[asterisk-users] call fail from audiocode to sip trunk

[satish patel]

Dear ALL

          I have audiocode MP -124 with configure in asterisk Endpoint configuration means every analog phone register in asterisk now thing is that i have one more SIP trunk with mediant 2000 

[auodiocode-mp-124]-----[ * ]------[mediant 2000]-----E1


When i call from audiocode MP -124 phone i got this error 

   -- Executing Dial("SIP/20-0889c4d8", "SIP/mediant/1") in new stack
    -- Called mediant/1
    -- SIP/mediant-088a1a18 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Congestion("SIP/20-0889c4d8", "") in new stack
  == Spawn extension (mysip, 111, 2) exited non-zero on 'SIP/20-0889c4d8'
    -- Executing Dial("SIP/24-0889c4d8", "SIP/mediant/0") in new stack
    -- Called mediant/0

my extension.conf file is 

exten => 43,1,Answer
exten => 43,2,Dial(SIP/43)
exten => 43,3,Hangup
exten => 777,1,Answer()
exten => 777,2,Dial(SIP/777)
exten => 777,3,Hangup()
exten => 888,1,Answer()
exten => 888,2,Dial(SIP/888)
exten => 55,1,Dial(SIP/55)
exten => 66,1,Dial(SIP/66)

exten => _11.,1,Dial(SIP/mediant/${EXTEN:2})
exten => _11.,2,Congestion

what is the problem

 
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