Take a look at /var/log/asterisk/main or full /if enabled. Perhaps
there is a file not found. try:
exten =
_367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
exten =
_367,n,Set(LIMIT_WARNING_FILE=/path_to_your_audiofiles/file)
# do not add any extension!
Hi,
it's me again with a cdr-issue. I have the following example
extensions.conf:
# dial in
exten = 100,1,Playback(hello)
exten = 100,n,Dial(local/200,20,rtg)
exten = 100,n,Playback(pleasewait)
exten = 100,n,wait(10)
exten = 100,n,Playback(goodbye)
exten = 100,n,Hangup
# for local dial
exten =
1.6.1.20 :-)
Am 12.11.2010 15:12, schrieb Olivier:
2010/11/12 Thorsten Gllner t...@ovm-group.com
Hi,
it's me again with a cdr-issue. I have the following example
extensions.conf:
# dial in
Exactly! The call duration is not correct in this case. That is "my"
problem.
Am 12.11.2010 15:23, schrieb Danny Nicholas:
From:
Hi,
our Asterisk is connected to an E1 port. So we are using the
DAHDI-Driver. Please , how do I tell the driver/Asterisk to wait for
overlap digits for in-calls? I found the option overlapdial=yes but I
did not try yet. Is that my option? Is there any option for setting an
timeout?
Thorsten
Am 12.12.2010 20:49, schrieb dave george:
I am using Asterisk 1.6.2.5-0
running on ubuntu and I have a problem passing called ID
on calls to the PSTN
list
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--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
Hi,
for dahdi-calls I can see the current calls with dahdi show channels.
But where can I see the current call-duration or the call-start-time?
dahdi show channel n does not show this info.
-Thorsten-
--
_
-- Bandwidth and
So simple - great, thank!!!
Am 17.12.2010 13:07, schrieb Vincius Fontes:
You probably want "core show channels
verbose".
Atenciosamente,
Vincius Fontes
Gerente de Segurana da
So simple - great, thank!!!
Am 17.12.2010 13:07, schrieb Vincius Fontes:
You probably want "core show channels
verbose".
Atenciosamente,
Vincius Fontes
Gerente de Segurana da
Am 20.12.2010 15:11, schrieb Shaun Ruffell:
On 12/20/10 6:50 AM, Max Alex wrote:
Hi All,
I have installed asterisk 1.6.2.8
Dahdi: 2.4.0
Digium card: Digium, Inc. Wildcard AEX800 8-port analog card
I have configured this card properly and it is working for calls too,
But there is issue of
Am 20.12.2010 15:11, schrieb Shaun Ruffell:
On 12/20/10 6:50 AM, Max Alex wrote:
Hi All,
I have installed asterisk 1.6.2.8
Dahdi: 2.4.0
Digium card: Digium, Inc. Wildcard AEX800 8-port analog card
I have configured this card properly and it is working for calls too,
But there is issue of
Am 20.12.2010 16:00, schrieb Thorsten Göllner:
Am 20.12.2010 15:11, schrieb Shaun Ruffell:
On 12/20/10 6:50 AM, Max Alex wrote:
Hi All,
I have installed asterisk 1.6.2.8
Dahdi: 2.4.0
Digium card: Digium, Inc. Wildcard AEX800 8-port analog card
I have configured this card properly
Am 20.12.2010 21:39, schrieb Ernie Dunbar:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until either
the Asterisk server is restarted (and the zombies die a natural death), or
the kernel runs out of PID
Am 10.01.2011 22:45, schrieb Shaun Ruffell:
On 1/10/11 3:07 PM, cjwstudios wrote:
I'm using libpri-1.4.11.5.
On Mon, Jan 10, 2011 at 2:05 PM, Shaun Ruffell sruff...@digium.com
mailto:sruff...@digium.com wrote:
What version of libpri are you using?
Others probably know better than I do
Am 12.01.2011 11:37, schrieb Duncan Turnbull:
Hi there
I have two different asterisk systems (both 1.4) whose dtmf tones are not being
picked up by a particular conference system users are dialling into. I can call
myself with the phones and hear the tones, but I am guessing perhaps they are
Am 14.01.2011 11:55, schrieb Jonas Kellens:
Hello list,
today I experienced the following for the first time :
[Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for
channel
'0x114af2c0'
[Jan 14
Am 14.01.2011 12:50, schrieb Jonas Kellens:
On 01/14/2011 12:44 PM, Thorsten Gllner wrote:
Am 14.01.2011 11:55, schrieb Jonas Kellens:
Hello list,
today I experienced the
07:47:24 UTC
LibPRI: 1.4.11.4
DAHDI: 2.3.0.1 Echo Canceller: MG2
Wanpipe-Driver: 3.5.15
Sangoma-Firmware: 43 (Board A104d with echo hardware chanceller)
Typically this problem can occur with echo cancellaltion. But I do not
think, that this is my problem.
Best regards,
-Thorsten Göllner
Am 19.01.2011 16:57, schrieb mancyb...@gmail.com:
Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for everything else)
Question:
how can I know if the call was not
Am 21.01.2011 12:21, schrieb Vitor Carlos Flausino:
Hello all.
Can you help me find where the CDR's are being stored?
The result of cdr show status is:
Call Detail Record (CDR) settings
--
Logging: Enabled
Mode: Simple
Log unanswered calls: No
*
Am 23.01.2011 18:38, schrieb Carlos Chavez:
On Sat, 22 Jan 2011 19:47:43 -0500, Mark Deneen wrote
On Sat, Jan 22, 2011 at 6:53 PM, Carlos Chavez
cur...@telecomabmex.com wrote:
On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote
On 22 Jan 2011, at 18:02, Carlos Chavez wrote:
Cannot
Hi,
I am using
Asterisk: 1.6.1.20
LibPRI: 1.4.11.4
DAHDI: 2.3.0.1 Echo Canceller: MG2
Wanpipe-Driver: 3.5.15
Sangoma-Firmware: 43 (A104d)
I handle some calls with my own PHP-AGI-Script. After a dial-command I
use GET FULL VARIABLE ${answeredtime} or GET FULL VARIABLE
${dialstatus} and get
Am 02.02.2011 04:34, schrieb Nikhil:
Hi everyone
How can I get the current calls details in asterisk.if I use cli
commad core show channels,there is two channels of each call.But the
requirement is, need to get caller ,calee,starttime ,duration of the
current calls.This value should be
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
Am 04.02.2011 10:53, schrieb DHAVAL INDRODIYA:
hi group ,
i am working on dahdi_monitor for measuring voice quality , so i want
to know that on which data i can tell that this PRI
lines are working properly, is there any measurement on basis of that
i can make MOS. i am working from last 2-3
I discussed this with sangoma support in the past. Sangoma says, it
is NOT recommended to disable echo cancellation there.
Am 04.02.2011 10:53, schrieb Gopalakrishnan A.N:
It seems to be you are using Sangoma T1/E1 card with
echo cancellation. If I am not wrong
Try it with your own AGI-Script - this is more flexible.
http://www.voip-info.org/wiki/view/Asterisk+AGI
Am 16.02.2011 11:45, schrieb shayne.al...@gmail.com:
Dear Mr,Ms;
I am planing for a custom IVR, for example to act as a
4 gsm-ports. Should be stable
and have an echo cancelltaion feature. And of course it should be cheap ;-)
Best regards
-Thorsten-
--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
://lists.digium.com/mailman/listinfo/asterisk-users
--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
--
_
-- Bandwidth and Colocation
Am 03.03.2011 16:02, schrieb satish patel:
Hey Guy,
I have quick question. I am purchasing Sangoma A102D card but i am
confused between PCI and PCI Express. Which card would be good for
me.
Definitely PCI Express is advance but i
After working fine for a week or so my new Quad E1 asterisk 1.8 system has
started rejecting outbound calls from the Nortel
BMC 450 it is connected to.
The cli fills up with these:
sig_pri.c: Ring requested on unconfigured channel 255/255 span 3
Is this likely to be a
1) config
After working fine for a week or so my new Quad E1 asterisk 1.8 system has
started rejecting outbound calls from the Nortel
BMC 450 it is connected to.
The cli fills up with these:
sig_pri.c: Ring requested on unconfigured channel 255/255 span 3
Is this likely to be a
1) config
Try:
cd /usr/src/dahdi
./Setup dahdi
That's it.
Am 22.03.2011 21:06, schrieb satish patel:
Hey!
I am installing Sangoma A102D wanpipe driver and i got following
error. what is this ? why dir isn't there ?
Should be installed here:
# which wancfg_dahdi
/usr/sbin/wancfg_dahdi
Try:
# find / -name "wancfg_dahdi"
Am 23.03.2011 15:17, schrieb satish patel:
Hey, I did ./Setup dahdi and everything went well but i didn't
find any command
Am 01.04.2011 14:27, schrieb Roger Burton West:
On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote:
Can anyone suggest which is the best scripting language for Asterisk or any
telecom device?
Depends on the other parameters. Perl is great for rapid development,
but I wouldn't
? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237
and about 2,5 GB free so swap space is not used
or exhausted.
Maybe the high load is not cause of this crashes but it's the only thing
the crashes can be reproduced with.
Thank you!
Maximilian Grobecker
Am 04.04.2011 16:03, schrieb Thorsten Göllner:
Take a look with top at your system when
Am 05.04.2011 18:50, schrieb vip killa:
I'm wondering if there is a simply way to perform a voicemail callback
feature using AGI.
For instance, a caller leaves a voicemail, the voicemail will then
call the owner of the voicemailbox determined by a database look up.
One possibility: look via
Am 13.04.2011 15:08, schrieb A J Stiles:
Hi. I just want to make sure I understand this before doing something that
might break things spectacularly for our users and customers :)
We are using Asterisk 1.6.2.9 and my programming language of choice is Perl.
I want, when a call comes in on
Am 02.05.2011 15:59, schrieb A E [Gmail]:
On Mon, May 2, 2011 at 3:15 AM, A E
[Gmail] all.efor...@gmail.com
wrote:
Hello All,
Probably a silly question, but we're
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
Maybe IO-Activity caused by intensive logging. Take a look at your
Log-Files. Maybe one or more log files a growing "rapidly"?
Am 20.05.2011 11:24, schrieb RSCL Mumbai:
CPU utilization is constantly above 24% without any
call activity..
top -
Hi,
"pri show version" should show you something like that:
libpri version: 1.4.11.4
loadzone should be "de".
Better you configure your system with the help of "./Setup dahdi"
(which can be found in your wanpipe-source-directory). Or after
://lists.digium.com/mailman/listinfo/asterisk-users
--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
--
_
-- Bandwidth and Colocation Provided by http
Ist the same operator connected to the pri-line? Perhaps another
telco-operator can not connect to the desired destination - for whatever
reason.
Am 08.06.2011 12:55, schrieb Satish Patel:
Thanks for reply,
But I'm able to call those number from my cell phone and othere pri.
I'm only having
Executing the query in MySQL-CLI is fine?
Am 05.07.2011 11:25, schrieb Ulrich Meckel:
Hi List
I tried to use SQL Query in my diaplan. If i only use one or two there
is no Problem but if i try to start the third one after the other it
hangup after the 2nd clear
exten = _123.,1,MYSQL(Connect
What does "but this didn't work" mean?
Am 20.07.2011 15:25, schrieb Danny Nicholas:
Hello,
Im putting Asterisk in to
replace an existing IVR and that PBX system uses * to
terminate number input instead of #.
no success:
exten = 123,1,dial(number,time,h(123)H(123))
Any idea? Or is it not possible?
I am using Asterisk 1.6.1.20 an have no problem with the "normal
hangup option" via pressing "*".
Thanks,
-Thorsten-
--
Thorsten Göllne
If the caller hangs up Asterisk sends a SIGHUP. You can catch the
signal and do whatever you want to do.
Am 21.11.2011 07:38, schrieb David Cunningham:
Hello,
We would like to continue a Perl AGI after a Dial() it has done
completes following caller
--
Thorsten Göllner
OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf
Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
Hi,
I use an AGI with PHP. Here is a short snippet:
[...]
declare(ticks = 1);
pcntl_signal(SIGHUP, array($this, "signal_handler"));
[...]
public function signal_handler($signal_number)
{
$this-log_message("debug", "Signal catched:
Hi,
since version 1.4.12 the libpri package supports ETSI Explicit Call
Transfer feature:
http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.12
Does anyone know, how to use this feature in the dialplan? I can not
find any hints in the asterisk doc.
Best regards,
-Thorsten-
Hi,
I am looking (for the best) solution to recognize *german* words or
simple phrases with a given number of words (eins, zwei drei etc. or
hauptmenü, zurück etc.). Can somebody give me a good link? Can I find
external service providers who can be accessed via ASR()?
Best regards,
What do you want to do? Sending and receiving SMS?
Am 03.06.2012 11:20, schrieb Michelle Konzack:
Hello Experts,
since connecting of 4 Huawei K3765-HV Sticks to my Server does not work,
I now use the Vodafone EasyBox 803A (cost less then 30 Euro on eBay) and
connect
Where can I find such ip-lists, please?
Am 05.06.2012 18:40, schrieb Alejandro Imass:
We use complete regional blocks from Wizcraft and blocking at minimum
all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We
block almost anything that is not our actual customer market and screw
Did you check "ulimits" in Asterisk CLI?
Am 14.06.2012 16:02, schrieb [Digital^Dude] :
Hello,
Asterisk under
90% load of SS7 calls can only withstand the voice
broadcasting for 30 minutes. After around 30 minutes, it stops
Hi,
I am trying now for over 4 hours setting up cdr-logging via odbc into a
mysql database. But with no success. Do you have any hint for me?
cat /etc/odbc.ini
--
[MySQL-asterisk]
Description = MySQL ODBC Driver
Driver = MySQL
Socket = /var/run/mysqld/mysqld.sock
Server =
Am 18.06.2012 21:49, schrieb James Sharp:
On 6/18/2012 11:52 AM, Thorsten Göllner wrote:
Hi,
I am trying now for over 4 hours setting up cdr-logging via odbc into a
mysql database. But with no success. Do you have any hint for me?
*SNIP*
But after a call hangup I get the following error
Am 19.06.2012 11:53, schrieb [Digital^Dude] ®:
Machine specs: CentOS release 5.5 (Final)
RAM: 4 GB
CPU: Dual Xeon 2.66 GHz
Asterisk 1.8.7.1 built by root on a x86_64 running Linux.
*CLI ulimit core
Core file size (core) is effectively unlimited.
*CLI ulimit data
Program data segment
Hi,
I need a fax-send - setup. I read the book Asterisk The Definitive
Guide chapter 19 (fax) and found 2 options listed there.
1) Using spandsp.
2) Using FFA (Digium Fax For Asterisk).
But the book nor any other article I read point out, what the
differences or drawbacks are.
Does anyone
Am 21.06.2012 11:30, schrieb [Digital^Dude] ®:
Asterisk 1.8.7.1 built by root on a x86_64 running Linux.
CentOS release 5.5 (Final)
RAM: 4 GB
CPU: Dual Xeon 2.66 GHz
Asterisk is running as root
data seg size (kbytes, -d) unlimited
file size (blocks, -f) unlimited
Am 29.06.2012 11:38, schrieb CDR:
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI-command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until
the new one is stable.
On Mon, Jul 2, 2012 at 4:57 PM, Thorsten Göllner t...@ovm-group.com wrote:
What Asterisk version?
Am 02.07.2012 15:14, schrieb CDR:
Thanks. I already solved it using this command. The only issue was
that it gives you as return the ASCII code of the digit pressed
instead
that information on the debug, but how do you bring it
inside a variable, so you may use it? I could not find a way. Maybe I
am missing something?
On Tue, Jul 3, 2012 at 9:20 AM, Thorsten Göllner t...@ovm-group.com wrote:
I just tried it on asterisk 1.8.13 with agi set debug on. The last log
line
Hi,
voicemail plays after hitting # as final file auth-thankyou. Is
there any possibility to change this behaviour? Custom soundfile or
disable it perhaps?
Thanks for your answer(s)!
-Thorsten-
--
_
-- Bandwidth and
Maybe a stupid answer ;-)
Did you make a "reload"?
Did you try from shell:
mysql -u myuser -pmysecret AsteriskHosted
?
Am 27.09.2012 11:00, schrieb Jonas
Kellens:
Hello,
this might seem a stupid question but I
Now I see: you try to use the wrong config file - try
/etc/asterisk/res_config_mysql.conf instead.
Am 27.09.2012 11:40, schrieb Jonas
Kellens:
On 27-09-12 11:27, Thorsten Gllner wrote:
Maybe a stupid answer ;-)
Now I see: you try to use the wrong config file - try
/etc/asterisk/res_config_mysql.conf instead.
Am 27.09.2012 11:40, schrieb Jonas
Kellens:
On 27-09-12 11:27, Thorsten Gllner wrote:
Maybe a stupid answer ;-)
Did you take a look at the asterisk log? With core set verbose 3 or more?
Am 01.10.2012 12:46, schrieb Gianluca Baù:
Hello guys,
my name is Gianluca and this is my first post in this ml.
i've a strange problem with my asterisk box. I'll try to explain you.
A (sip from ser) calls -- B (sip
the cause?
Thanks, regards
Gianluca
2012/10/1 Thorsten Göllner t...@ovm-group.com:
Did you take a look at the asterisk log? With core set verbose 3 or more?
Am 01.10.2012 12:46, schrieb Gianluca Baù:
Hello guys,
my name is Gianluca and this is my first post in this ml.
i've a strange problem
is the card sharing irq?
no. this the only card that uses IRQ 30
1b:00.0 Network controller: Digium, Inc. Device 1420 (rev 14)
Subsystem: Device 0005:
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop-
ParErr+ Stepping- SERR+ FastB2B- DisINTx-
Status:
Maybe you should give irqbalance a try:
https://irqbalance.org/
Maybe you also can assign irq 30 to a specific cpu (core):
https://cs.uwaterloo.ca/~brecht/servers/apic/SMP-affinity.txt
Am 06.11.2012 04:04, schrieb Edwin Lam:
On 11/5/12 11:59 AM, Vincent Swart wrote:
You're HDLC error is
Hi!
1) How long does the outdial take? Does the Dial-Command return immediatly?
2) Maybe dial-out is blocked by your carrier? Did you try to open a
trouble ticket there?
3) What number do you try to call? Did you try some different number?
Alway the same problem?
You receive
Am 10.12.2012 06:37, schrieb Chandrakant Solanki:
Hi All,
OS : CentOS 5 64bit OS Machine
Asterisk: 1.8.13.0
ODBC Packages:
unixODBC-2.2.11-7.1
mysql-connector-odbc-3.51.12-2.2
unixODBC-devel-2.2.11-7.1
res_odbc.conf
[telco-ops]
enabled = yes
dsn = telco-ops
username = dba
password =
= 3
On Mon, Dec 10, 2012 at 4:34 PM, Thorsten Göllner t...@ovm-group.com
mailto:t...@ovm-group.com wrote:
Am 10.12.2012 06:37, schrieb Chandrakant Solanki:
Hi All,
OS : CentOS 5 64bit OS Machine
Asterisk: 1.8.13.0
ODBC Packages:
unixODBC-2.2.11-7.1
Hi,
I am using:
Asterisk 11.2.0
libpri 1.4.12
Dahdi: 2.6.1
Sangoma E1-Card with Wanpipe-Drivers 3.5.28
I call my asterisk box via SIP and connect the call to an AGI-Script.
Within the script I do
EXEC SetCallerPres prohib
or
EXEC SetCallerPres prohib_not_screened
But I get the following
? If not, do a make menuselect and see
if something broke in the ability to make the application.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten
Göllner
Sent: Thursday, January 24, 2013 8:31 AM
To: Asterisk
Hi,
I am using Asterisk 11.2.0. Channel Event Logging (CEL) ist activated
and running. CEL entries are logged into an mysql database. So far so good.
I want to do some extra cel logging and try the following via an AGI-Script:
EXEC CELGenUserEvent test
In the asterisk logfile I can see the
Did you watch the cpu usage (for example with top)?
You have a board installed which does use dahdi? Did you check the
command dahdi_test?
Maybe a (performance) problem of the software ec?
Am 06.02.2013 11:13, schrieb Hristo Trendev:
Hi,
I have been experimenting with ConfBridge from the
Sorry - I just read you alsways checked the cpu usage. Are all cores at
100%? Is it the atserisk process which consumes it all?
Am 06.02.2013 13:54, schrieb Thorsten Göllner:
Did you watch the cpu usage (for example with top)?
You have a board installed which does use dahdi? Did you check
Did you check
asterisk -rx core show translation recalc 10
Am 06.02.2013 13:56, schrieb Thorsten Göllner:
Sorry - I just read you alsways checked the cpu usage. Are all cores
at 100%? Is it the atserisk process which consumes it all?
Am 06.02.2013 13:54, schrieb Thorsten Göllner:
Did you
Hi,
on this site
http://www.voip-info.org/wiki/view/Asterisk+func+callerid
you can read, that since Atserisk 1.8 the command (in dialplan) to hide
the caller id is:
Set(CALLERID(num-pres)=prohib)
I tried to implement it into my AGI-Script, but with no success. Can
please anyone give me a
Am 06.02.2013 16:02, schrieb Steve Edwards:
On Wed, 6 Feb 2013, Thorsten Göllner wrote:
I tried to implement it into my AGI-Script, but with no success. Can
please anyone give me a hint, what is wrong with it:
Set CALLERID(num-pres) prohib
or
Set CALLERID(num-pres)=prohib
Both commands lead
timing module).
BR,
Hristo
On Wed, Feb 6, 2013 at 1:57 PM, Thorsten Göllner t...@ovm-group.com
mailto:t...@ovm-group.com wrote:
Did you check
asterisk -rx core show translation recalc 10
Am 06.02.2013 13:56, schrieb Thorsten Göllner:
Sorry - I just read you alsways
timing module).
BR,
Hristo
On Wed, Feb 6, 2013 at 1:57 PM, Thorsten Göllner t...@ovm-group.com
mailto:t...@ovm-group.com wrote:
Did you check
asterisk -rx core show translation recalc 10
Am 06.02.2013 13:56, schrieb Thorsten Göllner:
Sorry - I just read you alsways
Am 08.02.2013 13:11, schrieb Doug Lytle:
Is there a way to slow down or speed up the speed at which SayDigits
core show application saydigits
[Synopsis]
Say Digits.
[Description]
This application will play the sounds that correspond to the digits of the
given number. This will use the
Hi again,
I did a try on my asterisk 11.2.1 compiled on Ubuntu 12.04 (64 bit)
with a simple Pentium 4 CPU (Intel(R) Pentium(R) D CPU 2.80GHz). I
connected 5 SIP-Users with a ConfBridge. This is my picture:
Please give a a hint where I can
Hi Olivier,
you have to edit /etc/security/limits.conf. Take a look at man
limits.conf.
Some users also modify the Asterisk-Start-Script. You can insert an
ulimit -n 8192 in the Start-Case.
Best regard
-Thorsten-
Am 15.02.2013 18:48, schrieb Olivier:
2013/2/15 Olivier oza_4...@yahoo.fr
Hi,
I am wondering, if there is any tool available, which performs a check
for suspicious entries in the dialplan. For example a non existing
AGI-Script or a double assigned extension ike that:
[context]
exten = *100*,1,AGI(test_app.pl)
...
exten = 190,1,Answer()
...
exten =
What exactly do you mean by crossing channels? Mixed audio? Can
callers hear each other?
Am 19.02.2013 02:07, schrieb Juan Carlos Agudelo:
Hi,
I have installed Asterisk 1.6.2.17-rc2 and I have a strange behavior,
because sometimes they are crossing channels, thus producing unwanted
calls
Ist one channel significant louder than the other? Maybe it is some sort
of crosstalking. Take a look here:
http://es.wikipedia.org/wiki/Diafon%C3%ADa
Am 19.02.2013 16:25, schrieb Juan Carlos Agudelo:
El 19/02/13 03:59, Thorsten Göllner escribió:
What exactly do you mean by crossing channels
Take a look here:
http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device
Am 06.03.2013 13:00, schrieb termo termosel:
Hi,
df -h output:
root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.
Try to set the tmp variable. In your case:
mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make
Am 06.03.2013 13:20, schrieb termo termosel:
Hi,
I read it but I don't find the solution. How Can I alocate more free
space in tmp?
Thanks,
Jordi
Did you execute the make command in the same environment so that make
really uses the TMPDIR directory? (no su or other shell)
Am 06.03.2013 13:37, schrieb termo termosel:
Hi,
the same error, I write your commands:
mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make
but the same error
That should be ok.
Try the following: open 2 shells. In the first one type watch df -h.
In the second one you start the compilation. While compilation is
running watch the first shell. The given command refreshes all 2 seconds
the display and shows the used/free disk space. _Perhaps_ it will
Hi,
I am using Asterisk 11.2.1. I am logging CDRs to a mysql database (via
odbc). The table contains the fields clid and src. Both fields are
varchar(100). But alls entries are without the leading 0. For example
0211 for Germany-Düsseldorf.
Where can I configure that behaviour, please?
Hi,
I have upgraded vom Atserisk 1.6.1.20 to 11.2.1. Most things went fine.
But 1 thing will not work: IAX. I used the same configuration but
Asterisk will not answer the incoming IAX-Call.
When enabling iax debugging I can see the following:
[Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:
I am sure, that my log configuration is correct. NO messages will be
logged other than the posted messages from iax debug.
Am 08.03.2013 16:44, schrieb Rusty Newton:
- Original Message -
From: Thorsten Göllner t...@ovm-group.com
I set verbose and debug to 100 but no(!) message
trigger IAX2 interop issues if your config file for chan_iax2 is not
setup properly. You can read more about it here:
http://downloads.asterisk.org/pub/security/IAX2-security.pdf
With regards to the CTOKEN addition. Hope that helps.
Matthew Fredrickson
Digium, Inc.
On 3/8/13 8:38 AM, Thorsten
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