[Asterisk-Users] SIP vs. SIP :-(

[Igor Barsanti]

I'v a sip client and a sip trunk to FWD:

[general]
port=5060
context=default
tos=reliability
disallow=all
allow=ulaw
careinvite=no

[freeworlddialup]
context=default
type=friend
username=MYUSERNAME
secret=MYPASSWORD
host=fwd.pulver.com

[igor]
type=friend
callerid="Me"
host=dynamic
dtmfmode=rfc2833
careinvite=no

When i try to call a FWD number from SIP client i obtain a lot of
build_route: messages from asterisk then the sip client die

.......
Stopping retransmission on
'569a2a5a77939bca491565ec50d0d3e5 at 82.51.138.189' of Request 104: Found
build_route: Record-Route hop:
<sip:421171 at 192.246.69.223;ftag=as61269cb9;lr=on>
build_route: Contact hop: <sip:421171 at 65.39.205.121>
..............

...with H.323 client all works perfectly. What's the problem ???

Igor