If you don't want to proxy the media through * the put this setting: canreinvite=yes this will allow the 2 end points to connect directly for the RTP bypassing you. otherwise I have noticed the same when I try to proxy I have to make sure everyone is using the same codec or it doesn't work well. .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Yves Sent: Thursday, 17 March, 2005 12:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Codec negociation Hi, I've got an Asterisk latest CVS head with oh323 installed. There is one thing I can't understand about the codec negociation. I receive calls in G723&G729, and send them to another gateway who can handle both codecs too. So all I want to do is just passthrou, for both. It seems that * only try to send with the first of the list, what is fine when it's the good one, but otherwise he complain about being unable to transcode instead of trying the second codec. I hope I've explained well my problem. Could someone explain me a little bit more about the negociation ? Or did someone have the same issue ? I didn't find much info, tried docs & google. Thank you. Yves _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
I read about this option. But does it work on a h323 channel ? (inAccessnetwork's one) Brian C. Fertig wrote:> If you don't want to proxy the media through * the put this setting: > > canreinvite=yes > > this will allow the 2 end points to connect directly for the RTP > bypassing > you. otherwise I have noticed the same when I try to proxy I have to > make sure everyone is using the same codec or it doesn't work well. > > > > .o-------------------------------------------------------o. > Brian Fertig > NOC/Network Engineer > Planet Telecom, Inc. > > > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Yves > Sent: Thursday, 17 March, 2005 12:28 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Codec negociation > > Hi, > > I've got an Asterisk latest CVS head with oh323 installed. There is one > thing I can't understand about the codec negociation. I receive calls in > > G723&G729, and send them to another gateway who can handle both codecs > too. So all I want to do is just passthrou, for both. It seems that * > only try to send with the first of the list, what is fine when it's the > good one, but otherwise he complain about being unable to transcode > instead of trying the second codec. > > I hope I've explained well my problem. Could someone explain me a little > > bit more about the negociation ? Or did someone have the same issue ? > I didn't find much info, tried docs & google. > > Thank you. > > Yves > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
After looking everywhere I still don't have any solution. Transcoding G729 is not a problem, Digium is selling licences at reasonable price, but transcoding G723 is a huge problem (look at the prices!). And the fact is that this is a quite often used codec. I receive G729 & G723 calls that I send to a provider who can handle both too, is it impossible to tell Asterisk to keep using the same codec for in & out ? It seems that he only follows the codec list in order. If we can handle this, it means less CPU need and less licences. Please tell me I'm not the only one who need this and there is a way to do it. It could be really good to have an option that gives a priority to the codec used in the incoming call when negociating the outgoing call. Yves