Hi! I have the following scenario: Asterisk INVITE-----> | <--200,ACK-- | Playback(Foo) | Dial(..) | ---------INVITE-----> | <-----404. ACK------> | As my extension configuration stops after the Dial command I expect Asterisk to hang up the call. Instead I see on the console: | | == Auto fallthrough, channel 'SIP/81.16.153.183-0883dea0' status is 'CONGESTION' | | Then I hear the congestion tone for 10 secondes, then Asterisk sends BYE | <-----BYE----|
Why does Asterisk not hangup immediately? Why 10 seconds congestion tone? Is this configurable? thanks klaus _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users