[asterisk-users] Help with asterisk and avaya SIP trunking

[Krishna Sumanth Chava]

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 10.10.8.2:5060 --->
ACK Tel:+32564 SIP/2.0
Via: SIP/2.0/UDP 10.10.8.2:5060
;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9
From: avayanew <sip:avayanew at avayanew>;tag=d60c0430c7b26cbd
To: Tel:+32564;tag=as51355066
Call-ID: 0182709d8c1d025f42dd3dd767c7e8b7 at 10.10.8.2
CSeq: 152795667 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0"

Note: 10.10.8.2 is avaya and 10.10.8.1 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

Regards
Krishna
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081107/0091e63b/attachment.htm