<div dir="ltr">...forgot to address your question about seeing the 200 ok sdp being sent. No it was not sending the reply to the caller. Which was making my call enter failure route.<div><br></div><div><br></div><div>Thanks,</div><div>Tito</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Dec 3, 2014 at 7:10 PM, Tito Cumpen <span dir="ltr"><<a href="mailto:tito@xsvoce.com" target="_blank">tito@xsvoce.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Vlad,<div><br><div>My apologies it seems like the pastebin has expired. Please take a look at this one.</div></div><div><br></div><div><br></div><div><a href="http://pastebin.com/XzfZReYB" target="_blank">http://pastebin.com/XzfZReYB</a><br></div><div><br></div><div><br></div><div>I have commented out cancel branch as it seems it is not necessary. Do you think this may have been the cause ? </div><div><br></div><div><br></div><div>
<p><span>onreply_route</span><span>[handle_nat]</span><span> {</span></p>
<p><span> if (nat_uac_test("1"))</span></p>
<p><span> fix_nated_contact()</span><span>;</span></p>
<p><span> if ( isflagset(NAT) )</span></p>
<p><span># rtpproxy_answer("ro");</span></p>
<p><span> if (t_check_status("200")) {</span></p>
<p><span> </span><span># use_media_proxy();</span></p>
<p><span></span><br></p>
<p><span> </span><span># no support for early media</span></p>
<p><span> </span><span># t_cancel_branch("o");</span></p>
<p><span>}</span></p><p><span><br></span></p><p><span><br></span></p><p><span>Thanks,</span></p><p><span>Tito</span></p></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Dec 2, 2014 at 10:27 AM, Vlad Paiu <span dir="ltr"><<a href="mailto:vladpaiu@opensips.org" target="_blank">vladpaiu@opensips.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>Hello,<br>
<br>
Looking at your OpenSIPS logs, it seems that the 200 OK was
succesfully relayed ( 2302 bytes written ) :<br>
<br>
<div>Nov 13 20:05:44 mediaproxy2 /sbin/opensips[311]:
DBG:core:tcp_send: after write: c= 0x7f80dc6230d0 n=2302 fd=41</div>
Nov 13 20:05:44 mediaproxy2 /sbin/opensips[311]:
DBG:core:tcp_send: buf=#012SIP/2.0 200 Ok#015#012Via: SIP/2.0/TCP
192.168.2.93:45442;received=24.186.16.85;branch=z9hG4bK.gLQAkbdo-;rport=45442#015#012From:
<br>
Nov 13 20:05:44 mediaproxy2 /sbin/opensips[311]:
DBG:tm:relay_reply: sent buf=0x7f80e70df3a8: SIP/2.0 2...,
shmem=0x7f80dc616370: SIP/2.0 2<br>
<br>
You do a ngrep trace on the OpenSIPS box, you do not see that 200
OK going out ?<span><br>
Best Regards,<br>
<pre cols="72">Vlad Paiu
OpenSIPS Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a> </pre></span><div><div>
On 02.12.2014 13:41, Tito Cumpen wrote:<br>
</div></div></div><div><div>
<blockquote type="cite">
<div dir="ltr">Any idea what would cause this issue?? Not sure
Michele and I are facing the same issue.</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Wed, Nov 26, 2014 at 3:53 AM,
Michele Pinassi <span dir="ltr"><<a href="mailto:michele.pinassi@unisi.it" target="_blank">michele.pinassi@unisi.it</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> A similar issue
happens to me, when from some voip phones i try to call
other voip phones.<br>
<br>
For example, when from 5002 i try to call 5023.<br>
<br>
Here's my config: <a href="http://pastebin.com/9gP9xncd" target="_blank">http://pastebin.com/9gP9xncd</a><br>
<br>
This is what happens on server <a href="http://172.20.1.2" target="_blank">172.20.1.2</a>:
<a href="http://pastebin.com/gZNSwyHB" target="_blank">http://pastebin.com/gZNSwyHB</a><br>
<br>
This is the log on 5002: <a href="http://pastebin.com/RXtqwMtz" target="_blank">http://pastebin.com/RXtqwMtz</a><br>
<br>
But on 5023 no INVITE arrive... :-( this happens only from
some phones while on others calling 5023 works as
expected.<br>
<br>
Michele<br>
<br>
<br>
<div>Il 25/11/2014 17:24, Vlad Paiu ha scritto:<br>
</div>
<div>
<div>
<blockquote type="cite">
<div>Hello,<br>
<br>
Please provide the SIP trace and the OpenSIPS full
debug log.<br>
<br>
Best Regards,<br>
<pre cols="72">Vlad Paiu
OpenSIPS Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a> </pre>
On 25.11.2014 11:50, Tito Cumpen wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>group</div>
<div><br>
</div>
<div><br>
</div>
I am having an issue in which opensips will
ignore the the 200 ok response to an invite . I
am using Opensips 1.11. This will cause the the
b leg to assume a call and the A leg to be
treated by the failure route. I cant find the
initial invite in the debug logs but see the 200
ok response. I have captures of a working call
vs a non working and cannot find a difference
that would be the reason behind the failure. Is
there anything I can provide to help identify
the cause of this issue?
<div><br>
</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Tito</div>
</div>
<br>
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<span><font color="#888888">
<pre cols="72">--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)2169 - fax: 0577.(23)2053
Per trovare una soluzione rapida ai tuoi problemi tecnici
consulta le FAQ di Ateneo, <a href="http://www.faq.unisi.it" target="_blank">http://www.faq.unisi.it</a> </pre>
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