asterisk users - Jun 2003 - outgoing calls with packet8 and dta310 problems

I'm trying to get asterisk working w/ packet8 (incoming and outgoing)
and a dta310 so I can have more control over voicemail.  I've looked
at the data stream coming from the dta310 and from packet8, but I
haven't managed to get the phone to actually place invites through
asterisk.  On the asterisk end with chan_oss.so, I can make it dial
and I hear ringing and the first second of conversation, but after
that it drops the call with a message along the lines of "nosupported
codecs in SDP" or somesuch.  If anyone wants it, I can get the full
message when I get home.  When I try to use the DTA310, asterisk
complains about authentication on SUBSCRIBE requests.

Here is the relevant info in my sip.conf file:

register =>0123456789@packet8.net/0123456789
;the 0123456789 is my activation number for packet8 service...
;the DTA seems to want to use it.

[packet8]
type=friend
disallow=gsm
secret=mypassword
allow=g723.1
;expirey=15
sip_codec=g723.1
username=0403531400
fromuser=0403531400
host=packet8.net

[0123456789]
type=friend
allow=g723.1
defaultip=192.168.1.247
user=0123456789
fromuser=0123456789
context=default
host=dynamic
secret=mypassword
careinvite=no
expirey=30
reinvite=no

and from the extensions.conf, I dial like this:

exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@packet8)
exten => _91NXXNXXXXXX,2,Congestion

In the dta310, I have the following SIP server settings:

Ip address: the asterisk machine's IP
Port: 5060
Domain Name: the asterisk machine's name
Phone number: blank
CallerID Name: blank
Port: 5060
User Name: 0123456789
Password: mypassword


Does anyone who has done this have any hints for me, or possibly an indication
of what I've screwed up?

Thanks,
Matthias
-- 
Matthias Granberry
matthias@utdallas.edu
(469) 371-0596