[Asterisk-Users] 2wav2mp3, monitor, mixmonitor, mpg123, queues

[g.delduca at webster.it]

Hello!
I'm using Asterisk for our office telephony, but we have some problems
that still we can't resolve about it. Here they are:

1) merge in/out call recording files

I also tried to use a script I found on the internet, called 2wav2mp3
In extensions.conf I added the following lines

; script to be executed when monitoring has been finished
MONITOR_EXEC=/usr/local/bin/2wav2mp3

exten => 102,1,SetVar(CALLFILENAME=${TIMESTAMP}-${EXTEN}-${CALLERID})
exten => 102,2,Monitor(wav,${CALLFILENAME},m)
;exten => 102,2,MixMonitor(${CALLFILENAME}.gsm)
;exten => 102,2,MixMonitor(test.wav,W(-3))
exten => 102,3,Ringing
exten => 102,4,Dial(Sip/giuseppedd,20,rtwW)

...but I always get two separate files.

As you can see I also tried the MixMonitor application but the resulting 
files
contain one channel that is clearly audible and the other seems to be noise.

2) an alternative to mpg123 becouse it generates a lot of errors like this:

Feb  3 19:50:08 WARNING[9568]: res_musiconhold.c:488 monmp3thread: 
Unable to spawn mp3player
Feb  3 19:58:28 NOTICE[9568]: res_musiconhold.c:507 monmp3thread: 
Request to schedule in the past?!?!
Feb  3 19:58:28 WARNING[9568]: res_musiconhold.c:421 spawn_mp3: Found no 
files in '/usr/share/asterisk/mohagents'


3) how to play different files to an agent before he picks up a call 
depending on which queue the call comes from

[qlu500]
musiconhold = qlu500
announce = vm-from-phonenumber  ; <<<---- here is the problem
context = qlu500out
wrapuptime=15
announce-frequency = 60
...


Comments or suggestions are greatly appreciated.

Thanks a lot.

Giuseppe