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
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
On Wed, Apr 6, 2011 at 5:54 AM, Silver Thorne <szilverthorne at gmail.com>wrote:> Hello Everyone; > > I am looking for a solution to record calls that come into our Asterisk > server. I am hoping for something that is easy to use - however, if I have > to modify it to make it easier to use, I do not mind. > > Does anyone 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 "MixMonitor"....Asterisk has built in call recording -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110406/48f1a51c/attachment.htm>
> 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> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110406/21b6762b/attachment.htm>
Hey Dan et al; 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/6000-0000002e [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF begin passthrough '*' on SIP/6000-0000002e [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end '*' received on SIP/6000-0000002e, duration 186 ms [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end accepted with begin '*' on SIP/6000-0000002e [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end passthrough '*' on SIP/6000-0000002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF begin '1' received on SIP/6000-0000002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF begin passthrough '1' on SIP/6000-0000002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end '1' received on SIP/6000-0000002e, duration 193 ms [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end accepted with begin '1' on SIP/6000-0000002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end passthrough '1' on SIP/6000-0000002e [Apr 10 12:05:16] DTMF[15005] channel.c: DTMF begin '*' received on SIP/6000-00000030 [Apr 10 12:05:16] DTMF[15005] channel.c: DTMF begin passthrough '*' on SIP/6000-00000030 [Apr 10 12:05:16] DTMF[15005] channel.c: DTMF end '*' received on SIP/6000-00000030, duration 185 ms [Apr 10 12:05:16] DTMF[15005] channel.c: DTMF end accepted with begin '*' on SIP/6000-00000030 [Apr 10 12:05:16] DTMF[15005] channel.c: DTMF end passthrough '*' on SIP/6000-00000030 [Apr 10 12:05:17] DTMF[15005] channel.c: DTMF begin '1' received on SIP/6000-00000030 [Apr 10 12:05:17] DTMF[15005] channel.c: DTMF begin passthrough '1' on SIP/6000-00000030 [Apr 10 12:05:17] DTMF[15005] channel.c: DTMF end '1' received on SIP/6000-00000030, duration 184 ms [Apr 10 12:05:17] DTMF[15005] channel.c: DTMF end accepted with begin '1' on SIP/6000-00000030 [Apr 10 12:05:17] DTMF[15005] channel.c: DTMF end passthrough '1' on SIP/6000-00000030 I assume that 185 ms is long enough for the application? How am I transmitting the tones? Simple - an Ekiga Softphone Useragent : Ekiga/3.2.7 Reg. Contact : sip:6000 at 10.0.1.5 Any more words of wisdom? I am still missing some minor detail - I must be. Glen On 4/10/2011 10:28, Dan Journo wrote:> > > 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 Systems <http://www.keshercommunications.com/> | Hosted > PBX <http://www.keshercommunications.com/hostedpbx.html> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided byhttp://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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110410/21cc9306/attachment.htm>