$$ is expanded at config parse/load and $ is expanded every time its encountered. In the example of sofia the domain usually won't change every call where the extension will. /b On Apr 17, 2008, at 10:16 AM, Pete Kay wrote:
Hi Brian,
What is the difference between $$ and $ in FS?
Thanks, Pete
On Thu, Apr 17, 2008 at 11:11 PM, Brian West <brian@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Pete Kay Sent: Thursday, April 17, 2008 6:16 PM To: freeswitch-users@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@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 No virus found in this incoming message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.23.0/1382 - Release Date: 16- Apr-08 5:34 PM
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