Re: [asterisk-users] MixMonitor not recording through transfer

On Tue, Nov 29, 2022 at 9:24 PM Carlos Chavez wrote: > I have the following scenario: > > Agent calls external number > > Mixmonitor starts recording call > > After agent speaks with customer they need to transfer them to an > extension that will simply play a message > > Customer hangs up

[asterisk-users] MixMonitor not recording through transfer

    I have the following scenario: Agent calls external number Mixmonitor starts recording call After agent speaks with customer they need to transfer them to an extension that will simply play a message Customer hangs up     The problem is that the recording stops the moment the agent

[asterisk-users] MixMonitor multiple times to the same file

Hello list, Hope you are all doing fine! While playing around with the MixMonitor, I've found out that it is possible to start the MixMonitor multiple times to the same output file. It is very easy to reproduce, via the CLI or via the dialplan. If the MixMonitor is called twice from the same

Re: [asterisk-users] MixMonitor and ChanSpy whisper

Thanks for the reply Joshua! I did look the code, but it's too complicated for my old C knowledge :( So I guess I am left with the Monitor app Also a ConfBridge would also work instead of ChanSpy with whisper Cheers, Patrick Wakano On 6 July 2018 at 08:51, Joshua Colp wrote: > On Thu,

Re: [asterisk-users] MixMonitor and ChanSpy whisper

On Thu, Jul 5, 2018, at 7:37 PM, Patrick Wakano wrote: > Hello Asterisk list, > Hope you are all doing well! > > We are using the MixMonitor application to record the calls and under some > situations the call can be spied using ChanSpy with whisper enabled. > Sometimes the spying channel is a

Re: [asterisk-users] MixMonitor and ChanSpy whisper

i am having same problem on Asterisk 13. MixMonitor does not record whisper or barge audio from ChanSpy. only call audio us recorded. hope someone can help. On Thu, Jul 5, 2018, 11:38 PM Patrick Wakano wrote: > Hello Asterisk list, > Hope you are all doing well! > > We are using the MixMonitor

[asterisk-users] MixMonitor and ChanSpy whisper

Hello Asterisk list, Hope you are all doing well! We are using the MixMonitor application to record the calls and under some situations the call can be spied using ChanSpy with whisper enabled. Sometimes the spying channel is a person who can interact in the call, and some other times it is a

[asterisk-users] MixMonitor recording when in the holding bridge

Hello list, Hope you all doing fine! We are using the MixMonitor to record the call with the option 'b' so the recording is done only when this call is bridged. Under some situations the call is put on a wait state and the recording should not happen anymore until the channel is bridged back.

Re: [asterisk-users] Mixmonitor with b option

>>>We have a server that records all calls so we set Mixmonitor with the b >>> option to only record calls that are actually bridged. I notice that we >>> have lost of 44 byte files in /var/spool/asterisk/monitor which correspond >>> to calls that were not answered. If a call is not

Re: [asterisk-users] Mixmonitor with b option

On 8 Jan 2018, at 18:44, Dovid Bender wrote: > On Jan 8, 2018 11:35 AM, "Carlos Chavez" wrote: >> We are running 13.8.4 at the moment which was the latest when deployed. >> I guess the patch never made it to the trunk. The problem with running a

Re: [asterisk-users] Mixmonitor with b option

Include in your script to only look at files that are x minutes old. For instance if a file has not been touched in over an hour chances are the call is over. On Jan 8, 2018 11:35 AM, "Carlos Chavez" wrote: > On 1/8/18 9:38 AM, Bertrand LUPART - Linkeo.com wrote: > > Hello

Re: [asterisk-users] Mixmonitor with b option

On 1/8/18 9:38 AM, Bertrand LUPART - Linkeo.com wrote: Hello Carlos, We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which

Re: [asterisk-users] Mixmonitor with b option

Hello Carlos, >We have a server that records all calls so we set Mixmonitor with the b > option to only record calls that are actually bridged. I notice that we have > lost of 44 byte files in /var/spool/asterisk/monitor which correspond to > calls that were not answered. If a call is

[asterisk-users] Mixmonitor with b option

We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls that were not answered. If a call is not answered I assume it

Re: [asterisk-users] MixMonitor Files Always Empty

-users@lists.digium.com Subject: [asterisk-users] MixMonitor Files Always Empty Hi all, I’m having an issue with MixMonitor in Asterisk 11.17.0. The gsm files are always created but no audio is being written to them regardless of whether or not I use the b option. I’ve tried switching to the Monitor

Re: [asterisk-users] MixMonitor Files Always Empty

Please don't top post. On Wed, 22 Apr 2015, Mark Farmer wrote: The file is always created but is always zero size. This is the dial plan that records the call: exten = _0[1-8]X.,1,Set(CALLFILENAME=/var/spool/asterisk/callrecordings/its/${STRFTIME(${EPOCH},,%Y/%m/%d)}/Outbound-${UNIQUEID})

Re: [asterisk-users] MixMonitor with b option recording all calls

On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: Hello I have an issue wit MixMonitor. I need to record only answered calls, so I set b option for this but calls still recording even call no answered My asterisk version 12.5.1, at my other servers with older versions of asterisk (11.8 for

Re: [asterisk-users] MixMonitor with b option recording all calls

SIP trunks 2014-09-22 12:12 GMT+04:00 A J Stiles asterisk_l...@earthshod.co.uk: On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: Hello I have an issue wit MixMonitor. I need to record only answered calls, so I set b option for this but calls still recording even call no answered My

Re: [asterisk-users] MixMonitor with b option recording all calls

THIS IS NOT WHERE YOUR REPLY BELONGS On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: 2014-09-22 12:12 GMT+04:00 A J Stiles asterisk_l...@earthshod.co.uk: On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: Hello I have an issue wit MixMonitor. I need to record only answered calls,

Re: [asterisk-users] MixMonitor with b option recording all calls

my sip.conf [kamailio_ext1] type=friend host = my.superprovider.com port = 5068 canreinvite = no insecure = invite,port transport=udp trustrpid=yes context = incoming videosupport=no directmedia=no dtlsenable = no tlsenable=no disallow=all allow=alaw allow=opus allow=ulaw connection goes great.

[asterisk-users] MixMonitor with b option recording all calls

Hello I have an issue wit MixMonitor. I need to record only answered calls, so I set b option for this but calls still recording even call no answered My asterisk version 12.5.1, at my other servers with older versions of asterisk (11.8 for example) MixMonitor works fine. --

[asterisk-users] mixmonitor - convert wav to mp3/aac

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 thank you -- --- Marek Cervenka

Re: [asterisk-users] mixmonitor - convert wav to mp3/aac

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:

Re: [asterisk-users] mixmonitor extension

for the record. info about opus from Lorenzo Mniero (author of Opus patch for asterisk) with his permission --cite-- Opus is just a codec. In order to save an audio file using Opus, you need a container, which for Opus is OGG. Asterisk supports OGG, but I think it is implemented to only dump

Re: [asterisk-users] mixmonitor extension

i'm talking about native mp3,opus support in mixmonitor application. read the first answer from Gareth Blades Dne 24.1.2014 1:39, Patrick Lists napsal(a): On 24-01-14 00:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? Iirc mp3 was in

Re: [asterisk-users] mixmonitor extension

On 23/01/14 23:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? what about patch for Opus? uncle google doesnt know MP3 is only supported for reading not writing. Its a patent issue as Asterisk cannot distribute the software to write to mp3 under

[asterisk-users] mixmonitor extension

hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --- Marek Cervenka === --

Re: [asterisk-users] mixmonitor extension

On 23/01/14 15:21, Marek Cervenka wrote: hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? core show file formats will give you a list of formats your system supports

Re: [asterisk-users] mixmonitor extension

can someone confirm that mp3 is unsupported? is patch available? what about patch for Opus? uncle google doesnt know Dne 23.1.2014 16:31, Gareth Blades napsal(a): On 23/01/14 15:21, Marek Cervenka wrote: hi, which file extensios are supported in mixmonitor application?

Re: [asterisk-users] mixmonitor extension

On 24-01-14 00:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? Iirc mp3 was in asterisk-addons in asterisk 1.4 and iirc in later versions of asterisk you can enable format_mp3 in make menuselect. what about patch for Opus? uncle google doesnt

[asterisk-users] MixMonitor inserting extra 20ms packets of silence (1.4.43)

I'm trying to diagnose potential causes of an issue with MixMonitor in 1.4.43 (I doubt it's very version-specific). I won't have hands on the kit until the end of the week, but I have listened to some recordings. It doesn't happen on every call - only sometimes. Basically, it is a call from one

Re: [asterisk-users] MixMonitor inserting extra 20ms packets of silence (1.4.43)

2012-09-10 18:13, Tony Mountifield skrev: I'm trying to diagnose potential causes of an issue with MixMonitor in 1.4.43 (I doubt it's very version-specific). I won't have hands on the kit until the end of the week, but I have listened to some recordings. It doesn't happen on every call - only

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

there… ** ** Regards, Ikka (Jakarta, Indonesia) ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Thorben Jensen *Sent:* Saturday, July 28, 2012 1:58 PM *To:* asterisk-users *Subject:* [asterisk-users

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

Thorben Jensen wrote: From: Thorben Jensen i...@thorben.dk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 3, 2012 4:15:58 AM Subject: Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b Hi

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

Hi Jonathan, If I set the MixMonitor option on a queue, it will not create an zero length file if the call is not bridged, and I just assumed it would be the case with option b. I have set the fileformat to raw, but if I set it to wav it will create a 64 byte file. Off course I can

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

I have read an earlier posts in this forum that said it was a bug in asterisk 10.4 and there's a patch to fix it. So it already reported as bug.  But i use asterisk 10.61 and that bug is still there, unfixed.  We upgrade to the newest version for the new feature that we can use,  and also

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

Ikka.vertika wrote: From: Ikka.vertika ikka.vert...@mitrakreasindo.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 3, 2012 9:20:03 AM Subject: Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

Jonathan Rose wrote: Thorben Jensen i...@thorben.dk wrote: Hi Jonathan, If I set the MixMonitor option on a queue, it will not create an zero length file if the call is not bridged, and I just assumed it would be the case with option b. I have set the fileformat to raw, but

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

I was looking over Queue and I don't think there is actually an option for Queue that will automatically start a MixMonitor. I see a few options involving mixmonitor (x and X), but they appear to be more about allowing the parties involved with the call to start MixMonitor through dialplan

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

MixMonitor creates the file before it starts recording. The b option simply waits until the bridge event to take audio frames and add them to the stream. I don't really see why this is a problem. It isn't like you are going to run out of space for zero length files, and more to the point if

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

Thorben Jensen wrote: I was looking over Queue and I don't think there is actually an option for Queue that will automatically start a MixMonitor. I see a few options involving mixmonitor (x and X), but they appear to be more about allowing the parties involved with the call to start

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

: [asterisk-users] MixMonitor creating file on non-bridged calls with option b I am using MixMonitor to record calls and I have set the b option as I don't want to get files for non-bridged calls. Mixmonitor always creates a file with 0 bytes even when the call is not bridged. Is it possible

[asterisk-users] MixMonitor creating file on non-bridged calls with option b

I am using MixMonitor to record calls and I have set the b option as I don't want to get files for non-bridged calls. Mixmonitor always creates a file with 0 bytes even when the call is not bridged. Is it possible to avoid this somehow? This is what I do:

Re: [asterisk-users] MixMonitor and ChanSpy

On 02/02/2012 11:24 AM, Jonas Kellens wrote: Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Anyone with some feedback ?! I notice that ongoing recordings are

Re: [asterisk-users] MixMonitor and ChanSpy

Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't chanspy a call because its recorded ! Which version of

Re: [asterisk-users] MixMonitor and ChanSpy

On 02/07/2012 01:07 PM, Sammy Govind wrote: Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't

Re: [asterisk-users] MixMonitor and ChanSpy

Oh Come on you are Using Asterisk 1.6.2.22. already. Atleast give it a shot and if this still persists then look for other methods or fixes. On Tue, Feb 7, 2012 at 5:44 PM, Jonas Kellens jonas.kell...@telenet.bewrote: ** On 02/07/2012 01:07 PM, Sammy Govind wrote: Hello, I've been

Re: [asterisk-users] MixMonitor and ChanSpy

that means that from 1.4.18 that issue is no longer present ? On 7 February 2012 12:44, Jonas Kellens jonas.kell...@telenet.be wrote: ** On 02/07/2012 01:07 PM, Sammy Govind wrote: Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or

Re: [asterisk-users] MixMonitor and ChanSpy

It's a good thing I never read that warning, since I've been using those in a call center environment for about seven years and never had that issue. Started with 1.2, went to 1.4 and 1.6 now. So I can't answer your question about when it was fixed but I've never had a problem doing it (70

Re: [asterisk-users] MixMonitor and ChanSpy

Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MixMonitor and ChanSpy It's a good thing I never read that warning, since I've been using those in a call center environment for about seven years and never had that issue. Started with 1.2, went to 1.4 and 1.6 now. So I can't

Re: [asterisk-users] MixMonitor and ChanSpy

On Tue, Feb 7, 2012 at 9:47 AM, Danny Nicholas da...@debsinc.com wrote: Only trust the wiki if it explicitly refers to your current version (and then you should still test it). THIS. Believe him, when it comes to Asterisk, don't trust the docs, try it. Or read the code. There is no other

[asterisk-users] MixMonitor and ChanSpy

Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Kind regards, Jonas. -- _ -- Bandwidth and

[asterisk-users] Mixmonitor command parameter problem on Asterisk 1.8.4

Dear all. I'm using MixMonitor command in my dialplan, and I used the command parameter to execute some thing after recording the file. I used the command parameter to convert the wav file that created earlier to MP3 and than deleted the WAV file. It worked fine with asterisk 1.4.21.2.

Re: [asterisk-users] MixMonitor and attended transfers [SOLVED]

On Tue, 2011-08-02 at 10:58 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called)

[asterisk-users] MixMonitor and attended transfers

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A

Re: [asterisk-users] MixMonitor and attended transfers

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B

Re: [asterisk-users] MixMonitor and attended transfers

On Tue, 2011-08-02 at 07:51 -0400, Dan Journo wrote: Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being

Re: [asterisk-users] MixMonitor and attended transfers

On Tue, 2011-08-02 at 10:58 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called)

[asterisk-users] Mixmonitor concept's question

[RecordPrompts] exten = ,1,Answer() exten = ,n,NoOp(WelCome to conference section) exten = ,n,Playback(ConfDemoWC) exten = ,n,MixMonitor(tmp/00Record/-${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}.wav,ab) exten = ,n,Konference(${EXTEN},ADRSV) Hi My basic doubt is that if 1 or

Re: [asterisk-users] Mixmonitor concept's question

On Monday, July 04, 2011 05:10:43 AM virendra bhati wrote: [RecordPrompts] exten = ,1,Answer() exten = ,n,NoOp(WelCome to conference section) exten = ,n,Playback(ConfDemoWC) exten = ,n,MixMonitor(tmp/00Record/-${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}.w av,ab) exten =

Re: [asterisk-users] Mixmonitor concept's question

Hi Your suggestion is right if we want different recording for all channels. But my problem is that I want to know if more user call the same conference at different time gape(difference) then mixmonitor will take single asterisk thread for recording or multiple thread for recoding. On Mon,

[asterisk-users] MixMonitor - garbled/corrupted WAV files

Hi, I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which are mostly fine. What can be the cause? The

Re: [asterisk-users] MixMonitor - garbled/corrupted WAV files

On Tue, Jun 28, 2011 at 10:30 AM, Mike l...@net-wall.com wrote: Hi, ** ** I’ve had problems with MixMonitor recordings. A lot (I’d say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as

Re: [asterisk-users] MixMonitor - garbled/corrupted WAV files

On Tue, Jun 28, 2011 at 11:30:54AM -0400, Mike wrote: I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which are

Re: [asterisk-users] MixMonitor - garbled/corrupted WAV files

On Tue, Jun 28, 2011 at 11:30:54AM -0400, Mike wrote: I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which

Re: [asterisk-users] MixMonitor - garbled/corrupted WAV files

I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which are mostly fine. What can be the cause? The conversation

Re: [asterisk-users] MixMonitor - garbled/corrupted WAV files

On Tue, Jun 28, 2011 at 01:24:43PM -0400, Mike wrote: On Tue, Jun 28, 2011 at 11:30:54AM -0400, Mike wrote: I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the

Re: [asterisk-users] MixMonitor - garbled/corrupted WAV files

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, June 28, 2011 1:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MixMonitor

[asterisk-users] MixMonitor

hello list, i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in order to record the conversation i can record all the calls inbound and outbound without problem. but when i receive an inbound call from customer in IAX(1000) and i want to transfer the call to other phone

Re: [asterisk-users] MixMonitor

On 16/06/11 07:36 AM, salaheddine elharit wrote: hello list, i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in order to record the conversation but when i receive an inbound call from customer in IAX(1000) and i want to transfer the call to other phone SIP(223) the

Re: [asterisk-users] MixMonitor

thanks for your response the call is going to IAX(1000), i have i DID Number when the customer call this number 0520XX the call is goint to agent IAX. in my dialplan i have exten = 223,1,MixMonitor(blah.wav) exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten = 223,n,Dial(SIP/223) and

Re: [asterisk-users] MixMonitor

On 16/06/11 09:20 AM, salaheddine elharit wrote: thanks for your response the call is going to IAX(1000), i have i DID Number when the customer call this number 0520XX the call is goint to agent IAX. in my dialplan i have exten = 223,1,MixMonitor(blah.wav) exten =

Re: [asterisk-users] MixMonitor

i have asterisk 1.4 and also i have aheeva applicaton also installed in my server in the consol this call may be monitored or recorded best regrads 2011/6/16 Leif Madsen leif.mad...@asteriskdocs.org On 16/06/11 09:20 AM, salaheddine elharit wrote: thanks for your response the call is

Re: [asterisk-users] MixMonitor

-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Thursday, June 16, 2011 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MixMonitor i have asterisk 1.4 and also i have aheeva applicaton also installed in my server

Re: [asterisk-users] MixMonitor

Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] MixMonitor i have asterisk 1.4 and also i have aheeva applicaton also installed in my server in the consol this call may be monitored or recorded best regrads 2011/6/16 Leif Madsen leif.mad...@asteriskdocs.org On 16

Re: [asterisk-users] MixMonitor

: Thursday, June 16, 2011 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MixMonitor hi Danny thank you for your response i switched the MixMonitor and i still have the same result any help please 2011/6/16 Danny Nicholas da

[asterisk-users] MixMonitor not recording in version 1.8

Greetings. Just updated from 1.4.22 to 1.8. Minor changes in dialplan and things work ok. Except for one thing. I have a call to MixMonitor. This is implementing a dictaphone kind of app. With forwarding recordings to email and storing them on the server. The process works so that we dial into

Re: [asterisk-users] MixMonitor

So it seems not related to MixMonitor. Are you 100% sure that your PHP-AGi script is not looping somewhere? You should try to understand which is the process that is taken you CPU. On Tue, Nov 9, 2010 at 2:32 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hi, After disabling

Re: [asterisk-users] MixMonitor

Hi, After disabling MixMonitor, I realize that my CPU saturates as always! What my script PHP-AGI is fairly simple! - I answer a call - Some menus - I send the call to another line $this-exec_dial (SIP/provider/NUMBER, ...) And I was 75-80% using an e4...@2.40ghz! It is not logic ! Please help

Re: [asterisk-users] MixMonitor

You think of a loop? This is possible because I use AGISIGHUP=no .. exten = s,1,set(AGISIGHUP=no); exten = s,2,AGI(myapp.agi) ; I will put lines and debug log file ... I do not think that Asterisk archive errors AGI script? 2010/11/9 Marino Punturieri map...@gmail.com So it seems not

Re: [asterisk-users] MixMonitor

Not sure, but you can try to increase debug log level and check whether you'll have more details On Tue, Nov 9, 2010 at 4:55 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: You think of a loop? This is possible because I use AGISIGHUP=no .. exten = s,1,set(AGISIGHUP=no); exten =

Re: [asterisk-users] MixMonitor

none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. -- _ -- Bandwidth and Colocation

Re: [asterisk-users] MixMonitor

On 5 Nov 2010, at 01:22, Mickael MONSIEUR wrote: Have you noticed a marked increase in CPU load when using MixMonitor? Since when? 1.6.2.9-1? 1.6.2.8? 1.0? S -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] MixMonitor

Hi, marked - noticed. I do not know where it comes from, my CPU goes from 2% to 60-70% at a command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU e4...@2.40ghz 2010/11/5 Norbert Zawodsky norb...@zawodsky.at Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: none ?

[asterisk-users] MixMonitor

Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

[asterisk-users] MixMonitor and StopMixMonitor

Hello list, how does StopMixMonitor know which 'monitoring channel' to stop when there are multiple conversations that are being monitored/recorded ?? I want to use StopMixMonitor in a macro, called from within applicationmap (features.conf). Jonas. --

[asterisk-users] MixMonitor stops audio in SIP to SIP call

Has anyone experienced this problem before? I am running Asterisk 1.4.21.2 If I run: MixMonitor(..) Dial(SIP/...) Both parties cannot hear each other. As soon as I comment out MixMonitor, the audio can be heard. I saw this issue on https://issues.asterisk.org/view.php?id=16256 It seems to

[asterisk-users] MixMonitor and Call Latency during conversation

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As

Re: [asterisk-users] MixMonitor and Call Latency during conversation

On Mon, Nov 16, 2009 at 9:40 AM, Bharath B. Reddy Bynagari bynag...@mavensphere.com wrote: We are using MixMonitor to record the call. When the call is bridged, the latency is significant. $ConversationFile = $ConversationPath.conv_.$CallQID-$ConversationID.wav; $self-agi-answer();

[asterisk-users] MixMonitor and Transcoding..

Can't find an answer to this, but maybe I've not looked hard enough ... Does MixMonitor work without transcoding? ie. if I have a g729 stream passing through and I'm recording it with e.g. MixMonitor(/dump/filename.g729,b) and specify g729 in the filename, does MixMonitor transcode both legs

Re: [asterisk-users] MixMonitor and Transcoding..

On Tue, 11 Aug 2009, Gordon Henderson wrote: Can't find an answer to this, but maybe I've not looked hard enough ... Does MixMonitor work without transcoding? ie. if I have a g729 stream passing through and I'm recording it with e.g. MixMonitor(/dump/filename.g729,b) and specify g729 in

Re: [asterisk-users] MixMonitor and Transcoding..

Gordon Henderson wrote: Transcoding is something that's not an option here. Hm. Maybe old fashioned 'monitor' and offline mixing although I'm open to suggestions here.. In general, it is not possible to mix compressed audio; it must be uncompressed first. -- Kevin P. Fleming Digium, Inc.

Re: [asterisk-users] MixMonitor/Queue and Tranfers

Thanks for the reply. 1. The extensions in the Queues are setup as Agent members, defined in Agents.conf, then within the definition of the queue in queues.conf they are made members of the queue. 2. As for the recording my diaplan is as follows: [main-line] exten = s,1,NoOp() exten =

Re: [asterisk-users] MixMonitor/Queue and Tranfers

Un-topposting... On Tue, Jul 7, 2009 at 7:08 PM, Miguel Molina mmol...@millenium.com.co wrote: Darrin Henshaw escribió: 2. The issue does seem to be limited to MixMonitor and the Queue application, as in testing I setup mixmonitor on my extension dialed it from outside the

[asterisk-users] MixMonitor/Queue and Tranfers

Hello, First off to lay the ground work, I’m running Asterisk 1.4.25, which was recently upgraded from 1.2 about one month ago. We are running it on CentOS 4.7, on Dell PoweEdge 1950’s. We are a small MSP(Managed Service Provider) providing Network/Server/Desktop support for companies based

Re: [asterisk-users] MixMonitor/Queue and Tranfers

Darrin Henshaw escribió: 2. The issue does seem to be limited to MixMonitor and the Queue application, as in testing I setup mixmonitor on my extension dialed it from outside the company(my cell phone) and transferred the call without stopping the recording. I have a couple of

Re: [asterisk-users] MixMonitor and ChanSpy strangeness...

Right after a bit of investigation i've found that it's because we're running a mysql database on the same server, it was fine all morning with a relatively low load on the server, now the rest of the agents have logged in the problem has returned! Time to buy a new database server... mystery

Re: [asterisk-users] MixMonitor and ChanSpy strangeness...

Doesn't look like anyone has any suggestions though, guess it's time to play until it's fixed then :) 2008/12/2 Thomas Kenyon [EMAIL PROTECTED] Geraint Lee wrote: Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be

[asterisk-users] MixMonitor and ChanSpy strangeness...

Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party) is conserned the

Re: [asterisk-users] MixMonitor and ChanSpy strangeness...

Geraint Lee wrote: Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party)

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