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 - 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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[asterisk-users] Call recording - methodology
Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I 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
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 ${UNIQUEID:0:10}.wav if you are recording multiple calls because
otherwise they'll be constantly saved to FILENAME.wav and you'll lose all the
previous calls.)
(please note, /var/lib/asterisk/sounds/rec1 is a beep tone so that the operator
knows that he's successfully started the recording.)
Then to recording every call, we add this before the DIAL(SIP/extension)
command in extensions.conf:-
exten = _9.,14,Macro(mixmon,nobeep)
If you don't want to record every call, you can give the operator the option of
press *1. We did this by adding the following to features.conf:-
MixMonApp = *1,self/both,Macro,mixmon
Hope that helps.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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asterisk-users mailing list
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Re: [asterisk-users] Call recording - methodology
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, April 06, 2011 6:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call recording - methodology
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I
have to modify it to make it easier to use, I do not mind.
Does anyone know of any opensource or otherwise solutions out there that
I can try out?
We give our clients to option of either recording all calls, or allowing the
operator to press *1 during a call to start recording manually.
Using Asterisk 1.4, this is what we do:-
We created a Macro in extensions.conf like this:-
[macro-mixmon]
exten = s,1,GotoIf($[${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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