Stale nonce and dropped registrations

Asterisk Asterisk Support
1

I have about 20 Aastra 9480i phones connected to Asterisk 1.6.1.1. I’m regularly seeing messages like these in the logs:

Correct auth, but based on stale nonce received from ‘sip:134@172.27.7.6:5060’

Typically, this doesn’t seem to create any problems. However, these phones seem to lose their registration about once a day or so (they go into a “No Service” display on the phone and then, a minute or two later, reregister fine). I’m not sure if this is related or not to the “stale nonce” issue. I’m also seeing some Registration Unauthorized messages too while the phones themselves are registered (I’m assuming these are re-register requests?):

<— SIP read from UDP://192.168.2.236:5060 —>
REGISTER sip:172.27.7.6:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.236:5060;branch=z9hG4bK14b1f713f12371077.6cef217f54542d463
Max-Forwards: 70
From: sip:562@172.27.7.6:5060;tag=41771d98c5
To: sip:562@172.27.7.6:5060
Call-ID: 69b72ca629bd37fd
CSeq: 27736 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: “Test User” sip:562@192.168.2.236:5060;transport=udp;+sip.instance="urn:uuid:00000000-0000-1000-8000-00085D21D8DA"
Supported: gruu, path
User-Agent: Aastra 9480i/2.5.3.18
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.2.236 : 5060 (no NAT)

<— Transmitting (NAT) to 192.168.2.236:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.236:5060;branch=z9hG4bK14b1f713f12371077.6cef217f54542d463;received=192.168.2.236
From: sip:562@172.27.7.6:5060;tag=41771d98c5
To: sip:562@172.27.7.6:5060;tag=as3d32ce64
Call-ID: 69b72ca629bd37fd
CSeq: 27736 REGISTER
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2e8f01d1"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘69b72ca629bd37fd’ in 32000 ms (Method: REGISTER)
social*CLI>
<— SIP read from UDP://192.168.2.236:5060 —>
REGISTER sip:172.27.7.6:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.236:5060;branch=z9hG4bKcf3478ae611a82361.aff33c09e9fbc736e
Max-Forwards: 70
From: sip:562@172.27.7.6:5060;tag=41771d98c5
To: sip:562@172.27.7.6:5060
Call-ID: 69b72ca629bd37fd
CSeq: 27737 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username=“562”,realm=“asterisk”,nonce=“2e8f01d1”,uri=“sip:172.27.7.6:5060”,response=“6587b203b40a2a17b6fd1647bcae309d”,algorithm=MD5
Contact: “Test User” sip:562@192.168.2.236:5060;transport=udp;+sip.instance="urn:uuid:00000000-0000-1000-8000-00085D21D8DA"
Supported: gruu, path
User-Agent: Aastra 9480i/2.5.3.18
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.2.236 : 5060 (NAT)
– Registered SIP ‘562’ at 192.168.2.236 port 5060
Reliably Transmitting (NAT) to 192.168.2.236:5060:
OPTIONS sip:562@192.168.2.236:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.27.7.6:5060;branch=z9hG4bK7a37964a;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@172.27.7.6;tag=as1f9468a4
To: sip:562@192.168.2.236:5060;transport=udp
Contact: sip:asterisk@172.27.7.6
Call-ID: 4715f07b2a5a9d0927ea03c94e8bbec4@172.27.7.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.1.6
Date: Fri, 19 Mar 2010 14:32:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<— Transmitting (NAT) to 192.168.2.236:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.236:5060;branch=z9hG4bKcf3478ae611a82361.aff33c09e9fbc736e;received=192.168.2.236
From: sip:562@172.27.7.6:5060;tag=41771d98c5
To: sip:562@172.27.7.6:5060;tag=as3d32ce64
Call-ID: 69b72ca629bd37fd
CSeq: 27737 REGISTER
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: sip:562@192.168.2.236:5060;transport=udp;expires=120
Date: Fri, 19 Mar 2010 14:32:46 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘69b72ca629bd37fd’ in 32000 ms (Method: REGISTER)
social*CLI>
<— SIP read from UDP://192.168.2.236:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.27.7.6:5060;branch=z9hG4bK7a37964a;rport=5060;received=172.27.7.6
From: “asterisk” sip:asterisk@172.27.7.6;tag=as1f9468a4
To: sip:562@192.168.2.236:5060;transport=udp;tag=835688800
Call-ID: 4715f07b2a5a9d0927ea03c94e8bbec4@172.27.7.6
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 9480i/2.5.3.18
Supported: gruu, timer, 100rel, replaces, path
Content-Length: 0

<------------->
— (10 headers 0 lines) —
[Mar 19 10:32:46] NOTICE[1826]: chan_sip.c:16784 handle_response_peerpoke: Peer ‘562’ is now Reachable. (106ms / 2000ms)
Really destroying SIP dialog ‘4715f07b2a5a9d0927ea03c94e8bbec4@172.27.7.6’ Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.2.236:5060:
OPTIONS sip:562@192.168.2.236:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.27.7.6:5060;branch=z9hG4bK48727a4f;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@172.27.7.6;tag=as736008c3
To: sip:562@192.168.2.236:5060;transport=udp
Contact: sip:asterisk@172.27.7.6
Call-ID: 5a9e508f2617a133664e958f1ea5fc47@172.27.7.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.1.6
Date: Fri, 19 Mar 2010 14:33:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP://192.168.2.236:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.27.7.6:5060;branch=z9hG4bK48727a4f;rport=5060;received=172.27.7.6
From: “asterisk” sip:asterisk@172.27.7.6;tag=as736008c3
To: sip:562@192.168.2.236:5060;transport=udp;tag=2056812882
Call-ID: 5a9e508f2617a133664e958f1ea5fc47@172.27.7.6
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 9480i/2.5.3.18
Supported: gruu, timer, 100rel, replaces, path
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘5a9e508f2617a133664e958f1ea5fc47@172.27.7.6’ Method: OPTIONS

Here is how the phones are configured in sip.conf:

[562]
dtmfmode=rfc2833
type=friend
nat=yes
canreinvite=no
host=dynamic
videosupport=yes
qualify=yes
maxcallbitrate=384
context=localcontext

Also, on the Aastra phones, Registration Period is set to 0, Registration Failed Retry Timer is set to 1800, Registration Timeout Retry Timer is set to 120, and Registration Renewal Timer is set to 15 (defaults).

Anyone have any ideas on this?

2

Hello,

We had the same problem (No Service) for Aastra 57xi phones and we used the following workaround:

  • Registration Period = 0 (in Aastra -> Global SIP)
  • qualify = no (in sip.conf for that extensions)

Is this working for you?

If no - have you the last firmware on the Aastra phones?

HTH,
Ioan.

3

The registration period value is what we have set. I’ll try qualify=no to see if it fixes the “No service” issue, but I know it doesn’t address the nonce issue or the Unauthorized login issue.