similar to: Lumenvox - Gentoo

The purpose of this patch is to add the matahari qmf agent package (http://git.et.redhat.com/?p=matahari.git;a=summary) to the node image. Matahari is a qmf agent that publishes hardware information to an amqp broker, replacing functionality from ovirt-identify-node. However, since the capability is not ovirt specific, the agent is maintained as a separate project (see link above). Dependencies

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

Just tried to install the Amazon-downloader on Centos 5.1 (the Fedora 8 version, which seems the closest match of any they offer). Of course it gets a bazillion unfulfilled dependencies. so I tried "yum localinstall ./a*m" and it trundles along for a while finding several of the packages available then spews out this: Error: Missing Dependency: libboost_date_time.so.3 is needed by

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 listers, I'm a 1.4 holdout who uses mostly Suse platforms. I bit the bullet and installed a Centos box to get Lumenvox up and running, but now see this "new" offering of Vestec that supports OpenSuse and Windows in addition to the Platforms supported by Lumenvox. Anybody out there working with Vestec that can give me a heads up on how it works or doesn't?

This Friday, February 15th, at 12 Noon EST, 9AM PST, 17:00 UTC, Lumenvox will be joining us on the VoIP Users Conference. This week, the last in a series about IVR, Lumenvox will be there to discuss and field your questions on their speech recognition solutions. http://www.VoipUsersConference.org - for info on the conference, how to connect, etc IRC freenode.net #voip-users-conference - to

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:

Does anyone have experience with this product?

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

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.

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?

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

Hi, This is a naive question, but is there a way in my AGI script to simultaneously play audio and listen for DTMF or voice responses? I've heard VOIP hackers call this "inbargeability;" it's the ability to "barge in" to a playing audio clip. I'm planning to use Lumenvox for the DTMF and voice recognition, BTW. Not sure if that matters. Many thanks to anyone who

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>