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

[Sriram]

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

 

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

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