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<div class="moz-cite-prefix">Hello,<br>
<br>
The <> are only required if you want to have SIP header
parameters for the TO header.<br>
Otherwise, if there are no <> , all the parameters are
considered to be SIP URI parameters.<br>
So, from what I see, that TO header is correct.<br>
<br>
Regards,
<pre class="moz-signature" cols="72">Vlad Paiu
OpenSIPS Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a>
</pre>
On 08/09/2012 06:13 AM, Duane Larson wrote:<br>
</div>
<blockquote
cite="mid:CAFcM1ErbvZ815U1pns5jpSoV4KC3HTcg_NVO-ubaC87M1OXc=Q@mail.gmail.com"
type="cite">
<div>I changed the following in the ctd.sh script</div>
<div> </div>
<div>Changed the default of</div>
<div>"`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"`</div>
<div> </div>
<div>To<br>
"`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"`</div>
<div> </div>
<div> </div>
<div>And now it is making it into the OpenSIPS/SBC's main route.
Not sure why.</div>
<div> </div>
<div>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)</div>
<div> </div>
<div>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 <> </div>
<div> </div>
<div>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.</div>
<div> </div>
<div> </div>
<div>U 2012/08/08 22:09:13.756976 <a moz-do-not-send="true"
href="http://192.168.88.1:5060">192.168.88.1:5060</a> -> <a
moz-do-not-send="true" href="http://192.168.88.13:3072">192.168.88.13:3072</a><br>
INVITE <a moz-do-not-send="true"
href="http://sip:9016XX6XX4@192.168.88.13:3072">sip:9016XX6XX4@192.168.88.13:3072</a>
SIP/2.0.<br>
Max-Forwards: 10.<br>
Record-Route: <sip:192.168.88.1;r2=on;lr>.<br>
Record-Route: <sip:99.XX.XX.161;r2=on;lr>.<br>
Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0.<br>
Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0.<br>
<strong>To: <a moz-do-not-send="true"
href="mailto:sip%3A9016XX6XX4@irck.com">sip:9016XX6XX4@irck.com</a>.</strong><br>
From: <<a moz-do-not-send="true"
href="mailto:sip%3Acontroller@ae.com">sip:controller@ae.com</a>>;tag=134448175329440.<br>
CSeq: 1 INVITE.<br>
Call-ID: 134448175329440.fifouacctd.<br>
Content-Length: 226.<br>
User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)).<br>
Contact: <<a moz-do-not-send="true"
href="http://sip:caller@50.57.54.156:5060">sip:caller@50.57.54.156:5060</a>>.<br>
Content-Type: application/sdp.<br>
.<br>
v=0.<br>
o=click2dial 0 0 IN IP4 50.XX.XX.156.<br>
s=click2dial call.<br>
c=IN IP4 173.XX.XX.111.<br>
t=0 0.<br>
m=audio 12790 RTP/AVP 0 8 18 3 4 97 98.<br>
a=rtpmap:0 PCMU/8000.<br>
a=rtpmap:18 G729/8000.<br>
a=rtpmap:97 ilbc/8000.<br>
a=rtpmap:98 speex/8000.</div>
<div>#<br>
U 2012/08/08 22:09:13.766974 <a moz-do-not-send="true"
href="http://192.168.88.13:3072">192.168.88.13:3072</a> ->
<a moz-do-not-send="true" href="http://192.168.88.1:5060">192.168.88.1:5060</a><br>
SIP/2.0 404 Not found.<br>
Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0.<br>
Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0.<br>
From: <<a moz-do-not-send="true"
href="mailto:sip%3Acontroller@ae.com">sip:controller@ae.com</a>>;tag=134448175329440.<br>
To: <<a moz-do-not-send="true"
href="mailto:sip%3A9016726924@irock.com">sip:9016726924@irock.com</a>>.<br>
Call-ID: 134448175329440.fifouacctd.<br>
CSeq: 1 INVITE.<br>
User-Agent: snom821/<a moz-do-not-send="true"
href="http://8.7.3.10">8.7.3.10</a>.<br>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY,
SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE.<br>
Allow-Events: talk, hold, refer, call-info.<br>
Supported: timer, replaces, from-change.<br>
Content-Length: 0.</div>
<div> </div>
<div><br>
</div>
<br>
</blockquote>
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