[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: chan_sip.c:8686 get_destination: Huh? Not
a SIP header (Tel:+32564 <tel:+32564>)?
[Nov 6 17:14:23] NOTICE: 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 10.10.8.2:5060 --->
ACK Tel:+32564 <tel:+32564> SIP/2.0
Via: SIP/2.0/UDP 10.10.8.2:5060
From: avayanew <sip:avayanew at avayanew>;tag=d60c0430c7b26cbd
Call-ID: 0182709d8c1d025f42dd3dd767c7e8b7 at 10.10.8.2
CSeq: 152795667 ACK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
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
Please advise. Any help is appreciated.
Thanks as always
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