[Freeswitch-users] newbie dialplan question

[Brian West]

You could also use user/${dialed_ext}@$${domain}

in addition you could do sofia/$${domain}/${dialed_ext}

/b



On Apr 17, 2008, at 9:45 AM, UV wrote:
> I’m not sure which build you’re using, but I had that problem on sub  
> version 7946 (RC1).
>
> There’s a “bug” in the conf/dialplan/default.xml example where under  
> the “Local_Extension” section is says:
>             <anti-action application="bridge" data="sofia/default/$ 
> {dialed_ext}@$${domain}"/>
> Where it should actually be:
>             <anti-action application="bridge" data="sofia/default/$ 
> {dialed_ext}%$${domain}"/>
>
> Check http://wiki.freeswitch.org/wiki/Sofia#Call_a_locally_registered_endpoint
>
> Cheers,
> UV
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Pete Kay
> Sent: Thursday, April 17, 2008 6:16 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] newbie dialplan question
>
> 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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