similar to: Microsoft Speech Server/UCMA Integration

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.

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

Hello asterisk users! Does asterisk (or any of its plugin) support MRCP to interface with 3rd party professional TTS engine? I've found very little information about this, for example: http://www.zen-turkey.com/blog/default.aspx?id=12&t=Asterisk-MRCP-Integr ation Thanks in advance! Regards, Claudio Internet Email Confidentiality Footer

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:

Hello, Is there anyone who has ever done a setup of VoiceXML combined with some licenses from Nuance for the ASR/TTS engine within Asterisk ? I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS engine, but we are having a couple of issues which I guess are caused by VoiceGenie. If there's an alternative, it would be very interesting for us. Thanks, -- Eric Rousse

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

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

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

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

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 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

Hi Reed, I've been using Speex to transmit TTS for years. It works very well with no tweaking. I use Microsoft TTS ("Microsoft Mike") with Speex at 16kHz wideband and VBR quality 6. Sometimes I forget that the sound is even coming from another computer and being compressed+decompressed. If anything, TTS seems easier for Speex to deal with than real voice. But I don't

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:

Hi, Xiph has done some really good work but since Speex things have seemed quite quiet. Devs to hobbyists have a huge battle with VoiceAI audio processing and even when available extremely similar routines are separated across libs so that much process duplication occurs and increases load. The Alsa-plugins of Speex echo just don’t seem to work whilst repos using the speex libs do.

Hi Everyone, Has anyone got any opensource speech recognition software to work with Asterisk? Please only list WORKING ones. Not the "theoretically" should work ones! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100821/4d11d6c0/attachment.htm

Briefly: I want to be able to have "press or say (number)", with Asterisk listening for a spoken number, but accepting a DTMF digit, too. I'm posting everything I found so far, here, partly to show working, but also in case anyone else finds it useful. So, moving on.... This looked hopeful for a moment until I realised that it doesn't do DTMF:

The Error Which I face I have attached. I need a clarification of Why I face this error and how to overcome this. Anybody know Please help...... -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150309/51d674d5/attachment.html> -------------- next part -------------- A non-text attachment was

Replace nospam with ssb22 to reply by email. Hi, Does anybody have pointers about using SAPI 5 speech synthesis engines (sometimes called a "SAPI 5 voices") under Wine? I've searched all over the Web but can't find any mention of it. Although English speech is well-supported in Linux, speech in languages like Chinese, Japanese and Korean is not. Commercial offerings are

I used Microsoft text to speech with speex. It sounded reasonably well. I didn't experiment with various quality settings (8KHz, default quality). I was capturing audio from a sound card, not the direct wave data from the TTS engine. .a -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Reed Hedges Sent: Friday, May 26, 2006 7:23 AM

Hello, Question for developers: what is the asterisk way to integrate with ASR (speech recognition)? Question to the community: has someone done anything in this direction? On the first glance that shouldn't be too hard. One part is delivering audio to the engine (for example, main ASR players Nuance and Speechworks will be happy with RTP) - can be done via RTP forking. The other part is