[asterisk-users] Asterisk does not accepts SIP registration

Yaroslav Panych panych.y at gmail.com
Tue Oct 25 06:30:53 CDT 2011


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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