similar to: Making Asterisk a "Voice Router"

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.

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 again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

Hi all, Anyone knows of any Voice Recognition applications which use VoIP? Preferably open source. I am basically trying to build a Voice Call Router, something to recognize a spoken name and then transfer the caller to the right party's extension? TIA. -Eric -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim

Is this replacing Astricon this year? If so it looks like a pretty poor showing in comparison to Astricon Dallas last year. Cheers, Dean ________________________________ From: Carl Ford [mailto:carlf at vonmag.com] Sent: Wednesday, 18 July 2007 9:09 AM To: Dean Collins Subject: Announcing Digium/Asterisk World's Conference Program

I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

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.

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 list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090802/abb21767/attachment.htm

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 List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels

Hi there. My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me to the FM on this. I would like to have either a web page or an application that I can view that whenever a call arrives on the Asterisk server the application will display the callerid information. I've found quite a few examples of the reverse of this. To where a script is

Hi folks, Is there anywhere I can possibly get a model of the exam itself, maybe possible scenarios for the prac, etc? To people who have done the exam....any helpful hints ? Thanks,

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

Hi all, I am looking for a voice recognition technology integrated to asterisk. Any suggestion about it? Ango

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