<div>Hi Guys,</div>
<div> </div>
<div>Thanks that did help to resolve my issue. i tried the ."@<a href="http://10.10.8.1">10.10.8.1</a>" and it worked and i had a successful call but i have the following 2 concerns.</div>
<div> </div>
<div>1. We have voice communication from avaya to asterisk now but avaya is forcing asterisk to use only codec G723. if i disable G723, it says no compatible codecs. While the calls from asterisk to avaya are being accepted as "alaw"</div>
<div>2. I am having issues with DTMF. DTMF is not being recognized or being sent from avaya to asterisk.</div>
<div> </div>
<div>I had connected an Analog phone to the POTS line of the IP Office for this experiment.</div>
<div> </div>
<div>Also i am having hard time for detecting Hangups. </div>
<div> </div>
<div>Please advise.</div>
<div> </div>
<div>Any help is appreciated as i am new to avaya IP office and am much familiar with asterisk.</div>
<div> </div>
<div>Regards</div>
<div>Krishna<br></div>
<div class="gmail_quote">On Sat, Nov 8, 2008 at 12:28 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 Robb,<br>> I had the checked the IP Office and i see that in the SIP Line<br>> Settings an option [checkbox] that says (Use Tel URI), which is<br>> unchecked. But i still get the Tel:+ in the SIP Header (even when it<br>
> is turned on or off).<br>><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>> do you mean that i need to program this in the ARS of the avaya IP office?<br>
><br>> i have version 4.1(9) firmware on the Avaya IP small Office. Can you<br>> share me on what Firmware version of avaya IP small Office, you got<br>> the Asterisk and avaya talking to each other.<br>><br>
> Thanks<br>> Krishna<br>><br>><br>><br>><br>> On Fri, Nov 7, 2008 at 2:59 PM, Robert Boardman <<a href="mailto:robb@boardman.me.uk">robb@boardman.me.uk</a><br></div>
<div class="Ih2E3d">> <mailto:<a href="mailto:robb@boardman.me.uk">robb@boardman.me.uk</a>>> wrote:<br>><br>> 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<br>> 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<br>> 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<br>> '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>
> <<a href="http://10.10.8.2:5060/" target="_blank">http://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>> <mailto:<a href="mailto:0182709d8c1d025f42dd3dd767c7e8b7@10.10.8.2">0182709d8c1d025f42dd3dd767c7e8b7@10.10.8.2</a>><br>
> > <mailto:<a href="mailto:0182709d8c1d025f42dd3dd767c7e8b7@10.10.8.2">0182709d8c1d025f42dd3dd767c7e8b7@10.10.8.2</a><br>> <mailto:<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></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>> <<a href="http://10.10.8.2/" target="_blank">http://10.10.8.2</a><br>
> <<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> <<a href="http://10.10.8.1/" target="_blank">http://10.10.8.1/</a>><br>
> > <<a href="http://10.10.8.1/" target="_blank">http://10.10.8.1</a> <<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<br>> 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<br>> strip<br>> > the 'Tel:+', so that the asterisk gets the number and thus<br>
> 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>> ><br>> ------------------------------------------------------------------------<br>> ><br>> > _______________________________________________<br>
> > -- Bandwidth and Colocation Provided by<br></div>> <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> <<a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com/</a>> --<br>
<div class="Ih2E3d">> ><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<br>> 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>
> <<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>
><br>><br>> ------------------------------------------------------------------------<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>
</div>sorry its something like<br><br>dial 9n;<br>feature dial<br>code n"@<a href="http://192.168.0.1/" target="_blank">192.168.0.1</a>"<br><br><br>where the ip address is the asterisk box<br>
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<div class="Wj3C7c"><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>
</div></div></blockquote></div><br>