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Since it seems nobody here has a clue I'll share what I eventually found out using the OpenStage 40 SIP Administrators Guide - seems the parameter names are fairly standardized except under SIP settings - what Siemens refers to as the "Voice Gateway" IP is what Cisco calls the "Outgoing Dial Proxy" and the "Incoming Proxy". That will get you dial tone, but you must also put a tick mark in the "Register with proxy" box to be able to use the phone.