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Greetings List.<br>I Have a small test server and i'm facing a small issue. <br>i have setup two SIP PEERS and they are able to do Video calls.<br>now I'm testing SET SIP_CODEC in a dial plan and when ever i'm setting the codec .. the inbound (=first) leg stops receiving or sending video and SIP SHOW CHANNELS shows only the Codec i set in the dialplan.<br>is it possible to avoid this problem? <br><br>Asterisk version <br>1.8.11.0<br><br>SIP.CONF<br>=======<br><br>[TK1000]<br>type=friend<br>secret=0jCiOdT81P<br>videosupport=yes<br>qualify=yes<br>host=dynamic<br>dtmfmode=rfc2833<br>context=DER-TEST<br>canreinvite=yes<br>disallow=all<br>allow=ulaw,alaw,gsm,h263,h263p<br><br>[TK1000]<br>type=friend<br>secret=0jCiOdT81P<br>videosupport=yes<br>qualify=yes<br>host=dynamic<br>dtmfmode=rfc2833<br>context=DER-TEST<br>canreinvite=yes<br>disallow=all<br>allow=ulaw,alaw,gsm,h263,h263p<br><br><br>EXTENSIONS.CONF<br>[DER-TEST]<br>;exten => _.,1,NoCDR()<br>exten => _.,1,Set(SIP_CODEC=alaw)<br>exten => _.,2,Set(SIP_CODEC_OUTBOUND=gsm)<br>;exten => _.,2,Set(SIP_CODEC_INBOUND=gsm)<br>exten => _.,n,DIAL(SIP/TK${EXTEN})<br>exten => h,1,Hangup()<br><br><br><br><br>Tarek Sawah<br><br>Information Technology Adviser<br><br>Integrated Digital Systems<br><br>CCNP, MCSE, RHCE, TELECOM<br><br>USA: +1 386 492 9993<br><br> </div></body>
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