similar to: Error running festival command

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

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

i have problem with dial-plan in php. I have 3 extention in dial plan, 555,551 and 551. the problem is that READ PIN||3 works in 555 and hangs in other extentions. (after timeout asterisk writes thar "user entered nothing"). i can't get what's wrong... here is my dial-plan [incoming] exten => 555,1,Answer(); exten => 555,2,DigitTimeout(2); exten => 555,3,Wait(1) exten

In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID "1000" and line 2 as SIP ID "2000". Basically I have this set up so that 1000 and 2000 are "lines in hunting" on incoming extension "555". I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here

hello, I got those extensions: exten => 555,1,MeetMeCount(500|count) exten => 555,2,Gotoif,$[${count} = 1]?6 exten => 555,3,Meetme,500|pMs|1234 exten => 555,4,Playback,goodbye exten => 555,5,Hangup exten => 555,6,Goto(from-internal-custom,556,1) exten => 555,7,hangup exten => 556,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/) exten =>

Hello, I am trying to run festival, it is running but I am getting this when I run tts_ping.agi WARNING[278546]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Enter the eye-p address you wish to ping. WARNING[278546]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... WARNING[278546]: File app_festival.c, Line 400 (festival_exec):

I have loaded festival-1.4.3 patched with the 1.4.3.diff file. Festival source is in /usr/src/festival dir. When I try to use it I get this from asterisk: -- Executing Answer("SIP/chad-57a4", "") in new stack -- Executing Festival("SIP/chad-57a4", ""I am talking"") in new stack == Parsing '/etc/asterisk/festival.conf': Found

The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension number&password could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there

Greetings and salutations to all... I'm having a bit of a problem getting a SIP phone (Xten) to call an H323 Cisco ATA-186. Both devices can call into the * and get the demo, voicemail, etc... I'm pretty sure my problem is in my configs as it feels like a stupid error and to prove this to myself I set tcpdump on the * box to capture all UDP traffic going to and from the ATA-186. If I

I just installed Festival 1.9.5 and all dependencies. It works properly from the command line, as I can play the contents of a text string or file from the Festival command line through PC speakers. I then added the code to the festival.scm file, as detailed in Method 1 of http://www.voip-info.org/wiki-Asterisk+Festival+installation I did not apply the patch used in Method 2, because I can not

Hello, I'm in the process of trying to get Festival to work with Asterisk. I followed the install process at http://www.voip-info.org/wiki-Asterisk+festival+installation. To get the Festival to compile I had to add the patch described in the comments. Once added, Festival and the Speech tools compiled without error. How ever, when ever I try to call the test extension, I get a busy

I have used this Asterisk at Home for a while now, 2 years. Only recently, I am trying Festival and on invoking festival --server I get these errors : /usr/share/festival/bin/festival: /usr/lib/libstdc++.so.5: version `CXXABI_1.2' not found (required by /usr/share/festival/bin/festival) /usr/share/festival/bin/festival: /usr/lib/libstdc++.so.5: version `GLIBCPP_3.2' not found (required by

I am using Festival to synthesize some menu Interaction with a caller and am having a problem. What I am working on is a remote callback where I can remotely call in to an extension, and enter a callback number (or use the CALLERID info) and a second outbound dialing number to connect to. Things work O.K. until I set usecache=yes in festival.conf. After doing this, things run well for

Do you think this should be "Bug Reported"? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Samaritoni Sent: Wednesday, October 06, 2004 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Festival, getting gethostbynamefailed error

I'm having trouble getting festival to work with asterisk. We are running debian (sarge) and got asterisk from CVS. Here's what I'm using as far as festival goes. Debian (Sarge) gcc version 3.3.4 (Debian 1:3.3.4-3) Connected to Asterisk CVS-HEAD-07/28/04-21:08:19 festival-1.4.3-release.tar.gz speech-tools_1.2.3.orig.tar.gz I got patches for both of these. Speech tools

Hello all! I googled lists.digium.com and ser mailing list, but did not find any working configuration of asterisk used as voicemail for SER. This is my config if (uri==myself) { if (method=="REGISTER") { save("location"); log (1, "Registered\n"); break; };

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

Hello, Suppose a dialplan such as: exten => 6004,1,Answer exten => 6004,n,Wait(1) exten => 6004,n,SayDigits(1) exten => 6004,n,Festival(This is a test of Festival) exten => 6004,n,Hangup When watching in the CLI, I see this: == Using SIP RTP CoS mark 5 -- Executing [6004 at internal:1] Answer("SIP/505-00000004", "") in new stack -- Executing [6004 at

Interestingly enough I had this same problem today.... 1. I created the directory and permissions for the directory " /var/lib/asterisk/festivalcache/ " (per the comment in the festival.conf file) 2. I had to comment out some things in the festival.conf file: the "host" line, the "port" line, and the "festivalcommand" line. I have also noticed the

Hi All, I have installed festival for work with asterisk, this work fine, but when I ran the festival server for languages spanish, in my PBX always I listen the english voice, in my extension.conf I have the following example: ; testing festival (text-to-speech app) exten => 556,1,Set(LANGUAGE()=es) exten => 556,2,Answer exten => 556,3,Festival(Es una prueba) exten => 556,4,Hangup