[Asterisk-Users] Record all phone calls
I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is what I have so far but isn't working. Can someone help me out? Thanks, [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten => _,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1001,1,Dial(SIP/one|20|tr) exten => 1001,2,VoiceMail,u1001 exten => 1001,102,VocieMail,b1001 exten => 2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001 exten => 1002,1,Dial(SIP/two|20|mtr) exten => 1002,2,VoiceMail,u1002 exten => 1002,102,VoiceMail,b1002 exten => 6001,1,Ringing exten => 6001,2,Wait(2) exten => 6001,3,VoicemailMain
RE: [Asterisk-Users] Record all phone calls
Here is what I have now. Where should the line " exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM})" go should it be under [sip]? Right now if I call sip to sip monitoring starts and the calls connect but I only get 44 byte files. If I call and iaxtel number monitoring starts but call never gets placed and again 44byte files with nothing in them. Thanks for the help. [iaxtel] exten => _1700XXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1888NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1877NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1866NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1800NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED]) [sip] include => iaxtel exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => s,1,Dial(SIP/one|20|tr) exten => 1001,1,Dial(SIP/one|20|tr) exten => 1001,2,VoiceMail,u1001 exten => 1001,102,VocieMail,b1001 exten => 2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001 exten => 1002,1,Dial(SIP/two|20|mtr) exten => 1002,2,VoiceMail,u1002 exten => 1002,102,VoiceMail,b1002 exten => 6001,1,Ringing exten => 6001,2,Wait(2) exten => 6001,3,VoicemailMain From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Mann Sent: January 10, 2004 12:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Record all phone calls - Original Message - From: Jimmy Riley To: '[EMAIL PROTECTED]' Sent: Saturday, January 10, 2004 10:01 AM Subject: [Asterisk-Users] Record all phone calls I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is what I have so far but isn't working. Can someone help me out? Thanks, [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten => _,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1001,1,Dial(SIP/one|20|tr) exten => 1001,2,VoiceMail,u1001 exten => 1001,102,VocieMail,b1001 exten => 2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001 exten => 1002,1,Dial(SIP/two|20|mtr) exten => 1002,2,VoiceMail,u1002 exten => 1002,102,VoiceMail,b1002 exten => 6001,1,Ringing exten => 6001,2,Wait(2) exten => 6001,3,VoicemailMain There are a few issues I can see with this but your two big problems are as follows. You never want to include a macro. include => macro-record-on So remove that line altogether. You show exten => _,1,macro(record-on,${EXTEN},${CALLERIDNUM}) the _ tells asterisk that you are going to want to match characters but then you dont tell it what you want to match. so exten => _.,1etc... See the . after _ this tells * to match the rest of the characters (digits) Those are your two big issues with not getting the recording to start. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Record calls where to put line?
Here is what I have now. Where should the line " exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM})" go should it be under [sip]? Right now if I call sip to sip monitoring starts and the calls connect but I only get 44 byte files. If I call and iaxtel number monitoring starts but call never gets placed and again 44byte files with nothing in them. Thanks for the help. [iaxtel] exten => _1700XXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1888NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1877NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1866NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => _1800NXX,1,Dial(IAX2/jmproductions:[EMAIL PROTECTED]/[EMAIL PROTECTED]) [sip] include => iaxtel exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => s,1,Dial(SIP/one|20|tr) exten => 1001,1,Dial(SIP/one|20|tr) exten => 1001,2,VoiceMail,u1001 exten => 1001,102,VocieMail,b1001 exten => 2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001 exten => 1002,1,Dial(SIP/two|20|mtr) exten => 1002,2,VoiceMail,u1002 exten => 1002,102,VoiceMail,b1002 exten => 6001,1,Ringing exten => 6001,2,Wait(2) exten => 6001,3,VoicemailMain From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Mann Sent: January 10, 2004 12:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Record all phone calls - Original Message - From: Jimmy Riley To: '[EMAIL PROTECTED]' Sent: Saturday, January 10, 2004 10:01 AM Subject: [Asterisk-Users] Record all phone calls I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is what I have so far but isn't working. Can someone help me out? Thanks, [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten => _,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1001,1,Dial(SIP/one|20|tr) exten => 1001,2,VoiceMail,u1001 exten => 1001,102,VocieMail,b1001 exten => 2001,1,Dial,IAX2/[EMAIL PROTECTED]/2001 exten => 1002,1,Dial(SIP/two|20|mtr) exten => 1002,2,VoiceMail,u1002 exten => 1002,102,VoiceMail,b1002 exten => 6001,1,Ringing exten => 6001,2,Wait(2) exten => 6001,3,VoicemailMain There are a few issues I can see with this but your two big problems are as follows. You never want to include a macro. include => macro-record-on So remove that line altogether. You show exten => _,1,macro(record-on,${EXTEN},${CALLERIDNUM}) the _ tells asterisk that you are going to want to match characters but then you dont tell it what you want to match. so exten => _.,1etc... See the . after _ this tells * to match the rest of the characters (digits) Those are your two big issues with not getting the recording to start. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how do i make this happen [macro-record-cleanup]
[macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [macro-record-cleanup] exten => s,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor) exten => s,2,GotoIf($[${CALLFILENAME} = ${FOO}]?6:3) exten => s,3,System(soxmix ${MONITORDIR}/${CALLFILENAME}-in.wav ${MONITORDIR}/${CALLFILENAME}-out.wav ${MONITORDIR}/${CALLFILENAME}.gsm) exten => s,4,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav ${MONITORDIR}/${CALLFILENAME}-out.wav) exten => s,5,NoOp [sip] exten => 1001,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1001,2,Dial(SIP/one|20|tr) Jimmy Riley Network Administrator VeriCore 985-626-1701 X1103 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] macro error "exited non-zero"
On this macro I keep getting exited non-zero on s,3, but s,3 is doing what it is suppose to do but the macro stops. Is there a way to make a macro ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4. Also if I just run this line from the command line I don't get an error. [EMAIL PROTECTED] monitor]# sox in.wav in-rev.wav reverse [EMAIL PROTECTED] monitor]# [macro-record-cleanup] exten => s,1,GotoIf($[${CALLFILENAME} = ${FOO}]?11:2) exten => s,2,SetVar(MONITORDIR=/var/spool/asterisk/monitor) exten => s,3,System(/usr/local/bin/sox ${MONITORDIR}/${CALLFILENAME}-in.wav ${MONITORDIR}/${CALLFILENAME}-in-rev.wav reverse) exten => s,4,System(/usr/local/bin/sox ${MONITORDIR}/${CALLFILENAME}-out.wav ${MONITORDIR}/${CALLFILENAME}-out-rev.wav reverse) exten => s,5,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav ${MONITORDIR}/${CALLFILENAME}-out.wav) exten => s,6,System(/usr/local/bin/soxmix ${MONITORDIR}/${CALLFILENAME}-in-rev.wav ${MONITORDIR}/${CALLFILENAME}-out-rev.wav ${MONITORDIR}/${CALLFILENAME}-rev.gsm) exten => s,7,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in-rev.gsm ${MONITORDIR}/${CALLFILENAME}-out-rev.gsm) exten => s,8,System(/usr/local/bin/sox ${MONITORDIR}/${CALLFILENAME}-rev.gsm ${MONITORDIR}/${CALLFILENAME}.gsm reverse) exten => s,9,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-rev.gsm) exten => s,10,System(sox ${MONITORDIR}/${CALLFILENAME}.gsm -g ${MONITORDIR}/${CALLFILENAME}.wav exten => s,11,NoOp == Spawn extension (sip, 18005551212, 2) exited non-zero on 'SIP/one-8e46' -- Executing Macro("SIP/one-8e46", "record-cleanup") in new stack -- Executing GotoIf("SIP/one-8e46", "0?11:2") in new stack -- Goto (macro-record-cleanup,s,2) -- Executing SetVar("SIP/one-8e46", "MONITORDIR=/var/spool/asterisk/monitor") in new stack -- Executing System("SIP/one-8e46", "/usr/local/bin/sox /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav reverse") in new stack WARNING[1209277232]: File app_system.c, Line 57 (system_exec): Unable to execute '/usr/local/bin/sox /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav reverse' == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on 'SIP/one-8e46' in macro 'record-cleanup' == Spawn extension (sip, s, 1) exited non-zero on 'SIP/one-8e46' Jimmy Riley Network Administrator VeriCore 985-626-1701 X1103 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] macro error "exited non-zero"
Jimmy Riley Network Administrator VeriCore 985-626-1701 X1103 -Original Message- From: Andres [mailto:[EMAIL PROTECTED] Sent: January 11, 2004 12:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] macro error "exited non-zero" On Sunday 11 January 2004 11:47, Jimmy Riley wrote: > On this macro I keep getting exited non-zero on s,3, but s,3 is doing > what it is suppose to do but the macro stops. Is there a way to make a > macro ignore errors and continue to s,4? I have the lattes ver of sox > 12.17.4. Are you using RedHat 9? Yes > > Also if I just run this line from the command line I don't get an error. > [EMAIL PROTECTED] monitor]# sox in.wav in-rev.wav reverse > [EMAIL PROTECTED] monitor]# > > > [macro-record-cleanup] > > exten => s,1,GotoIf($[${CALLFILENAME} = ${FOO}]?11:2) > exten => s,2,SetVar(MONITORDIR=/var/spool/asterisk/monitor) > > exten => s,3,System(/usr/local/bin/sox ${MONITORDIR}/${CALLFILENAME}-in.wav > ${MONITORDIR}/${CALLFILENAME}-in-rev.wav reverse) > exten => s,4,System(/usr/local/bin/sox > ${MONITORDIR}/${CALLFILENAME}-out.wav > ${MONITORDIR}/${CALLFILENAME}-out-rev.wav reverse) > > exten => s,5,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav > ${MONITORDIR}/${CALLFILENAME}-out.wav) > > exten => s,6,System(/usr/local/bin/soxmix > ${MONITORDIR}/${CALLFILENAME}-in-rev.wav > ${MONITORDIR}/${CALLFILENAME}-out-rev.wav > ${MONITORDIR}/${CALLFILENAME}-rev.gsm) > > exten => s,7,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in-rev.gsm > ${MONITORDIR}/${CALLFILENAME}-out-rev.gsm) > > exten => s,8,System(/usr/local/bin/sox > ${MONITORDIR}/${CALLFILENAME}-rev.gsm ${MONITORDIR}/${CALLFILENAME}.gsm > reverse) > > exten => s,9,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-rev.gsm) > > exten => s,10,System(sox ${MONITORDIR}/${CALLFILENAME}.gsm -g > ${MONITORDIR}/${CALLFILENAME}.wav > exten => s,11,NoOp > > > > == Spawn extension (sip, 18005551212, 2) exited non-zero on 'SIP/one-8e46' > -- Executing Macro("SIP/one-8e46", "record-cleanup") in new stack > -- Executing GotoIf("SIP/one-8e46", "0?11:2") in new stack > -- Goto (macro-record-cleanup,s,2) > -- Executing SetVar("SIP/one-8e46", > "MONITORDIR=/var/spool/asterisk/monitor") in new stack > -- Executing System("SIP/one-8e46", "/usr/local/bin/sox > /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav > /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav > reverse") in new stack > WARNING[1209277232]: File app_system.c, Line 57 (system_exec): Unable to > execute '/usr/local/bin/sox > /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav > /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav > reverse' > == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on > 'SIP/one-8e46' in macro 'record-cleanup' > == Spawn extension (sip, s, 1) exited non-zero on 'SIP/one-8e46' > > > Jimmy Riley > Network Administrator > VeriCore > 985-626-1701 X1103 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GUI client for windows for live monitoring/barge
I've seen a few but can't get them to work. I need one where I can drop a call into a conference without them knowing it to us it as a live monitor and barge function, anyone doing this are know of a gui client for windows I can use? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GUI client for windows for live monitoring/barge
I've seen a few but can't get them to work. I need one where I can drop a call into a conference without them knowing it to us it as a live monitor and barge function, anyone doing this are know of a gui client for windows I can use? Thanks, Jimmy Riley Network Administrator VeriCore 985-626-1701 X1103 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GUI client for windows for live monitoring/b arge
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: January 12, 2004 11:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] GUI client for windows for live monitoring/barge > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Jimmy Riley > Sent: Tuesday, 13 January 2004 13:02 > To: '[EMAIL PROTECTED]' > Subject: [Asterisk-Users] GUI client for windows for live > monitoring/barge > > I've seen a few but can't get them to work. I need one where > I can drop a > call into a conference without them knowing it to us it as a > live monitor > and barge function, anyone doing this are know of a gui > client for windows I > can use? > Thanks, This may be a wacky suggestion, might require more resources, but why don't you try setting up your dialplan so that all calls are in conference s with two members, then you can drop in any time you want... Cheers, Woody Thanks for the idea I'll look at doing that. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GUI client for windows for live monitoring/b arge
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: January 12, 2004 11:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] GUI client for windows for live monitoring/barge > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Jimmy Riley > Sent: Tuesday, 13 January 2004 13:02 > To: '[EMAIL PROTECTED]' > Subject: [Asterisk-Users] GUI client for windows for live > monitoring/barge > > I've seen a few but can't get them to work. I need one where > I can drop a > call into a conference without them knowing it to us it as a > live monitor > and barge function, anyone doing this are know of a gui > client for windows I > can use? > Thanks, This may be a wacky suggestion, might require more resources, but why don't you try setting up your dialplan so that all calls are in conference s with two members, then you can drop in any time you want... Cheers, Woody Any idea on how I would do that? I have meetme setup. But can't figure out how to make a person go in a conference room and dial a number out. Thanks exten => 1004,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1004,2,MeetMe,2004 exten => 1004,3,Dial(SIP/four|20|tr) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] hardware requirements - asterisk
What is your internet connection speed up and down? That could be your problem the traffic. Jimmy Riley Network Administrator VeriCore 985-626-1701 X1103 -Original Message- From: dkwok [mailto:[EMAIL PROTECTED] Sent: January 15, 2004 1:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] hardware requirements - asterisk In relation to voice degradation when having 2 or more connection to Asterisk. The comment on the network setup is quite possible. I am not too familiar with linux. How do I check whether the asterisk server's nic is running at full-duplex mode. Does Asterisk use the sound card on the box to do voice processing? I am running xlite on 2 pc and making calls through iax, FWD and back to my incoming call menu. Voice degradation happens. David Kwok ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe, conferencing questions
What would I use in my dial plan to make all calls made go into a conference room first then dial the number they called. I want to be able to do this so that someone could dial the conference room at anytime and listen into that persons call. I guess you could call it like live conference. Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting correct CDR info
I'd like to know how everyone else is going about getting correct CDR information for calls. Right now I notice that if a call come in and gets parked the CDR info doesn't how the correct info on who picked that call up, also when someone transfer a call there isn't a new record being made so the duration of the call shows up for who received the call and transferred it. I started playing around resetCDR and that does help in some cases. Just want to make sure I'm not just messing something. Thanks, Jimmy Riley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help - recording both sides of a conversati on
This is what I'm doing it gets you both sides of the phone call...small size...and playable on windows through a share. My notes: On redhat 9 I have to run the following command for asterisk to start LD_ASSUME_KERNEL=2.4.1 asterisk -gc [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) ;exten => s,3,ResetCDR(w) [macro-record-cleanup] exten => s,1,GotoIf($[${CALLFILENAME} = ${FOO}]?11:2) exten => s,2,SetVar(MONITORDIR=/var/spool/asterisk/monitor) exten => s,3,System(sox ${MONITORDIR}/${CALLFILENAME}-in.wav ${MONITORDIR}/${CALLFILENAME}-in-rev.wav reverse) exten => s,4,System(sox ${MONITORDIR}/${CALLFILENAME}-out.wav ${MONITORDIR}/${CALLFILENAME}-out-rev.wav reverse) exten => s,5,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav ${MONITORDIR}/${CALLFILENAME}-out.wav) exten => s,6,System(soxmix ${MONITORDIR}/${CALLFILENAME}-in-rev.wav ${MONITORDIR}/${CALLFILENAME}-out-rev.wav ${MONITORDIR}/${CALLFILENAME}-rev.wav) exten => s,7,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in-rev.wav ${MONITORDIR}/${CALLFILENAME}-out-rev.wav) exten => s,8,System(sox ${MONITORDIR}/${CALLFILENAME}-rev.wav ${MONITORDIR}/${CALLFILENAME}.wav reverse) exten => s,9,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-rev.wav) exten => s,10,System(sox ${MONITORDIR}/${CALLFILENAME}.wav -g ${MONITORDIR}/${CALLFILENAME}-done.wav) exten => s,11,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}.wav) exten => s,12,NoOp Jimmy Riley Network Administrator VeriCore 985-626-1701 X1103 -Original Message- From: Joelson S. Apon [mailto:[EMAIL PROTECTED] Sent: January 20, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] help - recording both sides of a conversation Hello Sirs.. I'm setting up a call-recording with my asterisk here and I do follow program which was post in this mailing list last Jan. 4 (program is also shown below), and I'm very much thankful for that.. However, I do have some errors, here is my output..Hope that someone could lighten me up for this..Thank you very much for the help.. Regards Joel *CLI> -- Starting simple switch on 'Zap/49-1' -- Executing Answer("Zap/49-1", "") in new stack -- Executing Macro("Zap/49-1", "record-enable") in new stack -- Executing AGI("Zap/49-1", "set-timestamp.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/set-timestamp.agi -- AGI Script set-timestamp.agi completed, returning 0 -- Executing Dial("Zap/49-1", "Zap/51|15") in new stack -- Called 51 -- Zap/51-1 is ringing -- Zap/51-1 answered Zap/49-1 -- Attempting native bridge of Zap/49-1 and Zap/51-1 -- Hungup 'Zap/51-1' == Spawn extension (test3, 2103, 3) exited non-zero on 'Zap/49-1' -- Executing Macro("Zap/49-1", "record-cleanup") in new stack -- Executing SetVar("Zap/49-1", "MONITORDIR=/var/spool/asterisk/conversations/") in new stack -- Executing GotoIf("Zap/49-1", " = ?6:3") in new stack -- Goto (macro-record-cleanup,s,3) Jan 20 13:43:37 WARNING[1256444864]: pbx.c:1173 pbx_extension_helper: No application 'System(/usr/scripts/mix_monitor_files.pl ${MONITORDIR} ${CALLFILENAME}-in.wav' for extension (macro-record-cleanup, s, 3) == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on 'Zap/49-1' in macro 'record-cleanup' == Spawn extension (test3, h, 1) exited non-zero on 'Zap/49-1' -- Hungup 'Zap/49-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of zoa Sent: Tuesday, January 06, 2004 1:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] help - recording both sides of a conversation You also don't need such a complicated perl script, just muxing them without cutting them is enough. (Timing was fixed) zoa. At 14:41 4/01/2004 -0600, you wrote: >you nolonger need set-timestamp.agi as we have ${TIMESTAMP} in that format >by default now. > >bkw > >On Sun, 4 Jan 2004, John Baker wrote: > > > Iain - > > > > First off, all of this is heavily borrowed from others. For those who see > > their code embedded here, I thank you and give you full credit. > > > > Here's how I do it. It's a bit convoluted, but I didn't want to record > > everything. So, if a call comes in and I want to record it, I send it > here: > > > > [ext-surrept] > > exten => _57XXX,1,Answer > > exten => _57XXX,2,Macro(record-enable) > > exten => _57XXX,3,BackGround(for-quality-purposes) > > exten => _57XXX,4,BackGround(this-call-may-be) > > exten => _57XXX,5,BackGround(recorded) > > exten =>
RE: [Asterisk-Users] Getting correct CDR info
-Original Message- From: Stephen J. Wilcox [mailto:[EMAIL PROTECTED] Sent: January 20, 2004 6:02 PM To: '[EMAIL PROTECTED]' Subject: Re: [Asterisk-Users] Getting correct CDR info Hi Jimmy, I have CDR issues also and was thinking of seeing if I could write a patch to fix, my particular problem which I think you include below alebit subtley is that there are numerous circumstances where you would expect multiple CDRs and you currently get one minus bits of vital detail eg Scenario1: IVR service A call comes in, caller gets a menu, hits an option and gets connected to an external pstn number. You need a CDR to cover the inbound call, showing the total time, the original CLI and the number that the call came in on (DNI) as the dst You also need a CDR to show the call made by the system showing only the time of the outbound leg and the external number as the dst. The reason for this is that you want to produce stats and possibly charges for the service eg this may be a toll free service that your customer pays the inbound call charges for, you may also wish to charge for the outbound call to the customer eg this is some kind of dialthro Scenario2: end user voip pstn service Ok simple, your providing pstn service to biz/home users giving each a phone as tho it were pstn.. lets say you have a call divert facility, one of your customers dials another customer (hence generating a charge + CDR). The other customer has setup a divert to their mobile phone (hence generating a charge + another CDR to the that customer). Theres more situations, especially when you start doing complex things involving multiple customers and services on your platform. In addition to being able to generate multiple CDRs there needs to be some additional controls to be able to alter the cdr fields or add more user fields.. Steve On Mon, 19 Jan 2004, Jimmy Riley wrote: > I'd like to know how everyone else is going about getting correct CDR > information for calls. Right now I notice that if a call come in and gets > parked the CDR info doesn't how the correct info on who picked that call up, > also when someone transfer a call there isn't a new record being made so the > duration of the call shows up for who received the call and transferred it. > I started playing around resetCDR and that does help in some cases. Just > want to make sure I'm not just messing something. > Thanks, > > > > Jimmy Riley > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have been testing using resetcdr. Is there a better way to do this? This is the extension config I'm using: exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,Background(vericore) exten => s,4,Wait,7 exten => s,5,Background(vericore) exten => s,6,DigitTimeout,5 exten => s,7,ResponseTimeout,10 exten => s,8,Goto,1006|1 exten => 1050,1,Goto,s|1 exten => 1001,1,ResetCDR(w) exten => 1001,2,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1001,3,Dial(SIP/one|20|tr) exten => 1001,4,Dial(SIP/two|20|tr) exten => 1001,5,VoiceMail,u1001 exten => 1001,102,Dial(SIP/two|20|tr) exten => 1001,103,VoiceMail,u1001 exten => 1002,1,ResetCDR(w) exten => 1002,2,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1002,3,Dial(SIP/two|20|tr) exten => 1002,4,VoiceMail,u1001 exten => 1002,102,VoiceMail,b1001 exten => 1003,1,ResetCDR(w) exten => 1003,2,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1003,3,Dial(SIP/three|20|tr) exten => 1003,4,Dial(SIP/four|20|tr) exten => 1003,5,VoiceMail,u1002 exten => 1003,102,Dial(SIP/four|20|tr) exten => 1003,103,VoiceMail,u1002 exten => 1004,1,ResetCDR(w) exten => 1004,2,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1004,3,Dial(SIP/four|20|tr) exten => 1004,4,VoiceMail,b1002 exten => 1004,102,VoiceMail,u1002 exten => 1005,1,ResetCDR(w) exten => 1005,2,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1005,3,Dial(SIP/five|20|tr) exten => 1005,4,Dial(SIP/six|20|tr) exten => 1005,5,VoiceMail,u1003 exten => 1005,102,Dial(SIP/six|20|tr) exten => 1005,103,VoiceMail,u1003 exten => 1006,1,ResetCDR(w) exten => 1006,2,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1006,3,Dial(SIP/six|20|tr) exten => 1006,4,VoiceMail,u1003 exten => 1006,102,VoiceMail,b1003 exten => #,1,Ringing exten => #,2,Wait(2) exten => #,3,VoicemailMain exten =>