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

Fri Feb 10 17:34:35 MST 2006

Hi,

For Question 3; we did this:

Queues.conf

[RLM_Admin]

music=default

strategy=rrmemory

timeout=3000

retry=1

wrapuptime=0

maxlen = 0

announce-frequency = 0

announce-holdtime = no

member => Local/203 at queue_admin_ext

member => Local/205 at queue_admin_ext

member => Local/208 at queue_admin_ext

member => Local/212 at queue_admin_ext

member => Local/501 at queue_admin_ext

member => Local/505 at queue_admin_ext

member => Local/511 at queue_admin_ext

Extensions.conf

[queue_admin_ext]

exten =>
_2XX,1,Dial(ZAP/g4/1111${EXTEN},20,mA(lm_features/Queues/pls_say_thankyou_ho
lding))

exten =>
_3XX,1,Dial(SIP/${EXTEN},20,mA(lm_features/Queues/pls_say_thankyou_holding))

exten =>
_7XX,1,Dial(ZAP/g4/1111${EXTEN},20,mA(lm_features/Queues/pls_say_thankyou_ho
lding))

So when the agent gets a call it plays
lm_features/Queues/pls_say_thankyou_holding file then puts the call through.

Thanks

James

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Darlon
Sent: Friday, 10 February 2006 9:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 2wav2mp3, monitor, mixmonitor, mpg123, queues

Answer for the question number 1:

Use it:

exten=XXXX,1,Macro(ramais-gravados,SIP/${EXTEN})

[macro-ramais-gravados]
exten=s,1,SetVar(CALLFILENAME=i${CALLERIDNUM}-${TIMESTAMP})
exten=s,2,Monitor(wav,${CALLFILENAME},m)
exten=s,3,Dial(${ARG1},20,Ttr) 
exten=s,4,Hangup

This script was changed

2wav2mp3 

#!/bin/sh
# create stereo mp3 out of two mono wav-files
# source files will be deleted
#
# 2005 05 23 dietmar zlabinger http://www.zlabinger.at/asterisk
#
# usage: 2wav2mp3 <wave1> <wave2> <mp3>
# designed for Asterisk Monitor(file,format,option) where option is "e" and
# the variable 
# MONITOR_EXEC/usr/bin/2wav2mp3

# location of SOX and SOXMIX
# (set according to your system settings, eg. /usr/bin)
SOX=/usr/bin/sox
SOXMIX=/usr/bin/soxmix
#lame is only required when sox does not support liblame
LAME=/usr/bin/lame

# command line variables
LEFT="$1"
RIGHT="$2"
OUT="$3"

#test if input files exist
test ! -r $LEFT && exit
test ! -r $RIGHT && exit

# convert mono to stereo, adjust balance to -1/1
# left channel
$SOX -c 1 $LEFT $LEFT-tmp.wav pan -1
# right channel
$SOX -c 1 $RIGHT $RIGHT-tmp.wav pan 1

# combine and compress
# this requires sox to be built with mp3-support.
# To see if there  is  support  for  Mp3  run sox -h and 
# look for it under the list of supported file formats as "mp3".
# $SOXMIX -v 1 $LEFT-tmp.wav -v 1 $RIGHT-tmp.wav -v 1 $OUT.mp3

# in case and old version of sox is used, the lame-encoding
# can be done afterwards
$SOXMIX -v 0.5 $LEFT-tmp.wav $RIGHT-tmp.wav $OUT

echo $OUT > final.dat
FINAL=`cat final.dat | sed 's/wav/mp3/g'`

$LAME --silent -V7 -B24 --tt $OUT --add-id3v2 $OUT $FINAL

#remove temporary files
test -w $LEFT-tmp.wav && rm $LEFT-tmp.wav
test -w $RIGHT-tmp.wav && rm $RIGHT-tmp.wav
test -w $OUT && rm $OUT

#remove input files if successfull
#test -r $OUT.mp3 && rm $LEFT $RIGHT
test -r $FINAL && rm $LEFT $RIGHT
rm -f final.dat

Darlon Ferreira Bortolini
Rede/Desenvolvimento
Betha Sistemas
Fone (48) 3431-0750/Ramal 1000

----- Original Message ----- 

From: g.delduca at webster.it 

To: asterisk-users at lists.digium.com 

Sent: Friday, February 10, 2006 7:13 AM

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

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

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