asterisk users - Nov 2008 - asterisk - avaya ip office 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 <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 <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
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