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