similar to: Old sound in new call.

We find that mpg123 0.59r works best. mpg123 0.59s-mh4 = the devil. What versions does everyone use without problems. 0.59r is PERFECT bkw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040506/937ae19c/attachment.htm

Hello All, I wrote a PHP function to parse the icecast stats log and return as much information in a structured array as possible (well, as much as I care about, anyway). Because the stats log is not easily machine-parseable, I thought this might be useful to somebody. The log is parsed using Perl-compatible regexps, so it should easily port. I have only my own setup to test this on, so

Hello, I have a problem with dovecot. seems that do not erase mail that mail client request to be erased. And I have this errors: > Error: Next message unexpectedly corrupted in mbox file Info: > dovecot-2.1.1-2.0.cf.fc16.i686 > root 5979 0.0 0.1 3208 1260 ? Ss 20:18 0:00 > /usr/sbin/dovecot -F > dovenull 5985 0.0 0.2 7060 2280 ? S 20:18

I'm using the current cvs version of wine and have been using Grand Theft Auto 3 for a month or so with only a few issues. After updating and recompiling wine last night I encountered a pretty major problem, cut scenes refuse to end. They play through alright, but when it comes to the end of the scene the game doesn't resume. Out of curiosity I backed up my ~/.wine directory and used

Hi, I'm trying to compile zttool from the Zaptel lib, but I just can't find the sorcecode for Libnewt. Anyone got a link? Since i'm using LFS, I can't use precompiled packages. -- Med venlig hilsen / Best regards Michael L?jtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 S?borg Tel (+45) 3955 0700 - Fax (+45) 3955 0707

Hi, I am trying to track down the reason to my problems with sending and reciving fax with my PRI and 2 TDM400P Cards: PSTN <-> PRI (E100P) <-> * <-> TDM400P <-> Fax Machine I have used Zapbarge to listen to the data stream, but I can't say if it really have some time slips - fax kinda noisy in itself. Using the zttool i saw the Sync source for the TDM are

Hi guys, Anyone seen something like below(see below the line)? Machine P2 w/512MB RAM Debian (testing) ; kernel 2.6.12-1-386 asterisk 1.2.1-n-all incl. astcc For many months now I went through * 1.07, 1.09 and never saw something like that. Even with 1.2.0, a month now, at the beginning everything was fine, and suddenly "codec_gsm.c:194 gsmtolin_framein: Invalid GSM data" thing

Hi, I have seen a couple of scripts that should be able to remotely reboot the 79xx phones, but I haven't been able to make it work for my 7940. Anyone able to guide me in the right direction? I am running the SIP 7.1 firmware. -- Med venlig hilsen / Best regards Michael L?jtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 S?borg Tel (+45) 3955 0700 - Fax (+45) 3955

Hi, Just started to get this error after updating to the latest CVS. Asterisk dies if it can't create a channel - not so good. -- Executing SetCallerID("SIP/750-2550", "39660426") in new stack -- Executing Dial("SIP/750-2550", "CAPI/39660426:22179808") in new stack Jun 22 13:52:05 NOTICE[262161]: chan_capi.c:1172 capi_request: didn't find

Typo in your OS79XX.TXT P00 ? instead of P0S !? -----Original Message----- From: Michael L?jtnant [mailto:ml@zyxel.dk] Sent: 17 August 2004 13:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released Hi Shaun, Saw you post, and rushed to their ftp-server and downloaded it :-) But, I can't make my phone (7940) upgrade, so maybe you

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

Hi, i am having problem on playing asterisk sound file on my new installed asterisk.. i have the following extension , if i call from any SIP / IAX phone playback or voicemail doesnt play anything .... but when i dial 102, I hear the MP3 music .. exten => 99,1,Answer() exten => 99,2,Playback(prepaid-welcome) exten => 99,3,Hangup() exten => 101,1,VoiceMailMain() exten =>

Hi all Asterisk 1.8.11.0 on Centos 6.5 My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom, South Africa). Unlicensed G729 codec version on server. 75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes into the recording. The server has been up for 7 months beforehand with no problems with recordings to .gsm format files. I noted

I'm testing an upgrade of an i686 production machine running 1.4.24.1 to 1.6.0.9. I've installed dahdi-linux-2.1.0.4. But: asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.

Hi, I am trying to recice a fax with * using SpanDSP - but it doesn't create the output file. (See the bottom of log file). * Loads both app_rxfax.so and app_txfax.so fine. Also I can't make * autodetect an incomming fax call (yes I have enabled faxdetect=both in zapata.conf - though it's not a Zap device) Any ideas are welcome :-) Best Regards Michael L?jtnant System Details:

Hi, If I try to transfer a user (user listens to MOH while I transfer) to eg. a queue, and the transfer occour while the MOH in the queue is playing, the MOH will stop, and the user hears nothing but scilence, but is in the queue. If I make the transfer to the queue, while still listening to the announcement, the user will hear the announcement, and then the MOH will start. Using latest CVS

Hi, Just made a simple test to see how the two extensions (s and i) worked but for some reason I can't seem to make then act as I would like them to. I pick up the phone and dials 100 or 200 - and in the CLI it prints out what ever I have put in the Noop() If i dial any other number, nothing happens - no indication in the CLI. Souldn't the s or i context be activated when I dial a

Guys, Ok - nowhere near as complex as most of the discussions on here ( ex telco engr for 18 years here).. But thought I'd ask for some assistance. Have just set up my first * Pbx - having a play with it and a couple of Cisco 7960 (configured as SIP) phones. The phones are tftp'ing into the server ok, and picking up the configs all ok. Everything _seems_ to be working, but I

Hi! I just installed Asterisk 0.4.0 with all the default options, and the configuration samples it has. When I try to dial from an h323 client (gnomemeeting) I get this message on the messages file: Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410 (ast_openstream): File demo-congrats does not exist in any format Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553 (ast_streamfile):

Hello list, I have been running Asterisk CVS for a good while. When I try to install 1.0.3, asterisk won't start. Below are the last few lines of output before Asterisk crashes. I ran "make samples" to start with a fresh setup. [app_read.so] => (Read Variable Application) == Registered application 'Read' [app_alarmreceiver.so] => (Alarm Receiver for Asterisk) ==