search for: voicexml

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 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...

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...

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

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

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 ( http://www.voip-info.org/wiki/index.php?...

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 Asterisk receives the SIP REFER message generated by the VoiceXML application, it sends back a NOTI...

...each other. So we would need a string list in this case. A driver can easily turn the string list back to a string easily, it would only help those drivers that would parse the the string for tags rather than passing it on to an xml-supporting engine. > I'd suggest using SSML instead of VoiceXML. If I'm not mistaken, SSML > is what is aimed at TTS, while the purpose of VoiceXML is different. > I thought that the GSAPI used some extention of VoiceXML, but maybe I am misinformed here. We should use the same syntax in any case. We can discuss the different possibilities on the l...

...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 much effort and cost? So far I have uncovered references to the following: 1) VoiceXML standards, and forums 2) OpenVXI - which supports VoiceXML, simulated speech, telephony 3) PublicVoiceXML 4) Sphinx - a Carnegie Mellon University Speech recognition project funded by DARPA >From what I can tell, I feel I am uncovering the tip of...

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

...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 * Standards (VoiceXML, MRCP and more) If anyone is interested please email me off list and we'll plan on having a meeting of minds. Thanks, Brian West FreeSWITCH.org

...6 1 0 18:33 ? 00:00:00 /bin/sh /usr/sbin/safe_openvxi root 2114 2076 0 18:33 ? 00:00:00 openvxi -channels 100 -config /etc/openvxi/client.cfg root 2606 2409 0 21:00 pts/2 00:00:00 grep openvxi [root at pabx002 tmp]# The /etc/asterisk/vxml.conf file contains: ; VoiceXML Configuration ; [general] wav_codec=gsm videosilence= audiosilence= [license] max=1 video=no key= And, finally here's my console output: -- Executing Vxml("SIP/xxx.201.84.142-b7600c30", "file:///tmp/menu.vxml") in new stack VoiceBrowser interface file:///tmp/menu.vxml...

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

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"/> <if cond="xfer == 'connected'"> <prompt>Call connec...

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 IPCC(ICM,CallManager,Unity,IP-IVR) 7 MS .Net for SALT voice 8 Programming including C/C++,J2EE,MS .Net I'm based Toronto,Canada,Thanks A lot! Howard --------------------------------- Do you Yahoo!? Exclusive Vid...

I'm hoping Asterisk can help me solve an unusual problem. I need two SIP endpoints (VoiceXML gateways) to transfer DTMF tones to each other. Both of them can detect DTMF according to rfc2833. However, one of them (host2) must generate DTMF inband. Happily, I came up with the following sip.conf to allow host1 to detect DTMF tones generated by host2. [in] type=peer host=host1 dtmfmode=rfc...

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:

Which is the best way around to integrate Asterisk with VoiceXML like VoiceGlue...! Am using Asterisk 11.2.1. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140205/6098117f/attachment.html>