February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP server? PBX vs. softswitch?
by Fred 28 Feb '09

28 Feb '09

Hello Even though I successfully set up an Asterisk voice server, I'm no telecom expert, and would like some clarification about the following things: - What is an SIP server as opposed to a IP PBX? - What is the different between a PBX like Asterisk and a softswitch? Thank you.

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freeswitch with celt on windows
by e schmidbauer 28 Feb '09

28 Feb '09

Hi. I was wondering if it is possible to compile freeswitch with the celt codec on windows? I have been able to compile celt for windows but when i compiled freeswitch i did not see the mod_celt in the solution explorer. If this is possible could someone point me in the right direction. Thank you.

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Using OpenZAP + FXO card just to get CID info?
by Fred 28 Feb '09

28 Feb '09

Hello I'd like to write a single-host CRM application, so I need to get the CallerID information when a call comes in. I don't actually need a PBX/softswitch. The user will have the FXO cards and a phoneset connected on the same line, and will pick up the phone once the CRM application has picked up the CID info and popped up a dialog box, etc. Is it possible to just use OpenZAP to get this information, or must I install Freeswitch and provide an IP phone as well? Thank you.

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Use XML dialplan and mod_perl
by Tchavdar Paskov 28 Feb '09

28 Feb '09

Hi, i have a quick question. is it possible to use both XML dial plan and mod_perl together. examlpe: default.xml - used as default context mod_perl - used to generate the public context If it is possible how i have to set perl.conf.xml and especially xml-handler-bindings ? is it possible in value="dialplan" to specify the name of the context? Regards Chav

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Variables from failed call to be exported to a a new B leg
by Tchavdar Paskov 27 Feb '09

27 Feb '09

Hi Everybody, this is what i' trying to do / unsuccessfully / so far: <extension name="test"> <condition field="destination_number" expression="^1(\d+)$" break="never"> <action application="set" data="bypass_media=true" /> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/gateway/gw_1/$1"/> <action application="info" /> -> at this point i'd like to collect some sip Vars from the failed call <action application="export" data="nolocal:sip_hangup_phrase" /> - from what i red in wiki i think this is the way to export the var to the Bleg <action application="bridge" data="sofia/gateway/gw_2/$0" /> </condition> </extension> the gw_2 does not seem to receive the sip_hangup_phrase. pls help me to figure out what i'm doing wrong. thank you in advance. regards Chav

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Thin Client VOIP setup?
by Erik Wickstrom 27 Feb '09

27 Feb '09

Hi, I've deployed Freeswitch as our phone system at work. We now want to use our new phonesystem in a phone room with thin clients (Terminal Server, possibly LTSP) for each agent. Ideally, we'd like to use x-lite or another softphone for each agent. The desired workflow for the agents is as follows: 1) A web based CRM with click to dial. (and customer data card etc) 2) Agent clicks dial button and is connected to customer 3) Interact with CRM... >From what I've read so far, there are some challenges that need to be overcome in deploying softphones over thin clients. Has anyone here had any success in setting up a system like this? I'm I asking for trouble trying to use softphones with thin clients (should I just use hardware phones? Do they support click to dial?) Thanks! Erik

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Console messages
by Nik Middleton 27 Feb '09

27 Feb '09

Hi Guys, Is there a way of displaying a console message not related to a log level? I've got the console only reporting errors now, but it would be nice to be able to display a message when a given condition exists. Yes, I could set it as an error level message, but I'd rather not do that. Regards,

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change session-timer value
by mashudi 27 Feb '09

27 Feb '09

Hi Folks , I have problem to change the session timer value that automatically created by Freeswitch for 120, I can change the value of session-timer in /sip_profiles/internal.xml <param name="session-timeout" value="120"/> become <param name="session-timeout" value="300"/> and it work only for INVITE message response, but for UPDATE message response still use default value of 120, please help me to solved this problem. Thank you in advanced. regards mashudi pls help me to figure out what i'm doing wrong. > ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya *****************************************

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session orignate freeswitch 1.0.3 segmentation error
by Gopalakrishnan A.N 27 Feb '09

27 Feb '09

Hi, I have installed Freeswitch 1.0.3. I am using event socket with Javascript. When I try to dial the script with below command, the call is not going thru it seems to be idle. and segmentation fault core dump error, (freeswitch hangs).....[?] new_session = new Session.originate(session, "sofia/default/<outbound-number>@foo.com"); bridge(session, new_session); I saw in the wiki http://wiki.freeswitch.org/wiki/FreeSwitch_Javascript_Session that the session is depreciated, earlier I was using like this in Freeswitch 1.0.2, it works fine....:) session = new Session(); session.originate(session, "{ignore_early_media=true}sofia/default/<outbound-number>@foo.com"); So something I am missing, please let me know where I am wrong? -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in

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snd_dummy setting for skype
by Henry Huang 27 Feb '09

27 Feb '09

I went through the wiki on mod_skypiax and see there should be a script to make skype work without sound card in linux. Does anyone know where to obtain that script to make sound work "without sound card"? I am currently creating a /etc/asound.conf for skype to load the "fake" sound driver. I do hear sound, but it's not perfect, it's very choppy and it gives me error message when starting skype. The following is my asound.conf setting. Hopefully someone can shed some light : pcm.plugfile{ type plug slave { pcm infile format S16_LE channels 1 rate 16000 } } pcm.infile { type file slave { pcm null } file /dev/dsp infile /dev/dsp } by using this configuration. skype spit out error messages as follow but still works: ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL plugfile ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL infile -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant

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