https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) | 11 lines
chan_sip: fix a deadlock in check_rtp_timeout.
Don't block doing silly deadlock avoidance. Just return and try again later.
The funciton gets called often enough that it's fine. Also, this change was
already made in trunk.
(closes issue #18791)
Reported by: irroot
Patches:
chan_sip.rtptimeout.patch uploaded by irroot (license 52)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317866
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (ast_rtp_instance_get_hold_timeout(dialog->rtp) && (t > dialog->lastrtprx + ast_rtp_instance_get_hold_timeout(dialog->rtp)))) {
/* Needs a hangup */
if (ast_rtp_instance_get_timeout(dialog->rtp)) {
- if(ast_channel_trylock(dialog->owner)) {
- /* Dont do a infinite deadlock avoidance loop.
- * Lets try this on next round (1 ms to 1000 ms later)
- * call is allready dead */
+ if (!dialog->owner || ast_channel_trylock(dialog->owner)) {
+ /*
+ * Don't block, just try again later.
+ * If there was no owner, the call is dead already.
+ */
return;
}
ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",