similar to: Announcing free (GPL) VXML for Asterisk - Voiceglue

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

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, 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 with path in asterisk server, are there any way to do this? --

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

Hello, I recently purchased the Cepstral 6 text-to-speech engine (swift), and am now wondering if I should have bought something else. I would like to use Cepstral text to speech like some people use the Festival() or Flite() applications. For example, when I do a "core show application flite" at the CLI, flite is available to me: localhost*CLI> core show application flite

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. --------------

Hello I found following issue while trying to load flite module from CLI module load app_flite.so Unable to load module app_flite.so Command 'module load app_flite.so ' failed. [Jun 19 11:31:12] WARNING[23507]: loader.c:384 load_dynamic_module: Module 'app_flite.so' did not register itself during load [Jun 19 11:31:12] WARNING[23507]: loader.c:662 load_resource: Module

Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the

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 have two questions ! 1. What is the best speech recognition engine for asterisk? I have searched and asked on forums and found that lumen vox is best for asterisk bala bla bla 2. For TTS (text to speech) which TTS engine will be better to use ? I have tested Flite , cepstral (i have not buyed lisence for it trial only) but still thinking may be i have a good option ? -- Best Regards

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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]

Version 2.1 of app-espeak and app-flite modules just got released. Changes include voice selection for flite, support for 16kHz file playback in asterisk 1.6.x, use of libsamplerate for faster resampling, better error handling and cleaner code. You can get them here: Asterisk espeak module: http://zaf.github.com/Asterisk-eSpeak/ Asterisk flite module: http://zaf.github.com/Asterisk-Flite/

Hi, After following the messages to install flite, I can find the following files. /usr/lib/asterisk/modules/app_flite.so /etc/asterisk/flite.conf That's mean flite is installed successfully. Then I restart asterisk but nothing found for that module. sip*CLI> core show application flite Your application(s) is (are) not registered The module doesn't load by asterisk even after

I am learning agi scripting using php. I m using phpagi 2.x on asterisk 1.2. I hve written a simple script that reads out the callerid using flite. My problem is that I seems the script is not getting the callerID. Bellow is the script _________________ #!/usr/bin/php -q <?php /** * @package phpAGI_examples * @version 2.0 */ set_time_limit(30);

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

I have a similar situation here, I want to include TTS in my asterisk IVR system. Could someone give suggestion(s) please, I prefer open-source thanks in advance! Chatila, A. C. P. O. Box 365, Kihesa Street, Njombe, Tanzania. *Mob:* +255 765 154 235 *Whatsapp:* +255 653 258 608 *Website:* chax.me.tz On Thu, Aug 27, 2015 at 9:07 PM, Steve Edwards <asterisk.org at sedwards.com> wrote:

On 07/05/2015 09:17 AM, linush at verizon.net wrote: > Someone please tell me what I did to screw this thing up so badly. On 07/05/15, Gordon Messmer<gordon.messmer at gmail.com> wrote: Have you looked at the log files in /mnt/sysimage/root/? ------------- Quoting broken in this mailer ------------ So I looked in /mnt/sysimage/var/log/anaconda and found this in anaconda.packaging.log:

During Astricon 2004, we'll have the first Asterisk developer's meeting. The Asterisk developer's meeting is a one day meeting with discussions, brainstorms and tests. For each session, we need a white paper produced that outlines the topic to be discussed. If controversial, several whitepapers on the same topic will be contributed. This is a list of suggested topics - please feel