similar to: asterisk having problem in playing sounds

Hi all, I m in a trouble using festival voices in asterisk. I am not able to change the default male voice of festival. Although i downloaded the us1 female voice and it iw working good in festival's CLI but it is not coming when i am usinf Festival in asterisk. I changed the default-voice-priority list directive and set us1_mbrole as first entry and also changed voice in festival CLI. But

Hi, I am using asterisk 1.2.1 for building an ippbx for my setup. I am having problem in accessing advanced options option that comes on pressing 5. It says to leave a msg and asks for extension. Now when i dial the extension the IVR silently goes into the top IVR menu. And the asterisk console show the message: "No entry for <EXTEN> in voicemail config file" As i

I am testing asterisk-1.2.1-15 on RedHat 9(i386) for SIP-to-SIP call and i found a strange problem. When an extension gets a ring and it picks up the call a "tick" sound comes at start. This happens on both sides. I tried Xten's softphones and also hardphones. A thing which was common in both(soft/hard phones) was the selected codec. When i used g711ulaw on both soft/hard phones

HAllo! This is Lokesh. I want to use speex codec in my project for encryption in GSM phones.Is it possible to use the availabale API in SPEEX in eclipse,i.e java IDE tool.Or else do u hav already available compatible package for java.Eagerly waiting for your reply.Please suggest me how to use speex in my project.Desperately waiting for your reply.Thanks for your time. regards,

Hi All, I am porting the vorbis encoder to Symbian platform,(to support ogg recording) everything works fine for the emulator(PC), but then on the Hardware whenever I am trying to record it gives KErrOverFlow(buffer OverFlow). We get this error from the driver when the latency of the codec is too high. The main reason is that the hardware does not have the floatpoint unit and the

Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the

Hi As per the Linphone, Readme.arm I tried to compile the speex. -------------------------------------readme.arm-------------------------------------------------- ........... Cross compiling speex for ARM: ******************************** First you need to remove ogg headers from your build system to avoid a dirty conflict between your build machine binaries and the arm binaries. They

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

Asterisk v1.0 is running on RH 9. I installed festival RPM (festival-1.4.2-16.i386.rpm) and edited the festival.scm file to add: (define (tts_textasterisk string mode) "(tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe

This script was developed by Mark Johnson. All I did (Dean Collins) was type up the instructions and make it easy to understand. This Script will allow you to dial and extension number on your Asterisk@home V0.6 or later pabx and have it read you the weather for your city How it does this is; 1/ When you dial your extension it causes the weather.agi script to run 2/ This makes the program

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

I cannot get Festival to work with asterisk. I have the following: exten => 555,1,Answer exten => 555,2,Festival(mary has a little lamb) exten => 555,3,Hangup I get the following from asterisk: "Festival returned ER" and the festival logs shows the following: client(1) Fri Jul 16 15:35:54 2004 : disconnected client(2) Fri Jul 16 15:40:26 2004 : accepted from localhost

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

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

Hi All, I was able to insert some extensions in Mysql DB and use them successfully. In Mysql extensions table the priority column is of type tinyint and when I give 's' value for it, it is not accepting that value as it takes only tinyints. Please tell how can I make that column accept values like t,s,i and make it work with asterisk in realtime without any problem? If I change the type

Hi, I created a script named example2.sh which goal is read some text from my HP Service Desk using an application in java and send this text to the text2wave application for TTS. example2.sh java -Xbatch Example10 | text2wave -f 8000 -o /var/lib/asterisk/sounds/my-sd.wav When I execute the script in prompt, everything is ok, but when I use the system() command in my extensions.conf it isn?t

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,

I followed the directions I found in the list to a tee http://www.marko.net/asterisk/archives/0209/0389.html The server starts fine, but when I call the festival extension it gives me an OID error variable tts_textasterisk I have RH7.3 festival 1.4.2 speech_tools 1.2.2 patched it with the festival-1.4.2.diff located in the /usr/src/asterisk/ folder . When I patched it, the patch was looking

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?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing