asterisk users - Jan 2010 - Problem with call transfer and Polycom 430

Hi all.

I have a (new) customer who is describing symptoms that I've not seen
before.

They have 12 Polycom 430's behind a NAT, which is working OK.  When phone A
is
on a call and phone B attempts to transfer another call to phone C, the 
conversation on phone A is interrupted for 15-20 seconds...

The server is hardly loaded, and we have plenty of bandwidth to support our 
call level.

I have these lines in the sip.cfg file:

=================================================nat = yes
canreinvite = no
=================================================
Has anyone seen these symptoms before?  Any clues as to how to fix it?

TIA,

-- 

Take care and have fun,
Mike Diehl.