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

[Vlad Paiu]

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 
> <http://192.168.88.1:5060> -> 192.168.88.13:3072 
> <http://192.168.88.13:3072>
> INVITE sip:9016XX6XX4 at 192.168.88.13:3072 
> <http://sip:9016XX6XX4@192.168.88.13:3072> SIP/2.0.
> Max-Forwards: 10.
> Record-Route: <sip:192.168.88.1;r2=on;lr>.
> Record-Route: <sip:99.XX.XX.161;r2=on;lr>.
> 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 <mailto:sip%3A9016XX6XX4 at irck.com>.*
> From: <sip:controller at ae.com 
> <mailto:sip%3Acontroller 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 
> <http://sip:caller@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 
> <http://192.168.88.13:3072> -> 192.168.88.1:5060 
> <http://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 
> <mailto:sip%3Acontroller at ae.com>>;tag=134448175329440.
> To: <sip:9016726924 at irock.com <mailto:sip%3A9016726924 at irock.com>>.
> Call-ID: 134448175329440.fifouacctd.
> CSeq: 1 INVITE.
> User-Agent: snom821/8.7.3.10 <http://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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