similar to: Text to speech modules (espeak, flite)

I have written a module for asterisk that uses the eSpeak speech synthesizer (http://espeak.sourceforge.net/) to render text to speech. The source is available here: http://zaf.github.com/Asterisk-eSpeak/ It's similar to app_festival and app_flite. It's only tested against asterisk 1.6.1 on x86 Linux but it must be working for other 1.6 branches too. Comments, fixes and suggestion are

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:

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:

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

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

Hi, a set of AGI scripts that provide ASR and TTS for asterisk using the iSpeech API (http://www.ispeech.org/) are available on this page: http://zaf.github.io/asterisk-ispeech/ This is the first public release, updates will soon follow. Feel free to test and report. Regards, Lefteris Zafiris -------------- next part -------------- An HTML attachment was scrubbed... URL:

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:

Hello, FastAGI doesn't get (or deserve) much love these days but a lot of people are still widely using it. Here is a small project of mine, started trying to scratch my own itch, that some might find useful. https://github.com/zaf/agitator It is a reverse proxy for the FastAGI protocol with some interesting features. Most noticeably request based routing, HA/failover, load balancing and

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

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

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

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

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, 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

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 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

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 <<

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

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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