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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I’m not sure which build you’re
using, but I had that problem on sub version 7946 (RC1).<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>There’s a “bug” in the conf/dialplan/default.xml
example where under the “Local_Extension” section is says:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'> <anti-action
application="bridge" data="sofia/default/${dialed_ext}</span></font><b><font
size=2 color=red face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:red;font-weight:bold'>@</span></font></b><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>$${domain}"/><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Where it should actually be:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'> <anti-action
application="bridge" data="sofia/default/${dialed_ext}</span></font><b><font
size=2 color=red face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:red;font-weight:bold'>%</span></font></b><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>$${domain}"/><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Check <a
href="http://wiki.freeswitch.org/wiki/Sofia#Call_a_locally_registered_endpoint">http://wiki.freeswitch.org/wiki/Sofia#Call_a_locally_registered_endpoint</a>
<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Cheers,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>UV<o:p></o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] <b><span
style='font-weight:bold'>On Behalf Of </span></b>Pete Kay<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, April 17, 2008
6:16 PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">freeswitch-users@lists.freeswitch.org</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> [Freeswitch-users] newbie
dialplan question</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>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-><font color=red><span style='color:red'>81001@default</span></font><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-><font color=red><span style='color:red'>1002@public</span></font><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></span></font></p>
<p><font size=2 face="Times New Roman"><span style='font-size:10.0pt'>No virus
found in this incoming message.<br>
Checked by AVG.<br>
Version: 7.5.519 / Virus Database: 269.23.0/1382 - Release Date: 16-Apr-08 5:34
PM</span></font><o:p></o:p></p>
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<P><FONT SIZE=2>No virus found in this outgoing message.<BR>
Checked by AVG.<BR>
Version: 7.5.519 / Virus Database: 269.23.0/1382 - Release Date: 16-Apr-08 5:34 PM<BR>
</FONT> </P>