[Asterisk-Users] Asterisk SIP + Grandstream 100 + sip.conf phone HELP

[pesb]

Hi there,
               I have been trying with asterisk, plus the h323 module with 
Grandstream's bt-100 IP phone. I want the asterisk to work as a SIP-proxy for 
these IP phones.
But, I am having trouble setting the /etc/asterisk/sip.conf file.
This is my file:

#############

;
; SIP Configuration for Asterisk
;
[general]
port = 5060   ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind to
;externip = 200.201.202.203 ; Address that we're going to put in SIP messages 
if we're behind a NAT
;localnet = 192.168.0.0         ; Internal NETWORK address
;localmask = 255.255.255.0      ; Internal netmask
context = default  ; Default for incoming calls
;srvlookup = yes  ; Enable SRV lookups on outbound calls
;pedantic = yes   ; Enable slow, pedantic checking for Pingtel
;tos=lowdelay
;tos=184
;maxexpirey=3600  ; Max length of incoming registration we allow
;defaultexpirey=120  ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes  ; Turn on support for SIP video
;disallow=all   ; Disallow all codecs
;allow=ulaw   ; Allow codecs in order of preference
;allow=ilbc
;
;register => 1234 at mysipprovider.com ; Register with a SIP provider
;register => 2345 at mysipprovider.com/1234 ; Register 2345 at sip provider as 
1234 here.
;

[243075]
type = friend
context = default
secret = gol
host = dynamic
callerid = fono75 <243075>

[243080]
type = friend
context = default
secret = gol
host = dynamic
callerid = fono80 <243080>


#############

and our SIP phones configuration are the following:

 SIP Server: 192.168.0.102

 Outbound Proxy:  <Empty>

SIP User ID:  243075

 Authenticate ID:  243075

 Authenticate Password:  gol

Name:  <Empty>

The IP phones can register to the proxy. But, when I try to dial (ie.:243080), 
The SIP-proxy answers with a 404 Not Found and I get a busy tone.

What I am doing wrong here?

Can someone that works with asterisk and these phones send me a sip.conf 
sample file, along with the scheme where it is set?

thanks in advance,
                              Pablo Salinas