Hello Always returns 401 Unauthorized, because of [Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on stale nonce received from '"L6" <sip:[email protected]:5060>;tag=31b9dc9e-684902'
L6 is realtime device of type FRIEND (DLINK DVG7022S) Reviewed SIP conversation - no results. SIP debug <--- SIP read from UDP:172.30.8.18:5060 ---> REGISTER sip:172.30.8.13:5060 SIP/2.0 v:SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bKe3a0c285855ae0e5 f:"L6" <sip:[email protected]:5060>;tag=31b9dc9e-684902 t:"L6" <sip:[email protected]:5060> i:BD2F-1923-466848179B9BEAA6258E-001@SipHost CSeq:23 REGISTER m:<sip:[email protected]:5060> Expires:0 Max-Forwards:70 User-Agent:dlink 12-36-9924913 l:0 <-------------> <--- Transmitting (no NAT) to 172.30.8.18:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bKe3a0c285855ae0e5;received=172.30.8.18 From: "L6" <sip:[email protected]:5060>;tag=31b9dc9e-684902 To: "L6" <sip:[email protected]:5060>;tag=as1a9dabcb Call-ID: BD2F-1923-466848179B9BEAA6258E-001@SipHost CSeq: 23 REGISTER Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1555e540" Content-Length: 0 <------------> REGISTER sip:172.30.8.13:5060 SIP/2.0 v:SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bK629009e8cee7f7e3 f:"L6" <sip:[email protected]:5060>;tag=31b9dc9e-684902 t:"L6" <sip:[email protected]:5060> i:BD2F-1923-466848179B9BEAA6258E-001@SipHost CSeq:24 REGISTER m:<sip:[email protected]:5060> Expires:0 Max-Forwards:70 Authorization:Digest username="L6",realm="asterisk",nonce="1555e540",uri="sip:172.30.8.13:5060",response="f47b23120619eb9e6d184bafa48b92c9",algorithm=MD5 User-Agent:dlink 12-36-9924913 l:0 <-------------> [Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on stale nonce received from '"L6" <sip:[email protected]:5060>;tag=31b9dc9e-684902' [Oct 25 11:59:48] VERBOSE[2501] chan_sip.c: <--- Transmitting (no NAT) to 172.30.8.18:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bK629009e8cee7f7e3;received=172.30.8.18 From: "L6" <sip:[email protected]:5060>;tag=31b9dc9e-684902 To: "L6" <sip:[email protected]:5060>;tag=as014cd348 Call-ID: BD2F-1923-466848179B9BEAA6258E-001@SipHost CSeq: 24 REGISTER Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="11195a41", stale=true Content-Length: 0 <------------> sip.conf [general] context = default allowguest = no bindport = 5060 bindaddr = 0.0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726nonstandard = no notifyringing = yes srvlookup = yes t1min = 100 t38pt_udptl = no ;tos_audio = none ;tos_sip = none ;tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = alaw type = friend host=dynamic context = noop-context dtmfmode=rfc2833 ;language = ru ;sipdebug=yes nat=no rtcachefriends=yes qualify=10000 deny=0.0.0.0/0.0.0.0 permit=172.30.8.0/255.255.255.0 sip show settings Global Settings: ---------------- UDP Bindaddress: 0.0.0.0:5060 TCP SIP Bindaddress: Disabled TLS SIP Bindaddress: Disabled Videosupport: No Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: No Match Auth Username: No Allow unknown access: No Allow subscriptions: Yes Allow overlap dialing: Yes Allow promisc. redir: No Enable call counters: No SIP domain support: Yes Realm. auth: No Our auth realm asterisk Use domains as realms: No Call to non-local dom.: No URI user is phone no: No Always auth rejects: No Direct RTP setup: No User Agent: Asterisk PBX SDP Session Name: Asterisk PBX 1.8.5.0 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify: No Legacy userfield parse: 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: Enabled Qualify Freq : 60000 ms Q.850 Reason header: No 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 Network Settings: --------------------------- SIP address remapping: Disabled, no localnet list Externhost: <none> externaddr: (null) Externrefresh: 10 Global Signalling Settings: --------------------------- Codecs: 0x8 (alaw) Codec Order: alaw:20 Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: 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: Yes 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 Max forwards: 70 Default Settings: ----------------- Allowed transports: UDP Outbound transport: UDP Context: noop-context Force rport: No DTMF: rfc2833 Qualify: 10000 Use ClientCode: No Progress inband: Never Language: MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk Realtime SIP Settings: ---------------------- Realtime Peers: Yes Realtime Regs: No Cache Friends: Yes Update: Yes Ignore Reg. Expire: No Save sys. name: No Auto Clear: 120 (Disabled) ---- When registering soft SIP client - all okay. What I'm doing wrong? regards, Yaroslav. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
