[asterisk-commits] tilghman: branch 1.6.1 r157255 - in /branches/1.6.1: ./ apps/app_dial.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 17 16:39:55 CST 2008
Author: tilghman
Date: Mon Nov 17 16:39:55 2008
New Revision: 157255
URL: http://svn.digium.com/view/asterisk?view=rev&rev=157255
Log:
Merged revisions 157253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r157253 | tilghman | 2008-11-17 16:25:06 -0600 (Mon, 17 Nov 2008) | 8 lines
Can't use items duplicated off the stack frame in an element returned from
a function: in these cases, we have to use the heap, or garbage will result.
(closes issue #13898)
Reported by: alecdavis
Patches:
20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: alecdavis
........
Modified:
branches/1.6.1/ (props changed)
branches/1.6.1/apps/app_dial.c
Propchange: branches/1.6.1/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.1/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.1/apps/app_dial.c?view=diff&rev=157255&r1=157254&r2=157255
==============================================================================
--- branches/1.6.1/apps/app_dial.c (original)
+++ branches/1.6.1/apps/app_dial.c Mon Nov 17 16:39:55 2008
@@ -962,7 +962,7 @@
play_to_caller = 1;
var = pbx_builtin_getvar_helper(chan, "LIMIT_WARNING_FILE");
- config->warning_sound = !ast_strlen_zero(var) ? ast_strdupa(var) : "timeleft";
+ config->warning_sound = !ast_strlen_zero(var) ? ast_strdup(var) : ast_strdup("timeleft");
/* The code looking at config wants a NULL, not just "", to decide
* that the message should not be played, so we replace "" with NULL.
@@ -971,10 +971,10 @@
*/
var = pbx_builtin_getvar_helper(chan, "LIMIT_TIMEOUT_FILE");
- config->end_sound = !ast_strlen_zero(var) ? ast_strdupa(var) : NULL;
+ config->end_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
var = pbx_builtin_getvar_helper(chan, "LIMIT_CONNECT_FILE");
- config->start_sound = !ast_strlen_zero(var) ? ast_strdupa(var) : NULL;
+ config->start_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
ast_channel_unlock(chan);
@@ -2060,6 +2060,15 @@
}
done:
+ if (config.warning_sound) {
+ ast_free((char *)config.warning_sound);
+ }
+ if (config.end_sound) {
+ ast_free((char *)config.end_sound);
+ }
+ if (config.start_sound) {
+ ast_free((char *)config.start_sound);
+ }
return res;
}
More information about the asterisk-commits
mailing list