similar to: Synchronous dialplan execution for feedback while processing speech recognition and voice synth, for example.

I am lost: > plot(testError, col="red") > lines(testVar, col="black") Only one plot (the red one) appear on the Window, the black line did not appear...what's wrong? Thanks a lot! [[alternative HTML version deleted]]

Here's what I'm doing: 1. I set __FOO=0 in the current context. 2. I use Dial() and the M(testvar) macro call. 3. The macro-testvar can see ${FOO} just fine. 4. the macro-testvar modifies FOO. I tried: Set(__FOO=1) Set(_FOO=1) Set(FOO=1) 5. When the Dial()ed call hangs up, and the original context continues, FOO is still 0.

Hi Arnaud, Hi Scott, Hi list, Here is a patch to support the Zinto D from ONLINE USV-Systeme AG. I already sent a version to Russell Kroll (2006-04-09), without no response and I cannot find support for Zinto in svn until now. I found a discussion on this list about the Xanto from ONLINE, but the Zinto seems to use different commands. The commands are quite similar to those for Opti-UPS, so I

Hi all, I''m getting an error and I really don''t know why, because I seem to be doing everything well..I pass "testvar" as a variable to render partial: <%= render(:partial => "main/categorysmallitem", :collection => group_of_four, :locals => {:testvar=>"AA"})%> But then, when I try to use testvar inside the partial, it

Hi Can anyone tell me what''s wrong with this one. Sorry I''m new and I just want to print some message to the the puppet log on the client side. --------------------------------------------------- $testvar = "Hello world!" class mytest { exec { "/bin/echo $testvar" } } I keep getting: err: Could not parse; using old configuration: Syntax error at

Related to http://lists.digium.com/pipermail/asterisk-users/2016-November/290384.html, at the moment I'm passing one variable via DIAL. Now I'd like to pass a whole bunch, and my idea was to rather than having a great string of b(synctest3b^setVar^1(something)^2(more things)^3(etc)) and then get them with ARG1..ARGn etc, I could bundle the whole lot into a HASH and then unbundle them at

Hi there, I try to understand the usage of environments but I'm not sure if I get it. I wrote a test script like this: testenv <- new.env(environment()) myfun <- function(x) { print(testvar) testenv$testvar_2 <- 20 } environment(myfun) <- testenv testenv$testvar <- 10 myfun("hello") ls(envir = testenv) Now, I was wondering if there is any way to create new

Hi, Greatly appreciate if some one help me with the application read(). asterisk*CLI> show application read asterisk*CLI> -= Info about application 'Read' =- [Synopsis]: Read a variable [Description]: Read(variable[|filename]): Reads a '#' terminated string of digits from the user, optionally playing a given filename first. Returns -1 on hangup or error and 0

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:

Hi, we've proof of conecpt system for speech recognition on Asterisk. We would like to send results of recognition back to user in standard way. Currently we're considering using sendtext command and it works with Firefly. But I'm curious what soft or hard ip phones that can connect to Asterisk support such feature ? Also what softphone would be most suitable for further work in

Does anyone have experience with this product?

I'm interested in taking a persons spoken recorded name (First, Last) and converting the two spoken words to text. Is there any solutions out there that would make this possible? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090130/1e90ba97/attachment.htm

This week Steve Sokol stops by to describe and field questions about Digium's new affordable speech recognition solution. Later on in the call, we'll also be looking at iVoIP, clients and uses for mobile VoIP. Join us on IRC anytime #voip-users-conference During the conference, call via SIP g711 or wideband g722 - or Try the web page widget to call in wideband. The details on all the

Hello! I need to: 1) call special number (or run special application) on mobile phone 2) establish connection between mobile phone and server 3) allow server to recognize spoken numbers (Polish language) and some other control words 4) let the server to say some short answers (prerecorded in mp3) according to some algorithm and recognized words 5) let the server to save little text file on its

Thanks Jurijs, Yes, in fact I'm already using that, and it works fine. The problem here is that I cannot find a way of recording speech AND listening for a DTMF digit being pressed as an alternative. That's where the problem lies. J.

Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS? Could anyone provide pros/cons for the various ASR options for Asterisk? We need the ability for very large grammars (over 100,000 options). Because of this, my initial thought is Nuance or Lumenvox. Does this sound correct? Have a great day! Dan -------------- next part -------------- An HTML attachment was

Hi, Can anyone tell me how can I achieve this: Say if I have a e-commerce site, I want to list all the products when user type in www.mysite.com Not www.mysite.com/products Please note there is no controller in the URL, how do I replace the default Rails index.html and display my products page. Hope my question makes sense. -- Best regards, Ming Ma

Hi Everyone, Has anyone got any opensource speech recognition software to work with Asterisk? Please only list WORKING ones. Not the "theoretically" should work ones! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100821/4d11d6c0/attachment.htm

All; Since I have interest in providing the capability for callers to speak the department, person or number they wish to call, as well as other IVR scenarios, I have been reviewing much of this lists email archives and searching the web for open source voice recognition that will work with the Asterisk PBX. What I am trying to determine, is what will it take to get it working on Asterisk? How

Thanks for your responses - it looks like I have the following options, in order of ease: 1: Modify and recompile app_record.c Change line 471 https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L471 from status_response = "DTMF"; to status_response = dtmf_integer; Pro: Free, easy Con: Have to remember to edit module each time a new Asterisk update comes out 2: