Avaya - Asterisk Integration

[mjm23]

Hi, this is my first time setting up a connectivity between Avaya and Asterisk. I used SM in between to manage the SIP routing. I can see that when Asterisk is sending OPTIONS messages, SM replies 200 OK response, but when SM sends OPTIONS msgs, Asterisk responds 404 Not Found though Link is already UP. Below is the SIP trace:

192.168.191.150:5062 ──UDP─► 10.173.66.29:5062 │
├─────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────────┤
│SIP/2.0 404 Not Found │
│Via: SIP/2.0/UDP 10.173.66.29:5062;branch=z9hG4bK551577600998340-AP;ft=10.173.66.29~13c6;received=10.173.66.29;rport=5062 │
│Via: SIP/2.0/UDP 127.0.0.2:15060;rport=15060;ibmsid=local.1587013557443_18253084_18253094;branch=z9hG4bK551577600998340 │
│Via: SIP/2.0/UDP 127.0.0.2:15060;rport;ibmsid=local.1587013557443_18253083_18253093;branch=z9hG4bK698364478032632 │
│Via: SIP/2.0/UDP 127.0.0.2:15060;rport;ibmsid=local.1587013557443_18253082_18253092;branch=z9hG4bK500063523934929 │
│From: <sip:10.173.66.29>;tag=7337654350150313_local.1587013557443_18253082_18253092 │
│To: <sip:192.168.191.150:5062>;tag=as5b7202da │
│Call-ID: 08999179236650479@127.0.0.2 │
│CSeq: 2 OPTIONS │
│Server: Asterisk PBX 1.8.32.3 │
│Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE │
│Supported: replaces, timer │
│Accept: application/sdp │
│Content-Length: 0

 

[kyle555]

That's fine. 404 means it's alive. I guess Asterisk doesn't know how to answer your OPTIONS TO header.

Try sending it a call.

 

[mjm23]

Still getting 404 error when I placed a test call. The Asterisk guy said he saw my call but he did not see something on the console. When he did the test call, I was able to see it in SM but he said he isn't hearing anything. We already set the access list to any-any for troubleshooting purposes. Is there any special configuration in Asterisk or are there any other stuff I need to check on my Avaya?

 

[kyle555]

Not really. You put the trace of an OPTIONS message. Any answer beyond a timeout is OK because it means the far end is answering - that's probably why your CM sig group is in service.

The asterisk guys need to look at the invite you sent and make heads or tails of it.

 

[mjm23]

From the trace, we are able to see calls hitting each sides. But I cant see a SIP Session in progress status. And also we hear no audio even though we opened ports 1024-65535/UDP in both directions.

 

[mjm23]

we are able to resolve this issue. it turns out that the network configuration was not replicated to the other network equipment hence blocking the RTP ports. thanks for your inputs @kyle555! cheers!