Older CMs don't support SIP directly. You'll need to set up a Session Border Controller or CUBE between the two, and run h323 or an Inter-Cluster Trunk to the SBC/CUBE, then SIP from there to your other PBX.
I can't think of anything specific, other than keeping your fingers crossed as you'll find that no two vendors' implementations of SIP are alike. We've even found instances where upgrading an application to a different release from the same vendor can present challenges on how SIP calls behave.
Regarding the need for an SBC/CUBE, although CUCM 7 supports SIP trunks without it, depending on your application, you might find that some vendors require it if you're interfacing with an outside carrier. Actually, this would be considered a "best practice" when trunking to an outside vendor as the CUBE acts as a sort of firewall between you and the external network.
Thanks for the tips. This will all be directly to Asterisk and CS1000's for internal testing only, so it will be a good chance to learn how to do this from scratch!
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