Hi * Users,<br> <br>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.<br> <br>Can someone provide me insight on how to address it or the path to resolve it.<br>
<br>The error I get is mentioned below: (dialing 32564 from avaya to asterisk)<br> <br>"[Nov 6 17:14:23] WARNING[6227]: chan_sip.c:8686 get_destination: Huh? Not a SIP header (Tel:+32564)?<br>[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."<br>
<br>A SIP Debug of the packet when this happens on asterisk CLI is<br> <br>"<--- SIP read from <a href="http://10.10.8.2:5060">10.10.8.2:5060</a> ---><br>ACK Tel:+32564 SIP/2.0<br>Via: SIP/2.0/UDP 10.10.8.2:5060;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9<br>
From: avayanew <sip:avayanew@avayanew>;tag=d60c0430c7b26cbd<br>To: Tel:+32564;tag=as51355066<br>Call-ID: <a href="mailto:0182709d8c1d025f42dd3dd767c7e8b7@10.10.8.2">0182709d8c1d025f42dd3dd767c7e8b7@10.10.8.2</a><br>
CSeq: 152795667 ACK<br>Max-Forwards: 70<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO<br>Content-Length: 0"<br> <br>Note: <a href="http://10.10.8.2">10.10.8.2</a> is avaya and <a href="http://10.10.8.1">10.10.8.1</a> is asterisk<br>
<br>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.<br>
<br>Please advise. Any help is appreciated.<br> <br>Thanks as always<br> <br>Regards<br>Krishna