C-Dot Class-4 Softswitch Data Guidelines

C-DOT
CLASS-4 SOFTSWITCH
Data Guidelines
This document has been authored/modified by

Shalini Gupta
with contributions from

Renu Gupta
Rekha Jain
Arvind Kumar
TOPICS COVERED
 Breif about C4 Soft-switch

 SIP Data
 ISUP Data
 MG Data
 M3UA Data
 Number Analysis (CIDP, AIRP & Route Case)
 Basic Data For One SIP To SIP Call
 Basic Data For One SIP To ISUP Call
 Basic Data For One ISUP To SIP Call
 Few Troubleshooting Guidelines
 Sample Field data for AIRP
Introduction
 A Class 4 Softswitch (C4 SS) is a central device in a telecommunications network which
connects telephone calls from one subscriber to another across the telecommunication
network.
 Call could be between IP - IP subscriber or between IP – PSTN subscriber.
 Class-4 Softswitch also provides interconnection between IP network and PSTN by

providing interworking between SIP and ISUP signalling protocols.
 SIP Routes are addeded between C4 SS & Remote nodes which are in IP N/W.
 ISUP Routes are added between C4 SS & Remote Nodes which are in PSTN N/W.
● PSTN nodes are connected to C4 SS via Signalling Gateway(SG) & Media

Gateway(MG)
● C4 SS & SG are connected over SCTP association.
● Over SCTP, there will be M3UA (MTP3 User Adaptation). And over M3UA is ISUP
Protocol.
● C4 SS & MG are connected over MEGACO Protocol.
● Physical E1 Trunks are connected between MG & Remote PSTN nodes.
 C4 SS do Number Analyses using signalling information received in incoming

INVITE/IAM message, to route the call across the network.
 CIDP ( Collect Information Dial Plan) & AIRP ( Analyse Information Dial Plan) are
defined for Number Analyses.
 Its must to define CIDP & AIRP on Incoming Route, to route the call.
 Generally Called Party Number should be used as criteria, to route the call, though
C4 SS do provides number of options to decide the outgoing routing.
 In case it get must to use Calling Party Number(CGPN) to be used as criteria to route
the call, do bifurcation on any I/C Route AIRP on the basis of CGPN itself, which
give outcome of new AIRPs. In those new AIRP's scenarios use Called Party
Number(CDPN) & other parameters as criteria to route the calls.
Basic Commands to make C4 Data:
SIP Commands
ADD-SIP-ROUTE: It adds a SIP route to the call agent.
SYNTAX: ADD-SIP-ROUTE:ROUTE=<Route>, HOST=<IP address>,

MAX-CALLS=<maximum number of connections> [, PORT=<port number>]
[, PROXY=<YES|NO>]
[, CIDP=<CI dial plan>]
[, AIRP=<AI routing plan>]
[, ORNP=<OR numbering plan>]
[, OPTIONS=<NONE|SEND|RECV|SENDRECV>]
[, CLIR=<RESTRICTED | ALLOWED | NONE>]
[, CLIP=<TRUE | FALSE>]
[, CLI-TRUSTED=<TRUE | FALSE>]
[, REMOTE-NODE=<CdotC5|C4SSP|BSNL|POI>]
[, SIPT=<YES | NO>]
[, ALLOW-REDIRECT=<YES|NO>]
Other SIP ROUTE Commands:

1)UBL-SIP-ROUTE : It locally maintenance unblocks a SIP route.
2)BLK-SIP-ROUTE : It locally maintenance blocks a SIP route.
3)MOD-SIP-ROUTE : It modifies a SIP route.
4)LST-SIP-ROUTE : It lists one or more SIP routes.
5)DIS-SIP-ROUTE : It displays a SIP route.
6)DEL-SIP-ROUTE : It deletes a SIP route.
SIP Route Status
The status field is either Unblocked (InService) or Blocked (OOS). If the status is
blocked, no calls can be made on that route.
There is a further status indicator in the form LMX_HX_RMX. LM stands for local
maintenance, H stands for hardware and RM stands for remote maintenace. The X is
either B for blocked or U for
unblocked. A route may be blocked in any combination of the three blocking
mechanisms.
Local Maintenance Blocked

A local maintenance block occurs when an operator has issued the BLK-SIP-ROUTE on
the call agent to block the route.
Hardware Blocked
In case while adding SIP Route, value of OPTIONS is given as “sendrecv”, C4 SS do
send OPTIONS message towrads remote node. Now when either the remote SIP entity
is not responding or it has sent a "403 FORBIDDEN" response to OPTIONS message,
status is set as Hardware Blocked (HU)
Remote Maintenance Blocked

In case while adding SIP Route, value of OPTIONS is given as “sendrecv”, C4 SS do
send OPTIONS message towrads remote node. Now when the remote SIP entity sent a
"480 TEMPORARILY UNAVAILABLE" response to OPTIONS message.
In case while adding SIP Route, value of OPTIONS is given as “recv”, C4 SS waits for
the OPTIONS message from remote SIP entity. If not received, status get set as Remote
Maintenace Blocked (RMB).
Status Examples :
 Status = Blocked (OOS) LMB_HU_RMU
 Status = Unblocked (InService)
 Status = Blocked (OOS) LMU_HB_RMU
 Status = Blocked (OOS) LMU_HU_RMB
SIP Configuration
DIS-SIP-CONFIG : It displays the details of the SIP configuration.
CONFIG-ITEM VALUE MODIFIABLE

--------------------------------------------------------------------------------
T1 500 ms YES
T2 4000 ms YES
OPTIONS-HEARTBEAT 60 s YES
OPTIONS-RETRY 20 s YES
SESSION-INTERVAL 1800 s YES
MIN-SESSION-INTERVAL 0 s YES
MOD-SIP-CONFIG : It modifies the SIP configuration items.

ISUP Route Commands
ADD-ISUP-ROUTE : It adds an ISUP route.
Syntax:
ADD-ISUP-ROUTE:ROUTE=<Route>,
OPC=<CA point code>,
DPC=<remote point code>
VARIANT=<isup protocol variant>
[, CIDP=<CI dial plan>]
[, AIRP=<AI routing plan>]
[, ORNP=<OR numbering plan>]
[, CLIR=<RESTRICTED | ALLOWED | NONE>]
[, REMOTE-NODE=<BSNL|POI>]
[, CLIP=<TRUE | FALSE>]
[, CLI-TRUSTED=<TRUE | FALSE>]
[, CODEC=<PCMU | PCMA | G721 | G729 | G7231>]
[, CODEC-CHOICE=<MANDATORY | PREFERRED | NONE>]
[, VOICE-CALLS=<voice calls count>]
Example :
ADD-ISUP-ROUTE: REMOTE-NODE=POI, CLI-TRUSTED=TRUE,
VARIANT=ETSI_V2, CLIR=NONE, CODEC-CHOICE=NONE, DPC=11736,
VOICE-CALLS=4096, OPC=8427, ROUTE=5111109, CLIP=TRUE
Other ISUP ROUTE Commands:
1)DIS-ISUP-ROUTE : It displays the details of an ISUP route.
2)MOD-ISUP-ROUTE : It changes the characteristics of an ISUP route.
3)DEL-ISUP-ROUTE : It deletes an ISUP route.
4)LST-ISUP-ROUTE : It lists the ISUP routes.

For each configured route, this command will display following information :
 ROUTE
 VARIANT
 OPC
 DPC
 CCTS
 IDLE&UBL
 BUSY
 PAUSED
 PEND
 DBL
 LOC-MT
 REM-MT
 L&R-MT
 LOC-HW
 REM-HW
 L&R-HW
ISUP Circuit Commands
ADD-ISUP-CIRCUIT : It adds circuits to an ISUP route.

An ISUP circuit is associated with a particular ISUP route and an endpoint on the
trunking media gateway. When a circuit is added, its initial state is disabled. The
circuit must be enabled before it can be used for a call.
Syntax:
ADD-ISUP-CIRCUIT: ROUTE=<Route>, CIC=<base Circuit>,
ENDPOINT=<base MG Endpoint>
[, END-CIC=<CIC of last circuit to add> | COUNT=<Number of
circuits to add>]
Examples :
ADD-ISUP-CIRCUIT: ENDPOINT="SH1/BR1/TR1/TS1@MG1", COUNT=1,
CIC=1, ROUTE=18
CIC=2, ROUTE=18
CIC=3, ROUTE=18
OR

CIC=1, ROUTE=18
LST-ISUP-CIRCUIT: It lists the ISUP circuits. For each configured circuit on
asked route, this command will display following information :
 ROUTE
 OPC
 DPC
 CIC
 ENDPOINT
 STATUS
 MT-STATUS
 HW-STATUS
Any one of following will be “STATUS” value:

● DISABLED : The circuit is disabled. Use “EBL-ISUP-CIRCUIT”
command to enable the circuit.
● PAUSED : A circuit is paused when the DPC is inaccessible. It

means either SCTP association between C4 SS & SG, is down OR
MTP3 link between SG & DPC is down.
● PENDING : A circuit is pending, if no response received from PSTN

side of sent RSC/GRS message.
● IDLE : The circuit is free and ready to be used for handling calls.
● OBUSY : The circuit is in being used as the originating circuit in a

call.
● TBUSY : The circuit is in being used as the terminating circuit in a

call.
● BLOCKED : The circuit is locally or remotely maintenance or

hardware blocked
MT-STATUS:
This field consists of two values separated by an ‘&’ - the local
maintenance block status and the remote maintenance block status
respectively.
The Local Maintenance block values are:

● NONE - not locally maintenance blocked
● LOC - locally maintenance blocked
The remote maintenance block values are:

● NONE - not remotely maintenance blocked
● REM - remotely maintenance blocked
HW-STATUS:
This field consists of two values separated by an ‘&’ - the local hardware
block status and the remote hardware block.
The local hardware block values are:

● NONE - not locally hardware blocked.
● LOC - locally hardware blocked.
The remote hardware block values are:

● NONE - not remotely hardware blocked.
● REM - remotely hardware blocked.
Example :
ROUTE OPC DPC CIC ENDPOINT STATUS MT-STATUS HW-STATUS
18 5555 1111 1 MG_EP Dbl NONE&NONE NONE&NONE
19 5555 2222 1 MG_EP Paused NONE&NONE NONE&NONE
Other ISUP Circuit Commands:
1)EBL-ISUP-CIRCUIT : It enables one or more circuits on an existing ISUP
route in the call agent. A circuit can only be Enabled if it is in Disabled state.
2)DBL-ISUP-CIRCUIT: It disable one or more circuits on an existing ISUP

route in the call agent. A circuit can only be Disabled if it is in one of the
following states:
• paused,
• locally maintenance blocked
• locally hardware blocked state
3)DEL-ISUP-CIRCUIT: It deletes one or more circuits on an existing ISUP

route in the call agent. A circuit can only be Deleted if it is in Disabled state.
4)BLK-ISUP-CIRCUIT: It initiates the local maintenance blocking of one or

more circuits on an existing ISUP route in the call agent. For a circuit Local
Maintenance Blocking can be initiated ,only if circuit was not in “Paused” Or
“Disabled” state.
5)UBL-ISUP-CIRCUIT : It initiates the local maintenance unblocking of one

or more circuits on an existing ISUP route in the call agent. For a circuit local
maintenance unblocking can be initiated, only if circuit was not in “Paused”
Or “Disabled” state.
6)RES-ISUP-CIRCUIT : It initiates the reset of one or more circuits on an

existing ISUP route in the call agent. For a circuit Reset can be initiated, only
if circuit was not in “Paused” / “Disabled” / “Locally Hardware Blocked”
state. Any existing Locally Maintenance Blocked circuits in the range will
become Locally Maintenance Unblocked as a result of the reset.
ISUP Configuration
DIS-ISUP-CONFIG : It displays the details of all the ISUP congiguration

items (ISUP )timers in the call agent. T1 to T36 timers are defined
Example:
T1 Time to wait for a response to a release message sent.
T2 Time to wait for a resume message after a suspend message
received.
T3 Time to wait for overload message to expire.
T4 Time to wait for call modification complete message.
T5 Time to wait for a response to initial release message sent.
T6 Time to wait for a resume message after a suspend (network)
message received.
T7 Time to wait for a response (for example, ACM, ANS, or CON)
to the latest address message sent.
T9 Time to wait for answer of outgoing call after ACM message
received.
MOD-ISUP-CONFIG: It modifies the ISUP configuration items.

Only T7, T9, T10 & T35 timers can be modified.
MEGACO Commands
ADD-VENDOR : It adds a vendor to the call agent.

A Vendor encapsulates configuration specific to a Vendor's equipment.
Syntax
ADD-VENDOR: VENDOR=vendor_name ,PROTOCOL=<MEGACO>
[, CLEAR-MODE=<CISCO|AUDIOCODES|OTHER>]
Example :
ADD-VENDOR: CLEAR-MODE=OTHER, PROTOCOL=MEGACO,
VENDOR=CDOT
ADD-MG : It adds an MEGACO Media Gateway to the call agent.

On media gateway creation, the call agent attempts to resolve the provided domain
name to an IP address.
Syntax
ADD-MG: MG-TYPE=<media gateway protocol>,
MG=<media gateway name>,
NC=<naming convention>,
VENDOR=<vendor name>
[, PREFERRED-MS=<media server name> ];
Example :
ADD-MG: NC=CDOT_E1, MG="MG_BPLCORE", PROTOCOL=MEGACO,
VENDOR=CDOT
Other MG Commands
LST-MG : It lists the details and status of one or more Media Gateway(s) in the
Call Agent.
Example :
MG PROTOCOL NC VENDOR PREFERRED-MS STATUS
---------------------------------------------------------------------------------------------------------------------
192.168.109.23 MEGACO CDOT_E1 CDOT OUT OF SERVICE
DEL-MG :This command may be used to delete a Media Gateway .

This command will fail if any endpoints are configured on the Media
Gateway.
MOD-MG: This command may be used to modify the parameters associated

with an MEGACO Media Gateway in the Call Agent.
M3UA MMI Commands
M3UA OVERVIEW
The M3UA protocol allows ISUP and other SS7 signalling traffic to be transported between the
call agent and an SS7 MTP network via a bridging entity called a signalling gateway process
(SGP).
The SS7 MTP network can be the public switched telephone network (PSTN) or any other SS7
MTP network. A set of SGPs providing the same bridging service form a signalling gateway. The
call agent uses SCTP protocol to connect to the SGPs.
Once an Call Agent is connected to an SGP, it requests the ability to send and receive SS7
messages that match one or more routing keys. A routing key consists of a local point code and a
set of remote point codes.
For an incoming SS7 message, a message matches a routing key if the message’s destination point
code (DPC) is equal to the local point code of the routing key and the message’s originating point
code (OPC) is equal to one of the routing key’s remote point codes.
Similarly, for an outgoing SS7 message, a message matches a routing key if the message’s OPC is
equal to the local point code of the routing key and the message’s DPC is equal to one of the
routing key’s remote point codes.
Each SS7 point code is unique within a specific MTP3 network.
Each MTP3 network is assigned a network indicator (NI) which can have one of four values:
0 => International MTP3 network
1 => Spare (reserved for international use only) 2 => National MTP3 network
3 => Reserved for national use
An NI is assigned to each routing key. Each outgoing SS7 message matching the routing key will
have the routing key’s NI value inserted in its MTP3 NI field. If the call agent M3UA has to send
and receive SS7 traffic to and from a particular SGP for a particular routing key, an M3UA route is
created.
A route has a routing context. A routing context is an integer identifier that the call agent and an
SGP use to uniquely identify the routing key associated with an M3UA route.
M3UA MMI Commands
ADD-M3UA-SG : It adds a signalling gateway into the M3UA subsystem of the call agent.
Syntax
ADD-M3UA-SG: SG=name
[, M3UA-VERSION={DRAFT_V5 | RFC}]
[, MTP={ITU | ANSI_92 | ANSI_96}];
ADD-M3UA-SGP : A signalling gateway process (SGP) is a process instance of a signalling gateway.

An SGP is identified by the name of the SGP in conjunction with the name of the signalling gateway. If
the local or remote port values are not specified, the default value of 2905 is used.
Syntax
ADD-M3UA-SGP: SGP=sgpName, SG=sgName, LOCAL-ADDRESS=hostName
[, LOCAL-PORT=port], REMOTE-ADDRESSES=hostNames [, REMOTE-PORT=port];
ADD-M3UA-RK : It adds a routing key that can associate a set of remote point codes with a local
point code of the call agent.
A routing key associates a set of remote point codes with a local point code of the call agent. It is used
when determining the signalling gateway through which the MTP traffic should be routed through.
Multiple routing keys can be added for the same local point code. Remote point codes can be
associated with more than one routing key, so long as the different routing keys have different local
point codes.
Syntax
ADD-M3UA-RK: ROUTING-KEY=name,
LOCAL-PC=pointCode
[, REMOTE-PC-SET=pointCodeSet]
[, NI=MTP3_NetworkIndicator];
ADD-M3UA-ROUTE : It adds a route to an SGP for a particular routing key.

Syntax
ADD-M3UA-ROUTE: ROUTING-KEY=rkName, SGP=sgpName, SG=sgName,
ROUTING-CONTEXT=context;
The routing context is associated with this routing key on this route.
Sample M3UA Data :
ADD-M3UA-SG: MTP=ITU, M3UA-VERSION=RFC, SG=SG_BPL
ADD-M3UA-SGP: SGP=SGP2_BPL, LOCAL-PORT=2905, REMOTE-ADDRESSES="SGP2_BPL",

REMOTE-PORT=2905, LOCAL-ADDRESS=myAsp, SG=SG_BPL
ADD-M3UA-SGP: SGP=SGP1_BPL, LOCAL-PORT=2905, REMOTE-ADDRESSES="SGP1_BPL",

REMOTE-PORT=2905, LOCAL-ADDRESS=myAsp, SG=SG_BPL
ADD-M3UA-RK: NI=2, ROUTING-KEY=rk_2709, LOCAL-PC=2709, REMOTE-PC-SET="2638,
11772, 2700, 2720, 2740, 2760, 5425, 7275, 8649, 10874, 11620"
ADD-M3UA-RK: NI=2, ROUTING-KEY=rk_8427, LOCAL-PC=8427, REMOTE-PC-SET="5184,

11566, 11736, 11762, 7275, 7486, 7565, 7580, 8549, 9247, 9469, 11257, 11286"
ADD-M3UA-ROUTE: SGP=SGP2_BPL, ROUTING-CONTEXT=1, ROUTING-KEY=rk_8427,

SG=SG_BPL

SG=SG_BPL

SG=SG_BPL
ADD-M3UA-ROUTE: SGP=SGP1_BPL, ROUTING-CONTEXT=2, ROUTING-

KEY=rk_2709, SG=SG_BPL
Number Analysis Overview
************************
Class-4 softswitch provides the capabilities for analysis of signalling information to route the call
across the network.
Number Analysis is done with the help of two dial plans, CIDP and AIRP.
CIDP : The Collect Information dial plan performs number length analysis on the incoming
destination number to determine when a complete number, or enough digits to proceed to
Analyse Information, has been received.
AIRP : The Analyse Information routing plan uses various criteria to find the outbound Routing
Case. AIRP can do modify (if required) the call signalling information too.
ROUTING CASE : A routing case is a set of routes to a particular destination. The selection of the
route within the routing case is based on prority and percentage distribution.
BLACK And WHITE Lists : Black and white lists are lists of calling or called party numbers that
can be matched against actual calling or called parties at runtime for the purpose of
providing (white list) or preventing (black list) access to a particular service.
ORNP : The Outbound Routing numbering plan can be defined for outbound Route. The outgoing call
data may be modified using the ORNP. ORNP uses various criteria (including call signalling
information after any modifications by AIRP) to analyse a call, to modify (if required) the call
signalling information. ORNP is not mandatory to define to route the call.
NOTE:
Within C4 Soft Switch, all CI dial plans, AI routing plans, OR numbering plans, black and white lists
and routing cases are managed independently. For example, the same CI dial plan can be used by many
routes and the same routing case can be selected from many AI routing plans.
Collect Information
A CI dial plan analyses call data to determine if sufficient data is available for the AI
routing plan to carry out further analysis.
ADD-CIDP : It adds a new inactive CI dial plan to the call agent.
Syntax
ADD-CIDP: CIDP=ci_dialplan_name;
Other CIDP Commands:

1)LST-CIDP: It display following information for each CI dial plan:
 The CIDP column displays the CI dial plan name.
 The MODIFICATION column shows the current modification state
of the dial plan. Valid states are:
• NEW, there is an inactive dial plan but no active dial plan
• NOT MODIFIED, the inactive dial plan is identical to the active

dial plan
• MODIFIED, the inactive dial plan differs to the active dial plan.
• DELETED, there is an active dial plan, but the inactive dial plan
has been deleted.
 The STATUS column indicates whether or not the dial plan is in use
by a route.
Example:
CIDP MODIFICATION STATUS

---------------------------------------------------
CIDP1 DELETED IN USE
CIDP2 NOT_MODIFIED IN USE
2)RES-CIDP: It resets an existing inactive CI dial plan. So it can be used to
undo changes made to an inactive CI dial plan so that it is identical to the active
CI dial plan.
3)UPD-CIDP: If no active version of inactive CIDP plan with same name is

existed, create the active version. Also the scenarios from the
inactive CI dial plan are copied to the active CI dial plan.
3)DEL-CIDP : It removes the inactive CI dial plan from the call agent.
To delete the corresponding active CI dial plan, run UPD-CIDP also.
But if that active CIDP is in use, UPD-CIDP command will get fail with reason
“Active CIDP is in use and cannot be deleted.”
ADD-CIDP-SCEN: It adds a scenario to an existing CI dial plan.
Syntax
ADD-CIDP-SCEN: CIDP=<cidp name>,
SCEN=<scenario name>,
CDPN=<digit pattern>,
LENGTH=<minimum length>;
A CI dial plan scenario consists of the following criteria:

 A string that defines a called party number (CDPN) digit pattern.
 A numeric value that defines the minimum length of the CDPN
To reflect these scenario’s in active CIDP, run the UPD-CIDP

command.
Example:
ADD-CIDP-SCEN: CIDP=DEFAULT_CIDP, CDPN=1, LENGTH=3,
SCEN=EMERGENCY
ADD-CIDP-SCEN: CIDP=DEFAULT_CIDP, CDPN=00[123456789],
LENGTH=10, SCEN=ISD
ADD-CIDP-SCEN: CIDP=DEFAULT_CIDP, CDPN=[2345678],
LENGTH=6, SCEN=LCL
Other CIDP-SCEN Commands
1)LST-CIDP-SCEN : It lists the scenarios in an existing active or

inactive CI dial plan.
2)MOD-CIDP-SCEN: It modifies an existing scenario in an inactive

CI dial plan.
3)DEL-CIDP-SCEN: It deletes an existing scenario from an inactive

CI dial plan.
Analyse Information
An Analyse information (AI) routing plan analyses the call data and determines
an outcome for a call during setup.
ADD-AIRP: It adds an inactive AI routing plan to the call agent.
Syntax
ADD-AIRP: AIRP=airp_name [, DEFAULT-REASON=release_reason_name];
Other AIRP Commands:
1)LST-AIRP: It display following information for each AI dial plan:

 The AIRP column displays the AI routing plan name.
 The DEFAULT-REASON column displays the name of the default

release reason for the AI routing plan.
 The MODIFICATION column shows the current modification state

of the routing plan. Valid states are:
• NEW, there is an inactive routing plan but no active routing plan.
• NOT MODIFIED, the inactive routing plan is identical to the
active routing plan.
• MODIFIED, the inactive routing plan differs to the active routing
plan
• DELETED, there is an active routing plan, but the inactive
routing plan has been deleted.
 The STATUS column indicates whether or not the routing plan is in

use by a route.
Example:
AIRP DEFAULT-REASON MODIFICATION STATUS
-------------------------------------------------------------------------------------------------------------------
AIRP1 UNALLOCATED_UNASSIGNED_NUMBER DELETED IN USE
AIRP2 UNALLOCATED_UNASSIGNED_NUMBER MODIFIED IN USE
2)MOD-AIRP : It modifies the default reason of an existing inactive AI routing
plan.
3)RES-AIRP: It resets an existing inactive AI dial plan. So it can be used to

undo changes made to an inactive AI dial plan so that it is identical to the
active AI dial plan.
4)UPD-AIRP: If no active version of inactive AIRP plan with same name is

existed, create the active version. Also the scenarios from the inactive AI dial
plan are copied to the active AI dial plan. Internal structures to optimise the
analysis process are created from the scenarios associated with the active AI
routing plan.
5)DEL-AIRP : It removes the inactive AI dial plan from the call agent.
To delete the corresponding active AI dial plan, run UPD-AIRP also.
But UPD-AIRP command will get fail, if another entity, such as a route in the
call agent, refers to the AI routing plan.
ADD-AIRP-SCEN: It adds an existing scenario to an AI routing plan.
Syntax
ADD-AIRP-SCEN: AIRP=ai_rp_name, SCEN=scenario_name,
CRITERIA=list_of_criteria,
OUTCOME=outcome [, MODIFICATION=list_of_modifications];
The criterias that can be specified are:
CDPN=<digit pattern>
CDPN-NAI=<nai value>
CDPN-NPI=<npi value>
CDPN-LENGTH=<length value>
CDPN-BWLIST=(<bwlist name>, <present>)
CGPN=<digit pattern>
CGPN-APRI=<apri value>
CGPN-NAI=<nai value>
CGPN-NPI=<npi value>
CGPN-LENGTH=<length value>
CGPN-BWLIST=(<bwlist>, <present>)
CLIR=<clir value>
CPC=<cpc value>
EMG-CALL=<emg call flag>
DAY-OF-WEEK=<dow value>
TIME-OF-DAY=hh:mm-hh:mm
The outcomes that can be specified are:
RTCASE=<routing case>
AIRP=<airp name>
REASON=<release reason>
The modifications that can be specified are:

CDPN=<mod digit string>
CDPN-NAI=<nai value>
CDPN-NPI=<npi value>
CGPN=<mod digit string>
CGPN-NAI=<nai value>
CGPN-NPI=<npi value>
CGPN-APRI=<apri value>
CPC=<cpc value>
EMG-CALL=<emg call flag>
TMR=<tmr value>
INSERT=(<cxpn>, <start pos>, <mod digit string>)
REMOVE=(<cxpn>, <start pos>, <end pos>)
REPLACE=(<cxpn>, <start pos>, <end pos>, <mod digit string>)
APPEND=(<cxpn>, <mod digit string>)
Criteria Precedence
CGPN (longer is higher precedence) (HIGHEST)

CDPN (longer is higher precedence)
CGPN-LENGTH
CDPN-LENGTH
CDPN-NAI
CGPN-NAI
CDPN-NPI
CGPN-NPI
CGPN-APRI
CPS
NC
CLIR
CPC
TMR
CGPN-BWLIST
CDPN-BWLIST
DAY-OF-WEEK (shorter range is higher precedence)
TIME-OF-DAY (shorter time range is higher precedence) (LOWEST)
Other AIRP-SCEN Commands
1)LST-AIRP-SCEN : It lists the scenarios in an existing active or

inactive AI dial plan.
2)MOD-AIRP-SCEN: It modifies an existing scenario in an inactive

AI dial plan.
3)DEL-AIRP-SCEN: It deletes an existing scenario from an inactive

AI dial plan.
Routing Case
A routing case consists of a routing case name and a collection of up to 10 route
options. A route option identifies a route. It contains a route name, a priority
value and a percentage distribution value.
A routing case can be specified as the outcome of an AI routing plan.
The route with the lowest prority value will be selected first.
The percentage distribution value indicates the percentage of calls that are directed to a
particular route when multiple routes have the
same priority value.
ADD-RTCASE : It adds a routing case to the call agent.
Syntax
ADD-RTCASE: RTCASE=routing_case_name,
RO-1="route_name,priority,percent_distribution"
[, RO-n="route_name,priority,percent_distribution"];
Example:
ADD-RTCASE: RO-1="50021,1,100", RO-2="50010,2,100", RTCASE=GMSCAMD_IMSAMD
ADD-RTCASE: RO-1="50020,1,100", RO-2="50000,2,100", RTCASE=GMSC_INDORE
ADD-RTCASE: RO-1="5017,1,100", RTCASE=BPL_CLIENT
Other Route Case Commands
DEL-RTCASE : It Deletes an existing routing case.
A routing case can not be deleted (DEL-RTCASE fails) if another entity in the
call agent refers to it, for example, if the routing case is used as the outcome of
an AI routing plan scenario.
LST-RTCASE: It displays the routing cases within the call agent.

For each routing case following is displayed:
 The ROUTING-CASE column displays the routing case

name.
 The STATUS column indicates if the routing case is in use

("IN USE", "NOT IN USE"). A routing case is in use if it is
being referenced from a parent entity, such as an AI routing
plan (as the outcome in a scenario).
MOD-RTCASE : It modifies an existing routing case.

Basic Data for SIP to SIP Call
*****************************
1)Add Inbound & Outbound SIP Routes

2)Define the Route Case.
3)Define CIDP & CIDP Scenario.
4)Define AIRP & AIRP Scenario to route the call .
Following is basic data to run Sip to Sip Call :

**************************************
ADD-SIP-ROUTE: HOST=196.1.110.24, ALLOW-REDIRECT=YES, CLI-TRUSTED=TRUE, MAX-
CALLS=100000, CLIR=NONE, SIPT=NO, PROXY=NO, ROUTE=server_SIP, CLIP=TRUE, PORT=5080

CALLS=100000, CLIR=NONE, SIPT=NO, PROXY=NO, ROUTE=client_SIP, CLIP=TRUE, PORT=5070
UBL-SIP-ROUTE: ROUTE=server_SIP
UBL-SIP-ROUTE: ROUTE=client_SIP
ADD-RTCASE: RO-1="server_SIP,1,100", RTCASE=to_server
ADD-AIRP: DEFAULT-REASON=UNALLOCATED_UNASSIGNED_NUMBER, AIRP=SIP_AIRP
UPD-AIRP: AIRP=SIP_AIRP
ADD-AIRP-SCEN: CRITERIA="CDPN=555", AIRP=SIP_AIRP, OUTCOME="RTCASE=to_server",

SCEN=client_to_server
UPD-AIRP: AIRP=SIP_AIRP
ADD-CIDP: CIDP=DEFAULT_CIDP
ADD-CIDP-SCEN: CIDP=DEFAULT_CIDP, CDPN=., LENGTH=8, SCEN=DEFAULT
UPD-CIDP: CIDP=DEFAULT_CIDP
MOD-SIP-ROUTE: CIDP=DEFAULT_CIDP, AIRP=SIP_AIRP, ROUTE=client_SIP

Basic Data for SIP to ISUP Call
*************************
1)Add a Signaling Gateway & Signaling Gateway Processor.
2)Add M3UA Routing Key & M3UA Route.
3)Add MG
4)Add Inbound SIP Routes & Outbound ISUP Route.
5)Define the Route Case.
6)Define CIDP & CIDP Scenario.
7)Define AIRP & AIRP Scenario to route the call .
ADD-M3UA-SG: MTP=ITU, M3UA-VERSION=RFC, SG=SG_1
ADD-M3UA-SGP: SGP=SGP_1, LOCAL-PORT=2905, REMOTE-ADDRESSES="196.1.110.24",

REMOTE-PORT=2905, LOCAL-ADDRESS=myAsp, SG=SG_1
EBL-M3UA-SGP: SGP=SGP_1, SG=SG_1
ADD-M3UA-RK: NI=2, ROUTING-KEY=rk5555, LOCAL-PC=5555, REMOTE-PC-SET="1111"
ADD-M3UA-ROUTE: SGP=SGP_1, ROUTING-CONTEXT=1, ROUTING-KEY=rk5555, SG=SG_1
EBL-M3UA-ROUTE: SGP=SGP_1, ROUTING-KEY=rk5555, SG=SG_1
ADD-VENDOR: CLEAR-MODE=OTHER, PROTOCOL=MEGACO, VENDOR=AUDIOCODES
ADD-MG: NC=CDOT_E1, MG="MG1", PROTOCOL=MEGACO, VENDOR=AUDIOCODES
ADD-ISUP-ROUTE: CLI-TRUSTED=TRUE, VARIANT=ETSI_V2, CLIR=NONE, ECHO-

CANCELLATION=NONE, VAD=NONE, CODEC-CHOICE=NONE, DPC=1111, VOICE-
CALLS=4096, OPC=5555, ROUTE=18, CLIP=TRUE
ADD-ISUP-CIRCUIT: ENDPOINT="SH1/BR1/TR1/TS1@MG1", COUNT=1, CIC=1, ROUTE=18

EBL-ISUP-CIRCUIT: COUNT=31, CIC=1, ROUTE=18

CALLS=1000, REGISTER=NONE, CLIR=NONE, SIPT=NO, PROXY=NO, ROUTE=client_SIP,
CLIP=TRUE, PORT=5070
UBL-SIP-ROUTE: ROUTE=client_SIP
ADD-RTCASE: RO-1="18,1,100", RTCASE=rcISUP_server
ADD-AIRP: DEFAULT-REASON=UNALLOCATED_UNASSIGNED_NUMBER, AIRP=test_AIRP

UPD-AIRP: AIRP=test_AIRP
ADD-AIRP-SCEN: CRITERIA="CDPN=555", AIRP=test_AIRP,

OUTCOME="RTCASE=rcISUP_server", SCEN=sip_to_isup
MOD-SIP-ROUTE: CIDP=DEFAULT_CIDP, AIRP=test_AIRP, ROUTE=client_SIP

Basic Data for ISUP to SIP Call
*************************
ADD-M3UA-SG: MTP=ITU, M3UA-VERSION=RFC, SG=SG_1
ADD-M3UA-SGP: SGP=SGP_1, LOCAL-PORT=2905, REMOTE-ADDRESSES="196.1.110.24",

REMOTE-PORT=2905, LOCAL-ADDRESS=myAsp, SG=SG_1
EBL-M3UA-SGP: SGP=SGP_1, SG=SG_1
ADD-M3UA-RK: NI=2, ROUTING-KEY=rk5555, LOCAL-PC=5555, REMOTE-PC-SET="1111"
ADD-M3UA-ROUTE: SGP=SGP_1, ROUTING-CONTEXT=1, ROUTING-KEY=rk5555, SG=SG_1
EBL-M3UA-ROUTE: SGP=SGP_1, ROUTING-KEY=rk5555, SG=SG_1
ADD-VENDOR: CLEAR-MODE=OTHER, PROTOCOL=MEGACO, VENDOR=AUDIOCODES
ADD-MG: NC=CDOT_E1, MG="MG1", PROTOCOL=MEGACO, VENDOR=AUDIOCODES
ADD-ISUP-ROUTE: CLI-TRUSTED=TRUE, VARIANT=ETSI_V2, CLIR=NONE, ECHO-

CANCELLATION=NONE, VAD=NONE, CODEC-CHOICE=NONE, DPC=1111, VOICE-
CALLS=4096, OPC=5555, ROUTE=18, CLIP=TRUE

EBL-ISUP-CIRCUIT: COUNT=31, CIC=1, ROUTE=18

CALLS=100, REGISTER=NONE, CLIR=NONE, SIPT=NO, PROXY=NO, ROUTE=to_SIP_server,
CLIP=TRUE, PORT=5080
UBL-SIP-ROUTE: ROUTE=to_SIP_server
ADD-RTCASE: RO-1="to_SIP_server,1,100", RTCASE=rcSIP_server
ADD-AIRP: DEFAULT-REASON=UNALLOCATED_UNASSIGNED_NUMBER, AIRP=test_AIRP
ADD-AIRP-SCEN: CRITERIA="CDPN=555", AIRP=test_AIRP,

OUTCOME="RTCASE=rcSIP_server", SCEN=isup_TO_sip
MOD-ISUP-ROUTE: CIDP=DEFAULT_CIDP, AIRP=test_AIRP, ROUTE=18

Troubleshooting Guidelines
Signalling (Heartbeat) Between Nodes
C4 Soft-switch & Remote SIP Node

 While adding SIP Route, in case “OPTIONS” was set as “sendrecv”,
OPTION message will be keep exchanging between C4 SS & SIP Node.
C4 Soft-switch & Signalling Gateway

 Till SCTP association between C4 & SG don't come up, C4 SS keep
sending “Init” towards SG. On receiving Ack, C4 stop sending Init.
 Once SCTP come up, C4 SS & SG, keep exchanging Hearbeat &
Heartbeat_ack.
 On disabling M3UA SGP(dbl-m3ua-sgp), message type: ASP down

(ASPDN) is send to SG. SG do reply ASP down ack (ASPDN_ACK).
 On enabling M3UA SGP (ebl-m3ua-sgp), message type: ASP up

(ASPUP)is send to SG. SG do reply ASP up ack (ASPUP_ACK). Also
after this C4 send DAUD Destination state audit to SG. SG do reply DAVA
(Destination available).
 When SG's connectivity get break with PSTN, remote Point Code become
unaccessible, SG do sends DUNA (Destination Unavailable) to C4. And
when remote Point Code become accessible, SG do sends DAVA
(Destination Available) to C4.
C4 Soft-switch & Media Gateway

 C4 keep exchanging in time interval of (3 min), Auditcapability messages
with MG.
 On receiving no response from MG of "AUDITCAPABILITY" mask MG

OOS i.e down and all PCMs (E1#) as OOS i.e LOC-HW (Local Hardware
block)
 Whenver C4 SS send GRS to PSTN node, after receving GRA, send

SUBTRACT ALL(Null Context) for individual circuit of that PCM to MG.
If SUBTRACT reply is not received, send NULL AUDIT VALUE, if no
reply received from MG, mark Circuits as OOS i.e LOC-HW
FIELD Data As Created By Field Group
Following Data has been made, with only one AIRP(SIP_AIRP)

CALLS=100000, CLIR=NONE, SIPT=NO, PROXY=NO, ROUTE=server_SIP, CLIP=TRUE,
PORT=5080
CALLS=100000, CLIR=NONE, SIPT=NO, PROXY=NO, ROUTE=client_SIP, CLIP=TRUE,
PORT=5070
MOD-SIP-ROUTE: CIDP=DEFAULT_CIDP, AIRP=SIP_AIRP, ROUTE=client_SIP
Below are Total Number of nodes created for SIP_AIRP, after adding each of following scenarios
***********************************************************************************
************
1)On creating folloiwng scenario, number of nodes created are : 4

2)On adding folloiwng scenario(in addition to above scenario), Total number of nodes created are : 222
ADD-AIRP-SCEN: CRITERIA="CGPN=91[12345678]...29[78],CDPN=77770", AIRP=SIP_AIRP,
OUTCOME="RTCASE=to_server", SCEN=scen2
ADD-AIRP-SCEN: CRITERIA="CDPN=44444,CGPN=6666666", AIRP=SIP_AIRP,
4)On adding folloiwng scenario(in addition to above scenario), Total number of nodes created are :541
ADD-AIRP-SCEN: CRITERIA="CDPN=77777777", AIRP=SIP_AIRP,
5)On adding folloiwng scenario(in addition to above scenario), Total number of nodes created are :612
ADD-AIRP-SCEN: CRITERIA="CDPN=999999,CGPN=444444", AIRP=SIP_AIRP,
6)On adding folloiwng scenario(in addition to above scenario), Total number of nodes created are :
2068
ADD-AIRP-SCEN: CRITERIA="CDPN=91[12345678]..29[78]", AIRP=SIP_AIRP,
4309
ADD-AIRP-SCEN: CRITERIA="CDPN=91[12345678]...29[78]", AIRP=SIP_AIRP,
Optimized Field Data created by Development Team
Following is data after ading two more AIRP

CALLS=100000, CLIR=NONE, SIPT=NO, PROXY=NO, ROUTE=server_SIP, CLIP=TRUE,
PORT=5080
CALLS=100000, CLIR=NONE, SIPT=NO, PROXY=NO, ROUTE=client_SIP, CLIP=TRUE,
PORT=5070
ADD-AIRP: DEFAULT-REASON=UNALLOCATED_UNASSIGNED_NUMBER, AIRP=AIRP_1
ADD-AIRP: DEFAULT-REASON=UNALLOCATED_UNASSIGNED_NUMBER, AIRP=AIRP_2
Below are Total Number of nodes created for SIP_AIRP, after adding each of following scenarios
***********************************************************************************
************
1)On creating folloiwng scenario, number of nodes created are : 4

Number of nodes for AIRP_1 : 6
ADD-AIRP-SCEN: CRITERIA="CGPN=91[12345678]...29[78]", AIRP=SIP_AIRP,
OUTCOME="AIRP=AIRP_1", SCEN=scen2
ADD-AIRP-SCEN: CRITERIA="CDPN=77770", AIRP=AIRP_1,
OUTCOME="RTCASE=to_server", SCEN=scen_1
ADD-AIRP-SCEN: CRITERIA="CGPN=6666666", AIRP=SIP_AIRP, OUTCOME="AIRP=AIRP_2",
SCEN=scen3
ADD-AIRP-SCEN: CRITERIA="CDPN=77777777", AIRP=SIP_AIRP,
ADD-AIRP-SCEN: CRITERIA="CGPN=444444", AIRP=SIP_AIRP, OUTCOME="AIRP=AIRP_2",
SCEN=scen5
1412
ADD-AIRP-SCEN: CRITERIA="CDPN=91[12345678]..29[78]", AIRP=SIP_AIRP,
2933
ADD-AIRP-SCEN: CRITERIA="CDPN=91[12345678]...29[78]", AIRP=SIP_AIRP,

C-Dot Class-4 Softswitch Data Guidelines

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C-Dot Class-4 Softswitch Data Guidelines

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 Breif about C4 Soft-switch

 Call could be between IP - IP subscriber or between IP – PSTN subscriber.

 Class-4 Softswitch also provides interconnection between IP network and PSTN by

● PSTN nodes are connected to C4 SS via Signalling Gateway(SG) & Media

● C4 SS & SG are connected over SCTP association.

● C4 SS & MG are connected over MEGACO Protocol.

● Physical E1 Trunks are connected between MG & Remote PSTN nodes.

 C4 SS do Number Analyses using signalling information received in incoming

SYNTAX: ADD-SIP-ROUTE:ROUTE=<Route>, HOST=<IP address>,

Other SIP ROUTE Commands:

Local Maintenance Blocked

Remote Maintenance Blocked

CONFIG-ITEM VALUE MODIFIABLE

MOD-SIP-CONFIG : It modifies the SIP configuration items.

2)MOD-ISUP-ROUTE : It changes the characteristics of an ISUP route.

3)DEL-ISUP-ROUTE : It deletes an ISUP route.

4)LST-ISUP-ROUTE : It lists the ISUP routes.

ADD-ISUP-CIRCUIT : It adds circuits to an ISUP route.

ADD-ISUP-CIRCUIT: ENDPOINT="SH1/BR1/TR1/TS1@MG1", COUNT=3,

Any one of following will be “STATUS” value:

● PAUSED : A circuit is paused when the DPC is inaccessible. It

● PENDING : A circuit is pending, if no response received from PSTN

● OBUSY : The circuit is in being used as the originating circuit in a

● TBUSY : The circuit is in being used as the terminating circuit in a

● BLOCKED : The circuit is locally or remotely maintenance or

The Local Maintenance block values are:

The remote maintenance block values are:

The local hardware block values are:

The remote hardware block values are:

2)DBL-ISUP-CIRCUIT: It disable one or more circuits on an existing ISUP

3)DEL-ISUP-CIRCUIT: It deletes one or more circuits on an existing ISUP

4)BLK-ISUP-CIRCUIT: It initiates the local maintenance blocking of one or

5)UBL-ISUP-CIRCUIT : It initiates the local maintenance unblocking of one

6)RES-ISUP-CIRCUIT : It initiates the reset of one or more circuits on an

DIS-ISUP-CONFIG : It displays the details of all the ISUP congiguration

MOD-ISUP-CONFIG: It modifies the ISUP configuration items.

ADD-VENDOR : It adds a vendor to the call agent.

ADD-MG : It adds an MEGACO Media Gateway to the call agent.

DEL-MG :This command may be used to delete a Media Gateway .

MOD-MG: This command may be used to modify the parameters associated

Each SS7 point code is unique within a specific MTP3 network.

ADD-M3UA-SGP : A signalling gateway process (SGP) is a process instance of a signalling gateway.

ADD-M3UA-ROUTE : It adds a route to an SGP for a particular routing key.

Sample M3UA Data :

ADD-M3UA-SG: MTP=ITU, M3UA-VERSION=RFC, SG=SG_BPL

ADD-M3UA-SGP: SGP=SGP2_BPL, LOCAL-PORT=2905, REMOTE-ADDRESSES="SGP2_BPL",

ADD-M3UA-SGP: SGP=SGP1_BPL, LOCAL-PORT=2905, REMOTE-ADDRESSES="SGP1_BPL",

ADD-M3UA-RK: NI=2, ROUTING-KEY=rk_8427, LOCAL-PC=8427, REMOTE-PC-SET="5184,

ADD-M3UA-ROUTE: SGP=SGP2_BPL, ROUTING-CONTEXT=1, ROUTING-KEY=rk_8427,

ADD-M3UA-ROUTE: SGP=SGP1_BPL, ROUTING-CONTEXT=1, ROUTING-KEY=rk_8427,

ADD-M3UA-ROUTE: SGP=SGP2_BPL, ROUTING-CONTEXT=2, ROUTING-KEY=rk_2709,

ADD-M3UA-ROUTE: SGP=SGP1_BPL, ROUTING-CONTEXT=2, ROUTING-

ADD-CIDP : It adds a new inactive CI dial plan to the call agent.

Other CIDP Commands:

• NOT MODIFIED, the inactive dial plan is identical to the active

CIDP MODIFICATION STATUS

3)UPD-CIDP: If no active version of inactive CIDP plan with same name is

A CI dial plan scenario consists of the following criteria:

To reflect these scenario’s in active CIDP, run the UPD-CIDP