There is no limit as far as asterisk goes. There can be other reasons such as T1 timers or rtptimeout being set. You need to start by enabling sip debug and seeing who sends the BYE then you need to figure out why they are hanging up.
Regards, Dovid -----Original Message----- From: Ikka Tirtawidjaja <[email protected]> Sender: [email protected]: Wed, 11 May 2016 18:26:48 To: asterisk-users<[email protected]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Subject: [asterisk-users] maximum call time -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
