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res_pjsip_refer: Fix crash from a REFER and BYE collision.

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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
This commit is contained in:
Richard Mudgett
2015-02-17 15:34:10 +00:00
parent d808eace5c
commit 09bfe4b208
5 changed files with 31 additions and 15 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
res/res_pjsip_caller_id.c
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@@ -361,7 +361,7 @@ static int caller_id_incoming_request(struct ast_sip_session *session, pjsip_rx_
if (!session->endpoint->id.self.number.valid) {
set_id_from_from(rdata, &session->id);
}
} else {
} else if (session->channel) {
/* Reinvite. Check for changes to the ID and queue a connected line
* update if necessary
*/