Time conditions causing audio delay

I have two problems that seems to have crept up out of nowhere.

First, when calls come in, there is a very long delay in audio on both endpoints. I did a tcpdump of rtp packets and the data flows immediately between the server and phone and server and trunk. Second, outbound calls for one user do not go out and get a busy signal.

The call flow is as follows: Inbound DID>time condition>ringgroup>all extensions ring. I started testing different routes and the problem cleared up. I finally narrowed it down to the time condition, when the condition is ''Destination if time matches"–this is the only time that audio delays occur. When destination does not match, the system goes to an IVR and rtp data from the ivr can be heard right away.
For the time being, I’m sending calls directly to the ring group for the time being.

As for the latter problem, it also cleared up after I bypassed time conditions (what in the world???). Very strange, any ideas?

You are using FreePBX terminology but this is not the FreePBX board.

In raw Asterisk, neither GotoIfTime nore time dependent contexts can have any conceivable effect on media handling.