[Asterisk-Users] Asterisk Not hanging up DS0 when number called is busy.

James Sizemore james at deny.org
Tue Feb 1 12:30:39 MST 2005


I have a PRI that if you dial a number that is busy, the channel does 
not hang up, it then sends "h|1"to the phone company which will then 
plays back to the end sip user "You don't need to dial a one or zero"
I am running stable CVS-v1-0-01/20/05-02:45:17. I have placed the 
important bit from the extension and sip configs below. Simplest 
possible example that will show the problem. Anyone run into this 
problem before?

     -- Executing 
Dial("SIP/192.168.69.254-08d76480","Zap/g1/5554441133") in new stack
     -- Called g1/5554441133
     -- Channel 0/1, span 1 got hangup
     -- Zap/1-1 is busy
     -- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time
     -- Timeout on SIP/192.168.69.254-08d76480
   == CDR updated on SIP/192.168.69.254-08d76480
     -- Executing Goto("SIP/192.168.69.254-08d76480", "h|1") in new stack
     -- Goto (default-out,h,1)
     -- Executing Hangup("SIP/192.168.69.254-08d76480", "") in new stack
   == Spawn extension (default-out, h, 1) exited non-zero on 
'SIP/192.168.69.254-08d76480'
     -- Executing Hangup("SIP/192.168.69.254-08d76480", "") in new stack
   == Spawn extension (default-out, h, 1) exited non-zero on 
'SIP/192.168.69.254-08d76480'

extensions.conf:
[trunk]
exten => _X.,1,Dial(${TRUNK}/${EXTEN})
exten => h,1,Hangup

[default-out]
include => trunk


sip.conf:
[office]
type=friend
host=192.168.69.254
context=default-out
canreinvite=no
dtmfmode=inband
accountcode=office






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