[OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

[duane.larson at gmail.com]

Thanks for the info. I'll check with Snom and see why the phone is  
rejecting the INVITE.




On , Vlad Paiu <vladpaiu at opensips.org> 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:9016XX6XX4 at 192.168.88.13:3072
> 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:9016XX6XX4 at irck.com.

> From: sip:controller at ae.com>;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:caller at 50.57.54.156:5060>.

> 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:controller at ae.com>;tag=134448175329440.

> To: sip:9016726924 at irock.com>.

> 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.
















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