From 80e2db47c25e7fb87d0fbc9d2bbde989f49cecba Mon Sep 17 00:00:00 2001
From: Mark Michelson <mmichelson@digium.com>
Date: Mon, 18 Feb 2008 20:53:25 +0000
Subject: [PATCH] Merged revisions 103786 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb 2008) | 10 lines
There was an invalid assumption when calculating the duration of a file that the filestream in question
was created properly. Unfortunately this led to a segfault in the situation where an unknown format was
specified in voicemail.conf and a voicemail was recorded. Now, we first check to be sure that the stream
was written correctly or else assume a zero duration.
(closes issue #12021)
Reported by: jakep
Tested by: putnopvut
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
main/app.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/main/app.c b/main/app.c
index b70ef022ff..99eaec3f63 100644
--- a/main/app.c
+++ b/main/app.c
@@ -760,7 +760,7 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
* message, otherwise we could get a situation where this stream is never
* closed (which would create a resource leak).
*/
- *duration = ast_tellstream(others[0]) / 8000;
+ *duration = others[0] ? ast_tellstream(others[0]) / 8000 : 0;
if (!prepend) {
for (x = 0; x < fmtcnt; x++) {
--
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