<html>
  <head>
    <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    Hello,<br>
    <br>
    run asterisk in debug mode to understand why is sending BYE.<br>
    <br>
    Cheers,<br>
    Daniel<br>
    <br>
    <div class="moz-cite-prefix">On 07/08/15 16:40, Loic Chabert wrote:<br>
    </div>
    <blockquote
cite="mid:CADLP+M0Nra9QeE8DC6cmvh-rAwpewFaj=zfSRh31zKuMBUbqyw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>
          <div>
            <div>
              <div>
                <div>
                  <div>
                    <div>
                      <div>
                        <div>
                          <div>Hello, <br>
                            <br>
                          </div>
                          I have set on the right place
                          "route(RTPPROXY")", and now it works for
                          internal calls and external calls.<br>
                        </div>
                        <div>Reason: my request passing througt RTPPROXY
                          twice ...<br>
                        </div>
                        <div><br>
                        </div>
                        One last problem:<br>
                      </div>
                      - 102 initiate a call to 101<br>
                    </div>
                    - 101 refuse call with a 486 response<br>
                  </div>
                  - as asterisk dialplan said: launch voicemail app<br>
                </div>
                - Sounds files has been read from asterisk, but after 5
                secondes, session has been cut with a BYE request sent
                by Asterisk.<br>
                <br>
              </div>
              Please find in attachement pcap trace file (91.x.x.x is
              wan kamailio interface, 10.0.247.197 is lan kamailio
              interface, facing to asterisk cluster)<br>
              <br>
            </div>
            Why asterisk send this BYE ? Kamailio does not force him to
            send this BYE...<br>
            <br>
          </div>
          Thanks,<br>
        </div>
        Loic.<br>
        <div>
          <div>
            <div>
              <div>
                <div>
                  <div>
                    <div>
                      <div>
                        <div>
                          <div><br>
                          </div>
                        </div>
                      </div>
                    </div>
                  </div>
                </div>
              </div>
            </div>
          </div>
        </div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">2015-08-07 10:34 GMT+02:00
          Daniel-Constantin Mierla <span dir="ltr"><<a
              moz-do-not-send="true" href="mailto:miconda@gmail.com"
              target="_blank">miconda@gmail.com</a>></span>:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div bgcolor="#FFFFFF" text="#000000"> Hello,<br>
              <br>
              look at the sip traffic and see what is in SDP, if you
              don't get audio, maybe the other ip is advertised.<br>
              <br>
              Cheers,<br>
              Daniel
              <div>
                <div class="h5"><br>
                  <br>
                  <div>On 07/08/15 09:16, Loic Chabert wrote:<br>
                  </div>
                  <blockquote type="cite">
                    <div dir="ltr">
                      <div>
                        <div>
                          <div>
                            <div>
                              <div>
                                <div>Hello Daniel, <br>
                                  <br>
                                </div>
                                I have changed my rtpproxy by rtpengine.
                                I have explicitly define public and
                                private interfaces, and now it work as
                                expected for external calls (througth
                                PSTN).<br>
                              </div>
                              But for now, after this change, internal
                              call (like 100 call 101), does not work
                              any more.<br>
                              <br>
                            </div>
                            I need more investigation to see what append
                            on my call flow.<br>
                            <br>
                          </div>
                          I will update you asap.<br>
                          <br>
                        </div>
                        Thanks,<br>
                      </div>
                      Regards.<br>
                      <div>
                        <div>
                          <div>
                            <div><br>
                            </div>
                          </div>
                        </div>
                      </div>
                    </div>
                    <div class="gmail_extra"><br>
                      <div class="gmail_quote">2015-08-07 9:04 GMT+02:00
                        Daniel-Constantin Mierla <span dir="ltr"><<a
                            moz-do-not-send="true"
                            href="mailto:miconda@gmail.com"
                            target="_blank">miconda@gmail.com</a>></span>:<br>
                        <blockquote class="gmail_quote" style="margin:0
                          0 0 .8ex;border-left:1px #ccc
                          solid;padding-left:1ex">
                          <div bgcolor="#FFFFFF" text="#000000"> Hello,<span><br>
                              <br>
                              <div>On 30/07/15 17:38, Loic Chabert
                                wrote:<br>
                              </div>
                              <blockquote type="cite">
                                <div dir="ltr">
                                  <div>
                                    <div>Hello everyone,<br>
                                      <br>
                                    </div>
                                    I'm trying put kamailio in front of
                                    asterisk server farm. Fow now, 2
                                    asterisk servers are running and i'm
                                    trying to make some basic calls
                                    between two UACc.<br>
                                    <br>
                                    All asterisk servers has been
                                    ofuscaded from public internet using
                                    <a moz-do-not-send="true"
                                      href="http://10.189.122.0/24"
                                      target="_blank">10.189.122.0/24</a>
                                    network.<br>
                                  </div>
                                  <div>All trafic must be passed
                                    throught asterisk so RTPproxy is
                                    used to (and used for rtp bridging).<br>
                                  </div>
                                  <div>Kamailio and rtpproxy is running
                                    with public IP address, and private
                                    ip address (mhomed=1)<br>
                                  </div>
                                  <div><br>
                                  </div>
                                  <div>But a wired thing append on my
                                    SDP body: c line have two rtpproxy
                                    public addresses concatenate (see my
                                    capture attached).<br>
                                    <br>
                                  </div>
                                  <div>Any reason for this ? Only invite
                                    method from my asterisk contains 2
                                    publics IP addresses concatenated.<br>
                                  </div>
                                  <div><br>
                                  </div>
                                  <div>Does it mean than rtp_manage as
                                    been executed twice ?<br>
                                    <br>
                                  </div>
                                </div>
                              </blockquote>
                            </span> It could be that it was executed
                            twice. As pointed in another response, look
                            at what is received on the network and in
                            the logs.<br>
                            <br>
                            You can enable cfgtrace for debugger module
                            in order to see what actions are executed
                            from configuration files -- it is good to
                            spot quickly errors in the logic of config
                            file.<br>
                            <br>
                            Cheers,<br>
                            Daniel<span><font color="#888888"><br>
                                <br>
                                <pre cols="72">-- 
Daniel-Constantin Mierla
<a moz-do-not-send="true" href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a moz-do-not-send="true" href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a moz-do-not-send="true" href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
                              </font></span></div>
                          <br>
_______________________________________________<br>
                          SIP Express Router (SER) and Kamailio
                          (OpenSER) - sr-users mailing list<br>
                          <a moz-do-not-send="true"
                            href="mailto:sr-users@lists.sip-router.org"
                            target="_blank">sr-users@lists.sip-router.org</a><br>
                          <a moz-do-not-send="true"
                            href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"
                            rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
                          <br>
                        </blockquote>
                      </div>
                      <br>
                    </div>
                  </blockquote>
                  <br>
                  <pre cols="72">-- 
Daniel-Constantin Mierla
<a moz-do-not-send="true" href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a moz-do-not-send="true" href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a moz-do-not-send="true" href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
                </div>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Daniel-Constantin Mierla
<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>
Book: SIP Routing With Kamailio - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a></pre>
  </body>
</html>