similar to: asterick and festival...Help!

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

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

I am totally new to this project, I have been trying to set it up now for almost a whole day, and can not understand what is wrong. The following error appears when my phone tries to register: Oct 6 19:08:30 NOTICE[98310]: chan_sip.c:7519 handle_request: Registration from '<sip:2201@192.168.0.5;user=phone>' failed for '192.168.0.253' What exactly does this error mean,

I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say "trying" and then hangup... Sep 24

Hello, I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk 1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine with SIP channels without Festival. I have written following context in extension.conf: [connect-to-me] exten => s,1,Answer Exten => s,n,SayDigits(?1?) exten => s,n,Festival(hello john) exten => s,n,Hangup I use call files to

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

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 am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an AT&T conf using regular analog phones. I don't have a great grasp as to what Asterick is capable of, but my

I am having trouble locating webvmail on my * server. Is this a seprate porgram or does it come with *. I am running version asterick*CLI> show version Asterisk CVS-03/26/04-17:08:20 built by root@localhost.localdomain on a i686 running Linux asterick*CLI> Thanks Kurt __________________________________ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢

Hi All... Can Asterick detect distinctive ringing on a POTS line and answer with different configurations? Thanks...

Hi all, I have been worriyng and googling a lot but I can't find my mistake. I am trying to regiter an X-Lite Softphone to Asterisk, but I am getting a SIP/2.0 403 Forbidden response: SEND TIME: 10157385 SEND >> 10.100.249.12:5060 REGISTER sip:10.100.249.12 SIP/2.0 Via: SIP/2.0/UDP 10.100.249.86:5060;rport;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22 From: Tester

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

Folks, My question concerns the SIP Notify that is being sent to ... device. You can see it in the following line: Voicemail: 0/0 Shows no Voice mail but I did leave a voice mail at the extension. Any suggestion on what I should look for in my * setup. I am not worried about the 481 coming back for the other side yet. Once I get a handle on the Notify, I'll work on the 481.

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

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

I am running Asterisk CVS-HEAD-07/14/04-16:28:29 and noticed that when I send a subscribe I get back a 403. This used to work in an old version which I forgot to record before upgrading to the above version. Any suggestion? I can register fine with the * server. Sip read: SUBSCRIBE sip:2486@192.168.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK46F2668 From:

Patrick hi. Asterisk can do that, and you don't need VOIP lines. If you connect Asterisk to the net, and all employees have a VOIP phone (either hardware or software) then you're good to go. What do you need? To begin with, install linux on an old pc (well, not too old). Then go to voip-info.org and take a look at the Asterisk wiki. Everything you need is there. And of course, we're

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

I've got a fresh FC2 install and I'm trying to get the symlinks right according to the /usr/src/zaptel/README.Linux26 instructions. I've created two symlinks: /usr/src/linux-2.6 -> /usr/src/linux-2.6.5-1.358 /lib/modules/linux-2.6 -> /lib/modules/2.6.7-1.494.2.2 When I do a "make linux26", I get a million warnings and errors with the result being: make[2]: ***

Hey folks, So I'm trying to get Festival() working on 1.2.17. I'm trying to use app_festival: Here's the show dialplan output from that extension: '3378' => 1. Answer() [pbx_config] 2. Festival(Hello Asterisk caller. How is your day?) [pbx_config] 3. Playback(vm-goodbye) [pbx_config] 4. Hangup()