From: Automerge script <automerge@asterisk.org>
Date: Wed, 12 Dec 2012 00:17:36 +0000 (+0000)
Subject: Merged revisions 377911 via svnmerge from
X-Git-Tag: 13.0.0-beta1~2194^2~68
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=7a7f9cba4324d576dfb59519915081793354ecd7;p=thirdparty%2Fasterisk.git

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
  ........

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index fa70dc807c..793e141794 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