search for: vxml

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...

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

Hi, I am working on vxml to record voice. I have trouble with getting url of recorded voice. I want to save and I am using java to get record parameter from url and it returns a string which is audio/basic:len(123123):p0x5a6e6241, but I want to get file object or base64 string with parameter or to relate returning string w...

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

The first release of Voiceglue is now available. Voiceglue provides a VXML interpreter using Asterisk telephony and the OpenVXI VXML parsing suite. It is released under the GPL, and thus compatible with Asterisk and OpenVXI licensing. The first release is available at the project website: http://www.voiceglue.org There is also a mailing list for those interested in...

I recently upgraded to asterisk 11 from 1.8. I had VXML working via AGI in 1.8 - from extensions.conf: [VXML] exten => s,1,Answer exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})}) exten => s,n,AGI(agi://localhost/url=${ENCODED}) exten => s,n,Hangup Using asterisk 11 on the same host with the same config in extensions.conf: -- Executin...

I'm using i6net's vxml browser in Asterisk. I'm trying to work out how I can return the inputs from a menu or form back into the Asterisk dial plan. Is there a variable? It's not documented if it is. The exit tag apparently can be used to return a value (still trying to work out how to do that), but what about...

Skipped content of type multipart/alternative-------------- next part -------------- This email was transferred using an evaluation version of AXIGEN Mail Server.

...] Sent: Tuesday, 24 May 2005 6:13 PM To: Dean Collins Subject: Re: TellMe pay-as-you-go? Well, for changes to the wiki, you might mention that the service is not likely to be for personal use. It's not totally clear from the entry. Also, I disagree. There's absolutely no reason that VXML hosting couldn't be scaled to providing individual users low-volume service. It's merely a matter of having a sufficient pool of resources, and removing all of the management cost-per-user, or making it very very small and mostly the burden of the user. In principle, this is exactly what Vo...

...re: 1. What is the estimated simultaneous call capacity of a single server placing calls and playing a message and allowing the called party to request a call transfer either to a PSTN number or another VoIP address? 2. Is it possible to transfer calls to as VXML server and then retrieve the call when the VXML browser hangs up? Similar to a recall type action. 3. What is the maximum number of conference participants possible in a single conference? Would it be best to have a separate * server for conferences. The conferences would be a mi...

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

...the list which actualy show the URI string in the debug message (at the To: field). Is there any setting I need to set or turn on during compilation of asterisk? I have the head version of asterisk and my extension.conf setting is proveded below: exten => 777,1,Answer exten => 777,2,SetVar(VXML_URL=play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml) exten => 777,3,Dial(SIP/1234@192.168.1.72,10,t) exten => 777,4,Hangup SIP Debug message: *CLI> dial 777 -- Executing Answer("OSS/dsp", "") in new stack << Console call has been answered >> --...

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

...ial number and variables from the call file request. 4. Pass that parameters to voiceglue 5. Catch the parameters (dialnumber and cli) in voiceglue.conf 6. Point to the voice.xml file dynamically by matching number I want to make the line in voiceglue.conf as <DEST_NUMBER> http://127.0.0.1/vxml/<FILE>.vxml so that I can catch parameters and call the file dynamically. Please advice me. Best Regards, Roy -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140629/115bc5d9/attachment.html&g...

...skype.com/partners/voice/> to do exactly what I was proposing. This was just plain dumb because although skype are set up to handle the front end billing (this is one of the main issues with Tellme setting up a SIP gateway) no real developer is going to be happy using a skype channel for their Vxml application. Of course they've basically made nothing out of it and the idea for a sip gateway is still sitting there unrealised BUT... thems the breaks. Besides they get to work for MS now and become zillionaires. Oh and in case anyone out there wants to tell me there is no money to be made...

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

...a current opportunity for a R&D Jr developer. The right candidate will have a background in developing and managing Linux based software systems, some experience in the IVR industry is a huge plus. Expertise in the following areas is a must: * VoIP: Asterisk * Languages: Ruby, C, PHP, MySQL, VXML * OS: GNU/Linux (Debian preferred) Position Requirements: * Analyze, design, develop, test, deploy and maintain key IVR components * Complete development/programming as described in design specifications * Create and document technical designs to achieve project requirements * Maintain integrity o...

Not open source, but free for 2 ports: Prophecy2006 by Voxeo. Supports CCXML, CallXML and VXML. Interops nicely w/ asterisk. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Neil Skowronek Sent: Wednesday, March 29, 2006 11:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subjec...

Hello asterisk users! Does asterisk (or any of its plugin) support MRCP to interface with 3rd party professional TTS engine? I've found very little information about this, for example: http://www.zen-turkey.com/blog/default.aspx?id=12&t=Asterisk-MRCP-Integr ation Thanks in advance! Regards, Claudio Internet Email Confidentiality Footer

Is anyone on this list using Voxeo hosted vxml application services? Are they really doing $24 million dollars worth of business in hosted services per annum? I thought with that kind of volume I would have heard them being mentioned more often. Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed.....