similar to: Changing the voice in Asterisk

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

I am using speex for encoding and decoding speech files in a speech synthesis program. I am using concatenative approach for speech synthesis using diphones (small speech chunks) as basic unit. Now in my application, the order of diphones to be joined after decoding is determined at runtime. Therefore I have to encode each diphone independent of others. I find out that when I encode diphones

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 -

Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan

Anybody has experirence with Festival + french voice ? I read "2.1.3 .--The French language Festival does not (at the moment) implement yet the French language but it's not difficult to make it work with the Mbrola synthesizer which offers high quality French voices." from the website http://www.pollock-nageoire.net/festival/festival-english003.html#toc5 but "not

Replace nospam with ssb22 to reply by email. Hi, Does anybody have pointers about using SAPI 5 speech synthesis engines (sometimes called a "SAPI 5 voices") under Wine? I've searched all over the Web but can't find any mention of it. Although English speech is well-supported in Linux, speech in languages like Chinese, Japanese and Korean is not. Commercial offerings are

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is

Here's part of the output of running an AGI file: -- Playing 'degrees' (escape_digits=) (sample_offset 0) -- Playing 'fahrenheit' (escape_digits=) (sample_offset 0) -- Playing 'wx/humidity' (escape_digits=) (sample_offset 0) -- <DAHDI/1-1> Playing 'digits/40.ulaw' (language 'en') -- <DAHDI/1-1> Playing

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

Hello, I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk 1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine with SIP channels without Festival. I have written following context in extension.conf: [connect-to-me] exten => s,1,Answer Exten => s,n,SayDigits(?1?) exten => s,n,Festival(hello john) exten => s,n,Hangup I use call files to

Hi, Is there IAX client in Applet JAVA which can be embeded in a web page ? Best regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031208/c388ef61/attachment.htm

FYI, I found a workaround with this. Festival w/ A@H comes with 4 voices. Here's a snippet from the siteinit.scm file: ;(set! voice_default 'voice_cmu_us_bdl_arctic_hts) (set! voice_default 'voice_cmu_us_slt_arctic_hts) ;(set! voice_default 'voice_cmu_us_jmk_arctic_hts) ;(set! voice_default 'voice_cmu_us_awb_arctic_hts) The default is to have the first line uncommented and

Hi, Andras, Thanks for the comments. Yes, I am aware of those issues. I probably should have been more accurate on my usage of terms. Actually in my project, the unit collection is a mixture of diphones and words. However seems to me, these synthesizer specific issue is irrelevant to my question about speex. As you said, i merely use speex as storage methods. All I ask for is to get the

Hi all, I need to simply use Asterisk to receive incoming calls in an IVR manner. It should authenticate users and read data from MySQL table that match their ID through Text-to-speech. I already have Asterisk 2.6 (Asterisk@home). I understand that I need to use Festival and AGI but do not know what to do exactly. Any help is appreciated. Thanks

Hello list, is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php Tofik Suleymanov

I'm using app_festival to speak some text to callers. I'm having two problems with this. The first is with IAX calls (I've not tried others) the first few seconds of the speech is garbled. The second problem I'm having is the the volume of the speech IS VERY LOUD. I tried putting the following in the siteinit.scm but it didn't seem to make any difference. (set!

I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say "trying" and then hangup... Sep 24

[At the risk of educating you about something you might already know] Natural speech in most human languages gradually changes from one phoneme to the next. Concatenating phonemes together from a fixed, prerecorded, unflexible set would give rise to abrupt changes between them (both in phoneme quality and in pitch), and thus make the resulting speech hard to understand and/or uncomfortable

Hello, Using asterisk-1.2.1 I am trying to convert my music-on-hold files from .wav to alaw: % sox moh.wav -r 8000 -c 1 moh.al resample -ql The file sounds fine when listened with: % sox mox.al -t ossdsp /dev/dsp But when listened through asterisk with an alaw SIP phone the sound is clicky and too fast. Did I forget something in my conversion command? -- ldm@apartia.fr