asterisk users - Apr 2004 - problem with SIP configuration AND EXTENSION.

When run 
asterisk ?vvvgc
 
IT show me this error

 
Asterisk Ready.
*CLI> Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout:
Registration for 'phone@192.168.0.6' timed out, trying again
Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration
from '<sip:phone@192.168.0.6>' failed for '192.168.0.6'
Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:4993 handle_response: Failed to
authenticate on REGISTER to
'<sip:phone@192.168.0.6>;tag=as4196dc97'
Apr 11 08:59:47 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration
for 'phone@192.168.0.6' timed out, trying again
Apr 11 08:59:47 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration
from '<sip:phone@192.168.0.6>' failed for '192.168.0.6'
Apr 11 08:59:47 NOTICE[81926]: chan_sip.c:4993 handle_response: Failed to
authenticate on REGISTER to
'<sip:phone@192.168.0.6>;tag=as3faa3b67'
 
I have in sip.conf 
 
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
register => phone@192.168.0.6/phone ; 192.168.0.6 it?s my server linux
ASTERISK.
 
[snomsip]
type=vozip
secret=blah
host=dynamic
dtmfmode=inband         
defaultip=192.168.0.9  ; phone grandstream
mailbox=1234,2345               
restrictcid=no          
 
 
how can configure the extensions.conf if  I want to sound my grandstream
when I have incoming calls.?
 
 
ANY IDEAS
 
Cheers.
 
vozip
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>Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration
from '<sip:phone@192.168.0.6>' failed for '192.168.0.6'
Are you sure your phone isn't registering?   These errors aren't related
to your grandstream.  Do a "sip show peers" at the Asterisk CLI and
see if it shows your phone registered.
>I have in sip.conf 
> 
>[general]
>port = 5060                     ; Port to bind to
>bindaddr = 0.0.0.0              ; Address to bind to
>register => phone@192.168.0.6/phone ; 192.168.0.6 it?s my server linux
ASTERISK.
Take this line out.  You don't need it.  That's only for remote SIP
providers.  You're telling your * box to register with itself.  And
obviously bad things are happening!

Sean