-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Michael Maier Sent: Wednesday, November 30, 2016 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Subject: [asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached
Hello all! I can see a strange problem during invite in dialog in the context of timer handling. Given is the following incoming call from provider at 8.195.88.234 (2@2) to my asterisk at 28.19.57.152 (1@1): After 900s suddenly *asterisk* starts the timer reinvite - I would have expected the reinvite started by the provider as usual. The expected reinvite by the provider is started during authentication of the reinvite started by asterisk and is answered immediately by asterisk with sip 481. The answer of the provider after the resend of the reinvite came about 0.5s later and is sip 481, too. => The session obviously isn't known on both sides! Asterisk therefore now drops the call (bye). Does anybody has any idea about the reason why both members don't recognize the existing session any more? I hope the attached sip trace can shed some light on the problem. Thanks, Michael HI Michael, You can set this in sip.conf: session-timers=refuse -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
