[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
More information about the asterisk-users
mailing list