I’m trying to register an Avaya 9630 IP phone with my Asterisk box. Asterisk is replying to the phone’s REGISTER request with a 401 Unauthorized message. I had successfully registered this phone, using the same phone configuration, with a previous install of Asterisk (AsteriskNOW 1.6 with FreePBX 2.5, upgraded to 2.6, 2.7, and finally to 2.8 using the Module Admin GUI) but decided to install a newer version of AsteriskNOW to see if it would resolve a problem with my Digium TDM800P (and it did).
I have included what I think is the pertinent information below. Please let me know if there is anything else I should include.
CentOS (not sure what version; uname -a gives kernel version 2.6.18-194.3.1.el5)
FreePBX 2.8.0.2
Asterisk 1.6.2.7
Version numbers of currently enabled modules:
Core 2.8.0.2
DAHDi Config 2.7.0
Feature Code Admin 2.8.0.0
FreePBX Framework 2.8.0.2
System Dashboard 2.8.0.3
Voicemail 2.8.0.0
Info Services 2.8.0.0
Music on Hold 2.8.0.2
Recordings 3.3.10.0
Custom Applications 2.8.0.0
Installation/configuration performed:
- booted from AsteriskNOW 64-bit version 1.7 ISO image downloaded from www.asterisk.org
- chose the “install with Asterisk 1.6 and FreePBX” option
- applied configuration changes
- updated all installed modules which had available online updates, using Module Admin
- applied configuration changes and rebooted
- installed 2.8 upgrade tool, upgraded to 2.8, applied configuration changes
- rebooted
- added SIP extension (6000) through GUI
SIP extension config from /etc/sip_additional.conf:
[6000]
deny=0.0.0.0/0.0.0.0
secret=secret01
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6000
accountcode=
mailbox=6000@default
permit=0.0.0.0/0.0.0.0
callerid=device <6000>
call-limit=50
faxdetect=no
Asterisk server is 192.168.2.10
Avaya 9630 IP phone is 192.168.2.11
SIP debug trace:
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
localhost*CLI>
<— SIP read from UDP:192.168.2.11:1032 —>
REGISTER sip:192.168.2.10 SIP/2.0
From: sip:[email protected];tag=-53c37645386d58bc3875b4fc_F192.168.2.11
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bK1_84bbe1ceb34653875a786_R
Content-Length: 0
Max-Forwards: 70
Contact: sip:[email protected];avaya-sc-enabled;transport=udp;q=1;expires=900;reg-id=1;+sip.instance="urn:uuid:00000000-0000-1000-8000-00040dec8565"
Allow: INVITE,CANCEL,BYE,ACK,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
User-Agent: Avaya one-X Deskphone
Supported: replaces
<------------->
— (12 headers 0 lines) —
Sending to 192.168.2.11 : 1032 (NAT)
<— Transmitting (NAT) to 192.168.2.11:1032 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bK1_84bbe1ceb34653875a786_R;received=192.168.2.11
From: sip:[email protected];tag=-53c37645386d58bc3875b4fc_F192.168.2.11
To: sip:[email protected];tag=as12704235
Call-ID: [email protected]
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="34dd9dfd"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
localhost*CLI>
<— SIP read from UDP:192.168.2.11:1032 —>
REGISTER sip:192.168.2.10 SIP/2.0
From: sip:[email protected];tag=-53c37645386d58bc3875b4fc_F192.168.2.11
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bK1_84bbe1ceb34653875a786_R
Content-Length: 0
Max-Forwards: 70
Contact: sip:[email protected];avaya-sc-enabled;transport=udp;q=1;expires=900;reg-id=1;+sip.instance="urn:uuid:00000000-0000-1000-8000-00040dec8565"
Allow: INVITE,CANCEL,BYE,ACK,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
User-Agent: Avaya one-X Deskphone
Supported: replaces
<------------->
— (12 headers 0 lines) —
Sending to 192.168.2.11 : 1032 (NAT)
localhost*CLI>
<— Transmitting (NAT) to 192.168.2.11:1032 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bK1_84bbe1ceb34653875a786_R;received=192.168.2.11
From: sip:[email protected];tag=-53c37645386d58bc3875b4fc_F192.168.2.11
To: sip:[email protected];tag=as12704235
Call-ID: [email protected]
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="34dd9dfd"
Content-Length: 0
It goes on like this until I cancel the login attempt on the phone. As I mentioned at the beginning of the message, I successfully registered this phone with a previous Asterisk install, so I don’t think the problem is with the phone’s configuration. I appreciate any help that can be provided in getting this to work.
Thanks!
Brian