Fix small leak on app_dial
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1171 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -305,6 +305,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
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if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF))) {
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if (ast_write(outgoing->chan, f))
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ast_log(LOG_WARNING, "Unable to forward voice\n");
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ast_frfree(f);
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}
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}
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if (!*to && (option_verbose > 2))
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