[Freeswitch-users] newbie dialplan question

[Pete Kay]

Hi,

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.

I installed with all the default config and xml setting.   Then, I bring up
two SiP clients - one in the same machine as freeswitch (192.168.1.104)  and
the other one on another machine ( 192.168.1.102).

When I dial an extension ( 1001, or 1002... etc ) from my SIP client on
192.168.1.102, I can make the call to the other client no problem.
2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name()
New Channel sofia/default/1002 at 192.168.1.104:5060[9e6d6146-0c98-11dd-bb92-f1e303d528b1]
2008-04-18 00:09:12 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing
1002->81001 at default
2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name()
New Channel sofia/default/1001 at 192.168.1.104:5061[9e93e564-0c98-11dd-bb92-f1e303d528b1]
2008-04-18 00:09:12 [NOTICE] sofia.c:1603 sofia_handle_sip_i_state()
Ring-Ready sofia/default/1001 at 192.168.1.104:5061!

However, when I dial extension from the other SIP client, the one on
192.168.1.104, the call can't be routed.

2008-04-18 00:10:34 [NOTICE] switch_channel.c:531 switch_channel_set_name()
New Channel sofia/default/1001 at 192.168.1.104:5061[cf710b58-0c98-11dd-bb92-f1e303d528b1]
2008-04-18 00:10:34 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing
1001->1002 at public
2008-04-18 00:10:34 [NOTICE] switch_core_state_machine.c:198
switch_core_standard_on_execute() Hangup sofia/default/
1001 at 192.168.1.104:5061 [CS_EXECUTE] [NORMAL_CLEARING]
2008-04-18 00:10:34 [NOTICE] switch_core_session.c:748
switch_core_session_thread() Session 20 (sofia/default/
1001 at 192.168.1.104:5061) Ended

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?

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?

Thanks alot for all your inputs and help.

Regards,
Pete
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