On Sun, 10 Jan 2016, Ian Harding wrote:
Inbound route: Don't Care
Queue: Yes
Extension: Don't Care
What front end are you using?
What version of Asterisk, OS, etc?
You may get more interest on a mailing list specific to that front end.
--
Thanks in advance,
Hello!
I inherited an asterisk setup that works fine, but I'd like to make a
change and it's not working the way I want.
Right now, our incoming calls are recorded at the "Queue" level. It
works but it records hold music, etc and when the call is sent to an
extension, the "Channel ID" (I think
I'm not super sure about the names for these various things.. but it's
PBX in a Flash FreePBX 12.0.76.2 on Centos6.5
2.6.32-431.1.2.0.1.el6.x86_64 (SMP) x86_64.
I see PBX in a Flash has a forum so I'll hit them up too. Thanks!
On 01/10/2016 01:39 PM, Steve Edwards wrote:
> On Sun, 10 Jan 2016,
Hi all,
on my atserisk box call recording and cdr doesn't work. In the log files I
have a strange entry - does this have something to do with that?
Version: Asterisk 13.1.0
Host: debian wheezy 7.7
Thanks a lot for a brief hint .
Walter.
***
[2015-Jan-26 11:34:04]
I have a call recording (audio) requirement that isn't addressed by
local Monitor/Record features.
All signalling and media currently pass through the Asterisk servers,
so that won't be an issue.
Instead of locally recording audio, for certain calls I need to add
what is effectively a 3rd leg to
Tom Browning wrote:
I have a call recording (audio) requirement that isn't addressed by
local Monitor/Record features.
All signalling and media currently pass through the Asterisk servers,
so that won't be an issue.
Instead of locally recording audio, for certain calls I need to add
what is
On Thu, Dec 13, 2012 at 9:45 AM, Joshua Colp jc...@digium.com wrote:
If you don't want to incur the overhead of a full blown conference bridge
you can use ChanSpy to spy on a channel. It will provide a mixed stream of
the incoming and outgoing part of the channel. So essentially use Originate
I am trying to record calls on demand both inbound and outbound calls. I can
record outbound calls just fine but not inbound calls or calls from an
internally between extensions. I am using the latest asterisk 1.8.x certified
version.
On an outbound call I see:
== Using SIP RTP CoS mark 5
You should simplify until you have something that works, then add your
conditions back in one line at a time.
On 12-08-28 11:05 AM, Josh Hopkins wrote:
-- Executing [s@macro-one-touch-record:3]
ExecIf(SIP/1010-0161, 1?MacroExit()) in new stack
This is where the inbound call is
-users] Call recording and transfer issue (asterisk
1.8)
On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote:
Hi
I'm having a problem with the entirety of a call being recorded in
the
following scenario
I'm using asterisk 1.8.7.0
Person A (asterisk peer) calls Person
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 18, 2012 9:58:47 AM
Subject: Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8)
On Thu
On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote:
Hi
I'm having a problem with the entirety of a call being recorded in the
following scenario
I'm using asterisk 1.8.7.0
Person A (asterisk peer) calls Person B (not on asterisk, real world
number via a SIP trunk)
Mixmonitor is
Hello,
I am able to get the call recording file path of each call in the CDR. How
can I get the realtime call recording streaming?
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
Hi
I'm having a problem with the entirety of a call being recorded in the
following scenario
I'm using asterisk 1.8.7.0
Person A (asterisk peer) calls Person B (not on asterisk, real world
number)
Mixmonitor is invoked by Person A in the outbound context and
AUDIOHOOK_INHERIT(MixMonitor)=yes is
Dan et al;
Okay - I have declared DYNAMIC_FEATURES=MixMonApp in the [global]
section of my extensions.conf
I dial into my trunk, the softphone rings, I answer and I press '*1' - I
hear the tones, but I see no indication in the Asterisk CLI and I see no
.wav file being created.
I must
What am I missing?
Not reading the DTMF tones. Thus not executing the macro.
Start by checking you are receiving the DTMF tones.
Edit logger.conf and add dtmf to the console line.
So it looks something like this:-
console = notice,warning,error,dtmf
Then see if you are receiving the tones
I set the logger.conf to show reading of DTMF tones as per your instructions
below. This is what I see:
[Apr 10 11:57:19] DTMF[14783] channel.c: DTMF begin '*' received on
SIP/6000-002e
[Apr 10 11:57:19] DTMF[14783] channel.c: DTMF begin passthrough '*' on
SIP/6000-002e
[Apr 10
Hi Dan et al;
I had actually done a sip reload, dialplan reload, module reload
res_features.so and logger reload.
However, upon seeing your email, I restarted the Asterisk server
completely to see if I had missed anything. I still see the same behaviour.
I am at a loss.
Glen
On 4/10/2011
I am at a loss.
Can you pastebin the following:-
- Run asterisk-cvvvddd and paste the output
- Pastebin your features.conf
- Pastebin your extensions.conf
I'll see if I can spot anything obvious.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/
Hey!
I did a little bit of digging - and I solved my issue!
Apparently, in my extensions.conf, I specified the wrong variable.
I had DYNAMIC_FEATURES=callrec (which is the name of my macro)
I changed it to DYNAMIC_FEATURES=MixMonApp, which is what is it aliased
to in the features.conf.
DYNAMIC_FEATURES=MixMonApp, either declared in your globals section of
extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion on a
per channel basis in extensions.conf.
Sorry, i forgot to mention that one.
Dan Journo
Kesher Communications (UK)
Business Phone
If you don't want to record every call, you can give the operator the option
of press *1. We did this by adding the following to features.conf:-
MixMonApp = *1,self/both,Macro,mixmon
As brought up in another post, I forgot to add the following:-
DYNAMIC_FEATURES=MixMonApp, either
Dan et al;
This looks like a perfect solution.
However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.
If I try to implement it adding the
Dan et al;
This looks like a perfect solution.
However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.
If I try to implement it adding the
On Fri, Apr 8, 2011 at 3:35 PM, Silver Thorne szilvertho...@gmail.comwrote:
Dan et al;
This looks like a perfect solution.
snip
It pretty much is. I've used it in similar situations. I was just about to
respond to your original post, but I see you reposted here, so I'll respond
here.
Hello Everyone;
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I
have to modify it to make it easier to use, I do not mind.
Does anyone know of any opensource or otherwise solutions out there that
I
On 6 Apr 2011, at 11:54, Silver Thorne wrote:
Does anyone know of any opensource or otherwise solutions out there that I
can try out?
Asterisk. Google it. If you're too lazy, Google MixMonitor. If you're too lazy
for that:
http://www.voip-info.org/wiki/view/MixMonitor
S
--
On Wed, Apr 6, 2011 at 5:54 AM, Silver Thorne szilvertho...@gmail.comwrote:
Hello Everyone;
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I have
to modify it to make it easier to use, I do not mind.
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I
have to modify it to make it easier to use, I do not mind.
Does anyone know of any opensource or otherwise solutions out there that
I can try
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, April 06, 2011 6:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call recording - methodology
I am
Tilghman,
When you say reformat the audio, do you mean sample rate and bits per
sample, etc...or do you mean the format in which each packet of data is
structured ? I just want to make sure I know which one I'd be dealing with
if recording a call that was using one of the higher quality codecs
On Wednesday 09 February 2011 03:50:51 Sherwood McGowan wrote:
Tilghman,
When you say reformat the audio, do you mean sample rate and bits per
sample, etc...or do you mean the format in which each packet of data is
structured ? I just want to make sure I know which one I'd be dealing
with
On Wed, Feb 9, 2011 at 12:31 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote:
On Wednesday 09 February 2011 03:50:51 Sherwood McGowan wrote:
Tilghman,
When you say reformat the audio, do you mean sample rate and bits per
sample, etc...or do you mean the format in which each packet of data
Hi
We're getting requests coming in for higher quality audio in our call
recordings. We currently use MixMonitor and everything is being saved in
it's native 8000Hz, 16 bit wav format.
I have seen information on using Monitor and specifying a conversion to
mp3 when the call ends and the 2
On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
We're getting requests coming in for higher quality audio in our call
recordings. We currently use MixMonitor and everything is being saved in
it's native 8000Hz, 16 bit wav format.
I have seen information on using
On Tue, 2011-02-08 at 05:40 -0600, Sherwood McGowan wrote:
On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik i...@pack-net.co.uk
wrote:
Hi
We're getting requests coming in for higher quality audio in
our call
recordings. We currently use MixMonitor and
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Tuesday, February 08, 2011 6:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call Recording audio file quality
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Call Recording audio file quality query
On Tue, 2011-02-08 at 05:40 -0600, Sherwood McGowan wrote:
On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik i...@pack-net.co.uk
wrote:
Hi
That answer was pretty much what I was expecting. Just wanted to make
sure.
Glad to be of service :D
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
On Tue, Feb 8, 2011 at 6:01 AM, fai...@vopium.com wrote:
But if you are getting calls all the way on VoIP then you can have calls in
HD audio using HD audio codec on all locations (Server and Client). In that
case you either need use some available 3rd party solution which uses packet
@lists.digium.com
Subject: Re: [asterisk-users] Call Recording audio file quality query
On Tue, Feb 8, 2011 at 6:01 AM, [mailto:fai...@vopium.com] fai...@vopium.com
wrote:
But if you are getting calls all the way on VoIP then you can have calls in HD
audio using HD audio codec on all locations (Server
: Re: [asterisk-users] Call Recording audio file quality query
On Tue, Feb 8, 2011 at 6:01 AM, fai...@vopium.com wrote:
But if you are getting calls all the way on VoIP then you can have calls
in HD audio using HD audio codec on all locations (Server and Client). In
that case you either need use
On Tuesday 08 February 2011 06:34:56 Sherwood McGowan wrote:
On Tue, Feb 8, 2011 at 6:01 AM, fai...@vopium.com wrote:
But if you are getting calls all the way on VoIP then you can have
calls in HD audio using HD audio codec on all locations (Server and
Client). In that case you either need
Hi All,
We have a requirement to record over 60 simultaneous calls. Our recording
facilities are implemented using Monitor() over AMI. The thing we have
noticed that making 60 simultaneous call recordings using wav CPU load is
significantly higher (around 2 times more) than using gsm. Even
What format are the actual calls in? Are they in G.711u/a format or
are they in something else (perhaps gsm?) format? I'm asking to find
out if Asterisk would need to transcode them.
On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius
vilius.adamkavic...@invade.net wrote:
Hi All,
We have a
Hi Joel,
We have a meetme on which we are landing two G.711 alaw calls, one coming
from TDM another from SIP. Once we those parties are in the conference we
are adding one more leg using Local channel and starting to record it.
Surely it would be logical if it would be less overhead recording
On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius
vilius.adamkavic...@invade.net wrote:
Hi All,
We have a requirement to record over 60 simultaneous calls. Our recording
facilities are implemented using Monitor() over AMI. The thing we have
noticed that making 60 simultaneous call
Hi David,
Looking at MOS G.711alaw wav most definitely has the higher score than gsm.
Moreover recording in gsm is more CPU intense than wav. Therefore your
suggestion to do more CPU intense recording and afterwards use system
resources to convert it back to wav is not a solution. Also some of
On Mon, Nov 22, 2010 at 03:28:27PM +, Vilius Adamkavicius wrote:
Hi David,
Looking at MOS G.711alaw wav most definitely has the higher score than gsm.
Moreover recording in gsm is more CPU intense than wav. Therefore your
suggestion to do more CPU intense recording and afterwards use
WAV or wav? One of these has GSM-encoding inside a WAV formatted
envelope. That said, I wouldn't expect that to have any noticeable
CPU utilization above that of GSM. If you are using the non-GSM
version of WAV, then I am as baffled as you - hopefully someone who
knows more about this can help.
We are using wav, not WAV. I believe WAV is the one with GSM. Its a very
good idea to compare WAV against wav, will run some tests and come back with
outcome, will try Tzafrir's suggestion as well.
Thanks guys
Vilius.
On 22 November 2010 16:31, Joel Maslak jmas...@antelope.net wrote:
WAV or
Hi,
I'm using the CallTime and a few other variables to name a recording so that I
can then take the wav file name and see when it was recorded, and what the
recording contains.
However, since ${CDR(start)} contains a space in part of the date, the filename
becomes corrupted when I use samba
Hi,
On 09/15/2010 09:02 PM, Dan Journo wrote:
Hi,
I'm using the CallTime and a few other variables to name a recording so that
I can then take the wav file name and see when it was recorded, and what the
recording contains.
However, since ${CDR(start)} contains a space in part of the
Is there any way to prevent the end user hearing the *1 key tones when the
touch recording is activated?
Thanks
Dan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Hi,
1) I want to create add *1 call recording and wanted to know whether the
file is created during recording or only after? I want to syncronise the
recorded files with my web server (on a different machine (Windows)) so I need
a way of telling when the recorded call has ended before
1) I want to create add *1 call recording and wanted to know whether the file
is created during recording or only after? I want to syncronise the
recorded files with my web server (on a different machine (Windows)) so I
need a way of telling when the recorded call has ended before copying
1) The file is written in real time. Personally I would add a dialplan
entry into the 'h' extension to move the recording into a different
directory when the call ends. That will make your syncronisation much
easier.
Dan Journo wrote:
Hi,
1) I want to create add *1 call recording
The DTMF mode can cause problems. The main rule is to make sure
everything is using the same method. I normally use SIP-Info as the
method as it allows to rtp stream to be switch directly between the two
end points but asterisk still sees all the dtmf digits.
Dan Journo wrote:
1) I want to
1) I use a bash script I wrote to check if call recordings are being
written to and if not then move them. I move them to a locally mounted
NFS share but this will work with any type of locally mounted share
(Samba for Windows). I run the script every minute with cron. It also
sorts the
On Thu, 2010-09-02 at 07:21 -0400, Dan Journo wrote:
1) I want to create add *1 call recording and wanted to know whether the
file is created during recording or only after? I want to syncronise the
recorded files with my web server (on a different machine (Windows)) so I
need a way of
How do you sort out the issue of having 2 wav files per call?
Also, when I press *1, asterisk thinks that both the caller and the callee have
pressed *1 and therefore it starts recording twice (therefore making 4 wav
files). Any idea what's going on there?
Heres the CLI output:-
-- Called
On Thu, 2010-09-02 at 08:20 -0400, Dan Journo wrote:
How do you sort out the issue of having 2 wav files per call?
Also, when I press *1, asterisk thinks that both the caller and the callee
have pressed *1 and therefore it starts recording twice (therefore making 4
wav files). Any idea
On 09/02/2010 01:09 PM, Ishfaq Malik wrote:
On Thu, 2010-09-02 at 07:21 -0400, Dan Journo wrote:
1) I want to create add *1 call recording and wanted to know whether the
file is created during recording or only after? I want to syncronise the
recorded files with my web server (on a
I have our recordings written to a solid state drive rather than straight to
storage disks then moved to long term storage to avoid this problem.
Not an option for me at the moment.
I'm running Asterisk on a cloud to reduce startup costs.
Once I reach around 1,000 extensions, I'll move over
In asterisk.conf, use these options:-
cache_record_files = yes ; Cache recorded sound files to another directory
during recording
record_cache_dir = /tmp ; Specify cache directory (used in cnjunction with
cache_record_files)
--
Regards,
Prince Singh
Drishti-Soft Solutions Pvt Ltd
W:
Dan Journo wrote:
Thanks for that.
I really appreciate it!
Dan
As pointed by the follow-ups, note that the recordings are not taken
from the monitor but from an upload folder inside, the dialplan takes
care to move there the files for the ended files only issuing a 'System'
command.
Steve Edwards wrote:
On Tue, 13 Oct 2009, Dan Journo wrote:
To avoid the problem of deleting/copying calls that are still being
recorded, I could record the call into a temp directory. Then using the
dial plan, I could copy the temp recording into the ftp root directory
once the call
thoughts.
-Elliot
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
Sent: Tuesday, October 13, 2009 3:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call
On Tue, 13 Oct 2009, Elliot Otchet wrote:
To Steve's other point, you could put all of this into an AGI
program/script, but you'll still also need a fallback mechanism to
actually copy the files to the remote server in the event that it is
unavailable/unreachable. To me, having two lines
You can try using NFS. Also you can pay some one to write script that would
move the files over on hang up.
- Original Message -
From: Dan Journo
To: asterisk-users@lists.digium.com
Sent: Monday, October 12, 2009 01:15
Subject: [asterisk-users] Call Recording and Posting
Dan Journo wrote:
I'm working on a call recording solution. I would like recordings to
either be automatically uploaded via FTP, or posted to a URL for
processing by our main server.
Is Asterisk capable of doing this or will I have to create a separate
application that monitors a temp
Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
Sent: 12 October 2009 12:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Recording and Posting
Dan Journo wrote:
I'm
Dan Journo wrote:
Thank you for replying. I hadn't thought about the problem of simultaneous
calls. It would be a problem if a number of calls ended at the same time.
If you can post it, the script would really be helpful as I'm only a beginner
with Linux
The script is very simple and
: Re: [asterisk-users] Call Recording and Posting
Dan Journo wrote:
Thank you for replying. I hadn't thought about the problem of simultaneous
calls. It would be a problem if a number of calls ended at the same time.
If you can post it, the script would really be helpful as I'm only a beginner
On Behalf Of Ivan Stepaniuk
The script is very simple and far from complete, it just moves the
content into the mounted FTP directory. It has some verbose output as it
is run from inside another script that redirects the output to a log
file.
What happens if the script is run while a
-Commercial Discussion
Subject: Re: [asterisk-users] Call Recording and Posting
On Behalf Of Ivan Stepaniuk
The script is very simple and far from complete, it just moves the
content into the mounted FTP directory. It has some verbose output as
it
is run from inside another script
On Tue, 13 Oct 2009, Dan Journo wrote:
To avoid the problem of deleting/copying calls that are still being
recorded, I could record the call into a temp directory. Then using the
dial plan, I could copy the temp recording into the ftp root directory
once the call has ended.
True, but if
Hello,
I'm working on a call recording solution. I would like recordings to
either be automatically uploaded via FTP, or posted to a URL for
processing by our main server.
Is Asterisk capable of doing this or will I have to create a separate
application that monitors a temp directory for
under enough load, you
might need another alternative.
-Elliot
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Sunday, October 11, 2009 7:15 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call
On Mon, 12 Oct 2009, Dan Journo wrote:
I'm working on a call recording solution. I would like recordings to
either be automatically uploaded via FTP, or posted to a URL for
processing by our main server.
Is Asterisk capable of doing this or will I have to create a separate
application that
On Sun, Jun 7, 2009 at 12:51 PM, Joao Gomes
Pereiragomespere...@startel.pt wrote:
Hello to all
I'm trying to record the calls going to my queues, but asterisk creates
2 files, one with the inbound and another with the outbound sound.
I know Sox should mix the 2 files automatically in the end,
David Backeberg escribió:
On Sun, Jun 7, 2009 at 12:51 PM, Joao Gomes
Pereiragomespere...@startel.pt wrote:
Hello to all
I'm trying to record the calls going to my queues, but asterisk creates
2 files, one with the inbound and another with the outbound sound.
I know Sox should mix the 2
You should look on the log for when the sox command is called, if the
invocation makes sense or not.
l.
2009/6/7 Joao Gomes Pereira gomespere...@startel.pt
Hello
I did as you told me, but the problem remains.
Im using Asterisk 1.2.x
and this is my config:
queues.conf
Hello to all
I'm trying to record the calls going to my queues, but asterisk creates
2 files, one with the inbound and another with the outbound sound.
I know Sox should mix the 2 files automatically in the end, but this
isn't happening.
I have sox installed in my server.
How can I force Sox
Hi,
I had similar issue which happened when record option was mentioned in
both agents.conf and queues.conf. When I commented the recordagentcalls
option in agents.conf, it started to work. Mention the monitor option
only in the queues.conf file. Do try.
Regards,
Kurian Thayil.
On Sun,
Hello
I did as you told me, but the problem remains.
Im using Asterisk 1.2.x
and this is my config:
queues.conf -
[general]
persistentmembers = no
[queue_1]
persistentmembers = no
monitor-format=wav
monitor-join=yes
monitor-type=MixMonitor
wrapuptime=3
I m recording every call, and i want to remove the recorded call at the end
of call, when the callee doesn't
want the call beeing recorded.
Maybe someone can point me in the right direction, having agents with
callbacklogin and recording enabled in
agents.conf. So if the callee doesn't want the
...@lists.digium.com] On Behalf Of Christian
Gansberger
Sent: Tuesday, April 28, 2009 4:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call recording - posible to remove recorded fileat
the end of the call
I m recording every call, and i want to remove
Where did I make mistake ?
On Thu, Jan 29, 2009 at 1:07 AM, David @ULC ucoms2...@gmail.com wrote:
http://download.vicidial-group.com/svn/agc_2-X/trunk/docs/SCRATCH_INSTALL.txt
vi /usr/local/apache2/conf/httpd.conf
add the following lines:
AddType
David @ULC schrieb:
Where did I make mistake ?
You posted (even re-posted) a question about Vicidial and Apache
configuration on asterisk-users.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com -
Modified httf.conf file and added :
--
Alias /recordings/ /var/spool/asterisk/monitorDONE/
Directory /var/spool/asterisk/monitorDONE
Options Indexes MultiViews
AllowOverride None
Order allow,deny
Allow from all
/Directory
Created a folder
dud please go to asterisk.org support and download the asterisk book and
then READ IT.
David
2009/1/28 David @ULC ucoms2...@gmail.com
Modified httf.conf file and added :
--
Alias /recordings/ /var/spool/asterisk/monitorDONE/
Thanks for your advice , but I asked for expert guidance as I read the doc
and it says like that. but somehow It didn't work out.
On Thu, Jan 29, 2009 at 12:13 AM, David @ULC ucoms2...@gmail.com wrote:
Modified httf.conf file and added :
--
http://download.vicidial-group.com/svn/agc_2-X/trunk/docs/SCRATCH_INSTALL.txt
vi /usr/local/apache2/conf/httpd.conf
add the following lines:
AddType application/x-httpd-php .php .phtml
LoadModule php4_module libexec/libphp5.so
http://download.vicidial-group.com/svn/agc_2-X/trunk/docs/SCRATCH_INSTALL.txt
vi /usr/local/apache2/conf/httpd.conf
add the following lines:
AddType application/x-httpd-php .php .phtml
LoadModule php4_module libexec/libphp5.so
, January 28, 2009 1:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Call Recording Alias
Thanks for your advice , but I asked for expert guidance as I read the doc
and it says like that. but somehow It didn't work out.
On Thu, Jan 29, 2009 at 12:13 AM, David
you aren't giving enough info
you should use vicidialnow it is an out of the box system.
http://vicidialnow.org/blog
and you should start whit Linux and asterisk with something more modest.
David
2009/1/28 David @ULC ucoms2...@gmail.com
Thanks for your advice , but I asked for expert
I wanted to setup Oreka to monitor calls on a trixbox box I have
setup. Oreka doesn't seem to be catching all of the calls
though I have port mirroring setup on the port that trixbox is
connected to, mirrored to the port Oreka is connected to.
I have read
Hello folks,
I wanted to setup Oreka to monitor calls on a trixbox box I have setup.
Oreka doesn't seem to be catching all of the calls though I have port
mirroring setup on the port that trixbox is connected to, mirrored to the
port Oreka is connected to.
I have read that Asterisk
Chris Rowson wrote:
I wanted to setup Oreka to monitor calls on a trixbox box I have
setup. Oreka doesn't seem to be catching all of the calls
though I have port mirroring setup on the port that trixbox is
connected to, mirrored to the port Oreka is connected to.
I
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