[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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