11 Mar
2006
11 Mar
'06
12:30 a.m.
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(a)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 ?
------------------------------
*From:* users-bounces(a)openser.org [mailto:users-bounces@openser.org] *On
Behalf Of *Script Head
*Sent:* Thursday, March 09, 2006 6:44 PM
*To:* users(a)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(a)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(a)openser.org [mailto: users-bounces(a)openser.org] *On
Behalf Of *Script Head
*Sent:* Thursday, March 09, 2006 12:59 PM
*To:* users(a)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(a)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(a)openser.org [mailto: users-bounces(a)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) -> 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