jonc
2005-Nov-29 13:41 UTC
[Asterisk-Users] route call based on codec? (g723 gets message, g729 goes to conf connection)
I have a rather curious integration problem. I need to direct a call connection based on the codec used for the connection. If my softswitch attaches to the Asterisk server using G729 I toss the connection into a requested conference - that works fine. On occasion my softswitch will attach to the Asterisk server using G723 (and request joining a conference that is using G729). When that happens I need to feed the connection a stock announcement (recorded in G723) and then hang up. Is there a way to direct a call based on the codec used to attach to the Asterisk server? === More detail for those scratching their heads... I'm using Asterisk servers to augment my Vocal Data softswitch. One of the many things that Asterisk does for me is act as a conference bridge. This works just dandy except that my softswitch uses the conference bridge to transcode Voicemail announcements. My Softswitch automagically transcodes all announcements into G711, G723, and G729. Whenever someone records a voicemail announcement the VM server opens a conference using each of the codecs - plays the announcement in G729 (our default) and then records on the other connections. Obviously the G723 connection does not work since Asterisk won't transcode G723. That's cool. We don't *ever* use G723 - it's just built into the softswitch. The problem comes with the fact that the softswitch won't give up on doing the transcoding to G723. It continues to try and try and try and try... There is nothing dumber than a machine doing a task it can never finish. Unless its a machine opening hundereds of connections to my conferencing bridge trying to do a task it can never complete. I need to feed it something - anything - in a G723 format. I've got plenty of G723 audio files. If I can simply play one to the g723 connection then it will be happy and go away. ;-) Any help is appreciated. Thanks - Jon Carnes
Giovanni Miano
2005-Nov-30 03:56 UTC
[Asterisk-Users] route call based on codec? (g723 gets message, g729 goes to conf connection)
If u are using 1.2 there is global var SIP_CODEC or IAX_CODE exten => 88,1,NoOP(${SIP_CODEC}) exten => 88,2,NoOP(${IAX_CODEC}) Try 29 Nov 2005 15:41:38 -0500, jonc <jonc@ftnc.net>:> I have a rather curious integration problem. I need to direct a call > connection based on the codec used for the connection. > > If my softswitch attaches to the Asterisk server using G729 I toss the > connection into a requested conference - that works fine. > > On occasion my softswitch will attach to the Asterisk server using G723 > (and request joining a conference that is using G729). When that happens > I need to feed the connection a stock announcement (recorded in G723) > and then hang up. > > Is there a way to direct a call based on the codec used to attach to the > Asterisk server? > > ===> > More detail for those scratching their heads... > > I'm using Asterisk servers to augment my Vocal Data softswitch. One of > the many things that Asterisk does for me is act as a conference bridge. > This works just dandy except that my softswitch uses the conference > bridge to transcode Voicemail announcements. > > My Softswitch automagically transcodes all announcements into G711, > G723, and G729. Whenever someone records a voicemail announcement the VM > server opens a conference using each of the codecs - plays the > announcement in G729 (our default) and then records on the other > connections. > > Obviously the G723 connection does not work since Asterisk won't > transcode G723. That's cool. We don't *ever* use G723 - it's just built > into the softswitch. > > The problem comes with the fact that the softswitch won't give up on > doing the transcoding to G723. It continues to try and try and try and > try... There is nothing dumber than a machine doing a task it can never > finish. Unless its a machine opening hundereds of connections to my > conferencing bridge trying to do a task it can never complete. > > I need to feed it something - anything - in a G723 format. I've got > plenty of G723 audio files. If I can simply play one to the g723 > connection then it will be happy and go away. ;-) > > Any help is appreciated. Thanks - Jon Carnes > > _______________________________________________ > --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 >-- Giovanni Miano