Configure your VOIP Phone

You’ve got asterisk installed, configured and running. Now it’s time to get your phone to connect to asterisk.

There are several major vendors of VOIP phones. I’m going to outline the steps for a SNOM 320 and an Aastra 9133i phone. Configuring phones isn’t hard, since you basically need to enter three pieces of information:

  1. username
  2. password
  3. asterisk server IP address

Despite only requiring these three pieces of information most VOIP phones seemed to have been designed to make this process as awkward as possible. So, take few deep breaths, and let’s get started.

Snom 320

Snom 320 Voip Phone SNOM phones are by far the easiest phones to setup. Documentation is freely available and you can do the setup directly from the phone.

Plug in the power and ethernet cables. Your router should have DHCP on and will assign the phone an IP address. After the phone boots it will enter a ‘wizard’ mode if it has not been configured before.

For the Account enter the username in sip.conf. For example: phone1

For the Registrar enter the IP address of your server running asterisk. (You can also enter a fully-qualified hostname such as if you have a domain name and have configured DNS for your server. If you don’t know what I’m talking about, use the IP address.)

The phone will contact the server and if all goes well, will ask for a Password which is what we defined as secret in the sip.conf config file.

And now you should be connected to the asterisk PBX. Dial 9 and you should be in the voicemail system.

If you see “NR”, that means Not Registered. More on how to debug connection problems below.

If you ever need to get into the administration menu, the default admin password is: 0000

Aastra 9133i

Aastra 9133i VOIP Phone
A nice, sturdy phone with better sound quality sound than the SNOM, but a bit more fiddly to configure.

Plug in the power and ethernet cables. Your router should assign the phone an IP address. If you don’t catch the IP address when booting, you can find the IP address in the phone’s menu system.

Press the Options button (top right) and use the arrow keys to navigate the menus:

Network Settings • Admin Password: 22222 • IP Address

Now, type the IP address into a web browser. The default admin username/password is: admin/22222

We need to configure two screens: Network Settings and Global SIP.

Network Settings

Most of these settings are automatic and defined by your router when the phone boots. The only field you need to define is the Time Server 1. There is a bug in Aastra firmware in their DNS resolution code that crashes the phone, so I suggest you use IP addresses instead of server names.

You can google for public NTP time servers or try one of the following servers below:  

Add an IP address for Time Server 1 and click Save Settings.

Global SIP

There are a lot of fields on this screen. Some are not important, and many are duplicated. The following works for me. This configuration assumes the username is phone1, the password is xxxsecretxxx and the asterisk server IP address is Unless specified, leave the default values as is.

# aastra 91331 - Global SIP Settings
Screen Name:         aastra
Phone Number:        phone1
Caller ID:           phone1
Authentication Name: phone1
Password:            xxxsecretxxx
Line Mode:           Generic

Proxy Server:
Proxy Port:            5060
Outbound Proxy Server:
Outbound Proxy Port:   5060
Registrar Server:
Registrar Port:        5060
Registration Period:   600

Once you click “Save Settings”, you’ll need to click the “Reset” menu option to reboot the phone before your settings will take effect.

If the phone boots, the red light goes off and you don’t see a “No Service” message, then you’re connected! To test, dial 9 to enter the voicemail system or any of the extensions we’ve defined.

Debugging Connection Problems

If you’re phones aren’t registering with the asterisk server, then you need to isolate the problem.

A command I often use to get an overview of all the sip phones is: sip show peers

On your asterisk server, as root, connect to asterisk:

asterisk -r

If all is well, you should see something like the following:

sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
phone1/phone1          D   N      5060     OK (115 ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]

If the status is UNKNOWN or there is host address, then you need to get more information about the registration process.

Turn on more verbose messages.

core set verbose 10

Now reboot your phone and watch for error messages on the asterisk console. If you don’t see any – then your phone isn’t contacting the server.

A successful registration message is:

Registered SIP 'phone1' at port 1025 expires 3600

If you want to see all of the SIP messages between your phone and the asterisk server, type:

sip set debug

But be warned that the raw SIP messages can be overwhelming! I find that cutting and pasting them into a text editor makes them a bit easier to decode.

Most of the time I had problems connecting a phone it was due to typos and having the wrong username or password entered into one of the fields. Double-check that the values in sip.conf match the ones you’ve entered into your phone’s configuration.


Now, you’re able to use your VOIP phone to call your PBX. That’s an important step, but in order to have a real phone system we need to be able to call the outside world, and let people call our PBX. To do that we need to create an account with a VOIP provider.

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