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Hello, <br>Is L6 a remote device? is there any firewall residing between the server and UA?<br><br><br>Tarek Sawah<br><br>Information Technology Adviser<br><br>Integrated Digital Systems<br><br>CCNP, MCSE, RHCE, TELECOM<br><br>USA: +1 386 492 9993<br><br><br><br><div>> From: panych.y@gmail.com<br>> Date: Tue, 25 Oct 2011 14:30:53 +0300<br>> To: asterisk-users@lists.digium.com<br>> Subject: [asterisk-users] Asterisk does not accepts SIP registration<br>> <br>> Hello<br>> <br>> Always returns 401 Unauthorized, because of<br>> [Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on<br>> stale nonce received from '"L6"<br>> <sip:L6@172.30.8.13:5060>;tag=31b9dc9e-684902'<br>> <br>> L6 is realtime device of type FRIEND (DLINK DVG7022S)<br>> <br>> Reviewed SIP conversation - no results.<br>> <br>> SIP debug<br>> <--- SIP read from UDP:172.30.8.18:5060 ---><br>> REGISTER sip:172.30.8.13:5060 SIP/2.0<br>> v:SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bKe3a0c285855ae0e5<br>> f:"L6" <sip:L6@172.30.8.13:5060>;tag=31b9dc9e-684902<br>> t:"L6" <sip:L6@172.30.8.13:5060><br>> i:BD2F-1923-466848179B9BEAA6258E-001@SipHost<br>> CSeq:23 REGISTER<br>> m:<sip:L6@172.30.8.18:5060><br>> Expires:0<br>> Max-Forwards:70<br>> User-Agent:dlink 12-36-9924913<br>> l:0<br>> <br>> <-------------><br>> <--- Transmitting (no NAT) to 172.30.8.18:5060 ---><br>> SIP/2.0 401 Unauthorized<br>> Via: SIP/2.0/UDP<br>> 172.30.8.18:5060;branch=z9hG4bKe3a0c285855ae0e5;received=172.30.8.18<br>> From: "L6" <sip:L6@172.30.8.13:5060>;tag=31b9dc9e-684902<br>> To: "L6" <sip:L6@172.30.8.13:5060>;tag=as1a9dabcb<br>> Call-ID: BD2F-1923-466848179B9BEAA6258E-001@SipHost<br>> CSeq: 23 REGISTER<br>> Server: Asterisk PBX<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,<br>> INFO, PUBLISH<br>> Supported: replaces, timer<br>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1555e540"<br>> Content-Length: 0<br>> <br>> <br>> <------------><br>> REGISTER sip:172.30.8.13:5060 SIP/2.0<br>> v:SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bK629009e8cee7f7e3<br>> f:"L6" <sip:L6@172.30.8.13:5060>;tag=31b9dc9e-684902<br>> t:"L6" <sip:L6@172.30.8.13:5060><br>> i:BD2F-1923-466848179B9BEAA6258E-001@SipHost<br>> CSeq:24 REGISTER<br>> m:<sip:L6@172.30.8.18:5060><br>> Expires:0<br>> Max-Forwards:70<br>> Authorization:Digest<br>> username="L6",realm="asterisk",nonce="1555e540",uri="sip:172.30.8.13:5060",response="f47b23120619eb9e6d184bafa48b92c9",algorithm=MD5<br>> User-Agent:dlink 12-36-9924913<br>> l:0<br>> <br>> <-------------><br>> [Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on<br>> stale nonce received from '"L6"<br>> <sip:L6@172.30.8.13:5060>;tag=31b9dc9e-684902'<br>> [Oct 25 11:59:48] VERBOSE[2501] chan_sip.c:<br>> <--- Transmitting (no NAT) to 172.30.8.18:5060 ---><br>> SIP/2.0 401 Unauthorized<br>> Via: SIP/2.0/UDP<br>> 172.30.8.18:5060;branch=z9hG4bK629009e8cee7f7e3;received=172.30.8.18<br>> From: "L6" <sip:L6@172.30.8.13:5060>;tag=31b9dc9e-684902<br>> To: "L6" <sip:L6@172.30.8.13:5060>;tag=as014cd348<br>> Call-ID: BD2F-1923-466848179B9BEAA6258E-001@SipHost<br>> CSeq: 24 REGISTER<br>> Server: Asterisk PBX<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,<br>> INFO, PUBLISH<br>> Supported: replaces, timer<br>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",<br>> nonce="11195a41", stale=true<br>> Content-Length: 0<br>> <br>> <br>> <------------><br>> <br>> sip.conf<br>> [general]<br>> context = default<br>> <br>> allowguest = no<br>> bindport = 5060<br>> bindaddr = 0.0.0.0<br>> <br>> allowexternaldomains = no<br>> allowoverlap = yes<br>> allowsubscribe = yes<br>> allowtransfer = yes<br>> alwaysauthreject = no<br>> autodomain = no<br>> callevents = no<br>> canreinvite = no<br>> checkmwi = 10<br>> compactheaders = no<br>> defaultexpiry = 120<br>> domain=sop-korniychuk<br>> domain=172.30.8.13<br>> domain=172.30.8.13:5060<br>> dumphistory = no<br>> externrefresh = 10<br>> g726nonstandard = no<br>> notifyringing = yes<br>> srvlookup = yes<br>> t1min = 100<br>> t38pt_udptl = no<br>> ;tos_audio = none<br>> ;tos_sip = none<br>> ;tos_video = none<br>> trustrpid = no<br>> useragent = Asterisk PBX<br>> usereqphone = no<br>> videosupport = no<br>> disallow = all<br>> allow = alaw<br>> type = friend<br>> host=dynamic<br>> context = noop-context<br>> dtmfmode=rfc2833<br>> ;language = ru<br>> ;sipdebug=yes<br>> nat=no<br>> rtcachefriends=yes<br>> qualify=10000<br>> deny=0.0.0.0/0.0.0.0<br>> permit=172.30.8.0/255.255.255.0<br>> <br>> sip show settings<br>> <br>> Global Settings:<br>> ----------------<br>> UDP Bindaddress: 0.0.0.0:5060<br>> TCP SIP Bindaddress: Disabled<br>> TLS SIP Bindaddress: Disabled<br>> Videosupport: No<br>> Textsupport: No<br>> Ignore SDP sess. ver.: No<br>> AutoCreate Peer: No<br>> Match Auth Username: No<br>> Allow unknown access: No<br>> Allow subscriptions: Yes<br>> Allow overlap dialing: Yes<br>> Allow promisc. redir: No<br>> Enable call counters: No<br>> SIP domain support: Yes<br>> Realm. auth: No<br>> Our auth realm asterisk<br>> Use domains as realms: No<br>> Call to non-local dom.: No<br>> URI user is phone no: No<br>> Always auth rejects: No<br>> Direct RTP setup: No<br>> User Agent: Asterisk PBX<br>> SDP Session Name: Asterisk PBX 1.8.5.0<br>> SDP Owner Name: root<br>> Reg. context: (not set)<br>> Regexten on Qualify: No<br>> Legacy userfield parse: No<br>> Caller ID: asterisk<br>> From: Domain:<br>> Record SIP history: Off<br>> Call Events: Off<br>> Auth. Failure Events: Off<br>> T.38 support: No<br>> T.38 EC mode: Unknown<br>> T.38 MaxDtgrm: -1<br>> SIP realtime: Enabled<br>> Qualify Freq : 60000 ms<br>> Q.850 Reason header: No<br>> <br>> Network QoS Settings:<br>> ---------------------------<br>> IP ToS SIP: CS0<br>> IP ToS RTP audio: CS0<br>> IP ToS RTP video: CS0<br>> IP ToS RTP text: CS0<br>> 802.1p CoS SIP: 4<br>> 802.1p CoS RTP audio: 5<br>> 802.1p CoS RTP video: 6<br>> 802.1p CoS RTP text: 5<br>> Jitterbuffer enabled: No<br>> <br>> Network Settings:<br>> ---------------------------<br>> SIP address remapping: Disabled, no localnet list<br>> Externhost: <none><br>> externaddr: (null)<br>> Externrefresh: 10<br>> <br>> Global Signalling Settings:<br>> ---------------------------<br>> Codecs: 0x8 (alaw)<br>> Codec Order: alaw:20<br>> Relax DTMF: No<br>> RFC2833 Compensation: No<br>> Symmetric RTP: No<br>> Compact SIP headers: No<br>> RTP Keepalive: 0 (Disabled)<br>> RTP Timeout: 0 (Disabled)<br>> RTP Hold Timeout: 0 (Disabled)<br>> MWI NOTIFY mime type: application/simple-message-summary<br>> DNS SRV lookup: Yes<br>> Pedantic SIP support: Yes<br>> Reg. min duration 60 secs<br>> Reg. max duration: 3600 secs<br>> Reg. default duration: 120 secs<br>> Outbound reg. timeout: 20 secs<br>> Outbound reg. attempts: 0<br>> Notify ringing state: Yes<br>> Include CID: No<br>> Notify hold state: No<br>> SIP Transfer mode: open<br>> Max Call Bitrate: 384 kbps<br>> Auto-Framing: No<br>> Outb. proxy: <not set><br>> Session Timers: Accept<br>> Session Refresher: uas<br>> Session Expires: 1800 secs<br>> Session Min-SE: 90 secs<br>> Timer T1: 500<br>> Timer T1 minimum: 100<br>> Timer B: 32000<br>> No premature media: Yes<br>> Max forwards: 70<br>> <br>> Default Settings:<br>> -----------------<br>> Allowed transports: UDP<br>> Outbound transport: UDP<br>> Context: noop-context<br>> Force rport: No<br>> DTMF: rfc2833<br>> Qualify: 10000<br>> Use ClientCode: No<br>> Progress inband: Never<br>> Language:<br>> MOH Interpret: default<br>> MOH Suggest:<br>> Voice Mail Extension: asterisk<br>> <br>> Realtime SIP Settings:<br>> ----------------------<br>> Realtime Peers: Yes<br>> Realtime Regs: No<br>> Cache Friends: Yes<br>> Update: Yes<br>> Ignore Reg. Expire: No<br>> Save sys. name: No<br>> Auto Clear: 120 (Disabled)<br>> <br>> ----<br>> <br>> <br>> When registering soft SIP client - all okay.<br>> What I'm doing wrong?<br>> <br>> regards, Yaroslav.<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div> </div></body>
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