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Can I use SIP outbound proxy to bypass NAT?

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512Kb dedicated for VOIP or a DSL line used for other stuff as well? This along with the CODEC used will determine the number of concurrent calls you can expect before bandwidth becomes a call quality issue. Keep in mind bandwidth is usually not the culprit for call quality but rather QoS. The main reason people usually have issues with VOIP is they try to use SIP trunks over public internet where neither the user nor the trunk provider have control over the middle network resulting in sporadic problems that neither end can do anything about. Don't get me wrong it is entirely possible to have good calls over the public internet - you just never know when something in the middle will cause an issue.

On a side note to the above caution - A VPN to the other server (TCP based with very lightweight encryption) can actually help voice quality over the public internet in some cases by reducing the number of lost packets - YMMV.

Another option to look at (or combined with a VPN):
Since you are Asterisk based on at least one end you could also have a local asterisk server that takes SIP from the phones and uses an AIX2 trunk to the far end server. AIX2 uses a single port for RTP and messaging which solves a lot of the NAT problems and also reduces some of the messaging bandwidth overhead from SIP - although not likely to make a difference in this case. You can also specify a port for AIX2 to use to get around a providers limitations.