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

[pesb]

Hi there,
I am still trying to make the asterisk SIP proxy server work with my 
Grandstream 100 IP phones.
I tried Stephen advice and it did not work. I stil got the 404 error message.
So, rigth now, I am trying the following configuration(sip.conf):

###########################
;
; 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
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
;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.
;
;[snomsip]
;type=friend
;secret=blah
;host=dynamic
;dtmfmode=inband  ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59
;mailbox=1234,2345  ; Mailbox for message waiting indicator
;restrictcid=yes  ; To have the callerid restriced -> sent as ANI

;[pingtel]
;type=friend
;username=pingtel
;secret=blah
;host=dynamic
;qualify=1000   ; Consider it down if it's 1 second to reply
;callgroup=1,3-4
;pickupgroup=1,3-4
;defaultip=192.168.0.60

;[cisco]
;type=friend
;username=cisco
;secret=blah
;nat=yes   ; This phone may be natted
;host=dynamic
;canreinvite=no   ; Cisco poops on reinvite sometimes
;qualify=200   ; Qualify peer is no more than 200ms away
;defaultip=192.168.0.4

;[cisco1]
;type=friend
;username=cisco1
;fromuser=markster  ; Specify user to put in "from" instead of callerid
;secret=blah
;host=dynamic
;defaultip=192.168.0.4
;amaflags=default  ; Choices are default, omit, billing, documentation
;accountcode=markster  ; Users may be associated with an accountcode tp ease 
billing


[1001]
type = friend
context = default
secret = gol
host = dynamic
callerid = "STREAM-1001" <1001>
;dtfmmode=inband
canreinvite=no
defaultip=192.168.0.105


[1002]
type = friend
context = default
secret = gol
host = dynamic
callerid = "STREAM-1002" <1002>
;dtfmmode=inband
canreinvite=no
defaultip=192.168.0.104
##############################

This is the configuration of my SIP-phones:


ipaddr=192.168.0.105
sipserver=192.168.0.102
sipserver_port=5060
outboundproxy=null
outboundproxy_port=null
userid=1001
authenticateid=1001
codec1=PCMU
codec2=PCMA
codec3=G723
codec4=G729
codec5=null
codec6=null
silence_supporession=no
voice_frames_per_tx=2
ipqos=48
vlantag=0
registration_expiration=10
local_sip_port=5060
local_rtp_port=5004
use_random_rtp_port=no
send_dtmf=in-audio
dtmf_payload_type=101
time_zone=GMT-0

ipaddr=192.168.0.104
sipserver=192.168.0.102
sipserver_port=5060
outboundproxy=null
outboundproxy_port=null
userid=1004
authenticateid=1004
codec1=PCMU
codec2=PCMA
codec3=G723
codec4=G729
codec5=null
codec6=null
silence_supporession=no
voice_frames_per_tx=2
ipqos=48
vlantag=0
registration_expiration=10
local_sip_port=5060
local_rtp_port=5004
use_random_rtp_port=no
send_dtmf=in-audio
dtmf_payload_type=101
time_zone=GMT-0


What's wrong here?? 

When I try to dial from one phone to the other, I get 404 error message.

Please, somebody help me.