Re: [OpenSIPS-Users] RTPProxy No Audio on Outbound Calls

2019-02-13 Thread John Quick
Mark, You can detect if the INVITE came from your Asterisk by testing the $si pseudo-variable. That will allow you to identify the direction of the call. I usually set a flag for this purpose. For example: If ($si == "my.ast.er.isk") setflag(DIR_OUT); At the point where you engage the rtpprox

[OpenSIPS-Users] RTPProxy No Audio on Outbound Calls

2019-02-13 Thread Mark Farmer
Hello everyone, all help gratefully received, I've been slogging away at this for ages! I have OpenSIPS 2.4.4 & RTPProxy behind 1:1 NAT's (different hosts). RTPProxy runs so: /usr/local/bin/rtpproxy -s unix:/var/run/rtpproxy/rtpproxy.sock -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -s