I created a setup on my LAN with that works great and forwards BYE all the way to the soft phone. I verified via tcpdump that RTP is flowing via the proxy. When I move this setup to the public Internet, it works as well, except the BYE isn't getting forwarded. The config file is attached.

On 3/10/06, Vitaly Nikolaev vnikolaev@intermedia.net wrote:

Add

if (!method=="REGISTER") record_route();

Somewhere in ur route (before relay but after loose_route)

do u have loose_route anyway ?

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*From:* users-bounces@openser.org [mailto:users-bounces@openser.org] *On Behalf Of *Script Head *Sent:* Thursday, March 09, 2006 6:44 PM *To:* users@openser.org

*Subject:* Re: [Users] forcing rtpproxy on a call

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.

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*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("");

}

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*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