Help with asterisk and Avaya SIP Trunking

Hi * Users,

I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk.

Can someone provide me insight on how to address it or the path to resolve it.

The error I get is mentioned below: (dialing 32564 from avaya to asterisk)

“[Nov 6 17:14:23] WARNING[6227]: chan_sip.c:8686 get_destination: Huh? Not a SIP header (Tel:+32564)?
[Nov 6 17:14:23] NOTICE[6227]: chan_sip.c:13774 handle_request_invite: Call from ‘avayanew’ to extension ‘Tel:+32564’ rejected because extension not found.”

A SIP Debug of the packet when this happens on asterisk CLI is

“<— SIP read from —>
ACK Tel:+32564 SIP/2.0
Via: SIP/2.0/UDP;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9
From: avayanew sip:avayanew@avayanew;tag=d60c0430c7b26cbd
To: Tel:+32564;tag=as51355066
Call-ID: 0182709d8c1d025f42dd3dd767c7e8b7@
CSeq: 152795667 ACK
Max-Forwards: 70
Content-Length: 0”

Note: is avaya and is asterisk

As I understand, we are getting a Tel URI and a “+” like in e.164 format and then the number dialed (32564)from avaya. These errors are coming on asterisk console when I try to dial a call from Avaya IP Phone over its SIP trunk on to the asterisk. We probably have to strip the ‘Tel:+’, so that the asterisk gets the number and thus follows the dialplan programmed in extensions file.

Please advise. Any help is appreciated.

Thanks as always


That is exactly what you have to do (remove the +). If in extensions.conf you have: exten => _X.,1,… then there is no room there for the. You want something like exten => _+X.,1,… and then remove the first part of the number which is the +.