Hi guys, My Asterisk box is connected to two different SIP providers. I have one interesting problem with both of them: from time to time, SIP provider would not send the BYE command, eventhough the person on the other side already hung up the call. So the line gets stuck active until someone on our side hangs up.
Is rtptimeout parameter in SIP configuration supposed to solve this problem, or are there any more advanced options available too? Regards, Alex _______________________________________________ -- 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
