similar to: 7.1 install with Areca arc-1224

Hi Asterisk guys, Flite is not working with asterisk 1.6.2.17. Flite is working with asterisk 1.4. Please help me how to use it with asterisk 1.6 ....... Thanks in advance. ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hello, I am working with a php/agi example now and really don't like the way flight sounds...I am just using it like below. Is there a better voice app to use? Also, I am wanting the agi to hit a webservice so it will return an array, is it possible to have asterisk read the array and allow the user to go to next element, skip back, etc? Thanks! exten => 711,5,Flite("At the beep

Hi List, I am confuse about this feature. When we use Wait(20) in active call session then it's work. But when we use after hangup the call then Asterisk don't wait from define time. Ex:- [call_log] exten => 4368,1,Answer() exten => 4368,n,Flite("Welcome") exten => 4368,n,Set(__StartTime=${STRFTIME(${EPOCH},Asia/Calcutta,%Y-%m-%d %H:%M:%S)}) exten =>

I just got cepstal working fine in the dial plan using code like: exten => 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep enter your zip code.) The php script it calls is based on the nerdvittles weather one so it calls a webpage which prints to the screen, the nerdvittles code uses system to generate the .wav file then has the dial plan call it via: //php script $retcode2 =

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, We are hitting an error in the centos 7 installation with our kickstart configuration. =============================================================================== An unknown error has occurred =============================================================================== anaconda 21.48.22.56-1 exception report Traceback (most recent call first): File

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

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);

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:

CentOS Errata and Bugfix Advisory 2015:1561 Upstream details at : https://rhn.redhat.com/errata/RHBA-2015-1561.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 384fbae3d29416385e6b0966556a63abc6db41f79a24095db2e6b967214649ba python-blivet-0.61.0.27-1.el7_1.noarch.rpm Source:

CentOS Errata and Bugfix Advisory 2020:5032 Upstream details at : https://access.redhat.com/errata/RHBA-2020:5032 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: ae536957e133aeb89aaad21eb5caebf2070b33547c21e967a1d51cc8ab213be4 python-blivet-0.61.15.76-1.el7_9.noarch.rpm Source:

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

This is probably a newbie question: I'm trying to get the asterisk weather app at http://nerdvittles.com/index.php?p=136 installed on Centos 4.2 x86_64 SMP ( Linux localhost.localdomain 2.6.9-42.0.3.ELsmp #1 SMP Fri Oct 6 06:28:26 CDT 2006 x86_64 x86_64 x86_64 GNU/Linux ) rpm -ihv flite-1.3-1.aah.i386.rpm seems to complete successfully rpm -ihv app_flite-0.3-1.aah.i386.rpm seems to

*** This response is my personal opinion and may not reflect that of my employer. *** It seems recent updates to Anaconda and in particular Blivet in the CentOS 7 .iso are restricting things rather than expanding the options. We have a scenario where we are doing a CentOS 6 to CentOS 7 upgrade (replacement) using a backup partition to hold the install image. Over the years, as we added cloud

For those who are using SuSE: At last they've managed to create ready-to-run packages for openSUSE_11.1. They are there since a couple of hours... (For other versions it was allready available for some time on the OBS) /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/asterisk16-devel-1.6.0.6-82.2.x86_64.rpm

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk

Gang, I'm having this error pop up when I do a ForkCDR, and I'm not sure how to get around it. Here are a few log lines: Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing ForkCDR("Zap/49-1", "") in new stack Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a CDR The scenario occurs like this: I use a .call file to generate a call on