<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">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 class=""><span class="">onreply_route</span><span class="">[handle_nat]</span><span class=""> {</span></p>
<p class=""><span class=""> if (nat_uac_test("1"))</span></p>
<p class=""><span class=""> fix_nated_contact()</span><span class="">;</span></p>
<p class=""><span class=""> if ( isflagset(NAT) )</span></p>
<p class=""><span class=""># rtpproxy_answer("ro");</span></p>
<p class=""><span class=""> if (t_check_status("200")) {</span></p>
<p class=""><span class=""> </span><span class=""># use_media_proxy();</span></p>
<p class=""><span class=""></span><br></p>
<p class=""><span class=""> </span><span class=""># no support for early media</span></p>
<p class=""><span class=""> </span><span class=""># t_cancel_branch("o");</span></p>
<p class=""><span class="">}</span></p><p class=""><span class=""><br></span></p><p class=""><span class=""><br></span></p><p class=""><span class="">Thanks,</span></p><p class=""><span class="">Tito</span></p></div></div><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 class=""><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 class="h5">
On 02.12.2014 13:41, Tito Cumpen wrote:<br>
</div></div></div><div><div class="h5">
<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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