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

[Dovid B]

What error are you getting on the Audio Codes side ? Set verbose to 5 on the Audio codes box and try running Syslog.
  ----- Original Message ----- 
  From: satish patel 
  To: asterisk-users at lists.digium.com 
  Sent: Tuesday, June 26, 2007 2:14 PM
  Subject: [asterisk-users] call fail from audiocode to sip trunk


  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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