similar to: Asterisk Cepstral TTS

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 have installed cepstral .... It works woderfull using an agi script but ..... when i try to use Swift("say this") is Dial plan .... I get the error [Aug 26 12:30:18] WARNING[7420]: pbx.c:3167 pbx_extension_helper: No application 'Swift' for extension (actdemo, 123, 2) Now i come to know to install app_swift Here is the issue... when i try to execute make command

Hi Shakeel, I had the same problem building app_swift (1.6..) myself and searched the web far-and-wide for a solution. I eventually contacted Darren Sessions -- who was maintaining that plug-in -- about a month ago. He was involved in another project and said he might be able get to it after a few weeks. But, since then, his website http://www.darrensessions.com/ has gone out of comission. I

All, Has anyone seen this before? This appears to be a Swift or app_swift bug. I'm having a difficult time finding any information or support on this. Asterisk version: Asterisk 11.6-cert4 built by asterisk @ ivrd02 on a x86_64 running Linux on 2014-08-11 13:55:25 UTC OS: Linux livrp03 2.6.32-431.11.2.el6.x86_64 #1 SMP Mon Mar 3 13:32:45 EST 2014 x86_64 x86_64 x86_64 GNU/Linux When

We are having some major problems with app_swift since we went live. It is regularly segfaulting. I don't know if this is my fault or not, but here's the story: Installed the cepstral voices (at the time, 4.0) on our test system (2.6.9-42.0.10.ELsmp) and later added some extra voices (now 4.2). All worked fine - we stress tested (20+ simultaneous calls). Move to live (

Hi, How can I integrate Asterisk to Nuance TTS engine instead of Cepstral? Has anybody done this? How is the architecture and can Java AGI be used to communicate between them? regards, Vela Sivasankaran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091021/56254a0e/attachment.htm

Hi, For an organization welcoming turists (in France), I would be curious to learn about successful use (with Asterisk) of Text-To-Speech in spanish (and english). I took a look at Cepstral's web site and saw there 2 "Americas Spanish" voices (along a bunch of english voices). 1. In this context, according to your experience, is it acceptable to use an "Americas Spanish"

Hi, Has anybody seen a non commercial, or freeware, or GPL, or even CHEAP... POP/IMAP to Text-to-speech? I have a working version for POP3 using festival. It DOES work... it even cleans the email contents to get the actual content. It works great with Outlook emails and similar, and skips non multipart/alternative (that would be mainly SPAM, where the email is just

Not being very happy with festival I would like ro get a better TTS engine. I looked at the listings at: http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international but I would like to get user input on suggested packages for Linux. Best performance vs. cost ???? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * *

Greetings, I've recently upgraded from Asterisk 1.2 to 1.4. I've been searching for a solution, but am also trying the easy way at the same time. I've now got David of Cepstral now speaking using app_swift from http://www.mezzo.net/asterisk/app_swift.html . The problem is, he sounds way worse than he did when it was asterisk version 1.2. I'm seriously considering either

I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I am trying to get cepstral to work with Asterisk for Text to Speech purposes. I got Asterisk to recompile with cepstral and after doing *CLI> show cepstral female voices Available Cepstral voices: Name Gender Age Language Sample Rate License ========== ========= ===== ==================== ============= ============ Amy female 30

Hey everyone, I was frustrated with the existing app_cepstral/app_swift TTS modules I've found on the net, so I hacked up my own. It's been working really well for me so I thought I'd share. In developing this, I wanted to avoid: * the startup delay incurred writing TTS output to a temp file then playing it back * gluing cepstral/swift to the asterisk module in a way that fubard

How do you get Cepstral working, they only offer windows versions. do I have to complie it to linux? http://www.cepstral.com

I just read about the partnership but was wondering what is actually going to happen? Is asterisk going to be bundled with cepstral voices for free :)? Or whats the deal?

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 I have recently configured TDM400P with four FXO ports. My next requirement is to configure for E1 line. which contain 30 phone lines and 2 for signalling information. The problem is I dont want to go for E1 line directly .....Is it possible to get simulation for E1 line ... so that i can develop a system for an E1 line. -- Best Regards Shakeel Abbas

Hello I am trying to configure TE420P but i am confused what to give chan_dahdi :( Below is configuration i am using for TDM400P Please help what changes to make in it... Please provide a link as well [trunkgroups] [channels] ;default for channels switchtype=national rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Hello I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection ....now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. this is a requirement .. Is it possible ?? what is the way forward ?? please give me a

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is