Salut et merci pour votre soutien.J'ai mis en place le serveur Asterisk pour mieux découvrir cette opensource .
A ce jour,je n'arrive pas à passer d’appels vers l’extérieur . J'utilise Xlite comme Client SIP sous Windows XP.
Par contre,lorsque je compose le 600 j'ai bien une réponse.
Voici la configuration de mes fichiers :
sip.conf
Code:
[general] 
context = asterisk        ; Default context for incoming calls 
allowguest = no        ; Allow or reject guest calls (default is yes, this can also be set to 'osp' 
realm=data4ict.com        ; Realm for digest authentication 
bindport = 5060         ; UDP Port to bind to (SIP standard port is 5060) 
bindaddr = 0.0.0.0        ; IP address to bind to (0.0.0.0 binds to all) 
srvlookup = yes          ; Enable DNS SRV lookups on outbound calls 
disallow = all        ; First disallow all codecs 
allow = ulaw        ; Allow codecs in order of preference 
allow = alaw 
allow = gsm 
dtmfmode = rfc2833        ; Set default dtmfmode for sending DTMF. 
canreinvite=no 
nat=yes 
videosupport = yes        ; Enable video 
allow = h263             ; H.263 is our video codec 
allow = h263p             ; H.263p is the enhanced video codec

register => mon_login:mon _mot_de_passe@voip.kiwak.net 

[authentication] 
[1001] 
type=friend 
context=asterisk 
username=1001 
secret=1001 
host=dynamic 
callerid="Phone1" 
language=fr

[kiwak] 
type=peer 
allow=all
host=voip.kiwak.net
secret=mon mot de passe
fromuser=mon login
username=mon login
fromdomain=kiwak.net
qualify=yes
extensions.conf
Code:
[general] 
; 
; If static is set to no, or omitted, then the pbx_config will rewrite 
; this file when extensions are modified.  Remember that all comments 
; made in the file will be lost when that happens. 
static=yes 
; 
; if static=yes and writeprotect=no, you can save dialplan by 
; CLI command 'save dialplan' too 
; 
writeprotect=yes 
; 
; If autofallthrough is set, then if an extension runs out of 
; things to do, it will terminate the call with BUSY, CONGESTION 
; or HANGUP depending on Asterisk's best guess (strongly recommended). 
; 
autofallthrough=yes 
; 
; If clearglobalvars is set, global variables will be cleared 
; and reparsed on an extensions reload, or Asterisk reload. 
; 
clearglobalvars=no 
; 
; If priorityjumping is set to 'yes', then applications that support 
; 'jumping' to a different priority based on the result of their operations 
; will do so (this is backwards compatible behavior with pre-1.2 releases 
; of Asterisk). Individual applications can also be requested to do this 
; by passing a 'j' option in their arguments. 
; 
priorityjumping=yes 
; 
;[globals] 
; 
[internal] 
exten => 1001,1,Dial(SIP/1001,20,Tr) 
exten => 1001,2,Hangup() 
exten => 1002,1,Dial(SIP/1002,20,Tr) 
exten => 1002,2,Hangup() 
[asterisk] 
include => internal 
; 
; Create an extension, 600, for evaluating echo latency. 
; 
exten => 600,1,Playback(demo-echotest)    ; Let them know what's going on 
exten => 600,2,Echo            ; Do the echo test 
exten => 600,3,Playback(demo-echodone)    ; Let them know it's over

[incoming] ; Context par défaut
exten => s,1,Dial(SIP/1000) 

[outgoing] ; Context sortant rattaché à votre compte SIP/IAX Asterisk (Ex : 1000)
exten => _X.,1,Dial(SIP/kiwak/$EXTEN)

exten => _0[123459]XXXXXXXX,1,Dial(SIP/${EXTEN}@kiwak_outbound,30,rT)
exten => _087XXXXXXX,1,Dial(SIP/${EXTEN}@kiwak_outbound,30,rT)
Voila,si vous avez besoin d'autres choses,merci de le faire savoir.