make sure to set the flags on your application to say it supports no media, you should find some examples in mod_dptools. Then you can set channel vars for either codecs, or the entire sdp string before calling switch_channel_pre_answer. Mike
I have a similar problem to the one posted above, which I got by searching the archieves: http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-May/003653.html
We have integrated our own IVR with FS. Our IVR handles all media (RTP) directly and FS handles only SIP.
When a call comes to FS with INVITE, in response we send 183 Session Progress with FS generated SDP (and RTP IP and port). In 200 OK, FS sends SDP with our application generated RTP IP and port. This works in most cases.
Is there any way to invoke my application before sending 183 Session Progress, so that our application generated SDP can be sent in the 183 Session Progress?