Again remove 

<anti-action application="set" data="sip_exclude_contact=${network_addr}"/> 

/b

On Apr 17, 2008, at 3:16 AM, Pete Kay wrote:
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@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@default
2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@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@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@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@public
2008-04-18 00:10:34 [NOTICE] switch_core_state_machine.c:198 switch_core_standard_on_execute() Hangup sofia/default/1001@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@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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