Ikka, Do a simple sip debug and see who sends the bye. You can also simply run tcpdump in a screened session and when the call is done analyze in wireshark. tcpdump -s0 host <IP of carrier> and port 5060 -w /tmp/my-trace.pcap
Regards, Dovid -----Original Message----- From: Ikka Tirtawidjaja <[email protected]> Sender: [email protected]: Thu, 12 May 2016 08:08:49 To: Asterisk Users Mailing List - Non-Commercial Discussion<[email protected]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Subject: Re: [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
