Hey Greger,
Thanks for your answer...
I put a register line into sip.conf in asterisk and it didnt work. The
same problem. Even when I call another X-lite, I have the same thing.
You told me that probably I haven't run use_media_proxy() on the OK
from Asterisk, but how do I do that? I didn't understand.
Here is my ser.cfg(only the route part):
route{
if (!mf_process_maxfwd_header("10")) {
if (method!="ACK") {
sl_send_reply("483", "Too many hops");
};
break;
};
if (msg:len >= max_len) {
if (method!="ACK") {
sl_send_reply("513", "Message too big");
};
break;
};
record_route();
if (loose_route()) {
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
# end media session for BYE and CANCEL is done above
# before entering the loose route. no need to call it here
t_relay();
break;
};
if (method=="REGISTER") {
# Mark as NAT'ed
if (client_nat_test("3")) {
setflag(2);
force_rport();
fix_contact();
};
if (!www_authorize("something", "subscriber")) {
www_challenge("something", "0");
break;
}
if (!save("location")) {
sl_reply_error();
};
break;
};
if (method == "BYE" || method == "CANCEL") {
end_media_session();
};
# Force subsequent messages to pass trough this proxy
if (method == "INVITE") {
record_route();
};
if (client_nat_test("3") && !search("^Record-Route:")) {
# Mark as NAT'ed
force_rport();
fix_contact();
};
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
if (!lookup("location")) {
if ( ( uri=~"^sip:[0-9]{4}@.*" ) ) {
t_relay_to_udp("X.X.X.X", "5060");
break;
}
sl_send_reply("404", "User not found");
break;
};
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
Thanks again,
Felipe
--
Master Student - Electrical Engineering Department
Computer Engineering and Telecommunications Research Group
Universidade Federal de Minas Gerais - Brazil
"For God so loved the world that he gave his one and only Son, that
whoever believes in him shall not perish but have eternal life."
John 3:16
On 9/23/05, Greger V. Teigre <greger(a)teigre.com> wrote:
Felipe,
Sounds like asterisk is trying to send directly to xlite (has not registered
with mediaproxy), i.e. you haven't run use_media_proxy() on the OK from
Asterisk.
g-)
----- Original Message -----
From: "Felipe Louback" <louback(a)gmail.com>
To: <serusers(a)lists.iptel.org>
Sent: Friday, September 23, 2005 02:52 AM
Subject: [Serusers] MediaProxy Not Showing The Traffic
I installed ser 8.14 and using mediaproxy 1.4.2.
I am trying to use X-lite to make a call to the PSTN through a PSTN
gateway (asterisk). The PSTN phone rings and get voice, but X-lite
only pass voice. I can hear nothing.
When I use the tool sessions.py available with mediaproxy, it shows me
the call but shows that it is inactive, although the PSTN phone can
hear audio. And after 60 seconds it drops the session, but the PSTN
end still can hear the audio sent from X-lite. And sessions.py does
not show the IP of asterisk....
[root@someplace mediaproxy]# ./sessions.py
Caller Via Called Status Duration
Codec Type Traffic
----------------------------------------------------------------------------------------------
200.150.x.x:8000 - 127.0.0.1:35008 - ?.?.?.?:? inactive 0'10"
Unknown Audio 0/0/0
If I run the rtpgenerator.py it shows the test rtp with traffic flowing.
Here it is what I do in ser.cfg:
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
if (!lookup("location")) {
if ( ( uri=~"^sip:[0-9]{8}@.*" ) ) {
t_relay_to_udp("X.X.X.X", "5060");
break;
}
sl_send_reply("404", "User not found");
break;
};
If I turn off mediaproxy and disable the use_media_proxy, everything works
fine.
Any ideas? I read every possible document and found nothing about this.
Thanks
Felipe
--
Master Student - Electrical Engineering Department
Computer Engineering and Telecommunications Research Group
Universidade Federal de Minas Gerais - Brazil
"For God so loved the world that he gave his one and only Son, that
whoever believes in him shall not perish but have eternal life."
John 3:16
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