[Asterisk-Users] in and out recorded audio mixing in queues

[Faris Raouf]

Way back I was still on Asterisk 1.0.7, I configured my systems to mix 
the incoming and outgoing audio call recordings into one file per call 
for both normal calls and queued calls using:

exten => 
_9.,1,Monitor(wav49,${TIMESTAMP}-${CALLERIDNUM}-to-${EXTEN:1}-${UNIQUEID},m) 
; m option merges audio into one file and deletes the parts

in extensions.conf

and

monitor-join = yes

in queues.conf

As far as I remember this worked perfectly.

But I was only on 1.0.7 for a very short while, and quickly updated. One 
system is on 1.0.9 Stable and the others are on the very latest SVN HEAD.

I recently decided to have a bit of a spring clean on the audio files, 
and to my horror found that only the very first few files on all 
machines were mixed (I presume those were the ones when I was still on 
1.0.7) while the rest were all still there as separate -in and -out 
files. This is despite the console showing that soxmix was being called 
to join the two files and remove the individual parts each time a call 
was made or received, and with no errors. Soxmix is installed, and does 
work - I can copy and paste the command from the console output at the 
command line and the single audio file is then properly created and the 
individual in and out parts deleted.

I've just tried changing to using MixMonitor in extensions.conf on the 
1.2x machines and this works perfectly for normal outgoing or incoming 
calls that don't involve queues.

But this obviously doesn't solve my queues mixing problem on the 1.2 
machines, nor any of the mixing problems on the 1.0.9 machine.

Has anyone else come across this issue? Any pointers please?

Thanks,

Faris.