similar to: Proprietary add-ons for Asterisk 1.8

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 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

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

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".

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?

Is there an issue with running it with the latest from the 1.6.2 branch? I did an svn update and make install and now when somebody comes in via Skype, I get an infinite loop of: [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]:

Am I correct that if I'm running an -rc or from an SVN release tree that there's no way I can use any commercial add-ons from Digium, such as Skype, Cepstral, or G.729?

Hello list, 'Just wondering if anyone can point to a very light-weight and easy to incorporate into Asterisk (v. 11.x) to handle a minimal set of responses, like: 0 - 9 yes no (maybe * and # for some people) The idea is that within an IVR menu, the caller could respond by speaking to the typical IVR options, like: For Archie, press or say 1 now For Veronica, press or say

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Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

There are some rather nice archive replacements for Mailman that provide full searching, etc. I can provide more info if wanted. Geoff

Dear R Help-Team, If I start R and type in search() the result is: [1] ".GlobalEnv" "package:ctest" "Autoloads" "package:base" I often need an Add-on-package. I can load it with "> library(package)" or include a require-command in the .First-function. The .First-function is fine, if I always use the same working directory. But I

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?

When clicking on the Tools->Add-ons link, the following. Running Excel without a Windows install.. Console output.. (Debug below) wine -- ./fake_windows/Program\ Files/Microsoft\ Office/Office/EXCEL.EXE fixme:ole:CoRegisterMessageFilter stub fixme:mpr:WNetOpenEnumA (00000002, 00000001, 00000000, (nil), 0x4067ea5c): stub fixme:hook:NotifyWinEvent (4,0x10021,-3,0)-stub!

How to upgrade h323 from Asterisk add-ons (from version 1.2.1 to 1.2.2)? In INSTALL they don't say anything about upgrade... Thank you for your time! -- Tomislav Parcina tparcina#lama.hr

Hi All, I have substantially expanded the table that compares SAS and SPSS add-on modules to somewhat equivalent R packages. This new version is at: http://r4stats.com/add-on-modules and I would very much appreciate any feedback you might have on it. The site http://r4stats.com is the replacement to http://RforSASandSPSSusers.com and includes the support files for both "R for SAS and SPSS

by "GUI" do you mean a configuration utility or a User Interface? MATT--- -----Original Message----- From: Jon Gabrielson [mailto:jon@directfreight.com] Sent: Thursday, February 10, 2005 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] asterisk GUI's that supports zap fxs extensions Are there any gui's that support zap fxs

I am trying to get more familiar with ASTCC, but I miss some tools, but I believe somebody has already thought about it: 1. I would like to send a standard letter to the users, as soon their balance drops below a certain value. E.g., Dear user, you have only 1.- left, .... consider to fill up... To make a php program which is started with cron every midnight should do the initial trick,

Greetings List, I'm looking for a way to add additional options to the VoiceMail system, specifically, we want to give the user the ability to transfer to an extension from the VoiceMail system. Do I have to really change app_voicemail.c to extend the functioanlity of the Voicemail system, what have I missed?

I have running Asterisk 1.2.5 with addons 1.2.1. on Fedora Core 4. I have installed ooh323 from 1.2.1 addons. How to upgrade addons to 1.2.2 version and install new ooh323 driver? Do I need to install addons 1.2.2 if I only need new ooh323 driver? Can I just untar addons, and run "make clean; make; make install" and then execute following cd asterisk-ooh323c ./configure make make