----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4417/ -----------------------------------------------------------
(Updated Feb. 17, 2015, 9:32 a.m.) Status ------ This change has been marked as submitted. Review request for Asterisk Developers. Changes ------- Committed in revision 431898 Bugs: ASTERISK-24700 https://issues.asterisk.org/jira/browse/ASTERISK-24700 Repository: Asterisk Description ------- Analyzing a one off crash on a busy system showed that processing a REFER request had a NULL session channel pointer. The only way I can think of that could cause this is if an outgoing BYE transaction overlapped the incoming REFER transaction in a collision. Asterisk sends a BYE while the phone sends a REFER to complete an attended transfer. * Made check the session channel pointer before processing an incoming REFER request in res_pjsip_refer. * Fixed similar crash potential for res_pjsip supplement incoming request processing for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE, res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail REFER messages. * Made res_pjsip_messaging respond to a message body too large with a 413 instead of ignoring it. Diffs ----- /branches/13/res/res_pjsip_send_to_voicemail.c 431735 /branches/13/res/res_pjsip_sdp_rtp.c 431735 /branches/13/res/res_pjsip_refer.c 431735 /branches/13/res/res_pjsip_messaging.c 431735 /branches/13/res/res_pjsip_caller_id.c 431735 Diff: https://reviewboard.asterisk.org/r/4417/diff/ Testing ------- Since this is a very timing dependent problem, I made some calls and did an attended transfer for a warm fuzzy that nothing serious broke. Thanks, rmudgett
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev