Mitel 6900 Series SIP Phones

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[[File:Mitel_6900_Series.png|right|none|Mitel 6900 Series IP Phones]] 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. = 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 = <div style="position: relative; width: 90%; /* height: 20px; */ background-color: #f0fbff; border: 2px solid #2f6185; border-radius: 8px; padding: 12px; overflow: hidden; display: flex; align-items: left; justify-content: left; flex-wrap: wrap; text-align: stretch; font-family: sans-serif; color: #333;"> <div style="position: absolute; margin-top:35px; margin-left: 20px; font-size: 150px; opacity: 0.1; z-index: 1; pointer-events: none; user-select: none; transform: rotate(-20deg);">ℹ️</div> <p style="position: relative; z-index: 2; margin: 0; line-height: 1.5;"> :'''The phone must be running SIP firmware.''' :'''If the phone was previously used with another platform or PBX, make sure it is running SIP firmware before attempting to register it with VoIP.ms.''' :'''If the phone is not running SIP firmware, please refer to the manufacturer documentation for firmware conversion or firmware upgrade instructions.''' </p></div> = Step 1 - Create or Verify the VoIP.ms Sub Account = Access your VoIP.ms Customer Portal: :# Go to <code>Sub Accounts</code> :# Create a new '''Sub Account''' or edit an existing one :# Take note of the '''SIP Username and SIP Password''' :# Make sure the selected codec matches the codec configured on the phone :# Save the changes :::Recommended Sub Account settings: ::::'''Device Type:''' ATA Device, IP Phone or Softphone ::::'''SIP Username:''' Your Sub Account username <code>100000_office</code> ::::'''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: :# Press the <code>Settings</code> button :# Navigate to <code>Status</code> :# Go to <code>Network</code> :# Locate the phone IP address :The path may appear as: :: <pre>Settings -> Status -> Network -> IP Address</pre> :: On some models, the IP address may be located under: :: <pre>Phone Status -> IP & MAC Addresses</pre> = Step 3 - Access the Phone Web Interface = From a computer connected to the same network as the phone: :# Open a web browser :# Enter the phone IP address in the address bar :# Log in as administrator Default administrator credentials are commonly: : '''Username:''' <code>admin</code> : '''Password:''' <code>22222</code> '''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: :# Go to <code>Advanced Settings</code> :# Select <code>Line 1</code> The SIP settings are usually divided into two main sections: :* Basic SIP Authentication Settings :* Basic SIP Network Settings == Basic SIP Authentication Settings == Under <code>Basic SIP Authentication Settings</code>, 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. :: [[File:Mitel_6900_ScreenName.png|none|350px]] :'''Phone Number:''' Enter your VoIP.ms SubAccount Username <code>######_SubAccount</code> :'''Authentication Name:''' Enter your VoIP.ms SubAccount Username <code>######_SubAccount</code> :'''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 <div style="position: relative; width: 90%; /* height: 20px; */ background-color: #fff3cd; border: 2px solid #ffeeba; border-radius: 8px; padding: 12px; overflow: hidden; display: flex; align-items: left; justify-content: left; flex-wrap: wrap; text-align: stretch; font-family: sans-serif; color: #856404;"> <div style="position: absolute; margin-top:35px; margin-left: 20px; font-size: 150px; opacity: 0.1; z-index: 1; pointer-events: none; user-select: none; transform: rotate(-20deg);">⚠️</div> <p style="position: relative; z-index: 2; margin: 0; line-height: 1.5;"> :- We suggest entering your outbound Caller ID Name must be in '''CAPITAL LETTERS'''. This will appears more clearly/visible on some devices. :- You must '''NOT''' use any special characters, they will not be displayed. :- '''Do not exceed 15 characters'''. Some of regular Canadian providers will '''not''' show more than 15 characters. We suggest shrinking or adapt your caller ID. :- Spaces are allowed and count as a character. </p></div> :: [[File:Mitel_6900_Line1_Basic_SIP_Auth.png|none|500px]] == Basic SIP Network Settings == Under <code>'''Basic SIP Network Settings'''</code>, configure the following fields: :'''Proxy Server:''' Enter your Preferred VoIP.ms PoP server. ''Example: <code>montreal3.voip.ms</code>'' :'''Proxy Port:''' <code>5060</code> ''(This is the default SIP port.)'' :'''Outbound Proxy:''' Enter your Preferred VoIP.ms PoP server. ''Example: <code>montreal3.voip.ms</code>'' :'''Proxy Port:''' <code>5060</code> ''(This is the default SIP port.)'' :'''Registrar Server:''' Enter your Preferred VoIP.ms PoP server. ''Example: <code>montreal3.voip.ms</code>'' :'''Registrar Port:''' <code>5060</code> ''(This is the default SIP port.)'' :'''Registration Period:''' <code>120</code> ''secondes'' :: Recommended registration expiry value. :: [[File:Mitel_6900_Line1_Basic_SIP_Network.png|none|500px]] == Step 5 - Save the Configuration == After entering the '''SIP authentication''' and '''SIP network settings''': :# Click''' <code>Save Settings</code>''' :# Reboot the phone if required :# 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 <code>RFC2833</code> or <code>AVT</code> :* Avoid using in-band DTMF :* Test using the VoIP.ms DTMF test number: <code>4747</code>

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