I want to originate calls from the asterisk machine directly, and connect them locally to different audio devices. (each in and out, which themselfes are redirected via rtp-streams)
The configuration for simple dialouts is working, but i have no clue how to connect multiple simultaneous calls to different audio devices.
Currently I use chan_oss to determine die audio device for local calls.
Is it somehow possible to do what I ask for, or is it one thing that can’t be done with asterisk? Another question, does anyone know whether it can be done with the cygwin port of asterisk?
I’d be really appreciated for any help in this topic.
btw, excuse my bad english