If Asterisk receives a SIP UPDATE request after a call has been terminated and
the channel has been destroyed but before the SIP dialog has been destroyed, a
condition exists where a connected line update would be attempted on a
non-existing channel. This would cause Asterisk to crash. The patch resolves
this by first ensuring that the SIP dialog has an owning channel before
attempting a connected line update. If an UPDATE request is received and no
channel is associated with the dialog, a 481 response is sent.
(closes issue ASTERISK-19770)
Reported by: Thomas Arimont
Tested by: Matt Jordan
Patches:
ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license 6283)
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Merged revisions 363106 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 363107 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363108
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
transmit_response(p, "501 Method Not Implemented", req);
return 0;
}
+ if (!p->owner) {
+ transmit_response(p, "481 Call/Transaction Does Not Exist", req);
+ return 0;
+ }
if (get_rpid(p, req)) {
struct ast_party_connected_line connected;
struct ast_set_party_connected_line update_connected;