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Yaroslav Panych

2011-Oct-25 11:30 UTC

[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.

Tarek Sawah

2011-Oct-25 12:53 UTC

head link

[asterisk-users] Asterisk does not accepts SIP registration

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.
> 
> --
> _____________________________________________________________________
> -- 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
> 
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Administrator TOOTAI

2011-Oct-25 15:22 UTC

head link

[asterisk-users] Asterisk does not accepts SIP registration

Le 25/10/2011 13:30, Yaroslav Panych a ?crit :> 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"
Change the local port from the DLInk (eg 5060 to 15060) and it should 
work. After few hours you should be able to go set again initial value.

[...]

-- 
Daniel

asterisk users - Oct 2011 - Asterisk does not accepts SIP registration

[asterisk-users] Asterisk does not accepts SIP registration

[asterisk-users] Asterisk does not accepts SIP registration

[asterisk-users] Asterisk does not accepts SIP registration