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, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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, 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. > -- Ian Harding IT Director Brown Paper Tickets 1-800-838-3006 ext 7186 http://www.brownpapertickets.com -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 effectively a 3rd leg to the in progress 2-leg call. This 3rd leg is a SIP dial to a URI and/or PSTN number. I'm thinking I have to do this with a conference bridge config and add a 3rd muted leg to the conference? 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 3rd leg and then have it execute ChanSpy with the correct criteria to get to the right leg. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call recording via 3rd INVITE/SIP leg
On Thu, Dec 13, 2012 at 9:45 AM, Joshua Colp jc...@digium.com wrote: If you don't want to incur the overhead of a full blown conference bridge you can use ChanSpy to spy on a channel. It will provide a mixed stream of the incoming and outgoing part of the channel. So essentially use Originate to call your 3rd leg and then have it execute ChanSpy with the correct criteria to get to the right leg. Thanks Joshua, I'll check that out! -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 exiting.
exten = s,n,ExecIf($[${CUT(CALLFILENAME,-,1)}=exten
${DB(AMPUSER/${THISEXTEN}/recording/ondemand)}!=enabled]?MacroExit())
This condition is causing MacroExit to be called, so fix these conditions.
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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 i...@pack-net.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 18, 2012 9:58:47 AM Subject: Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8) On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote: Hi I'm having a problem with the entirety of a call being recorded in the following scenario I'm using asterisk 1.8.7.0 Person A (asterisk peer) calls Person B (not on asterisk, real world number via a SIP trunk) Mixmonitor is invoked by Person A in the outbound context and AUDIOHOOK_INHERIT(MixMonitor)=yes is also set Person a transfers Person B to Person C (another asterisk peer) Person A is no longer involved in the call and the call is bridged between Person B and Person C The call recording stops as soon as Person A hangs up, even though AUDIOHOOK_INHERIT is set Is there any way we can get the entire call recorded in one file? Thanks in advance Ish Ish: Leif had a pretty good explanation of why AUDIOHOOK_INHERIT behaves this way in the comments of ASTERISK-16013. I'll quote it here: Well I just tested this scenario. ... after a bit of testing I determined the scenarios. Working: * Party A places a call to Party B * Party B places an attended transfer to Party C * Party A and C are not talking * Call recording works as expected Not working: * Party A places a call to Party B * Party A places an attended transfer to Party C Call recording works up to this point – the recording of the conversation between Party A and Party B, and the portion of the conversation between Party A and Party C is recorded * Party A now hangs up * Call recording is now stopped * Party B and Party C are now speaking (unrecorded) To me, this is actually the intended and expected behavior. The AUDIOHOOK_INHERIT() function is executed on the channel created by Party A, and thus the call recording is going to follow Party A around when it is transferred around the system. However, once Party A is kicked out of the conversation (i.e. they hangup) then the call recording stops because that is the channel the recording is associated with. Note that if you read the scenario description of AUDIOHOOK_INHERIT at https://wiki.asterisk.org/wiki/display/AST/Function_AUDIOHOOK_INHERIT, the function works in the transfer scenarios where the called party initiates the transfer, not the callee. For your scenario, you could try setting the MixMonitor on the called party channel as opposed to the callee channel, using one of the Dial GoSub/Macro options (U,M,b). https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial Note that Macro is deprecated in more recent versions of Asterisk, and the 'b' option will only be available in Asterisk 11. Thank you for the hints at the end, using the M option has sorted my issue out Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8)
- Original Message - From: Ishfaq Malik i...@pack-net.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 18, 2012 9:58:47 AM Subject: Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8) On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote: Hi I'm having a problem with the entirety of a call being recorded in the following scenario I'm using asterisk 1.8.7.0 Person A (asterisk peer) calls Person B (not on asterisk, real world number via a SIP trunk) Mixmonitor is invoked by Person A in the outbound context and AUDIOHOOK_INHERIT(MixMonitor)=yes is also set Person a transfers Person B to Person C (another asterisk peer) Person A is no longer involved in the call and the call is bridged between Person B and Person C The call recording stops as soon as Person A hangs up, even though AUDIOHOOK_INHERIT is set Is there any way we can get the entire call recorded in one file? Thanks in advance Ish Ish: Leif had a pretty good explanation of why AUDIOHOOK_INHERIT behaves this way in the comments of ASTERISK-16013. I'll quote it here: Well I just tested this scenario. ... after a bit of testing I determined the scenarios. Working: * Party A places a call to Party B * Party B places an attended transfer to Party C * Party A and C are not talking * Call recording works as expected Not working: * Party A places a call to Party B * Party A places an attended transfer to Party C Call recording works up to this point – the recording of the conversation between Party A and Party B, and the portion of the conversation between Party A and Party C is recorded * Party A now hangs up * Call recording is now stopped * Party B and Party C are now speaking (unrecorded) To me, this is actually the intended and expected behavior. The AUDIOHOOK_INHERIT() function is executed on the channel created by Party A, and thus the call recording is going to follow Party A around when it is transferred around the system. However, once Party A is kicked out of the conversation (i.e. they hangup) then the call recording stops because that is the channel the recording is associated with. Note that if you read the scenario description of AUDIOHOOK_INHERIT at https://wiki.asterisk.org/wiki/display/AST/Function_AUDIOHOOK_INHERIT, the function works in the transfer scenarios where the called party initiates the transfer, not the callee. For your scenario, you could try setting the MixMonitor on the called party channel as opposed to the callee channel, using one of the Dial GoSub/Macro options (U,M,b). https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial Note that Macro is deprecated in more recent versions of Asterisk, and the 'b' option will only be available in Asterisk 11. -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 is invoked by Person A in the outbound context and AUDIOHOOK_INHERIT(MixMonitor)=yes is also set Person a transfers Person B to Person C (another asterisk peer) Person A is no longer involved in the call and the call is bridged between Person B and Person C The call recording stops as soon as Person A hangs up, even though AUDIOHOOK_INHERIT is set Is there any way we can get the entire call recorded in one file? Thanks in advance Ish Has anyone else encountered this as it's becoming a real problem. Does anyone know a way of getting continuity of call recording in this scenario? -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 still be missing some subtle little thing. Wow, this is taking on a life of it's own. What am I missing? Not reading the DTMF tones. Thus not executing the macro. Keep in mind, that if I execute the macro manually (put in right in my extension declaration in extensions.conf, it works) Let me know if you want to see anything (parameters, etc) Thanks Glen On 4/9/2011 20:51, Dan Journo wrote: If you don't want to record every call, you can give the operator the option of press *1. We did this by adding the following to features.conf:- MixMonApp = *1,self/both,Macro,mixmon As brought up in another post, I forgot to add the following:- DYNAMIC_FEATURES=MixMonApp, either 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. Thanks to Warren Selby from http://www.selbytech.com for pointing that out. Dan Journo Kesher Communications (UK) Business Phone Systems http://www.keshercommunications.com/ | Hosted PBX http://www.keshercommunications.com/hostedpbx.html -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 tones correctly. What method are you using to transmit the dtmf tones? Regards Dan Journo Kesher Communications (UK) Business Phone Systemshttp://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.html -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end '*' received on SIP/6000-002e, duration 186 ms [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end accepted with begin '*' on SIP/6000-002e [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end passthrough '*' on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF begin '1' received on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF begin passthrough '1' on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end '1' received on SIP/6000-002e, duration 193 ms [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end accepted with begin '1' on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end passthrough '1' on SIP/6000-002e It looks like Asterisk hasnt added the new details from features.conf. You may need to fully restart Asterisk in order to get this to work. Dan Journo Kesher Communications (UK) Business Phone Systemshttp://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.html -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 14:37, Dan Journo wrote: I set the logger.conf to show reading of DTMF tones as per your instructions below. This is what I see: [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF begin '*' received on SIP/6000-002e [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF begin passthrough '*' on SIP/6000-002e [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end '*' received on SIP/6000-002e, duration 186 ms [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end accepted with begin '*' on SIP/6000-002e [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end passthrough '*' on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF begin '1' received on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF begin passthrough '1' on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end '1' received on SIP/6000-002e, duration 193 ms [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end accepted with begin '1' on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end passthrough '1' on SIP/6000-002e It looks like Asterisk hasnt added the new details from features.conf. You may need to fully restart Asterisk in order to get this to work. Dan Journo Kesher Communications (UK) Business Phone Systems http://www.keshercommunications.com/ | Hosted PBX http://www.keshercommunications.com/hostedpbx.html -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Systemshttp://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.html -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. Looking back through the email trail, I think I must have overlooked that. My bad. However, I thank all of you for your patience and help. Nice to have friends in high places! Thank you again. Guinness for everyone! Glen On 4/10/2011 17:09, Dan Journo wrote: 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 http://www.keshercommunications.com/ | Hosted PBX http://www.keshercommunications.com/hostedpbx.html -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Systemshttp://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.html -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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, 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. Thanks to Warren Selby from http://www.selbytech.com for pointing that out. Dan Journo Kesher Communications (UK) Business Phone Systemshttp://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.html -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 MixMonApp =
*1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I
cannot get it to work.
Steps.
1. added the example macro to the dialplan in extensions.conf
2. added the line MixMonApp = *1,self/both,Macro,mixmon to the
features.conf file under [applicationmap]
3. sip reload / dialplan reload / reload res_features
4. see the message that 'Mapping Feature 'apps' to app 'Macro(callrec)'
5. make incoming call - answer with SIP phone
6. I press *1 on the keypad, I hear the tones, but it does not begin
recording
7. see nothing in the CLI and no new files get created in
/var/spool/asterisk/monitor directory.
What am I missing? Probably something simple.
Any words of wisdom?
Glen
On 4/6/2011 07:29, Dan Journo wrote:
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 out?
We give our clients to option of either recording all calls, or
allowing the operator to press *1 during a call to start recording
manually.
Using Asterisk 1.4, this is what we do:-
We created a Macro in extensions.conf like this:-
[macro-mixmon]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} =
]?startrec:donothing)
exten = s,n(startrec),GotoIf($[${ARG1}=]?beep:nobeep)
exten = s,n(beep),Playback(/var/lib/asterisk/sounds/rec1)
exten = s,n(nobeep),Set(XAD=1)
exten = s,n,MixMonitor(FILENAME.wav,b)
exten = s,n(donothing),MacroExit
(please note, FILENAME.wav is usually ${UNIQUEID:0:10}, but I changed
it to make it easier for you to understand. You'll need to change it
back to something like ${UNIQUEID:0:10}.wav if you are recording
multiple calls because otherwise they'll be constantly saved to
FILENAME.wav and you'll lose all the previous calls.)
(please note, /var/lib/asterisk/sounds/rec1 is a beep tone so that the
operator knows that he's successfully started the recording.)
Then to recording every call, we add this before the
DIAL(SIP/extension) command in extensions.conf:-
exten = _9.,14,Macro(mixmon,nobeep)
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
Hope that helps.
Dan Journo
Kesher Communications (UK)
Business Phone Systems http://www.keshercommunications.com/ | Hosted
PBX http://www.keshercommunications.com/hostedpbx.html
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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 szilvertho...@gmail.comwrote: Dan et al; This looks like a perfect solution. snip It pretty much is. I've used it in similar situations. I was just about to respond to your original post, but I see you reposted here, so I'll respond here. Steps. 1. added the example macro to the dialplan in extensions.conf 2. added the line MixMonApp = *1,self/both,Macro,mixmon to the features.conf file under [applicationmap] 3. sip reload / dialplan reload / reload res_features 4. see the message that 'Mapping Feature 'apps' to app 'Macro(callrec)' 5. make incoming call - answer with SIP phone 6. I press *1 on the keypad, I hear the tones, but it does not begin recording 7. see nothing in the CLI and no new files get created in /var/spool/asterisk/monitor directory. What am I missing? Probably something simple. 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. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording - methodology
On Wed, Apr 6, 2011 at 5:54 AM, Silver Thorne szilvertho...@gmail.comwrote: Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hrm Try googling MixMonitorAsterisk has built in call recording -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 out?
We give our clients to option of either recording all calls, or allowing the
operator to press *1 during a call to start recording manually.
Using Asterisk 1.4, this is what we do:-
We created a Macro in extensions.conf like this:-
[macro-mixmon]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} = ]?startrec:donothing)
exten = s,n(startrec),GotoIf($[${ARG1}=]?beep:nobeep)
exten = s,n(beep),Playback(/var/lib/asterisk/sounds/rec1)
exten = s,n(nobeep),Set(XAD=1)
exten = s,n,MixMonitor(FILENAME.wav,b)
exten = s,n(donothing),MacroExit
(please note, FILENAME.wav is usually ${UNIQUEID:0:10}, but I changed it to
make it easier for you to understand. You'll need to change it back to
something like ${UNIQUEID:0:10}.wav if you are recording multiple calls because
otherwise they'll be constantly saved to FILENAME.wav and you'll lose all the
previous calls.)
(please note, /var/lib/asterisk/sounds/rec1 is a beep tone so that the operator
knows that he's successfully started the recording.)
Then to recording every call, we add this before the DIAL(SIP/extension)
command in extensions.conf:-
exten = _9.,14,Macro(mixmon,nobeep)
If you don't want to record every call, you can give the operator the option of
press *1. We did this by adding the following to features.conf:-
MixMonApp = *1,self/both,Macro,mixmon
Hope that helps.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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Re: [asterisk-users] Call recording - methodology
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, April 06, 2011 6:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call recording - methodology
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I
have to modify it to make it easier to use, I do not mind.
Does anyone know of any opensource or otherwise solutions out there that
I can try out?
We give our clients to option of either recording all calls, or allowing the
operator to press *1 during a call to start recording manually.
Using Asterisk 1.4, this is what we do:-
We created a Macro in extensions.conf like this:-
[macro-mixmon]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} =
]?startrec:donothing)
exten = s,n(startrec),GotoIf($[${ARG1}=]?beep:nobeep)
exten = s,n(beep),Playback(/var/lib/asterisk/sounds/rec1)
exten = s,n(nobeep),Set(XAD=1)
exten = s,n,MixMonitor(FILENAME.wav,b)
exten = s,n(donothing),MacroExit
(please note, FILENAME.wav is usually ${UNIQUEID:0:10}, but I changed it to
make it easier for you to understand. You'll need to change it back to
something like ${UNIQUEID:0:10}.wav if you are recording multiple calls
because otherwise they'll be constantly saved to FILENAME.wav and you'll
lose all the previous calls.)
(please note, /var/lib/asterisk/sounds/rec1 is a beep tone so that the
operator knows that he's successfully started the recording.)
Then to recording every call, we add this before the DIAL(SIP/extension)
command in extensions.conf:-
exten = _9.,14,Macro(mixmon,nobeep)
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
Hope that helps.
Dan Journo
Kesher Communications (UK)
Business Phone Systems http://www.keshercommunications.com/ | Hosted PBX
http://www.keshercommunications.com/hostedpbx.html
[Danny Nicholas]
Good solution, Dan - 2 additions - asterisk has a beep sound built in to
most sound sets and there is also a nice disclaimer file you can use
this-call-may-be-monitored-or-recorded
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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 that was metioned earlier. I *think* you mean just the structure version of the format options I presented, because for example: Microsoft PCM (wav) files can be of varying quality levels (192Khz, 256Khz..8bit 16 bit 24...32)..This is true (as you know, I'm more than sure) of almost every audio file format... So, is it Structure of data/packets or sample rate, bitrate, etc' ? Thanks mate! -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 with if recording a call that was using one of the higher quality codecs that was metioned earlier. I *think* you mean just the structure version of the format options I presented, because for example: Microsoft PCM (wav) files can be of varying quality levels (192Khz, 256Khz..8bit 16 bit 24...32)..This is true (as you know, I'm more than sure) of almost every audio file format... So, is it Structure of data/packets or sample rate, bitrate, etc' ? That would be structure of data stored in the file. At the point where the file format comes into play, the samples are already in their final stage of computation. The only thing that remains is how the samples are wrapped for storage. -- Tilghman -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording audio file quality query
On Wed, Feb 9, 2011 at 12:31 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Wednesday 09 February 2011 03:50:51 Sherwood McGowan wrote: Tilghman, When you say reformat the audio, do you mean sample rate and bits per sample, etc...or do you mean the format in which each packet of data 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 that was metioned earlier. I *think* you mean just the structure version of the format options I presented, because for example: Microsoft PCM (wav) files can be of varying quality levels (192Khz, 256Khz..8bit 16 bit 24...32)..This is true (as you know, I'm more than sure) of almost every audio file format... So, is it Structure of data/packets or sample rate, bitrate, etc' ? That would be structure of data stored in the file. At the point where the file format comes into play, the samples are already in their final stage of computation. The only thing that remains is how the samples are wrapped for storage. -- Tilghman thanks for confirming! -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording audio file quality query
On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi We're getting requests coming in for higher quality audio in our call recordings. We currently use MixMonitor and everything is being saved in it's native 8000Hz, 16 bit wav format. I have seen information on using Monitor and specifying a conversion to mp3 when the call ends and the 2 channels get mixed but surely the 2 channels are already saved as 16bit 8000Hz wav files so the quality is lost already? Is there any way of making high quality recordings of call content? Have you ever heard of the saying You can't polish a turd ? It doesn't matter if you have an app capable of recording 196Khz 24bit recordings (or capable of upsampling to that sample rate)...if the call itself is native at 8Khz 16bit, you'd just be making a bigger recording file with no literal improvement in quality. You can't create more samples of audio from nothing. it's like taking a new box of, say, 50 paperclips... Now, go get an empty box that says it contained 250 paperclips when it was purchased... Now, throw all 50 paperclips from the little box into the big box marked 250..now, imagine REALLY REALLY hard that you think you can perceive about 5 more paperclips somewhere all mixed up in the jumble...(Extrapolation) that, my friend, is an over simplified metaphor, but in essence it's close enough to get the point across.. Sorry bud :( If you don't believe me, I can refer you to my old audio production school ;-D ) Slainte! the Mick -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording audio file quality query
On Tue, 2011-02-08 at 05:40 -0600, Sherwood McGowan wrote: On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi 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 channels get mixed but surely the 2 channels are already saved as 16bit 8000Hz wav files so the quality is lost already? Is there any way of making high quality recordings of call content? Have you ever heard of the saying You can't polish a turd ? It doesn't matter if you have an app capable of recording 196Khz 24bit recordings (or capable of upsampling to that sample rate)...if the call itself is native at 8Khz 16bit, you'd just be making a bigger recording file with no literal improvement in quality. You can't create more samples of audio from nothing. it's like taking a new box of, say, 50 paperclips... Now, go get an empty box that says it contained 250 paperclips when it was purchased... Now, throw all 50 paperclips from the little box into the big box marked 250..now, imagine REALLY REALLY hard that you think you can perceive about 5 more paperclips somewhere all mixed up in the jumble...(Extrapolation) that, my friend, is an over simplified metaphor, but in essence it's close enough to get the point across.. Sorry bud :( If you don't believe me, I can refer you to my old audio production school ;-D ) Slainte! the Mick That answer was pretty much what I was expecting. Just wanted to make sure. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 channels get mixed but surely the 2 channels are already saved as 16bit 8000Hz wav files so the quality is lost already? Is there any way of making high quality recordings of call content? We're currently using asterisk 1.4 and soon upgrading to 1.8 Thanks in Advance Switch everything to ulaw/alaw codecs, and stop using highly compressed codecs As for 16bit, 8khz, that is as high as your going to get in the telephone world. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording audio file quality query
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 the calls and record call using packet capture and assembling regardless of server as asterisk still will not be able to record call in HD but some other switches like FreeSWITCH can do it or you need to write your own app like it. -Original Message- From: Ishfaq Malik i...@pack-net.co.uk Sent: Tuesday, February 8, 2011 6:47am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call Recording audio file quality query On Tue, 2011-02-08 at 05:40 -0600, Sherwood McGowan wrote: On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi 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 channels get mixed but surely the 2 channels are already saved as 16bit 8000Hz wav files so the quality is lost already? Is there any way of making high quality recordings of call content? Have you ever heard of the saying You can't polish a turd ? It doesn't matter if you have an app capable of recording 196Khz 24bit recordings (or capable of upsampling to that sample rate)...if the call itself is native at 8Khz 16bit, you'd just be making a bigger recording file with no literal improvement in quality. You can't create more samples of audio from nothing. it's like taking a new box of, say, 50 paperclips... Now, go get an empty box that says it contained 250 paperclips when it was purchased... Now, throw all 50 paperclips from the little box into the big box marked 250..now, imagine REALLY REALLY hard that you think you can perceive about 5 more paperclips somewhere all mixed up in the jumble...(Extrapolation) that, my friend, is an over simplified metaphor, but in essence it's close enough to get the point across.. Sorry bud :( If you don't believe me, I can refer you to my old audio production school ;-D ) Slainte! the Mick That answer was pretty much what I was expecting. Just wanted to make sure. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording audio file quality query
On Tue, Feb 8, 2011 at 6:01 AM, fai...@vopium.com wrote: But if you are getting calls all the way on VoIP then you can have calls in HD audio using HD audio codec on all locations (Server and Client). In that case you either need use some available 3rd party solution which uses packet capturing to trace the calls and record call using packet capture and assembling regardless of server as asterisk still will not be able to record call in HD but some other switches like FreeSWITCH can do it or you need to write your own app like it. It's not difficult at all to perform what you're referring to..If you have the hardware... A simple way is to have a port on your main network switch/router that will firehose the traffic the device interacts with In case someone reading this doesn't know, I'm talking about having a port that just makes a copy of EVERY PACKET that the device sees and sends those copies out over the port that you've set up for the purpose..It just GUSHES data over that port...like a firehose just gushes out all the water it possibly can... LOL Anyway, once your data is being mirrored over that firehose, send it to a dedicated recording server...all it has to do is find the signaling packets for each call and then just dump the payload from the RTP. It'll come out exactly as it was transported within RTP...in the codec the call set up I may be wrong, but I'm fairly sure that Asterisk can write a filetype for almost any of it's codecs...I know it can READ audio files that are encoded in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... If the DECoding portion is there, there's almost GOT to be the enCOding functionality... -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 sherwood.mcgo...@gmail.com Sent: Tuesday, February 8, 2011 7:34am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call Recording audio file quality query On Tue, Feb 8, 2011 at 6:01 AM, [mailto:fai...@vopium.com] fai...@vopium.com wrote: But if you are getting calls all the way on VoIP then you can have calls in HD audio using HD audio codec on all locations (Server and Client). In that case you either need use some available 3rd party solution which uses packet capturing to trace the calls and record call using packet capture and assembling regardless of server as asterisk still will not be able to record call in HD but some other switches like FreeSWITCH can do it or you need to write your own app like it. It's not difficult at all to perform what you're referring to..If you have the hardware... A simple way is to have a port on your main network switch/router that will firehose the traffic the device interacts with In case someone reading this doesn't know, I'm talking about having a port that just makes a copy of EVERY PACKET that the device sees and sends those copies out over the port that you've set up for the purpose..It just GUSHES data over that port...like a firehose just gushes out all the water it possibly can... LOL Anyway, once your data is being mirrored over that firehose, send it to a dedicated recording server...all it has to do is find the signaling packets for each call and then just dump the payload from the RTP. It'll come out exactly as it was transported within RTP...in the codec the call set up I may be wrong, but I'm fairly sure that Asterisk can write a filetype for almost any of it's codecs...I know it can READ audio files that are encoded in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... If the DECoding portion is there, there's almost GOT to be the enCOding functionality... -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording audio file quality query
yep..that would be what i said, using the nifty slang my peeps use in the datacenters I just wanted to be cool like them...*hangs head*... great...now I gotta transfer to another school... LOL, have a good one mate! On Tue, Feb 8, 2011 at 7:23 AM, fai...@vopium.com wrote: Yes. The technology need to be used on LAN switches is port mirroring or line tapping -Original Message- From: Sherwood McGowan sherwood.mcgo...@gmail.com Sent: Tuesday, February 8, 2011 7:34am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call Recording audio file quality query On Tue, Feb 8, 2011 at 6:01 AM, fai...@vopium.com wrote: But if you are getting calls all the way on VoIP then you can have calls in HD audio using HD audio codec on all locations (Server and Client). In that case you either need use some available 3rd party solution which uses packet capturing to trace the calls and record call using packet capture and assembling regardless of server as asterisk still will not be able to record call in HD but some other switches like FreeSWITCH can do it or you need to write your own app like it. It's not difficult at all to perform what you're referring to..If you have the hardware... A simple way is to have a port on your main network switch/router that will firehose the traffic the device interacts with In case someone reading this doesn't know, I'm talking about having a port that just makes a copy of EVERY PACKET that the device sees and sends those copies out over the port that you've set up for the purpose..It just GUSHES data over that port...like a firehose just gushes out all the water it possibly can... LOL Anyway, once your data is being mirrored over that firehose, send it to a dedicated recording server...all it has to do is find the signaling packets for each call and then just dump the payload from the RTP. It'll come out exactly as it was transported within RTP...in the codec the call set up I may be wrong, but I'm fairly sure that Asterisk can write a filetype for almost any of it's codecs...I know it can READ audio files that are encoded in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... If the DECoding portion is there, there's almost GOT to be the enCOding functionality... -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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, fai...@vopium.com wrote: But if you are getting calls all the way on VoIP then you can have calls in HD audio using HD audio codec on all locations (Server and Client). In that case you either need use some available 3rd party solution which uses packet capturing to trace the calls and record call using packet capture and assembling regardless of server as asterisk still will not be able to record call in HD but some other switches like FreeSWITCH can do it or you need to write your own app like it. It's not difficult at all to perform what you're referring to..If you have the hardware... A simple way is to have a port on your main network switch/router that will firehose the traffic the device interacts with In case someone reading this doesn't know, I'm talking about having a port that just makes a copy of EVERY PACKET that the device sees and sends those copies out over the port that you've set up for the purpose..It just GUSHES data over that port...like a firehose just gushes out all the water it possibly can... LOL Anyway, once your data is being mirrored over that firehose, send it to a dedicated recording server...all it has to do is find the signaling packets for each call and then just dump the payload from the RTP. It'll come out exactly as it was transported within RTP...in the codec the call set up I may be wrong, but I'm fairly sure that Asterisk can write a filetype for almost any of it's codecs...I know it can READ audio files that are encoded in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... If the DECoding portion is there, there's almost GOT to be the enCOding functionality... Actually, the writing of encoded voice has nothing to do with codecs. The format modules simply expect a particular type of packet to be fed in, and they simply reformat the audio (without transcoding) to be stored on disk. One caveat is that the format in which they are stored on disk is not guaranteed to be a standard format that is at all useful to outside utilities; just that Asterisk can read it off disk and reassemble the packets. -- Tilghman -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 alaw wav since we are using alaw on both parties, but its not. Thanks, Vilius. On 22 November 2010 14:19, Joel Maslak jmas...@antelope.net wrote: What format are the actual calls in? Are they in G.711u/a format or are they in something else (perhaps gsm?) format? I'm asking to find out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a 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 call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vilius Adamkavicius InVADE Technical Support 3 Berkeley Crescent, Bristol United Kingdom BS8 1HA Company Registration Number: 3660482 Registered in England and Wales this email, and any attachment, is intended only for the attention of the addressee. Its unauthorised use, disclosure, storage or copying is not permitted. If you are not the intended recipient, please destroy all copies and inform the sender by return email. If you have received this email in error, please return it to the sender and highlight the error. We accept no legal liability for the content of the message. Any opinions or views presented are solely the responsibility of the author and do not necessarily represent those of InVADE. We cannot guarantee that this message has not been modified in transit, and this message should not be viewed as contractually binding. Although we have taken reasonable steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. international phone number +44(0) 117 33 555 00 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording format
On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. Ignoring your real questions, and asking an alternate question: Why not just record in gsm? If your answer is that you have to play these back on Windows, you can build an on-the-fly gsm-to-wav converter using sox. My understanding is that recording in wav doesn't exactly make you have higher audio quality in your recordings, although the experts at codecs could better answer that. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 customers require call recordings to be done in wav. Thanks, Vilius. On 22 November 2010 15:03, David Backeberg dbackeb...@gmail.com wrote: On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. Ignoring your real questions, and asking an alternate question: Why not just record in gsm? If your answer is that you have to play these back on Windows, you can build an on-the-fly gsm-to-wav converter using sox. My understanding is that recording in wav doesn't exactly make you have higher audio quality in your recordings, although the experts at codecs could better answer that. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 system resources to convert it back to wav is not a solution. Also some of our customers require call recordings to be done in wav. wav with signed linear payload? I wonder what would happen if you record it as .sl (raw signed linear) and convert it to wav at the end of the call (while mixing). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. On Mon, Nov 22, 2010 at 7:58 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: 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 alaw wav since we are using alaw on both parties, but its not. Thanks, Vilius. On 22 November 2010 14:19, Joel Maslak jmas...@antelope.net wrote: What format are the actual calls in? Are they in G.711u/a format or are they in something else (perhaps gsm?) format? I'm asking to find out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a 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 call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vilius Adamkavicius InVADE Technical Support 3 Berkeley Crescent, Bristol United Kingdom BS8 1HA Company Registration Number: 3660482 Registered in England and Wales this email, and any attachment, is intended only for the attention of the addressee. Its unauthorised use, disclosure, storage or copying is not permitted. If you are not the intended recipient, please destroy all copies and inform the sender by return email. If you have received this email in error, please return it to the sender and highlight the error. We accept no legal liability for the content of the message. Any opinions or views presented are solely the responsibility of the author and do not necessarily represent those of InVADE. We cannot guarantee that this message has not been modified in transit, and this message should not be viewed as contractually binding. Although we have taken reasonable steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. international phone number +44(0) 117 33 555 00 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 jmas...@antelope.net wrote: 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. On Mon, Nov 22, 2010 at 7:58 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: 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 alaw wav since we are using alaw on both parties, but its not. Thanks, Vilius. On 22 November 2010 14:19, Joel Maslak jmas...@antelope.net wrote: What format are the actual calls in? Are they in G.711u/a format or are they in something else (perhaps gsm?) format? I'm asking to find out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a 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 call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vilius Adamkavicius InVADE Technical Support 3 Berkeley Crescent, Bristol United Kingdom BS8 1HA Company Registration Number: 3660482 Registered in England and Wales this email, and any attachment, is intended only for the attention of the addressee. Its unauthorised use, disclosure, storage or copying is not permitted. If you are not the intended recipient, please destroy all copies and inform the sender by return email. If you have received this email in error, please return it to the sender and highlight the error. We accept no legal liability for the content of the message. Any opinions or views presented are solely the responsibility of the author and do not necessarily represent those of InVADE. We cannot guarantee that this message has not been modified in transit, and this message should not be viewed as contractually binding. Although we have taken reasonable steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. international phone number +44(0) 117 33 555 00 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk?
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 ${CDR(start)} contains a space in part of the date, the filename
becomes corrupted when I use samba and share the file over a network.
Therefore I need to replace the spaces with another valid character.
Any ideas how I can do this (simply)?
Here is the macro that i'm using to trigger call recording when the user
presses *1.
[macro-mixmon]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} = ]?startrec:donothing)
exten = s,n(startrec),Playback(beep)
exten = s,n,Set(XAD=1)
exten =
s,n,MixMonitor(/var/lib/asterisk/clientsounds/${CDR(start)}~${CALLFROM}~${CDR(channel):4}.wav,b)
exten = s,n(donothing),MacroExit
Thanks
Dan
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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 ${CDR(start)} contains a space in part of the date, the
filename becomes corrupted when I use samba and share the file over a network.
Therefore I need to replace the spaces with another valid character.
Any ideas how I can do this (simply)?
Here is the macro that i'm using to trigger call recording when the user
presses *1.
[macro-mixmon]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} = ]?startrec:donothing)
exten = s,n(startrec),Playback(beep)
exten = s,n,Set(XAD=1)
exten =
s,n,MixMonitor(/var/lib/asterisk/clientsounds/${CDR(start)}~${CALLFROM}~${CDR(channel):4}.wav,b)
Are you sure it is the space which is corrupting it? The space is not
incompatible with either Samba or Linux filesystem. However, is the ~
character part of the filename you are creating? If it is, that is
definitely an illegal/reserved character in the Linux file systems.
Sebastian
exten = s,n(donothing),MacroExit
Thanks
Dan
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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 introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude. Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in. However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working. Can anyone assist? Thanks Dan -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I’ve added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don’t know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Thanks Dan -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude. Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in. However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working. Can anyone assist? Thanks Dan -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 recordings in directories based on date. If you just want to sync files rather than move, just change the mv commands to cp commands. Script attached. On 09/02/2010 12:27 PM, Gareth Blades wrote: 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 create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude. Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in. However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working. Can anyone assist? Thanks Dan p style=margin: 0; padding: 0; border-collapse: collapse; font-family: Tahoma, Arial, Sans-Serif; font-size: 10px; color: #33; ---DISCLAIMERbr / The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free./p MoveCallRecs.sh Description: Bourne shell script -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude. Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in. However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working. Can anyone assist? Thanks Dan We've mounted a separate storage device onto both the web server and asterisk server. The recorded calls are saved directly onto the storage device and the web server can read off it directly too. This has the added advantage of allowing the web server to create sub directories on the monitor directory if you have more than one client using the same asterisk server -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 01615556...@supplier -- SIP/supplier-0055 is making progress passing it to SIP/clientone_201-0054 -- SIP/supplier-0055 answered SIP/clientone_201-0054 -- SIP/kesher_201-0054 Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1283429941-SIP-clientone_201-0054-01615556607|m -- SIP/supplier-0055 Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1283429941-01615556607-SIP-clientone_201-0054|m Thanks Dan -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 what's going on there?
Heres the CLI output:-
-- Called 01615556...@supplier
-- SIP/supplier-0055 is making progress passing it to
SIP/clientone_201-0054
-- SIP/supplier-0055 answered SIP/clientone_201-0054
-- SIP/kesher_201-0054 Playing 'beep' (language 'en')
-- User hit '*1' to record call. filename:
wav|auto-1283429941-SIP-clientone_201-0054-01615556607|m
-- SIP/supplier-0055 Playing 'beep' (language 'en')
-- User hit '*1' to record call. filename:
wav|auto-1283429941-01615556607-SIP-clientone_201-0054|m
Thanks
Dan
Sounds like it's using Monitor rather than MixMonitor.
I had a quick look at this:
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf
And it looks like you might be better off creating your own macro for
one touch recording and adding it to the features.conf as shown in this
part of that web page
Examples
One Touch Recording (applicationmap) with WAV to MP3 Conversion Macro.
extensions.conf :
[macro-apprecord]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} = ]?startrec:stoprec)
exten = s,n(startrec),Playback(startmonitor)
exten = s,n,Set(XAD=1)
exten = s,n,Set(FILENAME=${TIMESTAMP}-OUT
${CALLERID(number)}-^-${UNIQUEID})
exten = s,n,Set(MONITOR_EXEC_ARGS= nice -n 19 /usr/local/bin/lame -b
96 -t -F -m m --bitwidth 16 --quiet
/var/spool/asterisk/monitor/${FILENAME}.wav
/var/spool/asterisk/monitor/${FILENAME}.mp3 rm -f
/var/spool/asterisk/monitor/${FILENAME}.wav)
exten = s,n,Monitor(wav,${FILENAME},m)
exten = s,n,MacroExit
exten = s,n(stoprec),StopMonitor
exten = s,n,Set(XAD=0)
exten = s,n,Playback(stopmonitor)
exten = s,n,MacroExit
features.conf :
apps = *9,caller,Macro,apprecord
but using MixMonitor rather than monitor.
Let me know how you got on with it as I think I'm going to be asked to
do this in the next month or 2.
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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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 different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude. Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in. However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working. Can anyone assist? Thanks Dan We've mounted a separate storage device onto both the web server and asterisk server. The recorded calls are saved directly onto the storage device and the web server can read off it directly too. This has the added advantage of allowing the web server to create sub directories on the monitor directory if you have more than one client using the same asterisk server Beware that if you have lots of concurrent calls writing all these simultaneously to disk can be heavy on the disk I/O load. 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. You could use a ram disk if you have enough memory, this is probably cheaper. Also if the share you're writing directly to goes down call recordings will stop being written, where as if you try and copy/move them after they're finished then the cp/mv will just fail but your recordings will still be written locally and stored up until the share is available again. And yes, sounds like its using Monitor() and not MixMonitor(). p style=margin: 0; padding: 0; border-collapse: collapse; font-family: Tahoma, Arial, Sans-Serif; font-size: 10px; color: #33; ---DISCLAIMERbr / The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free./p -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 the physical servers. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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.drishti-soft.com B: http://blog.drishti-soft.com On Thu, Sep 2, 2010 at 7:22 PM, Dan Journo d...@keshercommunications.comwrote: 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 the physical servers. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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' command. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 the call has ended. True, but if you need to execute a process at the end of the call, why not make it an AGI and hide all the ugly details and keep your dialplan nice and clean and shiny and maintainable? Your recordings will be instantly available and the correct operation of your system does not depend on an externally scheduled external process involving clear-text passwords and obscure packages. Your successor will thank you :) This is true, doing everything from inside an AGI script would be nicer, the ugly part comes if you are tied to an old and ugly FTP server, specially if its from a hosting provider that limits the connection count to 2, or so. AGI+sshfs/scp/nfs/whateverfs... would be much more cleanshiny (tm). -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Steve's spot on about wanting to move the calls when they're available. We're the instant gratification society. Anytime I've linked multiple independent systems together I've always had to plan for one system being offline - either for maintenance, upgrades, etc. so please consider that in your solution. Your successor will thank you for that too! If you were only posting per call via AGI, then you might have some issues if the receiving server was unavailable. Ivan's solution handles that cleanly (it would simply try moving the file again when the script is invoked again). What to do?...Combine approaches!! You could add some error checking to Ivan's script that tests if the file system is mounted (or another copy of the script is running) before invoking the other commands and exit if it is (e.g. look at the exit status code of curlftpfs, maybe?). Doing so gives you three major benefits: Prevents multiple copies of the script from running and trying to process the same file (low probability, but theoretical) Lets you call the script more frequently without having to worry about multiple processes running simultaneously (the need to manage your concurrency). Gives you a way to safely call this on a per call basis from the dialplan (right after the h,1,System(move the file to the upload directory)) to get the trigger for instant gratification. To Steve's other point, you could put all of this into an AGI program/script, but you'll still also need a fallback mechanism to actually copy the files to the remote server in the event that it is unavailable/unreachable. To me, having two lines in the dialplan versus one is no big deal. Just make sure you add comments for it so your successors know 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 Discussion Subject: 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 the call has ended. True, but if you need to execute a process at the end of the call, why not make it an AGI and hide all the ugly details and keep your dialplan nice and clean and shiny and maintainable? Your recordings will be instantly available and the correct operation of your system does not depend on an externally scheduled external process involving clear-text passwords and obscure packages. Your successor will thank you :) This is true, doing everything from inside an AGI script would be nicer, the ugly part comes if you are tied to an old and ugly FTP server, specially if its from a hosting provider that limits the connection count to 2, or so. AGI+sshfs/scp/nfs/whateverfs... would be much more cleanshiny (tm). -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message is intended only for the use of the individual (s) or entity to which it is addressed and may contain information that is privileged, confidential, and/or proprietary to Calling Circles LLC and its affiliates. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, forwarding or copying of this communication is prohibited without the express permission of the sender. If you have received this communication in error, please notify the sender immediately and delete the original message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 lines in the dialplan versus one is no big deal. Just make sure you add comments for it so your successors know the logic behind the code. My AGI logs to syslog so I can track and follow up on failures. The simple task of can you upload a recording quickly spiraled out of dialplan territory. It kind of went like this: 1) Can you upload the recording? 2) It takes to long to upload and download the recording. Can you encode it to a low bitrate WMA? 3) Some of the recordings are too [loud|quiet]. Can you normalize them? 4) There's too much dead air at the [beginning|end] of the recording. Can you trim off the cruft? 5) It takes too long to listen to all of the questions. (My recordings are a sequence of pre-recorded questions and caller voice and DTMF responses). Can you also upload a recording of just the responses? 6) Can you save all the DTMF responses in a database so we can do some reporting. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 new recordings? I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. Many thanks Dan Journo -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 directory for new recordings? As said by others, there is no such built-in capability I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. It is really important for you that the recordings are available asap on your FTP destination? I had a similar task and I found problematic to upload the files from inside the dialplan, Is not that it can't be done, but if the FTP is slow and your number of connections limited, you may run into a problem with simultaneous calls ending and asterisk trying to upload 20 files at the same time. In my case the files could be uploaded every hour, so I made a simple bash script that uploads the new files to the FTP using 'curlftpfs', a nice command that mounts the remote FTP on a local mount point using FUSE, then the script just moves the files from the local folder to the FTP, and voila. Asterisk just takes care of moving recordings that ended to the desired path. I can post the bash script if you are interested. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Hi Iván, 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. Many thanks Dan Journo -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 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 directory for new recordings? As said by others, there is no such built-in capability I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. It is really important for you that the recordings are available asap on your FTP destination? I had a similar task and I found problematic to upload the files from inside the dialplan, Is not that it can't be done, but if the FTP is slow and your number of connections limited, you may run into a problem with simultaneous calls ending and asterisk trying to upload 20 files at the same time. In my case the files could be uploaded every hour, so I made a simple bash script that uploads the new files to the FTP using 'curlftpfs', a nice command that mounts the remote FTP on a local mount point using FUSE, then the script just moves the files from the local folder to the FTP, and voila. Asterisk just takes care of moving recordings that ended to the desired path. I can post the bash script if you are interested. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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. The script has a password inside so remember to 'chown root' and 'chmod 700' the file to protect it from other users. You have to set up a cron so the script is run every hour, normally putting the script or a link to it inside '/etc/cron.hourly' is enough. There is also a /etc/crontab file you can use to setup something more complicated if needed (ie: runing it every 2 hours, running at tea time...). read 'man cron', and 'man crontab'. You also need to install the magic part, 'curlftps'. in Debian that's the name of the package too. I use version curlftpfs 0.9.1 libcurl/7.18.2 fuse/2.5 Be careful that in *nix, file.WAV and file.wav are different files. Here is the script: #!/bin/sh MOUNT_POINT=/mnt/remote_ftp FTP_HOST=www.ftphost.com/htdocs/recordings FTP_USER=ftpusername:difficultpassword RECORDINGS=/var/spool/asterisk/monitor/upload echo Starting upload `date` echo Connecting to $FTP_HOST... curlftpfs -o user=$FTP_USER $FTP_HOST $MOUNT_POINT echo Transfering `du -hc $RECORDINGS/*.wav | grep total` ... mv -vf $RECORDINGS/*.wav $MOUNT_POINT echo Disconnecting. umount $MOUNT_POINT exit 0 -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Thanks for that. I really appreciate it! Dan -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 22:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: 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 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. The script has a password inside so remember to 'chown root' and 'chmod 700' the file to protect it from other users. You have to set up a cron so the script is run every hour, normally putting the script or a link to it inside '/etc/cron.hourly' is enough. There is also a /etc/crontab file you can use to setup something more complicated if needed (ie: runing it every 2 hours, running at tea time...). read 'man cron', and 'man crontab'. You also need to install the magic part, 'curlftps'. in Debian that's the name of the package too. I use version curlftpfs 0.9.1 libcurl/7.18.2 fuse/2.5 Be careful that in *nix, file.WAV and file.wav are different files. Here is the script: #!/bin/sh MOUNT_POINT=/mnt/remote_ftp FTP_HOST=www.ftphost.com/htdocs/recordings FTP_USER=ftpusername:difficultpassword RECORDINGS=/var/spool/asterisk/monitor/upload echo Starting upload `date` echo Connecting to $FTP_HOST... curlftpfs -o user=$FTP_USER $FTP_HOST $MOUNT_POINT echo Transfering `du -hc $RECORDINGS/*.wav | grep total` ... mv -vf $RECORDINGS/*.wav $MOUNT_POINT echo Disconnecting. umount $MOUNT_POINT exit 0 -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 while a recording is being written? Will it copy the incomplete file and then delete it? The script has a password inside so remember to 'chown root' and 'chmod 700' the file to protect it from other users. If you used sshfs you could use public keys and eliminate the password hassles. Personally, I'd still vote for uploading the file at the completion of the recording via an AGI. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Hi, 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. Dan Thank you for contacting Kesher Communications Ltd. IT Maintenance Clients can now receive a faster response by using our Live Chat and Support Service: Click Here This email and any files transmitted with it are confidential and intended solely for the recipient(s). If you are not the named addressee you should not disseminate, copy or alter this email. Under no circumstances may this email be distributed without written permission from the sender. Warning: Although the Company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. All prices exclude VAT unless otherwise stated. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: 12 October 2009 23:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Recording and Posting On Behalf Of Ivan Stepaniuk The script is very simple and far from complete, it just moves the content into the mounted FTP directory. It has some verbose output as it is run from inside another script that redirects the output to a log file. What happens if the script is run while a recording is being written? Will it copy the incomplete file and then delete it? The script has a password inside so remember to 'chown root' and 'chmod 700' the file to protect it from other users. If you used sshfs you could use public keys and eliminate the password hassles. Personally, I'd still vote for uploading the file at the completion of the recording via an AGI. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 if you need to execute a process at the end of the call, why not make it an AGI and hide all the ugly details and keep your dialplan nice and clean and shiny and maintainable? Your recordings will be instantly available and the correct operation of your system does not depend on an externally scheduled external process involving clear-text passwords and obscure packages. Your successor will thank you :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Dan, You can do this directly in the dialplan. See the System command. It allows you to call any program on the system (ftp, scp, mv, etc). Keep in mind that depending on the volume of calls you're handling, you might run into I/O issues on the disk side. If you're talking about 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-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 new recordings? I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. Many thanks Dan Journo This message is intended only for the use of the individual (s) or entity to which it is addressed and may contain information that is privileged, confidential, and/or proprietary to Calling Circles LLC and its affiliates. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, forwarding or copying of this communication is prohibited without the express permission of the sender. If you have received this communication in error, please notify the sender immediately and delete the original message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. There is no built in facility -- but there are all the parts. There is the curl() application, but I don't know if it exposes enough curl to upload files. There is the system() application which will let you execute any command line you can construct. There is the agi() application which lets an external program interact with the dialplan. Monitoring a temp directory with an external program would be the worst way. Personally, I would wrap up the entire call recording solution in an AGI so you have a full featured language (my preference is C) and can hide all the ugly details and keep your dialplan simple and maintainable. I've done these kind of applications where either a control file needed to be written and uploaded with the recording or a database needed updating. Both of these can get ugly in a dialplan. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording in - out
On Sun, Jun 7, 2009 at 12:51 PM, Joao Gomes Pereiragomespere...@startel.pt wrote: Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? On the Hangup context, you can invoke sox directly from the dialplan, feeding it the names of the files to mix and the name of the output file. Or you can use MixMonitor(), which mixes the recording on the fly into a single common file, but I don't know whether that works in 1.2. I'm using 1.6 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording in - out
David Backeberg escribió: On Sun, Jun 7, 2009 at 12:51 PM, Joao Gomes Pereiragomespere...@startel.pt wrote: Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? On the Hangup context, you can invoke sox directly from the dialplan, feeding it the names of the files to mix and the name of the output file. Or you can use MixMonitor(), which mixes the recording on the fly into a single common file, but I don't know whether that works in 1.2. I'm using 1.6 Th The old Monitor() application has the 'm' option that launches Sox automatically to mix both call leg recordings when the call hangs up. This has the great disadvantage of generating CPU spikes if the recordings are too long/simultaneous. I saw machines with a lot of pending sox processes (adjusted with a very low nice value) so the recording mixing didn't affect normal asterisk operation, or sometimes we used to do mixing on a separate server during off-hours. That was years away, before the MixMonitor() application bugs was fixed on 1.2 and it's quite stable since then. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 gomespere...@startel.pt Hello I did as you told me, but the problem remains. Im using Asterisk 1.2.x and this is my config: queues.conf - [general] persistentmembers = no [queue_1] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=MixMonitor wrapuptime=3 timeout=15 strategy=roundrobin retry=5 queue-youarenext= queue-thereare= queue-thankyou= queue-callswaiting= member = Agent/600 member = Agent/601 agents.conf - [general] persistentagents=no [agents] updatecdr=no custom_beep=beep group=1 wrapuptime=19 ackcall=no musiconhold = music group=1 agent = 600,1234,Jose agent = 601,1234,Maria The calls are recordedbut always produces 2 separated files, with in and out. What could be missing? Do I need to create a line in crontab to mix the 2 files? Thanks regards Joao Pereira Kurian Thayil wrote: 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, 2009-06-07 at 17:51 +0100, 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 happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf- [general] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor [queue_1] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor wrapuptime=3 timeout=15 strategy=roundrobin retry=5 member = Agent/600 member = Agent/601 agents.conf- [general] persistentagents=no [agents] updatecdr=no recordagentcalls=yes recordformat=wav monitor-join=yes savecallsin=/var/www/html/recordings/ custom_beep=beep group=1 wrapuptime=19 ackcall=no group=1 agent = 600,1234,Jose agent = 601,1234,Maria Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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, 2009-06-07 at 17:51 +0100, 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 happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf- [general] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor [queue_1] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor wrapuptime=3 timeout=15 strategy=roundrobin retry=5 member = Agent/600 member = Agent/601 agents.conf- [general] persistentagents=no [agents] updatecdr=no recordagentcalls=yes recordformat=wav monitor-join=yes savecallsin=/var/www/html/recordings/ custom_beep=beep group=1 wrapuptime=19 ackcall=no group=1 agent = 600,1234,Jose agent = 601,1234,Maria Thanks Regards Joao Pereira -- Kurian Mathew Thayil. (GPG KeyID: E232394F) signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 timeout=15 strategy=roundrobin retry=5 queue-youarenext= queue-thereare= queue-thankyou= queue-callswaiting= member = Agent/600 member = Agent/601 agents.conf - [general] persistentagents=no [agents] updatecdr=no custom_beep=beep group=1 wrapuptime=19 ackcall=no musiconhold = music group=1 agent = 600,1234,Jose agent = 601,1234,Maria The calls are recordedbut always produces 2 separated files, with in and out. What could be missing? Do I need to create a line in crontab to mix the 2 files? Thanks regards Joao Pereira Kurian Thayil wrote: 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, 2009-06-07 at 17:51 +0100, 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 happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf- [general] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor [queue_1] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=mixmonitor wrapuptime=3 timeout=15 strategy=roundrobin retry=5 member = Agent/600 member = Agent/601 agents.conf- [general] persistentagents=no [agents] updatecdr=no recordagentcalls=yes recordformat=wav monitor-join=yes savecallsin=/var/www/html/recordings/ custom_beep=beep group=1 wrapuptime=19 ackcall=no group=1 agent = 600,1234,Jose agent = 601,1234,Maria Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording - posible to remove recorded fileat the end of the call
Here are two ways to handle this scenario:
1. call an AGI at the end of the call
sample dialplan
exten = s,1,Set(Callid=time)
exten = s,2,Mixmonitor(${Callid}.wav)
exten = s,3,background(zeroorone)
exten = s,4,waitexten(2)
exten = s,n,hangup
exten = 0,1,system(/bin/rm /var/lib/asterisk/sounds/blah/${callid}.wav)
or
exten = 0,1,AGI(killcall.agi | ${callid}.wav)
exten = 0,2,hangup
exten = 1,1,hangup
2. present the calls in a web page and let the agent zap them there.
_
From: asterisk-users-boun...@lists.digium.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 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 recording, the agents is
pressing 0 for deleting the file or 1 for
leave the file stored.
thanks
christian gansberger
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Re: [asterisk-users] Call Recording Alias
Where did I make mistake ? On Thu, Jan 29, 2009 at 1:07 AM, David @ULC ucoms2...@gmail.com wrote: http://download.vicidial-group.com/svn/agc_2-X/trunk/docs/SCRATCH_INSTALL.txt vi /usr/local/apache2/conf/httpd.conf add the following lines: AddType application/x-httpd-php .php .phtml LoadModule php4_module libexec/libphp5.so or LoadModule php4_module modules/libphp5.so modify the index.html line and add index.php to the list to disable logging, change: CustomLog logs/access_log common to this: CustomLog /dev/null common to enable web browsing of Recordings on Asterisk server, add this: Alias /RECORDINGS/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all files *.mp3 Forcetype application/forcedownload /files /Directory - /usr/local/apache2/bin/apachectl start - go to http://your-new-asterisk-server-ipaddress/ to see if it worked - you are done NOTE: If using PHP5 you may need to add the following line to php.ini: short_open_tag = On On Thu, Jan 29, 2009 at 12:13 AM, David @ULC ucoms2...@gmail.com wrote: Modified httf.conf file and added : -- Alias /recordings/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all /Directory Created a folder under vicidial as recordings. FULL_RECORDING is also enabled. But I don't see recordings under recording folder. Any guidance ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 ucoms2...@gmail.com Modified httf.conf file and added : -- Alias /recordings/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all /Directory Created a folder under vicidial as recordings. FULL_RECORDING is also enabled. But I don't see recordings under recording folder. Any guidance ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 ucoms2...@gmail.com wrote: Modified httf.conf file and added : -- Alias /recordings/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all /Directory Created a folder under vicidial as recordings. FULL_RECORDING is also enabled. But I don't see recordings under recording folder. Any guidance ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 or LoadModule php4_module modules/libphp5.so modify the index.html line and add index.php to the list to disable logging, change: CustomLog logs/access_log common to this: CustomLog /dev/null common to enable web browsing of Recordings on Asterisk server, add this: Alias /RECORDINGS/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all files *.mp3 Forcetype application/forcedownload /files /Directory - /usr/local/apache2/bin/apachectl start - go to http://your-new-asterisk-server-ipaddress/ to see if it worked - you are done NOTE: If using PHP5 you may need to add the following line to php.ini: short_open_tag = On On Thu, Jan 29, 2009 at 12:13 AM, David @ULC ucoms2...@gmail.com wrote: Modified httf.conf file and added : -- Alias /recordings/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all /Directory Created a folder under vicidial as recordings. FULL_RECORDING is also enabled. But I don't see recordings under recording folder. Any guidance ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 or LoadModule php4_module modules/libphp5.so modify the index.html line and add index.php to the list to disable logging, change: CustomLog logs/access_log common to this: CustomLog /dev/null common to enable web browsing of Recordings on Asterisk server, add this: Alias /RECORDINGS/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all files *.mp3 Forcetype application/forcedownload /files /Directory - /usr/local/apache2/bin/apachectl start - go to http://your-new-asterisk-server-ipaddress/ to see if it worked - you are done NOTE: If using PHP5 you may need to add the following line to php.ini: short_open_tag = On On Thu, Jan 29, 2009 at 12:13 AM, David @ULC ucoms2...@gmail.com wrote: Modified httf.conf file and added : -- Alias /recordings/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all /Directory Created a folder under vicidial as recordings. FULL_RECORDING is also enabled. But I don't see recordings under recording folder. Any guidance ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Alias
Asterisk is not Apache. This being said, you can get to where you are going by hook and crook. Asterisk HTTP files live in /var/lib/asterisk/static-http, so if you do a ln -ns /var/spool/asterisk/monitorDONE /var/lib/asterisk/static-http/monitorDONE, you can access the directory with this command: http://1.2.3.4:8088/asterisk/static/monitordone http://1.2.3.4:8088/asterisk/static/monitordone%20where%201.2.3.4 where 1.2.3.4 is your * IP. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 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 29, 2009 at 12:13 AM, David @ULC ucoms2...@gmail.com wrote: Modified httf.conf file and added : -- Alias /recordings/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all /Directory Created a folder under vicidial as recordings. FULL_RECORDING is also enabled. But I don't see recordings under recording folder. Any guidance ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 ucoms2...@gmail.com Thanks for your advice , but I asked for expert guidance as I read the doc and it says like that. but somehow It didn't work out. On Thu, Jan 29, 2009 at 12:13 AM, David @ULC ucoms2...@gmail.com wrote: Modified httf.conf file and added : -- Alias /recordings/ /var/spool/asterisk/monitorDONE/ Directory /var/spool/asterisk/monitorDONE Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all /Directory Created a folder under vicidial as recordings. FULL_RECORDING is also enabled. But I don't see recordings under recording folder. Any guidance ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this meant that some phones, after checking in with Asterisk would simply communicate via RTP between each other, without going media transport going through trixbox itself? If this is the case then I guess I'd need to mirror the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason that I'm missing here? Chris, Make sure that all of your SIP clients are set to canreinvite=no in sip.conf. The default is canreinvite=yes, which allows RTP to bypass Asterisk. Certain things (codec translation, playback of audio files, etc.) require Asterisk to be in the RTP path, which may explain why you're recording some of the calls. If you're still missing calls, make sure Oreka is configured properly in config.xml. In particular, the AllowedIpRanges and BlockedIpRanges settings provide IP address filtering at the Oreka level. In general, I've had to configure these to prevent getting two recordings of each call (since Asterisk acts as a B2BUA) but your configuration may be too strict. Running tcpdump/Wireshark on the Oreka server will let you see exactly what's being mirrored. There is even a setting in Oreka named PcapFile that will let you playback the packet capture file over and over until you're satisfied with your configuration. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Matthew, Thank you so much for your advice. It's really appreciated - I'll go through it and see where I get. Thanks again Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording - Asterisk
Hello folks, I wanted to setup Oreka to monitor calls on a trixbox box I have setup. Oreka doesn't seem to be catching all of the calls though I have port mirroring setup on the port that trixbox is connected to, mirrored to the port Oreka is connected to. I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this meant that some phones, after checking in with Asterisk would simply communicate via RTP between each other, without going media transport going through trixbox itself? If this is the case then I guess I'd need to mirror the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason that I'm missing here? Just trying to get this sussed out in my head! Thanks for your time. Chris Hi again, didn't get a reply to this one. I'm a bit stumped so I thought I'd try the list one more time to see if anyone has an answer. If not, thanks for reading anyway! Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording - Asterisk
Chris Rowson wrote: I wanted to setup Oreka to monitor calls on a trixbox box I have setup. Oreka doesn't seem to be catching all of the calls though I have port mirroring setup on the port that trixbox is connected to, mirrored to the port Oreka is connected to. I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this meant that some phones, after checking in with Asterisk would simply communicate via RTP between each other, without going media transport going through trixbox itself? If this is the case then I guess I'd need to mirror the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason that I'm missing here? Chris, Make sure that all of your SIP clients are set to canreinvite=no in sip.conf. The default is canreinvite=yes, which allows RTP to bypass Asterisk. Certain things (codec translation, playback of audio files, etc.) require Asterisk to be in the RTP path, which may explain why you're recording some of the calls. If you're still missing calls, make sure Oreka is configured properly in config.xml. In particular, the AllowedIpRanges and BlockedIpRanges settings provide IP address filtering at the Oreka level. In general, I've had to configure these to prevent getting two recordings of each call (since Asterisk acts as a B2BUA) but your configuration may be too strict. Running tcpdump/Wireshark on the Oreka server will let you see exactly what's being mirrored. There is even a setting in Oreka named PcapFile that will let you playback the packet capture file over and over until you're satisfied with your configuration. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording problems from queue
Ex Vito [EMAIL PROTECTED] writes: I don't have access to an asterisk system right now (nor any other sort of information source) but I seem to recall that from 1.4 onwards the config option for recording queue calls is named differently... Is it mixmonitor ? Check you 1.4 queues.conf sample. PS: I'm not really sure about this one! Hi exvito, Mysteriously it started working today. Maybe Asterisk just needed a restart after playing with the configuration all day, I'll see if it keeps working. Thanks! ---Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording problems from queue
I don't have access to an asterisk system right now (nor any other sort of information source) but I seem to recall that from 1.4 onwards the config option for recording queue calls is named differently... Is it mixmonitor ? Check you 1.4 queues.conf sample. PS: I'm not really sure about this one! -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording on Hanup
What makes you decide wehter or not you want to keep the recorded file? Is the fact that the user hangup the call before the 30 seconds or the fact that he really talked? As far as I know current version of asterisk doesn't allow you to detect who end up the call. Of course you may use some tricks for these, I mean you may set the IVR to record a bit more than 30 seconds, then when the call hangs up when you reach the h extension in you diaplan you may check the answered time of the call. If your call has an answered time duration lower than 30 seconds, for sure was the caller who hangup the call. Hope it helps. On Dec 18, 2007 8:04 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: yes, the senario is this when user gets a call IVR starts playing and after hearing beep user starts recording message for 30 seconds(call duration is for 30 seconds). What i want is During 30 seconds if user does hangup his/her call then message should be recorded otherwise(after timeout) message is discarded. Is there any thing that will help me...??? currently I am doing the same thing on pressing 1 with php agi script and its working fine. On 12/19/07, Marco Mouta [EMAIL PROTECTED] wrote: What do you mean with record a call on hangup? If the calling party ends the call you want to keep recorded file? On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording on Hanup
sure it will really help me thanx for responding. On 12/19/07, Marco Mouta [EMAIL PROTECTED] wrote: What makes you decide wehter or not you want to keep the recorded file? Is the fact that the user hangup the call before the 30 seconds or the fact that he really talked? As far as I know current version of asterisk doesn't allow you to detect who end up the call. Of course you may use some tricks for these, I mean you may set the IVR to record a bit more than 30 seconds, then when the call hangs up when you reach the h extension in you diaplan you may check the answered time of the call. If your call has an answered time duration lower than 30 seconds, for sure was the caller who hangup the call. Hope it helps. On Dec 18, 2007 8:04 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: yes, the senario is this when user gets a call IVR starts playing and after hearing beep user starts recording message for 30 seconds(call duration is for 30 seconds). What i want is During 30 seconds if user does hangup his/her call then message should be recorded otherwise(after timeout) message is discarded. Is there any thing that will help me...??? currently I am doing the same thing on pressing 1 with php agi script and its working fine. On 12/19/07, Marco Mouta [EMAIL PROTECTED] wrote: What do you mean with record a call on hangup? If the calling party ends the call you want to keep recorded file? On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording on Hanup
What do you mean with record a call on hangup? If the calling party ends the call you want to keep recorded file? On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording on Hanup
yes, the senario is this when user gets a call IVR starts playing and after hearing beep user starts recording message for 30 seconds(call duration is for 30 seconds). What i want is During 30 seconds if user does hangup his/her call then message should be recorded otherwise(after timeout) message is discarded. Is there any thing that will help me...??? currently I am doing the same thing on pressing 1 with php agi script and its working fine. On 12/19/07, Marco Mouta [EMAIL PROTECTED] wrote: What do you mean with record a call on hangup? If the calling party ends the call you want to keep recorded file? On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording on demand...
Look at features.conf /b On Oct 10, 2007, at 10:48 AM, Reggie Payne wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording on demand...
Ok. I know you have to use touch monitor but what I am after is the variables that need to be specified and where in the extensions.conf to configure for users? Brian West [EMAIL PROTECTED] 10/10/2007 12:00 PM Look at features.conf /b On Oct 10, 2007, at 10:48 AM, Reggie Payne wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording on demand...
And is there a way the automon can send the result to voicemail? I hadn't found that yet. Moj Reggie Payne wrote: Ok. I know you have to use touch monitor but what I am after is the variables that need to be specified and where in the extensions.conf to configure for users? Brian West [EMAIL PROTECTED] 10/10/2007 12:00 PM Look at features.conf /b On Oct 10, 2007, at 10:48 AM, Reggie Payne wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording on demand...
On 10/10/2007, Reggie Payne [EMAIL PROTECTED] wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg Are you suggesting that all of a call is recorded and if a certain key sequence is not entered during the call, the recording is completely discarded otherwise the complete call is saved. Or are you suggesting the call is only recorded from the point you enter a specific key sequence? Ray ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording on demand...
The call is recorded after a key sequence has been pressed. Example: SIP/101 makes an outbound call to 5551212 5551212 starts to get rowdy SIP/101 enters *99 to start recording the call After the call is ended the recording is sent to the voicemail of 101 Razza [EMAIL PROTECTED] 10/10/2007 1:56 PM On 10/10/2007, Reggie Payne [EMAIL PROTECTED] wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg Are you suggesting that all of a call is recorded and if a certain key sequence is not entered during the call, the recording is completely discarded otherwise the complete call is saved. Or are you suggesting the call is only recorded from the point you enter a specific key sequence? Ray ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording on demand...
Reggie Payne wrote: The call is recorded after a key sequence has been pressed. Example: SIP/101 makes an outbound call to 5551212 5551212 starts to get rowdy SIP/101 enters *99 to start recording the call After the call is ended the recording is sent to the voicemail of 101 Except for the sending to voicemail bit, I have some scripts I put together at http://horanappraisals.com/asterisk/recordings/ that provide a simple web interface to asterisk's recordings directory. Depending on the version of asterisk installed, the parsing of the name of the monitor filename might be a little off, but it shouldn't be hard to straighten out. Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording on demand...
Reggie Payne wrote: The call is recorded after a key sequence has been pressed. Example: SIP/101 makes an outbound call to 5551212 5551212 starts to get rowdy SIP/101 enters *99 to start recording the call After the call is ended the recording is sent to the voicemail of 101 Use a script run regularly from cron to detect new recordings in the monitor directory, determine who the recipient should be, and mail them out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording on demand...
Awesome. Thanks all. I am still gonna work on some other possible logic. It would really be cool to have all of that functionality in Asterisk. Reg Mojo with Horan Company, LLC [EMAIL PROTECTED] 10/10/2007 3:24 PM Reggie Payne wrote: The call is recorded after a key sequence has been pressed. Example: SIP/101 makes an outbound call to 5551212 5551212 starts to get rowdy SIP/101 enters *99 to start recording the call After the call is ended the recording is sent to the voicemail of 101 Except for the sending to voicemail bit, I have some scripts I put together at http://horanappraisals.com/asterisk/recordings/ that provide a simple web interface to asterisk's recordings directory. Depending on the version of asterisk installed, the parsing of the name of the monitor filename might be a little off, but it shouldn't be hard to straighten out. Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording filename
I have figured out a way to include dialed number in recorded
voicefile in freepbx . You have to edit
/var/lib/asterisk/agi-bin/recordingcheck
add this lines after $agi=new AGI()
$temp= $agi-get_variable(DIAL_NUMBER);
$agi-verbose(Number to be dialled is -{$temp[data]});
After this you can use variable {$temp[data]} in outfile names ( set
few line below in same file ) . This is only required for freepbx .
On 30/11/06, Vicky [EMAIL PROTECTED] wrote:
No response at all :( . I did a temporary solution . I made cdr mysql to
store unique id into database from this wiki . So i now atleast have
uniquefield common in callfilename and sql records to tally .
Storing the Unique ID
Q: It would appear that the uniqueid field is not being populated in the
MySQL CDR DB. Is this an obsolete field or is a bug?
A: You need to define MYSQL_LOGUNIQUEID at compile time for it to use that
field.
You have two options in /usr/src/asterisk-addons:
1. Add CFLAGS+=-DMYSQL_LOGUNIQUEID to the Makefile.
2. Add a #define MYSQL_LOGUNIQUEID to the top of cdr_addon_mysql.c.
Finally perform the usual make clean, make, make install. Be sure to check
the Makefile for the presence of this flag after having done a CVS update!
You will most probably also want to index the uniqueid field in your cdr
table to improve performance.
On 30/11/06, Nick Hoffman [EMAIL PROTECTED] wrote:
On Wed November 29 2006 05:17, Vicky [EMAIL PROTECTED] wrote:
I am using asterisk along with freepbx . When recording is enabled for a
extension the call record file made in /var/spool/asterisk/monitor
contains information like OUT(extension
number)-(timestamp)-(uniqueid).wav . This can be a big
mess if there are
more than 1000-2000 files in that folder and very hard to locate a call
recording based on call time and extension number who dialled. I need to
put something like outgoing number dialled within call file name instead
of uniqueid .. After watching in console i opened up
/var/lib/asterisk/agi-bin/recordingcheck and saw that
it is setting
callfilename variable with extension number,time,unique id , etc. so i
edited and instead of $uniqueid i put $DIALEDPEERNUMBER ( saw in
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
) but
its just not giving dialed number and hence callfilename doesnt contain
outgoing number . Any suggestions how can i get outgoing call number in
recording file ?
Hi Vicky. Did you receive any responses to your email? I'd be interested
in
anything people suggested.
Cheers,
-- Nick
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