HowTo configure Avaya CM 5.1 with sipX

From SIPfoundry sipXecs IP PBX, The Open Source SIP PBX for Linux - Calivia

This is currently a work in progress.

I can dial "extensions" from the Avaya to sipXecs, send calls to voicemail and host a conference call (8 parties tested so far)

blind transfers from an automated attendant to a conference work, there may be other parts that do not yet work...

On the Avaya CM

change node-names ip

In the example below sipXecs is the IP node-name used, and domain.local is the SIP domain used

and add the sipeXecs name and address

now add a signalling group, the far end domain is deliberately blank

change signaling-group 60                                       Page   1 of   1
                                SIGNALING GROUP

 Group Number: 60             Group Type: sip
                        Transport Method: tcp
                         Co-Resident SES? y





   Near-end Node Name: procr                 Far-end Node Name: sipXecs
 Near-end Listen Port: 5062                Far-end Listen Port: 5060
                                        Far-end Network Region: 1
       Far-end Domain: 

                                             Bypass If IP Threshold Exceeded? n

         DTMF over IP: rtp-payload            Direct IP-IP Audio Connections? n
                                                        IP Audio Hairpinning? n
         Enable Layer 3 Test? n
 Session Establishment Timer(min): 3              Alternate Route Timer(sec): 6

The "Near-end Listen Port" needs to be unique on the CM system

Now add a trunk group using the signalling group

change trunk-group 60                                           Page   1 of  22
                                TRUNK GROUP

Group Number: 60                   Group Type: sip           CDR Reports: y
  Group Name: loewy-conf-2                COR: 1        TN: 4        TAC: 460
   Direction: two-way        Outgoing Display? n
 Dial Access? n                                   Night Service:
Queue Length: 0
Service Type: public-ntwrk          Auth Code? n

                                                       Signaling Group: 60
                                                     Number of Members: 30

Set the number of members to be the number of channels desired on the link

change trunk-group 60                                           Page   2 of  22
      Group Type: sip

TRUNK PARAMETERS

     Unicode Name? y

                                            Redirect On OPTIM Failure: 5000

            SCCAN? n                               Digital Loss Group: 18
                      Preferred Minimum Session Refresh Interval(sec): 600

change trunk-group 60                                           Page   3 of  22
TRUNK FEATURES
          ACA Assignment? n            Measured: none
                                                          Maintenance Tests? y



                     Numbering Format: public
                                                UUI Treatment: shared
                                              Maximum Size of UUI Contents: 128
                                                 Replace Restricted Numbers? n
                                                Replace Unavailable Numbers? n



               Send UCID? n



 Show ANSWERED BY on Display? y

change trunk-group 60                                           Page   4 of  22
                           SHARED UUI FEATURE PRIORITIES

                             ASAI: 1

         Universal Call ID (UCID): 2

MULTI SITE ROUTING (MSR)

                      In-VDN Time: 3
                         VDN Name: 4
                 Collected Digits: 5
            Other LAI Information: 6

change trunk-group 60                                           Page   5 of  22
                              PROTOCOL VARIATIONS

                      Mark Users as Phone? n
            Prepend '+' to Calling Number? n
      Send Transferring Party Information? n

             Telephone Event Payload Type:

Obviously the the "extensions" that exist on the sipXecs side need to be suitable configured on the Avaya side. I did this by putting them into the uniform dialplan as aar numbers, and in the aar analysis table to route to a route pattern that points at the SIP trunk.

On the sipXecs server

In System | Servers | Configure

Ensure that SIP Trunking and Conferencing have ticks (I tick them all...)

In Devices | Gateways

Add an entry for CM, using the the port defined for the CM signalling group "Far-end Listen Port", set the transport to be TCP

You should now be able to dial from the Avaya to the sipXecs