- FreeSWITCH-dev - FreeSWITCH Mailing Lists

FSComm PortAudio error
by Paul Li 30 Jan '10

30 Jan '10

Most likely, your Vista 64-bit machine has a generic Vista 64-bit (High Definition) audio driver, instead of a specific audio driver for your machine's sound card. It seems PortAudio does not like the generic audio driver from Microsoft, even though you can play audio/video fine with Windows media player or Real Player. This is from my experience. I installed Windows 7 on my old PC. At first, PortAudio failed to pick up sound device before I installed proper audio driver. When I finally installed proper audio driver, PortAudio works perfectly. Good luck to find a proper audio driver to your OS!

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Seeking Advice on SIP Phones like Aastra 6739i or Snom 870 that have good interoperability with FreeSWITCH
by Darren C. 30 Jan '10

30 Jan '10

Hello, I'm working on a project and are interested in using one of the newer SIP Phones with color displays, perhaps touchscreen, etc. We want to send some information about a call to the SIP Phone either via FS or our own Web Service. These two phones have XML browsers that we may be able to utilize. I prefer to use phones that work well with FS but these are too new to show up on interoperability page. Anyone have experience using these, any advise would be appreciated: Aastra 6739i http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-EAA51F83/03/6739i_pd… http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-EAA51F83/03/6739i_pd… http://www.voipsupply.com/aastra-6739i SNOM 870 http://www.snom.com/uploads/docu/data_snom870_en.pdf http://www.snom.com/products/ip-phones/snom-870-touchscreen-voip-phone/ http://www.voipsupply.com/snom-870 Thanks, teldev

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FSComm PortAudio error
by vip killa 29 Jan '10

29 Jan '10

Hi, when i run FSComm on Vista 64-bit. I get an error message saying Error querying audio devices when i click on File -> Preferences. Then under PortAudio (in preferences) there are no device. Is there a specific OS that FSComm will work on?

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Question about switch_cond_next()
by Alex To 29 Jan '10

29 Jan '10

Hi, would someone clarify this for me please Is it right to say that switch_cond_next() does only one thing which is to put the current thread to sleep for a period of time (1 second) ? Is it equivalent to, let's say, Thread.Sleep(xxx); or it also does something else besides putting the current thead to sleep ? Is switch_cond_yield the same as switch_cond_next, except that switch_cond_yield accepts parameter as the input interval ? Thanks Regards Alex To

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FreeSWITCH Conference Call Agenda
by Michael Collins 29 Jan '10

29 Jan '10

Greetings, This week's conference call agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_29 Please add your items as the agenda is very light this week. We do have a few things to discuss, though, so please hop on and bring a friend! Talk to you all soon, Michael

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.deb sound installation files
by Mark Sobkow 28 Jan '10

28 Jan '10

It looks like the sound files are no longer part of the standard build, as per emails that were sent out some time ago. Is there any word on .deb support for building the sound files? Our install process requires that we do .deb files for ubuntu rather than doing a source build. -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com

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fscomm Call failed with reason NO_ROUTE_DESTINATION.
by vip killa 27 Jan '10

27 Jan '10

Hi, I've downloaded and installed the latest FSComm for windows. I am able to register to the freeswitch server but i cannot make a call. When i try an extension i know works (because i use it with ekiga) i get "Call with (####) failed with reason NO_ROUTE_DESTINATION". I've tried calling ####(a)server.ip. But nothing works. I can receive calls to the FSComm application. I'm running FSComm on windows xp 64.

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FreeSWITCH 1.0.5 and Buying Dinner For The Dev Team
by Michael Collins 26 Jan '10

26 Jan '10

Hello all, The FreeSWITCH development team is planning to meet in one place during the week of February 8 to release version 1.0.5! We would like to invite everyone to show their appreciation by donating a few dollars to help pay for dinner for the development team. More information is available here: http://www.freeswitch.org/node/234 Let's all show the team how much we appreciate them by giving them a well-deserved dinner! -Michael

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Estimated release date for 1.0.5 final
by Roland Zagler 26 Jan '10

26 Jan '10

Hi everyone, does anyone already have an idea on when 1.0.5 final will be released? Or at least a timeframe... Thanks, Roland Zagler

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No sound with Gizmo5 after transfer from a dial plan
by Paul Li 24 Jan '10

24 Jan '10

It is a very interesting and challenging issue! What I am trying to achieve is playback a message (notice) to PSTN telephones with Gizmo5 as service provider with the originate command as follows: $ originate sofia/gateway/gizmo/0019876543210 5001 (Case 1) $ originate sofia/gateway/gizmo/0019876543210 5002 (Case 2) I can hear the playback from extension 5001 in the first case, while I could not hear anything from extension 5002 in the second case. The difference between them lies in that extension 5002 transfer the call when the PSTN phone is connected. Please see the dial plan below. Both dial plans are basically adapted from examples found in the online documents. Both 5001 and 5002 work for my two other voip service providers. On the surface, it appears to be Gizmo5's problem. But, I am wondering if there is any issue on the FreeSwitch side. My main goal is to delay playback until the end user has picked up the phone. I have been looking for a way to do this but ended up with nothing so far. I have tried setting {ignore_early_media=true}, but that did not help either. I am welcome any suggestion on how to achieve this. In the meanwhile, I would like point out this issue.  <extension name="direct_ivr_demo1"> <condition field="destination_number" expression="^5001$"> <action application="start_dtmf" /> <action application="playback" data="test.wav"/> </condition> </extension>   <extension name="indirect_ivr_demo2"> <condition field="destination_number" expression="^5002$"> <action application="set" data="execute_on_answer=transfer ANSWERCALL XML default"/> <action application="sleep" data="60000"/> <action application="transfer" data="NOANS XML default"/> </condition> </extension>  <extension name="Call_Answered"> <condition field="destination_number" expression="^ANSWERCALL$"> <action application="start_dtmf" /> <action application="playback" data="test.wav"/> </condition> </extension>  <extension name="Call_Not_Answered"> <condition field="destination_number" expression="^NOANS$"> <action application="hangup" data="NO_USER_RESPONSE"/> </condition> </extension>

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