[Freeswitch-users] newbie dialplan question

[Pete Kay]

Hi Brian,

What is the difference between $$ and $ in FS?

Thanks,
Pete

On Thu, Apr 17, 2008 at 11:11 PM, Brian West <brian at freeswitch.org> wrote:

> 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
> > No virus found in this incoming message.
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> > Apr-08 5:34 PM
> >
> >
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> >
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