asterisk users - Feb 2006 - Sip - no peer or user found on incoming call

Hi list,

I try to connect to a GW which have one domain eg sip.mydomain.com and 
have few IPs related to this domain. I register * to this domain with 
host=sip.mydomain.com and type=user. So DNS will decide on which IP of 
my domain I will register (or redirection on the GW side).

If an incoming call arrive, I would guess that, as type=user, it will 
not try to match the IP from INVITE as I want to match on username. But 
this is not true, I always have in logs "Found no matching peer or user 
for 'xxx.xxx.xxx.xxx:5060'" and asterisk then try to find a
<MyUserName>
extension in the SIP default context. I tried to play with deny/permit 
without luck.

The call is finishing properly _only_ when the IP which with my * is 
registred to the GW match this from the incoming call, and then doesn't 
matter if type=user or type=peer, which is normal according to 
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer.

I'm running Asterisk SVN-trunk-r8643M built by root @ keewi on a i686 
running Linux on 2006-01-25 14:50:51 UTC

Here is relevant part of my sip.conf

register =>
<MyUserName>:<MySecret>@sip.mydomain.com/<MyUserName>

[IN-UserName]
type=user
username=<MyUserName>
fromuser=<MyUserName>
fromdomain=<MyFromDomainName>
secret=<MySecret>
context=incoming-GW
;deny=0.0.0.0/0.0.0.0
;permit=xxx.xxx.xxx.xx0/32
;permit=xxx.xxx.xxx.xx1/32
;permit=xxx.xxx.xxx.xx2/32
;permit=xxx.xxx.xxx.xx3/32
;permit=xxx.xxx.xxx.xx4/32
;permit=xxx.xxx.xxx.xx5/32
host=sip.mydomain.com
;insecure=invite,port	;very
;nat=yes
;canreinvite=no
;qualify=1000
disallow=all
allow=g726

Thanks for any clue.

-- 
Daniel