<div class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Hi * Users,</font></div>
<div class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"></font> </div>
<div class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">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.</font></div>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"> </font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Can someone provide me insight on how to address it or the path to resolve it.</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"> </font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">The error I get is mentioned below: (dialing 32564 from avaya to asterisk)</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"> </font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">"[Nov 6 17:14:23] WARNING[6227]: chan_sip.c:8686 get_destination: Huh? Not a SIP header (</font><a href="tel:+32564"><font face="Calibri" color="#0000ff" size="3">Tel:+32564</font></a><font face="Calibri" size="3">)?</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">[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."</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"> </font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">A SIP Debug of the packet when this happens on asterisk CLI is</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"> </font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">"<--- SIP read from <a href="http://10.10.8.2:5060">10.10.8.2:5060</a> ---></font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">ACK </font><a href="tel:+32564"><font face="Calibri" color="#0000ff" size="3">Tel:+32564</font></a><font face="Calibri" size="3"> SIP/2.0</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Via: SIP/2.0/UDP 10.10.8.2:5060;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">From: avayanew <sip:avayanew@avayanew>;tag=d60c0430c7b26cbd</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">To: Tel:+32564;tag=as51355066</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Call-ID: <a href="mailto:0182709d8c1d025f42dd3dd767c7e8b7@10.10.8.2">0182709d8c1d025f42dd3dd767c7e8b7@10.10.8.2</a></font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">CSeq: 152795667 ACK</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Max-Forwards: 70</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Content-Length: 0"</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"> </font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">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</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"> </font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">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.</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"> </font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Please advise. Any help is appreciated.</font></p>
<p class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"> </font></p>
<div class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Thanks as always</font></div>
<div class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3"></font> </div>
<div class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Regards</font></div>
<div class="MsoNormal" style="MARGIN: 0cm 0cm 0pt"><font face="Calibri" size="3">Krishna</font></div>