Now that my rtpproxy actually passes traffic I stumbled on another problem. When the called party hangs up the call (asterisk command Hangup()) the soft phone remains connected. Yet, when I click the Hangup button on the softphone, SER receives BYE messages.

On 3/9/06, Vitaly Nikolaev <vnikolaev@intermedia.net> wrote:

Looks like forward includes relay in it. And by putting force_rtpproxy AFTER forward you does not give it a chance J on_reply route is also MUST be there.

 

 

From: users-bounces@openser.org [mailto: users-bounces@openser.org] On Behalf Of Script Head
Sent: Thursday, March 09, 2006 12:59 PM
To: users@openser.org
Subject: Re: [Users] forcing rtpproxy on a call

 

Thank you guys, it's working now.

Apparently, rewritehostport("<ip>:<port>") works great with rptproxy while forward does exactly that, forwards the call to the destination bypassing the force_rtp_proxy request. This should be documented somewhere.

ScriptHead

On 3/9/06, Vitaly Nikolaev <vnikolaev@intermedia.net> wrote:

  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@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