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
extensions.confCode:[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
Voila,si vous avez besoin d'autres choses,merci de le faire savoir.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)
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