Hi, I have a queue with one agent added using AddQueueMember (FAO|Local/[EMAIL PROTECTED]|0||Agent/602). My extensions.conf is
[general] static=yes writeprotect=yes autofallthrough=no clearglobalvars=no priorityjumping=no [from-sip] exten => 100001000,1,Dial(SIP/100001000,,t) exten => 1001,1,Dial(SIP/1001,,t) exten => 1002,1,Dial(SIP/1002,,t) exten => 1003,1,Dial(SIP/1003,,t) exten => 1004,1,Dial(SIP/1004,,t) exten => 2001,1,agi,login.php exten => 2002,1,Queue(FAO|tT) exten => 2004,1,MusicOnHold exten => 2004,2,Hangup When I call from 100001000 to 1001, I can press # and type 2004 to transfer and 100001000 gets MOH. When I dial 2002 (queue) from 100001000, 1001 rings and I am able to talk both ways, but nothing happens when I press # at 1001. No logs appears at asterisk console in verbose 3 level. I am using asterisk 1.4.15. All the docs indicate that I just need to invoke Queue application with tT to enable call transfer. But that does not seems to work in my case. queues.conf [general] persistentmembers = no eventwhencalled = yes autofill = yes monitor-type = MixMonitor [FAO] musiconhold = default strategy = roundrobin servicelevel = 60 eventmemberstatus = yes eventwhencalled = yes timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes monitor-format = gsm sip.conf [general] context=from-sip allowguest=no bindport=5060 bindaddr=192.168.3.36 srvlookup=yes [100001000] host=dynamic type=friend dtmfmode=RFC2833 username=100001000 secret=masked context=from-sip disallow=all allow=ulaw allow=alaw incominglimit=1 canreinvite=no [1001] host=dynamic type=friend dtmfmode=RFC2833 username=1001 secret=masked context=from-sip disallow=all allow=ulaw allow=alaw incominglimit=1 canreinvite=no Thanks and regards, raj _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users