I will give it a try. Thanks.
--- original message --- From: "Bogdan-Andrei Iancu" <[email protected]> Subject: Re: [OpenSIPS-Users] opensips rtpproxy Date: February 24, 2010 Time: 7:0:38 AM Hi, First of all , be sure that rtpproxy is called two time for each call - once a INVITE time, second at 200 OK time. Put xlogs in the script check this first. Regards, Bogdan bradleyd wrote: > 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 > > -- Bogdan-Andrei Iancu www.voice-system.ro _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
