Hello,
from your config snippets I don't figure out what version are you using
and what flavour, does not look like latest ser or kamailio, and they
are not relevant for troubleshooting.
Maybe you want to start with the default config, for kamailio flavor
there is a tutorial to install it from git and enable authentication:
http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git
Then, if you look in kamailio.cfg, you see option to enable other
features like nat traversal, presence, etc...
Cheers,
Daniel
On 9/29/11 12:09 AM, JJ Zeng wrote:
Thanks Daniel. I got it basically working with the following config.
But there is one last bug. If client B connected and registered, then
disconnected without proper unregister, it seems the location table
didn't get prompt update. If client A sends Invite right after that,
the sip server will try to connect to B(which will surely fail since B
is behind the firewall), instead of sending 480 to A right away. Is
there a way to fix this?
Thanks.
JJ
if (is_method("REGISTER"))
{
# authenticate the REGISTER requests (uncomment to
enable auth)
if (!www_authorize("", "subscriber"))
{
www_challenge("", "0");
exit;
}
if (!db_check_to())
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
if (!save("location", "m"))
sl_reply_error();
exit;
}
if (!lookup("location")) {
#if (!registered("location")) {
sl_send_reply("480","Not Found");
exit;
}
if (!registered("location")) {
sl_send_reply("480","Not Found");
exit;
}
if(!t_relay()) {
sl_send_reply("500", "relaying failed");
}
exit;
*From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com]
*Sent:* Wednesday, September 28, 2011 2:49 AM
*To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
Users Mailing List
*Cc:* JJ Zeng
*Subject:* Re: [SR-Users] simple question but can't find the answer
Hello,
On 9/28/11 2:43 AM, JJ Zeng wrote:
Hi,
I am really new to ser(just installed it today). My situation is here:
Both A and B make tcp connection to ser proxy server. If A sends
"Invite" to B, ser would look up the tcp connection of B, and relay
the sip request. If B is not connected, then 480 is returned to A.
When B sends back the response, ser would relay it to A.
It looks really simple, but I just can't figure out how to do it.
what default config file you started with?
Apart of that when the user is not online, the default config replies
with 404 Not Found, the relaying between phones when they are online
is done there, being a combination of calling function from registrar
and tm modules. So, if you start with the default config, no matter is
Kamailio or SER flavour, routing of the calls between online users is
provided there, just go through it in order to understand better.
Cheers,
Daniel
--
Daniel-Constantin Mierla --http://www.asipto.com
Kamailio Advanced Training, Oct 10-13,
Berlin:http://asipto.com/u/kat
http://linkedin.com/in/miconda --
http://twitter.com/miconda
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--
Daniel-Constantin Mierla --
http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin:
http://asipto.com/u/kat
http://linkedin.com/in/miconda --
http://twitter.com/miconda