That would be reinvite= and canreinvite= in the user entry for each SIP endpoint. Asterisk will allow the endpoints to talk directly to each other if both those settings are = yes (the default, I think) AND both endpoints use the same protocol (SIP) AND the same codec.
I tried to document this on the Wiki. Did I get it right? Please check! http://tinyurl.com/mvny
(If it's wrong, just change it, it's a Wiki)
/O
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