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Hello,<br>
<br>
can you gran the SIP trace on kamailio server for such case?<br>
<br>
You can use ngrep, like:<br>
<br>
ngrep -d any -qt -W byline port 5060<br>
<br>
and send the output to the mailing list. You can replace any
sensitive information (e.g., ip address) before sending to mailing
list.<br>
<br>
The typical call drop after 30-40 secs is when ACK is not routed
properly, but we have to see that in the sip trace.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div class="moz-cite-prefix">On 25/06/14 18:50, Carlos Rangel wrote:<br>
</div>
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<p class="MsoNormal">Hello<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span lang="EN-US">I have successfully (I
believe) implemented Kamailio 4.1.4 integration with Freepbx
5.2.11 taking as a guide Daniel’s tutorial <a
moz-do-not-send="true"
href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb">http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb</a>.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">I just did not create
the voicemail tables because voice mail is handled by
Freepbx. I installed the system in a separate box for
testing and connected to the Freepbx Production server via
IAX trunk. <o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">The system is behind a
Cisco Firewall and looks like this<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"> Remote
User
Internet Internal network<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Cisco 7960 ----DSL
router ---|------Internet --------|-----Cisco ASA 5500
FW--------------Kamailio/Freepbx (Same Box)------IAX
Trunk----------Freepbx Production Server --------|------
PSTN<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"> I have configured the
FW to allow UDP and TCP traffic from the corresponding IP as
well as tfpt that is needed for the Ciscos to pick up the
configuration from the server. I have a few remotes Cisco
7960 phones that can register remotely in Kamailio as long
as the user is added with kamctl add user password and as
long as the extension is created in Freepbx. <o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">The problem that I have
is when try to make a call from the remote Ciscos the call
is dropped after 30 or 40 seconds. I can see from the logs
that the problem appears to be that the server is not
receiving responses from the phone<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal" style="text-autospace:none"><span
style="font-size:10.0pt;font-family:"Lucida
Console"" lang="EN-US">06-25 10:57:30] WARNING[1814]
chan_sip.c: Retransmission timeout reached on transmission
<a class="moz-txt-link-abbreviated" href="mailto:000653dc-39400006-2579bbcd-13d9adcb@192.168.0.22">000653dc-39400006-2579bbcd-13d9adcb@192.168.0.22</a> for seqno
102 (Critical Response) -- See
<a class="moz-txt-link-freetext" href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a><o:p></o:p></span></p>
<p class="MsoNormal" style="text-autospace:none"><span
style="font-size:10.0pt;font-family:"Lucida
Console"" lang="EN-US">Packet timed out after 32001ms
with no response<o:p></o:p></span></p>
<p class="MsoNormal" style="text-autospace:none"><span
style="font-size:10.0pt;font-family:"Lucida
Console"" lang="EN-US">[2014-06-25 10:57:30]
WARNING[1814] chan_sip.c: Hanging up call
<a class="moz-txt-link-abbreviated" href="mailto:000653dc-39400006-2579bbcd-13d9adcb@192.168.0.22">000653dc-39400006-2579bbcd-13d9adcb@192.168.0.22</a> - no reply
to our critical packet (see
<a class="moz-txt-link-freetext" href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Is this something that
we can adjust in kamailio or could it be related to the FW
configuration?? Sorry but I am very new to kamailio and
sip.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Thanks<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Carlos<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
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<pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
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<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a></pre>
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