- FreeSWITCH-users - FreeSWITCH Mailing Lists

Serbian DID number.
by Cesar Bermudez 02 Jun '10

02 Jun '10

Sorry list, i'know this its no the correct spot to ask this, but i'dont have more places to ask, i'need a serbian did, anyone have any place to buy one? Best regards.

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Robust Affordable Speech Recognition
by Kashif Kahn 02 Jun '10

02 Jun '10

Dear All, All those who have wanted a speech recognition solution for Freeswitch but found the software cost too expensive or the recognition accuracy unsatisfactory, please consider Vestec Speech Engine for Freeswitchat: http://www.vestec.ca/products A starter kit - which is a specially priced one port (ie. one channel) license for the standard engine - is available for only $25. Additional ports (channels) licenses can be purchased for $99/port. The engine comes with a free-of-charge Freeswitch connector, thereby allowing direct interaction via Dialplan. Best regards, -Kashif Kashif Kahn VP, Business Development Vestec, Inc. Waterloo, ON Canada phone: (519) 885-7615

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How to do a "bridging" conference
by Damjan Jovanovic 02 Jun '10

02 Jun '10

Hi I am looking for a way to have a single conference distributed over multiple Freeswitch servers. For example callers 1, 2 and 3 use a conference on Freeswitch server A, and callers 4, 5 and 6 use a conference on Freeswitch server B. These servers mix the local audio individually and only share a single audio channel between them. Is this what a "bridging" conference does? If not, what is it for? Also how could I efficiently make a large listen-only "conference" where there is only ever 1 talker, with very many listeners distributed over lots of servers? Thank you Damjan

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Channel variables difference
by Nagalenoj H. 02 Jun '10

02 Jun '10

Dear friends, I need to know the difference between these two values, Channel-Destination-Number and Caller-Destination-Number. I've found these two in the CHANNEL_DATA event header in event socket(when FS connects to an "Event Socket Outbound" handler). can someone help me please? -- Regards, Nagalenoj H.

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time format
by Tihomir Culjaga 02 Jun '10

02 Jun '10

hello, I need to convert seconds into hh:mm:ss format, any hint ? T.

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does session:read flush digits?
by Neil Patel 01 Jun '10

01 Jun '10

Before lua's session:read returns, does it do session:flushDigits so that the next call won't have the input carried over? The reason I ask is because it seems that after session:streamFile, any dtmf input hangs around for a subsequent session:read call, unless I explicitly invoke session:flushDigits after the streamFile call. Thanks, Neil

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Portaudio: Call For Input
by Mitch Capper 01 Jun '10

01 Jun '10

Recently I have started attempting to replace an existing sip client with freeswitch. In doing so I have started looking more towards portaudio. My goal is to add some robustness to portaudio to make it better suited to being used by sip clients. Some of the features I have currently worked to implement: *Event generation for call holding and resuming *Ability to switch input/output devices during calls with live audio going on (useful for say speakerphone support) *Ability to switch both the indev and outdev at once rather than separately (time savings) *Ability to keep the audio stream initialized rather than each time a call is made/ device is rang (time savings) *Only init codecs during load / config reload rather than on audio stream init (time savings) *Ability to call play (for sounds) with active calls going on Most of the improvements so far have been to make portaudio a bit faster at doing things and less restricted than previously coded. Most changes have been made through additional configuration variables so out of the box portaudio will not function any differently than currently. My request for input is on next steps and warnings. Right now portaudio keeps track of two streams, the call audio stream and the ring audio stream (and both are inited on demand only accept with my always active stream change). I was thinking of taking the always active streams to a slightly higher level: I am thinking about moving to have it keep track of an arbitrary number of streams, a linked list of streams it is keeping track of with no more than one stream per input/output pair. This would allow for initing a device prior to use to allow for near instant use of that stream. This would remove the small delay that still exists (under 1 second currently I would estimate) when say switching on the speaker phone. Or allow for very quick playing of audio on a specific device. Aside from the speed increases in stream switching this would allow for you to play audio on a device that isnt currently the primary audio device. While I do not plan to take it this far currently, it would actually allow for a much easier time of handling multiple calls on different devices at the same time. The other way to go is to look towards trying to speed up initing streams more, there are some yields in the code that look like safety things that may be possible to remove without negative affects. Also I looked at the last merge of portaudio from upstream into trunk (end of 2007) and the changes that were made back then. The good news is there were not actually a lot of base code changes and most of them have actually been merged into upstream now so updating to the latest may not be a very hard thing to do. This is in part to see if it helps with the audio quality issues that people of portaudio seem to report as it certainly is not something you want in your client. If anyone has any input into the current state of portaudio or purposed changes please let me know. In addition if anyone has input into getting better call quality out of PA that would be extremely advantageous, as other than updating to trunk I doubt my changes will result in much of a quality improvement. I am specifically also interested if anyone knows of any of the reasons some of the safeguards that are there are in place, some of which I don't see a technical reason for them to be there unless there is funnyness in libportaudio itself (which certainly could be and may be partially resolved with updating to the changes from the last 3 years). If anyone knows any stress testing etc that presented issues for PA previously that would be helpful. I am working with PA in windows, do have a linux box I will do some limited testing on, but overall if you just can test various changes let me know as that will be helpful too as I believe removing some of the safety steps may not present as an issue right away. I could also add a "faster" option to portaudio config that would result in the optimizations rather than just removing them completely. ~Mitch

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Read and Set the UUI in PRI
by velusamy Krishnan 01 Jun '10

01 Jun '10

Dear All, How do I set and read the UUI(User-User-Information) in FreeSWITCH? Thanks, Velusamy.

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RE FER with Replaces
by blacked 31 May '10

31 May '10

hi everyone well about my last post now I'll be more specific did anyone know how can I make a transfer, a REFER with replaces to repleace FS in the loop of the call and let the two legs with out FS. hope someone can help me -- View this message in context: http://old.nabble.com/REFER-with-Replaces-tp28712663p28712663.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] alias folder
by Madovsky 31 May '10

31 May '10

found the problem apparently FS didn't restarted well. I removed the pid and started again and everything is fine. Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, May 31, 2010 12:50 PM Subject: alias folder Hi, is it a problem to have an alias folder (ex default directory in conf/directory) ? because it seems that reloadxml doesn't work or maybe it's a permission problem... Thanks Franck

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