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Merged revisions 317865 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.8

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  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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2011-05-06 19:48:06 +00:00
parent 426aa5e09b
commit ae8dbde4a8
1 changed files with 5 additions and 4 deletions
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9
channels/chan_sip.c
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@@ -24933,10 +24933,11 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
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",