Basically, (and it's a simple problem) if a user taps the hook switch quickly they get dialtone again but it does not hangup the existing call. The user can then make another call, however, i have incominglimit=1 in sip.conf so they cannot. This means the original call get's lost. Does anyone know how to retrieve the call? Or at least where there is some documentation on this 'feature'?
TIA
|
-- Jamie Carl <[EMAIL PROTECTED]> Resident Geek Achieve Corp. +61262648200 |
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
