similar to: Text to Speech In R

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:

Why hasn't someone found 50 people who sound alike, put them in sound studios and record the 10,000 most commonly used words. You would all differnent forms of the 1,000 most words, i.e. leading, trailing, question etc. You can synthesize the other 0.05% when you run into them. With hard drives so big, processors so fast and EXT3 that can handle 30,000+ files in a single directory that

On Mar 13, 2013 10:16 PM, "Amit Salunkhe" <amitsalunkhe21 at gmail.com> wrote: > Hi > > I want to know asterisk 1.8 as text to speech server. > > If we can use as TTS server then it support SSML. > > Any sample configuration available for this requirement. Plz help me with > support asterisk as tts server. > > Amit-- > -------------- next part

Hi, I'm trying to get a specific program to work under Wine. It is NextUp TextAloud. It gives some error when I try to run it. I've also tried to install the Windows version of espeak and it won't work under Wine either. So I think there is something different that the text to speech programs use that Wine doesn't like. Does anyone have any ideas of how to get this to work?

I had been using google tts, but it started requiring a captcha for my browser, and via linux I can't access http://translate.google.com/translate_tts?q=test (redirects to captcha) as so, its not reliable On 27 August 2015 at 17:16, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 8/26/15 1:15 PM, Tech Support wrote: > > All; > > I have a customer who is

All; I have a customer who is looking for a good speech to text solution, either open source or reasonably priced commercial product, I'm open to suggestions. Thanks; John V -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150826/64d6c224/attachment.html>

Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the

Hello I have two questions ! 1. What is the best speech recognition engine for asterisk? I have searched and asked on forums and found that lumen vox is best for asterisk bala bla bla 2. For TTS (text to speech) which TTS engine will be better to use ? I have tested Flite , cepstral (i have not buyed lisence for it trial only) but still thinking may be i have a good option ? -- Best Regards

Version 2.1 of app-espeak and app-flite modules just got released. Changes include voice selection for flite, support for 16kHz 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/

Many of you are familiar with how lousy Festival sounds. AT&T has a product, NaturalVoices, that sounds much better. There are male & female voice fonts for US/UK/Indian English, French, Spanish, and German. I am considering offering a linux-based text-to-speech engine based on the NaturalVoices runtime. An asterisk module would also be provided, making it easy to add natural sounding

Thank You . But now i get solved with that error since I had some mistakes in installing googletts.agi Now when calling from my softphone i have written dialplan with an AGI script to convert from text to speech. It get executed without error but there is no sound getting played. My output, == Using SIP RTP CoS mark 5 -- Executing [1310 at Client-dial-Menu:1]

Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS? Could anyone provide pros/cons for the various ASR options for Asterisk? We need the ability for very large grammars (over 100,000 options). Because of this, my initial thought is Nuance or Lumenvox. Does this sound correct? Have a great day! Dan -------------- next part -------------- An HTML attachment was

I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: <PJSIP/2001-0000006e>AGI Tx >> 200 result=99981 (timeout) endpos=22720 <PJSIP/2001-0000006e>AGI Rx << VERBOSE "Unable to get recognition

I have done research and reading around text to speech, and was wanting to get an updated query of where everyone is with this. I have installed festival 1.4 on CentOS 3.5 ( system was installed using Asterisk at Home ISO ). I also changed the directory php application to use the festival.pl to read the names of those who have not installed a greeting. It works well enough, though the only voices

Hello Luca, Thank you for your response. I?m familiar with speech recognition and TTS, but new to MRCP. Yes, the 100k options is used for names in a directory listing. In the pre-MRCP support, Nuance ASR used API events/methods for the application to tell ASR when the prompt was playing and when it stopped. If ASR detected speech, it would signal an event so we would stop playing the prompt.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Milan! Thanks for your comments ont he requirement list. [Milan Zamazal, Dienstag, 26. Oktober 2004 20:58] > [Since the mailing list apparently hasn't been created yet, I continue > in private not to freeze the discussion for too long.] > I have just asked David Stone when we can start using the list. > BTW, this might be

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

Hi I want to can we use asterisk as TTS server. Which can support mrcpv2 and ssml. Im looking for tts server with above requirement will asterisk 1.8 is useful for me. Any configuration available. Any opensource tts available. Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm interested in taking a persons spoken recorded name (First, Last) and converting the two spoken words to text. Is there any solutions out there that would make this possible? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090130/1e90ba97/attachment.htm

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: