Sriram escribió:
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100
& 101 ) in a queue..When a caller arrives in queue , it lands on
first 100 , 100 then does a blind transfer to 101 .. so that the
caller can converse with 101 .. strangely enough the queue_log shows :
1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123
1253814093|1253814090.12|55365|SIP/100|CONNECT|3|1253814090.13|2
1253814104|1253814090.12|55365|SIP/100|TRANSFER|101|from-internal|3|11|1
Hi,
The queue_log content you are showing is the expected behavior. Once the
call is transfered, is out of the domain of the Queue() application (it
ends handling of the call and passes the control to the context and
extension), so you get no more queue_log events. That is, the TRANSFER
event counts as the last event on that queue for that call. The only
chance to have queue_log events for the transferred call is to transfer
to another queue, for example:
;Transfer context with queue
[from-internal]
exten => _X.,1,Answer
exten => _X.,2,*Queue*(transfer-queue,tT,,,90)
That way you will see the transferred call as an incoming call to
transfer-queue. Now how to make the queue ring the desired extension is
another issue. If using queues to transfer to any other available (not a
specific one) agent suits your needs, it works like a charm. Obviously
if someone transfers to the queue with no available members, it would be
queued again in the transfer-queue until someone is ready to take the
call. You can use attended transfer and blind transfer, it works the
same way.
The third leg of the call that is the CALLERCOMPLETED part (Caller's
talk time with 101) is not at all reflecting in the queue log...I;ve
tried the same with lot many calls ...I also tried with asterisk 1.6.0
version but same problem persists.. my dial plan is ttached below
along with sip.conf.
Extensions.conf
[incoming]
exten = _X.,1,*Queue*(55365,tT,,,90)
exten = _X.,2,Hangup
[from-internal]
exten => _X.,1,Answer
exten => _X.,2,Dial(SIP/{EXTEN},20,tT)
queues.conf
[general]
persistentmembers = yes
autofill = yes
Canreinvite=yes ; (tried with NO also)
monitor-type = MixMonitor
[55365]
fullname = Frontdesk
strategy = roundrobin
context=from-internal
ringinuse=no
setinterfacevar=yes
setqueueentryvar=yes
timeout = 10
wrapuptime =
autofill = yes
autopause = no
maxlen =
joinempty = no
leavewhenempty = no
reportholdtime = no
musicclass =
call-limit = 20
member = SIP/100
member = SIP/101
member = SIP/102
Please help , I m in a total mess ...Thanks Sriram
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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