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    <p>Thank you for your reply Sergiu.</p>
    <p>Following url with the pcap capture of the two servers,
      kamailio.pcap and asterisk.pcap.</p>
    <p>asterisk.pcap:
<a class="moz-txt-link-freetext" href="https://drive.google.com/file/d/1DP--4NnSBsEQiN4n_yfYoJzsmkv5fr1m/view?usp=sharing">https://drive.google.com/file/d/1DP--4NnSBsEQiN4n_yfYoJzsmkv5fr1m/view?usp=sharing</a></p>
    <p>kamailio.pcap:
<a class="moz-txt-link-freetext" href="https://drive.google.com/file/d/1agcA5Y3MuECdMl0mjnQ1dN4_ht4YHDJU/view?usp=sharing">https://drive.google.com/file/d/1agcA5Y3MuECdMl0mjnQ1dN4_ht4YHDJU/view?usp=sharing</a></p>
    <p><br>
    </p>
    <p>thanks a lot <br>
    </p>
    <p>-- <br>
      <img src="cid:part1.2260CFAF.BF65E329@servicescobrancas.com.br"
        border="0"></p>
    <br>
    <div class="moz-cite-prefix">On 05/02/2018 09:27 PM, Sergiu Pojoga
      wrote:<br>
    </div>
    <blockquote type="cite"
cite="mid:CAJZsGC=HA+x=FafBZtgWrzUr7rMbwYwB3-C4WruU==zmYWckqA@mail.gmail.com">
      <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"
                moz-do-not-send="true">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"
                  moz-do-not-send="true" 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"
                        moz-do-not-send="true">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"
                        moz-do-not-send="true">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"
                      moz-do-not-send="true">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"
                      moz-do-not-send="true" border="0"></div>
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