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Hello,<br>
<br>
On 9/28/11 2:43 AM, JJ Zeng wrote:
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<p class="MsoNormal"><span style="font-size:14.0pt">Hi,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:14.0pt"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:14.0pt">I am really
new to ser(just installed it today). My situation is here:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:14.0pt"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:14.0pt">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. <o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:14.0pt"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size: 14pt;">It looks
really simple, but I just can’t figure out how to do it.</span></p>
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what default config file you started with?<br>
<br>
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.<br>
<br>
Cheers,<br>
Daniel<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla -- <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
Kamailio Advanced Training, Oct 10-13, Berlin: <a class="moz-txt-link-freetext" href="http://asipto.com/u/kat">http://asipto.com/u/kat</a>
<a class="moz-txt-link-freetext" href="http://linkedin.com/in/miconda">http://linkedin.com/in/miconda</a> -- <a class="moz-txt-link-freetext" href="http://twitter.com/miconda">http://twitter.com/miconda</a></pre>
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