similar to: music on hold

Hello list, I'm using asterisk 1.4.30 and realtime sip. I notice that the field "musiconhold" is not working as when putting someone on hold, the default musiconhold class is always used. musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes my realtime sip peers have the

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug

I get the following error when trying to play wav files for my IVR menu. Does anyone know what this means or how to fix it? [Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not say fmt Thanks! David

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on

Hi gang, I have a "fun" one for you. I'm not getting the quality of sound I want out of GSM, so I'm trying to make my files into .WAV (.wav) format. Here is the "file" output for 5 files: file *.WAV cents.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft

Hi Everyone, I am currently running Trixbox 2.6 and I have a problem with Asterisk. /etc/init.d/asterisk status Asterisk dead but subsys locked I deleted all files in /var/run/asterisk folder and asterisk restart... It's ok for a while. But some days after Asterisk again is dead. Can anybody help me? Rgs / budacsik

Hello, I would like to have your opinions on this. *Problem:* Currently, there are no ways to perform hypercalls into LLVM (they transfer execution from the program being executed to the LLVM infrastructure to perform compilation). The goal of this project is to cleanly integrate an API into the LLVM code/execution engine to create custom callbacks. The “lli” executable should allow having a

On Thu, Oct 31, 2013 at 11:39 PM, sumeeth kc <sumeethkc at gmail.com> wrote: > Hello, > > I would like to have your opinions on this. > > *Problem:* > > Currently, there are no ways to perform hypercalls into LLVM (they > transfer execution from the program being executed to the LLVM > infrastructure to perform compilation). The goal of this project is to >

Dear list, I am trying to implement a publish-subscribe mechanism in for an embedded R interpreter. But somehow my registered closures seem to get collected by the GC, even though I have protected them. I have reducted my code to the following sample. Sorry if it is a little verbose. The first couple of call of calls still work, but at some point one of the callbacks (callback1 in my

Does anyone know what this means "Warning [65542]: chan_sip:c:501 retrans_plct: Maximum retries exceeded on call 7438737dc873850@172.16.0.52 for seqno102 (Non-critical Request. 172.16.0.52 is the Asterisk Server I'm guessing that I have something miss configured just not sure what it is. If you need more info just ask.

I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

So as y'all know, with your help I managed to get Opus installed at last. Yay! With excitement, I wrote my dialplan, dialled in, and.... [Jan 28 21:30:11] ERROR[29977][C-0000001d]: format_ogg_opus.c:95 ogg_opus_rewrite: Cannot write OGG/Opus streams. Sorry :( [Jan 28 21:30:11] WARNING[29977][C-0000001d]: file.c:468 fn_wrapper: Unable to rewrite format ogg_opus Any idea where I'm going

Somebody knows if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [*40 at liberado15:15] Record("SIP/1201-083453c8", "/var/spool/asterisk/alarme/alarme-1201-200905121212:mp3") in new stack ??? -- <SIP/1201-083453c8> Playing 'beep'

Hi i got lots of this from the asterisk console what does this mean? format_wav.c:247 update_header: Unable to find our position asterisk console: Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[2812]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[4430]: format_wav.c:247 update_header:

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

Hello, As an exercice, I installed sipp on the same box as a Asterisk 11.4 instance (to keep network equipements out of the equation). I'm focusing on the maximum number of new calls this Asterisk instance can deal with. Here is the dialplan (AEL) I'm playing with: _X. => { Verbose(0,Incoming call from ${CALLERID(num)} to ${EXTEN} in ${CONTEXT} - case A);

Hi, all I have a medium sized data, 6 years. Each observation is a case with a date variable, such as '2004-08-02'. Some of the months didn't occur a case. I want to plot the 6 years data by month, and the Y_axis is the freqency of cases for each month, meaning 12*6=72 bars or points in the figure. I though of a method, 1st, using the months function, then ploting. But I need to

I have a persistent problem with a PBX I commissioned recently. After a few days it goes into a spasm, creating thousand of log files and giving the message below on the CLI. Dell PE 1600 with Sangoma A200. pbtpbx*CLI> show version Asterisk 1.2.14 built by root @ pbtpbx.local on a i686 running Linux on 2007-01-13 18:31:56 UTC Asterisk Queue Logger restarted Rotated Logs Per SIGXFSZ

Hi All, Many Thanks to Underwood for her excellent review of our big trouble which prevent LMS-based AEC algorithms to be used in most computer. Maybe it can be summaried as follows: 1. Different sample rate of sampling and rendering does exists in most low-cost soundcards (In my experiments over more than 20 soundcards, the differences range from 0.5Hz to more than 50Hz when sample rate is set