On 10/17/21 12:59 PM, cio-al...@playerschool.edu wrote:
I did test manually and the NFS mount works fine. I do create a
directory and it shows at the server.
I am using containers, indeed. How can it be affecting Asterisk that I
am using LXC containers?
I'm by no means an expert in containers
I did test manually and the NFS mount works fine. I do create a
directory and it shows at the server.
I am using containers, indeed. How can it be affecting Asterisk that I
am using LXC containers?
On 2021-10-16 11:34, Dave Platt wrote:
I did not explain myself well, for this I apologize.
The
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of Dave Platt
Sent: Saturday, October 16, 2021 1:35 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] recording not working to NFS
> I did not explain myself well, for this I apologize.
> The
I did not explain myself well, for this I apologize.
The files never appear on the NFS mount, only in the local drive.
Restarting Asterisk with the mount on does not fix it.
Asterisk simply ignores the mount and writes to the local drive.
But the mount is fine, I can create a dir and it appears
Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Telium Technical Support
Subject: Re: [asterisk-users] recording not working to NFS
I did not explain myself well, for this I apologize.
The files never appear on the NFS mount, only in the local drive.
Restarting Asterisk with the mount on
@lists.digium.com
Subject: [asterisk-users] recording not working to NFS
I have an NFS mount and I am trying to record to it. The mount works
fine, I create a directory and it shows on the server, I delete it and
it gets deleted at the server, but Asterisk 16-latest is always
recording to the local drive, it
?
-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of cio-al...@playerschool.edu
Sent: Wednesday, October 13, 2021 1:37 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] recording not working to NFS
I have an NFS mount and I am trying to
I have an NFS mount and I am trying to record to it. The mount works
fine, I create a directory and it shows on the server, I delete it and
it gets deleted at the server, but Asterisk 16-latest is always
recording to the local drive, it ignores the NFS mount.
Once I unmount the directory, the r
I'd recommend trying the support resources for the GUI that is managing
your asterisk installation.
This looks like it might be FreePBX dialplan logic to me, Most likely it
won't be something that the list will be able to help you modify.
http://community.freepbx.org/ is the FreePBX Community sup
This must be something simple. We have recording starting on all our
inbound calls and it works fine. I'd like to have the option to stop
recording when transferred to certain extensions.
I set extension 7077 to "Never" and the log makes it look like the
recording was stopped but it was not.
An
Leandro Dardini wrote:
Hi,
I'd like to record the barged call... but whichever leg of the call I
try to barge, my speaking is never recorded using MixMonitor. Any idea
about the reason?
The only suggestion I have really is to insert a Local channel in the
mix and record on the real one while b
Hi,
I'd like to record the barged call... but whichever leg of the call I try
to barge, my speaking is never recorded using MixMonitor. Any idea about
the reason?
Leandro
--
_
-- Bandwidth and Colocation Provided by http://www.ap
On Fri, Jul 17, 2015 at 9:09 AM, Luca Bertoncello
wrote:
> Rusty Newton schrieb:
>
> > Perhaps the incoming calls are routed through different dialplan and in
> > that Dial you do not have the proper options? The dialplan you posted
> > appears to be for dialing an explicit outbound number.
>
>
Rusty Newton schrieb:
> Perhaps the incoming calls are routed through different dialplan and in
> that Dial you do not have the proper options? The dialplan you posted
> appears to be for dialing an explicit outbound number.
YES!! That was the problem!
I just added "xX" to the previous Dial and
On Thu, Jul 16, 2015 at 3:37 AM, Luca Bertoncello
wrote:
> Hi list!
>
> I'm trying to configure Asterisk to record incoming calls, if the called
> press *3.
> I added in features.conf:
>
> automixmon => *3
>
> then, in my dialplan:
>
> exten => 1,n,Dial(SIP/004935,20,RcxX)
>
> Well, if I
Hi list!
I'm trying to configure Asterisk to record incoming calls, if the called
press *3.
I added in features.conf:
automixmon => *3
then, in my dialplan:
exten => 1,n,Dial(SIP/004935,20,RcxX)
Well, if I **CALL** a number I'm able to record the call, but if I'll be
called, and press
Anurag Rana wrote:
Hi All,
Kia ora,
I am calling mobile numbers from Soft-phone and recording the call.
In recording the level of sound from the receiver's side is perfect
(loud enough) but my voice's sound level is very weak. I barely can hear
it.
During the call receiver is able to hear me
Hi All,
I am calling mobile numbers from Soft-phone and recording the call.
In recording the level of sound from the receiver's side is perfect (loud
enough) but my voice's sound level is very weak. I barely can hear it.
During the call receiver is able to hear me. But in recording my part of
con
converting wav to mp3
[sub-Monitor-Init]
exten => s,1,NoOp(Monitor Init)
exten => s,n,Set(_X-SRC_CHANNEL=${CHANNEL})
exten =>
s,n,Set(recMonitorFName=${STRFTIME(${EPOCH},,%Y_%m_%d)}/${STRFTIME(${EPOCH},,%Y_%m_%d_%H_%M_%S)}-${FILTER(0-9a-zA-Z,${ARG1})}-${FILTER(0-9a-zA-Z,${ARG2})})
exten => s,n,
On Tue, Jul 1, 2014 at 9:39 PM, binary wrote:
>
> i would go for recording into wav.
> then at regular intervals eg every night at 01:00 i would start a script to
> convert the wav to mp3 and then delete the wav files.
> it is really easy.
>
>
This method works for us too.
We get around 40 GB o
Please don't top-post.
Please trim irrelevant posts.
From: Tiago Geada
mixmonitor has a argument that is a script ran just as the recording is
finished.
we use this to move the file from ramfs to final destination.
you can use it to use sox and convert it...
On Thu, 3 Jul 2014, andrew Co
Can you explain?
Sent from Samsung Mobile
Original message From: Tiago Geada
Date:03/07/2014 9:04 PM (GMT+02:00)
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
recording in mp3
no need.
mixmonitor has a argument that is a
no need.
mixmonitor has a argument that is a script ran just as the recording is
finished.
we use this to move the file from ramfs to final destination.
you can use it to use sox and convert it...
On 2 July 2014 18:54, Dave Platt wrote:
>
> > Problem with this is client needs to listen to t
> Problem with this is client needs to listen to the call recordings and my
> interface will only display .wav or .mp3 so they will moan if they have to
> wait until the next day for today's recordings
If you're up to writing a bit of shell script, and are running
on Linux, you could automate t
Currently using tikal crystal call recording
Do you guys know of any better ones?
Sent from Samsung Mobile
Original message From: binary
Date:01/07/2014 6:33 PM (GMT+02:00)
To: asterisk-users@lists.digium.com Subject: Re:
[asterisk-users] recording in mp3
what is your
obile
Original message
From: binary
Date:01/07/2014 6:09 PM (GMT+02:00)
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] recording in mp3
i would go for recording into wav.
then at regular intervals eg every night at 01:00 i would start a
script to convert the wav t
ssage
> From: binary
> Date:01/07/2014 6:09 PM (GMT+02:00)
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] recording in mp3
>
> i would go for recording into wav.
> then at regular intervals eg every night at 01:00 i would start a script
> to co
6:09 PM (GMT+02:00)
To: asterisk-users@lists.digium.com Subject: Re:
[asterisk-users] recording in mp3
i would go for recording into wav.
then at regular intervals eg every night at 01:00 i would start a script to
convert the wav to mp3 and then delete the wav files.
it is really easy.
On
i would go for recording into wav.
then at regular intervals eg every night at 01:00 i would start a script
to convert the wav to mp3 and then delete the wav files.
it is really easy.
On 30/6/2014 23:30, Scott Griepentrog wrote:
You will not be able to able to save much space if any by using
please help in packet2 packet bridging concept
I want peer2peer call and replay all the audio traffic via rtp directly to
destination on public network behind the nat
On Tue, Jul 1, 2014 at 1:41 AM, andrew Colin wrote:
> Hey guys
>
> Is it possible to record with mixmonitor straight into mp3.
>
You will not be able to able to save much space if any by using MP3
instead of ulaw or wav -- at least not without expending a lot of CPU time
to encode the file at a very low bitrate which sounds pretty bad even with
just speech. One of the better space savings options for recordings or
voicemai
Hey guys
Is it possible to record with mixmonitor straight into mp3.
I am trying to reduce disk space and want my calls to be recorded in mp3
Instead of wav.
Sent from Samsung Mobile
Original message From: Sameer Rathod
Date:30/06/2014 9:23 PM (GMT+02:00)
To: asterisk-
I'm running 10.7.1 (yes, I know it's old, but this may be a problem in
later versions too) and had a conference being recorded via:
Set(CONFBRIDGE(bridge,record_conference)=yes)
The bridge started out at 8KHz despite one HD device. But when the
second came in (G.722), it switched to 16KH
mailto:asterisk-users@lists.digium.com>>
Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI
hi,
the "music" heard by MoH is configurable... so if you want silence...
But "hold" could e.g. also be done by transferring a caller into a dynamic
meetme room...
yves
lto:asterisk-users@lists.digium.com>>, Henrik Westerberg
mailto:henrik.westerb...@ain.se>>
Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI
Hi,
so if your are ok with the way you solved part 1... alright, lets go
to part 2..
but again... hu.. I don´t understand..
what do you m
ers Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>,
Henrik Westerberg mailto:henrik.westerb...@ain.se>>
Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI
Hi,
so if your are ok with the way you solved part 1... alright, lets go to part 2..
but again
case.
Any suggestions?
Regards,
Henrik
Från: "Yves A." mailto:yves...@gmx.de>>
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Datum: torsdag 7 mars 2013 20:10
Till: Asterisk Users Mailing List - Non-Commercial Di
Hi,
Ok but when I use the macro the recording doesn´t start until the call is
answered which is a plus. It´s easy to trim away silence of course though.
But according to the documentation it seems like DeadAgi is obsolete in
Asterisk 1.6 and later, that AGI should be used instead.
Regards,
Henri
irst case.
Any suggestions?
Regards,
Henrik
Från: "Yves A." mailto:yves...@gmx.de>>
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Datum: torsdag 7 mars 2013 20:10
Till: Asterisk Users Mailing List - Non-Commer
As far as i understand your requirements, there is no need to use
macro for recording, You can directly call mixmonitor before Dial
application in your dialplan with required options. For transfer of
file, you are using AGI in h priority. However, you have to use
DeadAgi in h extension. As your ch
hi,
hard to understand, what your objective is... at least for me ;-)
so you want to establish a call (triggered by ami) between two partys,
record the conversation
and save the file to a(nother) server (afterwards), right?
and another task is to establish (also ami triggered) a call to a mob
Hi,
I am developing a call recording application on Asterisk 11.2 and have this
configuration in my dialplan:
[macro-ccdev2-rec]
exten => s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten => _X.,1,NoOp(Will send call to ${EXTEN})
exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z)
[outgoing-originate-r
""]?Set(THISEXTEN=${IF($["${REALCALLERIDNUM}"=""]?${DIALEDPEERNUMBER}:${FROMEXTEN})}))
exten => s,n(stopped),Playback(beep&beep)
exten => s,n,MacroExit()
;--== end of [macro-one-touch-record] ==--;
From: asterisk-users-boun...@lists.digium.com
[mailto:a
you need to provide dial plan for macro-one-touch-record.
i think there is something which records outgoing only
On Wed, Aug 22, 2012 at 6:39 AM, Josh Hopkins wrote:
> I am trying to record calls on demand both inbound and outbound calls. I
> can record outbound calls just fine but not inbound
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
-users-boun...@lists.digium.com] On Behalf Of Torintino T
Sent: Friday, February 03, 2012 10:39 AM
To: Asterisk-users
Subject: [asterisk-users] Recording the follow-me calls
I want please to record the forwarded calls in the not working hours,
so how can i record the follow-me calls to external
I want please to record the forwarded calls in the not working hours,so how can
i record the follow-me calls to external numbers like mobile numbers?.
Thank you --
_
-- Bandwidth and Coloca
I want please to record the forwarded calls in the not working hours,so how can
i record the follow-me calls to external numbers like mobile numbers?.
Thank you --
_
-- Bandwidth and Coloca
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, October 27, 2011 1:11 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording a meetme conference
apes me at the
moment.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, October 27, 2011 1:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users
ling List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording a meetme conference
Danny,
You'd think so (my half-asleep brain saw this after I posted) but it didn't
work. I tried using MixMonitor too, the conference call does not get
recorded. But otherwise,
Danny,
You'd think so (my half-asleep brain saw this after I posted) but it didn't
work. I tried using MixMonitor too, the conference call does not get
recorded. But otherwise, it works fine.
I also made sure to us StopMixMonitor BEFORE the conference or the call to
MixMonitor, in case it
k Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Recording a meetme conference
Hi,
I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:
r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECO
On Thu, Oct 27, 2011 at 11:53 AM, Mike wrote:
> I am trying to record a MeetMe conference, and this is what is relevant in
> the 1.8 manual:
>
>
>
> r - Record conference (records as MEETME_RECORDINGFILE using format
> MEETME_RECORDINGFORMAT. Default filename is
> meetme-conf-rec-${CONFNO}-${UNIQU
Hi,
I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:
r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.
Which is fi
Hi, can anyone help with this?
Thanks
Lee
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 05 July 2011 16:27
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Recording SIP history
Hi all, can
Hi all, can someone explain what siphistory is supposed to do as it
appears to record nothing to my log files. When I sip show history
it's fine but it's not logging anything. My logger.conf has
debug => debug and the debug file grows. Is my understanding correct in
that at the end of the call
Thanks.
Thats perfect!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users m
Non-Commercial Discussion'
Subject: Re: [asterisk-users] Recording
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Monday, October 18, 2010 2:37 PM
To: Asterisk Users Mailing L
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Monday, October 18, 2010 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Recording
Hi,
Does anyone have a
Hi,
Does anyone have a professional recording of someone saying "Recording" so I
can let the operator know that the one-touch recording has started successfully?
Thanks
Dan
[cid:image001.gif@01CB6F04.2EEFF060]
Dan Journo
IT Business Consultant
Kesher Communi
Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Recording maximum time and stop on silence
>
>
>
> All,
>
> Two questions:
>
> 1. Is there a limit on how long a call can be recorded for? For example is
> 4 hours a problem?
>
> 2. Can r
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Wednesday, September 22, 2010 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Recording maximum time and stop on silence
All,
Two questions:
1. Is there a limit on how long a call can be recorded for? For example is 4
hours a problem?
2. Can recording be stopped after a configured period of silence?
Thanks in advance,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0)
Hi,
Never tried it, but you can take a look to the AUDIOHOOK_INHERIT
function that allows MixMonitor to continue the recording in the same
file after the transfer.
http://www.voip-info.org/wiki/view/Asterisk+func+AUDIOHOOK_INHERIT
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium
Hello list,
A calls B, B transfers to C, A speaks with C.
Anyone knows how I can record a conversation where the call is transfered ?!
The recording of the call (which begins when B answers) stops when A and
C are connected together.
Can I keep the recording going ?!
Kind regards,
Jonas.
Hello.
I notice that when a call that is recorded with MixMonitor is transfered
to another co-worker, the recording ends.
exten => 409,n,Macro(SDstartrecording,external,${DID})
the incoming call then goes to a queue...
[macro-startrecording]
; ARG1 = incoming DID or CALLERID(name)
; ARG2 = o
f Of Leif Madsen
> [leif.mad...@asteriskdocs.org]
> Sent: Tuesday, July 27, 2010 9:49 PM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
>
> On 10-07-27 08:39 PM, Michelle Dupuis wrote:
>> Is there a prebuild module/dialplan whic
...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
[paul.belan...@polybeacon.com]
Sent: Tuesday, July 27, 2010 10:10 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
On Tue, Jul 27, 2010 at 10:05 PM, Michelle Dupuis
On Tue, Jul 27, 2010 at 10:05 PM, Michelle Dupuis wrote:
> Hopefully I don't have to read the source code to figure out the features;(
>
*CLI> core show application Record
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polyb
users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan
[sherwood.mcgo...@gmail.com]
Sent: Tuesday, July 27, 2010 8:47 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
There's an app_record, and I believe app_dictate
On 7/27/2010 7:39 PM
-boun...@lists.digium.com] On Behalf Of Leif Madsen
[leif.mad...@asteriskdocs.org]
Sent: Tuesday, July 27, 2010 9:49 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
On 10-07-27 08:39 PM, Michelle Dupuis wrote:
> Is there a prebuild module/dialplan which gives m
On 10-07-27 08:39 PM, Michelle Dupuis wrote:
> Is there a prebuild module/dialplan which gives me a nice interface to
> recording messages? Assuming I can't use the voicemail command, I need to
> offer users a way to record, playback, erase, rerecord, etc.
>
> I can probably do it through dialpl
There's an app_record, and I believe app_dictate
On 7/27/2010 7:39 PM, Michelle Dupuis wrote:
> Is there a prebuild module/dialplan which gives me a nice interface to
> recording messages? Assuming I can't use the voicemail command, I need to
> offer users a way to record, playback, erase, rere
Is there a prebuild module/dialplan which gives me a nice interface to
recording messages? Assuming I can't use the voicemail command, I need to
offer users a way to record, playback, erase, rerecord, etc.
I can probably do it through dialplan but it feels like I'm reinventing the
wheel.
Than
On Wed, 14 Jul 2010, Jonas Kellens wrote:
> On 07/14/2010 03:41 PM, Gordon Henderson wrote:
>> It's the default codec used in DECT phones. I trialled it for a while for
>> some backhaul applications - the users didn't notice anything different
>> and CPU overhead seemed very low, but I've since go
On 07/14/2010 03:41 PM, Gordon Henderson wrote:
> It's the default codec used in DECT phones. I trialled it for a while for
> some backhaul applications - the users didn't notice anything different
> and CPU overhead seemed very low, but I've since gone back to alaw. It
> does save 32Kb/sec per cal
On Wed, 14 Jul 2010, Jonas Kellens wrote:
> On 07/14/2010 01:39 PM, Gordon Henderson wrote:
>> And it's nice to have a choice of vendors to buy G729 from now too.
>> Doesn't help on weedy hardware though.
>
> I thought you could only buy licenses from Digium ? Can you install
> other G729-licenses
On 07/14/2010 01:39 PM, Gordon Henderson wrote:
> And it's nice to have a choice of vendors to buy G729 from now too.
> Doesn't help on weedy hardware though.
>
> Gordon
>
I thought you could only buy licenses from Digium ? Can you install
other G729-licenses on Asterisk ?
I need the MixMoni
On Wed, 14 Jul 2010, Jonas Kellens wrote:
> On 07/14/2010 08:55 AM, Gordon Henderson wrote:
>> On Tue, 13 Jul 2010, Paul Belanger wrote:
>>
>>> On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens
>>> wrote:
>>>
I have no licenses and I want to avoid transcoding all together.
>>> For termina
On 07/14/2010 08:55 AM, Gordon Henderson wrote:
> On Tue, 13 Jul 2010, Paul Belanger wrote:
>
>> On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens
>> wrote:
>>
>>> I have no licenses and I want to avoid transcoding all together.
>>>
>> For terminating a call into Asterisk, you need
On Tue, 13 Jul 2010, Paul Belanger wrote:
> On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens
> wrote:
>> I have no licenses and I want to avoid transcoding all together.
>>
> For terminating a call into Asterisk, you need g729 licenses. It is
> that simple.
The sounds package is avalable in g729
On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens wrote:
> I have no licenses and I want to avoid transcoding all together.
>
For terminating a call into Asterisk, you need g729 licenses. It is
that simple.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC:
I have no licenses and I want to avoid transcoding all together.
When the phone supports G729 and the SIP provider support G729, then the
audio can just pass through...
However, in some cases the audio is recorded. Any change that we can
record in G729 format then ??
And how about voicemail
On Tue, Jul 13, 2010 at 4:29 AM, Jonas Kellens wrote:
> when the conversation is using the G729-codec and the conversation is
> recorded with the Monitor()-application in wav-format, will there be
> transcoding (and thus a need for licenses ?)
>
I believe so, Yes. You can check your license use b
Hello list,
when the conversation is using the G729-codec and the conversation is
recorded with the Monitor()-application in wav-format, will there be
transcoding (and thus a need for licenses ?)
Kind regards,
Jonas.
--
_
Hi!
I am recording with asterisk and so far so good. Now I need to use in the name
of recording wich extension that takes the call and the agent in the queue that
takes the call/
Is there a way to know what extension and the agent that take the call in a
queue for recording???
*---
On Fri, 2010-04-16 at 09:00 -0500, Jason Walker wrote:
> I know that this is a “feature” but I would like to have the hold
> music recorded while a person is on hold. So I know the agent put
> them on hold and not just muted.
>
> I have
>
> monitor-join=yes
>
> monitor-format=wav
>
> in my qu
On Behalf Of Jason Walker
Sent: Friday, April 16, 2010 9:00 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Recording music in Queue
I know that this is a "feature" but I would like to have the hold music
recorded while a person is on hold. So I know the agent put
I know that this is a "feature" but I would like to have the hold music
recorded while a person is on hold. So I know the agent put them on
hold and not just muted.
I have
monitor-join=yes
monitor-format=wav
in my queues.conf
any ideas?
Per
http://www.asteriskguru.com/tutorials/queue
Hello everyone
I have a central Avaya S8300 with G450 Gateway, now all calls go
through the Avaya, but I need to record all calls, my questions are:
1- Can I to interconnect Asterisk with Avaya
?
2- With that tool might Asterisk record calls.
I hope your suggestions.
Thanks
Greeting
On Thu, 2009-12-24 at 18:53 +0100, Gergo Csibra wrote:
> Thursday, December 24, 2009, 5:41:46 PM, Danny wrote:
>
> > Just my opinion; unless you are recording long or many long calls, you
> > should record to your local drive, then copy the files to the USB drive.
> > Asterisk is a very good tool
Thursday, December 24, 2009, 5:41:46 PM, Danny wrote:
> Just my opinion; unless you are recording long or many long calls, you
> should record to your local drive, then copy the files to the USB drive.
> Asterisk is a very good tool - you don't need to mess it up by introducing
> an easy "point of
On Thu, Dec 24, 2009 at 11:24:24AM -0500, Krishna Sumanth Chava wrote:
> Hi Guys,
>
> Merry Christmas and Happy new Year.
>
> I am looking for some assistance from the group as i think this might
> already have been tried before.
>
> i have an asterisk server with a external USB Harddisk Drive,
risk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Krishna
Sumanth Chava
Sent: Thursday, December 24, 2009 10:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Recording the Calls to a USB Drive
Hi Guys,
Merry Chri
Hi Guys,
Merry Christmas and Happy new Year.
I am looking for some assistance from the group as i think this might
already have been tried before.
i have an asterisk server with a external USB Harddisk Drive, just to store
recordings. I am using the mixmonitor application for doing the recording
Daniel Stefanus wrote:
> I want to rebuild my mixmonitor file.But this time I just want the
> recording is from the time when the client answer the call,not from the
> beginning. Anybody can help?
>
> Daniel
>
> ___
> -- Bandwidth and Colocation Provide
I want to rebuild my mixmonitor file.But this time I just want the
recording is from the time when the client answer the call,not from the
beginning. Anybody can help?
Daniel
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
a
hello
AGI is a good option to handle such complexities
On Fri, Oct 23, 2009 at 6:33 PM, Mail list wrote:
> Hello everyone. I have a client with specific requirement, here's the
> scenario:
>
> Call comes in
> Ivr menu, press 1 for new record 2 for existing 3 for operation blabla..
> on pressing
Hello everyone. I have a client with specific requirement, here's the
scenario:
Call comes in
Ivr menu, press 1 for new record 2 for existing 3 for operation blabla..
on pressing 1, list of 5 categories A,B,C,D .. when customer selects a
category a 4 digit pin needs to be generated and a recording
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