July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS Support for LinuxMCE
by EdPimentl 07 Jul '08

07 Jul '08

LinuxMCE now supports *sterisk ..... however it would be better with FS. LinuxMCE uses an online(avatar) person to walk one through the configuration of asterisk voip in their home. http://video.google.com/videoplay?docid=2176025602905109829&hl=en LinuxMCE is installed on the Kubuntu OS, and utilizes open source applications such as Asterisk, Xine, MythTV, VDR, Firefox, VideoLAN and SlimServer. OS: Kubuntu TV/PVR: MythTV VoIP : Asterisk (Better alternative would be FS) HomeAutomation: Pluto Serveillance Camera Recording: Motion Is anyone currently working on FS support for LinuxMCE..?

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Time out issue and static noise with originate command
by Diego Viola 07 Jul '08

07 Jul '08

Hi all... I have a user account on FS, sometimes when I register with Zoiper or X-Lite it works, but after some time it unregisters itself, I'm not sure why... this didn't happened on Asterisk for me. Does FS handles NAT automatically? This is how I have it setup: softphone => NAT (router) => FS In Asterisk I used to have nat=yes in sip.conf, is there a equivalent for this in FS? Why could my softphone be deregistering itself? This is my user account on FS: Maitei:/usr/local/freeswitch# cat conf/directory/default/diego.xml <include> <user id="diego" mailbox="1000"> <params> <param name="password" value="1234"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> </variables> </user> </include> Another issue... when I use the originate command, with a WAV or GSM file... all I get is static noise, is there a way that this could work?, this also works on Asterisk just fine for me: originate sofia/internal/diego &playback('/root/nin.mp3.wav') But I get static noise on FS. Thanks, Diego

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Rerouting bridge based on error response from far end
by Joseph Bajin 07 Jul '08

07 Jul '08

I will use xml_curl to query a db for the cheapest routes to use. It will come back with a couple of different gateways to use. (LCR idea) Based on the response I get from the first gateway (404,503,500, 486), I want to either reroute to another gateway or send back to the originator the result that I found (486 in this case, other messages should be rerouted unless we hit the last and then send back a final response). How can I do that in a dialplan? Can you point me some documention maybe?

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help with: [ERR] mod_local_stream.c:266 local_stream_file_open() unknown source moh
by Diego Viola 07 Jul '08

07 Jul '08

Hi When I dial a conference I get this: 2008-07-07 00:19:45 [ERR] mod_local_stream.c:266 local_stream_file_open() unknown source moh I think I have all the MOH files: http://pastebin.com/m6a463dfa http://pastebin.com/m7f24dce Maitei:~# grep hold_music /usr/local/freeswitch/conf/vars.xml <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/> Sorry if I'm missing something... I'm still very new in FreeSWITCH, but I'm really enjoying the experience. Thanks, Diego

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Invalid Application bridge
by Grey Man 06 Jul '08

06 Jul '08

Hi, I've just installed FreeSwitch on Windows XP and have been able to get a SIP account registered and placing calls to the FreeSwitch server. The problem I'm having is when I try and place an outgoing call I get an error message of: "switch_core_sessions.c:964 switch_core_session_execute_application() Invalid Application bridge" The dial plan I am using is: <include> <context name="default"> <extension name="test"> <condition field="destination number" expression="^.*$"> <action application="bridge" data="sofia/sip/300(a)sip.mytestserver.com" /> </condition> </extension> </context> </include> It seems as if the "bridge" application cannot be found. Is there anything I can check to see why that may be so? Regards, Greyman.

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IVR XML and phrase macros all in one single file
by Diego Viola 06 Jul '08

06 Jul '08

Hi, I been playing with the ivr.conf.xml in FS, is there a way to put the phrase macros inside the ivr.conf.xml, the reason for this is that I would like to have all my ivr config in one single file. Something like this: http://pastie.org/228307 Thanks, Diego

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Using Mod_fifo
by Faraz R. Khan 05 Jul '08

05 Jul '08

Is there any example on how to use mod_fifo? I am trying to implement a call centre queue as follows (much like Asterisk queues) : Inbound call-> press 0 for operator -> mod_fifo -> 3 agents of whom any one can get the call (doing round robin or whatever) I checked out: http://wiki.freeswitch.org/wiki/Mod_fifo I understand how to park the call, I understand how to retrieve the call from the fifo- but I dont understand how this will happen automatically. Is the expected way to write some kind of JS to run periodically, check if any of the 3 specified agents are free and send them (originate) to the fifo 'pop' extension? Pointers would be appreciated! If anyone has sample JS to do something close to the above it would give me a great headstart. Thanks! -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz

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core dump on shutdown in 1.0 trunk build 8899
by Alex Kinch 04 Jul '08

04 Jul '08

Hi guys, Happy Independence Day and all that. Having some probs in 8899 which I hadn't seen in previous builds - when running with console and shutting down I get a stack of error messages that certain modules don't have a shutdown routine. For example: 2008-07-05 19:51:04 [CONSOLE] switch_loadable_module.c:1105 do_shutdown() mod_say_en has no shutdown routine 2008-07-05 19:51:04 [CONSOLE] switch_loadable_module.c:1105 do_shutdown() mod_tone_stream has no shutdown routine 2008-07-05 19:51:04 [CONSOLE] switch_loadable_module.c:1105 do_shutdown() mod_local_stream has no shutdown routine It eventually gets to the end and core dumps. Full log is on the pastebin - http://pastebin.freeswitch.org/4747 It doesn't appear to be causing any particular grief right now but thought I'd flag it. Alex

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Bridge to Conference and Outbound calling
by Richard Open Source 04 Jul '08

04 Jul '08

Hi, I am interested in being able to convert a bridged call into a conference and from within the conference, invite others into the conference. >From http://wiki.freeswitch.org/wiki/Specsheet, it seems that mod_conference delivers these out of the box (i.e Bridge to Conference trasition and Outbound Calling). However, I can't get it to work. Here is what I have so far: 1. From XLite, I dial out to an external number thru my Asterisk GW. 2. Set hangup_after_bridge to false as mentioned here http://wiki.freeswitch.org/wiki/Mod_commands#uuid_transfer 3. When the caller (XLite) press *1, transfer both legs to a conference room. 4. Can't make outbound calls from conference (yet). Result: 1. The bridging works. But when I try to press *1 to transfer the call to a conference, it does not work. The log I get is below. However, when I type the command into the console freeswitch@dev-vpbx> uuid_transfer 95bd9744-8c27-4b4b-a785-b915d6640a0b -both adhoc_conf it works. == Pressing *1 log == 2008-07-04 15:01:36 [DEBUG] switch_core_session.c:386 switch_core_session_receive_message() Kill sofia/internal/1012(a)10.98.7.132[BREAK] 2008-07-04 15:01:36 [NOTICE] sofia.c:2015 nua() Channel [sofia/internal/ 1012(a)10.98.7.132] has been answered 2008-07-04 15:01:36 [DEBUG] switch_core_session.c:386 switch_core_session_receive_message() Kill sofia/internal/1012(a)10.98.7.132[BREAK] 2008-07-04 15:01:36 [DEBUG] sofia.c:1701 nua() Channel sofia/internal/1012(a)10.98.7.132 entering state [completed] 2008-07-04 15:01:36 [DEBUG] sofia.c:1701 nua() Channel sofia/internal/1012(a)10.98.7.132 entering state [ready] 2008-07-04 15:01:40 [DEBUG] switch_ivr_async.c:1387 meta_on_dtmf() sofia/internal/1012(a)10.98.7.132 Processing meta digit '1' [uuid_transfer::95bd9744-8c27-4b4b-a785-b915d6640a0b -both adhoc_conf] 2008-07-04 15:01:40 [DEBUG] switch_core_session.c:567 switch_core_session_queue_private_event() Kill sofia/internal/1012(a)10.98.7.132 [BREAK] 2008-07-04 15:01:40 [DEBUG] switch_core_session.c:386 switch_core_session_receive_message() Kill sofia/internal/1012(a)10.98.7.132[BREAK] 2008-07-04 15:01:40 [DEBUG] switch_core_session.c:386 switch_core_session_receive_message() Kill sofia/internal/1012(a)10.98.7.132[BREAK] 2008-07-04 15:01:40 [DEBUG] switch_ivr_bridge.c:168 audio_bridge_thread() Kill sofia/external/916135551234(a)10.98.7.202 [BREAK] == Outbound Calling: I create a caller-controls group to be able to dial out based on intructions here http://wiki.freeswitch.org/wiki/Mod_conference (see caller-control section). However the, <control action="dial" digits="888" ... is not a recognized control. Any ideas on how I can proceed? Below is the relevant section in my dialplan. One more thing, what to these features mean from the Conferencing application (http://wiki.freeswitch.org/wiki/Specsheet)? Can someone give a scenario where these can be used? 1. Relationships 2. Transfers 3. Outbound Calling 4. Configurable Key Lay -- Is this the caller-controls? 5. Bridge to Conference trasition 6. Multi Party outbound dialing. Thank you in advance. Looking forward to playing around some more with FS. Thanks for the wonderful software. R === dialplan default.xml == <extension name="ast_extens"> <condition field="destination_number" expression="^(9\d{11})$"> <action application="set" data="hangup_after_bridge=false"/> <action application="bind_meta_app" data="1 a s uuid_transfer::${uuid} -both adhoc_conf"/> <action application="bridge" data=" sofia/external/$1(a)10.98.7.202"/ <sofia/external/$1(a)10.98.7.202%22/>> </condition> </extension> <extension name="adhoc_conf"> <condition field="destination_number" expression="^adhoc_conf$"> <action application="answer"/> <action application="conference" data="${uuid}@adhoc"/<$%7Buuid%7D@adhoc%22/> > </condition> </extension> ===

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originate dialplan numbers from JS
by Ghulam Mustafa 04 Jul '08

04 Jul '08

i am trying to write a freeswitch javascript app for the first time, it's easy to call a sip endpoint using session.originate, but i don't understand how to originate a dialplan number, e.g. 5900, examples in wiki are just referencing how to originate call to a sofia profile. Regards, Mustafa

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