[asterisk-users] asterisk - avaya ip office SIP trunking

Krishna Sumanth Chava kschava at gmail.com
Fri Nov 7 03:22:17 CST 2008

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

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 <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

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

"<--- SIP read from --->

ACK Tel:+32564 <tel:+32564> SIP/2.0

Via: SIP/2.0/UDP

From: avayanew <sip:avayanew at avayanew>;tag=d60c0430c7b26cbd

To: Tel:+32564;tag=as51355066

Call-ID: 0182709d8c1d025f42dd3dd767c7e8b7 at

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

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