diff options
author | Automerge script <automerge@asterisk.org> | 2012-12-12 00:17:36 +0000 |
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committer | Automerge script <automerge@asterisk.org> | 2012-12-12 00:17:36 +0000 |
commit | 7a7f9cba4324d576dfb59519915081793354ecd7 (patch) | |
tree | d1d1632a57ec3af79a3e6792554e91bce62e223f | |
parent | 686cdd0e79b01bfaf130389a1a948b2dc3d5efb0 (diff) |
Merged revisions 377911 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk
................
r377911 | mmichelson | 2012-12-11 18:02:31 -0600 (Tue, 11 Dec 2012) | 22 lines
Fix a potential deadlock in chan_sip during transfers.
The issue comes from the fact that transfers may perform
a redirecting update on a channel. The issue is that lock
inversion between the channel and its tech_pvt occurs since
the channel lock is released during the transfer process.
The fix is to move when the redirecting update occurs to a
place where neither the tech_pvt or the channel is locked so
that the two can be locked in the proper order.
(closes issue ASTERISK-20708)
reported by Mark Michelson
patches:
ASTERISK-20708-3.patch uploaded by Mark Michelson (License #5049)
Tested by:
Tim Ringenbach at Asteria Solutions Group
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Merged revisions 377910 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rw-r--r-- | channels/chan_sip.c | 31 |
1 files changed, 18 insertions, 13 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index fa70dc807..793e14179 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -26331,6 +26331,24 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, uint if (!ast_strlen_zero(referred_by)) { pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REFERER", referred_by); } + + /* When a call is transferred to voicemail from a Digium phone, there may be + * a Diversion header present in the REFER with an appropriate reason parameter + * set. We need to update the redirecting information appropriately. + */ + ast_channel_lock(p->owner); + sip_pvt_lock(p); + ast_party_redirecting_init(&redirecting); + memset(&update_redirecting, 0, sizeof(update_redirecting)); + change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE); + + /* Do not hold the pvt lock during a call that causes an indicate or an async_goto. + * Those functions lock channels which will invalidate locking order if the pvt lock + * is held.*/ + sip_pvt_unlock(p); + ast_channel_unlock(p->owner); + ast_channel_update_redirecting(current.chan2, &redirecting, &update_redirecting); + ast_party_redirecting_free(&redirecting); } sip_pvt_lock(p); @@ -26378,20 +26396,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, uint } ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */ - /* When a call is transferred to voicemail from a Digium phone, there may be - * a Diversion header present in the REFER with an appropriate reason parameter - * set. We need to update the redirecting information appropriately. - */ - ast_party_redirecting_init(&redirecting); - memset(&update_redirecting, 0, sizeof(update_redirecting)); - change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE); - - /* Do not hold the pvt lock during a call that causes an indicate or an async_goto. - * Those functions lock channels which will invalidate locking order if the pvt lock - * is held.*/ sip_pvt_unlock(p); - ast_channel_update_redirecting(current.chan2, &redirecting, &update_redirecting); - ast_party_redirecting_free(&redirecting); /* For blind transfers, move the call to the new extensions. For attended transfers on multiple * servers - generate an INVITE with Replaces. Either way, let the dial plan decided |