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    Cual es el resultao de:<br>
    <br>
    <b>sip show settings</b><br>
    <br>
    <div class="moz-cite-prefix">El 29/04/13 9:10, Moosa Khalid
      escribi&oacute;:<br>
    </div>
    <blockquote
cite="mid:CAH+n-jXgfKEaeD8zDrSm6ZwBLUVXC_3ncMp6P9qcLoYq_Ri+NQ@mail.gmail.com"
      type="cite">
      <div dir="ltr">I'm trying to make a video call between two SIP
        peers registered on a single asterisk box. Following is the
        config of my sip peers
        <div><br>
        </div>
        <div>
          <div>[107]</div>
          <div>defaultuser=107</div>
          <div>secret=107</div>
          <div>type=friend</div>
          <div>host=dynamic</div>
          <div>context=default</div>
          <div>canreinvite=yes</div>
          <div>videosupport=yes</div>
          <div>dtmfmode=rfc2833</div>
          <div>qualify=yes</div>
          <div>disallow=all</div>
          <div>allow=ulaw</div>
          <div>
            allow=alaw</div>
          <div>allow=gsm</div>
          <div>allow=h263</div>
          <div><br>
          </div>
          <div>[701]<br>
          </div>
          <div>defaultuser=701</div>
          <div>secret=701</div>
          <div>type=friend</div>
          <div>host=dynamic</div>
          <div>context=default</div>
          <div>videosupport=yes</div>
          <div>qualify=yes</div>
          <div>canreinvite=yes</div>
          <div>dtmfmode=rfc2833</div>
          <div>disallow=all</div>
          <div>allow=ulaw</div>
          <div>allow=alaw</div>
          <div>allow=gsm</div>
          <div>allow=h263</div>
        </div>
        <div style=""><br>
        </div>
        <div style="">I've also enabled videosupport in the global sip
          settings.</div>
        <div style="">I'm using the softphone Xlite 4.5 which supports
          h263 codec on for both clients. <b>Asterisk version is 11.3.0
            LTS</b>. Clients registered are on same network. Asterisk
          shows the following output on console on a call b/w peers.
          Audio is fine but of course no video thanks to the following
          warning.&nbsp;</div>
        <div style=""><br>
        </div>
        <div style="">
          <div><b>&nbsp; == Using SIP VIDEO CoS mark 6</b></div>
          <div><b>&nbsp; == Using SIP RTP CoS mark 5</b></div>
          <div><b>&nbsp; &nbsp; -- Executing [96107@default:1]
              Dial("SIP/701-00000014", "SIP/107,20,rt") in new stack</b></div>
          <div><b>&nbsp; == Using SIP VIDEO CoS mark 6</b></div>
          <div><b>&nbsp; == Using SIP RTP CoS mark 5</b></div>
          <div><b>&nbsp; &nbsp; -- Called SIP/107</b></div>
          <div><b>&nbsp; &nbsp; -- SIP/107-00000015 is ringing</b></div>
          <div><b>&nbsp; &nbsp; -- SIP/107-00000015 is ringing</b></div>
          <div><b>[Apr 29 18:35:12] WARNING[11363][C-0000000d]:
              chan_sip.c:10141 process_sdp: Ignoring video stream offer
              because port number is zero</b></div>
          <div><b><br>
            </b></div>
          <div style="">This happens regardless of which sip peer
            originates the call.</div>
          <div><br>
          </div>
        </div>
      </div>
      <br>
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      <br>
      <pre wrap="">--
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    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Alberto Llamas
Ingeniero de Telecomunicaciones
Digium Certified Asterisk Administrator
Digium Certified Asterisk Professional
Linux Administrator</pre>
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