Index: channels/chan_sip.c
===================================================================
--- channels/chan_sip.c	(revision 376570)
+++ channels/chan_sip.c	(working copy)
@@ -25820,6 +25820,24 @@
 		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);
@@ -25867,21 +25885,6 @@
 	}
 	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
 	 * indicate before masquerade so the indication actually makes it to the real channel