[asterisk-users] Asterisk 1.6 Transfer issue[Edited]

[Miguel Molina]

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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