similar to: espeak app for asterisk 1.6

Hi, I'm giving a look at [1] with this: cd /tmp git clone https://github.com/zaf/Asterisk-eSpeak.git cd Asterisk-eSpeak ln -s <path-to-asterisk-folder>/include/asterisk.h ln -s <path-to-asterisk-folder>/include/asterisk make I'm getting this: gcc -pipe -fPIC -Wall -Wextra -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o

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:

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 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 written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first

Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file:

Hi, I recently discovered http://zaf.github.com/asterisk-mstts/ . In the page above, it is mentioned you have to subscribe to Microsoft Translator API on Azure Marketplace. In Azure Marketplace, I found something called "Microsoft Translator". This API is free within a 2 000 000 characters per onth limit. Is this the API needed for MS TTS ? If not, where and how can I find the good

Hi, The fact that apparently the last tigervnc update from upstream was missed triggered me to check for missing updates and packages in 6.3. Here are my results. Sorry for any false positives that might have crept in, but note that some of the 6_x updates actually are updates and not a parsing error. And perhaps an occasional false positive due to having to compare upstream SRPMS vs downstream

CentOS Errata and Bugfix Advisory 2012:1118 Upstream details at : http://rhn.redhat.com/errata/RHBA-2012-1118.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) i386: a421c7cd4b3ff004887dc4da9e1cb1f1cf89334c1987bb712644cad2e7ae8d88 espeak-1.40.02-4.el6_3.i686.rpm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, First, as far as I can tell, this topic hasn't been mentioned in the mailing lists in awhile (atleast not as far as google can tell). Is there any client/program that can send "net start" requests to windows boxes. The ability to start a program as a service on a windows box remotely would obviously be very handy. I am aware of

For reasons best known to myself, I call a python agi (PYST2 - love it!) which streams a series of very short files in quick succession. Like this: escape_digits = str("0") agi.stream_file(promptFile,escape_digits) and this is what I see on the AGI debug: <Local/s at root-00000061;2>AGI Tx >> 200 result=0 endpos=6784 <Local/s at root-00000061;2>AGI Rx <<

On Tue, 2016-02-23 at 17:06 +0000, Steve Howes wrote: > Google?... Yeah... searched "google voice recognition api asterisk", browsed though various results. Nothing helpful for a beginner, very confusing bla bla... Thanks anyway for your help. F.

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

Hi Marcelo, Thanks for replying, I do not know what this branch is. Could you please let me know. Also, I enabled google cloud speech API only from the console. Do I need more API enabled? On Wed, Jul 19, 2017 at 3:41 PM, Marcelo Terres <mhterres at gmail.com> wrote: > Did you already tried the cloud_api branch? > > Regards, > > Marcelo H. Terres <mhterres at

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

Hi, I'm trying to get a specific program to work under Wine. It is NextUp TextAloud. It gives some error when I try to run it. I've also tried to install the Windows version of espeak and it won't work under Wine either. So I think there is something different that the text to speech programs use that Wine doesn't like. Does anyone have any ideas of how to get this to work?

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

Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -------------- next part -------------- An HTML attachment was

I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: <PJSIP/2001-0000006e>AGI Tx >> 200 result=99981 (timeout) endpos=22720 <PJSIP/2001-0000006e>AGI Rx << VERBOSE "Unable to get recognition

Send CentOS-announce mailing list submissions to centos-announce at centos.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.centos.org/mailman/listinfo/centos-announce or, via email, send a message with subject or body 'help' to centos-announce-request at centos.org You can reach the person managing the list at centos-announce-owner at centos.org When