this is the scenario:
asterisk 220.127.116.11 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?
here is what my GW gets as response message from asterisk:
"Received Non Invite Response code: 481 to method: CANCEL "
are there any params i can tune?