similar to: VoiceXML + Nuance

Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the

Hi, I'm currently working on a setup between Asterisk and VoiceGenie (which is a IVR system). The way my setup is done, is that I have a PRI line coming in my Asterisk server, and then VoiceGenie is connected to Asterisk via SIP, like any other softphone basically. I'm able to receive calls in Asterisk and then link them with VoiceGenie. But one of my issues is that when I get an

Does anyone know if X-Ten or SJPhone support multiple cordless handsets for multiple lines? I have an office with multiple roaming users(nurses) that are in and out. I'd like to provide them telephones, and my idea is to have a PC sitting in a corner somewhere running a softphone client. When a nurse comes in she just picks up any available handset(anywhere from 2-5 per office) and starts

hi, is there anything going with VoiceXML in asterisk??? is this the list to query regarding this or should I put this on the dev list? thanks, dave cantera

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

Firstly can I ask when the documentation site will be online again? I'm struggling here without it. Further to my recent post I have tried to simplify things a little. I have used a VoiceXML app to simple call an asterisk extension. EG: <form id="transfer"> <block> <call name="xfer" dest="sip:101 at 10.0.4.147:5060"/>

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

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 (

-----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

Everyone, I will be attending Astricon in Phoenix and would like to have a little get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Exchange ideas and go over some use cases and see how we can all work together to improve our understanding of the dynamics of how everything works together. * Scaleability * Reusability of code

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

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

Hi I need to know if Asterisk supports the full features of the SIP REFER message (i.e blind and supervised transfers). I'm trying to do a supervised transfer through Asterisk from a VoiceXML application using the <transfer> tag and setting bridge="true" (i.e <transfer name="transfer1" bridge="true" connecttimeout="10s"> ) but as soon as

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

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

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

Hi there, yesterday I came across the "Vocera Communication Badge" and now I'd like to know if anyone here has played with that thing (or even just seen it in real life), and if a price tag can be found for this device? Too bad they don't use SIP... ;-( http://www.vocera.com/ http://www.heise.de/newsticker/data/tol-25.12.03-001/ Cheers, Philipp ** Wireless Specifications

Hello, I am running a samba 2.2.7 server that currently belongs to a NT4 domain (security = domain and password server = <pdc name>). There is a plan to roll out a Windows 2003 server with it's own active directory domain. Can my current samba server allow users from both domains to mount its shares? If so how? Thanks, -- Neil Watson Network Administrator www.voicegenie.com

Hi, As Freelance programmer/consultant I'm looking for project/task of IVR/CTI/CRM/IP-based, my skils are as following, 1 Dialogic-based CTI/IVR software programming 2 Intervoice IVR development 3 Siebel CRM integration and development 4 IBM DirectTalk and WebShpere Voice(VoiceXML,...) 5 IP-based development(VoIP,h323,sip,...) 6 Cisco