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...d in the README file: https://raw.github.com/zaf/asterisk-speech-recog/master/README The script is available here: https://github.com/zaf/asterisk-speech-recog The code is still young and not roughly tested so comments, suggestions and bug reports are more than welcome. ---------------- Lefteris Zafiris

...tts-0.2.tar.gz To get a sample of the speech synthesis quality try this link: http://translate.google.com/translate_tts?tl=en&q=this+is+a+test+for+google+text+to+speech+engine The code is still very young so suggestions, comments and bug reports are more than welcome. -------------- Lefteris Zafiris -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111201/6ec87862/attachment-0001.htm>

Hi Asterisk guys, Flite is not working with asterisk 1.6.2.17. Flite is working with asterisk 1.4. Please help me how to use it with asterisk 1.6 ....... Thanks in advance. ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I recently discovered http://zaf.github.com/asterisk-mstts/ . In the page above, it is mentioned you have to subscribe to Microsoft Translator API on Azure Marketplace. In Azure Marketplace, I found something called "Microsoft Translator". This API is free within a 2 000 000 characters per onth limit. Is this the API needed for MS TTS ? If not, where and how can I find the good

For reasons best known to myself, I call a python agi (PYST2 - love it!) which streams a series of very short files in quick succession. Like this: escape_digits = str("0") agi.stream_file(promptFile,escape_digits) and this is what I see on the AGI debug: <Local/s at root-00000061;2>AGI Tx >> 200 result=0 endpos=6784 <Local/s at root-00000061;2>AGI Rx <<

I have a similar situation here, I want to include TTS in my asterisk IVR system. Could someone give suggestion(s) please, I prefer open-source thanks in advance! Chatila, A. C. P. O. Box 365, Kihesa Street, Njombe, Tanzania. *Mob:* +255 765 154 235 *Whatsapp:* +255 653 258 608 *Website:* chax.me.tz On Thu, Aug 27, 2015 at 9:07 PM, Steve Edwards <asterisk.org at sedwards.com> wrote:

...ly request based routing, HA/failover, load balancing and TLS encryption for FastAGI sessions (something asterisk unfortunately is still lacking). It is written in Go and it is quite fast and light on resources. First public release, feedback and patches are more than welcome ;) Regards, Lefteris Zafiris

On Tue, 2016-02-23 at 17:06 +0000, Steve Howes wrote: > Google?... Yeah... searched "google voice recognition api asterisk", browsed though various results. Nothing helpful for a beginner, very confusing bla bla... Thanks anyway for your help. F.

I have written a module for asterisk that uses the eSpeak speech synthesizer (http://espeak.sourceforge.net/) to render text to speech. The source is available here: http://zaf.github.com/Asterisk-eSpeak/ It's similar to app_festival and app_flite. It's only tested against asterisk 1.6.1 on x86 Linux but it must be working for other 1.6 branches too. Comments, fixes and suggestion are

Im looking for wifi sip phones that support auto provisioning and work flawlessly with atserisk. Can anyone suggest me some models?

Greetings list, Has anyone compiled (or could point me at) a list of the minimum required modules and conf files for a very basic 1.8 deployment? We have lots of 1.4 boxes in production, and I'm currently setting up a pair of 1.8 boxes to bounce calls coming in via IAX over IPv6 over to the existing 1.4 boxes. All the new installs need to do is receive calls via IAX and send them out via

...6kHz file playback in asterisk 1.6.x, use of libsamplerate for faster resampling, better error handling and cleaner code. You can get them here: Asterisk espeak module: http://zaf.github.com/Asterisk-eSpeak/ Asterisk flite module: http://zaf.github.com/Asterisk-Flite/ ---------------- Lefteris Zafiris

Hi, a set of AGI scripts that provide ASR and TTS for asterisk using the iSpeech API (http://www.ispeech.org/) are available on this page: http://zaf.github.io/asterisk-ispeech/ This is the first public release, updates will soon follow. Feel free to test and report. Regards, Lefteris Zafiris -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130522/d89395e1/attachment.htm>

I am not aware of any alive project on this field. My client is interested in it. He told me that Google Translate web service used to provide pronunciation of hebrew texts, but now speech button is disabled for Hebrew, and also direct requesting for speech generation (by Lefteris Zafiris's awsome scripts) fails. I can work on Asterisk integration by myself, i'd be happy to know of such engine(s) at all. -- Andrey Utkin

Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -------------- next part -------------- An HTML attachment was