[Asterisk-Users] help to configure sip server asterisk

[serge perreard]

hi everybody
I'm a new Asterisker.
I have a very simple configuration : 1 Sip proxy and 2 grandstream 102 in 
ethernet with
private adress
sip proxy : 192.168.2.194
ip phone address : 192.168.2.144
192.168.2.195
I want to make a communication between 2 ip phone with the SIP proxy but i 
have 2 different problems :

1 -> a grandstream phone (192.168.2.144 ) can't register
with this error :
192.168.2.144 -->192.168.2.194 register
192.168.2.194-->192.168.2.144 100 trying
192.168.2.194-->192.168.2.144 401 unauthorized

Registration for 'grandstream1 at 192.168.2.144' timed out, trying again

2 -> I can ring the phone with the sip proxy but phone can't make a phone 
between us and phone can't call the proxy.
invite
484 address incomplete

someone can help me to find the problem, please
I join my config :
---------------sip.conf--------------------
[general]
;----------- general setup
port = 5060
bindaddr = 192.168.2.194
tos = none
;----------- codecs setup
disallow = all
allow = ulaw ;autorise PCMU
allow = alaw ;autorise PCMA
allow = ilbc ;autorise ILBC
;----------- other options

;------------NETWORK
;localnet = 192.168.2.0
fromdomain = 192.168.2.1
;-----------CONTEXT
context = from-sip-external
;context = from-sip-internal
;context = default

maxexpirey = 3600
srvlookup = yes
nat = no
;promiscredir = no
;useragent = Asterisk PBX
defaultexpirey = 120
;trustrpid = no
;musicclass = default


[grandstream1]
type = friend
username = grandstream1
accountcode = grandstream1
dtmfmode = info
host =dynamic
defaultip = 192.168.2.144
port = 5061
secret = monpassword
context = from-sip-internal
canreinvite = yes
nat = no
reinvite = no
qualify = yes
;rtnoupdate=no

[grandstream2]
type = friend
username = grandstream2
accountcode = grandstream2
;callerid=<101>
dtmfmode = info
secret = password
host = dynamic
defaultip = 192.168.2.195
context = from-sip-internal
port = 5060
auth=md5
canreinvite = no
nat = no
reinvite = no
qualify = yes


------------extensions.conf-------------
[general]
static=yes
writeprotect=yes

;------------------------
[local]
exten => 100,1,Dial(SIP/100 at grandstream1,10)
exten => 101,1,Dial(SIP/101 at grandstream2,10)
;exten => 2,1,Dial(SIP/2/@192.168.2.194,10)
;-------------------------
[from-sip-internal]
include => local
;------------------------

rudy73

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