has anybody made a patch for asterisk 1.4*?<br><br><div><span class="gmail_quote">On 6/4/07, <b class="gmail_sendername">Jaswinder Singh</b> <<a href="mailto:vicky.r@gmail.com">vicky.r@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Asterisk by default uses the codec preferred by other device/client .<br>Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough<br>to check if it can avoid transcoding by forcing same codec on other<br>side of conversation . If both sides prefer g729 then asterisk does
<br>not do transcoding but if one side prefer gsm and other prefers g729<br>and the gsm side can also support g729 still asterisk will transcode .<br>Someone posted a patch to this in mantis bug tracking system at digium<br>
for 1.2 .. google for it and maybe you can find .<br><br>On 31/05/07, Rizwan Hisham <<a href="mailto:rizwanhasham@gmail.com">rizwanhasham@gmail.com</a>> wrote:<br>> Does anybody has any documentation on codec negotiation within asterisk?
<br>><br>> Well im using free g729 codec for testing purposes. i mentioned g729 just as<br>> an example. whatever codec is mentioned in user perefernce, asterisk uses<br>> ulaw to throw out the call.<br>><br>
><br>> On 5/30/07, Marco Mouta <<a href="mailto:marco.mouta@gmail.com">marco.mouta@gmail.com</a>> wrote:<br>> > so you r sure you have g729 licences installed and ur * is transcoding<br>> your RTP streaming?
<br>> ><br>> > Test the work flow with disallow=all and allow=g729, can be my mistake but<br>> I remember to read somewhere on the net any issue about codec negotiating<br>> precedence when you use allow=all.
<br>> ><br>> > good luck<br>> ><br>> ><br>> ><br>> > On 5/30/07, Rizwan Hisham < <a href="mailto:rizwanhasham@gmail.com">rizwanhasham@gmail.com</a>> wrote:<br>> > ><br>> > > Hi all,
<br>> > > My configuration is:<br>> > > USER ----(connects to)----> ASTERISK---(connects to)--->CARRIER-OUT<br>> > ><br>> > > i want the user preffered codec to pass thru asterisk to carrier-out.
<br>> what i mean is:<br>> > > USER ----(user uses g729)----> ASTERISK---(asterisk should use g729 for<br>> dialing out)--->CARRIER-OUT<br>> > ><br>> > > instead, this is what happens
<br>> > > USER ----(user uses g729)----> ASTERISK---(asterisk uses<br>> g711u)--->CARRIER-OUT<br>> > ><br>> > > How can i force asterisk to use user preffered codec for dialing out so<br>
> that my asterisk machine saves time by no conversion<br>> > > USER PREFERENCE IS<br>> > > disallow=all<br>> > > allow=g729<br>> > ><br>> > > CARRIER PREFERENCE IS<br>> > > allow=all
<br>> > ><br>> > > Anybody who can help?<br>> > ><br>> > > --<br>> > > Rizwan Hisham<br>> > > Software Engineer<br>> > > AXVOICE Inc.<br>> > > _______________________________________________
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<br>><br>><br>><br>> --<br>> Rizwan Hisham<br>> Software Engineer<br>> AXVOICE Inc.<br>> _______________________________________________<br>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">
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<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>Rizwan Hisham<br>Software Engineer<br>
AXVOICE Inc.