In SIP, there’s not a physical tying of one trunk to one simultaneous call, as there would be with analog telephony (one circuit to one call) or digital telephony (one bearer channel per call). If you want 50 simultaneous calls, you tell your provider you want to do 50 simultaneous calls.
I might not be fully understanding your proposal, but if you are indeed tying into a traditional PBX and wanting to push it out to SIP, and that traditional PBX does not already support SIP (you’re using Asterisk to get there), then you’ll need some kind of connection from the traditional PBX to Asterisk that can support 50 or 100 simultaneous connections. In this case, you’re going to be pushing E1s (most of the world, 30 channels / simultaneous calls per E1) or T1s (North America, 23 or 24 channels / simultaneous calls per T1) from that traditional PBX to Asterisk. For that, I, naturally, recommend Digium’s multi-port digital E1/T1 interface cards (store.digium.com/telephony_card_selector.php)