[asterisk-users] Asterisk does not accepts SIP registration
Tarek Sawah
tareksawah at hotmail.com
Tue Oct 25 07:53:33 CDT 2011
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: panych.y at gmail.com
> Date: Tue, 25 Oct 2011 14:30:53 +0300
> To: asterisk-users at lists.digium.com
> 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:L6 at 172.30.8.13:5060>;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:L6 at 172.30.8.13:5060>;tag=31b9dc9e-684902
> t:"L6" <sip:L6 at 172.30.8.13:5060>
> i:BD2F-1923-466848179B9BEAA6258E-001 at SipHost
> CSeq:23 REGISTER
> m:<sip:L6 at 172.30.8.18:5060>
> 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:L6 at 172.30.8.13:5060>;tag=31b9dc9e-684902
> To: "L6" <sip:L6 at 172.30.8.13:5060>;tag=as1a9dabcb
> Call-ID: BD2F-1923-466848179B9BEAA6258E-001 at 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:L6 at 172.30.8.13:5060>;tag=31b9dc9e-684902
> t:"L6" <sip:L6 at 172.30.8.13:5060>
> i:BD2F-1923-466848179B9BEAA6258E-001 at SipHost
> CSeq:24 REGISTER
> m:<sip:L6 at 172.30.8.18:5060>
> 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:L6 at 172.30.8.13:5060>;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:L6 at 172.30.8.13:5060>;tag=31b9dc9e-684902
> To: "L6" <sip:L6 at 172.30.8.13:5060>;tag=as014cd348
> Call-ID: BD2F-1923-466848179B9BEAA6258E-001 at 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.
>
> --
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