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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I have noticed this, too. I can open up ekiga on the machine FS
is installed on, and it can accept calls, but, if I remember correctly, FS
doesn’t show that the call is being routed, meaning ekiga is just accepting all
the calls itself. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>What I also found, in relation to this, is that if you have
another sip contact outside the FS box, and you try to call them with a sip
client also outside the FS box, then the client that is running on the FS box (in
my case, Ekiga) will accept the calls that are supposed to be for other users.
<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>So, basically, I don’t think you can have a client on the same
machine as FS. I suppose that is to be expected, as a sip client is also a
server. Am I right?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>A related problem I have is if you have multiple sip accounts on
the same machine (registering from the same ip address), they can’t call each other.
It may stem from the problem above, but it may not.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I’m also a quite a newbie in FS, so hopefully we can figure out
how to do these things. I would gladly add the answer to the wiki for others if
there’s something specific we need to do to do these kind of things.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>-Dan<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Pete
Kay<br>
<b>Sent:</b> Thursday, April 17, 2008 3:16 AM<br>
<b>To:</b> freeswitch-users@lists.freeswitch.org<br>
<b>Subject:</b> [Freeswitch-users] newbie dialplan question<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Hi,<br>
<br>
I am working on some test on seeing how I can port my exist Asterisk stuff to
Freeswitch. I am just getting started and I am hoping someone can give me
some help to get started. <br>
<br>
I installed with all the default config and xml setting. Then, I bring
up two SiP clients - one in the same machine as freeswitch (<a
href="http://192.168.1.104">192.168.1.104</a>) and the other one on
another machine ( <a href="http://192.168.1.102">192.168.1.102</a>).<br>
<br>
When I dial an extension ( 1001, or 1002... etc ) from my SIP client on <a
href="http://192.168.1.102">192.168.1.102</a>, I can make the call to the other
client no problem. <br>
2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name() New
Channel sofia/default/<a href="http://1002@192.168.1.104:5060">1002@192.168.1.104:5060</a>
[9e6d6146-0c98-11dd-bb92-f1e303d528b1]<br>
2008-04-18 00:09:12 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing
1002-><span style='color:red'>81001@default</span><br>
2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name() New
Channel sofia/default/<a href="http://1001@192.168.1.104:5061">1001@192.168.1.104:5061</a>
[9e93e564-0c98-11dd-bb92-f1e303d528b1]<br>
2008-04-18 00:09:12 [NOTICE] sofia.c:1603 sofia_handle_sip_i_state() Ring-Ready
sofia/default/<a href="http://1001@192.168.1.104:5061">1001@192.168.1.104:5061</a>!<br>
<br>
However, when I dial extension from the other SIP client, the one on <a
href="http://192.168.1.104">192.168.1.104</a>, the call can't be routed. <br>
<br>
2008-04-18 00:10:34 [NOTICE] switch_channel.c:531 switch_channel_set_name() New
Channel sofia/default/<a href="http://1001@192.168.1.104:5061">1001@192.168.1.104:5061</a>
[cf710b58-0c98-11dd-bb92-f1e303d528b1]<br>
2008-04-18 00:10:34 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing
1001-><span style='color:red'>1002@public</span><br>
2008-04-18 00:10:34 [NOTICE] switch_core_state_machine.c:198
switch_core_standard_on_execute() Hangup sofia/default/<a
href="http://1001@192.168.1.104:5061">1001@192.168.1.104:5061</a> [CS_EXECUTE]
[NORMAL_CLEARING]<br>
2008-04-18 00:10:34 [NOTICE] switch_core_session.c:748
switch_core_session_thread() Session 20 (sofia/default/<a
href="http://1001@192.168.1.104:5061">1001@192.168.1.104:5061</a>) Ended<br>
<br>
It seems like the the call is being routed to the wrong context. How come
this happens? I am using the standard default config xml files. Can
anyone please help me?<br>
<br>
With freeswitch, is there anyway to debug/trace the processing of the call so I
can see which condition it is in, and where it is routed to? That way, I
can debug the config easier? <br>
<br>
Thanks alot for all your inputs and help.<br>
<br>
Regards,<br>
Pete<o:p></o:p></p>
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