Hello, Is L6 a remote device? is there any firewall residing between the server and UA?
Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 > From: [email protected] > Date: Tue, 25 Oct 2011 14:30:53 +0300 > To: [email protected] > Subject: [asterisk-users] Asterisk does not accepts SIP registration > > 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
-- _____________________________________________________________________ -- 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
