Hi all, My configuration is: USER ----(connects to)----> ASTERISK---(connects to)--->CARRIER-OUT i want the user preffered codec to pass thru asterisk to carrier-out. what i mean is: USER ----(user uses g729)----> ASTERISK---(asterisk should use g729 for dialing out)--->CARRIER-OUT instead, this is what happens USER ----(user uses g729)----> ASTERISK---(asterisk uses g711u)--->CARRIER-OUT How can i force asterisk to use user preffered codec for dialing out so that my asterisk machine saves time by no conversion USER PREFERENCE IS disallow=all allow=g729 CARRIER PREFERENCE IS allow=all Anybody who can help? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070530/1cbabf1e/attachment.htm
so you r sure you have g729 licences installed and ur * is transcoding your RTP streaming? Test the work flow with disallow=all and allow=g729, can be my mistake but I remember to read somewhere on the net any issue about codec negotiating precedence when you use allow=all. good luck On 5/30/07, Rizwan Hisham <rizwanhasham@gmail.com> wrote:> > Hi all, > My configuration is: > USER ----(connects to)----> ASTERISK---(connects to)--->CARRIER-OUT > > i want the user preffered codec to pass thru asterisk to carrier-out. what > i mean is: > USER ----(user uses g729)----> ASTERISK---(asterisk should use g729 for > dialing out)--->CARRIER-OUT > > instead, this is what happens > USER ----(user uses g729)----> ASTERISK---(asterisk uses > g711u)--->CARRIER-OUT > > How can i force asterisk to use user preffered codec for dialing out so > that my asterisk machine saves time by no conversion > USER PREFERENCE IS > disallow=all > allow=g729 > > CARRIER PREFERENCE IS > allow=all > > Anybody who can help? > > -- > Rizwan Hisham > Software Engineer > AXVOICE Inc. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- Esta mensagem (incluindo quaisquer anexos) pode conter informa??o confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070530/9fabeb5d/attachment.htm
Does anybody has any documentation on codec negotiation within asterisk? Well im using free g729 codec for testing purposes. i mentioned g729 just as an example. whatever codec is mentioned in user perefernce, asterisk uses ulaw to throw out the call. On 5/30/07, Marco Mouta <marco.mouta@gmail.com> wrote:> > so you r sure you have g729 licences installed and ur * is transcoding > your RTP streaming? > > Test the work flow with disallow=all and allow=g729, can be my mistake but > I remember to read somewhere on the net any issue about codec negotiating > precedence when you use allow=all. > > good luck > > On 5/30/07, Rizwan Hisham <rizwanhasham@gmail.com> wrote: > > > Hi all, > > My configuration is: > > USER ----(connects to)----> ASTERISK---(connects to)--->CARRIER-OUT > > > > i want the user preffered codec to pass thru asterisk to carrier-out. > > what i mean is: > > USER ----(user uses g729)----> ASTERISK---(asterisk should use g729 for > > dialing out)--->CARRIER-OUT > > > > instead, this is what happens > > USER ----(user uses g729)----> ASTERISK---(asterisk uses > > g711u)--->CARRIER-OUT > > > > How can i force asterisk to use user preffered codec for dialing out so > > that my asterisk machine saves time by no conversion > > USER PREFERENCE IS > > disallow=all > > allow=g729 > > > > CARRIER PREFERENCE IS > > allow=all > > > > Anybody who can help? > > > > -- > > Rizwan Hisham > > Software Engineer > > AXVOICE Inc. > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > -- > Esta mensagem (incluindo quaisquer anexos) pode conter informa??o > confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio > pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se recebeu > esta mensagem por engano, por favor informe o emissor e elimine-a > imediatamente. Obrigado. > > This e-mail message is intended only for individual(s) to whom it is > addressed and may contain information that is privileged, confidential, > proprietary, or otherwise exempt from disclosure under applicable law. If > you believe you have received this message in error, please advise the > sender by return e-mail and delete it from your mailbox. Thank you. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070531/b2545a40/attachment.htm
Asterisk by default uses the codec preferred by other device/client . Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough to check if it can avoid transcoding by forcing same codec on other side of conversation . If both sides prefer g729 then asterisk does not do transcoding but if one side prefer gsm and other prefers g729 and the gsm side can also support g729 still asterisk will transcode . Someone posted a patch to this in mantis bug tracking system at digium for 1.2 .. google for it and maybe you can find . On 31/05/07, Rizwan Hisham <rizwanhasham@gmail.com> wrote:> Does anybody has any documentation on codec negotiation within asterisk? > > Well im using free g729 codec for testing purposes. i mentioned g729 just as > an example. whatever codec is mentioned in user perefernce, asterisk uses > ulaw to throw out the call. > > > On 5/30/07, Marco Mouta <marco.mouta@gmail.com> wrote: > > so you r sure you have g729 licences installed and ur * is transcoding > your RTP streaming? > > > > Test the work flow with disallow=all and allow=g729, can be my mistake but > I remember to read somewhere on the net any issue about codec negotiating > precedence when you use allow=all. > > > > good luck > > > > > > > > On 5/30/07, Rizwan Hisham < rizwanhasham@gmail.com> wrote: > > > > > > Hi all, > > > My configuration is: > > > USER ----(connects to)----> ASTERISK---(connects to)--->CARRIER-OUT > > > > > > i want the user preffered codec to pass thru asterisk to carrier-out. > what i mean is: > > > USER ----(user uses g729)----> ASTERISK---(asterisk should use g729 for > dialing out)--->CARRIER-OUT > > > > > > instead, this is what happens > > > USER ----(user uses g729)----> ASTERISK---(asterisk uses > g711u)--->CARRIER-OUT > > > > > > How can i force asterisk to use user preffered codec for dialing out so > that my asterisk machine saves time by no conversion > > > USER PREFERENCE IS > > > disallow=all > > > allow=g729 > > > > > > CARRIER PREFERENCE IS > > > allow=all > > > > > > Anybody who can help? > > > > > > -- > > > Rizwan Hisham > > > Software Engineer > > > AXVOICE Inc. > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > -- > > Esta mensagem (incluindo quaisquer anexos) pode conter informa??o > confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio > pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se recebeu > esta mensagem por engano, por favor informe o emissor e elimine-a > imediatamente. Obrigado. > > > > This e-mail message is intended only for individual(s) to whom it is > addressed and may contain information that is privileged, confidential, > proprietary, or otherwise exempt from disclosure under applicable law. If > you believe you have received this message in error, please advise the > sender by return e-mail and delete it from your mailbox. Thank you. > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > -- > Rizwan Hisham > Software Engineer > AXVOICE Inc. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >
That just seems really, REALLY dumb for a program of this magnitude. I know this has been patched here and there by one person or another, but does anyone know if any of these patches to make CODEC negotiation actually, you know, negotiate a CODEC will ever make it into the core src? Jaswinder Singh wrote:> Asterisk by default uses the codec preferred by other device/client . > Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough > to check if it can avoid transcoding by forcing same codec on other > side of conversation . If both sides prefer g729 then asterisk does > not do transcoding but if one side prefer gsm and other prefers g729 > and the gsm side can also support g729 still asterisk will transcode . > Someone posted a patch to this in mantis bug tracking system at digium > for 1.2 .. google for it and maybe you can find . > > On 31/05/07, Rizwan Hisham <rizwanhasham@gmail.com> wrote: >> Does anybody has any documentation on codec negotiation within asterisk? >> >> Well im using free g729 codec for testing purposes. i mentioned g729 >> just as >> an example. whatever codec is mentioned in user perefernce, asterisk >> uses >> ulaw to throw out the call. >> >> >> On 5/30/07, Marco Mouta <marco.mouta@gmail.com> wrote: >> > so you r sure you have g729 licences installed and ur * is transcoding >> your RTP streaming? >> > >> > Test the work flow with disallow=all and allow=g729, can be my >> mistake but >> I remember to read somewhere on the net any issue about codec >> negotiating >> precedence when you use allow=all. >> > >> > good luck >> > >> > >> > >> > On 5/30/07, Rizwan Hisham < rizwanhasham@gmail.com> wrote: >> > > >> > > Hi all, >> > > My configuration is: >> > > USER ----(connects to)----> ASTERISK---(connects to)--->CARRIER-OUT >> > > >> > > i want the user preffered codec to pass thru asterisk to >> carrier-out. >> what i mean is: >> > > USER ----(user uses g729)----> ASTERISK---(asterisk should use >> g729 for >> dialing out)--->CARRIER-OUT >> > > >> > > instead, this is what happens >> > > USER ----(user uses g729)----> ASTERISK---(asterisk uses >> g711u)--->CARRIER-OUT >> > > >> > > How can i force asterisk to use user preffered codec for dialing >> out so >> that my asterisk machine saves time by no conversion >> > > USER PREFERENCE IS >> > > disallow=all >> > > allow=g729 >> > > >> > > CARRIER PREFERENCE IS >> > > allow=all >> > > >> > > Anybody who can help? >> > > >> > > -- >> > > Rizwan Hisham >> > > Software Engineer >> > > AXVOICE Inc. >> > > _______________________________________________ >> > > --Bandwidth and Colocation provided by Easynews.com -- >> > > >> > > asterisk-users mailing list >> > > To UNSUBSCRIBE or update options visit: >> > > >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > >> > > >> > >> > >> > >> > -- >> > Esta mensagem (incluindo quaisquer anexos) pode conter informa??o >> confidencial para uso exclusivo do destinat?rio. Se n?o for o >> destinat?rio >> pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se >> recebeu >> esta mensagem por engano, por favor informe o emissor e elimine-a >> imediatamente. Obrigado. >> > >> > This e-mail message is intended only for individual(s) to whom it is >> addressed and may contain information that is privileged, confidential, >> proprietary, or otherwise exempt from disclosure under applicable >> law. If >> you believe you have received this message in error, please advise the >> sender by return e-mail and delete it from your mailbox. Thank you. >> > _______________________________________________ >> > --Bandwidth and Colocation provided by Easynews.com -- >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> > >> >> >> >> -- >> Rizwan Hisham >> Software Engineer >> AXVOICE Inc. >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
has anybody made a patch for asterisk 1.4*? On 6/4/07, Jaswinder Singh <vicky.r@gmail.com> wrote:> > Asterisk by default uses the codec preferred by other device/client . > Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough > to check if it can avoid transcoding by forcing same codec on other > side of conversation . If both sides prefer g729 then asterisk does > not do transcoding but if one side prefer gsm and other prefers g729 > and the gsm side can also support g729 still asterisk will transcode . > Someone posted a patch to this in mantis bug tracking system at digium > for 1.2 .. google for it and maybe you can find . > > On 31/05/07, Rizwan Hisham <rizwanhasham@gmail.com> wrote: > > Does anybody has any documentation on codec negotiation within asterisk? > > > > Well im using free g729 codec for testing purposes. i mentioned g729 > just as > > an example. whatever codec is mentioned in user perefernce, asterisk > uses > > ulaw to throw out the call. > > > > > > On 5/30/07, Marco Mouta <marco.mouta@gmail.com> wrote: > > > so you r sure you have g729 licences installed and ur * is transcoding > > your RTP streaming? > > > > > > Test the work flow with disallow=all and allow=g729, can be my mistake > but > > I remember to read somewhere on the net any issue about codec > negotiating > > precedence when you use allow=all. > > > > > > good luck > > > > > > > > > > > > On 5/30/07, Rizwan Hisham < rizwanhasham@gmail.com> wrote: > > > > > > > > Hi all, > > > > My configuration is: > > > > USER ----(connects to)----> ASTERISK---(connects to)--->CARRIER-OUT > > > > > > > > i want the user preffered codec to pass thru asterisk to > carrier-out. > > what i mean is: > > > > USER ----(user uses g729)----> ASTERISK---(asterisk should use g729 > for > > dialing out)--->CARRIER-OUT > > > > > > > > instead, this is what happens > > > > USER ----(user uses g729)----> ASTERISK---(asterisk uses > > g711u)--->CARRIER-OUT > > > > > > > > How can i force asterisk to use user preffered codec for dialing out > so > > that my asterisk machine saves time by no conversion > > > > USER PREFERENCE IS > > > > disallow=all > > > > allow=g729 > > > > > > > > CARRIER PREFERENCE IS > > > > allow=all > > > > > > > > Anybody who can help? > > > > > > > > -- > > > > Rizwan Hisham > > > > Software Engineer > > > > AXVOICE Inc. > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > > > asterisk-users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > > > > > > > -- > > > Esta mensagem (incluindo quaisquer anexos) pode conter informa??o > > confidencial para uso exclusivo do destinat?rio. Se n?o for o > destinat?rio > > pretendido, n?o dever? usar, distribuir ou copiar este e-mail. Se > recebeu > > esta mensagem por engano, por favor informe o emissor e elimine-a > > imediatamente. Obrigado. > > > > > > This e-mail message is intended only for individual(s) to whom it is > > addressed and may contain information that is privileged, confidential, > > proprietary, or otherwise exempt from disclosure under applicable law. > If > > you believe you have received this message in error, please advise the > > sender by return e-mail and delete it from your mailbox. Thank you. > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > -- > > Rizwan Hisham > > Software Engineer > > AXVOICE Inc. > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070604/98b9b2df/attachment.htm