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<tt>Hi Duane,<br>
<br>
That is really strange. Logically speaking, a change in SDP should
have no impact on receiving (or not) a message - the SDP is just
payload and is not even parsed by opensips (unless you ask it from
script by using rtpproxy/mediaproxy/sipmsgops modules). Anyhow,
the request should make it to the main route.<br>
<br>
I rather suspect that something else is filtering your SIP
traffic, dropping the INVITEs in the first format :-/ . <br>
<br>
Regarding the second issue:<br>
<br>
1) the angle brackets are not mandatory by RFC - they should be
used only if you have URI params or a complex display name (which
is not your case here)<br>
<br>
2) assuming that the phone does not like TO - it should reply with
a 400 Bad request, not a 404 - a 404 represents a routing
indication and routing is done based RURI not TO.<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<br>
On 08/09/2012 06:13 AM, Duane Larson wrote:
<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: <a class="moz-txt-link-rfc2396E" href="sip:192.168.88.1;r2=on;lr"><sip:192.168.88.1;r2=on;lr></a>.<br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:99.XX.XX.161;r2=on;lr"><sip:99.XX.XX.161;r2=on;lr></a>.<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>
<div class="gmail_quote">On Fri, Jul 27, 2012 at 3:24 PM, <span
dir="ltr"><<a moz-do-not-send="true"
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: 1px solid rgb(204, 204, 204);"
class="gmail_quote">Very sure. Normal calls are working with
clients behind the OpenSIPS/SBC.
<div class="im">
<br>
<br>
<br>
<br>
On , Bogdan-Andrei Iancu <<a moz-do-not-send="true"
href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>>
wrote:<br>
> <br>
> <br>
> <br>
> <br>
> <br>
> <br>
</div>
<div>
<div class="h5">
> Duane,some stupid question : are you sure your
opensips is<br>
> listening on the given IP:port ? have you check with
netstat ?<br>
> also have you checked with netstat also if there is
traffic queued<br>
> on the sockets ?<br>
> <br>
> <br>
> <br>
> Regards,<br>
> <br>
> <br>
> Bogdan-Andrei Iancu<br>
> OpenSIPS Founder and Developer<br>
> <a moz-do-not-send="true"
href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a><br>
> <br>
> <br>
> On 07/27/2012 12:48 AM, Duane Larson wrote:<br>
> Oh yeah. My first email has the SIPTrace from the<br>
> OpenSIPS/SBC. So I am logged into the OpenSIPS/SBC
and did the<br>
> NGREP. So I see it SIP invite (99.XX.XX.161 is the
IP of the<br>
> OpenSIPS/SBC). I would even let you log into the
OpenSIPS/SBC<br>
> and see it for yourself. Makes no sense.<br>
> <br>
> <br>
> <br>
> U 2012/07/19 18:20:13.486847 50.XX.XX.156:5060 ->
99.XX.XX.161:5060<br>
> <br>
> INVITE <a class="moz-txt-link-freetext" href="sip:9016XXXXX@192.168.88.13:3072;line=g2hfphrk">sip:9016XXXXX@192.168.88.13:3072;line=g2hfphrk</a>
SIP/2.0.<br>
> <br>
> Via: SIP/2.0/UDP
50.57.54.156;branch=z9hG4bKaab1.6c7ffd84.0.<br>
> <br>
> To: <a moz-do-not-send="true"
href="mailto:sip%3A9016XXXXX@irock.com" target="_blank">sip:9016XXXXX@irock.com</a>.<br>
> <br>
> From: <a moz-do-not-send="true"
href="mailto:sip%3Acontroller@ae.com" target="_blank">sip:controller@ae.com</a>>;tag=134274001013257.<br>
> <br>
> CSeq: 1 INVITE.<br>
> <br>
> Call-ID: 134274001013257.fifouacctd.<br>
> <br>
> Content-Length: 155.<br>
> <br>
> User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)).<br>
> <br>
> Contact: <a moz-do-not-send="true"
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=click-to-dial 0 0 IN IP4 0.0.0.0.<br>
> <br>
> s=session.<br>
> <br>
> c=IN IP4 0.0.0.0.<br>
> <br>
> b=CT:1000.<br>
> <br>
> t=0 0.<br>
> <br>
> m=audio 9 RTP/AVP 8 0.<br>
> <br>
> a=rtpmap:8 PCMA/8000.<br>
> <br>
> a=rtpmap:0 PCMU/8000. <br>
> </div>
</div>
</blockquote>
</div>
</blockquote>
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