My Aastra 480i CT is running firmware version 1.2.2.1004.
There are several problems I have noted with this version.
It doesn’t pick up the NNTP settings from the DHCP server.
It doesn’t get the next-address setting for the TFTP server from DHCP.
For some reason the factory defaults have the primary DNS IP as 0.0.0.0 and the secondary as 128.111.235.128. This is not coming from DHCP.
It grabs the mac.cfg file, but doesn’t attempt to register
Manually specifying those IPs in the phone’s config resolves the problem.
In aastra.cfg I have the following lines relevant to SIP:
sip registrar ip: 192.168.42.10
sip registrar port: 5060
sip digit timeout: 4
sip dial plan: "[2-9]11|9911|1[01]xx|[2-9]xxxxxx|1[2-9]xxxxxxxxx|011x+#|xx*|*xx$
In the mac.cfg file I have the following lines relevant to SIP:
sip line1 auth name: nocphone1
sip line1 password: nocpass
sip line1 user name: nocphone1
sip line1 display name: Disp L1-100
sip line1 screen name: Scrn L1-100
My asterisk sip.cfg file contains:
[nocphone1]
type=friend
host=dynamic
defaultip=192.168.42.40
secret=nocpass
dialmode=rfc2833
context=sip
callerid=“NOC Phone 1” <100>
I start asterisk with -cvvv and type the following:
*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
nocphone1 (Unspecified) D 0 Unmonitored
1 sip peers [1 online , 0 offline]
When the phone tries to register I get no messages on the asterisk display and the phone displays ‘no service’.
If I enable sip debugging and boot the phone, I get this:
*CLI> sip debug ip 192.168.42.40
SIP Debugging Enabled for IP: 192.168.42.40
<-- SIP read from 192.168.42.40:5060:
REGISTER sip:192.168.42.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.42.40;branch=z9hG4bK241cb2a3c
Max-Forwards: 70
Content-Length: 0
To: Disp L1-100 sip:nocphone1@
From: Disp L1-100 sip:nocphone1@;tag=a2922b24e5dda06
Call-ID: 40e00760c0f69e9658fb75bdd5fe1046@192.168.42.40
CSeq: 1899175569 REGISTER
Contact: Disp L1-100 sip:nocphone1@192.168.42.40;expires=0
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Expires: 0
User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
— (13 headers 0 lines)—
Using latest REGISTER request as basis request
Sending to 192.168.42.40 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.42.40:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.42.40;branch=z9hG4bK241cb2a3c;received=192.168.42.40
From: Disp L1-100 sip:nocphone1@;tag=a2922b24e5dda06
To: Disp L1-100 sip:nocphone1@
Call-ID: 40e00760c0f69e9658fb75bdd5fe1046@192.168.42.40
CSeq: 1899175569 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:nocphone1@192.168.42.10
Content-Length: 0
Transmitting (no NAT) to 192.168.42.40:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.42.40;branch=z9hG4bK241cb2a3c;received=192.168.42.40
From: Disp L1-100 sip:nocphone1@;tag=a2922b24e5dda06
To: Disp L1-100 sip:nocphone1@;tag=as095944f6
Call-ID: 40e00760c0f69e9658fb75bdd5fe1046@192.168.42.40
CSeq: 1899175569 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:nocphone1@192.168.42.10
WWW-Authenticate: Digest realm=“asterisk”, nonce="1820352f"
Content-Length: 0
Scheduling destruction of call ‘40e00760c0f69e9658fb75bdd5fe1046@192.168.42.40’ in 15000 ms
show peers
Name/username Host Dyn Nat ACL Port Status
nocphone1 (Unspecified) D 0 Unmonitored
1 sip peers [1 online , 0 offline]
*
I noticed this gives me an access denied error, so I toss ‘autocreatepeer=yes’ into sip.conf, comment out the entry for my phone and restart asterisk and the phone.
The following shows up in asterisk.
*CLI> – Registered SIP ‘nocphone1’ at 192.168.42.40 port 5060 expires 120
– Saved useragent “Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26” for peer nocphone1sip show peers
Name/username Host Dyn Nat ACL Port Status
nocphone1/nocphone1 192.168.42.40 D 5060 Unmonitored
1 sip peers [1 online , 0 offline]
*CLI>
So the phone shows up. But the display still says ‘No Service’.
I am at a total loss as to why the phone refuses to register.
I must admit I am not too familiar with SIP–but from what I understand, the phone should register with asterisk and depending on what context I put it in, it should be able to call numbers in that context.
I also have a Sipura 841 sitting on my desk. I’m going to boot it and see if I have as much trouble with that one. That should be the test to see if the trouble lies with me, or the aastra.