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@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@gmail.com To: serusers@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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