Hi Nandini,
The parameters and modules that you need to turn ON in your opensips.cfg file:
loadmodule "nat_traversal.so"
The above line load the module and the given below paragraph will set to test
the parameters.
route[nat_check] {
if (client_nat_test("3")) {
force_rport();
fix_contact();
nat_keepalive();
}
}
Everytime you route a call first test the calls through the route(nat_check)
which will fix all the NAT handling parameters.
For e.g. if you are gonna route INVITE request then you need to do it like this:
if(is_method("INVITE")) {
route(invite_requests);
exit;
}
route[invite_requests] {
route(nat_check);
if(!lookup("location")) {
sl_send_reply("404", "User Not registered");
exit;
}
t_on_reply("user_reply");
t_relay();
exit;
}
Its just an example that how I do it and always you can explore things and read
the modules provided by OpenSIPS and upgrade yourself to use this server in all
possible cases.
Regards,
Aamir Chougule
Cell: 09167989111
________________________________
From: Aamir <[email protected]>
To: OpenSIPS users mailling list <[email protected]>
Sent: Tuesday, 7 May 2013 1:58 PM
Subject: Re: [OpenSIPS-Users] one way audio problem
Hi Nandini,
You actually need to turn on the nat_traversal module I guess, will pass you
the parameters if I get time to do so.
--Aamir
--- Sent from My BlackBerry ---
-----Original Message-----
From: sermj 2012 <[email protected]>
Sender: [email protected]
Date: Tue, 7 May 2013 13:49:04
To: <[email protected]>
Reply-To: OpenSIPS users mailling list <[email protected]>
Subject: [OpenSIPS-Users] one way audio problem
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