Hi all,
this is the scenario:
asterisk 1.8.16.0 from sources
ISDN lines --> ISDN/SIP gateway --> ASTERISK --> SIP PHONES
“A” calls from ISDN, my Gateway send correctly the call to asterisk.
if “A” hangups before someone answers, ASTERISK still got the call as ACTIVE and phones still ring: if someone answers that call, it’s active and, obviously, mute.
why does this happen?
ideas?
here is what my GW gets as response message from asterisk:
"Received Non Invite Response code: 481 to method: CANCEL "
The problem is most probably in your ISDN/SIP gateway. If the remote caller ends the call before it is answered on a phone connected to the Asterisk box, ISDN/SIP gateway should indicate the event with a SIP message RIGH AWAY. Otherwise Asterisk can not know if the call has been cancelled by the remote end.