Hooman Fazaeli writes: > Hi > > I have noticed that asterisk returns 'SIP 603' when the called party does > not answer. > > My test setup is simple: two SIP phones (extensions: 100 and 111) > registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. > When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to > 111 (expected) and a '603 Decline' response to 100 (unexpected & > misleading). > It seems to me that a'480 Temporarily unavailable' response is more > suitable in this situation. > > Is this a normal behavior of asterisk or a bug? > > Thanks.
That sounds like you are not doing a Hangup(). What is the dialplan that you are using? -- Joris Engbers -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
