Probleme de configuration derniere un NAT

[HClO]

Merci pour ton aide,j'apprend plein de chose

alors

jn-serveur*CLI> sip show peers

Name/username              Host            Dyn Nat ACL Port     Status
Fixe                       (Unspecified)    D          5060     Unmonitored
HClO/HClO                  82.124.**.***   D   N      1024     Unmonitored
SPA/SPA                    192.168.1.**    D          5060     OK (9 ms)
fixe-voip/fixe-voip        82.124.**.**    D   N      5060     Unmonitored

------------
peer HClO

jn-serveur*CLI>

 * Name       : HClO
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : local
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : 3484
  Insecure     : no
  Nat          : Always
  ACL          : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : 82.124.42.*** Port 1024
  Defaddr->IP  : 0.0.0.0 Port 5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: HClO
  SIP Options  : (none)
  Codecs       : 0x90c (ulaw|alaw|g726|g729)
  Codec Order  : (alaw:20,ulaw:20,g726:20,g729:20)
  Auto-Framing :  No
  100 on REG   : No
  Status       : Unmonitored
  Useragent    : Ekiga/3.2.7
  Reg. Contact : sip:HClO@192.168.1.10
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  Parkinglot   :

------------

Global Settings:
----------------

  UDP SIP Port:           5060
  UDP Bindaddress:        0.0.0.0
  TCP SIP Port:           Disabled
  TLS SIP Port:           Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        No
  Match Auth Username:    No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promsic. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          asterisk
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    No
  Direct RTP setup:       No
  User Agent:             Asterisk PBX 1.6.2.5-0ubuntu1.3
  SDP Session Name:       Asterisk PBX 1.6.2.5-0ubuntu1.3
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          -1
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS0
  IP ToS RTP audio:       CS0
  IP ToS RTP video:       CS0
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No
  Jitterbuffer forced:    No
  Jitterbuffer max size:  -1
  Jitterbuffer resync:    -1
  Jitterbuffer impl:
  Jitterbuffer log:       No

Network Settings:
---------------------------
  SIP address remapping:  Enabled using externip
  Externhost:             <none>
  Externip:               78.226.**.**:5060
  Externrefresh:          10
  Internal IP:            192.168.1.100:5060
  Localnet:               192.168.1.0/255.255.255.0
                          192.168.1.0/255.255.255.0
  STUN server:            0.0.0.0:0

Global Signalling Settings:
---------------------------
  Codecs:                 0x90c (ulaw|alaw|g726|g729)
  Codec Order:            alaw:20,ulaw:20,g726:20,g729:20
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   No
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set>
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     Yes

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:     UDP
  Context:                local
  Nat:                    Always
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   asterisk

Voila

Sinon concernant la sécurité de mon serveur:
J'ai fais que les redirections de port nécessaire, et je me pencherais la dessus plus sérieusement quand tout marchera bien