[asterisk-users] FXO ports locking up

[Shawn L]

I have a situation where I have an Asterisk box which receives 8
analog lines from a
Mitel PBX and then drives 8 cordless SIP phones in a 1-to-1 mapping (a
call coming in
on port 1 of the digium FXO board is delivered to SIP phone 1, an
outgoing call on SIP
phone 2 goes out FXO line 2, etc.

This works fine normally, but every once in a while (no set time, or
pattern that I can
see -- It may be caused by the wifi sip phone going out of range of an
access point and
not coming back into range fast enough) the FXO port does not hangup
after the call is
terminated and just sits in an in-use state.  Since it's a 1-to-1
mapping, the SIP phone
associated with the in-use line now produces a fast busy when you
attempt to make a
call because it cannot get an outbound line.

Is there a way to detect that there is no longer really an active call
happening and force a
hangup or reset the channel?  It'd be great if this could happen
automatically.  Or as a
temporary fix , is there a way to setup and extension that the SIP
phone could dial which
would clear any active calls associated with it?  Right now if this
happens, I need to login
to the Asterisk CLI and issue a hangup command.  If I don't, the
channel appears to be
in-use forever.

Thanks

Shawn


The setup is fairly straight-forward

Extensions
[in-phone2]
exten => s,1,Answer()
exten => s,n,Noop(CALLERID(name))
exten => s,n,Noop(CALLERID(num))
exten => s,n,Dial(SIP/cordless2,25,tTo)
exten => s,n,Hangup

[out-phone2]
exten => _[*#0-9]!,1,Dial(${LINE2}/${EXTEN})
exten => _[*#0-9]!,2,Congestion()
exten => _[*#0-9]!,102,Congestion()

[cordless2]
type=friend
qualify=yes
rtptimeout=1
secret=XXXX
call-limit=1
nat=no
host=dynamic
canreinvite=no
context=out-phone2
callerid="cordless2" <102>