there are a few variable that you can set in /usr/local/freeswitch/conf/vars.xml.<br><br><i><b><span style="font-family: courier new,monospace;"> <X-PRE-PROCESS cmd="set" data="call_debug=false"/></span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"> <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/></span><br style="font-family: courier new,monospace;"></b></i><br>You can change it to something like (and then restart FS),<br>
<br><i><b><span style="font-family: courier new,monospace;"> <X-PRE-PROCESS cmd="set" data="call_debug=true"/></span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;"> <X-PRE-PROCESS cmd="set" data="console_loglevel=debug"/></span><br style="font-family: courier new,monospace;">
</b></i><br>Usually it will give you enough information about call processing, however just in case you are looking for SIP trace of a call only then you can enable it on per-profile basis at run-time,<br><br>for example,<br>
<br><b>sofia profile internal siptrace on</b><br><br>this will enable SIP trace for all calls to / from sofia internal profile (which also includes directory users).<br><br>You can run following command on FS console to get information on what profile etc. are available as well as their status.<br>
<br><b>sofia status</b><br><br>For more info consult Wiki page at,<br><br><a href="http://wiki.freeswitch.org/wiki/Sofia">http://wiki.freeswitch.org/wiki/Sofia</a><br><br>Thank you.<br><br><br><div class="gmail_quote">On Sun, Sep 20, 2009 at 4:44 PM, Klaus Teller <span dir="ltr"><<a href="mailto:klaus.teller@gmx.net">klaus.teller@gmx.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi T.,<br>
<br>
I just tried that but i don't see anything different on the console. My test call is going via callcentric and <a href="http://les.net" target="_blank">les.net</a>, but besides the final hop which i normally see in the channel name, there is nothing else.<br>
<br>
Any idea what i might be doing wrong here?<br>
<br>
Thanks,<br>
Klaus.<br>
-------- Original-Nachricht --------<br>
> Datum: Sun, 20 Sep 2009 10:33:01 +0200<br>
> Von: Tihomir Culjaga <<a href="mailto:tculjaga@gmail.com">tculjaga@gmail.com</a>><br>
> An: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
> Betreff: Re: [Freeswitch-users] Call Tracing<br>
<br>
> switch.conf.xml (btw: in console you can enable/disable logging on the fly<br>
> -<br>
> F8/F7)<br>
><br>
> <param name="loglevel" value="debug"/><br>
><br>
><br>
> your relevant sip profile:<br>
><br>
> <param name="sip-trace" value="yes"/><br>
><br>
> T.<br>
><br>
><br>
> On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller <<a href="mailto:klaus.teller@gmx.net">klaus.teller@gmx.net</a>><br>
> wrote:<br>
><br>
> > Hi,<br>
> ><br>
> > Say i have an inbound VoIP/SIP call that hits my FS box. Is it possible<br>
> to<br>
> > to extract information about the intermediate hops that the call or the<br>
> > signaling went through? If so, what information can i get?<br>
> ><br>
> > Thanks,<br>
> > Gregoire.<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Muhammad Shahzad<br>-----------------------------------<br>CISCO Rich Media Communication Specialist (CRMCS)<br>CISCO Certified Network Associate (CCNA)<br>Cell: +92 334 422 40 88<br>
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