From: Richard Mudgett <rmudgett@digium.com>
Date: Wed, 9 Nov 2011 20:53:16 +0000 (+0000)
Subject: Fix deadlock during dialplan reload.
X-Git-Tag: 10.1.0-rc1~105
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=eca7e363377efd7bb54ef9bf69e0ed1c88fbfe71;p=thirdparty%2Fasterisk.git

Fix deadlock during dialplan reload.

Another deadlock between the conlock/hints and channels/channel locking
orders.

* Don't hold the channel and private lock in sip_new() when calling
ast_exists_extension().

(closes issue ASTERISK-18740)
Reported by: Byron Clark
Patches:
      sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky
      ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark
Tested by: Byron Clark
........

Merged revisions 344268 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6818bd222d..39b6fbfa97 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6972,7 +6972,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 	int needvideo = 0;
 	int needtext = 0;
 	char buf[SIPBUFSIZE];
-	char *decoded_exten;
+	char *exten;
 
 	{
 		const char *my_name;	/* pick a good name */
@@ -7117,14 +7117,15 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 	 * we should decode the uri before storing it in the channel, but leave it encoded in the sip_pvt
 	 * structure so that there aren't issues when forming URI's
 	 */
-	if (ast_exists_extension(NULL, i->context, i->exten, 1, i->cid_num)) {
-		/* encoded in dialplan, so keep extension encoded */
-		ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));
-	} else {
-		decoded_exten = ast_strdupa(i->exten);
-		ast_uri_decode(decoded_exten, ast_uri_sip_user);
-		ast_copy_string(tmp->exten, decoded_exten, sizeof(tmp->exten));
+	exten = ast_strdupa(i->exten);
+	sip_pvt_unlock(i);
+	ast_channel_unlock(tmp);
+	if (!ast_exists_extension(NULL, i->context, i->exten, 1, i->cid_num)) {
+		ast_uri_decode(exten, ast_uri_sip_user);
 	}
+	ast_channel_lock(tmp);
+	sip_pvt_lock(i);
+	ast_copy_string(tmp->exten, exten, sizeof(tmp->exten));
 
 	/* Don't use ast_set_callerid() here because it will
 	 * generate an unnecessary NewCallerID event  */