August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Conferencing performance
by Fernando Testa 12 Aug '08

12 Aug '08

Hi list, I'm impressed by the quality of FS and I'm considering to suggest our company to use it for the conference platform. So, since I don't have currently a way to stress it, the concerns are: Q1: how is the quality under load? Q2 what are the limits of a conference bridge? Usually we get from 200 to 500 people in a system. could it be done? Q3 what about echo cancellation? Is that present on FS or is not an issue? (related to Q.1) Thank you! Fernando Gregianin Testa Voice Technnology Ltda

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DTMF recognition and nbound calls
by Ilan Perez 12 Aug '08

12 Aug '08

Dear All, I have setup my inbound extension (in the public.xml) diaplan to transfer to extension 5000 à the demo IVR The recordings come through nicely. From an internal call the demo fine including DTMF recognition but with inbound calls from external sourceie pstn line Two problems occur. 1. There is no DTMF recognition 2. If I hangup during the playback the system still plays out the whole IVR Here is my code for the extension. Hopefully someone can help me out <extension name="public_did"> <condition field="destination_number" expression="^(0283472006)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=C:/Program Files/FreeSWITCH/sounds/en/us/callie/voicemail/8000/vm-press.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 2008 Diagnostic Devices, Inc."/> <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/> <action application="set" data="RECORD_ARTIST=Ian Curtis"/> <action application="set" data="RECORD_COMMENT=Love will tear us apart"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=false"/> <action application="set" data="playback_terminators=#*"/> <action application="record_session" data="C:/Program Files/FreeSWITCH/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_num ber}_${caller_id_number}.wav"/> <action application="set" data="ringback=${us-ring}"/> <action application="transfer" data="5000 XML default"/>  </condition> </extension> Anyone? Ilan Perez

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AppGen or GUI tool for FS ?
by Marek Gorecki 12 Aug '08

12 Aug '08

Hi FS community, I remember, that GUI issue, and actually many other FS control access issues are of high priority, and seems there is something what might/should help: http://adhearsion.com/ It is not straitforward since it was made with Asterisk in mind, but I believe it shouldn't be big issue to port it to FS too. I just trust in Anthony The Second persuasion and influence to convince Adhearsion authors/creators to make it also FS compatible - if not directly, then maybe they can open some API or other method of integration, and FS community directly, or via bounties, will bring it to reality. Its the first thought, before reading any further about Adhearsion. What are other opinions ? /\/\arekg

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RIngback and sample rate
by Dome Charoenyost 12 Aug '08

12 Aug '08

Dear All, Is posible ro play 32000 bit wav for ringback to gsm codec channel ? I try and got message and shoppy sound 2008-08-11 08:27:54 [DEBUG] switch_ivr_originate.c:1101 switch_ivr_originate() Play Ringback File [/home/ring.wav] 2008-08-11 08:27:54 [WARNING] switch_core_file.c:111 switch_core_perform_file_open() Sample rate doesn't match Dome C.

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ANNOUNCEMENT: Weekly conf call Wednesday 13:00 CST
by Michael Collins 11 Aug '08

11 Aug '08

Here's the page for this week's documentation call. As usual it is currently thin, but I'm sure there are things people can add. We'll be picking a new "mod of the week" so if you have an opinion then please hop on the call. http://wiki.freeswitch.org/wiki/Wiki_meet_2008_08_13 Thanks, Michael

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Cannot create outgoing channel of type [user] cause
by Adeel Ansari 11 Aug '08

11 Aug '08

Hi Guys, I am having this error on Freeswitch console. Any ideas. *[ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL]* I am using Twnikle Phone. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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what does this mean?
by Ilan Perez 11 Aug '08

11 Aug '08

2008-08-11 17:40:53 [DEBUG] sofia.c:194 sofia_event_callback() event [nua_r_options] status [501][Not Implemented] session: n/a What have I not setup properly that this message constantly is displayed ona loglevel 7 mode Ilan Perez

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Having problem in loading Java and Python modules
by Adeel Ansari 11 Aug '08

11 Aug '08

Hello Folks, Having problem in loading Java and Python modules. The problem is mod_java.so and mod_python.so not found. I configured Freeswitch with mod_java enabled. Everything seems fine, until it says these 2 files are not found, while starting up. Any idea? Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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How to run OS commands
by Klaus Teller 11 Aug '08

11 Aug '08

Hi, is there a way to execute OS commands (e.g. audio conversion with ffmpeg) from a diaplan or from javascript script? thanks, Klaus L'Imbecile. -- GMX Kostenlose Spiele: Einfach online spielen und Spaß haben mit Pastry Passion! http://games.entertainment.gmx.net/de/entertainment/games/free/puzzle/61691…

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Fwd: Asterisk and FreeSWITCH | OS-VoIP
by Michael S Collins 10 Aug '08

10 Aug '08

Hey guys, check out the link below. It is a good article by someone who really likes Asterisk but who also has good things to say about FS. Also I'd like someone who knows more about FS conferences to address the issues raised in this blog's first comment. I want to know if what the commenter says is true or not and if it is then perhaps we could investigate the possibility of addressing those issues. Thanks, MC Sent from my iPhone Begin forwarded message: > > > http://www.os-voip.com/2008/08/asterisk-and-freeswitch/

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