[asterisk-commits] kpfleming: branch kpfleming/agi_multiple-api-fix r157288 - in /team/kpfleming...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Nov 17 18:10:52 CST 2008


Author: kpfleming
Date: Mon Nov 17 18:10:52 2008
New Revision: 157288

URL: http://svn.digium.com/view/asterisk?view=rev&rev=157288
Log:
bring up to date

Modified:
    team/kpfleming/agi_multiple-api-fix/   (props changed)
    team/kpfleming/agi_multiple-api-fix/apps/app_dial.c

Propchange: team/kpfleming/agi_multiple-api-fix/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Nov 17 18:10:52 2008
@@ -1,1 +1,1 @@
-/trunk:1-157222
+/trunk:1-157287

Modified: team/kpfleming/agi_multiple-api-fix/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/kpfleming/agi_multiple-api-fix/apps/app_dial.c?view=diff&rev=157288&r1=157287&r2=157288
==============================================================================
--- team/kpfleming/agi_multiple-api-fix/apps/app_dial.c (original)
+++ team/kpfleming/agi_multiple-api-fix/apps/app_dial.c Mon Nov 17 18:10:52 2008
@@ -1170,7 +1170,7 @@
 		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.
@@ -1179,10 +1179,10 @@
 	 */
 
 	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);
 
@@ -2264,6 +2264,15 @@
 	}
 
 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