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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