From: Corey Farrell <git@cfware.com>
Date: Sun, 2 Nov 2014 08:13:52 +0000 (+0000)
Subject: Fix ast_writestream leaks
X-Git-Tag: 14.0.0-beta1~1501
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=509c04ef381cd525fe92b1f23a7031f6cf20fb54;p=thirdparty%2Fasterisk.git

Fix ast_writestream leaks

Fix cleanup in __ast_play_and_record where others[x] may be leaked.
This was caught where prepend != NULL && outmsg != NULL, once
realfile[x] == NULL any further others[x] would be leaked. A cleanup
block was also added for prepend != NULL && outmsg == NULL.

11+: Fix leak of ast_writestream recording_fs in
app_voicemail:leave_voicemail.

ASTERISK-24476 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4138/
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Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 427024 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 427025 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 427026 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c
index af4e7f0815..d99bced3d4 100644
--- a/apps/app_voicemail.c
+++ b/apps/app_voicemail.c
@@ -6158,6 +6158,7 @@ static int msg_create_from_file(struct ast_vm_recording_data *recdata)
 				ast_log(LOG_ERROR,"Unable to determine sample rate of recording %s\n", recdata->recording_file);
 			}
 		}
+		ast_closeframe(recording_fs);
 	}
 
 	/* If the duration was below the minimum duration for the user, let's just drop the whole thing now */
diff --git a/main/app.c b/main/app.c
index fd19beeafc..dea5ee6ed7 100644
--- a/main/app.c
+++ b/main/app.c
@@ -1782,18 +1782,20 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 			ast_truncstream(others[x]);
 			ast_closestream(others[x]);
 		}
-	}
-
-	if (prepend && outmsg) {
+	} else if (prepend && outmsg) {
 		struct ast_filestream *realfiles[AST_MAX_FORMATS];
 		struct ast_frame *fr;
 
 		for (x = 0; x < fmtcnt; x++) {
 			snprintf(comment, sizeof(comment), "Opening the real file %s.%s\n", recordfile, sfmt[x]);
 			realfiles[x] = ast_readfile(recordfile, sfmt[x], comment, O_RDONLY, 0, 0);
-			if (!others[x] || !realfiles[x]) {
+			if (!others[x]) {
 				break;
 			}
+			if (!realfiles[x]) {
+				ast_closestream(others[0]);
+				continue;
+			}
 			/*!\note Same logic as above. */
 			if (dspsilence) {
 				ast_stream_rewind(others[x], dspsilence - 200);
@@ -1810,7 +1812,15 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 			ast_verb(4, "Recording Format: sfmts=%s, prependfile %s, recordfile %s\n", sfmt[x], prependfile, recordfile);
 			ast_filedelete(prependfile, sfmt[x]);
 		}
+	} else {
+		for (x = 0; x < fmtcnt; x++) {
+			if (!others[x]) {
+				break;
+			}
+			ast_closestream(others[x]);
+		}
 	}
+
 	if (rfmt && ast_set_read_format(chan, rfmt)) {
 		ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_format_get_name(rfmt), ast_channel_name(chan));
 	}