similar to: AGI and embargeability

Hi, We've run into an interesting (to us) problem with SpeechBackground. Inside a AGI script, we're playing some extended audio?basically, like a podcast?and we want playback to stop if and only if the speech recognized matches something in our grammar. If there's speech that doesn't match, we just want to go right on playing. (We're using LumenVox as our speech recognition

Which speed recognition products will also recognize DTMF? In other words, I want to say "Please speak or dial the conference number". Does Vestec allow that? LumenVox? Any other way?

I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent problem?

Hi Guys I am new working with lumenvox products, and unfortunately I had not been able to install it properly, I follow the steps in lumenvox site and it looks like it works I mean: ========================================================= [root at pbx-millenium examples]# ./example 127.0.0.1 Connecting to 127.0.0.1 Interpretation 1: 8587070707 count=0, decode returns 1 Interpretation 1:

Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh

Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm

Is anyone running Lumenvox on Gentoo? My asterisk install has been running like a champ for a few years now and I really hate the thoughts of changing distros just for Lumenvox. Here is my issue: The engine needs the libs from boost. I emerged boost and noticed that there were four libs that the engine were looking for that were not installed via portage. libboost_regex.so.2

Hello! I need to: 1) call special number (or run special application) on mobile phone 2) establish connection between mobile phone and server 3) allow server to recognize spoken numbers (Polish language) and some other control words 4) let the server to say some short answers (prerecorded in mp3) according to some algorithm and recognized words 5) let the server to save little text file on its

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 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.

Hello all, how are you? I would like to know from someone uses or has used the engines of LumenVox for integration with the asterisk for voice recognition. Best Regards Josu?

I just made a change to one of my *.gram files for my LumenVox IVR. I was just wondering if anyone knows the command in Asterisk to reload the .gram files. Thanks for your help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100608/22a0fc65/attachment.htm

Hi all, Yes, this might not be the proper list for this, but i have a question about Asterisk and voice recognition. If I want to create a menu system where the user can say different things in the Swedish language what should I look at? For example, i want the user to be able to say something simular in Swedish: connect disconnect help and so on. Best regards and thanks, Christian

When I was looking for a Java FastAGI interface for Asterisk I came across asterisk-java first and didn't realize there was more than one out there. It seems to work fine and I've got my first project working with it, but I was wondering which Java FastAGI implementation is the most popular and how they compare against each other. So I'm aware of: asterisk-java JastAGI

http://www.boingboing.net/2006/06/05/play_zork_by_phone.html Let me preface this idea with one comment: I don't have the time to do this - I don't even have time to eat these days. But someone out there has the cycles to do this... and it would be very cool. OK, so now Zork is attached to Asterisk, but using the less-than-clear Festival engine. There are beta tests of the LumenVox

Hello list, I was wondering if it were possible for asterisk to do a voice recognition type IVR? Like you know how most banks and stuff do, where they ask you to say your selection or key it in? If this is possible, how can I set this up? I'm using FreePBX 2.11 on Linux, CentOS 6.7 32-bit, asterisk version 1.8.32.3. Thanks.

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>

Hi Gang, I'm working on a lumenvox app and am having "fun" with the Gotoif's on speech/DTMF recognition. If you're using DTMF to enter a number instead of speech to enter a numeric value, the engine will often return a "confidence score" of 1000 instead of 1-999. Therefore this Gotoif fails: exten => s,n,GotoIf($["${SPEECH_SCORE(0)}"

Hi everybody, I have a Fedora Core 4 x86 32 bit install, which I recently upgraded from asterisk 1.2 to the office 1.4.4 tarball. In the process of doing that I had to upgrade some autoconf/automake stuff, but it worked fine, and my new asterisk works fine. Except that anytime I receive a fax with spandsp and app_rxfax, asterisk seg faults. I have applied the spandsp

Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play