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    <div class="moz-cite-prefix">Hi Joshua, thanks for answer.<br>
      In this particular test my extension is on a simple network. There
      is no NAT, just an asterisk running on a virtual machine on a 7-
      64bit CentOs. I am simulating an environment to be able to use
      PJSIP on my client. And even in this small environment, my
      extension does not call.<br>
      <br>
      My problem with NAT was with SIP "one way audio" on a client. All
      of this testing is to replace SIP with PJSIP on this client. But
      as the queue is unable to call a PJSIP extension, the migration
      project on the client is stopped.</div>
    <div class="moz-cite-prefix"><br>
    </div>
    <div class="moz-cite-prefix"><br>
      I tried to separate the debug file, but it seems to me that in
      asterisk 17.16.0, there is a problem or I did not know how to
      configure it, because the log did not generate it either.<br>
    </div>
    <div class="moz-cite-prefix">on console:<br>
    </div>
    <div class="moz-cite-prefix">"pjsip set logger on"</div>
    <div class="moz-cite-prefix">"pjsip set history on"</div>
    <div class="moz-cite-prefix"><br>
    </div>
    <div class="moz-cite-prefix">on file Logger.conf:<br>
      debbuger => debug, trace</div>
    <div class="moz-cite-prefix"><br>
    </div>
    <div class="moz-cite-prefix">asterisk -rx "reload"</div>
    <div class="moz-cite-prefix"><br>
    </div>
    <div class="moz-cite-prefix">Make same calls, and opening the file
      only the following appears:</div>
    <div class="moz-cite-prefix"><br>
    </div>
    <div class="moz-cite-prefix">[2020-08-18 08:46:47.778] Asterisk
      17.6.0 built by root @ asterisk-homolog on a x86_64 running Linux
      on 2020-08-13 22:40:11 UTC\</div>
    <div class="moz-cite-prefix"><br>
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    <div class="moz-cite-prefix">Em 17/08/2020 18:57, Joshua C. Colp
      escreveu:<br>
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cite="mid:CAM0A2Z0kmCu7+O83=x2K1ktnnto03M2Nq7L_bdgzRtLeOJaNgw@mail.gmail.com">
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        <div dir="ltr">On Mon, Aug 17, 2020 at 6:16 PM Roberto <<a
            href="mailto:roberto.medola@gasparimsantos.com.br"
            moz-do-not-send="true">roberto.medola@gasparimsantos.com.br</a>>
          wrote:<br>
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style="margin:0px;padding:0px;color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:14px;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial">Hello.</p>
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style="margin:0px;padding:0px;color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:14px;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial"><br>
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              <p
style="margin:0px;padding:0px;color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:14px;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial">I
                am having a lot of problems with SIP through NAT. So, I
                decided to adopt PJSIP. However, I am not able to make
                the extensions ring when receiving a call from the
                queue. I'm using telnet to include the extension and on
                the asterisk console, it even shows Called PJSIP/6001,
                but the extension doesn't ring. If I call from extension
                to extension, it works normally.</p>
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        <div>Can you describe the actual network setup further? Is the
          endpoint behind NAT or merely Asterisk? I ask because there is
          no NAT configuration for the endpoint, which if it is behind
          one can be problematic. Failing that you'll need to provide a
          SIP trace using "pjsip set logger on" to show the actual SIP
          traffic flowing (and where to).</div>
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        -- <br>
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                          <div style="font-family:tahoma,sans-serif"><font
                              color="#073763">Joshua C. Colp</font></div>
                          <div style="font-family:tahoma,sans-serif"><font
                              color="#073763">Asterisk Technical Lead</font></div>
                          <div style="font-family:tahoma,sans-serif"><font
                              color="#073763">Sangoma Technologies</font></div>
                          <div style="font-family:tahoma,sans-serif"><font
                              color="#073763">Check us out at <a
                                href="http://www.sangoma.com"
                                target="_blank" moz-do-not-send="true">www.sangoma.com</a>
                              and <a href="http://www.asterisk.org"
                                target="_blank" moz-do-not-send="true">www.asterisk.org</a></font><br>
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