Asterisk leitet Anruf nicht weiter

[default]
include => 101
include => demo
;include => from_sipgate
exten => ,1,Dial(SIP/${EXTEN},60)
exten => 101,2,Congestion
exten => 101,102,Busy

[sipgate]
include => default
exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate.de,60,trg)
exten => _9.,2,Congestion
exten => _9.,102,Busy

[from-sipgate]
exten => 1959064,1,Ringing
exten => 1959064,2,Wait,2
exten => 1959064,3,Dial(SIP/101,20,tr)
exten => 1959064,4,Hangup

Zum Vergrößern anklicken....

-- parse_srv: SRV mapped to host proxy.de.sipgate.net, port 5060
-- Executing Ringing("SIP/1959064-84d6", "") in new stack
-- Executing Wait("SIP/1959064-84d6", "2") in new stack
-- Executing Dial("SIP/1959064-84d6", "SIP/101") in new stack
Mar 27 18:23:04 NOTICE[1152990128]: app_dial.c:743 dial_exec: Unable to create channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing Hangup("SIP/1959064-84d6", "") in new stack
== Spawn extension (from-sipgate, 1959064, 4) exited non-zero on 'SIP/1959064-84d6'

Zum Vergrößern anklicken....