[Asterisk-Users] What does the error "stale nonce' mean?

Tue Oct 4 03:26:13 MST 2005

On 10/3/05, Morten Isaksen <misaksen at gmail.com> wrote:
>
>
> On 10/3/05, Olle E. Johansson <oej at edvina.net> wrote:
> >
> > > Does anyone know what "stale nonce" is?
> > I've answered this question many times, so you should be able to find
> > the answer...
> >
> > A stale nonce is when a device tries to re-authenticate with a nonce
> > that is no longer valid. We are telling them that the nonce they used is
> > invalid, and re-issue a new challenge and a fresh nonce. It's just an
> > informative message, that I propably should move away to a debug level
> > of some kind.
>
>   I get this error when I use a Audiocodes MP-124 against Asterisk
> 1.2beta1 and asterisk refuses the call. When I use
> CVS-D2005.02.12.14.37.11-04/13/05-16:14:03 it works fine.
>  I do not have access to the debug and log file now, but I will send them
> tomorrow.
>
 Here is the output from sip debug. I hope someone can explain what is
wrong.

<-- SIP read from 10.131.2.1:5060 <http://10.131.2.1:5060>:
INVITE sip:*2 at 10.131.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.2.1 <http://10.131.2.1>;branch=z9hG4bKaciipncbQ
Max-Forwards: 70
From: <sip:070001 at 10.131.0.1>;tag=1c1850211233
To: <sip:*2 at 10.131.0.1;user=phone>
Call-ID: 195411465Zwlj at 10.131.2.1
CSeq: 1 INVITE
Contact: <sip:070001 at 10.131.2.1>
Supported: em,100rel,timer,replaces,path
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006
Content-Type: application/sdp
Content-Length: 242

v=0
o=AudiocodesGW 644554 101011 IN IP4 10.131.2.1 <http://10.131.2.1>
s=Phone-Call
c=IN IP4 10.131.2.1 <http://10.131.2.1>
t=0 0
m=audio 6070 RTP/AVP 8 0 96
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv

--- (13 headers 12 lines)---
Using INVITE request as basis request - 195411465Zwlj at 10.131.2.1
Sending to 10.131.2.1 <http://10.131.2.1> : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.131.2.1:5060 <http://10.131.2.1:5060>:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.131.2.1 <http://10.131.2.1>;branch=z9hG4bKaciipncbQ
From: <sip:070001 at 10.131.0.1>;tag=1c1850211233
To: <sip:*2 at 10.131.0.1;user=phone>;tag=as6a339401
Call-ID: 195411465Zwlj at 10.131.2.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:*2 at 10.131.0.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="22a96479"
Content-Length: 0


---
Scheduling destruction of call '195411465Zwlj at 10.131.2.1' in 15000 ms
Found user '070001'
localhost*CLI>
<-- SIP read from 10.131.2.1:5060 <http://10.131.2.1:5060>:
ACK sip:*2 at 10.131.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.2.1 <http://10.131.2.1>;branch=z9hG4bKaciipncbQ
Max-Forwards: 70
From: <sip:070001 at 10.131.0.1>;tag=1c1850211233
To: <sip:*2 at 10.131.0.1;user=phone>;tag=as6a339401
Call-ID: 195411465Zwlj at 10.131.2.1
CSeq: 1 ACK
Contact: <sip:070001 at 10.131.2.1>
Supported: em,timer,replaces,path
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006
Content-Length: 0


--- (12 headers 0 lines)---
localhost*CLI>
<-- SIP read from 10.131.2.1:5060 <http://10.131.2.1:5060>:
INVITE sip:*2 at 10.131.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.2.1 <http://10.131.2.1>;branch=z9hG4bKaclMBIpvu
Max-Forwards: 70
From: <sip:070001 at 10.131.0.1>;tag=1c1850211233
To: <sip:*2 at 10.131.0.1;user=phone>
Call-ID: 195411465Zwlj at 10.131.2.1
CSeq: 2 INVITE
Proxy-Authorization: Digest
username="070001",realm="asterisk",nonce="22a96479" ",uri="sip:*2 at 10.131.0.1
",algorithm=MD5,response="41cc6e74fc333e770fa28a7db158a495"
Contact: <sip:070001 at 10.131.2.1>
Supported: em,100rel,timer,replaces,path
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006
Content-Type: application/sdp
Content-Length: 242

v=0
o=AudiocodesGW 644554 101011 IN IP4 10.131.2.1 <http://10.131.2.1>
s=Phone-Call
c=IN IP4 10.131.2.1 <http://10.131.2.1>
t=0 0
m=audio 6070 RTP/AVP 8 0 96
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv

--- (14 headers 12 lines)---
Using INVITE request as basis request - 195411465Zwlj at 10.131.2.1
Sending to 10.131.2.1 <http://10.131.2.1> : 5060 (non-NAT)
Oct 4 13:20:51 NOTICE[4078]: chan_sip.c:5710 check_auth: stale nonce
received from '<sip:*2 at 10.131.0.1;user=phone>'
Reliably Transmitting (no NAT) to 10.131.2.1:5060 <http://10.131.2.1:5060>:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.131.2.1 <http://10.131.2.1>;branch=z9hG4bKaclMBIpvu
From: <sip:070001 at 10.131.0.1>;tag=1c1850211233
To: <sip:*2 at 10.131.0.1;user=phone>;tag=as6a339401
Call-ID: 195411465Zwlj at 10.131.2.1
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:*2 at 10.131.0.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="0e317db4"
Content-Length: 0


---
Scheduling destruction of call '195411465Zwlj at 10.131.2.1' in 15000 ms
Found user '070001'
localhost*CLI>
<-- SIP read from 10.131.2.1:5060 <http://10.131.2.1:5060>:
ACK sip:*2 at 10.131.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.2.1 <http://10.131.2.1>;branch=z9hG4bKaclMBIpvu
Max-Forwards: 70
From: <sip:070001 at 10.131.0.1>;tag=1c1850211233
To: <sip:*2 at 10.131.0.1;user=phone>;tag=as6a339401
Call-ID: 195411465Zwlj at 10.131.2.1
CSeq: 2 ACK
Contact: <sip:070001 at 10.131.2.1>
Supported: em,timer,replaces,path
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006
Content-Length: 0


--- (12 headers 0 lines)---


--
Morten Isaksen
http://www.misak.dk/blog/
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