Aastra 480i not connecting with Asterisk @ home


#1

Hello guys, I recently setup asterisk @ home with Digium’s Wildcard TDM400P REV I BOARD I (2 modules) with Aastra 480i (really nice phone). Everything is running perfectly, but my Aastra phone keeps saying “no service”. I left the settings on the phone at default, as I noticed that it automatically detected the DHCP settings. Also I see FXSKS=1 and FXSKS=2 are opened and plugged into a regular landline. Here is my configuration in Asterisk…

SIP.CONF

[phone1]
type=friend
host=dynamic
defaultip=192.168.0.156 (ip from phone)
secret=1234
dtmfmode=rfc2833
mailbox=1000
context=sip
callerid=“Phone 1” <1234>

and here is my config for the EXTENSIONS.CONF

exten => 1234,1,Dial(SIP/phone,20)

Anyone know how I can test the connection on the phone? I dialed 7777 and hear nothing. Thank you for your help!

Regards,
Tabo


#2

I am having the same issue with a 480i CT that I purchased to demo.

From what I can tell the 480i is sending a malformed SIP registration string–but I’m not totally sure. I’m not an expert on SIP.

I have three other phones that all connect to asterisk, and I’ve looked at the SIP debugging information on all of them. The 480i seems to send a totally different version of the registration string.

I called Aastra about 30 minutes ago and whoever answered the phone said a tech would ‘call me right back’.

Who knows how long that will be–but I’m almost ready to ship the phone back and never buy from them again.

As soon as I get a response or get it figured out, I will let you know.


#3

Hi,

Please refer to the documentation here on how to configure the 480i phone
voip-info.org/tiki-index.php … astra+480i

and here: voip-info.org/wiki/index.php … T+Cordless
for the 480i cordless phones.

Are you sure you have the aastra.cfg and mac.cfg files at your tftp server? You can also log on to the phone’s web-interface to configure the phone. Just type in the IP address of the phone in your web-browser. Hope this will solve the current problems that you have. Also please do not forget to mention what firmware version your are using on the phone and what REGISTER sip message the phone sends to your asterisk server.


#4

Thanks for the pointers–but I have poured over those docs many times.

The phones are getting the config files from the TFTP server–I can see it connect and download aastra.cfg and the (macaddress).cfg file

I have logged in to the phones interface configured it–as well as through the console itself.

I am about ready to tackle it again after downloading the newly released version of Asterisk–I’ll grab the SIP debug info from the console and also post my config.


#5

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.


#6

Hi darkpixel,

From the phone’s configuration files I see that you do not have the proxy address and ip set. Can you please try setting the following parameters:

#Phone will use this as proxy (send out INVITE and other packets)
sip proxy ip: 192.168.42.10
sip proxy port: 5060

#Phone will use this to send REGISTER requests to
sip registrar ip: 192.168.42.10
sip registrar port: 5060

I will let you know about the other issues that you have:

  1. It doesn’t pick up the NNTP settings from the DHCP server.
  2. It doesn’t get the next-address setting for the TFTP server from DHCP.
  3. 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.

Let me know if setting the proxy ip and address makes the phone in service.

Thanks.


#7

Hi darkpixel,

Can you tell me what kind of DHCP server you are running and on which platform (OS)? It would be helpful to see a capture of the network traffic that the phone receives from the DHCP server. If you know how to capture network traffic using any sniffing tool (such as ethereal) can you please capture and post it?

Also Aastra will be releasing a new firmware 1.3 soon. The phone not getting the time server IP address was a known issue and is resolved in 1.3. Hopefully the other issues that you have may also have been resolved. Please update your firmware to 1.3 whenever that is released and see if it fixes all your problems.

Thanks.


#8

I’ve been having the exact same problems listed above with my Aastra/Sayson 480 CT phone, with firmware version 1.3

I do NOT have these same problems with firmware version 1.2.1.1002 that the phone shipped with.

I will post my aastra.cfg file when I get a chance.


#9

Okay, so I just set my aastra.cfg file to look like the above example. These settings mostly work for any soft phones, as well as the 480 CT with older firmware.

sip proxy ip: 192.168.0.27
sip proxy port: 5060

#Phone will use this to send REGISTER requests to
sip registrar ip: 192.168.0.27
sip registrar port: 5060


#10

Sorry for the delay in replying.

I was originally working on compiling a list of everything that failed to work on this phone.

In the end, I gave up. Everything from not grabbing DHCP settings correctly (NTP address, TFTP address) , to not registering with the asterisk box. Even when it grabbed the aastra.cfg and mac.cfg file, it would ignore several of the settings.

My solution was to use the keypad on the phone to manually enter everything I could. I was finally able to get it to register with my asterisk box. Definitely not a phone for a corporate environment where you need them to use DHCP and TFTP to self-configure.

I really like the looks of the phone. It’s very professional. But the network side of it is hell. Once the new firmware comes out, I’ll give it another shot. For now, the phone is going to sit on the test bench.


#11

We have used these phones (480i and 9133) for a while now. The latest firmware is crucial. We have several customers using the 9133’s remotely and picking up their cfg files via FTP. The one thing besides the BLF not dialing I don’t like about the latest firmware is that they have changed the DTMF speed and slowed it down. While this may sound like a minor issue, it delays speedial keys (Such as a *8 or long phone number) enough to be irritating.

The only real issue I have lately with the config files is that I can’t seem to get a 480i register remotely while registered on a local Asterisk server.

For example, line 1: Local Asterisk box (192.168.xxx.xxx)
Line 4: Remote Asterisk box (199.227.xxx.xxx)

We can get other phones to work fine with this type of config, but not the 480i. Maybe someone else can shed some light on this?