In that case you should turn on sip trace for profile where your callcentric peer is configured. By default FS comes with two profiles namely internal and external. If you haven't created any new profile and configured your users and peers in these two profiles then you should try turning on sip trace for external profile too (or just external profile alone).<br>
<br><b>sofia profile external siptrace on</b><br><br>Please check your peer configuration and turn on sip trace on appropriate profile.<br><br>Thank you.<br><br><br><div class="gmail_quote">On Sun, Sep 20, 2009 at 5:49 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;">Thanks. I tried that and what it shows me is the trace between my peer and the SIP provider (i.e. <a href="http://les.net" target="_blank">les.net</a>). The call is actually coming from callcentric and i don't see that in the trace. Is it supposed to show this?<br>
<br>
Klaus.<br>
<br>
-------- Original-Nachricht --------<br>
> Datum: Sun, 20 Sep 2009 17:11:50 +0600<br>
> Von: Muhammad Shahzad <<a href="mailto:shaheryarkh@googlemail.com">shaheryarkh@googlemail.com</a>><br>
<div><div></div><div class="h5">> An: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
> Betreff: Re: [Freeswitch-users] Call Tracing<br>
<br>
> there are a few variable that you can set in<br>
> /usr/local/freeswitch/conf/vars.xml.<br>
><br>
> * <X-PRE-PROCESS cmd="set" data="call_debug=false"/><br>
> <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/><br>
> *<br>
> You can change it to something like (and then restart FS),<br>
><br>
> * <X-PRE-PROCESS cmd="set" data="call_debug=true"/><br>
> <X-PRE-PROCESS cmd="set" data="console_loglevel=debug"/><br>
> *<br>
> Usually it will give you enough information about call processing, however<br>
> just in case you are looking for SIP trace of a call only then you can<br>
> enable it on per-profile basis at run-time,<br>
><br>
> for example,<br>
><br>
> *sofia profile internal siptrace on*<br>
><br>
> this will enable SIP trace for all calls to / from sofia internal profile<br>
> (which also includes directory users).<br>
><br>
> You can run following command on FS console to get information on what<br>
> profile etc. are available as well as their status.<br>
><br>
> *sofia status*<br>
><br>
> For more info consult Wiki page at,<br>
><br>
> <a href="http://wiki.freeswitch.org/wiki/Sofia" target="_blank">http://wiki.freeswitch.org/wiki/Sofia</a><br>
><br>
> Thank you.<br>
><br>
><br>
> On Sun, Sep 20, 2009 at 4:44 PM, Klaus Teller <<a href="mailto:klaus.teller@gmx.net">klaus.teller@gmx.net</a>><br>
> wrote:<br>
><br>
> > Hi T.,<br>
> ><br>
> > I just tried that but i don't see anything different on the console. My<br>
> > test call is going via callcentric and <a href="http://les.net" target="_blank">les.net</a>, but besides the final<br>
> hop<br>
> > 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<br>
> > 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<br>
> possible<br>
> > > to<br>
> > > > to extract information about the intermediate hops that the call or<br>
> the<br>
> > > > signaling went through? If so, what information can i get?<br>
> > > ><br>
> > > > Thanks,<br>
> > > > Gregoire.<br>
> > > > --<br>
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><br>
><br>
><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>
> MSN: <a href="mailto:shari_786pk@hotmail.com">shari_786pk@hotmail.com</a><br>
> Email: <a href="mailto:shaheryarkh@googlemail.com">shaheryarkh@googlemail.com</a><br>
<br>
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</div></div></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>
MSN: <a href="mailto:shari_786pk@hotmail.com">shari_786pk@hotmail.com</a><br>Email: <a href="mailto:shaheryarkh@googlemail.com">shaheryarkh@googlemail.com</a><br>