Displaying 20 results from an estimated 1100 matches similar to: "Weird problem with beep.wav!"
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
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 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
2004 Feb 03
0
upgrade problems
I upgraded to 0.7.1 from a cvs version from a few weeks before 0.7.1 was
relesed.
now I am having troubles with my dialing plan and voice mail.
As part of the upgrade I re-built the machine so there was a blank slate
however after installing 0.7.1 I had no mail box creation script and
could not figure out how to go about creating a mailbox, any suggestions
would be usefull.
I have looked at
2005 Oct 11
5
help with broken voicemail
I can not figure out what the heck is going on. I went back to my old
version and I still get errors when the voicemail system tries to load
any of the greetings, unavail messages, etc. the normal voicemail
prompts work, but any user recording don't work. Leaving a new message
appears to work, but the system wont replay them, it throws errors.
Here is an example of the errors:
Oct 11
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
2004 Sep 03
7
Dropping incompatible voice frame
Hi: i have a problem.
Mi extensions.conf:
exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN})
exten => _N.,2,SetAccount(${customer})
exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1})
exten => _N.,4,ResponseTimeout(5)
exten => _N.,5,Background(ifyou)
exten => _N.,6,Background(silence/1)
exten => _N.,7,Background(ifyou)
exten => _N.,8,Background(silence/5)
exten
2012 May 04
0
Sound file format and Asterisk 1.8.11-cert1
Hi All;
I installed Asterisk 1.8.11-cert1, and it look like the default is ulaw for the sound files. How I can fix this?
Athough file beep.gsm is existed under path (/var/lib/asterisk/sounds/en), but when I used the Record function, it gave me the following (so I am sure there is something that can let asterisk accept beep.gsm), what could be?
[May 5 00:44:16] WARNING[2262]: file.c:663
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
2006 Jan 27
0
ODBC Problem with voicemail.
I've installed the last released asterisk 1.2.2 on my own HLFS system
with a 2.6.14.3 kernel. I've also a 2 FXO/ 1 FXS digium card on it.
Every thing is working correctly.
For ODBC, I'm using UnixODBC with pgsql. The voice messages are
correctly written to the database and also their number is correctly
reported by VoicemailMain dialogue.
However, after reading the time/day of the
2003 Jun 19
1
Unable to find a path
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):
2003 Jun 13
1
strace shows that files are not accessed
strace on file access in asterisk shows that * is not even attempting to
access the voice files.
If I *manually* load app_playback.so, app_macro.so, and then
pbx_config.so, I they will load and I get a dialplan. Ok, that's a
problem -- autoconf is clearly not working, or there's some other
related issue.
So I try to use the demo and do "dial 500". This should connect and
2004 May 18
1
VoiceMailMain dumps user back into my incoming context after leaving a message
I have a dial plan that includes a company phone directory as a main menu
option. If they just sit at the main menu, after 20 seconds, they are
transferred to the operator. If the user picks an extension from the
directory, they are transferred to the proper extension. If the called
number is not available, they are transferred into VoiceMailMain. They
leave a message, and hang up. The hang
2007 Jun 25
2
two channels, each dropping into the same context, different behavior.
So, incoming calls on zap work just as I expect them - an intro is played, the
caller hits 1 for sale 2 for support or dials an extension. I'm using the
privacy option for all extensions. When calls come in from zap, they caller
is played the priv-recordintro recording, they say their name, and everything
happens normally from there on out. However, when the call comes in from sip
and
2007 May 17
2
Blacklist
Hello All,
I was wondering where does Asterisk stores the blacklist numbers?
I looked into the dialplan and it shows that it
*"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB?
hyperion*CLI> show dialplan app-blacklist-add
[ Context 'app-blacklist-add' created by 'pbx_config' ]
'1' => 1.
2004 Aug 03
0
ZyXEL 2000w In Call Menu/Hold configs
Hi Everyone,
After a fair amount of faffing ive managed to get the 2000w working with
asterisk for IP -> PSTN calls (i.e. get the phone to make and receive calls
over our BT line). The final solution is to set up outgoing VoIP calls but
I now know that without a SIP aware router I can think again! (damn you
iptables!)
In the mean time I'm trying to figure out why I can't get the
2005 Mar 21
2
Ext matching problems
Hello everyone...
I'm trying to get up a testing pbx installation. Following instructions
of what've read from the handbook and from asterisk's wiki, I wrote the
dial plan as follows:
[general]
;
;
static = yes
;[globals]
;
[default]
;
exten => 0,1,Answer()
exten => 0,2,Playback(fcopba1)
exten => 0,3,Hangup()
exten => *0,1,Answer()
exten => *0,2,Record(fcopba1:gsm)
2003 Jul 07
1
Dial plan doesn't seem to save properly
When I first to the "add extension" the "show dialplan" has the lines that
say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and
restart the lines with "SIP/" are gone.
************************
"Show dialplan" before:
************************
asterisk01*CLI>
[ Context 'default' created by
2003 Mar 29
1
How does * process the extensions??
Hi,
How does * read and process the extension.conf file??
The reason I ask is that I think it probably has a very large impact on how the calls are routed and processed by the system especially when it comes to least cost routing..
Let me explain...with an example..
I am using the * Devkit to get to grips with the system, so I have and X100P (Zap/1) and and S100U (Zap/2)..
Below is my