Intel/Dialogic software (HMP) does not support
these parameters.
AudioCodes hardware boards support these
parameters.
35 Troubleshooting
35.1 Tracing
The flowing are trace topics for the possible different issues. Use
traceconfig.exe to set these topics to 61, which is one of the Notes levels.
• TsServer | TsServer – Turn on for any TS related investigation.
Turn on the following depending on this problem:
• TsServer | SIPEngine - All SIP protocol related problems. This contains the
SIP protocol messages and SIP engine state information. This one is very
important if debugging problems with SIP end-devices (phones, gateways,
proxies, etc)
• TsServer | SIPUrl - SIP URL problems. This shouldn't be turned on unless
directed.
• TsServer | SIPMessage - SIP parser problems. This shouldn’t be turned on
unless directed.
• TsServer | AudioHub - All audio related problems.
35.3 Echo
• Understand echo (see section 9.2 “Echo”).
• Note the direction of the echo. Are the agents hearing their own voice or are the remote
callers hearing the echo?
• Use the RTP Audio Monitor and Analysis Guide to record the audio directly from the
network (see section 36.5 “RTP Audio Monitor and Analysis Guide”)
• Adjust the gain parameters. For EIC 2.2 SR-A and CIC 2.2 SR-C this is done with
server parameters AudioCodes Network Gain and AudioCodes Bus Gain. For CIC 2.2
SR-D, these values are in Interaction Administrator line, station, and global station
configuration dialog boxes.
1. If the phones are transmitting too “hot”, the agent could hear their own voice. In
this case, the Network Gain can be slowly turned down to help overcome the
phone’s transmit levels.
• Some head phones generate echo.
• Some phones generate echo if their volume is turned too high.
• DTMF, when sent inband, might not retain it frequencies through the network.
This is especially true when compression is used. Out-of-band DTMF (RFC2833) is
available. If having DTMF problems, the options are:
• Try using RFC2833. This is the best solution. For configuring DTMF type
to RFC2833, see sections 17.2.1 “SIP Configuration Page” for lines and
19”Creating and Configuring SIP stations in Interaction Administrator”.
• Try using G.711.
• If using Intel HMP, make sure the IP address of your network card is the same
that is in the registry (see section 16.3.2 “IP addresses”).
35.5.3 DTMF from Managed Phone not being recognized by remote system
• See IVR DTMF Recognition Problem above. The phones and Interaction Center
should be set to RFC2833 if possible.
35.6 Miscellaneous
35.6.1 Selecting hold on the Interaction client puts the call in Held, put the IP
phone still shows connected.
• A SIP call can be held by either endpoint, and since the phone did not put the call
on hold, it can not take it off hold (since its side was never held). Thus, the hold
state will not show on the phone. The same goes for a call held by the phone
and unheld by the client. It must still be unheld by the phone for the complete
audio path to be connected.
35.6.3 External Call made from SIP phone hears IVR rather than making the
intended call
• Make a call from the phone to an external number. If you hear the IVR then the
call is being treated as a normal inbound call and is not being identified as a call
from a managed station. Solution: Check the “Line Details” page, then verify that
the call’s Number field exactly matches the value configured as the SIP
Identification Address of the station in Interaction Administrator (see section 0
“Creating and Configuring SIP stations in Interaction Administrator”).
35.6.4 Internal Call made from SIP phone is placed correctly, but does not show
up on client.
• Make a call from the phone to an internal number. If the call completes correctly
but you do not see the call on the Interaction Tab in the client, then this call is
not being identified as a call from a managed station. Solution: Check the “Line
Details” page, then verify that the call’s Number field exactly matches the value
configured as the SIP Identification Address of the station in Interaction
Administrator (see section 0 “Creating and Configuring SIP stations in Interaction
Administrator”).
35.6.5 Calls made from SIP phones do not show on Line Details Page
• If the call does not show on the Line Details page (see section Error! Reference
source not found. “Error! Reference source not found.”), the SIP message
in not making it to the Interaction Center Server. The Interaction Center must
35.6.6 Phone rings when I use the MakeCall button in the Interaction Client
• This is normal. The Interaction Center server must establish an audio path to the
SIP phone. This is accomplished by making a call to the phone. When you
make a call from a client, and your phone is a SIP phone (and not a SIP soft
phone running with the audio-enabled client), and you do not have a persistent
connection, then the IC server call the phone.
35.6.9 Microsoft Messenger window pops for every incoming call with using the
SIP enabled Interaction Client
• If using the Interaction Client with the audio option (section 26.2 “Configuring
the Interaction Client for Audio”, the Interaction Client must be started BEFORE
Microsoft Messenger. Messenger must have been loaded on the system, but does
not need to be active. If you desire Messenger and the client to run on the same
system, the client should be started before Messenger. If not, Messenger will
process the incoming calls to your station.
35.6.10 “Station Not Reached” error when making calls from the Interaction
Client (when using a SIP station)
• The Interaction Center must be able to contact the workstation. See section
26.2.1 “Special Messenger Considerations for SIP Enabled Interaction Client”.
• Verify that this station can be reached. Call if via the station’s Connection
Address in IA, ping it,…..
• Verify that the station is not set to “Do Not Disturb”. Each SIP station has unique
setting for this configuration.
This is a SIP limitation. If you select the Pickup, Listen, or MakeCall buttons in the
Interaction Client, the Interaction Center will first call your SIP device to establish an
audio path. Typically, your phone will ring and you must answer the phone (by
picking up the handset or hit the speaker button). There is no way, via the SIP
protocol, to make the phone “answer the call”. The Pickup, Listen, or Makecall
buttons on the Interaction Client will stay depressed until you pick up the handset, or
the call times out.
With Microsoft Messenger and the audio enabled Interaction Client: We have solved
this limitation by having the Interaction Client answer the call via Microsoft’s APIs.
36 Tools
36.3 NetIQ
There has been success with users using SIP over their home DSL or cable connection. Listed
below are potential problems with a setup over a cable or DSL connection:
• Most ISPs do not provide QOS. Your voice traffic is NOT guaranteed and either is your
bandwidth.
• For cable connections, upload speeds could be very low, even though download speeds are
high.
• DSL and cable connections are shared (at some point in the network). At 3PM when the kids
come home from school, the extra traffic can compromise your bandwidth.
36.4 Speakeasy
Test upload and download speeds: http://chi.speakeasy.net/