Hello Everyone, I had one way audo (out) for weeks, which was ok for our testing. Plans were to integrate RTPProxy to manage two way audo, but for now one way was ok.
Not sure what the hell I changed, and now I have no audio at all. This is for an OpenSIPS -> Asterisk integration. I'm really streesed about this and not sure where to start debugging this thing anymore. Please Help, Nick. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
