Re: [asterisk-users] Call Recording

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, I

Re: [asterisk-users] Call Recording

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, -

[asterisk-users] Call Recording

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 it

[asterisk-users] Call Recording doesn't work

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] [PHP-DEPRECAT

Re: [asterisk-users] call recording via 3rd INVITE/SIP leg

On Thu, Dec 13, 2012 at 9:45 AM, Joshua Colp 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 > to call your

Re: [asterisk-users] call recording via 3rd INVITE/SIP leg

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 eff

[asterisk-users] call recording via 3rd INVITE/SIP leg

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

Re: [asterisk-users] Call Recording

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

[asterisk-users] Call Recording

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

Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8)

On Mon, 2012-07-30 at 08:39 -0500, Matthew Jordan wrote: > > - Original Message - > > From: "Ishfaq Malik" > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > Sent: Wednesday, July 18, 2012 9:58:47 AM > > S

Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8)

- Original Message - > From: "Ishfaq Malik" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Wednesday, July 18, 2012 9:58:47 AM > Subject: Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8) > >

Re: [asterisk-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 B (not on asterisk, real world > number via a SIP trunk) > Mixmonitor

Re: [asterisk-users] Call recording and hosted PBX platform using Asterisk

On 7/2/2012 5:15 PM, Carlos Alvarez wrote: We are a hosted PBX service provider using Asterisk (primarily 1.6, moving to 1.8 soon). In the past, when we've been asked to provide call recording, we deploy a custom server just for that customer. I'd like to bring call recording to our standard ho

[asterisk-users] Call recording and hosted PBX platform using Asterisk

We are a hosted PBX service provider using Asterisk (primarily 1.6, moving to 1.8 soon). In the past, when we've been asked to provide call recording, we deploy a custom server just for that customer. I'd like to bring call recording to our standard hosted system so we can provide it at a lower c

[asterisk-users] Call Recording Stream

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? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] Call recording and transfer issue (asterisk 1.8)

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 a

Re: [asterisk-users] Call recording - methodology

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. Lookin

Re: [asterisk-users] Call recording - methodology

> 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 Systems

Re: [asterisk-users] Call recording - methodology

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 1

Re: [asterisk-users] Call recording - methodology

> 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 > [A

Re: [asterisk-users] Call recording - methodology

> 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 ton

Re: [asterisk-users] Call recording - methodology

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 stil

Re: [asterisk-users] Call recording - methodology

> 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, eit

Re: [asterisk-users] Call Recording using MixMonitor - close, but would like some more words of wisdom.

> 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 Systems

Re: [asterisk-users] Call Recording using MixMonitor - close, but would like some more words of wisdom.

On Fri, Apr 8, 2011 at 3:35 PM, Silver Thorne wrote: > Dan et al; > > This looks like a perfect solution. > > 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. > Steps. > >1. added

[asterisk-users] Call Recording using MixMonitor - close, but would like some more words of wisdom.

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 MixMon

Re: [asterisk-users] Call recording - methodology

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 MixMon

Re: [asterisk-users] Call recording - methodology

_ 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 >

Re: [asterisk-users] Call recording - methodology

> 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

Re: [asterisk-users] Call recording - methodology

On Wed, Apr 6, 2011 at 5:54 AM, Silver Thorne wrote: > 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 kn

Re: [asterisk-users] Call recording - methodology

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 -- ___

[asterisk-users] Call recording - methodology

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

Re: [asterisk-users] Call Recording audio file quality query

On Wed, Feb 9, 2011 at 12:31 PM, Tilghman Lesher wrote: > 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 > > str

Re: [asterisk-users] Call Recording audio file quality query

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 >

Re: [asterisk-users] Call Recording audio file quality query

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 t

Re: [asterisk-users] Call Recording audio file quality query

On Tuesday 08 February 2011 06:34:56 Sherwood McGowan wrote: > On Tue, Feb 8, 2011 at 6:01 AM, 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 av

Re: [asterisk-users] Call Recording audio file quality query

-Commercial Discussion" < > asterisk-users@lists.digium.com> > Subject: Re: [asterisk-users] Call Recording audio file quality query > > On Tue, Feb 8, 2011 at 6:01 AM, wrote: > >> But if you are getting calls all the way on VoIP then you can have calls >> in HD au

Re: [asterisk-users] Call Recording audio file quality query

Yes. The technology need to be used on LAN switches is "port mirroring" or "line tapping" -Original Message- From: "Sherwood McGowan" Sent: Tuesday, February 8, 2011 7:34am To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subj

Re: [asterisk-users] Call Recording audio file quality query

On Tue, Feb 8, 2011 at 6:01 AM, 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 > capturing to trace th

Re: [asterisk-users] Call Recording audio file quality query

> > > 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 introduc

Re: [asterisk-users] Call Recording audio file quality query

;Asterisk Users Mailing List - Non-Commercial Discussion" 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 > wrote: > Hi > > We're getting req

Re: [asterisk-users] Call Recording audio file quality query

> -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 Re

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 > wrote: > Hi > > We're getting requests coming in for higher quality audio in > our call > recordings. We currently use MixMonitor and everything is >

Re: [asterisk-users] Call Recording audio file quality query

On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik 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 Monitor and s

[asterisk-users] Call Recording audio file quality query

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 channel

Re: [asterisk-users] Call recording format

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 wrote: > WAV or wav? One of these has

Re: [asterisk-users] Call recording format

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.

Re: [asterisk-users] Call recording format

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

Re: [asterisk-users] Call recording format

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 our

Re: [asterisk-users] Call recording format

On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius 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 recordings using wav CPU load is > s

Re: [asterisk-users] Call recording format

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 ala

Re: [asterisk-users] Call recording format

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 wrote: > Hi All, > We have a requirement to record over 60

[asterisk-users] Call recording format

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 writing

Re: [asterisk-users] Call Recording Questions

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 t

Re: [asterisk-users] Call Recording Questions

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 a

Re: [asterisk-users] Call Recording Questions

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

Re: [asterisk-users] Call Recording Questions

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: http://www.

Re: [asterisk-users] Call Recording Questions

> 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

Re: [asterisk-users] Call Recording Questions

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

Re: [asterisk-users] Call Recording Questions

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

Re: [asterisk-users] Call Recording Questions

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

Re: [asterisk-users] Call Recording Questions

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

Re: [asterisk-users] Call Recording Questions

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 record

Re: [asterisk-users] Call Recording Questions

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 c

Re: [asterisk-users] Call Recording Questions

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 recor

Re: [asterisk-users] Call Recording Questions

> 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 copyin

[asterisk-users] Call Recording Questions

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 cop

Re: [asterisk-users] Call Recording and Posting

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 lin

Re: [asterisk-users] Call Recording and Posting

ow the logic behind the code. Just some more 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

Re: [asterisk-users] Call Recording and Posting

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

Re: [asterisk-users] Call Recording and Posting

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' comman

Re: [asterisk-users] Call Recording and Posting

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 i

Re: [asterisk-users] Call Recording and Posting

-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

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 that redirects the output to a log > file. What happens if the script is run whil

Re: [asterisk-users] Call Recording and Posting

: 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 help

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 > with Linux The script is very simple a

Re: [asterisk-users] Call Recording and Posting

--Original 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 wr

Re: [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 dir

Re: [asterisk-users] Call Recording and Posting

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

Re: [asterisk-users] Call Recording and Posting

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

Re: [asterisk-users] Call Recording and Posting

a machine 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-u

[asterisk-users] Call Recording and Posting

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 ne

Re: [asterisk-users] Call recording in - out

David Backeberg escribió: On Sun, Jun 7, 2009 at 12:51 PM, Joao Gomes Pereira 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 e

Re: [asterisk-users] Call recording in - out

On Sun, Jun 7, 2009 at 12:51 PM, Joao Gomes Pereira 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, but this > isn't h

Re: [asterisk-users] Call recording in - out

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 > Hello > I did as you told me, but the problem remains. > Im using Asterisk 1.2.x > > and this is my config: > > queues.conf

Re: [asterisk-users] Call recording in - out

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 tim

Re: [asterisk-users] Call recording in - out

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, 2009-06-

[asterisk-users] Call recording in - out

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 to

Re: [asterisk-users] Call recording - posible to remove recorded fileat the end of the call

.com [mailto:asterisk-users-boun...@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 recordin

[asterisk-users] Call recording - posible to remove recorded file at the end of the call

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 re

Re: [asterisk-users] Call Recording Alias

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 - http://das-aste

Re: [asterisk-users] Call Recording Alias

Where did I make mistake ? On Thu, Jan 29, 2009 at 1:07 AM, David @ULC 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 application/x-htt

Re: [asterisk-users] Call Recording Alias

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 > > > Thanks for your advice , but I asked for expert guidance as I read the

Re: [asterisk-users] Call Recording Alias

David @ULC Sent: Wednesday, 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

Re: [asterisk-users] Call Recording Alias

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"

Re: [asterisk-users] Call Recording Alias

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"

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 @ULC wrote: > > Modified httf.conf file and added : > -- > > Alias /record

Re: [asterisk-users] Call Recording Alias

dud please go to asterisk.org support and download the asterisk book and then READ IT. David 2009/1/28 David @ULC > > Modified httf.conf file and added : > -- > > Alias /recordings/ "/var/spool/asterisk/monitorDONE/" > > > Options Indexes

[asterisk-users] Call Recording Alias

Modified httf.conf file and added : -- Alias /recordings/ "/var/spool/asterisk/monitorDONE/" Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all Created a folder under vicidial as recordings. FULL_RECORDING is also

Re: [asterisk-users] Call Recording - Asterisk

> > > > > > 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. > > > >

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