Merci pour ton aide,j'apprend plein de chose
alors
jn-serveur*CLI> sip show peers
------------Code: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>
------------Code:* 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:
----------------
VoilaCode: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
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
Répondre avec citation
Envoyé par HClO
