[asterisk-users] Problems with 9133i config

Wed Feb 4 14:29:44 CST 2009

I am unable to get my 9133i to register with my asterisk server. I am
including config files below, this a simple test network so there's nothing
secret in the config files. I have upgraded the phone to the latest software
version (1.4.3) I'm not sure what the problem is. I can call the phone from
a softphone, but the 9133i says "no service" on the screen and I can't dial
anything on it.

configs:
Aastra.cfg
dhcp: 1                           # DHCP enabled.
sip silence suppression: 2        # "0" = off, "1" = on, "2" = default
sip proxy port: 5060              # 5060 is set by default.
sip registrar ip: 192.168.0.94    # IP of registrar
sip registrar port: 5060          # 5060 is set by default.
sip digit time out: 6
time server disabled: 0           # Time server disabled.
time server1: 192.168.0.90        # Enable time server and enter at

<mac>.cfg - this is the correct mac address in all uppercase
sip line1 auth name: phone1
sip line1 password: 1234
sip line1 registrar ip: 192.168.0.94
sip line1 user name: phone1
sip line1 display name: "myname"
sip line1 screen name: "myname"

sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context=tutorial

[phone1]
type=friend
username=phone1
secret=1234
host=dynamic
canreinvite=no
permit=192.168.0.0/24
allow=all
qualify=yes

extensions.conf
[tutorial]
exten => 1234,1,Answer
exten => 1234,n,SayDigits(123456789)

exten => 3001,1,Dial(SIP/phone1,18)

exten => 3002,1,Dial(SIP/phone2,18)

sip debug output
<--- SIP read from 192.168.0.11:5060 --->
REGISTER sip:192.168.0.94 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869
Max-Forwards: 70
Content-Length: 0
To: myname <sip:phone1@>
From: myname <sip:phone1@>;tag=24b6354e352ab62
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
Contact: myname <sip:phone1 at 192.168.0.11:5060;transport=udp>
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
User-Agent: Aastra 9133i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.11 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11
From: myname <sip:phone1@>;tag=24b6354e352ab62
To: myname <sip:phone1@>
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:phone1 at 192.168.0.94>
Content-Length: 0

<------------>
<--- Transmitting (no NAT) to 192.168.0.11:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11
From: myname <sip:phone1@>;tag=24b6354e352ab62
To: myname <sip:phone1@>;tag=as51ded290
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f250e11"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog
'a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11' in 32000 ms (Method:
REGISTER)
<--- SIP read from 192.168.0.11:5060 --->
REGISTER sip:192.168.0.94 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869
Max-Forwards: 70
Content-Length: 0
To: myname <sip:phone1@>
From: myname <sip:phone1@>;tag=24b6354e352ab62
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
Contact: myname <sip:phone1 at 192.168.0.11:5060;transport=udp>
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
User-Agent: Aastra 9133i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.11 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11
From: myname <sip:phone1@>;tag=24b6354e352ab62
To: myname <sip:phone1@>
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:phone1 at 192.168.0.94>
Content-Length: 0

<------------>
<--- Transmitting (no NAT) to 192.168.0.11:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11
From: myname <sip:phone1@>;tag=24b6354e352ab62
To: myname <sip:phone1@>;tag=as51ded290
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f250e11"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog
'a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11' in 32000 ms (Method:
REGISTER)
<--- SIP read from 192.168.0.11:5060 --->
REGISTER sip:192.168.0.94 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869
Max-Forwards: 70
Content-Length: 0
To: myname <sip:phone1@>
From: myname <sip:phone1@>;tag=24b6354e352ab62
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
Contact: myname <sip:phone1 at 192.168.0.11:5060;transport=udp>
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
User-Agent: Aastra 9133i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.11 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11
From: myname <sip:phone1@>;tag=24b6354e352ab62
To: myname <sip:phone1@>
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:phone1 at 192.168.0.94>
Content-Length: 0

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

<--- Transmitting (no NAT) to 192.168.0.11:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11
From: myname <sip:phone1@>;tag=24b6354e352ab62
To: myname <sip:phone1@>;tag=as51ded290
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f250e11"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog
'a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11' in 32000 ms (Method:
REGISTER)
<--- SIP read from 192.168.0.11:5060 --->
REGISTER sip:192.168.0.94 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869
Max-Forwards: 70
Content-Length: 0
To: myname <sip:phone1@>
From: myname <sip:phone1@>;tag=24b6354e352ab62
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
Contact: myname <sip:phone1 at 192.168.0.11:5060;transport=udp>
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
User-Agent: Aastra 9133i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.11 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11
From: myname <sip:phone1@>;tag=24b6354e352ab62
To: myname <sip:phone1@>
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:phone1 at 192.168.0.94>
Content-Length: 0

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

<--- Transmitting (no NAT) to 192.168.0.11:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11
From: myname <sip:phone1@>;tag=24b6354e352ab62
To: myname <sip:phone1@>;tag=as51ded290
Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11
CSeq: 1006065354 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f250e11"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog
'a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11' in 32000 ms (Method:
REGISTER)

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer	Safe Data, Inc.
(910) 285-7200	david at safedatausa.com