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 transferred part 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090924/99940f37/attachment.htm