[Freeswitch-users] newbie dialplan question

[UV]

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 HYPERLINK
"http://wiki.freeswitch.org/wiki/Sofia#Call_a_locally_registered_endpoint"ht
tp://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 (HYPERLINK
"http://192.168.1.104"192.168.1.104)  and the other one on another machine (
HYPERLINK "http://192.168.1.102"192.168.1.102).

When I dial an extension ( 1001, or 1002... etc ) from my SIP client on
HYPERLINK "http://192.168.1.102"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/HYPERLINK
"http://1002@192.168.1.104:5060"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 at default
2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name()
New Channel sofia/default/HYPERLINK
"http://1001@192.168.1.104:5061"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/HYPERLINK
"http://1001@192.168.1.104:5061"1001@192.168.1.104:5061!

However, when I dial extension from the other SIP client, the one on
HYPERLINK "http://192.168.1.104"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/HYPERLINK
"http://1001@192.168.1.104:5061"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 at public
2008-04-18 00:10:34 [NOTICE] switch_core_state_machine.c:198
switch_core_standard_on_execute() Hangup sofia/default/HYPERLINK
"http://1001@192.168.1.104:5061"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/HYPERLINK
"http://1001@192.168.1.104:5061"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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