[Asterisk-Users] help - recording both sides of a conversation
Brian West
brian at bkw.org
Sun Jan 4 13:41:13 MST 2004
you nolonger need set-timestamp.agi as we have ${TIMESTAMP} in that format
by default now.
bkw
On Sun, 4 Jan 2004, John Baker wrote:
> Iain -
>
> First off, all of this is heavily borrowed from others. For those who see
> their code embedded here, I thank you and give you full credit.
>
> Here's how I do it. It's a bit convoluted, but I didn't want to record
> everything. So, if a call comes in and I want to record it, I send it here:
>
> [ext-surrept]
> exten => _57XXX,1,Answer
> exten => _57XXX,2,Macro(record-enable)
> exten => _57XXX,3,BackGround(for-quality-purposes)
> exten => _57XXX,4,BackGround(this-call-may-be)
> exten => _57XXX,5,BackGround(recorded)
> exten => _57XXX,6,Dial(SIP/${EXTEN:1},120,tm)
> exten => _57XXX,7,Macro(rg-inbound,10,tr)
> exten => _57XXX,8,Goto(aa-nooneavail,s,1)
>
> By transferring a call to 5 + the extension I'm at, I enable the call
> recording, let the caller know he might be recorded and then send the call
> right back to myself.
>
> Here's the Macro:
>
> [macro-record-enable]
> exten => s,1,AGI(set-timestamp.agi)
> exten => s,2,SetVar(CALLFILENAME=${timestamp}-${CALLERIDNUM}-${MACRO_EXTEN})
> exten => s,3,Monitor(wav,${CALLFILENAME})
>
> It starts the recording and calls set-timestamp.agi
>
> Here's the agi file:
>
> #!/bin/sh
> longtime=`date +%Y%m%d-%H%M%S`
> echo SET VARIABLE timestamp $longtime
>
> It sets a timestamp, which if you scour the asterisk list, you'll see that
> it is necessary for mixing the in and out audio later.
>
> I have one hangup extension set for my internal phones; it looks like this:
>
> exten => h,1,Macro(record-cleanup)
>
> And the record-cleanup macro looks like this:
>
> [macro-record-cleanup]
> exten => s,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor)
> exten => s,2,GotoIf($[${CALLFILENAME} = ${FOO}]?6:3)
> exten => s,3,System(/usr/scripts/mix_monitor_files.pl ${MONITORDIR}
> ${CALLFILENAME}-in.wav ${CALLFILENAME}-out.wav ${CALLFILENAME}.wav)
> exten => s,6,NoOp
>
> Don't forget to make the /var/spool/asterisk/monitor directory!
>
> Finally, mix_monitor_files.pl does the mixing job and combines the in and
> out files:
>
> #!/usr/bin/perl
>
> $monitordir = shift;
> $infile = shift;
> $outfile = shift;
> $finishfile = shift;
>
> chdir($monitordir);
>
>
> $infile_output = `sox $infile -e stat 2>&1`;
> $outfile_output = `sox $outfile -e stat 2>&1`;
>
> $infile_output =~ /Samples read:\s+(\d+)/;
> $infile_samples = $1;
>
> $outfile_output =~ /Samples read:\s+(\d+)/;
> $outfile_samples = $1;
>
>
> if($outfile_samples > $infile_samples)
> {
> $diff_samples = $outfile_samples - $infile_samples;
> system("sox -v 3 $outfile temp${outfile} trim ${diff_samples}s");
> system("wmix $infile temp${outfile} > $finishfile");
> system("rm -f $infile temp${outfile} $outfile");
> }
> elsif($infile_samples > $outfile_samples)
> {
> $diff_samples = $infile_samples - $outfile_samples;
> system("sox -v 3 $infile temp${infile} trim ${diff_samples}s");
> system("wmix temp${infile} $outfile > $finishfile");
> system("rm -f temp${infile} $outfile $infile");
> }
> else
> {
> system("wmix $infile $outfile > $finishfile");
> system("rm -f $infile $outfile");
> }
>
>
> You'll need wmix from http://tph.tuwien.ac.at/~oemer/wavetools.html and
> sox, which was already on my system and is pretty standard.
>
> The only problem I've found is that my in channel is a bit low, with respect
> to volume. It's probably a sox issue, but I haven't had time to mess with
> the settings yet. It's only an annoyance; you can definitely hear both
> sides of the conversation.
>
> John
>
> P.S. I record my outbound calls by prefixing my outbound calls with a 5,
> which similiarly call record-enable. In that case, the other party doesn't
> know they're being recorded. IANAL. Check your state laws first! In some
> states both parties must know about calls being recorded. In mine, TX, only
> the calling party must know, but it must be first person. For this reason,
> I do not let asterisk record everything, because my employees must
> themselves determine what they're going to record.
>
>
> ----- Original Message -----
> From: "Iain Stevenson" <iain at iainstevenson.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Sunday, January 04, 2004 12:51 PM
> Subject: Re: [Asterisk-Users] help - recording both sides of a conversation
>
>
> >
> > * always records both sides of the conversation - but stores them in
> > separate files in
> > /var/spool/asterisk/monitor/. You need to combine the "in" and "out"
> parts
> > using soxmix.
> >
> > Iain
> >
> >
> >
> > --On Sunday, January 4, 2004 9:59 am -0800 Paul Mahler
> > <pmahler at signate.com> wrote:
> >
> > > Does some kind Asterisk soul have an example from extensions.conf that
> > > shows how to record both sides of a conversation?
> > >
> > > Thanks!
> > >
> > >
> > > Paul Mahler
> > > mail:pmahler at signate.com
> > > phone: 650.207.9855
> > > fax: 877.408.0105
> > >
> > > -----Original Message-----
> > > From: asterisk-users-admin at lists.digium.com
> > > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Philipp von
> > > Klitzing
> > > Sent: Sunday, January 04, 2004 9:23 AM
> > > To: asterisk-users at lists.digium.com
> > > Subject: Re: [Asterisk-Users] CAPI, transfering thru a 2nd PBX - keep
> > > original CallerID
> > >
> > > Hi!
> > >
> > >> I want to have Asterisk as my gateway to the outside world and use
> > >> another PBX to connect my existing phones.
> > >>
> > >> exten => ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN}
> > >>
> > >> How do I transfer the caller Id information initially coming in?
> > >
> > > I have strong doubts that this can be done at all. One way would be to
> > > set your ${MSN2ndPBX} to ${CALLERIDNUM}, but that would require that
> > > capi.conf has that CALLERIDNUM listed as one of the valid outgoing MSNs.
> > > Since you won't know in advance who'll call that'll be a problem - also
> I
> > > don't think you can reconfigure capi.conf in the midst of processing a
> > > call...
> > >
> > > Besides: I suppose your ISDN PBX (which brand exactly?) supports CLIP
> (or
> > > comes with an internal S0 bus) and you have an analog CLIP phone (or
> ISDN
> > > phone) connected?
> > >
> > > Workaround: See my last posting and other very recent discussions
> > > concerning a simple tool that shows the current caller ID and name on
> > > your PC using either Flash, HTML or Java. Or use astman/ gastman.
> > > As of now I am storing the caller data through AGI in mySQL and display
> > > that on a web page that the user needs to re-load manually when desired.
> > >
> > > Cheers, Philipp
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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