Bonjour,

J'ai un gros problème depuis quelques jours je n'arrive pas à configurer mon serveur astérisk.

Je vous fourni le débug SIP :
sogebat*CLI> sip set debug peer ovh
SIP Debugging Enabled for IP: 91.121.129.17:5060
Audio is at 31.35.74.10 port 19718
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 91.121.129.17:5060:
INVITE sip:006XXXXXXX@sip.ovh.net SIP/2.0
Via: SIP/2.0/UDP 31.35.74.10:5060;branch=z9hG4bK571e19fe;rport
From: "poste3" <sip:0033484XXXXX@sip.ovh.net>;tag=as1dcf877f
To: <sip:006XXXXXXX@sip.ovh.net>
Contact: <sip:0033484XXXXX@31.35.74.10>
Call-ID: 1fb45ebf12c9fbf627aefcf645a1b404@sip.ovh.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 06 Feb 2012 11:14:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 3580 3580 IN IP4 31.35.74.10
s=session
c=IN IP4 31.35.74.10
t=0 0
m=audio 19718 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
sogebat*CLI>
<--- SIP read from 91.121.129.17:5060 --->
SIP/2.0 407 authentication required
Allow: UPDATE,REFER,INFO
Call-ID: 1fb45ebf12c9fbf627aefcf645a1b404@sip.ovh.net
Contact: <sip:006XXXXXXX@91.121.129.17:5060;user=phone>
CSeq: 102 INVITE
From: "poste3" <sip:0033484XXXXX@sip.ovh.net>;tag=as1dcf877f
Proxy-Authenticate: Digest realm="sip.ovh.net",nonce="087810e40230830212aeb38 e385d1d05",opaque="086a62bd0df8de0",stale=false,al gorithm=MD5
Server: Cirpack/v4.42a (gw_sip)
To: <sip:006XXXXXXX@sip.ovh.net>;tag=00-08024-08781df5-34ef51d17
Via: SIP/2.0/UDP 31.35.74.10:5060;received=31.35.74.10;rport=5060;b ranch=z9hG4bK571e19fe
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 91.121.129.17:5060:
ACK sip:006XXXXXXX@sip.ovh.net SIP/2.0
Via: SIP/2.0/UDP 31.35.74.10:5060;branch=z9hG4bK571e19fe;rport
From: "poste3" <sip:0033484XXXXX@sip.ovh.net>;tag=as1dcf877f
To: <sip:006XXXXXXX@sip.ovh.net>;tag=00-08024-08781df5-34ef51d17
Contact: <sip:0033484XXXXX@31.35.74.10>
Call-ID: 1fb45ebf12c9fbf627aefcf645a1b404@sip.ovh.net
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Feb 6 11:14:32] NOTICE[3634]: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to '"poste3" <sip:0033484XXXXX@sip.ovh.net>;tag=as1dcf877f'
Really destroying SIP dialog '1fb45ebf12c9fbf627aefcf645a1b404@sip.ovh.net' Method: INVITE
Ma configuration :
BBOX ==> serveur Asterisk (configurer en 192.168.36.X) ==> 5 adaptateur PAP2T avec 2 téléphone sur chaque sorti en RJ11

Ma configuration SIP.conf :
[general]
language=fr
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no
defaultexpiry=3600
registertimeout=30
registerattempts=0
disallow=all
allow=alaw
allowguest=yes
nat=yes
externip=31.35.XXX.XX
localnet=192.168.36.0/255.255.255.0

register => 0033484XXXXXX:*******@sip.ovh.net/901

[ovh]
type=peer
host=sip.ovh.net
fromuser=0033484XXXXXX
fromdomain=sip.ovh.net
secret=XXXXXXX
canrenvite=no
insecure=invite
;qualify=yes
nat=yes
context=outside
;dtmfmode=inband


[poste1]
type=friend
username=poste1
host=dynamic
context=inside

[poste2]
type=friend
username=poste2
host=dynamic
context=inside

[poste3]
type=friend
username=poste3
host=dynamic
context=inside

[poste4]
type=friend
username=poste4
host=dynamic
context=inside

[poste5]
type=friend
username=poste5
host=dynamic
context=inside

[poste6]
type=friend
username=poste6
host=dynamic
context=inside

[poste7]
type=friend
username=poste7
host=dynamic
context=inside

[poste8]
type=friend
username=poste8
host=dynamic
context=inside

[poste9]
type=friend
username=poste9
host=dynamic
context=inside

[poste10]
type=friend
username=poste10
host=dynamic
context=inside

[poste11]
type=friend
username=poste11
host=dynamic
context=inside
et le Extention.conf
[outside]

; Ovh
exten => _90X,1,Set(ligne=${EXTEN})
exten => _90X,n,Goto(910,1)

exten => 910,1,Dial(SIP/poste1&SIP/poste2&SIP/poste3&SIP/poste4&SIP/poste5&SIP/poste6&SIP/poste7&SIP/poste8&SIP/poste9&SIP/poste10,,A(custom/ligne${ligne}))

exten => 911,1,Ringing
exten => 911,n,Wait(30)
exten => 911,n,Answer
exten => 911,n,Wait(10)
exten => 911,n,Hangup

[inside]
;Include parkedcalls (or the context you define in features conf)
;to enable call parking.
;include => parkedcalls

exten => 501,1,Dial(SIP/poste1)
exten => 502,1,Dial(SIP/poste2)
exten => 503,1,Dial(SIP/poste3)
exten => 504,1,Dial(SIP/poste4)
exten => 505,1,Dial(SIP/poste5)
exten => 506,1,Dial(SIP/poste6)
exten => 507,1,Dial(SIP/poste7)
exten => 508,1,Dial(SIP/poste8)
exten => 509,1,Dial(SIP/poste9)
exten => 510,1,Dial(SIP/poste10)
exten => 511,1,Dial(SIP/poste11)

exten => 520,1,Dial(SIP/poste1&SIP/poste2&SIP/poste3&SIP/poste4&SIP/poste5&SIP/poste6&SIP/poste7&SIP/poste8&SIP/poste9&SIP/poste10)

exten => 560,1,Ringing
exten => 560,n,WaitExten(1)
exten => 560,n,Answer
exten => 560,n,Echo
exten => 561,1,Dial(SIP/123@ovh)
exten => 562,1,Dial(SIP/900@ovh)

exten => _X.,1,Dial(SIP/0${EXTEN}@ovh)
Je n'arrive pas à comprendre d'ou vient le problème si vous pouvez m'aider.
Merci.