Fix deadlock in SIP transfers that involve a REFER request

In r367163, "send to voicemail" functionality was added to the SIP channel
driver.  This required updating the party redirecting information for the
channel based on the headers provided in the REFER request.  When the
redirecting party information is updated on the channel, a call to
ast_indicate_data occurs.  Because handle_request_refer still had the sip_pvt
locked, a deadlock could occur between the pbx_thread and the do_monitor thread
servicing the REFER request.

This patch preserves the proper locking order between the channel and the
sip_pvt by ensuring that the sip_pvt is unlocked prior to updating the party
redirecting information on the channel.

(closes issue AST-903)
Reported by: Matt Jordan
patches:
  jira_ast_903_trunk.patch by rmudgett (license 5621)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368793 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5fd1bc1..274e978 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -24858,16 +24858,18 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
        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);
 
-       /* Do not hold the pvt lock during the indicate and async_goto. Those functions
-        * lock channels which will invalidate locking order if the pvt lock is held.*/
        /* 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
         * when using local channels with MOH passthru */
-       sip_pvt_unlock(p);
        ast_indicate(current.chan2, AST_CONTROL_UNHOLD);
        res = ast_async_goto(current.chan2, refer_to_context, refer_to, 1);