similar to: Astricon Meetup

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.

Hi to all, I am sending a SIP REFER message to Asterisk from a VoiceXML application using the <Transfer> element to do a Transfer through Asterisk. I need to know if Asterisk supports the full features of the SIP REFER message because if i set 'bridge=true' in the <transfer> element of the VoiceXML application to supervise the call, Asterisk sends a NOTIFY message with

-----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

A couple of weekends ago I decided to see if I could get Asterisk to play nice with TellMe's VoiceXML studio. They provide the VoiceXML studio for free, and you can access it through SIP, so I thought this would be a fun and cheap way to integrate voice recognition into my IVR. I have posted a brief tutorial with code and examples on the voip-info.org wiki (

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

hi, is there anything going with VoiceXML in asterisk??? is this the list to query regarding this or should I put this on the dev list? thanks, dave cantera

Hi All, Do any of you know what the status is for VoiceXML support in * ? Is it already existing, or is it planned for the future? If it's not in now, do you know on what type of scale the work would be to integrate VXML into * ? Thanks in advance

All; Since I have interest in providing the capability for callers to speak the department, person or number they wish to call, as well as other IVR scenarios, I have been reviewing much of this lists email archives and searching the web for open source voice recognition that will work with the Asterisk PBX. What I am trying to determine, is what will it take to get it working on Asterisk? How

Thank you Lars for the reply. Your instructions were dead on. Yes I have considered the security risks; however, I think scaleability and rapid deployment out weigh the security risks for me. On Aug 18, 2015, at 6:31 AM, Lars Kruse <lists at sumpfralle.de> wrote: > Hi, > >> Is it possible to generate one key pair? > > Maybe this is all you want to know? > openssl

When I look at the lastest UniMRCP manual, they only mention as high as Asterisk 13. Does anybody know if I need to do anything to allow it to work on Asterisk 14 and, if so, what that is?

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

So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten => _X.,1,Answer() exten => _X.,n,Wait(1) exten => _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: <?xml version="1.0"?> <vxml version="1.0"> <form> <block><audio

Anyone know of anybody doing VXML with Asterisk and/or Linux? Tia Kevin

Hi I need to know if Asterisk supports the full features of the SIP REFER message (i.e blind and supervised transfers). I'm trying to do a supervised transfer through Asterisk from a VoiceXML application using the <transfer> tag and setting bridge="true" (i.e <transfer name="transfer1" bridge="true" connecttimeout="10s"> ) but as soon as

Hi all, We are organising another open source search social evening (OSSSE?) in London on Tuesday the 12th of January. The plan is to get together and chat about search technology, from Lucene to Solr, Hadoop, Mahout, Xapian, Ferret and the like - bringing together people from across the field to discuss ideas and ask questions over a quiet drink. We've also created a group on Meetup.com

I'd like to announce the release of an open source connector bridge for Asterisk and UniMRCP. The connector bridge is an implementation of Asterisk's Generic Speech API using UniMRCP client stack. This module allows Asterisk to connect to MRCPv2 or MRCPv1 compliant servers for speech recognition. It also allows to offload Asterisk using client/server architecture MRCP provides. Moreover,

Firstly can I ask when the documentation site will be online again? I'm struggling here without it. Further to my recent post I have tried to simplify things a little. I have used a VoiceXML app to simple call an asterisk extension. EG: <form id="transfer"> <block> <call name="xfer" dest="sip:101 at 10.0.4.147:5060"/>

I will as well be interested in attending the meetup (in Hyderabad). IMHO we can start with Meetup and select a city. How are Meetups expenses generally handled elsewhere ? Regards,Pankaj Date: Wed, 30 May 2018 14:59:21 +0530 From: Madhur Amilkanthwar via llvm-dev <llvm-dev at lists.llvm.org> To: Siddharth Bhat <siddu.druid at gmail.com> Cc: Siddharth Shankar Swain <h2015096 at

Hello, We''re organizing a meetup group for Silicon Valley Ruby on Rails engineers. If you''re in the Valley and interested in meeting with local Rails programmers, we would like to invite you to join the group. We haven''t set the date and time for our first meeting, but we''re hoping to have our first get together sometime in the next few weeks. To join,