[Asterisk-Users] route call based on codec? (g723 gets message, g729 goes to conf connection)

[jonc]

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