- FreeSWITCH-users - FreeSWITCH Mailing Lists

How To - Simple Outbound IVR
by Aloysius Thevarajah Lloyd 26 Mar '08

26 Mar '08

Hello, Can some one guide me how to do the following in Freeswitch.... *simple outbound caller for a small group of people with very simple IVR (press 1 to ack, press 2 to replay). * Thank you Lloyd* *

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How to execute a dialplan extension from a javascript file
by Leonardo Alves 25 Mar '08

25 Mar '08

Is there anyway that I can execute a dialplan extension from a javascript file ? Thanks Leonardo Alves

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Help with ignore_early_media
by Leonardo Alves 24 Mar '08

24 Mar '08

I need some help using the ignore_early_media. I am using a javascript file to dial a number from a asterisk server and play a file to the person who answers the phone. The problem I am having is relatated to the early_media. I think I am doing something wrong. But I cannot discover what it is. here is my .js file: ---------------------------------------------------------------------------------------------------------------------------------------------------- var languageCode = "en"; var soundDir = "sounds/"; function playFile(fileName, callBack, callBackArgs) { new_session.streamFile(soundDir + languageCode + "/" + fileName, callBack, callBackArgs); } new_session = new Session(); new_session.setCallerData("caller_id_name", "6130356060"); new_session.originate(session, "{ignore_early_media=true}sofia/gateway/asterisk/9133803764"); new_session.waitForAnswer(); if(new_session.ready()){ playFile("HelloWorld.wav"); } exit(); ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ Here is the Freeswitch output: jsrun HelloWorld.js API CALL [jsrun(HelloWorld.js)] output: OK 2008-03-23 10:21:08 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/9133803764 [ff67046e-f8db-11dc-ad03-7f3820494c31] freeswitch@debianfs01> 2008-03-23 10:21:08 [NOTICE] switch_channel.c:1380 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/default/9133803764! 2008-03-23 10:21:08 [NOTICE] sofia_glue.c:1760 sofia_glue_tech_media() Pre-Answer sofia/default/9133803764! 2008-03-23 10:21:19 [NOTICE] sofia.c:1848 sofia_handle_sip_i_state() Channel [sofia/default/9133803764] has been answered 2008-03-23 10:22:02 [NOTICE] sofia.c:1946 sofia_handle_sip_i_state() Hangup sofia/default/9133803764 [CS_HOLD] [NORMAL_CLEARING] 2008-03-23 10:22:02 [ERR] HelloWorld.js:7 mod_spidermonkey() TypeError: callBack is not a function 2008-03-23 10:22:02 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 24 (sofia/default/9133803764) Ended 2008-03-23 10:22:02 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/9133803764 [CS_HANGUP] The problem is: If in the originate command I use the {ignore_early_media=true} option. The script to play the file is executed when the call is answered. But the person who answers the phone cannot hear the .wav file. If I do not use the {ignore_early_media=true} option. The script to play the file starts when the call is preAnswered. And when the person in the phone answer the call, can hear the sound of the .wav file, but the audio has already started (when the preanswer is executed). ps: the computer that is making the call is a asterisk server, connected with the FS by sip, and the call is made by zaptel whe pri isdn signalling. I would like to know if I am doing something wrong or this is a bug in the system. Thanks for the help. Leonardo Alves

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Re: [Freeswitch-users] Aastra BLA/SLA
by Tim Ferguson 24 Mar '08

24 Mar '08

My apologies, I meant the status/presence is *not* shown between the phones. P.S. Brian you should be getting that aastra 480i today or tomorrow (if you hadn't caught my message in the channel). ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Sent: Mon Mar 24 06:50:30 2008 Subject: Re: [Freeswitch-users] Aastra BLA/SLA On Mar 21, 2008, at 11:05 AM, Tim Ferguson wrote: When making a call, both phones ring but the status/presence is shown across both phones when one phone picks up. Isn't this what it should do? /b

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Aastra BLA/SLA
by Tim Ferguson 24 Mar '08

24 Mar '08

Hello, I was wondering if anyone has any experiencing using the Aastra phones, specifically the 480i's in the BLA/SLA setup with freeswitch? I currently have two registered with a test box, multiple registrations enabled and managed presence enabled. When making a call, both phones ring but the status/presence is shown across both phones when one phone picks up. I've tried setting the BLA line mode to BLA in the aastra configs as well as setting the BLA number to the extension number the phones are registered as. Am I missing something? Thanks! -Tim

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Help with ignore early media
by Leonardo Alves 23 Mar '08

23 Mar '08

I need some help using the ignore_early_media. I am using a javascript file to dial a number from a asterisk server and play a file to the person who answers the phone. The problem I am having is relatated to the early_media. I think I am doing something wrong. But I cannot discover what it is. here is my .js file: ---------------------------------------------------------------------------------------------------------------------------------------------------- var languageCode = "en"; var soundDir = "sounds/"; function playFile(fileName, callBack, callBackArgs) { new_session.streamFile(soundDir + languageCode + "/" + fileName, callBack, callBackArgs); } new_session = new Session(); new_session.setCallerData("caller_id_name", "6130356060"); new_session.originate(session, "{ignore_early_media=true}sofia/gateway/asterisk/9133803764"); new_session.waitForAnswer(); if(new_session.ready()){ playFile("HelloWorld.wav"); } exit(); ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ Here is the Freeswitch output: jsrun HelloWorld.js API CALL [jsrun(HelloWorld..js)] output: OK 2008-03-23 10:21:08 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/9133803764 [ff67046e-f8db-11dc-ad03-7f3820494c31] freeswitch@debianfs01> 2008-03-23 10:21:08 [NOTICE] switch_channel.c:1380 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/default/9133803764! 2008-03-23 10:21:08 [NOTICE] sofia_glue.c:1760 sofia_glue_tech_media() Pre-Answer sofia/default/9133803764! 2008-03-23 10:21:19 [NOTICE] sofia.c:1848 sofia_handle_sip_i_state() Channel [sofia/default/9133803764] has been answered 2008-03-23 10:22:02 [NOTICE] sofia.c:1946 sofia_handle_sip_i_state() Hangup sofia/default/9133803764 [CS_HOLD] [NORMAL_CLEARING] 2008-03-23 10:22:02 [ERR] HelloWorld.js:7 mod_spidermonkey() TypeError: callBack is not a function 2008-03-23 10:22:02 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 24 (sofia/default/9133803764) Ended 2008-03-23 10:22:02 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/9133803764 [CS_HANGUP] The problem is: If in the originate command I use the {ignore_early_media=true} option. The script to play the file is executed when the call is answered. But the person who answers the phone cannot hear the .wav file. If I do not use the {ignore_early_media=true} option. The script to play the file starts when the call is preAnswered. And when the person in the phone answer the call, can hear the sound of the .wav file, but the audio has already started (when the preanswer is executed). ps: the computer that is making the call is a asterisk server, connected with the FS by sip, and the call is made by zaptel whe pri isdn signalling.. I would like to know if I am doing something wrong or this is a bug in the system. Thanks for the help. Leonardo Alves Devenez un meilleur amigo grâce à Yahoo! Courriel http://cf.promos.yahoo..com/courriel/visiteguidee2/

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not able to login wikipbx.com
by ram 23 Mar '08

23 Mar '08

Hi \ iam not able to loging to wikipbx.com using same user email and password of freeswitch any one suggestion, how can i get in and look the documentation ram

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Re: [Freeswitch-users] Cannot Initialize [[error near line 1621]: missing >]
by Henk Oegema 22 Mar '08

22 Mar '08

Chris, you were NOT wrong. It pointed me exactly to the line where I made the mistake. After correcting it, everything was OK. :) Big lesson (for (yet to be) a total beginner). Thanks Rgds, Henk >Guys correct me if I am wrong here but, another tip for resolving this >problem can by looking in the file: > /usr/local/freeswitch/log/freeswitch.xml.fsxml >(this is the compiled version of all the files, this may contain your >error... on the line mentioned) > >Cheers, >Chris Henk Oegema wrote: >> My system is Kubuntu 7.10 >> I downloaded an compiled (without errors) freeswitch-1.0.rc1 >> >> (There was an earlier version installed before I compiled this one. >> There I got the same error as what I got now. That was actually the reason >>for >> the new install. but the error remains, also after the new install) >> > >When I want to start FS I get following error:

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Cannot Initialize [[error near line 1621]: missing >]
by Henk Oegema 21 Mar '08

21 Mar '08

My system is Kubuntu 7.10 I downloaded an compiled (without errors) freeswitch-1.0.rc1 (There was an earlier version installed before I compiled this one. There I got the same error as what I got now. That was actually the reason for the new install. but the error remains, also after the new install) When I want to start FS I get following error: henkoegema@kubuntu:/usr/local/freeswitch/bin$ sudo ./freeswitch Error including /usr/local/freeswitch/conf/dialplan/extensions/*.xml Cannot Initialize [[error near line 1621]: missing >] <------------ :( In which file does the error occur ? Rgds Henk

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new socket module
by Jonas Gauffin 21 Mar '08

21 Mar '08

Hello, I'm trying to do a new socket module and I got a problem that is driving me mad. It's probably a really stupid thing. I got a (tcp) server that the module connects to. Then it sends a "hello" message that the server responds to. The module reads the response OK, but after that everything fails with error 730038 ("An operation was attempted on something that is not a socket."). I can't understand why, the socket descriptor is the same all the time. Got any general hints on what might cause it? 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:859 ivr_socket() Connecting to: localhost:8123. 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:639 socket_send_event() send 'event;session;voicemail;sofia/default/jonas@192.168.0.58:5060;acf42337-a1ad-f84e-af01-4b7c633840dc;XML;CS_EXECUTE;jonas;jonas gaff;192.168.0.58;;;8331; ', len 152 on 21485136 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:641 socket_send_event() Bytes sent 152 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:708 socket_read() Reading, buffersize 4195 bytes from 21485136. 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:713 socket_read() Bytes received 15. 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:775 socket_loop() Read: action;answer; 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:708 socket_read() Reading, buffersize 4195 bytes from 21485136. 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:713 socket_read() Bytes received 0. 2008-03-21 09:23:59 [ERR] mod_ivr_socket.c:717 socket_read() socket receive failed: 730038. //Jonas

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