similar to: Vxml record voice parameter

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 continued evolution of

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

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

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 multiple values, such as with a form? Doug. --------------

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Hi Freinds, I am trying to do the following. 1. Accept the call from call ifle. 2, Answer it 3. Extract the dial 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>

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

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 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

Hi, I can't make outgoing calls via Broadvoice. I have tried each and every configuration that was posted to list previously. I am able to receive incoming calls fine. I get the following in asterisk console: ===================================================== asterisk*CLI> show version Asterisk CVS-HEAD-03/10/05-22:51:28 built by vicky@asterisk on a i686 running Linux

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

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

I have not been able to access the UJS site, http://www.ujs4rails.com/ for two days. Does that mean there are problems or that it has been incorporated into Rails core? In general, what do you think of using UJS instead of RJS? -- Posted via http://www.ruby-forum.com/. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google

Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of

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: -- Executing [s at VXML:1]

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. I am new to asterisk. I have an ivr application with asterisk and voiceglue. I make a call from asterisk (say to A) and when callee press a button voiceglue transfer the callee to another number (say to B). When I look cdr records the billsec between A and B always 0 and billsec with A shows the billsec for A and B. I am confused. Is there a reliable way to get the real call durations? Best

Hi, I'm trying to use SFU to set user IDs, shells, etc. but I can't work out the right magic - it always seems to use default template shell and home directory (but the UID seems correct) The global section of my smb4.conf looks like so: # Global parameters [global] log level = all:2 netbios name = GATEWAY2 realm = BEGER.COM.AU server role = active