You should be able to get active/active SIP services that present calls randomly to one or the other for incoming calls unless one fails in which case all calls go to the other.
For outgoing calls, I'm not sure that a Route List actually works if a Route is congested - the call will just fail, not use the second choice Route. Maybe I'm wrong? Either way, I always go the active/active and then use the order on my Route Lists to juggle outgoing calls until they are goinhg out roughly evenly among the SIP trunks.
You can get rid of the physical box and upgrade it to a virtual. The time spent making it virtual will be paid back many times over when you do your next upgrade.