Quick and drity:
1) Meetme has to be configured to record the media stream.
2) You have to install a streaming server. Maybe ffmpeg could do the job:
https://trac.ffmpeg.org/wiki/StreamingGuide
3) Then your website should be able to get the stream from the streaming
server.
You should be able
Take a look here:
http://asteriskfaqs.org/tag/confbridge/page/2
Am 02.12.2014 03:37, schrieb Bryant Zimmerman:
I am doing dynamic conference bridges using confbridge in asterisk 11.
Is there a way to toggle off an on recording of an ongoing conference call
I have figured out how to record a
On Nov 26, 2014, at 6:12 PM, Thorsten Göllner t...@ovm-group.com
mailto:t...@ovm-group.com wrote:
Am 26.11.2014 11:37, schrieb Antoine Megalla:
Hi,
I am struggling with a very strange issue I have been facing for
the past week;
I have a fresh install of CENTOS 5.11 and I have installed
Am 26.11.2014 11:37, schrieb Antoine Megalla:
Hi,
I am struggling with a very strange issue I have been facing for the
past week;
I have a fresh install of CENTOS 5.11 and I have installed asterisk
1.8.32 form sources.
The asterisk installation went fine but as soon as I start asterisk
Am 27.10.2014 08:54, schrieb Olivier:
2014-10-25 19:33 GMT+02:00 Thorsten Göllner t...@ovm-group.com:
Am 25.10.2014 00:09, schrieb Olivier:
Hello,
I need to play some musiconhold content starting at a random duration
from the start.
Thanks to mode=custom option and either madplay
Am 26.10.2014 00:43, schrieb lee:
Hi,
how can I make asterisk do something when an outgoing call is picked up?
The background is that I would like to record incoming and outgoing
phone calls. In order to do this, I need to play an announcement
telling the person calling or being called
Am 25.10.2014 00:09, schrieb Olivier:
Hello,
I need to play some musiconhold content starting at a random duration
from the start.
Thanks to mode=custom option and either madplay or mpg123 programs, I
could successfully get what I was after on a Debian Wheezy system.
Now I realized sox
Hi there,
I am running
Asterisk 11.9.0
WANPIPE Release: 7.0.10
DAHDI Version: 2.9.0 Echo Canceller: HWEC
libpri version: 1.4.12
When I start the ConfBridge application I get the following warning:
[2014-10-24 14:36:21] WARNING[29177][C-6934]: config_options.c:790
uint_handler_fn: Attempted
Hi,
I have Asterisk 11 with DAHDI (Sangoma E1-Card) running on Ubuntu 12.04
LTS. Asterisk and DAHDI-Drivers are installed from source.
When doing an apt-get upgrade the system packages will be update but
sometimes Asterisk is broken. Which packages do I have to exclude when I
do not have time to
Am 18.09.2014 11:06, schrieb Marek Cervenka:
hi,
i want convert mixmonitor recorded speech audio from wav to mp3 or aac
can you recommend your settings for speech audio? filters, noise
elimination, compression ratio, ...
i will probably use lame
Give sox with compiled mp3-support a try:
Hi there,
I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine
:-) But I am wondering if there is a solution/application which will
enable me to implement voice recognition while playing a voice file
(barge in). So that the caller hears a voice file and can interrupt it
with
Am 04.09.2014 16:44, schrieb motty cruz:
Hi All,
I see this kind of attack on our Asterisk Server, do you know how to
block that IP?
[Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375
handle_request_invite: Call from '' (213.136.81.166:9306
http://213.136.81.166:9306) to extension '34422'
Am 02.09.2014 07:09, schrieb Bryant Zimmerman:
Hey All
We have several AGI scripts that access databases. These work well
most of the time.
The issue we are having is that on rare occasion our script must fail
to a backup database server.
When this occurs it may take up to two seconds to do
Am 14.08.2014 17:22, schrieb Mitch Claborn:
Is it possible (and advisable) to copy menuselect options from
Asterisk 11 to Asterisk 12? If so, is menuselect.makeopts the only
file to copy?
I am not sure - but I would'nt do that. Make a hardcopy from your
console and transcribe the settings
Hi,
I have a question about this here:
Asterisk-Version: 11.10.2
File: res/res_agi.c
Line: 2377
[...]
static int handle_recordfile(struct ast_channel *chan, AGI *agi, int
argc, const char * const argv[])
2304 {
2305 struct ast_filestream *fs;
2306 struct ast_frame *f;
2307
Hi,
we implemented ispeech for voice recognition. I works fine. But you have
to develop an app of your own to do it.
Take a look at http://www.ispeech.org/api (Section 3 Automated Speech
Recognition).
ispeech let you upload a recorded speex file via http-upload and will
return the result
Look for irqbalancer for your distribution:
http://www.tutorialspoint.com/unix_commands/irqbalance.htm
Am 14.05.2014 09:00, schrieb Chandrakant Solanki:
Hello All,
I have 2 Digium card configure on Single machine, which can't share
interrupt across all CPUs and sometimes asterisk reach 100%
That's correct. When you update the kernel package youhave also to
recompile dahdi package.
Am 12.05.2014 07:05, schrieb Lee, John (Sydney):
Hi,
I have noticed it for a while but I just thought about confirming this
with the Asterisk community.
As the compilation of DAHDI will need to
Hi,
it seems, that the caller hangs up immediatly after calling. Try to
reproduce it by yourself. Dial the number (to reach your asterisk
server) and hangup after ~ 0.5 sec (or whatever).
Best regards,
-Thorsten-
Am 30.04.2014 01:11, schrieb Bryce Lowe:
Hello,
I am trying to diagnose an
Is your script really so simple?
Enable agi debugging (agi set debug on) and take look at it when this
happens.
-Thorsten-
Am 30.04.2014 11:47, schrieb Hoggins!:
Hello all,
I have a strange problem with a very simple AGI script, using the GET
DATA command.
When using this command,
Am 28.03.2014 10:32, schrieb Haider Khalil:
Hello Experts,
I want to know if there is any way to modify welcome banner on
asterisk console when I connect using asterisk -r
Hi,
did you compile asterisk from source? Take a look at main/asterisk.c
(line 174 in asterisk v 11.5.1). I think
Take a look at http://www.ispeech.org/
I implemented Speech-Recognition. The API is well documented and easy.
Am 10.01.2014 21:16, schrieb Jai Rangi:
Hello,
Anyone know good quality text to speach engine for building IVRs for
asterisk. Open-source will be nice, but I wont mind paying for
11:56, schrieb Henrik Andresen:
All calls are sip--sip
On 19/12/13 11:32, Thorsten Göllner wrote:
Am 19.12.2013 10:37, schrieb Henrik Andresen:
I have a problem with asterisk. I got ~15 asterisk servers on new
hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1
and 10. No disk
Am 19.12.2013 11:56, schrieb Henrik Andresen:
All calls are sip--sip
On 19/12/13 11:32, Thorsten Göllner wrote:
Am 19.12.2013 10:37, schrieb Henrik Andresen:
I have a problem with asterisk. I got ~15 asterisk servers on new
hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1
Am 19.12.2013 10:37, schrieb Henrik Andresen:
I have a problem with asterisk. I got ~15 asterisk servers on new
hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1 and
10. No disk activity, no ram or swap problem. But asterisk main
process is using up to 300-500% cpu. This happens
Hi,
I made good experienes with Siemens Gigaset C610 IP. This model is about
90 Euro. Configuration via web interface. But encryption (SIPS/SRTP) is
*not* possible with this phones.
-Thorsten-
Am 11.12.2013 11:30, schrieb Mario Giammarco:
Hello,
I need to setup this configuration:
-
Hi,
I am facing a (for me) strange problem. When placing a SIP-Call I
normally get connected and the dialplan is executed. The Call-Flow is
controlled by a PHP-Agi-Script. The script answers the call (via
AGI-Command) and a simple voicefile is played. SOMETIMES(!) I get
disconnected
Maybe this could help you:
http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf
Am 13.09.2013 11:49, schrieb Jonas Kellens:
Hello,
and when I define 11500 - 11954 it should use a random port in this range.
Where is it stated that you MUST use 1-2 ???
Someone else please ?
Did you open a ticket at Sangoma-Site?
What wanpipe driver version do you use?
Is it a production machine? Or can you test it in that way, that you
crossover lines from one card to the other?
Am 04.09.2013 10:48, schrieb DHAVAL INDRODIYA:
Hello List,
I have configure 2 sangoma card each with
Hi,
I am using Asterisk 11.5.1. As far as I understood, the following
configuration allows a sip client only to receive calls (type=peer) but
not to place calls
(http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I place
calls though with this config?
sip.conf
...
[thorsten]
Hi,
I use Asterisk 11.5.1 and it works fine. :)
Now I want to use TLS and media encryption. I followed this guide:
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
When I place a call via Blink-Client (0.5.0) I get connected and Blink
shows 2 locks. The blue lock shows
Thanks a lot. Seems to be a good hidden page, isn't it? ;-)
Am 03.09.2013 14:30, schrieb Steve Totaro:
On Tue, Sep 3, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com
mailto:t...@ovm-group.com wrote:
Hi,
I am using Asterisk 11.5.1. As far as I understood, the following
Permissions: take a look at /etc/udev/rules.d/dahdi.rules. Last line.
OWNER and GROUP should be the same as the user running the asterisk
process (root or asterisk?).
Am 29.08.2013 11:47, schrieb bilal ghayyad:
Hello;
I am installing asterisk and dahdi on ubuntu and I used my username
You should take a look at this options:
type=friend
context=my_context
host=ip_address
Am 26.07.2013 16:52, schrieb jin jan:
Hi all,
I've tried to sen calls to asterisk from different soft switch.
I want to define ip authentication(not register) to an extension for
make call through asterisk.
Additionally you shoudl take a look at sip set debug on (in cli) and
then place a call.
Am 26.07.2013 17:14, schrieb Thorsten Göllner:
You should take a look at this options:
type=friend
context=my_context
host=ip_address
Am 26.07.2013 16:52, schrieb jin jan:
Hi all,
I've tried to sen calls
Enter CLI via /usr/sbin/asterisk -r and execute dialplan reload. Any
errors?
BTW: you should think about upgrading to 1.8 (for example).
Am 25.07.2013 08:49, schrieb Kamlesh Kumar:
Hello
Asterisk version 1.6.2.9.
I want to know is there any limitation on number of contexts or
including
Why not use ODBC?
Am 24.07.2013 13:41, schrieb Prashant Abhang:
Hi,
I was having question about mysql driver support ( not odbc).
Do we still need the asterisk-add-on to be installed for mysql
support. If yes, Which version should be used and from where I should
get it?
Thanks in
Depends on used kernel and perhaps on other hardware you are using.
Am 23.07.2013 00:09, schrieb bilal ghayyad:
Hello
I need to deploy asterisk on production and same thing for DAHDI,
which version is recommended for this?
Regards
Bilal
--
Hi,
where did you change the ulimit? The following command should show you,
if your setting is correct:
asterisk -rx ulimit descriptors
In my installation I edited the limits here:
vi /etc/security/limits.conf
[...]
asterisksoftnofile 8192
asteriskhard
Hi Satish
:) You reminded me of my teacher of old school days.
Very well explained.
I have somewhat similar requirement where I need to play some
announcements to entertain a caller while passing/processing some data
through webservice call ().
do you want to use C or PHP?
-Thorsten-
Take a look here:
http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/
Am 16.06.2013 09:43, schrieb Olivier CALVANO:
Hi
we have a small problems.
We have a Asterisk 1.6.1 old server with music on old.
we have updated to AsteriskNow
Hi there,
does anyone have experience with Asterisk-AGI-Scripts in PHP while using
pthreads in PHP? Are there any limitations or problems known?
Best regards
-Thorsten-
--
_
-- Bandwidth and Colocation Provided by
Is the subdir Horaires readable/executable for User Asterisk/Asterisk?
Did you try to convert it to wav?
Am 17.06.2013 09:47, schrieb Thorsten Göllner:
Take a look here:
http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/
Am 16.06.2013 09
Hi,
some month ago we installed a VoiceRec-Module from Vestec
(https://www.vestec.com/) on Asterisk 11.x. It works so far and you will
find examples for your dialplan. It should be ok for your needs.
-Thorsten-
Am 13.06.2013 23:19, schrieb asterisk users:
Hello list,
'Just wondering if
Hi,
I use a Sangoma A104d-Card (with 4 x germany E1). I process some calls
via an AGI-Script. When parsing the AGI-Variables I can see one that
look like that:
[agi_channel] = DAHDI/i3/211123456-89c
What hat do the values mean in detail, please?
DAHDI : this is clear
i3 : does it mean,
Hi,
I configured in features.conf, that the Dial-App may be cancelled by
pressing the pound key. That works fine. The caller can cancel the
bridged call. BUT can I configure it that way, that the dialing itself
can NOT be cancelled? My dial should only be cancelled by the timeout
or by the
Hi,
I tried to use Early Media:
exten = 1,1,Playback(demo-thanks,noanswer)
same = n,Hangup()
But when calling my extension I do not hear the voicefile - I only hear
the ring tone. In the Asterisk-Log I can see, that the voicefile is played.
I got the same result when using Progress() in
Well, the question is, what your secretary wants to do. Only see the
CDRs or more? Realtime? One simple method would be to mail her the
CSV-File, so she can open it with Excel or Calc (Open Office).
Am 23.04.2013 16:35, schrieb aristidis tsitras:
Hi. i am running asterisk in a low powered
Hi,
I have the following setup:
Ubuntu 12.04.02 LTS (64 bit)
Asterisk 11.2.1
Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected)
WANPIPE Release: 3.5.28
DAHDI Version: 2.6.1 Echo Canceller: HWEC
libpri version: 1.4.12
I call via sip into the dialplan. Then I do a
a call use
*
**hangup request channel*
where channel is the exact id of your channel as you would receive it via
*core show channels*
yves
Am 11.04.2013 10:56, schrieb Thorsten Göllner:
Hi,
I have the following setup:
Ubuntu 12.04.02 LTS (64 bit)
Asterisk 11.2.1
Sangoma 4-Port-Card (A104d
What hardware do you use? Do your have some E1 or T1 Ports? Maybe one or
more of this ports is down.
Am 26.03.2013 17:57, schrieb Salaheddine Elharit:
Hello,
i have all the time this warning i use asterisk 1.4 all works without
issue i don't have any problem (i can use the inbound and
You do use only span 1 and 6? So the other ports are not plugged? That
is the cause for the warnings. I use a Sangoma E1-Card. The configure
script gives me the option unused for any port. Maybe your configure
script offers you the same option.
Am 27.03.2013 11:54, schrieb Salaheddine
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
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:
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
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
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
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
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 =
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
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
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
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
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
= 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
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 =
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
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
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:
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
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
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 ;-)
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
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
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
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
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 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
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 =
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
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
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,
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-
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