[asterisk-users] trouble recording calls

[ahester]

Hi all,

I am having the following trouble with recording calls:
When calls come into the support line did number, the call starts to
record on the first queue, but appears to hang up when the call actually
connects to the engineer (ie I see "got hangup request" on the cli and
then mixmonitor ends.)  I am guessing this has to do with the announce
file that is played to the engineer before the call is connected.  It
seems that if the call rolls to the next queue because of timeout,
asterisk doesn't even try to record it. (I don't see any mixmonitor on
the cli for the next queue). 

I would appreciate any help with this.  I have to have all calls
recorded and I have to do announcements so that the callee knows how to
answer the phone.

Thanks,
Andy


The configs are as below:

>From extensions.conf:

#after various menu stuff, send to support
exten => 214xxxxxxx,13,SetGlobalVar(ORIGIN=support)
exten => 214xxxxxxx,14,Queue(support1|tr|||10)
exten => 214xxxxxxx,15,Queue(support2|tr|||)

#dial command for sip extensions that are in the queues
exten =>
_72XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
exten => _72XXX,2,Dial(SIP/${EXTEN})
exten =>
_73XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
exten => _73XXX,2,Dial(SIP/${EXTEN})


queues from queues.conf:

[support1]
; Support call queue
announce = 16
strategy = rrmemory
timeout = 15
retry = none
wrapuptime=15
announce-frequency = 0
joinempty = no
leavewhenempty = yes
member => Agent/2008
member => Agent/2009
member => Agent/2014
member => Agent/2015
member => Agent/2017
member => Agent/2018
member => Agent/2019
member => Agent/3520
member => Agent/3521
member => Agent/3522
member => Agent/3524
member => Agent/3529

[support2]
; Support2 call queue
announce = 16
strategy = ringall
announce-frequency = 0
; Added below for testing because the second queue was not even trying
to record
; according to the asterisk console (still doesn't)
Set(MONITOR_FILENAME=support/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
monitor-format = wav
monitor-join = yes
joinempty = yes
member => SIP/72008
member => SIP/72009


 

-- 
Andy Hester
Network Engineer
Architel