[asterisk-users] canreinvite question

[Vincent]

On Fri, 22 Feb 2008 18:50:16 +0800, Ron <ron at silverbackasp.com> wrote:
>If i set, canreinvite=yes on all ext, assuming all ip phones have the 
>same codec, if 100 calls 101, or vice versa will rtp still go thru 
>asterisk? and same scenario for 200 to 202 or vice versa.

... and I'd like to add to this question: If the phones have the
option "Enable NAT", I expected them to be able to talk to each other
directly, but they didn't, and I had to set them to "canreinvite=no"
in sip.conf. Why is that?