Hello, opensips ver: 1.6. (not svn) rtpproxy ver: 1.2.1 (not svn) Debian lenny
I am trying to setup opensips with rtpproxy. I cant seem to get it to work. I either have no audio or one way--depending on the changes. We are not registering any users and do not require authorization for any incoming calls. The incoming calls will go either to a set of ivr's or to a set of gateway's via dialplan and dynamic routing. I can get the audio/dtmf to work if I set a src_ip conditional in on_reply route; but this seems like a awful idea. I am unsure where to go from here, any help would be appreciated. Here is a link to my config http://pastie.org/837255 Thanks _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
