RE: [Users] forcing rtpproxy on a call

1. I never used forward, see my example, I do not know if it actually relay call or not 2. if you do not have NAT between client and server you do not need force_rport, and try to avoid any nat_uac_test, etc if you are actually working on private ips without nat 3. you MUST enable proxy also for reply route[x] { ..... force_rtp_proxy(); t_on_reply("1"); rewritehostport("x.x.x.x:5060"); if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (!(status=~"183" || status=~"200")) break; force_rtp_proxy(""); } ________________________________ From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Script Head Sent: Wednesday, March 08, 2006 6:29 PM To: users(a)openser.org Subject: [Users] forcing rtpproxy on a call Hello everyone, I am trying to debug why my rtpproxy isn't working. I have the following setup, on my LAN. softphone (192.168.1.100) -> openser/rtpproxy ( 192.168.1.10 <http://192.168.1.10> ) -> asterisk (192.168.1.12) The rtpproxy is running and I see commands flying thru it. the following route works if(method=="INVITE") { if(uri=~"^sip:[0-9]{6}1[0-9]*{10}@") { forward(192.168.1.12,5060); }; } when I replace it with this route if(method=="INVITE") { if(uri=~"^sip:[0-9]{6}1[0-9]*{10}@") { forward(192.168.1.12,5060); }; force_rport(); force_rtp_proxy(); } I get dead air while asterisk logs show that my test message is playing. How should I proceed to debug this? ScriptHead