Merged revisions 89599 via svnmerge from
authorJoshua Colp <jcolp@digium.com>
Mon, 26 Nov 2007 18:04:46 +0000 (18:04 +0000)
committerJoshua Colp <jcolp@digium.com>
Mon, 26 Nov 2007 18:04:46 +0000 (18:04 +0000)
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6 lines

Add module counting removal for error conditions.
(closes issue #11333)
Reported by: Laureano
Patches:
      res_features_v2.c.patch uploaded by Laureano (license 265)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res/res_features.c

index 31ce64b..59ba1cf 100644 (file)
@@ -2336,6 +2336,7 @@ static int park_exec(struct ast_channel *chan, void *data)
                        if (error) {
                                ast_log(LOG_WARNING, "Failed to play courtesy tone!\n");
                                ast_hangup(peer);
+                               ast_module_user_remove(u);
                                return -1;
                        }
                } else
@@ -2345,6 +2346,7 @@ static int park_exec(struct ast_channel *chan, void *data)
                if (res < 0) {
                        ast_log(LOG_WARNING, "Could not make channels %s and %s compatible for bridge\n", chan->name, peer->name);
                        ast_hangup(peer);
+                       ast_module_user_remove(u);
                        return -1;
                }
                /* This runs sorta backwards, since we give the incoming channel control, as if it
@@ -2370,6 +2372,7 @@ static int park_exec(struct ast_channel *chan, void *data)
                /* Simulate the PBX hanging up */
                if (res != AST_PBX_NO_HANGUP_PEER)
                        ast_hangup(peer);
+               ast_module_user_remove(u);
                return res;
        } else {
                /*! \todo XXX Play a message XXX */