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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