IPCM SIP trunk to Avaya CM

[hdbeavis]

All,
I am trying to implement a SIP Trunk between an Avaya Communicaiton Manager System R5.2 and an IPCM SIP server. Calls come in the Avaya and route across the SIP trunk to the IPCM just fine. However, when a user tries to call outbound from the system on the other side of the IPCM I don't see the call hitting the trunk. The trunk is currently setup directly from a Clan to the IPCM. I was given homemade config notes from the IPCM vendor showing the direct connection from the Clan to the IPCM. It was my belief that Avaya Clan's didn't support direct SIP trunks, without the use of an SES server, until CM6.0. The IPCM vendor is sure they have this setup at other customer sites without the use of a SIP Server but I am just not convinced. Has anyone by chance set this similar configuration up?

 

[cjgovMIL]

is the IPCM considered a gateway? I have Avaya 5.2 and direct connect to a NET SIP Gateway with no problems.

 

[hdbeavis]

I believe it is, don't really know much about it to be honest. I have only worked with them via phone. I have a basic SIP Trunk/Signaling Group setup. We can see the SIP invite come out of the IPCM, via wireshark, and appears correctly addressed to the Clan but the Clan never acknowledges the request.

 

[cjgovMIL]

Try this on the Avaya:

list trace tac <tac of the SIP trunk group>

And see if you see it hitting the Avaya.

 

[hdbeavis]

No I never see anything. That's what prompted the Wireshark trace. I can see it all the way to switch the Clan is connected to byt I get an error something like "invalid ip address/port". It is trying to connect via port 5060, as signaling group is programmed and again I can complete calls from the Avaya to the IPCM so I know the connectivity is there. I can telnet from the actual IPCM, via port 5060, directly to the Clan so again I know there isn't an ip address/port problem. The Clan just doesn't seem to ackowledge the SIP request.

 

[cjgovMIL]

What version is your Avaya?

 

[cjgovMIL]

Full version.. 5.2.??

 

[hdbeavis]

5.2.1.016.4

 

[cjgovMIL]

Ok.. try these tricks (it is how I made it work to my SIP Gateway)

On the trunk form:
Service Type: public-ntwrk (Page 1)
Numbering Format: unk-pvt (Page 3)
(Page 4)
Mark Users as Phone? n
Prepend '+' to Calling Number? n (YOU MAY NEED THIS Y DEPENDING ON HOW YOU HAVE YOUR SIP GATEWAY SETUP)
Send Transferring Party Information? y
Network Call Redirection? n
Send Diversion Header? y
Support Request History? n
Telephone Event Payload Type: 127

SIGNALING GROUP:
Group Type: sip
Transport Method: tcp
Near-end Node Name: <YOUR C-LAN BOARD NODE NAME>
Far-end Node Name: <YOUR SIP GATEWAY NODE NAME>
Near-end Listen Port: 5060
Far-end Listen Port: 5060
Far-end Domain: <IP ADDRESS OF THE SIP GATEWAY>
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate
RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload
Direct IP-IP Audio Connections? n
Session Establishment Timer(min): 3
IP Audio Hairpinning? y
Enable Layer 3 Test? n
Alternate Route Timer(sec): 6

Then, you will have to modify your AAR and ARS depending on how you are making calls. but this should at least get the SIP trunks talking to one another.

 

[hdbeavis]

Thanks for the responses cjgovMIL....it turns out that the PBX was fine, the IPCM programmer did not have their outbound transport method set to TCP. As soon as they changed that all was good. Thanks again and sorry I didn't get this updated sooner.