Re: [asterisk-users] Streaming for ASR

t complicated but that > is a way to get media out. > There is also EAGI, not very flexible but still an option. -- Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- J

Re: [asterisk-users] AGI: How to break out of AGI when stream_file escape_digits are detected in middle of long sequence of files?

to try something like: pressed_digit = agi.stream_file(promptFile,escape_digits) in you case the raw value 48 is converted to its ascii equivalent so pressed_digit will have the value 0 -- Lefteris Zafiris -- _ -- B

Re: [asterisk-users] Phone Number Validation

On Tue, 29 Mar 2016 09:53:15 +0100 Rizwan H Qureshi wrote: > Hi Everyone, > I need to develop a service which tells me whether a given phone number is > in service and is valid or not. It can be international number. This is > basically to clean the list of leads we have.

Re: [asterisk-users] Voice recognition IVR Is it possible?

On Tue, 23 Feb 2016 22:56:50 +0100 Frank wrote: > On Tue, 2016-02-23 at 17:06 +, Steve Howes wrote: > > > Google?... > > Yeah... searched "google voice recognition api asterisk", browsed though > various results. Nothing helpful for a beginner, very confusing

Re: [asterisk-users] Anyone doing speech to text?

Vestec: http://www.asteriskexchange.com/listings/113 Regards, Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] agitator - FastAGI reverse proxy

are more than welcome ;) Regards, Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Asterisk-eSpeak and Asterisk 12

://zaf.github.io/Asterisk-eSpeak/ Hello, please make sure that you are using the latest trunk code and not some older 'stable' release. You can get it from here: http://github.com/zaf/Asterisk-eSpeak/tarball/master Regards, Lefteris Zafiris

[asterisk-users] Automatic Speech Recognition and Text To Speech using iSpeech

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

Re: [asterisk-users] Voicemail to text for Asterisk

against using this into production since google haven't yet defined the terms of service for speech recognition, and its more or less a hack for now. [1] http://zaf.github.com/asterisk-speech-recog/ Lefteris Zafiris

Re: [asterisk-users] html/js/flash/air SIP clients?

On Thu, 2 Aug 2012 10:27:59 +0800 Arstan Jusupov arst...@gmail.com wrote: 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

Re: [asterisk-users] OT - mstts.agi - Where to find API key ?

here suggestions that may help others to use this script ? That's what this list is all about. Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available

On 03/05/2012 09:52 PM, Jason Parker wrote: On 03/05/2012 01:49 PM, Eric Germann wrote: Will a 1.8.10.0 build be imminent or should we go ahead and push this in to production with testing? Thanks! EKG ~20 minutes Some packages seem to lag behind, eg asterisk18-addons-mysql is

Re: [asterisk-users] Speech recognition in asterisk using google voice API

be found here: https://github.com/zaf/asterisk-speech-recog/tarball/master Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] 44Khz files in Asterisk 10

the frequency of the codec you are using, eg 8kHz or 16kHz for wideband codecs. And if you have to resample use another resampler like sox (with dithering, lowpass etc) instead of letting asterisk do this for you. Lefteris Zafiris

Re: [asterisk-users] 44Khz files in Asterisk 10

On Mon, 9 Jan 2012 14:40:47 -0600 Danny Nicholas da...@debsinc.com wrote: What do I need to set to play 16 Khz wav files? Rename them to .wav16 Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Speech recognition in asterisk using google voice API

://wiki.asterisk.org/wiki/display/AST/Speech+Recognition+API) This will make the application actually usable for real case scenarios and not a proof of concept as it is now. Lefteris Zafiris -- _ -- Bandwidth

Re: [asterisk-users] Speech recognition in asterisk using google voice API

versions of sox. Anyway I'm not sure audio normalization and the rest we use sox for is really needed. My tests so far didn't show any improvements in detection rates. Keep in mind that all this is still WIP and the option to use sox is more for testing than for serious use. Lefteris

Re: [asterisk-users] Speech recognition in asterisk using google voice API

this. I have replaced sox with flac and it seems to work now on older platforms too (tested on Centos 5 with asterisk 1.4). You can get the updated code here: https://github.com/zaf/asterisk-speech-recog/tarball/master Lefteris Zafiris

Re: [asterisk-users] Speech recognition in asterisk using google voice API

particular reason you want the googletts.agi data in flac? Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Speech recognition in asterisk using google voice API

be on your distro repos already. Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Speech recognition in asterisk using google voice API

? This is a gray area at the moment. Voice recognition is undocumented in google's API and i guess not officially supported yet. I hope it gets covered by the general TOS of google services: http://www.google.com/accounts/TOS Lefteris Zafiris

Re: [asterisk-users] Speech recognition in asterisk using google voice API

On Wed, Jan 4, 2012 at 8:27 PM, isr...@gmail.com wrote: Does anyone know what languages are supported? For sure english and spanish, since its undocumented i don't have a complete list yet. Lefteris Zafiris

Re: [asterisk-users] Speech recognition in asterisk using google voice API

level etc. I have also read that normalizing the recording and setting the gain to -5 db improves detection rates. I m experimenting with this at the moment and there will be some new code soon (as soon as i get sox working in RHEL/Centos 5 :P ). Lefteris Zafiris

Re: [asterisk-users] Speech recognition in asterisk using google voice API

, this can be done by both sox and flac encoder. For now the script uses flac encoder for compatibility with older distros (mainly RHEL 5). Sox is a bit more flexible and also gives you the option to edit the sound data (normalizing, changing levels etc). Lefteris Zafiris

Re: [asterisk-users] Speech recognition in asterisk using google voice API

-speech-recog/asterisk-speech-recog-0.3.tar.gz Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] Speech recognition in asterisk using google voice API

, suggestions and bug reports are more than welcome. Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] AGI script that uses google's text to speech engine

release, documentation and dialplan examples can be found here: http://zaf.github.com/asterisk-googletts/ A big thank you to all the users that contributed with feedback, bug reports and suggestions. Lefteris Zafiris

Re: [asterisk-users] AGI script that uses google's text to speech engine

that gets from google to raw slinear. In that case mpg123 or sox failed to run. It would be very helpful if you could send the full console output with verbosity set to 3. Please reply to my mail address so we don't pollute the list. Thanks for the feedback Lefteris Zafiris

Re: [asterisk-users] AGI script that uses google's text to speech engine

by these 2 variables in the script: $usecache = 1; $cachedir = /tmp; Voice data gets stored in the cachedir for future use so we don't have to fetch it from google each time. Lefteris Zafiris -- _ -- Bandwidth

Re: [asterisk-users] AGI script that uses google's text to speech engine

format_mp3 module isn't available in many installations. Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] AGI script that uses google's text to speech engine

on. Thanks for the feedback. Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] AGI script that uses google's text to speech engine

including this fix can be obtained here: http://github.com/zaf/asterisk-googletts/tarball/master Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] AGI script that uses google's text to speech engine

://translate.google.com/translate_tts?tl=enq=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 -- _ -- Bandwidth

Re: [asterisk-users] IAX MOS Score measuring solution

data like rtt, jitter and packet loss from the dialplan with something like: ${CHANNEL(rtpqos|audio|all)} Based on these u can calculate R and MOS using the formulas on this page: http://www.nessoft.com/kb/50 Lefteris Zafiris

[asterisk-users] Flite module for asterisk

with asterisk 1.6 , 1.8 , 10. http://zaf.github.com/Asterisk-Flite/ Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

[asterisk-users] espeak module for asterisk

, Vietnamese, Welsh. It supports 8kHz and 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. Works with asterisk 1.6 , 1.8 , 10. http://zaf.github.com/Asterisk-eSpeak/ Lefteris Zafiris

Re: [asterisk-users] use ILBC installed from asterisk yum repositories

this in numerous systems that use the digium rpms and it works flawlessly. This method can also be used to build other modules that are missing from the digium rpms. --- Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Asterisk 1.8 minimum modules/configuration

load = res_rtp_asterisk.so load = res_timing_timerfd.so load = codec_ulaw.so load = format_pcm.so load = app_dial.so load = pbx_config.so load = chan_local.so load = chan_sip.so Lefteris Zafiris

Re: [asterisk-users] Flite issue

: http://zaf.github.com/Asterisk-Flite/ Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Flite issue

. Install flite either from your distros repos (judging by the hostname i assume you are running centos so you have to search for it in 3rd party repos) or from source. Lefteris Zafiris -- _ -- Bandwidth

Re: [asterisk-users] Flite issue

) is to download the source of flite, compile it and install it. Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

[asterisk-users] Wi-Fi sip phones with auto provisioning

Im looking for wifi sip phones that support auto provisioning and work flawlessly with atserisk. Can anyone suggest me some models? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or

Re: [asterisk-users] Wi-Fi sip phones with auto provisioning

Fred Posner wrote: On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote: Im looking for wifi sip phones that support auto provisioning and work flawlessly with atserisk. Can anyone suggest me some models? Don't know of any wifi phone that works flawlessly whatsoever. Best to consider

[asterisk-users] espeak app for asterisk 1.6

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

Re: [asterisk-users] espeak app for asterisk 1.6

Steve Edwards wrote: On Tue, 1 Sep 2009, Lefteris Zafiris wrote: 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/ I hope it sounds

Re: [asterisk-users] Flite module for asterisk 1.6.x

Klaus Darilion wrote: Lefteris Zafiris schrieb: I have written a simple application for asterisk 1.6 that uses the Flite tts engine to render text to speech. Source is available here: http://zaf.github.com/Asterisk-Flite/ It works more or less like the festival app, can use cache etc. Its

[asterisk-users] Flite module for asterisk 1.6.x

I have written a simple application for asterisk 1.6 that uses the Flite tts engine to render text to speech. Source is available here: http://zaf.github.com/Asterisk-Flite/ It works more or less like the festival app, can use cache etc. Its only tested against asterisk 1.6.1 on X86 linux but i