New subject: [Users] forcing rtpproxy on a call

9 Mar 2006

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

RE: [Users] forcing rtpproxy on a call