Displaying 20 results from an estimated 700 matches similar to: "playback windows recorded sound"
2006 May 31
0
AGI MySql
thanks Billy. I replaced
print "STREAM FILE $filename \"\"\n";
with
print "EXEC PLAYBACK $filename \n";
and it worked fine. Interestingly when I did
print "STREAM FILE beep \"\"\n";
within the script, it worked.
If I wasnt a newbie to asterisk I wouldve thought this to be strange.
>From: "William Piper"
2007 Jun 13
3
WAV file best sound quality
Hi,I have been using wav files with sample rate of 8khz and 8 bits and I find the sound quality really poor.What is the best sound quality I can achieve on Asterisk?Responses would be appreciated.Rgds,Akpome
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2006 Feb 03
0
varion card
I've been using it in a test environment with no problems. However, I
haven't used it in production yet. I'm doing some voice broadcasting
with a PRI and so far I'm content with the performance.
-MC
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Akpome
Akpoguma
Sent: Friday, February 03,
2009 Aug 14
1
i have a error in ivr
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
2007 Oct 17
2
Help Needed - Error when playing wav files in 1.4.11
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
2008 Nov 11
7
music on hold
hii guys:
i get the message from the asterisk:
Started music on hold, class 'default', on Local/s at skype-web-callback-dial-263to263-1775,1
[2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025
[2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav
[2008-11-11 14:32:41] WARNING[1781]:
2006 Apr 14
22
attended transfer issue
Hi!
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!
At this point the attended transfer should go into a blind transfer. The
phone of Person B should still be ringing and the
2005 Oct 11
1
migrated to new ver on voip connection vs1 server voicemail problems
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
2006 May 31
5
Converting .wav to .WAV
Hi,
how can I convert .wav files to .WAV:
# file greet.*
greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz
greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
using 'sox'?
Thanks
--
Domenico Viggiani
2006 Jun 13
1
VOCAL + Asterisk
I want to start a community based voip network projcet and am thinkimg of
using VOCAL and asterisk gateways..... my question is, has anyone bench
marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or
Asterisk all the way.........am expecting 1000 -> 5000 users..
your thoughts would be appreciated.
_________________________________________________________________
Don't
2010 Oct 12
1
sound file debug
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
2003 Nov 06
6
Error in Incoming SIP call
When I get a SIP call, I get the following error:
*CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is
'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp'
WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No
application '' for extension (incoming, 5147771111, 1)
== Spawn extension (incoming,
2009 Mar 12
2
compiling ffmpeg with --enable-libspeex (was Re: from Adobe Flex / Flash Player 10 .flv Speex via Red5 to .wav PCM?)
I am having trouble compiling ffmpeg to support speex, which didn't
work with the ubuntu libspeex-dev package, but looks like it might
with the Speex version 1.2rc1 tarball from http://speex.org/downloads/
How do I tell ffmpeg's configure and/or make to use the 1.2rc1 version
of libspeex in /usr/local/include instead of the older debian/ubuntu
libspeex-dev package in /usr/include/speex?
2003 Dec 17
0
issue recording files in wav49 from AGI
Following is a log from an attempt to record and playback a file in
wav49 format from an AGI script.
COMMAND: stream file aa/after_the_tone "" 0
RESULT_LINE: 200 result=0 endpos=41920
RESULT_DICT: {'result': ('0', ''), 'endpos': ('41920', '')}
COMMAND: record file /activity_alerts/wavs/123456_1_1_0.745781945801 wav49 "#" 20000 0
2006 Mar 02
3
Native music on hold - Error
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
2006 Oct 16
0
Weird problem with beep.wav!
This is really doing my head in!
For some reason, my asterisk box can't playback beep.wav.
I have this extension defined in my internal context:
'10001' => 1. Answer() [pbx_config]
2. Wait(2) [pbx_config]
3. Record(/tmp/asterisk/10001:gsm) [pbx_config]
2010 Aug 13
2
realtime sip peers : musiconhold class
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
2007 May 18
1
web app to playback recorded phone calls.
1 of our customers records all phone calls and needs to be able to be
played back via a searchable web app. I tried ARI but it is very
limited.
Anyone have any ideas?
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2006 May 16
0
Re: [Astlinux-users] British English Female files ready for download
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
2003 Sep 26
0
Unable to find a path from ULAW to G723
Hello,
I just CVS'd today and now I'm getting these errors when I call one
grandstream phone to another both using 711U:
NOTICE[1225991360]: File channel.c, Line 1476 (ast_set_read_format): Unable
to find a path from ULAW to G723
NOTICE[1225991360]: File channel.c, Line 1446 (ast_set_write_format): Unable
to find a path from G723 to ULAW
NOTICE[1225991360]: File channel.c, Line 1476