similar to: System Application with java

I currently have an AGI that calls the Festival text2wave app to write a wav file that my dialplan plays into a call with the Background() command. But the voice sounds terrible: like SAM, the 1980s 6502 voice synthesizer. I tried to slow it down by calling (text2wav -eval "(Parameter.set 'Duration_Stretch 1.4)" -scale 2.0 [...]), but it still sounds like it's talking while

At time to time I receive some junk faxes from some advertising companies that play smart and don't provide any TSI number so I can not bock them by the number in Hylafax. Despite calling their Fax Removal Service 1-800-... number several time they refuse to obey my request. So I would like to setup a small script or context loop in extension.conf if possible and maybe run it overnight; maybe

Hi, Sorry for my bad english but I?ll try to explain my problem I got an Asterisk running in my house with ADSL... I?m using X100P and TDM400P cards.... My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here?s the map with Firewalls Call for anyone to my house => PSTN => X100P => EXTENSIONS => SIP/RTP => ISA MICROSOFT

Hi, I?m trying to make * work with Brazilian analog signalling.. I?m using the following in indications.conf file... [br] description = Brasil ringcadence = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 callwaiting = 425/60,0/250,425/60,0/5000 I changed zaptel.conf to loadzone=br #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no

I have this code which was taken from the phpagi project page along with the following in extensions_conf and the output from the asterisk CLI. When I call the 311 extension, I does nothing then hangs up. What am I doing wrong?? ----php code------------ #!/usr/local/bin/php -q <?php set_time_limit(30); require('phpagi.php'); $agi = new AGI(); $agi->answer(); $cid =

Does the System() dialplan application have a limit on how long it can run? Either a time limit, or server load limit? I'm trying to pipe the output of Sphinx2 into Text2Wave, but Asterisk just runs by it to the next extension priority, with no errors. If I run the same command via the system shell, all is good, though it does take a few seconds, probably about 5 seconds to run. Yes,

Hi, I'm banging my head with Perl AGI script to produce wav file to be streamed with Festival's text2wave utility. I have two lines of code: ------------------------------------ system('echo "' . $text . '" | text2wave -scale 1.5 -F 8000 -o ' . $filename . ' - '); system('echo "' . $text . '" | text2wave -eval

Can anyone suggest a shell script in linux bash for encodimg raw audio output from text to speech app into an ogg file. I am running festival speech. The command line for basic tts is festival> bin/festival --tts myfile.txt Thanks -- Raena Lea-Shannon --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this

Hi I have installed Festival on the same box as asterisk and followed the instructions to integrate it with asterisk. Festival seems to work fine on its own performing text to speech from the command line or via a file. Asterisk answers the call but there is no speech. I can see no errors in the Festival log file The asterisk console shows --Executing Answer("SIP/81801-c091",

Hi, I am need of "voice to text" converter for one of my project. I need to use it on Linux OS. I got "text2wave" converter but didnt find any "wave2text" equivalent. Then I saw "speexenc" and "speexdec" on linux. It basically does encoding and decoding the the WAV file. Thus I wanted to know if I can use these utilities to convert a WAV file

Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that

Hi Keith, Thanks a lot for your reply. Actually I am trying to build an application in linux & C for converting a voice file(that is WAV) to a text message. This is required for one of our voice project. So just wanted to know if there is any existing application already there in linux to do that. Otherwise I have to write my own. Now to right my own "wav to text" converter I

I'm trying to capture faxes on 1.6.1-beta4. AFAICT, app_fax is working OK. I'm then using fax2mail to send the fax. That wasn't working, so i posted for help using the System() cmd, since fax2mail did work from the command line. But now I realize it's fax2mail and mime-construct itself. I set up a fax-test context: [fax-test] exten=>666,1,NoOp( fax-test )

Hi, Does anybody know where I can find any open source ITU G.107 implementation available? I'm looking a way to measure the voice quality in my projects.. Thanks in Advanced, My Best Regards, Andre Lomonaco -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, i want to to this thing with php AGI: #!/usr/local/bin/php -q <?php set_time_limit(30); require('phpagi.php'); error_reporting(E_ALL); $agi = new AGI(); $agi->answer(); $cid = $agi->parse_callerid(); $agi->text2wav("Hello, {$cid['name']}."); $agi->text2wav('Enter some numbers and then press the pound key. Press 1 1 1 followed by the pound key

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

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

Is there any way to use Festival from script called by the ExternalIVR() dialplan function? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david@safedatausa.com

Can you also post information such as: Type of phone (model Number would be idela) How is it conencted, SIP, ZAP, IAX, Channel Bank. Corresponding config files would also help. Help us help you. >>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >>Paul A Brown >>Sent:

Hi all, I am trying to use a "Siemens optiPoint 300" IPPhone (H.323 only) with Asterisk (1.0-RC2). So far I have been using the H.323 channel included in the tarball (Nufone ?). I encountered a strange behaviour when I try to make a call from the IPPhone to my Asterisk box : =====> here is the H.323 configuration for the incoming calls (192.168.1.50 is the IP of the