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<font size="-1"><font face="Verdana">Hi,<br>
<br>
you can use ESL (</font></font><a
href="http://wiki.freeswitch.org/wiki/Event_Socket_Library">http://wiki.freeswitch.org/wiki/Event_Socket_Library</a>)
<font size="-1"><font face="Verdana">to receive certain events </font></font><a
href="http://wiki.freeswitch.org/wiki/Event_list">http://wiki.freeswitch.org/wiki/Event_list</a><br>
<br>
<font size="-1"><font face="Verdana">or you can use information
stored in XML CDR data to do your analysis: </font></font><a
href="http://wiki.freeswitch.org/wiki/Mod_xml_cdr">http://wiki.freeswitch.org/wiki/Mod_xml_cdr</a><font
size="-1"><font face="Verdana"><br>
<br>
</font></font>
<pre class="moz-signature" cols="72">Best regards / Mit freundlichen Grüßen,
Daniel Neubert
</pre>
Am 01.01.2011 22:46, schrieb Frank Park:
<blockquote
cite="mid:AANLkTi=OP-eVAr2QpxEghVN1v-ASFHUgC9n_3MyPRsm1@mail.gmail.com"
type="cite">
<pre wrap="">Happy New Years!
This is merely a wish list for me, but I wanted to know if there's a
way for me to log all calls that disconnects without the proper SIP
termination..
I would like to build some level of debug tool where I can track
dropped calls due to freeswitch, or due to any other external issues
(which I assume most are).
In the same statistics, I am also interested in logging who initially
sent the BYE message (the caller, the callee, or the switch).
A direction to do this might be sufficient at this point.
Thank you!
Frank
----=======================----
Frank Park
Telonium Communications, LLC
<a class="moz-txt-link-abbreviated" href="mailto:frank@telonium.com">frank@telonium.com</a>
<a class="moz-txt-link-freetext" href="http://www.telonium.com">http://www.telonium.com</a>
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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</pre>
</blockquote>
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