From VoIP.ms Wiki

This guide explains how to configure a Mitel 6900 Series Open SIP phones with VoIP.ms. Please note that menu names and available options may vary depending on the phone model, firmware version, and provisioning method being used.

Contents 1 Supported Models 2 Firmware Note 3 Step 1 - Create or Verify the VoIP.ms Sub Account 4 Step 2 - Obtain the Phone IP Address 5 Step 3 - Access the Phone Web Interface 6 Step 4 - Configure SIP Line 1 6.1 Basic SIP Authentication Settings 6.2 Basic SIP Network Settings 6.3 Step 5 - Save the Configuration 7 Codec Settings 8 Firewall and NAT Considerations 9 Test Calls 10 Troubleshooting 10.1 The Phone Does Not Register 10.2 Incoming Calls Do Not Reach the Phone 10.2.1 One-Way Audio 10.2.2 DTMF Is Not Working

Supported Models

This guide applies generally to Mitel 6800 and 6900 Series IP phones, including but not limited to:

• 6863i
• 6865i
• 6867i
• 6869i
• 6873i
• 6905
• 6910
• 6920
• 6930
• 6940
• 6970

Firmware Note

Step 1 - Create or Verify the VoIP.ms Sub Account

Access your VoIP.ms Customer Portal:

1. Go to Sub Accounts
2. Create a new Sub Account or edit an existing one
3. Take note of the SIP Username and SIP Password
4. Make sure the selected codec matches the codec configured on the phone
5. Save the changes

Recommended Sub Account settings:

Device Type: ATA Device, IP Phone or Softphone

SIP Username: Your Sub Account username 100000_office

Password: Your SIP password

Protocol: SIP

Allowed Codecs: Select the codecs you will use on the Mitel phone

Step 2 - Obtain the Phone IP Address

Connect the phone to the network and power it on. From the phone keypad or screen:

1. Press the Settings button
2. Navigate to Status
3. Go to Network
4. Locate the phone IP address

The path may appear as:

Settings -> Status -> Network -> IP Address

On some models, the IP address may be located under:

Phone Status -> IP & MAC Addresses

Step 3 - Access the Phone Web Interface

From a computer connected to the same network as the phone:

1. Open a web browser
2. Enter the phone IP address in the address bar
3. Log in as administrator

Default administrator credentials are commonly:

Username: admin

Password: 22222

Note:

If these credentials do not work, the password may have been changed by a previous administrator. In that case, please refer to the device documentation or reset procedure.

Security recommendation:

It is strongly recommended to change the default administrator password after completing the configuration.

Step 4 - Configure SIP Line 1

Once logged into the phone Web Interface:

1. Go to Advanced Settings
2. Select Line 1

The SIP settings are usually divided into two main sections:

• Basic SIP Authentication Settings
• Basic SIP Network Settings

Basic SIP Authentication Settings

Under Basic SIP Authentication Settings, configure the following fields:

Screen Name: Name that displays on the idle screen

Screen Name 2: Custom text message that displays on the idle screen.

Phone Number: Enter your VoIP.ms SubAccount Username ######_SubAccount

Authentication Name: Enter your VoIP.ms SubAccount Username ######_SubAccount

Password: Use the SIP password from the Main Account or Sub Account.

Display Name: Enter your outbound Caller ID Name that will be displayed for Canadian Numbers

Basic SIP Network Settings

Under Basic SIP Network Settings, configure the following fields:

Proxy Server: Enter your Preferred VoIP.ms PoP server. Example: montreal3.voip.ms

Proxy Port: 5060 (This is the default SIP port.)

Outbound Proxy: Enter your Preferred VoIP.ms PoP server. Example: montreal3.voip.ms

Proxy Port: 5060 (This is the default SIP port.)

Registrar Server: Enter your Preferred VoIP.ms PoP server. Example: montreal3.voip.ms

Registrar Port: 5060 (This is the default SIP port.)

Registration Period: 120 secondes

Recommended registration expiry value.

Step 5 - Save the Configuration

After entering the SIP authentication and SIP network settings:

1. Click Save Settings
2. Reboot the phone if required
3. Wait for the phone to register

The phone should show the line as registered once the configuration is successful.

Codec Settings

You can change the default codecs selection on the phone to match the codecs enabled in your VoIP.ms account or Sub Account.

Note: This is optional, as the Basic Codecs Includes G.711u.

Firewall and NAT Considerations

Recommended router settings:

• Disable SIP ALG
• Make sure the phone can resolve DNS hostnames
• Make sure the phone can reach the selected VoIP.ms server
• Avoid multiple SIP devices using the same local SIP port when possible

Test Calls

After the phone registers, test the configuration.

VoIP.ms test numbers:

• 4443: Echo test
• 4747: DTMF test
• 822: Caller ID playback test
• *98 or *97: Voicemail access

It is also recommended to test:

• Outbound calls
• Inbound calls
• Two-way audio
• DTMF through an IVR
• Voicemail access

Troubleshooting

The Phone Does Not Register

If the phone does not register, verify the following:

• The SIP Username is correct
• The SIP Password is correct
• The correct VoIP.ms PoP server is being used
• The SIP port is correct
• The selected transport protocol matches the account configuration
• SIP ALG is disabled on the router
• The account or Sub Account is not blocked due to repeated failed registration attempts
• If using SIP-TLS, encrypted SIP traffic is enabled in the VoIP.ms portal and the phone is configured for TLS/SRTP

Incoming Calls Do Not Reach the Phone

If outbound calls work but inbound calls do not, verify the following:

• The DID is routed to the correct SIP account or Sub Account
• The DID PoP matches the server used by the phone
• The phone is currently registered
• Failover routing is not sending the call elsewhere
• The firewall is not blocking SIP or RTP traffic

One-Way Audio

If one-way audio occurs, verify the following:

• SIP ALG is disabled
• RTP ports are not blocked by the firewall
• The phone is not behind multiple NAT routers
• The selected codec is enabled in both the phone and the VoIP.ms portal
• The device is registering to the correct VoIP.ms server

DTMF Is Not Working

If DTMF tones are not detected properly:

• Set DTMF to RFC2833 or AVT
• Avoid using in-band DTMF
• Test using the VoIP.ms DTMF test number: 4747