Fix small leak on app_dial

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1171 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_dial.c b/apps/app_dial.c
index 89deefa..38f70bc 100755 (executable)
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -305,6 +305,7 @@ 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);
                        }
                }
                if (!*to && (option_verbose > 2))