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authormarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2005-08-03 20:17:53 +0000
committermarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2005-08-03 20:17:53 +0000
commit248b4e7facbd79a128c86593e150a2d9b2801572 (patch)
tree17eaa0231cff424013e1eaead6a878976678f0d8 /apps
parent5b604821d5cc01e2507108b85fcc339158e5c0c1 (diff)
Be sure not to leak frames in certain cases
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6275 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'apps')
-rwxr-xr-xapps/app_dial.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c
index f56f08907..8fe9fc43a 100755
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -599,8 +599,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF))) {
if (ast_write(outgoing->chan, f))
ast_log(LOG_WARNING, "Unable to forward voice\n");
- ast_frfree(f);
}
+ ast_frfree(f);
}
if (!*to && (option_verbose > 2))
ast_verbose( VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", orig);