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Fix a potential deadlock in chan_sip during transfers.

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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
........

Merged revisions 377910 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Michelson
2012-12-12 00:02:31 +00:00
parent 9c74a60ba5
commit 607a5d898c
1 changed files with 18 additions and 13 deletions
Download Patch File Download Diff File Expand all files Collapse all files

31
channels/chan_sip.c
View File

@@ -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