res_pjsip_refer: Fix crash from a REFER and BYE collision.
authorRichard Mudgett <rmudgett@digium.com>
Tue, 17 Feb 2015 15:34:10 +0000 (15:34 +0000)
committerRichard Mudgett <rmudgett@digium.com>
Tue, 17 Feb 2015 15:34:10 +0000 (15:34 +0000)
commit09bfe4b2088e61a085004f5cd679040532533054
tree50feade55641576dbcc8a1b199bcbf6051bb3d3e
parentd808eace5c308bafc9b592d94d7b7c2b98b1e84c
res_pjsip_refer: Fix crash from a REFER and BYE collision.

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.

ASTERISK-24700 #close
Reported by: Zane Conkle

Review: https://reviewboard.asterisk.org/r/4417/
........

Merged revisions 431898 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res/res_pjsip_caller_id.c
res/res_pjsip_messaging.c
res/res_pjsip_refer.c
res/res_pjsip_sdp_rtp.c
res/res_pjsip_send_to_voicemail.c