[Freeswitch-users] newbie dialplan question

Daniel Hefti dhefti at metropark.com
Thu Apr 17 07:57:18 PDT 2008

I have noticed this, too.  I can open up ekiga on the machine FS is installed on, and it can accept calls, but, if I remember correctly, FS doesn't show that the call is being routed, meaning ekiga is just accepting all the calls itself.

What I also found, in relation to this, is that if you have another sip contact outside the FS box, and you try to call them with a sip client also outside the FS box, then the client that is running on the FS box (in my case, Ekiga) will accept the calls that are supposed to be for other users.

So, basically, I don't think you can have a client on the same machine as FS.  I suppose that is to be expected, as a sip client is also a server.  Am I right?

A related problem I have is if you have multiple sip accounts on the same machine (registering from the same ip address), they can't call each other.  It may stem from the problem above, but it may not.

I'm also a quite a newbie in FS, so hopefully we can figure out how to do these things.  I would gladly add the answer to the wiki for others if there's something specific we need to do to do these kind of things.


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 3:16 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] newbie dialplan question


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 (<>)  and the other one on another machine (<>).

When I dial an extension ( 1001, or 1002... etc ) from my SIP client on<>, 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<http://1002@> [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<http://1001@> [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<http://1001@>!

However, when I dial extension from the other SIP client, the one on<>, 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<http://1001@> [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<http://1001@> [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<http://1001@>) 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.

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