Connecting Avaya S8720 via SIP TCP

I am having problems setting up SIP trunks with an Avaya S8720. We are not using the SIP Enablement Server (SES) but are using the native SIP in CM5.1. It talks either TLS or TCP but no UDP. It is currently set to TCP. We are using FreePBX 2.5.1.0 and Asterisk 1.6.0.9 in order to get get TCP on Asterisk.

I have tried various searchs without finding applicable results. :frowning:

I can call station to station with no problem. When I try to call to the Asterisk from the Avaya, I get “ss-noservice” intercept so I know the two switches are talking on at least some level. I do hear the announcement from the Avaya end.

Following some of the other posts on this forum, I set up the debug on the channel. I tried to use verbosity set to 0 but got nothing so I went back to 3 and got this output. Sorry if it is too long.

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] NoOp(“SIP/invalid.unknown.domain-1664caf0”, “Received incoming SIP connection from unknown peer to 10999”) in new stack
– Executing [[email protected]:2] Set(“SIP/invalid.unknown.domain-1664caf0”, “DID=10999”) in new stack
– Executing [[email protected]:3] Goto(“SIP/invalid.unknown.domain-1664caf0”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [[email protected]:1] GotoIf(“SIP/invalid.unknown.domain-1664caf0”, “0?from-trunk,10999,1”) in new stack
– Executing [[email protected]:2] Set(“SIP/invalid.unknown.domain-1664caf0”, “TIMEOUT(absolute)=15”) in new stack
Channel will hangup at 2009-06-03 11:46:52.000 EDT.
– Executing [[email protected]:3] Answer(“SIP/invalid.unknown.domain-1664caf0”, “”) in new stack
– Executing [[email protected]:4] Wait(“SIP/invalid.unknown.domain-1664caf0”, “2”) in new stack
– Executing [[email protected]:5] Playback(“SIP/invalid.unknown.domain-1664caf0”, “ss-noservice”) in new stack
– <SIP/invalid.unknown.domain-1664caf0> Playing ‘ss-noservice.ulaw’ (language ‘en’)
== Spawn extension (from-sip-external, s, 5) exited non-zero on ‘SIP/invalid.unknown.domain-1664caf0’
– Executing [[email protected]:1] NoOp(“SIP/invalid.unknown.domain-1664caf0”, “Hangup”) in new stack
– Executing [[email protected]:2] Set(“SIP/invalid.unknown.domain-1664caf0”, “DID=s”) in new stack
– Executing [[email protected]:3] Goto(“SIP/invalid.unknown.domain-1664caf0”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [[email protected]:1] GotoIf(“SIP/invalid.unknown.domain-1664caf0”, “0?from-trunk,s,1”) in new stack
– Executing [[email protected]:2] Set(“SIP/invalid.unknown.domain-1664caf0”, “TIMEOUT(absolute)=15”) in new stack
Channel will hangup at 2009-06-03 11:46:57.000 EDT.
– Executing [[email protected]:3] Answer(“SIP/invalid.unknown.domain-1664caf0”, “”) in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on ‘SIP/invalid.unknown.domain-1664caf0’

Incoming settings are
user context is set to from-trunk
host=XXX.XX.XXX.X with that IP being the address of the CLAN on the S8720
type=peer
context=from-trunk

The Avaya does not seem to support the registration feature. The Avaya guru has some notes from a meeting he attended bbut I don’t know how to really interpret them based on what I see in FreePBX. His notes are:

• In the Asterisk /etc/asterisk folder
o ;#sipid=???
Enter a User ID
o secret=avaya
it doesn’t matter what is put here because the insecure=very later overrides it but I believe something is needed
o callerid=000000
o type=friend
o context=sip-pbx
o host=SVP CLAN
o nat=no
o canreinvite=no
o insecure=very
o username=000000

I am very new to this and don’t know where to go from here. I am also having issues with calling out but one thing at a time! :slight_smile: I appreciate any help you can give me. Please let me know if there are other pieces of information that might help.

Trying to do the same thing. I saw this post:
http://www.trixbox.org/forums/trixbox-forums/open-discussion/tcp-sip-asterisk-1-4

Specifically:
Next add the following two lines to sip.conf (or sip_custom.conf if you’re using freepbx);

tcpenable=yes
transport=tcp

and, in the trunk peer details, I added transport=tcp

TCP is established, trunk is up, and I am able to originate calls from Asterisk to CM.

Could you please share your successful configurations. Thanks.