January 2010 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] When is 1.0.5 due out?
by Mark Sobkow 02 Mar '10

02 Mar '10

I've been advised that the Ubuntu/.deb packaging won't be getting updated until the 1.0.5 release (1.0.4 slipped through the cracks.) I need to know when 1.0.5 is scheduled for release, as we can't ship source builds to our customers, only .deb installations. (The boxes our customers run don't have space on them for doing source builds, and we don't have compilers and such installed on them by default.) Alternatively, if someone could point me to a repo with control files for 1.0.3 or 1.0.4, we could do our own source builds and packaging. I've no fear of tweaking and hacking control files; I just haven't done .deb builds before and starting off from scratch with a project as big as Freeswitch is a bit overwhelming (especially as a "proper" packaging needs to split out the mods, sound packagings, etc.)

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Re: [Freeswitch-dev] Openzap and intermitent ringing
by Moises Silva 25 Feb '10

25 Feb '10

Hi again François, Please keep the conversation in the freeswitch-dev mailing list, unless you want private paid support ;-) See the answer to your question below ... On Wed, Jan 20, 2010 at 3:57 AM, François Legal <devel(a)thom.fr.eu.org>wrote: > Do you know if there is a way (except using an oscilloscope to sample the > line signal) to get a dump of the data sent to the card by openzap (is it > samples or something) and vice versa. > > > > What I would like to do is see what is returned to openzap when a call > arrives on an FXO port (CLIP is working in that direction) and compare it to > what is being sent to the card when sending CLIP to and FXS port. > Once the call gets into the computer we're talking about digital audio samples. In a typical configuration a Sangoma board will give you 160 samples of audio each 20ms (most likely encoded in ulaw, that can be verified in /etc/wanpipe/wanpipex.conf for the FXO/FXS board, look at the TDMV_LAW parameter). There is an openzap API command to record the digital audio read from the board, check ZAP_COMMAND_TRACE_INPUT and ZAP_COMMAND_TRACE_OUTPUT. What we currently lack is a FreeSWITCH command in mod_openzap to enable/disable the audio tracing. It should be trivial to hack mod_openzap.c and add that command, if you need help with that ping me on IRC (moy). -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy(a)sangoma.com

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[Freeswitch-dev] Call for help on FSComm
by João Mesquita 08 Feb '10

08 Feb '10

FreeSWITCH™ers, I believe that everyone already heard about FSComm by now. Development is moving fast and features are being added very fast. Nonetheless, we need help to get this software the way we want. If you are a GUI Designer or have any design skills and want to contribute, please, get in touch with me by email, IM, IRC on #fscomm or even pigeon if you like. I believe everyone wants a sexy looking softphone with all the features FreeSWITCH™ is able to provide. Regards, João Mesquita FSComm Developer GTalk: jmesquita(a)gmail.com PayPal: jmesquita(a)gmail.com IRC: jmesquita on #fscomm @freenode

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[Freeswitch-dev] FSComm script folder
by Paul Li 02 Feb '10

02 Feb '10

While FreeSWITCH set switch_directories.script_dir to "scripts", FSComm set switch_directories.script_dir to "script". Shouldn't it be "scripts" to be consistent?

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[Freeswitch-dev] FSComm capable of multiple lines?
by Paul Li 02 Feb '10

02 Feb '10

With FSComm, may you put a call on hold, while taking or making another call? How about multiple lines? Say, you have more than two incoming lines, you take one connection, while leaving the rest on MOH (Music on Hold). I think this is achievable when FreeSwitch runs as standalone. Thanks a lot for your attention.

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[Freeswitch-dev] getting out of proxy media
by Paweł Pierścionek 30 Jan '10

30 Jan '10

Hi, I have a stupid case with proxy media. (I need it so T.38 works, bypass is not an option). I set proxy_media to true, do a simple bridge to a sip destination and then, once it fails, I need to play a prompt that it failed. The bridge fails with SIP cause 404 after a second worth of G.729 early media (can't ignore the early media) Tried: a) pre_answer + playback b) answer + playback c) redirect to a sip destination with a proper prompt d) deflect to a sip destination with a proper prompt e) bridge to a sip destination with a proper prompt while forcing PCMA codec A & B fail with Codec PROXY PASS-THROUGH encoder error! C - hangs the channel indefinitely D - hangups the channel without sending SIP REDIRECT (call is never fully answered) E - fails with Codec PROXY PASS-THROUGH encoder error! Any ideas how can I enjoy proxy_media and yet be able to play anything to the user if the bridge setup fails ? Bounty? Pawel,

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[Freeswitch-dev] ClueCon MMX - Save the Date!
by Michael Collins 30 Jan '10

30 Jan '10

ClueCon MMX (2010) will be here before you know it! Please mark your calendars: August 3-5, 2010. Start talking up ClueCon with your peers, coworkers, business owners, CEOs, potential sponsors, and anyone else you can think of. It's coming fast, so start getting ready now. Lots more information will be coming soon. Looking forward to seeing everyone this August! -ClueCon Team

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[Freeswitch-dev] 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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[Freeswitch-dev] 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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[Freeswitch-dev] 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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