<div>HI Robb,<br></div>
<div>I had the checked the IP Office and i see that in the SIP Line Settings an option [checkbox] that says (Use Tel URI), which is unchecked. But i still get the Tel:+ in the SIP Header (even when it is turned on or off).</div>
<div> </div>
<div>"you need to make sure the sip dial command in the ipoffice is set to<br>dial 9n;<br>feature dial<br>code n"</div>
<div> </div>
<div>do you mean that i need to program this in the ARS of the avaya IP office?</div>
<div> </div>
<div>i have version 4.1(9) firmware on the Avaya IP small Office. Can you share me on what Firmware version of avaya IP small Office, you got the Asterisk and avaya talking to each other.</div>
<div> </div>
<div>Thanks</div>
<div>Krishna<br></div>
<div> </div>
<div> </div>
<div><br> </div>
<div class="gmail_quote">On Fri, Nov 7, 2008 at 2:59 PM, Robert Boardman <span dir="ltr"><<a href="mailto:robb@boardman.me.uk">robb@boardman.me.uk</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div class="Ih2E3d">Krishna Sumanth Chava wrote:<br>> Hi * Users,<br>><br>> I ran into a problem when I was trying to communicate an avaya IP<br>> Office talk to asterisk with SIP Trunking. I had successful calls from<br>
> 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<br>> 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:<br>> Huh? Not a SIP header (Tel:+32564)?<br>> [Nov 6 17:14:23] NOTICE[6227]: chan_sip.c:13774<br>> handle_request_invite: Call from 'avayanew' to extension 'Tel:+32564'<br>
> rejected because extension not found."<br>><br>> A SIP Debug of the packet when this happens on asterisk CLI is<br>><br></div>> "<--- SIP read from <a href="http://10.10.8.2:5060/" target="_blank">10.10.8.2:5060</a> <<a href="http://10.10.8.2:5060/" target="_blank">http://10.10.8.2:5060</a>> ---><br>
<div class="Ih2E3d">> ACK Tel:+32564 SIP/2.0<br>> Via: SIP/2.0/UDP<br>> 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></div>> <mailto:<a href="mailto:0182709d8c1d025f42dd3dd767c7e8b7@10.10.8.2">0182709d8c1d025f42dd3dd767c7e8b7@10.10.8.2</a>><br>
<div class="Ih2E3d">> CSeq: 152795667 ACK<br>> Max-Forwards: 70<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO<br>> Content-Length: 0"<br>><br></div>> Note: <a href="http://10.10.8.2/" target="_blank">10.10.8.2</a> <<a href="http://10.10.8.2/" target="_blank">http://10.10.8.2</a>> is avaya and <a href="http://10.10.8.1/" target="_blank">10.10.8.1</a><br>
> <<a href="http://10.10.8.1/" target="_blank">http://10.10.8.1</a>> is asterisk<br>
<div class="Ih2E3d">><br>> As I understand, we are getting a Tel URI and a "+" like in e.164<br>> format and then the number dialed (32564)from avaya. These errors are<br>> coming on asterisk console when I try to dial a call from Avaya IP<br>
> Phone over its SIP trunk on to the asterisk. We probably have to strip<br>> the 'Tel:+', so that the asterisk gets the number and thus follows the<br>> dialplan programmed in extensions file.<br>><br>
> Please advise. Any help is appreciated.<br>><br>> Thanks as always<br>><br>> Regards<br>> Krishna<br></div>> ------------------------------------------------------------------------<br>><br>> _______________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>you need to make sure the sip dial command in the ipoffice is set to<br>
dial 9n;<br>feature dial<br>code n<br><br>in system<br>the set the dial delay timer to 4 seconds<br><br>and the dial delay count to 1<br><br>this will allow 4 seconds in between each digit<br><br>there is a setting on the ipo to change the TEL:+ setting to url setting<br>
<br>cannot remember wher it is but it in the sip trunk settings<br><br><br>robb<br><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
<br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
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