similar to: SAPI (Speech Synthesis API)

Hi, I upgraded my kernel from 2.4.23 to 2.6.19.2 and Wine still works fine for most apps, but it won't run Bider Speech Plus anymore. It pops up the splash screen, says "Initializing TTS system" for a while, and then says wine: Unhandled page fault on read access to 0x00000000 at address 0x403c67 (thread 0009), starting debugger... Usage: winedbg [ [ --gdb ] [ prog-name [

> Has anyone had any luck in changing the voices for Festival and > Asterisk? > > I have Festival installed and working, but can not get the voice > different > from the default. > > Thanks, > > Jason > Jason-- Assuming you follow the installation instructions, and install the Mbrola and other goodies for all the possible different voices, then you can,

Hello Asterisk, noticed that the mbrola really adds a extra dimension to tts; anyone got any experience with running this together with the festival and * ? -- Best regards, Danny mailto:dannyz@belgonet.com belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 -

Just and FYI. In RPM form, works on 4.x and I will try on 5.x too. Speech is tacky, but if I can get something but the Gnome Festival synthesizer, maybe it will improve? http://www.adobe.com/products/acrobat/readstep2.html?promoid=BONRM -- Bill

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Hello-- In the interests of playing around and wasting time, I've installed the latest version of the Festival stuff, 1.95beta. And, in the interests of future Asterisk-Festival connectivity, I applied the 1.4.3 patch to put in the asterisk related routines. I did it by hand, but, it looks like the patch will apply with no comment. Asterisk works with the new server... BUT the speed of

Hello List, I just turned up a T1 with Qwest. All the channels came up and inbound and outbound work and sound great. The only issue I have is this message "Handling message for SAPI/TEI=0/0" repeated on the console a couple times a second. I tried googling it but only found links to code in libpri. I am running Asterisk SVN-trunk-r76015M. I am going to upgrade to the latest

Hi, in the earlier version there was a sounds.txt with the transcript of the soundfiles. Does this still exist somewhere? Is there a plan to make speech synthesis available the same way as soundfiles, ie. instead of playing language/soundfile.wav, send the text to the speechengine and play the output...? Jay... -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello, I have a Junghanns QuadBRI card in my asterisk server. I'm able to dial & connect to local numbers through the ZAP interfaces however when I try to dial national numbers with the according area code the connection fails, an intense BRI debug is shown below. The error being Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) about half way through the

Many of you are familiar with how lousy Festival sounds. AT&T has a product, NaturalVoices, that sounds much better. There are male & female voice fonts for US/UK/Indian English, French, Spanish, and German. I am considering offering a linux-based text-to-speech engine based on the NaturalVoices runtime. An asterisk module would also be provided, making it easy to add natural sounding

Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net I've posted my demp weather report Asterisk AGI script at http://www.fnords.org/~eric/asterisk/downloads/ I have no affiliation with Cepstral. Below is the README: Contact: Eric Wieling <eric@fnords.org> If you want a demo of this AGI script you may call via IAXtel 1-700-923-3645 extension 2101. Option 23 is

Hi All, I am currently working to intergrate the MS SAPI engine into Ubuntu Linux, I want to harness its TTS and Speech recognition features. At the moment i have managed to get a version working in wine but only by using an older version that was included in Gutsy to install the package and then migrate it into the newer version of wine. I copied reg files and drive_c to a new 1.1.9 version

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:

The Wine development release 2.18 is now available. What's new in this release (see below for details): - Radial gradients in Direct2D. - Asynchronous message support in WebServices. - Virtual memory write watches improvements. - FreeType 2.8.1 compatibility fixes. - Various bug fixes. The source is available from the following locations:

Our application is 32-bit big-endian ARM and we use -O3 with LTO. clang optimizes certain initialization of structures to zero with calls to memset, which are not further lowered to move instructions. Investigating perf reports, it looks like it may be beneficial to disable this optimization that introduces a function call to memset in certain hot paths. I tried passing -fno-builtin, but that

I have done research and reading around text to speech, and was wanting to get an updated query of where everyone is with this. I have installed festival 1.4 on CentOS 3.5 ( system was installed using Asterisk at Home ISO ). I also changed the directory php application to use the festival.pl to read the names of those who have not installed a greeting. It works well enough, though the only voices

Hello, I try to synthesize harmonic functions with a subset of frequencies determined by fast fourier transform. I wrote two different functions, a looped version and a matrix multiplication version. As an example the looped version takes 5 sec and the matrix-version takes 3-4 sec (R 1.4, Athlon 1.2 GHz, 256 MB, Win ME), but the latter needs huge amount of memory due to the matrices. So, none of

I have a working system with asterisk 1.4.26.2 libpri 1.4.1 and zaptel 1.4.12.1 With a digium TE205p. I am trying to update to libpri 1.4.10.2. When I do, incoming calls work but outgoing does not. When I do this I "rm /usr/lib/libpri*" then just install libpri-1.4.10.2 as normal. I then do a make clean in asterisk and make distclean ,then configure, make and make install. I do

I have an application where I need to send outbound prerecorded messages. The Asterisk "call file" process works fine if I am sending the call via SIP or IAX, but not via ZAP over a PRI channel. The destination device (my cell phone) never rings. The only unusual thing I see is on the fifth line of the "full" log, below, "channel.c: Don't know what to do with