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@openser.org [mailto:users-bounces@openser.org] On Behalf Of Script Head
Sent: Wednesday, March 08, 2006
6:29 PM
To: users@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) ->
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