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author | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2005-08-03 20:22:27 +0000 |
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committer | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2005-08-03 20:22:27 +0000 |
commit | 1454c2f045bce799e26854d962f1af3d99581710 (patch) | |
tree | 5ed3df404906be1909330a5b39e6bbb180517e74 | |
parent | dceb7e21ffebc96a6eeb56ffe057894c0d73d12b (diff) |
Fix memory leak
git-svn-id: http://svn.digium.com/svn/asterisk/branches/v1-0@6276 f38db490-d61c-443f-a65b-d21fe96a405b
-rwxr-xr-x | apps/app_dial.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c index 0d16d0483..c5d9c92b4 100755 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -414,8 +414,8 @@ 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); |