similar to: VoiceXML? question

Hello, Is there anyone who has ever done a setup of VoiceXML combined with some licenses from Nuance for the ASR/TTS engine within Asterisk ? I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS engine, but we are having a couple of issues which I guess are caused by VoiceGenie. If there's an alternative, it would be very interesting for us. Thanks, -- Eric Rousse

I have a customer looking for an automated way to provide his customers information. He found some windows software called Active Call Center - but I believe that he did the 40 day trial and it crashed his windows machine to much. So he wants something that can do a similar task running on linux. The Users expierience should go something like this One of his customers calls in the IVR asks

Hi All, Do any of you know what the status is for VoiceXML support in * ? Is it already existing, or is it planned for the future? If it's not in now, do you know on what type of scale the work would be to integrate VXML into * ? Thanks in advance

Hi, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten => _X.,1,Answer() exten => _X.,n,Wait(1) exten => _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: <?xml version="1.0"?> <vxml version="1.0"> <form> <block><audio

I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the

Anyone know of anybody doing VXML with Asterisk and/or Linux? Tia Kevin

Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

A couple of weekends ago I decided to see if I could get Asterisk to play nice with TellMe's VoiceXML studio. They provide the VoiceXML studio for free, and you can access it through SIP, so I thought this would be a fun and cheap way to integrate voice recognition into my IVR. I have posted a brief tutorial with code and examples on the voip-info.org wiki (

hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070320/227a3b32/attachment.htm

Dear all, I have two system Sys A and Sys X. Sys A is normal PC. Sys X have installed asterisk 1.6 and i want register(or reserved) sip extension(like 4001,4002,4003..) through Sys A(Sys A have some ip address) but i don't use any soft-phone means i want to write Perl or php(any language) script to register sip extension. Suppose to 4001 is reserved with Sys A. 4002 is reserved with

http://deancollinsblog.blogspot.com/2007/03/microsoft-buys-tellme.html I thought I would email this post I made on my blog from yesterday as a way of stimulating discussion on this. It looks like the Asterisk community is no closer to getting a Pre-Paid 'Offboard Speech Recognition Processing' SIP gateway than ever. I've left the

Hi to all, I am sending a SIP REFER message to Asterisk from a VoiceXML application using the <Transfer> element to do a Transfer through Asterisk. I need to know if Asterisk supports the full features of the SIP REFER message because if i set 'bridge=true' in the <transfer> element of the VoiceXML application to supervise the call, Asterisk sends a NOTIFY message with

Hello, I've seen that Nuance SIP audio provider supports additional information (parameters and extra headers) in SIP URIs, using the format: sip:user:password@host:port;uri-param1;uri-param2?header1&header2 For example, sip:1234@myserver.com;extra_header=Uui?Uui=Hello Does Asterisk support this format? Is there a way to retrieve the value of these additional headers, and then decide

Hi Greeting to all you out there. I am new at asterisk, I have been working with PLMN platforms telecommunication for 5 years with NSN and Huawei. We have recently built an asterisk PBX with Trixbox and connected it to our MSS using Digium E1 cards(ISDN). Everything went smoothly as there are tons of information out there, except for the TON number. If you have worked in Telecommunication you

Dave, Docos for Comedian Mail? Justin >From: dave cantera <david.cantera at iacnet.net> >Subject: Re: [asterisk-users] Need good voicemail documentation > >An HTML attachment was scrubbed... >URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080208/501668f8/attachment.htm __________________________________________________ Do You Yahoo!? Tired of spam?

Hello everyone. Is it possible to do some very basic voice recognition from within Asterisk's dialplan? What I'm aiming at is the ability to speak the digits I want to dial from my mobile phone. Dialing digits on my mobile phone while driving is not all that safe... Thanks for any input, Cosmin Prund

I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Milan! Thanks for your comments ont he requirement list. [Milan Zamazal, Dienstag, 26. Oktober 2004 20:58] > [Since the mailing list apparently hasn't been created yet, I continue > in private not to freeze the discussion for too long.] > I have just asked David Stone when we can start using the list. > BTW, this might be

All; Since I have interest in providing the capability for callers to speak the department, person or number they wish to call, as well as other IVR scenarios, I have been reviewing much of this lists email archives and searching the web for open source voice recognition that will work with the Asterisk PBX. What I am trying to determine, is what will it take to get it working on Asterisk? How