Merged revisions 373466,373468 via svnmerge from
file:///srv/subversion/repos/asterisk/branches/10
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r373466 | rmudgett | 2012-09-24 15:44:27 -0500 (Mon, 24 Sep 2012) | 33 lines
Fix potential reentrancy problems in chan_sip.
Asterisk v1.8 and later was not as vulnerable to this issue.
* Made find_call() lock each private as it processes the found dialogs.
(Primary cause of ABE-2876)
* Made the other functions that traverse the dialogs container lock each
private as it examines them.
* Fix race condition in sip_call() if the thread that sent the INVITE is
held up long enough for a response to be processed. The p->initid for the
INVITE retransmission could be added after it was canceled by the response
processing.
* Made __sip_destroy() clean up resource pointers after freeing. This is
primarily defensive in case someone has a stale private pointer.
* Removed redundant memset() in reqprep(). The call to init_req() already
does the memset() and is the first reference to req in reqprep().
* Removed useless set of req.method in transmit_invite(). The calls to
initreqprep() and reqprep() have to do this because they memset() the req.
JIRA ABE-2876
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Merged -r373423 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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Merged revisions 373424 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r373468 | jrose | 2012-09-24 16:05:44 -0500 (Mon, 24 Sep 2012) | 13 lines
func_audiohookinherit: Document some missed sources.
This patch also mentions that AUDIOHOOK_INHERIT can be used to
transfer MixMonitor audiohooks. There is also wiki that addresses
audiohooks and the use of AUDIOHOOK_INHERIT at the following link:
https://wiki.asterisk.org/wiki/display/AST/Audiohooks
(closes issue ASTERISK-18220)
Reported by: Ishfaq Malik
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Merged revisions 373467 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373498
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