<div>For anyone else that runs into this issue here is what I found to fix it.</div><div> </div><div> </div><div> </div><div>This is what Snom said to try and it worked</div><div> </div><div>Our first suggestion is to simply set the "Support Broken Registrar" to <on></div>
<div>Web Interface --> Identity X --> SIP 'tab' --> "Support broken Registrar" = ON</div><div>This should resolve the 404 Not Found error.<br><br></div><div> </div><div> </div><div class="gmail_quote">
On Thu, Aug 9, 2012 at 10:20 AM, <span dir="ltr"><<a href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</a>></span> wrote:<br><blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">
Thanks for the info. I'll check with Snom and see why the phone is rejecting the INVITE.<div><div class="h5"><br><br><br><br><br>On , Vlad Paiu <<a href="mailto:vladpaiu@opensips.org" target="_blank">vladpaiu@opensips.org</a>> wrote:<br>
> <br>> <br>> <br>> <br>> <br>> Hello,<br>> <br>> <br>> <br>> The <> are only required if you want to have SIP header<br>> parameters for the TO header.<br>
> <br>> Otherwise, if there are no <> , all the parameters are<br>> considered to be SIP URI parameters.<br>> <br>> So, from what I see, that TO header is correct.<br>> <br>> <br>
> <br>> Regards,<br>> Vlad Paiu<br>> OpenSIPS Developer<br>> <a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a><br>> <br>> On 08/09/2012 06:13 AM, Duane Larson wrote:<br>
> <br>> <br>> <br>> <br>> I changed the following in the ctd.sh script<br>> <br>> <br>> <br>> Changed the default of<br>> <br>> "`printf "v=0\r\no=click-to-dial 0 0 IN IP4<br>
> 0.0.0.0\r\ns=session\r\nc=IN IP4 0.0.0.0\r\nb=CT:1000\r\nt=0<br>> 0\r\nm=audio 9 RTP/AVP 8 0\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:0<br>> PCMU/8000\r\n"`<br>> <br>> <br>> <br>
> To<br>> <br>> "`printf "v=0\r\no=click2dial 0 0 IN IP4<br>> 50.XX.XX.156\r\ns=click2dial call\r\nc=IN IP4<br>> 173.XX.XX.111\r\nt=0 0\r\nm=audio 12790 RTP/AVP 0 8 18 3 4 97<br>
> 98\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:18<br>> G729/8000\r\na=rtpmap:97 ilbc/8000\r\na=rtpmap:98<br>> speex/8000\r\n"`<br>> <br>> <br>> <br>> <br>> <br>
> And now it is making it into the OpenSIPS/SBC's main route. <br>> Not sure why.<br>> <br>> <br>> <br>> I noticed another issue now. My snom phone is receiving the<br>> INVITE but it is replying with a "404 Not Found" error. (If I<br>
> test with a Jitsi client I don't have the 404 issue)<br>> <br>> <br>> <br>> This shouldn't happen since the TO header is the correct SIP<br>> URI. The only thing that can be wrong is that the To: URI is<br>
> not in <> <br>> <br>> <br>> <br>> I think the TM MI function t_uac_dlg isn't placing the<br>> <> around the TO: header URI. Reading the RFC I am not<br>> 100% sure if the <> are required.<br>
> <br>> <br>> <br>> <br>> <br>> U 2012/08/08 22:09:13.756976 <a href="http://192.168.88.1:5060" target="_blank">192.168.88.1:5060</a> -> <a href="http://192.168.88.13:3072" target="_blank">192.168.88.13:3072</a><br>
> <br>> INVITE <a href="http://sip:9016XX6XX4@192.168.88.13:3072" target="_blank">sip:9016XX6XX4@192.168.88.13:3072</a><br>> SIP/2.0.<br>> <br>> Max-Forwards: 10.<br>> <br></div></div>
> Record-Route: .<br>> <br>> Record-Route: .<div class="HOEnZb"><div class="h5"><br>> <br>> Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0.<br>> <br>> Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0.<br>
> <br>> To: <a href="mailto:sip%3A9016XX6XX4@irck.com" target="_blank">sip:9016XX6XX4@irck.com</a>.<br>> <br>> From: <a href="mailto:sip%3Acontroller@ae.com" target="_blank">sip:controller@ae.com</a>>;tag=134448175329440.<br>
> <br>> CSeq: 1 INVITE.<br>> <br>> Call-ID: 134448175329440.fifouacctd.<br>> <br>> Content-Length: 226.<br>> <br>> User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)).<br>
> <br>> Contact: <a href="http://sip:caller@50.57.54.156:5060" target="_blank">sip:caller@50.57.54.156:5060</a>>.<br>> <br>> Content-Type: application/sdp.<br>> <br>> .<br>> <br>
> v=0.<br>> <br>> o=click2dial 0 0 IN IP4 50.XX.XX.156.<br>> <br>> s=click2dial call.<br>> <br>> c=IN IP4 173.XX.XX.111.<br>> <br>> t=0 0.<br>> <br>> m=audio 12790 RTP/AVP 0 8 18 3 4 97 98.<br>
> <br>> a=rtpmap:0 PCMU/8000.<br>> <br>> a=rtpmap:18 G729/8000.<br>> <br>> a=rtpmap:97 ilbc/8000.<br>> <br>> a=rtpmap:98 speex/8000.<br>> <br>> #<br>> <br>
> U 2012/08/08 22:09:13.766974 <a href="http://192.168.88.13:3072" target="_blank">192.168.88.13:3072</a> -><br>> <a href="http://192.168.88.1:5060" target="_blank">192.168.88.1:5060</a><br>> <br>
> SIP/2.0 404 Not found.<br>> <br>> Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0.<br>> <br>> Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0.<br>> <br>> From: <a href="mailto:sip%3Acontroller@ae.com" target="_blank">sip:controller@ae.com</a>>;tag=134448175329440.<br>
> <br>> To: sip:<a href="tel:9016726924" target="_blank" value="+19016726924">9016726924</a>@<a href="http://irock.com" target="_blank">irock.com</a>>.<br>> <br>> Call-ID: 134448175329440.fifouacctd.<br>
> <br>> CSeq: 1 INVITE.<br>> <br>> User-Agent: snom821/<a href="http://8.7.3.10" target="_blank">8.7.3.10</a>.<br>> <br>> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY,<br>
> SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE.<br>> <br>> Allow-Events: talk, hold, refer, call-info.<br>> <br>> Supported: timer, replaces, from-change.<br>> <br>> Content-Length: 0.<br>
> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>></div></div></blockquote></div><br><br clear="all">
<br>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br>--<br>