Cisco cant register

Hello,

let me explain my situation.
Two days ago, I decided to move the asterisk to the another computer, so I downloaded the latest 1.4.26.1 relase, installed it and copied configuration files from old 1.4.15.
The problem is, Cisco 7940 and 7960 phones (SIP Firmware 8.11) suddenly cant register to Asterisk, other phones can, softphones can too.
I thought, its some incompatibility issue or bug, so i downgraded to 1.4.15.
But Cisco phones still cant register. I even tried to downgrade firmware in the phones down to 8.2.
Strange thing is, it is working on the original asterisk box.
The only difference in software is architecture (original is x86, new is x86_64) and kernel version (old 2.6.22, new 2.6.29), both gentoo, both 1.4.15, same configuration.
Phones and asterisk are on the same subnet.
Below is debug log from old and new install.
Is here someone with similar configuration and working Cisco phones?
Im really desperate for any help.

Thank you.

Old (works fine)

<--- SIP read from 192.168.0.81:51217 --->
REGISTER sip:192.168.0.80 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK51978974
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
Max-Forwards: 70
CSeq: 135 REGISTER
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:101@192.168.0.81:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000f2321c7fd>";+u.sip!model.ccm.cisco.com="8"
Content-Length: 0
Expires: 3500


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.81 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.81:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK51978974;received=192.168.0.81
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
CSeq: 135 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.0.80>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.0.81:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK51978974;received=192.168.0.81
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>;tag=as36fc87f6
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
CSeq: 135 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d5995ec"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81' in 32000 ms (Method: REGISTER)
asterisk*CLI>
<--- SIP read from 192.168.0.81:51218 --->
REGISTER sip:192.168.0.80 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK3eb3a934
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
Max-Forwards: 70
CSeq: 136 REGISTER
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:101@192.168.0.81:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000f2321c7fd>";+u.sip!model.ccm.cisco.com="8"
Authorization: Digest username="101",realm="asterisk",uri="sip:192.168.0.80",response="b78a5674ba48fca962597b5d9daf13a1",nonce="6d5995ec",algorithm=MD5
Content-Length: 0
Expires: 3500


<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.81 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.81:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK3eb3a934;received=192.168.0.81
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
CSeq: 136 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.0.80>
Content-Length: 0


<------------>
asterisk*CLI>
<--- Transmitting (no NAT) to 192.168.0.81:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK3eb3a934;received=192.168.0.81
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>;tag=as36fc87f6
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
CSeq: 136 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3500
Contact: <sip:101@192.168.0.81:5060;transport=udp>;expires=3500
Date: Fri, 21 Aug 2009 12:21:02 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81' in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog '55a291d93834b85a36b2b73124545c94@192.168.0.80' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.0.81:5060:
NOTIFY sip:101@192.168.0.81:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.80:5060;branch=z9hG4bK424e99a8;rport
From: "asterisk" <sip:asterisk@192.168.0.80>;tag=as10806650
To: <sip:101@192.168.0.81:5060;transport=udp>
Contact: <sip:asterisk@192.168.0.80>
Call-ID: 55a291d93834b85a36b2b73124545c94@192.168.0.80
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.0.80
Voice-Message: 0/0

---
asterisk*CLI>
<--- SIP read from 192.168.0.81:51219 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.80:5060;branch=z9hG4bK424e99a8;rport
From: "asterisk" <sip:asterisk@192.168.0.80>;tag=as10806650
To: <sip:101@192.168.0.81:5060;transport=udp>
Call-ID: 55a291d93834b85a36b2b73124545c94@192.168.0.80
CSeq: 102 NOTIFY
Content-Length: 0

New (cisco cant register)

<--- SIP read from 192.168.0.81:50851 --->
REGISTER sip:192.168.0.53 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK74855e0e
From: <sip:101@192.168.0.53>;tag=000f2321c7fd00022043af91-6b9af7a0
To: <sip:101@192.168.0.53>
Call-ID: 000f2321-c7fd0002-07a198fc-68d94e03@192.168.0.81
Max-Forwards: 70
CSeq: 101 REGISTER
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:101@192.168.0.81:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000f2321c7fd>";+u.sip!model.ccm.cisco.com="8"
ontent-Length: 0
Expires: 3500


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.81 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.81:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK74855e0e;received=192.168.0.81
From: <sip:101@192.168.0.53>;tag=000f2321c7fd00022043af91-6b9af7a0
To: <sip:101@192.168.0.53>
Call-ID: 000f2321-c7fd0002-07a198fc-68d94e03@192.168.0.81
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
pbx*CLI>
<--- Transmitting (no NAT) to 192.168.0.81:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK74855e0e;received=192.168.0.81
From: <sip:101@192.168.0.53>;tag=000f2321c7fd00022043af91-6b9af7a0
To: <sip:101@192.168.0.53>;tag=as22cd17fd
Call-ID: 000f2321-c7fd0002-07a198fc-68d94e03@192.168.0.81
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2b67d58c"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '000f2321-c7fd0002-07a198fc-68d94e03@192.168.0.81' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '000f2321-c7fd0002-07a198fc-68d94e03@192.168.0.81' Method: REGISTER

Well, I sorted this out, just make some changes in kernel configuration (in Networking Support -> Networking options -> disabled not needed things for me, fe. Netfilter, Multicast support, GARP etc.) and now its working :smiley:.

[quote=“zipiju”]Hello,

let me explain my situation.
Two days ago, I decided to move the asterisk to the another computer, so I downloaded the latest 1.4.26.1 relase, installed it and copied configuration files from old 1.4.15.
The problem is, Cisco 7940 and 7960 phones (SIP Firmware 8.11) suddenly cant register to Asterisk, other phones can, softphones can too.
I thought, its some incompatibility issue or bug, so i downgraded to 1.4.15.
But Cisco phones still cant register. I even tried to downgrade firmware in the phones down to 8.2.
Strange thing is, it is working on the original asterisk box.
The only difference in software is architecture (original is x86, new is x86_64) and kernel version (old 2.6.22, new 2.6.29), both gentoo, both 1.4.15, same configuration.
Phones and asterisk are on the same subnet.
Below is debug log from old and new install.
Is here someone with similar configuration and working Cisco phones?
Im really desperate for any help.

Thank you.

Old (works fine)

<--- SIP read from 192.168.0.81:51217 --->
REGISTER sip:192.168.0.80 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK51978974
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
Max-Forwards: 70
CSeq: 135 REGISTER
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:101@192.168.0.81:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000f2321c7fd>";+u.sip!model.ccm.cisco.com="8"
Content-Length: 0
Expires: 3500


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.81 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.81:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK51978974;received=192.168.0.81
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
CSeq: 135 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.0.80>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.0.81:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK51978974;received=192.168.0.81
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>;tag=as36fc87f6
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
CSeq: 135 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d5995ec"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81' in 32000 ms (Method: REGISTER)
asterisk*CLI>
<--- SIP read from 192.168.0.81:51218 --->
REGISTER sip:192.168.0.80 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK3eb3a934
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
Max-Forwards: 70
CSeq: 136 REGISTER
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:101@192.168.0.81:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000f2321c7fd>";+u.sip!model.ccm.cisco.com="8"
Authorization: Digest username="101",realm="asterisk",uri="sip:192.168.0.80",response="b78a5674ba48fca962597b5d9daf13a1",nonce="6d5995ec",algorithm=MD5
Content-Length: 0
Expires: 3500


<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.81 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.81:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK3eb3a934;received=192.168.0.81
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
CSeq: 136 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.0.80>
Content-Length: 0


<------------>
asterisk*CLI>
<--- Transmitting (no NAT) to 192.168.0.81:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK3eb3a934;received=192.168.0.81
From: <sip:101@192.168.0.80>;tag=000f2321c7fd001e05140bd7-3814f0e0
To: <sip:101@192.168.0.80>;tag=as36fc87f6
Call-ID: 000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81
CSeq: 136 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3500
Contact: <sip:101@192.168.0.81:5060;transport=udp>;expires=3500
Date: Fri, 21 Aug 2009 12:21:02 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '000f2321-c7fd000c-7922c2e5-5a453a4c@192.168.0.81' in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog '55a291d93834b85a36b2b73124545c94@192.168.0.80' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.0.81:5060:
NOTIFY sip:101@192.168.0.81:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.80:5060;branch=z9hG4bK424e99a8;rport
From: "asterisk" <sip:asterisk@192.168.0.80>;tag=as10806650
To: <sip:101@192.168.0.81:5060;transport=udp>
Contact: <sip:asterisk@192.168.0.80>
Call-ID: 55a291d93834b85a36b2b73124545c94@192.168.0.80
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.0.80
Voice-Message: 0/0

---
asterisk*CLI>
<--- SIP read from 192.168.0.81:51219 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.80:5060;branch=z9hG4bK424e99a8;rport
From: "asterisk" <sip:asterisk@192.168.0.80>;tag=as10806650
To: <sip:101@192.168.0.81:5060;transport=udp>
Call-ID: 55a291d93834b85a36b2b73124545c94@192.168.0.80
CSeq: 102 NOTIFY
Content-Length: 0

New (cisco cant register)

[code]
<— SIP read from 192.168.0.81:50851 —>
REGISTER sip:192.168.0.53 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK74855e0e
From: sip:101@192.168.0.53;tag=000f2321c7fd00022043af91-6b9af7a0
To: sip:101@192.168.0.53
Call-ID: 000f2321-c7fd0002-07a198fc-68d94e03@192.168.0.81
Max-Forwards: 70
CSeq: 101 REGISTER
User-Agent: Cisco-CP7940G/8.0
Contact: sip:101@192.168.0.81:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-000f2321c7fd”;+u.sip!model.ccm.cisco.com="8"
ontent-Length: 0
Expires: 3500

<------------->
— (11 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.81 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.0.81:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK74855e0e;received=192.168.0.81
From: sip:101@192.168.0.53;tag=000f2321c7fd00022043af91-6b9af7a0
To: sip:101@192.168.0.53
Call-ID: 000f2321-c7fd0002-07a198fc-68d94e03@192.168.0.81
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
pbx*CLI>
<— Transmitting (no NAT) to 192.168.0.81:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.81:5060;branch=z9hG4bK74855e0e;received=192.168.0.81
From: sip:101@192.168.0.53;tag=000f2321c7fd00022043af91-6b9af7a0
To: sip:101@192.168.0.53;tag=as22cd17fd
Call-ID: 000f2321-c7fd0002-07a198fc-68d94e03@192.168.0.81
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2b67d58c"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘000f2321-c7fd0002-07a198fc-68d94e03@192.168.0.81’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘000f2321-c7fd0002-07a198fc-68d94e03@192.168.0.81’ Method: REGISTER
[/code][/quote] :smiley:

Sorry to dig up an old post, but the OP doesn’t appear to be logging on any more. I have this exact issue, but with AsteriskNOW (freepbx) 2.7 + Asterisk 1.4.41.

I fear an update must have created an issue (found this ticket here:https://issues.asterisk.org/jira/browse/ASTERISK-17535) suggesting as much.

What did the OP mean when they said [quote]just make some changes in kernel configuration (in Networking Support -> Networking options -> disabled not needed things for me, fe. Netfilter, Multicast support, GARP etc.) and now its working[/quote]

Any help is appreciated.

He means that the software firewall was breaking things.

Thanks, I thought that might be the case, but I hadn’t changed any settings on the Linux firewall. It has been on (to allow some bandwidth reporting) but everything is set to accept. Are there any other firewalls I don’t know about?

As there is no firewall active, and no config changes were made, I can’t find anything to “fix” (they just failed to register after a reboot).

Has anyone else seen this Cisco 7940 + Asterisk 1.4.1 reg fail issue pop up in the last month or so?

These phones have been happily registered and used for over a year, and just suddenly stopped, though X-Lite and polycom 331 phones still register and work fine.

Thanks in advance

Howdy,

Try 1.4.42-rc1.

Thanks. Is this safe to try on a production (albeit busted) box?

Would I just use yum to search, or are there more specific instructions for the RC?

Thanks in advance for your help.

Chris

I don’t believe there are any release candidate packages available on packages.asterisk.org.

You’re best off to roll back a version and wait for a 1.4.42 package, or instead build it yourself from source.

Malcom (or any others) - any suggestions / guides on how to roll the asterisk version back on the free pbx box?

Also, as an aside, I have a freepbx 2.7 / asterisk 1.6 box (64bit) in another location that the cisco phones will register to. When building the same version box (32bit) the ciscos will not register. The only diff is that I have not taken the new server to the other location to see if they will register.

Thanks again for any help - this is a nasty issue…