[Asterisk-Users] All calls in queue go to agent that is down??
Chuck Bunn
chuck.bunn at networkdoc.com
Mon Mar 13 13:42:00 MST 2006
Hi,
When an agent is logged in and his phone goes off the network (using an
SJPhone on a portable PC) and the user had forgotten to log out of the
queue all calls go to this agent that is no longer connected to the
system. I have tried training and retraining but I need some way to fix
this. This just seems like really odd behavior but I realize that SIP
has no way of telling if a connection is alive or not. I would think
that the unregistering (ie disconnection of the SIP device) of the SIP
device would be enough to fix this (why would asterisk send all calls to
a sip connection that was unregistered???) Perhaps I need to make the
SIP registration time shorter? I am using the AgentCallBackLogin command
in my configs. Here is a copy of some the files involved:
queues.conf ************************
[general]
[default]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
leavewhenempty=strict
context=mainmenu
member => Agent/3000
member => Agent/3001
member => Agent/3002
member => Agent/3003
member => Agent/3004
member => Agent/3005
member => Agent/3006
member => Agent/3007
member => Agent/3008
member => Agent/3009
member => Agent/3010
member => Agent/3011
*************************
agents.conf ***********************
[agents]
wrapuptime=0
musiconhold => default
updatecdr=yes
recordagentcalls=no
;Operator - Ageless
group=1
agent => 3000,3000,xxx
agent => 3001,3001,xxx
agent => 3002,3002,xxx...
************************************
extensions-home.incl included in extensions.conf ************
[default]
;Operator queue, Operator Console, and Receptionist Phone
exten => s,1,Answer()
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout(5)
exten => s,4,ResponseTimeout(30)
exten => s,5,GotoIfTime(8:00-21:00|*|*|*?default,s,7)
exten => s,6,Goto(mainmenu,s,1)
exten => s,7,Queue(extensions-home|tn|||25)
exten => s,8,Goto(mainmenu,s,1)
include => mainmenu
;Ageless
exten => _400,1,Voicemail(u400 at default&413 at default)
exten => _405,1,Voicemail(u405 at default&411 at default)
exten => _41[0-3],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten => _499,1,Macro(novmail,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten => _500,1,Voicemail(u500 at default&510 at default)
exten => _51[0],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten => _590,1,Macro(novmail,${EXTEN},ZAP/3)
;Chicken
;exten => _60[0],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Resedential
;exten => _70[0-3],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Voicemail Main
exten => 800,1,Answer
exten => 800,2,VoicemailMain(@default)
;Agent Login
exten => 801,1,AgentCallbackLogin(||@default)
;Recording Interface
exten => 820,1,Goto(phrase,s,1)
;Voice Conferencing
exten => _85X,1,Answer
exten => _85X,2,MeetMe(${EXTEN})
;Music on Hold
exten => 870,1,Answer
exten => 870,2,SetMusicOnHold(default)
exten => 870,3,WaitMusicOnHold(420)
exten => 870,4,Hangup
*********************
Thanks
More information about the asterisk-users
mailing list