March 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

directory with several xl files
by Ivan C Myrvold 27 Mar '08

27 Mar '08

I am not sure if this is a bug, or if this is how it should behave: If you in conf/directory have two files, say n1.xml and n2.xml, and two directories n1 and n2, and n1.xml includes files from n1 and n2.xml includes files from n2, then registrations are not working properly. Is this correct? Maybe it is supposed to only be one xml file in conf/ directory? Ivan

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Can't follow Getting Started Guide SIP profile instructions
by Erland Lewin 26 Mar '08

26 Mar '08

Hello, I just built Freeswitch from SVN, and am looking at the wiki Getting Started guide documentation: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Default_FreeSWITCH_Co… Some things are unclear to me: "The sofia.conf.xml file includes other xml files to define multiple "profiles"" Looking at the "sofia.conf.xml" file generated by the default install, I see no inclusion of other xml files. "The default profiles are located in: $PREFIX/conf/sip_profiles/PROFILE_NAME.xml -- where PROFILE_NAME is the name of the profile." No sip_profiles directory was generated when I installed FreeSwitch, am I expected to create this myself, and add configuration file lines to include these files from sofia.conf.xml? sofia.conf.xml doesn't seem to reference a sip_profiles directory? I'm new to FreeSwitch, but it seems like the documentation and the default config files don't agree completely, which makes it hard for a noob to get this sorted out. Thanks for any help! /Erland

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SIP trunks with FS
by michael mendel 26 Mar '08

26 Mar '08

Hello I have a question , how I use SIP TRUNKS with FS. Regards, Michael Mendel

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How to start with the windows console demo?
by Ritesh Singh 26 Mar '08

26 Mar '08

Hi, I am very new to the FreeSwitch. Can someone tell me in detail how to start with the console demo of freeswitch on windows. It will be great if I can get some information on how to make test call. My aim is to make a demo in which 2 users talking to each other using freeswitch as a bridge. Also I would like to have all the call details. Any help regarding this is highly appreciated. Thanks and Regards Ritesh

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Answering machine detection
by Aloysius Thevarajah Lloyd 26 Mar '08

26 Mar '08

Hello , Thank you for the answer to my previous question. Is there a way to identify the Answering machine detection in FreeSwitch? Thank you Lloyd

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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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