[asterisk-users] Full queue issues

[Vinícius Fontes]

Oops, seems like I didn't realized something: the queue size can't be zero. I solved the problem by setting maxlen=1 and defining a timeout on the Queue() app. That way when all the agents are busy, the call gets diverted after [TIMEOUT] seconds, which is ok to me.


Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.

----- "Vinícius Fontes" <vinicius at canall.com.br> escreveu:

> Hello everyone.
> 
> I got a little problem in here: I want to set up a queue so that if
> anything of these happens:
> 
> a) No agents logged in
> b) All agents busy
> 
> Then the user gets diverted somewhere. I used this (for testing
> purposes only, of course):
> 
> exten => 7080,1,Answer()
> exten => 7080,n,Queue(teste)
> exten => 7080,n,Goto(${QUEUESTATUS})
> exten => 7080,n(ERROR),NoOp(${QUEUESTATUS})
> exten => 7080,n,Hangup()
> exten => 7080,n(LEAVEEMPTY),Goto(ERROR)
> exten => 7080,n(TIMEOUT),Goto(ERROR)
> exten => 7080,n(JOINUNAVAIL),Goto(ERROR)
> exten => 7080,n(LEAVEUNAVAIL),Goto(ERROR)
> exten => 7080,n(JOINEMPTY),Goto(ERROR)
> exten => 7080,n(TIMEOUT),Goto(ERROR)
> 
> exten => *210,1,AddQueueMember(teste,SIP/${CALLERID(num)})
> exten => *210,n,UserEvent(RefreshQueue)
> exten => *210,n,Playback(agent-loginok)
> 
> exten => *220,1,RemoveQueueMember(teste,SIP/${CALLERID(num)})
> exten => *220,n,UserEvent(RefreshQueue)
> exten => *220,n,Playback(agent-loggedoff)
> 
> 
> 
> In queues.conf:
> 
> [teste]
> strategy=roundrobin
> music=default
> timeout=10
> retry=0
> maxlen=1
> ringinuse=no
> leavewhenempty=strict
> joinempty=strict
> 
> 
> Then I have those scenarios:
> 
> a) There is no agents logged in, a call tries to enter the queue, the
> ${QUEUESTATUS} variable is set to LEAVEEMPTY and the call is
> disconnected. Everything fine in here.
> 
> b) There is only one agent logged in, he's in a call (InUse), the call
> enters the queue and stays there. I would like the call NOT to enter
> the queue and the ${QUEUESTATUS} variable to be set to something
> different.
> 
> Am I missing something or it's just not possible? I'm using SIP phones
> for the agents and Asterisk 1.4.15.
> 
> 
> Att
> Vinícius Fontes
> Desenvolvimento
> Canall Tecnologia em Comunicações Ltda.
> 
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