[SOLVED] SIP hangup issue 1.8.16.0 [481 to method: CANCEL]

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 "

my sip_gw.conf
are there any params i can tune?

[SIPGW]
username=SIPGW
secret=SIPGWSECRET
type=friend
qualify=no
context=from_external
dtmfmode=rfc2833
pickupgroup=10
callgroup=10
insecure=port
disallow=all
call-limit=100
allow=g729
allow=alaw
allow=ulaw
host=SIPGW_IP
canreinvite=yes

[quote]21.4.19 481 Call/Transaction Does Not Exist

This status indicates that the UAS received a request that does not
match any existing dialog or transaction.[/quote]

You are going to have to provide a SIP trace including Call-ID, and the tags on the To and From lines, starting from the INVITE.

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.

thank you,
i think this is the point becouse with another version of ISDN/SIP gateway firmware it is working.

i’m trying to get traces.
thank you again.

Hi,
placing an ANSWER() on top of dialplan is a workaround for this problem.

it was due to a gateway issue.

thanks everybody.