I highly recommend that you set aside this endeavor for the time being and use the default configuration. Once you get familiar with the default config then you'll realize how to make changes to registered users and to the dialplan. Don't let the size of the default configuration scare you off. It is very well designed, and much of it is compartmentalized, which means you can changes in a single file without affecting the rest of the configuration. Now for the usual questions: What platform are you on? Linux? Did you use SVN? (We highly recommend using SVN) Have you seen the wiki pages on installing FS? If you're running on Linux then I recommend a clean install using the method documented here: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install Let us know how it goes. -MC On Mon, Apr 13, 2009 at 9:56 AM, Martin Fiala <fialkam@gmail.com> wrote:
Hello.
I'm trying to use freeswitch, was able to compile it without problems, which is very nice. Then studying the configurations etc., I managed to set up SIP accounts those register properly. But now, if I want to call one registered account from the other one, I get error 404 - not found. I tried to set up a minimalistic dialplan using xml syntax as well as asterisk syntax but neither worked for me. I changed just a few thing, I'll list them later. I'm trying to make calls using ip addresses and ports instead of domain names..
This is the error freeswitch outputs: 2009-04-13 18:35:48 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/02@192.168.2.100 [19bad83a-ec9a-4b59-8457-cd76f1eaef65] 2009-04-13 18:35:48 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 02->01 in context default 2009-04-13 18:35:48 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/02@192.168.2.100 to enum[01@default] 2009-04-13 18:35:50 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting 2009-04-13 18:35:50 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup sofia/internal/02@192.168.2.100 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 1 (sofia/internal/02@192.168.2.100) Ended 2009-04-13 18:35:50 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/02@192.168.2.100 [CS_HANGUP]
My users are added in file users.xml in directory/ : <include> <user id="01" mailbox="01"> <params> <param name="password" value="01"/> <param name="vm-password" value="01"/> </params> <variables> <variable name="accountcode" value="01"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="01"/> <variable name="effective_caller_id_number" value="01"/> </variables> </user> <user id="02" mailbox="02"> <params> <param name="password" value="02"/> <param name="vm-password" value="02"/> </params> <variables> <variable name="accountcode" value="02"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="02"/> <variable name="effective_caller_id_number" value="02"/> </variables> </user> </include>
I've added the file dialplan/default/000_default.xml with contents: <extension name="internal"> <condition field="source" expression="mod_sofia" /> <condition field="destination_number" expression="^(4\d+)"> <action application="bridge" data="sofia/internal/$0@192.168.2.100:5060" /> </condition> </extension> That's from sample configs, I wonder, if the IP address can be used like that. I understand it that way, the ip address specified is of registrar server. I've added the port as I'm testing it on local loop and thus am running different sip services on the same ip (freeswitch and calling softfones). Is that ok?
extensions.conf I've tried to use: [default]
; Things you're used to.... ;exten => music,n,Dial(SIP/1234@conference.freeswitch.org|120)
;exten => _1XXXXX,n,set(cool=${EXTEN}) ;exten => _1XXXXX,n,set(myvar=true) ;exten => _1XXXXX,n,Goto(default|music) ;exten => 2137991400/1000,n,Goto(default|music)
; Some new magic you can do.... ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,enum($1) ;exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,bridge(${enum_auto_route})
; instead of exten, put anything about the call you would rather match on. ; either the names of a field in caller_profile or a string of variables to expand. ;caller_id_number => 2137991400,n,Goto(default|music) ;${sip_from_user} => bill,n,Goto(default|music)
[pbx] exten => 01,1,Dial(SIP/01,20) exten => 02,1,Dial(SIP/02,20)
When using extensions.conf I've changed this line in sip_profiles/internal.xml from: <param name="dialplan" value="XML"/> to <param name="dialplan" value="asterisk,XML"/> I didn't make any other changes in that file.
I didn't change anything else.
I'm trying to use two sip phones - one using port 6001 (user "01") and the other one 5000 (user "02"). After registration succeeds, calling this sip uri : sip:01@192.168.2.100:5060, where 192.168.2.100:5060 is IP:PORT of freeswitch (the IP is same for softphones.. the same machine).
Thanks for any help. Fiala
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