Be sure not to leak frames in certain cases

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6275 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_dial.c b/apps/app_dial.c
index f56f089..8fe9fc4 100755 (executable)
--- 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);