<div dir="auto"><div>32 seconds is the default asterisk T2 timer. Probably some ACK is not being relayed following BYE. </div><div dir="auto"><br></div><div dir="auto">Would help to see some sip traces. <br><br><div class="gmail_quote" dir="auto"><div dir="ltr">On Wed, May 2, 2018, 5:40 PM Jeferson Oliveira, <<a href="mailto:oliveira1.jeferson@servicescobrancas.com.br">oliveira1.jeferson@servicescobrancas.com.br</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<p>Btw, the version of kamailio is 4.2.3. </p>
<p>Thank you.<br>
</p>
-- <br>
<img src="cid:part1.97F3DEDB.15D31152@servicescobrancas.com.br" border="0">
<div class="m_3548078539755489536moz-cite-prefix">On 05/02/2018 06:29 PM, Jeferson
Oliveira wrote:<br>
</div>
<blockquote type="cite">
Hello everyone,<br>
<br>
I have an error that I have not yet been able to solve and would
like the help of colleagues to indicate a correct path.<br>
The problem that is occurring is that when the client disconnects
the call kamailio is not sending the BYE forward until arriving at
the asterisk.<br>
<br>
Both in the test scenario and in the production scenario the
problem is the same and the message I see in the capture is 404
Not here, msg this coming from kamailio.<br>
<br>
Production scenario.<br>
<br>
PSTN <----------> Dialer --------->kamailio
-----------> asterisk1<br>
-----------> asterisk2 <br>
<br>
Test scenario.<br>
<p>sipp generated calls ------> kamailio -------> asterisk1</p>
<p>
-------> asterisk2</p>
<br>
<br>
When this occurs, the calls that are disconnected by the client
are in a "zombie" state in asterisk, and end up being terminated
by timeout with the following message in the asterisk CLI:<br>
<br>
<font size="-1"><i>[Apr 25 17:49:59] WARNING[2121]:
chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on
transmission <a class="m_3548078539755489536moz-txt-link-abbreviated" href="mailto:22-6073@10.110.7.242" target="_blank" rel="noreferrer">22-6073@10.110.7.242</a>
for seqno 1 (Critical Response) -- See <a class="m_3548078539755489536moz-txt-link-freetext" href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank" rel="noreferrer">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a></i><i><br>
</i><i>Packet timed out after 31999ms with no response</i></font><br>
<br>
In the sipp panel I see in the Retransmission column several
incrementing counters, as per the attachment.<br>
<br>
If I take the kamailio from the move and point the sipp to only
one of the asterisk, the retransmissions do not happen and BYE
follows normally.
<p>My kamailio.cfg configuration file can be downloaded from this
url:
<a class="m_3548078539755489536moz-txt-link-freetext" href="https://drive.google.com/file/d/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=sharing" target="_blank" rel="noreferrer">https://drive.google.com/file/d/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=sharing</a></p>
<p><br>
</p>
<p>Thank you very much.</p>
<div class="m_3548078539755489536moz-signature">-- <br>
<img src="cid:part1.97F3DEDB.15D31152@servicescobrancas.com.br" border="0"></div>
</blockquote>
<br>
<div class="m_3548078539755489536moz-signature"><br>
</div>
</div>
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