[OpenSIPS-Users] opensips rtpproxy

[bradleyd]

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