similar to: Vestec vs Lumenvox

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, I installed the Vestec module to one of my development Asterisk servers a few months ago but now I need to move the license to another host. Does anyone know how to do this? I've had a look on my Account page on the Digium website but it only shows the Language Pack, and I can't do anything with this either. Can anyone point me in the right direction please? Thanks Lee

I'm having a lot of problem with it recognizing "oh" for zero. I've tried both "o" and "oh". In one case, I just tried: $digit = o { out = "0"; } | fundamental {out = "2"; }; So I gave it a choice that was VERY far away from what I said. But when I said "o o o o o", more than 75% of the time, it had a bunch of

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

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 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

Greetings listers, I have been running 1.4.21 for about 7 months now, but have been told I have to move up the 1.4 food chain or into the 1.6 chain because 1.4.21 is too flaky for our POTS line handling (does funny things with echo, doesn't connect to external conference calls, etc.). Which release will give me the most joy/least headache/closest performance to

Greetings Listers, I'm running 1.4.21.2 on SUSE 11.0 with and zaptel 1.4.12.1 on a TDM400P. Most of my calls work great, but occasionally we try to connect to a customer or vendor external conference call and the call will drop after 60-65 seconds unless I have an Answer before the Dial in the dialplan. Isn't this solution a hack and what would be a better one?

Does it exist? Sending email to their support address appears to be a black hole. They reference a forum, but Google can't find it. I keep having problems in any grammar than has a an "o" for "zero": it breaks recognition anywhere NEAR it. For example, if I say "two o five", it gets recognized as "one o five".

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a

I have a delay of "0" on SpeecBackGround, but when I enter DTMF, there's an almost-exactly five second delay before it returns. Where is this delay controlled? How can I shorten it? Is there a way to set the maximum number of digits to look for?

Max and List, Could you advise me if I am using the proper caret syntax to carry out leave-one-out cross validation. In the example below, I use example data from the rda package. I use caret to tune over a grid and select an optimal value. I think I am then using the optimal selection for prediction. So there are two rounds of resampling with the first one taken care of by caret's train

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)}"

Does anyone have experience with this product?

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:

Greetings listers, I am running Asterisk 1.4.21.2 on Suse 11.0 on a Dual Processor Dell Poweredge 1650. I recently attempted to update the BIOS and now have this happen: When the machine starts up, Asterisk runs fine. When I do a large wget or scp, the local SIP to SIP quality goes to heck in a handbasket. The only resolution I've found so far is to completely

Hi Listers, I'm running 1.4.25-rc1 on opensuse 11.0 with dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and snapdsp.0.0.2. Incoming calls work fine. Outgoing calls made directly (exten => s,1,Dial(DAHDI/G1) then number work fine. The problem I have is trying to let Asterisk make the call (exten => s,1,Dial(DAHDI/G1/5551212,,r). If I use "m" (moh) the

expected context is valid (may not work on 1.2, I started this ride at 1.4 and therefore have no backward knowledge). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David Nickel Sent: Wednesday, May 05, 2010 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hash Dial

--- span_1 = DAHDI/g1 1,1,dial(${span_1}/${EXTEN:0}) --- I can only presume some form of precedence overrides the group configuration in the recent asterisk installs and not for the servers set up earlier. On 26/5/09 4:01 PM, "Kal Feher" <kalman.feher at melbourneit.com.au> wrote: > Ok I've solved the problem. I do not think it was as switchtype issue after > all as