Displaying 20 results from an estimated 300 matches similar to: "Another Segmentation Fault (Recording sound)"
2009 Apr 23
3
Record in mp3
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'
2011 Dec 20
1
File Convert
Hi users,
I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file
to G729 using file convert, but I am facing error as follows,
file convert /tmp/welcome.gsm /tmp/welcome.g729
Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729!
Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed.
[Dec 20 17:24:18] WARNING[2221]: translate.c:256
2004 Jul 23
1
chan_alsa record problem
Some unsuccessfull attempts to make console calls working.
If a sip phone is called, the other side will hear nothing.
If I try to record some sound the application will not finish. There
is a sound file, but it is empty (0 bytes). "Record(${FILE}:gsm|10|30|skip)"
is used in the dial plan. After hangup the following error messages
show up:
NOTICE[]: channel.c:1683 ast_set_read_format:
2005 Jul 30
1
Record() permission problem
Hi All...
I'm trying to use the record() app and it complains that it can't open it's
file because permission was denied. I'm running the released Asterisk on
Debian Linux. The target directory is workd writable. Here is the
relevant part of the dialplan:
exten => 1,1,Playback(leave-message)
exten => 1,2, Record(/var/local/whois-messages/whois-${contactid}:wav|6|120)
2004 Jun 28
2
AGI->Exec Problem
Hello,
I am having some trouble with the Asterisk::AGI perl library. It seems
that the AGI->Exec() command is causing me a problem.
Here's the line in my AGI code: $AGI->exec('Record',"$vmfile:wav, 30");
I'm trying to record voicemail to the file name stored in $vmfile with
a silence timeout of 30. However, this is not being parse by AGI or
Asterisk correctly,
2003 Nov 11
4
Registering an application
Hello..
Maybe I'm asking something silly but..... How can I register my own app with * ?
I've made a simple .so , but I cannot find it in asterisk when i type "show applications"
Here is the code:
#include <asterisk/lock.h>
#include <asterisk/file.h>
#include <asterisk/logger.h>
#include <asterisk/channel.h>
#include <asterisk/pbx.h>
#include
2006 Dec 15
2
call from h323 to SIP
Hi
i am trying to do the same thing:
receive a call from a cisco callmanager and forward it to a SIP user.
Asterisk is compiled with h323 support, and is configured as a gateway
in the cisco callmanager.
h323.conf:
[general]
port = 1720
bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP
address for this machine
allow=all
extension.conf:
exten = 3298,1,Answer
exten =
2017 Dec 06
2
Simple speech recognition for driving IVR - "press or say one".
Thanks for your responses - it looks like I have the following
options, in order of ease:
1: Modify and recompile app_record.c
Change line 471
https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L471
from
status_response = "DTMF";
to
status_response = dtmf_integer;
Pro: Free, easy
Con: Have to remember to edit module each time a new Asterisk update comes out
2:
2003 Apr 03
6
tc problem
Hello..
I have a linux box and I want to make priority on traffic generated by my
LAN''s computers..
I don''t have a guaranted bandwidth, so I wanna use sfq...
I want to make traffic to port 80 , 443 , 25 & 110 PRIORITY 1
Traffic src or dest 192.168.0.2 to make priority 2
And the rest to put it in proiority 3..
I did the following :
tc qdisc add dev eth0 root handle 1:
2003 Nov 07
2
Modem as a FXO
Can I use a modem and a soundcard as an fxo ?
I've read in the documentation something , but how can I do that ?
Regards
Alex
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2014 Jan 24
2
IOPS required by Asterisk for Call Recording
Hi
What are the disk IOPS required for Asterisk call recording?
I am trying to find out number of disks required in RAID array to record
500 calls.
Is there any formula to calculate IOPS required by Asterisk call
recording? This will help me to find IOPS for different scale.
If I assume that Asterisk will write data on disk every second for each
call, I will need disk array to support minimum
2004 Jan 17
3
SS7 over Asterisk ?
Hello..
I have a customer who wants to connect 2 PBX's over IP..
The setup should look like this:
[PBX] <-- SS7 --> [Asterisk] <-- IAX --> [Asterisk] <-- SS7 --> [PBX]
Since there are no SS7 cards , I was thinking at a way of carrying the E1 data as bulk...Can I do that ? How ?
Is possible a scenario like this ? I'm thinking of IAX because I don't
2003 Jul 23
4
Problems with g729
I am having some problems with g729 with SIP and ZAP channels.
1)
I have two g729 licences. Very frequetnly (I don?t know what triggers the error) I get the following warnings and error when I try to place a call via SIP to my X100P. The only way to get out of this is through a restart of *. When the error ocurrs there are no other calls in place. Any ideas?
Error Opening channel:2 not
2004 Nov 20
1
Asterisk dead but pid file exists - gdb asterisk core.13089
Dear ALL,
Any clues or tips for the following gdb messages.
[root@localhost asterisk]# uname -a
Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct
29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux
localhost*CLI> show version
Asterisk CVS-HEAD-09/22/04-11:19:09 built by
root@localhost on a i686 running Linux
[root@localhost asterisk]# gdb asterisk core.13089
GNU gdb Red Hat Linux
2018 Jan 20
2
Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?
Just a quick and dirty thought, try the MONITOR application.
Pseudo-code:
Anchor-point
PLAYBACK ("press or say")
MONITOR (use the split audio files mode, not the mixed - this way you can
roughly separate which side did the "talking")
READ (audio file "1 to 5", try to grab one digit)
STOPMONITOR
IF (READ variable timed-out, send the incoming half of the monitor file
2003 Apr 15
4
call announce?
using a zap fxo and zap fxs card how can I set up caller announce? like
this.
1 call comes in and a prompt asks the called to identify themselves.
2 the system would then put the caller on hold and pick up the FXS and
play the message for the users prompting them to hit 1 to accept the
call and have it connected or hit 2 to dump the live caller to
voicemail.
Can this be done with *
Dave
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
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]
2003 May 27
1
Monitor with Mp3 format
Hi all,
i want to monitor channels with the "Monitor" app in mp3 formatted files.
But, if i use mp3 as first argument for the Monitor app, errors occurs....
WANRING[28689]: File file.c, Line 193 (ast_writestream): Unable to
translate to format mp3, source format 64
...following more "Warnings" .....
Is it not possible to monitor to mp3-files?
What's wrong?
Thanks
2018 Jan 20
2
Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?
On Saturday 20 January 2018 at 18:45:49, Jonathan H wrote:
> Oh, what a good idea! That's exactly the kind of lateral thinking I
> was hoping someone would come up with.
>
> I thought it was called MixMonitor, and tried to wrap my head around
> it but couldn't.
MixMonitor is related, but different (and as the name suggests, automatically
mixes the two channels, so I