VoIP.ms Wiki - User contributions [en]

Mitel 6900 Series SIP Phones

2026-07-07T19:53:30Z

Scomtois:

[[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
:* 6907
:* 6910
:* 6915
:* 6920/6920w
:* 6930/6930w
:* 6940/6940w
:* 6970
:* M695 PKM ''(If supported)''

= 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>

Mitel 6900 Series SIP Phones

2026-07-07T13:34:56Z

Scomtois: /* Supported Models */

[[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
:* 6907
:* 6910
:* 6915
:* 6920/6920w
:* 6930
:* 6940
:* 6970
:* M695 PKM ''(If supported)''

= 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>

3CX Phone System - v20

2026-07-06T19:48:34Z

Scomtois: /* Sub-Account Setup */

[[File:3CX_Logo.jpg|300px|right|link=https://www.3cx.com/|3CX home page]]

3CX offers a complete Unified Communications solution out of the box.
As an open standards software PBX, installation and management of your phone system have never been easier.

== Pre-requisites ==

=== Latest 3CX V20 Update ===
:- Ensure you have the '''latest update''' of 3CX V20 installed.

=== Sub-Account Setup ===
Before proceeding, create a sub-account with the following settings:
:- '''Authentication type''': (x) User/Password Authentication
:- '''Device type''': (x) Asterisk, IP PBX, Gateway or VoIP Switch
:- '''CallerID Number''': (x) I use a system capable of passing its own CallerID
:- '''Allowed Codecs''': Only '''select G711''' or you can select G722 for HD voice
:- '''rInstance Parameter''': Set to '''"Yes"'''

= Trunk Configuration =

If you do not already have set a VoIP.ms trunk in your instance, you will need to create one.
: [1] Log into your 3CX web interface.
: [2] Click the '''"Admin"''' icon located at the bottom left corner.
: [3] In the left menu, select '''"Voice & Chat"'''.
: [4] Click the '''"[+ Add Trunk]"''' button.

== General Tab ==

Before completing the fields, load the VoIP.ms Trunk template:
: [1] Under '''"General"''' tab, find the '''"SIP Trunk Details"''' section.
: [2] Country: Select '''"CA"'''.
: [3] Trunk: Select '''"VoIP.ms"'''.

Now configure the trunk:
: [1] '''Name''': Customize as needed.
: [2] '''Default Route''': Leave it as '''"End calls"'''.
: [3] '''Limit To''': Choose the user group to limit trunk access, default is '''"DEFAULT"'''.
: [4] '''Outbound Rules''': Check '''"[x] Create an outbound rules for this SIP Trunk"''' if no outbound route is created yet.
: [5] '''Registrar/Server''': Enter the POP server matching your DID settings.
: [6] '''Main Trunk Number''': Enter the default DID for this trunk.
: [7] '''Simultaneous Calls''': Set trunk limit based on your 3CX license.
: [8] '''Authentication ID (SIP User ID)''': Enter sub-account username: ''eg: #####_SubAccount''.
: [9] '''Authentication Password''': Enter sub-account password.

== DID Numbers Tab ==

If using multiple DIDs:
: [1] Go to the '''"DID Numbers"''' tab.
: [2] Click '''"[+ Add]"''' button.
: [3] Type VoIP.ms DID '''without hyphen, dot, or parentheses'''.
: [4] Press '''Enter'''.

== Options Tab ==

If setting up outbound caller ID Name:
: [1] Go to the '''"Options"''' tab.
: [2] Click '''"Caller ID Control"'''.
: [3] Enter your Caller ID Name by following the requirements '''below'''.
<div style="margin-left:50px;
padding:8px;
border-left: 6px solid rgb(73, 111, 199, 0.8);
box-shadow: 5px 4px 4px 4px #f5f5f5;
background-color: rgba(73, 111, 199, 0.08);
width:70%;">

'''IMPORTANT - Caller ID Name'''<br/>
* Entering outbound Caller ID Name, it must be in '''CAPITAL LETTERS'''. ''This will appears more clearly/visible on some devices.'' <br/>
* You must '''NOT use any special characters'''. ''They will not be displayed and it prevents potential outbound call issues.'' <br/>
* Do '''NOT exceed 15 characters'''. ''Traditional canadian providers will '''not show more than 15 characters'''. We suggest shrinking or adapt the name.''<br/>
* Spaces are allowed in a caller ID name but count as a characters. ''(do not exceed 15 chars. in total)''.

</div>

Click '''"[SAVE]"''' button at the top.

== SMS Tab ==
===Service Limitations===

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 56, 45, 0.08);
width:70vw;">

🏗️ ''' Temporary Limitation'''
: ➜ '''Temporarily''' the current integration supports a maximum of '''160 characters per sent message'''. We are currently working on a way to permit more than 160 characters and will be available soon.
: ➜ Messages longer than 160 characters will result in an error, noticeable by a ❌ where your message. The error can be retrieved in your 3CX Admin console under '''Event Log'''.
</div>

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

⚠️ '''LIMITATIONS'''
: ➜ VoIP.ms supports ''Application-to-Person (A2P)'' delivery for business text messaging. However, requirements and fees vary due to industry regulations. The default limit is 100 SMS per day, which can be raised upon request. Contact us at '''messaging@voip.ms''' to start your verification process.
<br/>
: ➜ The limit of sending SMS/MMS using the '''Bearer token''' is limited to the '''same limit''' as our '''SOAP and REST/JSON API''' which is '''<u>100 per day</u>'''. For any modification to the limit, please contact our support team through the live chat or open a ticket via the customer portal.
: ➜ The SMS/MMS service is only available for <u>'''US and Canadian DID and Toll-Free Numbers'''</u> marked with the distinctive '''SMS/MMS Icon'''.
: ➜ The SMS/MMS service can only be used to send SMS/MMS Messages to <u>'''Canadian and US'''</u> 10 digit numbers at this time.
: ➜ We cannot guarantee that '''accents or special characters''' including non-Latin letters will be properly delivered.
: ➜ Public URL shorteners such as '''''bit.ly, tinyurl.com''''' etc, <font style="color:red">'''are widely restricted'''</font> by mobile providers as a response from the industry to combat SPAM campaigns and protect the consumer. Including this type of links into your messages '''<u>may result in a blocked or undelivered message</u>'''.
: ➜ VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS/MMS service from any account that may present marketing patterns/automation patterns/bulk sending or receiving.

</div>

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 NOTE '''
: When there is an error while sending a message, you will notice a ❌ where your message. You can retrieve the error details in your 3CX Admin console under '''Event Log'''.
</div>

===Generate a Bearer Token===

If you do not already have a "Bearer Token" generated that you have copied somewhere, click on "Generate New Token" to obtain one.
You will need to generate a new one.

: 1) Go to your VoIP.ms client portal '''"Main Menu"''' > '''"SOAP and REST/JSON API"'''
: 2) Click on '''"Enable"''' the API if its not already made.
: 3) Where '''Bearer Token''' click on '''Generate New Token'''

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 149, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 144, 45, 0.08);
width:70vw;">

🔐 '''ATTENTION: Store your generated token securely!''' ⚠️
: When you generate a Bearer Token, be sure to copy and store it in a secure place, as it won't be visible again after generation.
: It will only be available to be copied once it being generated. If you lost your Bearer Token, you need to regenerate a new one.'''

</div>

=== Add your Bearer Token to your 3CX trunk ===
Once the trunk is created and configured
:1) Go to the '''"SMS"''' tab.
:2) '''Enable SMS''': [x] Enable the service by checking the checkbox.
:3) In the '''"API Key"''' field, paste your VoIP.ms ''[[#Generate a Bearer Token|generated Bearer Token]]''.

:: [[File:VOIPMS_3CX_V20_SMS_TAB.png|none]]

:4) Then, copy the Webhook URL, you can click on the '''copy 📋 icon '''.

=== Set the 3CX trunk Webhook URL to your VoIP.ms DID ===
:1) Go to your VoIP.ms client portal and navigate to '''"DID Number"''' tab > '''"Manage DID"''' >
:2) click the '''yellow Edit button''' of your DID you wish to use for SMS.
:3) Scroll down to the SMS section until you see the '''Message Service (SMS/MMS)''' section.
:4) '''Message Service (SMS/MMS)''': [x] '''Enable'''
:5) In the '''"SMS/MMS Webhook URL (3CX)"''' field, paste the '''Webhook URL''' from the new 3CX trunk.
:6) '''Dialing Mode for SMS (3CX)''': Select '''NANPA'''

:: [[File:3CXv20_DID_Edit_SMS.png|700px]]

Done! Simply associate the DID to your Queue or your extension.

=[[#SMS_Tab | Configuring VoIP.ms SMS/MMS service]]=
[[#SMS_Tab |Click here to see the section SMS/MMS Tab]]

Catalaya - Orchid Link SBC

2026-06-23T18:32:48Z

Scomtois: moved Catalaya - Orchid Link SBC to Cataleya - Orchid Link SBC

#REDIRECT [[Cataleya - Orchid Link SBC]]

Cataleya - Orchid Link SBC

2026-06-23T18:32:48Z

Scomtois: moved Catalaya - Orchid Link SBC to Cataleya - Orchid Link SBC

[[File:Cataleya logo.png|400px|right|none]]

This article provides general configuration guidelines for connecting an '''Orchid Link Session Border Controller (SBC)''' with [[VoIP.ms]] SIP trunking. The configuration described below focuses on two Orchid Link zones:

:* '''Access Public''': VoIP.ms-facing side of the SBC.
:* '''Access Core''': customer network, SIP platform, IP-PBX, softswitch, or core-facing side of the SBC.

'''Audience:''' Network and voice engineers familiar with SIP, VoIP.ms accounts, and basic Orchid Link administration.

'''Vendor confirmation pending:''' Supported Orchid Link software versions and deployment models should be confirmed with Cataleya before adding a component information table to this article.

'''Note:''' The screenshots and field values shown in this article are examples only. Adjust the configuration according to your own network design, security policy, licensing, and Orchid Link deployment.

'''The Microsoft Teams section is a high-level overview and may require additional configuration and testing.'''

__TOC__

= Prerequisites =

Before starting the configuration, make sure you have the following information available:

:* Access to the Orchid Link administrative interface.
:* A configured customer and partition in Orchid Link.
:* SIP and media interfaces already created for both the public and core sides.
:* The VoIP.ms server or POP information that will be used by the trunk.
:* The trusted VoIP.ms source IP addresses or POPs that will be allowed through the SBC security rules.
:* SIP credentials or trunk information required for the VoIP.ms side.
:* The internal IP address and SIP port of the customer-side PBX, softswitch, or SIP platform.
:* Any required SIP Parameters, Transparency, Media, Call Admission Control, and Routing Policy profiles.

= Configuration Overview =

The Orchid Link SBC configuration is separated into two main zones:

:* Configure the '''Access Public''' zone first. This zone represents the VoIP.ms-facing side of the SBC.
:* Configure the '''Access Core''' zone after the Access Public zone. This zone represents the customer network or PBX-facing side of the SBC.

Once both zones are created, routing is configured so that calls can pass between the VoIP.ms side and the core-side SIP environment.

= Access Public Zone Configuration =

The Access Public zone is the side of the SBC that communicates with VoIP.ms. This zone should be configured first.

== Step 1 - General ==

In the '''General''' tab, configure the mandatory fields shown in the Orchid Link interface.

[[File:Orchid_Link_Access_Public_General.png|thumb|600px|Access Public zone - General tab]]

:* '''Name''': Enter a descriptive name for the zone, for example, '''VoIPMS_Public'''.
:* '''Customer''': Select the customer associated with this zone.
:* '''Partition''': Select the partition allocated to this customer.
:* '''Zone Type''': Select '''AccessPublic'''.
:* '''Time Zone''': Select the appropriate time zone for the customer.
:* '''Trunk Group Type''': Leave this set to '''None''', unless your deployment requires another value.
:* '''Service Level''': Keep the default service level, for example, '''Gold''', unless your policy requires otherwise.

== Step 2 - Signaling ==

In the '''Signaling''' step, select the SIP-related profiles and interfaces used for the VoIP.ms-facing side.

[[File:Orchid_Link_Access_Public_Signaling.png|thumb|600px|Access Public zone - Signaling step]]

:* '''SIP Profile''': Select the preconfigured SIP Parameters Profile from the drop-down list.
::* This profile is created under '''Profiles > SIP Parameters'''.

[[File:Orchid_Link_SIP_Parameters_Profile.png|thumb|600px|SIP Parameters Profile example]]

:* '''Transparency Profile''': Select the transparency profile created for this deployment.
::* This profile can be created under '''Profiles > Transparency'''.

[[File:Orchid_Link_Transparency_Profile.png|thumb|500px|Transparency Profile example]]

:* '''SIP Interface''': Select the SIP Interface configured for the public access side and add it to the zone.

== Step 3 - Media ==

In the '''Media''' step, select the media interface and media profile used for VoIP.ms traffic.

[[File:Orchid_Link_Access_Public_Media.png|thumb|600px|Access Public zone - Media step]]

:* '''Media Interface''': Select the Media Interface configured for the public access side and add it to the zone.
:* '''Media Profile''': Select the Media Profile used for VoIP.ms traffic.
::* The Media Profile is created under '''Profiles > Media'''.

[[File:Orchid_Link_Media_Profile.png|thumb|500px|Media Profile example]]

== Step 4 - Call Admission Control ==

In the '''Call Admission Control''' step, select the CAC profile that will control the number of concurrent sessions allowed for this zone.

[[File:Orchid_Link_CAC_Profile_Zone.png|thumb|450px|Zone CAC Profile example]]

:* '''CAC Profile Zone''': Select the CAC profile created for this zone.
::* The profile is created under '''Profiles > Call Admission Control'''.
::* Configure the session count according to the licensed capacity allocated for VoIP.ms traffic.
:* '''CAC Profile UE''': Select the UE CAC profile if you want to apply per-endpoint limits.

[[File:Orchid_Link_CAC_Profile_UE.png|thumb|450px|UE CAC Profile example]]

== Step 5 - Security ==

In the '''Security''' step, configure the allowed sources and any required security parameters according to your security policy.

[[File:Orchid_Link_Access_Public_Security.png|thumb|600px|Access Public zone - Security step]]

:* Restrict access to known and trusted VoIP.ms IP addresses or POPs whenever possible.
:* Avoid allowing unrestricted public SIP access unless this is specifically required and properly secured.
:* Apply any additional SBC security settings required by your deployment.

== Step 6 - Remote Peers ==

No configuration is required in this step for the Access Public zone.

:* Leave the default values unless a specific deployment requires otherwise.

== Step 7 - Routing ==

In the '''Routing''' step, define where traffic from the Access Public zone should be sent.

[[File:Orchid_Link_Access_Public_Routing.png|thumb|600px|Access Public zone - Routing step]]

:* '''Egress Zone''': Select the '''Access Core''' zone as the egress destination for this Access Public zone.

== Step 8 - Transcoding ==

If transcoding is not required, no configuration is needed in this step.

:* Configure transcoding only if codec interworking is required between VoIP.ms and the customer-side SIP environment.
:* The transcoding profile should match your network design and licensing.

= Access Core Zone Configuration =

The Access Core zone represents the customer network or PBX-facing side of the SBC. Configure this zone after the Access Public zone has been created.

== Step 1 - General ==

In the '''General''' tab, configure the core-side zone information.

:* '''Zone Type''': Select '''AccessCore'''.
:* '''Name''': Use a clear name that identifies the customer or core-side environment.
:* '''Customer''': Select the relevant customer.
:* '''Partition''': Select the partition allocated to the customer.
:* '''Time Zone''': Select the appropriate time zone.
:* '''Trunk Group Type''': Leave as '''None''', unless your deployment requires otherwise.
:* '''Service Level''': Use the appropriate service level for the customer or deployment.

== Step 2 - Signaling ==

In the '''Signaling''' step, configure the SIP interface and profiles used toward the customer core network.

:* '''SIP Interface''': Select the SIP Interface configured for the core side.
:* '''SIP Profile''': Select the appropriate SIP Parameters Profile for the core network.
::* This profile may be different from the profile used on the VoIP.ms-facing side.
:* '''Transparency Profile''': Select the appropriate transparency profile.
::* This profile can be created under '''Profiles > Transparency'''.

== Step 3 - Media ==

In the '''Media''' step, configure the media interface and media profile used toward the core network.

:* '''Media Interface''': Select the Media Interface configured for the core side.
:* '''Media Profile''': Select the Media Profile used for traffic toward the core network (customer network or PBX).

== Step 4 - Call Admission Control ==

In the '''Call Admission Control''' step, select the CAC profile used for the core-facing side.

:* '''CAC Profile Zone''': Select the core-side CAC Profile Zone.
::* Configure the session count according to the capacity you want to reserve for core-side calls.
:* '''CAC Profile UE''': Select the UE CAC profile if per-endpoint limits are required on the core side.

== Step 5 - Security ==

In the '''Security''' step, configure the allowed source addresses for the core-side signaling devices.

:* Only allow traffic from trusted internal networks, or core devices
:* Do not expose the core-side zone to untrusted networks unless this is part of the intended design and is properly secured.

== Step 6 - Remote Peers ==

In the '''Remote Peers''' step, configure the customer-side SIP endpoint information.

[[File:Orchid_Link_Access_Core_Remote_Peers.png|thumb|600px|Access Core zone - Remote Peers step]]

:* '''Remote Endpoints''': Configure the IP address and port of the remote core endpoint.
::* Examples include an IP-PBX, softswitch, SIP application server, or another SBC.
::* The Orchid Link SBC will send SIP signaling to these addresses.

== Step 7 - Routing ==

In the '''Routing''' step, select the routing policy that will be used for traffic toward the core side.

[[File:Orchid_Link_Access_Core_Routing.png|thumb|600px|Access Core zone - Routing step]]

:* '''Routing Policy''': Select the appropriate Session Routing Policy from the drop-down list.
::* The policy is configured under '''Policy Configuration > Session Routing Policies'''.

[[File:Orchid_Link_Session_Routing_Policy.png|thumb|600px|Session Routing Policy example]]

== Step 8 - Transcoding ==

Configure transcoding only if codec interworking is required between VoIP.ms and your core network.

:* If transcoding is not required, no changes are needed in this step.
:* If transcoding is required, select or create the appropriate transcoding profile according to the deployment requirements.

= Final Checklist =

After completing the configuration, review the following items:

:* The Access Public zone is configured as '''AccessPublic'''.
:* The Access Core zone is configured as '''AccessCore'''.
:* SIP and media interfaces are assigned to the correct zones.
:* The Access Public zone routes traffic toward the Access Core zone.
:* The Access Core zone has the correct remote endpoint configured.
:* The Session Routing Policy is selected and points to the intended routing behavior.
:* Security rules only allow trusted VoIP.ms sources and trusted core-side devices.
:* CAC limits match the licensed and intended concurrent call capacity.
:* Transcoding is configured only when required.

Catelaya - Orchid Link SBC

2026-06-23T18:32:26Z

Scomtois: moved Catelaya - Orchid Link SBC to Catalaya - Orchid Link SBC

#REDIRECT [[Catalaya - Orchid Link SBC]]

Cataleya - Orchid Link SBC

2026-06-23T18:32:26Z

Scomtois: moved Catelaya - Orchid Link SBC to Catalaya - Orchid Link SBC

Cataleya - Orchid Link SBC

2026-06-23T18:28:39Z

Scomtois: Created page with "none This article provides general configuration guidelines for connecting an '''Orchid Link Session Border Controller (SBC)''' with [[VoI..."

File:Orchid Link Session Routing Policy.png

2026-06-23T13:44:36Z

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File:Orchid Link Access Core Routing.png

2026-06-23T13:44:21Z

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File:Orchid Link Access Core Remote Peers.png

2026-06-23T13:44:08Z

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File:Orchid Link Access Public Routing.png

2026-06-23T13:43:57Z

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File:Orchid Link Access Public Security.png

2026-06-23T13:43:43Z

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File:Orchid Link CAC Profile UE.png

2026-06-23T13:43:27Z

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File:Orchid Link CAC Profile Zone.png

2026-06-23T13:43:12Z

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File:Orchid Link Media Profile.png

2026-06-23T13:42:58Z

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File:Orchid Link Access Public Media.png

2026-06-23T13:42:36Z

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File:Orchid Link Transparency Profile.png

2026-06-23T13:37:32Z

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File:Orchid Link SIP Parameters Profile.png

2026-06-23T13:37:14Z

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File:Orchid Link Access Public Signaling.png

2026-06-23T13:36:54Z

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File:Orchid Link Access Public General.png

2026-06-23T13:36:16Z

Scomtois:

Grandstream HT801v2

2026-06-18T20:33:39Z

Scomtois:

This guide explains how to configure a '''Grandstream HandyTone HT801v2''' for use with VoIP.ms services. The HT801v2 is a compact, single-port Analog Telephone Adapter (ATA) that allows users to connect a traditional analog phone or fax machine to the VoIP.ms network. Along with the two-port HT802v2 and their previous-generation counterparts, it has remained one of the [[ATA Devices#Most Popular ATA Devices|most commonly used ATA devices]] among VoIP.ms customers in recent years, based on internal network analytics.

=== Device details ===
The HT801v2 is an easy-to-use, single-port Analog Telephone Adapter (ATA) designed for residential users, small offices, and large-scale commercial IP voice deployments.
Featuring an ultra-compact form factor measuring 10×10×2.95cm and weighing 102g, the device provides a simple and cost-effective way to integrate a traditional analog telephone or fax machine into a modern VoIP environment. Despite its small size, the HT801v2 offers a comprehensive feature set, including:

* Support for a single SIP profile through a dedicated FXS (RJ11) port, compatible with both [[Order a DID Number|VoIP.ms DIDs]] and [[Sub Accounts|sub-accounts]]
* Incoming and outgoing [[Caller ID]] with support for a wide range of caller ID formats
* A 10/100 Mbps auto-sensing Ethernet (RJ45) network port
* Automatic SIP server failover to maintain service continuity should the primary server become unavailable
* Advanced telephony features including call transfer, [[Call Forwarding|call forwarding]], call waiting, hold, do-not-disturb, message waiting indication, multilingual prompts, and flexible dialplans
* Optional [[Call Encryption - TLS/SRTP|TLS and SRTP encryption protocols]] to secure calls and accounts
* [[Virtual Fax|Fax over IP capability]] using T.38 fax relay with automatic fallback to G.711 pass-through

<gallery caption="Grandstream HandyTone HT801v2 photos (click to expand)" heights="224" perrow="3" widths="224" style="margin: 1.5em auto; text-align: center; width: fit-content;">
File:Grandstream HandyTone HT801v2 - Front View.png|Front view
File:Grandstream HandyTone HT801v2 - Top View.png|Top view, showing status LEDs for power, network connection, and in-use indicator for both analog phone ports
File:Grandstream HandyTone HT801v2 - Rear View.png|Rear view with Ethernet and dual phone ports, USB Micro-B power input and pinhole reset
</gallery>

== Configuring your device ==
[[File:HT801v2 FXS1 General Settings AccountRegistration.png|thumb|right|512px|FXS Port 1 General Settings > Account Registration]]

=== Prerequisites ===
* Grandstream HT801v2
:* Latest firmware installed ([https://www.grandstream.com/support/firmware|check Grandstream site]) ''(We you use firmware version 1.0.11.4 or later.)''
:* VoIP.ms account with a configured DID and a sub-account
:* Internet access via Ethernet
:* Phone or fax device connected to FXS Port 1

=== Logging into the web interface ===
: Connect the HT801v2 to your network and power it on.
::* Pick up the connected phone, dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">* * * 0 2</code> (Star Star Star 0 2)

The ATA will read its IP address.
Open a browser and go to that IP (e.g., http://192.168.1.50).
: Login
:* Username: <code>admin</code>
:* Password: found on device label (or as set by you)

=== General Settings (FXS PORT1) ===
[[File:HT801v2 FXS1 General Settings NetworkSettings.png|thumb|right|512px|FXS Port 1 General Settings > Network Settings]]

==== Account Registration ====
:* '''Account Active:''' ✅ Checked
:* '''Primary SIP Server''': popserver.voip.ms (or your nearest VoIP.ms POP) ''eg: montreal1.voip.ms''
:* '''Prefer Primary SIP Server''' Yes
:* '''Outbound Proxy''' popserver.voip.ms (Same as the Primary SIP Server)
:* ''Backup Outbound Proxy This is optional you can enter a different VoIP.ms PoP Server location. eg: toronto2.voip.ms''
::* ''Prefer Primary Outbound Proxy: Check the box, only if you entered a backup Outbound Proxy''
:* '''SIP User ID''' Your VoIP.ms sub-account username
:* '''SIP Authenticate ID''' Same as SIP User ID
:* '''SIP Authentication Password''' Your sub-account password

==== Network Settings ====
:* '''DNS Mode''': A Record
:* '''NAT Traversal''' Keep-Alive
: Click '''Save and Apply'''.

=== SIP Settings Tab ===
[[File:Grandstream HT801v2 FXS Port1 Advanced Settings.png|thumb|right|512px|FXS Port 1 Advanced Settings > Security Settings]]

==== SIP Basic settings ====
:* '''SIP Registration''': Yes
:* '''Register Expiration''': 2 (minutes)
:* '''Enable SIP OPTIONS Keep Alive''': Yes
:* Local SIP Port: 5060 (change if multiple ATAs)

: Disable extra headers:
:* '''Privacy Header''' → No
:* '''P-Preferred-Identity Header''' → No
:* '''P-Access-Network-Info''' → No
:* '''P-Emergency-Info''' → No

==== Codec Settings Tab ====
: '''Preferred Vocoder''' (in all sessions): G722, (G.711 μ-law) then set G711 for the rest.

==== Advanced Settings ====
: '''Check SIP User ID for Incoming INVITE''': Enabled
: '''Allow incoming SIP messages from SIP proxy only''': Enabled

==== Faxing ====
[[File:Grandstream HT80v2 FXS Port1 Call Freatures Settings.png|thumb|right|512px|FXS Port 1 Call Features Settings]]
: Ensure options like Enable‑Call‑Waiting, Caller‑ID, and Tone are unchecked — avoiding disruptions during fax sessions.
: Security Settings: Enable Allow incoming packets from SIP proxy only to restrict access to VoIP.ms servers.

: Under '''Call Settings'''
: '''Disable Local Call Features''': Yes
:* '''Enable-Call-Waiting''': uncheck
:* '''Enable-Call-Waiting-Caller-ID''': uncheck
:* '''Enable-Call-Waiting-Tone''': uncheck

: Under '''Call Features Settings''':
:* Set ''' Enable Local Call Features''' '''No''' to prevent unintended feature code usage via the fax machine.

: Under '''Codec Settings''':
::* '''Fax Mode''': T.38 (if you are using it for faxing with a VoIP.ms FAX POP server)
::* '''T.38 Max Bit Rat'''e: 14400 bps
::* '''Jitter Buffer''': Fixed, High

: Under '''Analog Signal Line Configuration'''
::* '''Enable Line Echo Canceller (LEC)''' → Check

== Testing ==
: Reboot the ATA after applying settings.
: Check '''Status tab''' → Registration should be '''Yes'''.
: Dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">4443</code> for a Voice Echo Test.

== External links ==
* [https://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht801v2 Official product page]
* [https://www.grandstream.com/hubfs/Product_Documentation/HT801%20V2/Datasheet_HT801_V2_English.pdf Product datasheet and specifications] (PDF)
* [https://documentation.grandstream.com/article-categories/ht8xx-series Grandstream Documentation Center]
** [https://documentation.grandstream.com/knowledge-base/ht801-v2-quick-installation-guide/ HT801v2 Quick Installation Guide]
** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-administration-guide/ HT801v2 Administration Guide]
** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-user-guide/ HT801v2 User Guide]
* [https://www.grandstream.com/support/firmware Firmware download site]

[[Category:Analog Telephone Adapters]]

File:HT801v2 FXS1 General Settings AccountRegistration.png

2026-06-18T20:14:09Z

Scomtois:

Grandstream HT801v2

2026-06-18T19:52:45Z

Scomtois: /* Device details */

This guide is intended to assist with the process of configuring a '''Grandstream HandyTone HT801v2''' for use with [[VoIP.ms]] services. This device and its single-line counterpart (HT801v2)—along with their "v1" predecessors—have been the [[ATA Devices#Most Popular ATA Devices|most widely-used ATAs]] by our customers in recent years, according to internal network analytics.

=== Device details ===
The HT801v2 is an easy-to-use, single-port Analog Telephone Adapter (ATA) designed for residential users, small offices, and large-scale commercial IP voice deployments.
Featuring an ultra-compact form factor measuring 10×10×2.95cm and weighing 102g, the device provides a simple and cost-effective way to integrate a traditional analog telephone or fax machine into a modern VoIP environment. Despite its small size, the HT801v2 offers a comprehensive feature set, including:

* Support for a single SIP profile through a dedicated FXS (RJ11) port, compatible with both [[Order a DID Number|VoIP.ms DIDs]] and [[Sub Accounts|sub-accounts]]
* Incoming and outgoing [[Caller ID]] with support for a wide range of caller ID formats
* A 10/100 Mbps auto-sensing Ethernet (RJ45) network port
* Automatic SIP server failover to maintain service continuity should the primary server become unavailable
* Advanced telephony features including call transfer, [[Call Forwarding|call forwarding]], call waiting, hold, do-not-disturb, message waiting indication, multilingual prompts, and flexible dialplans
* Optional [[Call Encryption - TLS/SRTP|TLS and SRTP encryption protocols]] to secure calls and accounts
* [[Virtual Fax|Fax over IP capability]] using T.38 fax relay with automatic fallback to G.711 pass-through

<gallery caption="Grandstream HandyTone HT801v2 photos (click to expand)" heights="224" perrow="3" widths="224" style="margin: 1.5em auto; text-align: center; width: fit-content;">
File:Grandstream HandyTone HT801v2 - Front View.png|Front view
File:Grandstream HandyTone HT801v2 - Top View.png|Top view, showing status LEDs for power, network connection, and in-use indicator for both analog phone ports
File:Grandstream HandyTone HT801v2 - Rear View.png|Rear view with Ethernet and dual phone ports, USB Micro-B power input and pinhole reset
</gallery>

== Configuring your device ==
[[File:HT801v2 FXS1 General Settings AccountRegistration.png|thumb|right|512px|FXS Port 1 General Settings > Account Registration]]

=== Prerequisites ===
* Grandstream HT801v2
:* Latest firmware installed ([https://www.grandstream.com/support/firmware|check Grandstream site]) ''(We you use firmware version 1.0.11.4 or later.)''
:* VoIP.ms account with a configured DID and a sub-account
:* Internet access via Ethernet
:* Phone or fax device connected to FXS Port 1

=== Logging into the web interface ===
: Connect the HT801v2 to your network and power it on.
::* Pick up the connected phone, dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">* * * 0 2</code> (Star Star Star 0 2)

The ATA will read its IP address.
Open a browser and go to that IP (e.g., http://192.168.1.50).
: Login
:* Username: <code>admin</code>
:* Password: found on device label (or as set by you)

=== General Settings (FXS PORT1) ===
[[File:HT801v2 FXS1 General Settings NetworkSettings.png|thumb|right|512px|FXS Port 1 General Settings > Network Settings]]

==== Account Registration ====
:* '''Account Active:''' ✅ Checked
:* '''Primary SIP Server''': popserver.voip.ms (or your nearest VoIP.ms POP) ''eg: montreal1.voip.ms''
:* '''Prefer Primary SIP Server''' Yes
:* '''Outbound Proxy''' popserver.voip.ms (Same as the Primary SIP Server)
:* ''Backup Outbound Proxy This is optional you can enter a different VoIP.ms PoP Server location. eg: toronto2.voip.ms''
::* ''Prefer Primary Outbound Proxy: Check the box, only if you entered a backup Outbound Proxy''
:* '''SIP User ID''' Your VoIP.ms sub-account username
:* '''SIP Authenticate ID''' Same as SIP User ID
:* '''SIP Authentication Password''' Your sub-account password

==== Network Settings ====
:* '''DNS Mode''': A Record
:* '''NAT Traversal''' Keep-Alive
: Click '''Save and Apply'''.

=== SIP Settings Tab ===
[[File:Grandstream HT801v2 FXS Port1 Advanced Settings.png|thumb|right|512px|FXS Port 1 Advanced Settings > Security Settings]]

==== SIP Basic settings ====
:* '''SIP Registration''': Yes
:* '''Register Expiration''': 2 (minutes)
:* '''Enable SIP OPTIONS Keep Alive''': Yes
:* Local SIP Port: 5060 (change if multiple ATAs)

: Disable extra headers:
:* '''Privacy Header''' → No
:* '''P-Preferred-Identity Header''' → No
:* '''P-Access-Network-Info''' → No
:* '''P-Emergency-Info''' → No

==== Codec Settings Tab ====
: '''Preferred Vocoder''' (in all sessions): G722, (G.711 μ-law) then set G711 for the rest.

==== Advanced Settings ====
: '''Check SIP User ID for Incoming INVITE''': Enabled
: '''Allow incoming SIP messages from SIP proxy only''': Enabled

==== Faxing ====
[[File:Grandstream HT80v2 FXS Port1 Call Freatures Settings.png|thumb|right|512px|FXS Port 1 Call Features Settings]]
: Ensure options like Enable‑Call‑Waiting, Caller‑ID, and Tone are unchecked — avoiding disruptions during fax sessions.
: Security Settings: Enable Allow incoming packets from SIP proxy only to restrict access to VoIP.ms servers.

: Under '''Call Settings'''
: '''Disable Local Call Features''': Yes
:* '''Enable-Call-Waiting''': uncheck
:* '''Enable-Call-Waiting-Caller-ID''': uncheck
:* '''Enable-Call-Waiting-Tone''': uncheck

: Under '''Call Features Settings''':
:* Set ''' Enable Local Call Features''' '''No''' to prevent unintended feature code usage via the fax machine.

: Under '''Codec Settings''':
::* '''Fax Mode''': T.38 (if you are using it for faxing with a VoIP.ms FAX POP server)
::* '''T.38 Max Bit Rat'''e: 14400 bps
::* '''Jitter Buffer''': Fixed, High

: Under '''Analog Signal Line Configuration'''
::* '''Enable Line Echo Canceller (LEC)''' → Check

== Testing ==
: Reboot the ATA after applying settings.
: Check '''Status tab''' → Registration should be '''Yes'''.
: Dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">4443</code> for a Voice Echo Test.

== External links ==
* [https://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht801v2 Official product page]
* [https://www.grandstream.com/hubfs/Product_Documentation/HT801%20V2/Datasheet_HT801_V2_English.pdf Product datasheet and specifications] (PDF)
* [https://documentation.grandstream.com/article-categories/ht8xx-series Grandstream Documentation Center]
** [https://documentation.grandstream.com/knowledge-base/ht801-v2-quick-installation-guide/ HT801v2 Quick Installation Guide]
** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-administration-guide/ HT801v2 Administration Guide]
** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-user-guide/ HT801v2 User Guide]
* [https://www.grandstream.com/support/firmware Firmware download site]

[[Category:Analog Telephone Adapters]]

File:Grandstream HandyTone HT801v2 - Rear View.png

2026-06-18T19:40:53Z

Scomtois:

File:Grandstream HandyTone HT801v2 - Top View.png

2026-06-18T19:40:45Z

Scomtois:

File:Grandstream HandyTone HT801v2 - Front View.png

2026-06-18T19:40:33Z

Scomtois:

Grandstream HT801v2

2026-06-18T16:00:58Z

Scomtois: Created page with "This guide is intended to assist with the process of configuring a '''Grandstream HandyTone HT801v2''' for use with VoIP.ms services. This device and its single-line counterp..."

This guide is intended to assist with the process of configuring a '''Grandstream HandyTone HT801v2''' for use with [[VoIP.ms]] services. This device and its single-line counterpart (HT801v2)—along with their "v1" predecessors—have been the [[ATA Devices#Most Popular ATA Devices|most widely-used ATAs]] by our customers in recent years, according to internal network analytics.

=== Device details ===
The HT801v2 is an entry-level model intended for residential and light SOHO (Small Office/Home Office) installations, capable of leveraging a single network link into two simultaneously usable analog phone ports. The device itself is very compact, measuring 10×10×2.95cm and weighing 114g, and utilizes a ubiquitous USB Micro-B port for AC power input (expecting 5VDC/1A)—allowing it to be powered by all but the most feeble of mobile phone chargers manufactured in the last decade. Despite its small form factor it provides a broad range of features, including:
* Support for two SIP profiles, compatible with both [[Order a DID Number|VoIP.ms DIDs]] and [[Sub Accounts|sub-accounts]], each mapped to a dedicated FXS (RJ11) phone port
* Incoming and outgoing [[Caller ID]] with support for custom formatting and ringtones
* A T-Base10/100 Ethernet (RJ45) port
* Optional [[Call Encryption - TLS/SRTP|TLS and SRTP encryption protocols]] to secure calls and accounts
* [[Audio Conferencing|Three-way call conferencing]]
* Call transfers, [[Call Forwarding|forwarding]], hold and do-not-disturb
* SIP server failover to ensure your device stays connected to the service even if the default server goes offline
* [[Virtual Fax|Fax over IP capability]] (using either T.38 or G.711 passthrough)

<gallery caption="Grandstream HandyTone HT801v2 photos (click to expand)" heights="224" perrow="3" widths="224" style="margin: 1.5em auto; text-align: center; width: fit-content;">
File:Grandstream HandyTone HT801v2 - Front View.png|Front view
File:Grandstream HandyTone HT801v2 - Top View.png|Top view, showing status LEDs for power, network connection, and in-use indicator for both analog phone ports
File:Grandstream HandyTone HT801v2 - Rear View.png|Rear view with Ethernet and dual phone ports, USB Micro-B power input and pinhole reset
</gallery>

== Configuring your device ==
[[File:HT801v2 FXS1 General Settings AccountRegistration.png|thumb|right|512px|FXS Port 1 General Settings > Account Registration]]

=== Prerequisites ===
* Grandstream HT801v2
:* Latest firmware installed ([https://www.grandstream.com/support/firmware|check Grandstream site]) ''(We you use firmware version 1.0.11.4 or later.)''
:* VoIP.ms account with a configured DID and a sub-account
:* Internet access via Ethernet
:* Phone or fax device connected to FXS Port 1

=== Logging into the web interface ===
: Connect the HT801v2 to your network and power it on.
::* Pick up the connected phone, dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">* * * 0 2</code> (Star Star Star 0 2)

The ATA will read its IP address.
Open a browser and go to that IP (e.g., http://192.168.1.50).
: Login
:* Username: <code>admin</code>
:* Password: found on device label (or as set by you)

=== General Settings (FXS PORT1) ===
[[File:HT801v2 FXS1 General Settings NetworkSettings.png|thumb|right|512px|FXS Port 1 General Settings > Network Settings]]

==== Account Registration ====
:* '''Account Active:''' ✅ Checked
:* '''Primary SIP Server''': popserver.voip.ms (or your nearest VoIP.ms POP) ''eg: montreal1.voip.ms''
:* '''Prefer Primary SIP Server''' Yes
:* '''Outbound Proxy''' popserver.voip.ms (Same as the Primary SIP Server)
:* ''Backup Outbound Proxy This is optional you can enter a different VoIP.ms PoP Server location. eg: toronto2.voip.ms''
::* ''Prefer Primary Outbound Proxy: Check the box, only if you entered a backup Outbound Proxy''
:* '''SIP User ID''' Your VoIP.ms sub-account username
:* '''SIP Authenticate ID''' Same as SIP User ID
:* '''SIP Authentication Password''' Your sub-account password

==== Network Settings ====
:* '''DNS Mode''': A Record
:* '''NAT Traversal''' Keep-Alive
: Click '''Save and Apply'''.

=== SIP Settings Tab ===
[[File:Grandstream HT801v2 FXS Port1 Advanced Settings.png|thumb|right|512px|FXS Port 1 Advanced Settings > Security Settings]]

==== SIP Basic settings ====
:* '''SIP Registration''': Yes
:* '''Register Expiration''': 2 (minutes)
:* '''Enable SIP OPTIONS Keep Alive''': Yes
:* Local SIP Port: 5060 (change if multiple ATAs)

: Disable extra headers:
:* '''Privacy Header''' → No
:* '''P-Preferred-Identity Header''' → No
:* '''P-Access-Network-Info''' → No
:* '''P-Emergency-Info''' → No

==== Codec Settings Tab ====
: '''Preferred Vocoder''' (in all sessions): G722, (G.711 μ-law) then set G711 for the rest.

==== Advanced Settings ====
: '''Check SIP User ID for Incoming INVITE''': Enabled
: '''Allow incoming SIP messages from SIP proxy only''': Enabled

==== Faxing ====
[[File:Grandstream HT80v2 FXS Port1 Call Freatures Settings.png|thumb|right|512px|FXS Port 1 Call Features Settings]]
: Ensure options like Enable‑Call‑Waiting, Caller‑ID, and Tone are unchecked — avoiding disruptions during fax sessions.
: Security Settings: Enable Allow incoming packets from SIP proxy only to restrict access to VoIP.ms servers.

: Under '''Call Settings'''
: '''Disable Local Call Features''': Yes
:* '''Enable-Call-Waiting''': uncheck
:* '''Enable-Call-Waiting-Caller-ID''': uncheck
:* '''Enable-Call-Waiting-Tone''': uncheck

: Under '''Call Features Settings''':
:* Set ''' Enable Local Call Features''' '''No''' to prevent unintended feature code usage via the fax machine.

: Under '''Codec Settings''':
::* '''Fax Mode''': T.38 (if you are using it for faxing with a VoIP.ms FAX POP server)
::* '''T.38 Max Bit Rat'''e: 14400 bps
::* '''Jitter Buffer''': Fixed, High

: Under '''Analog Signal Line Configuration'''
::* '''Enable Line Echo Canceller (LEC)''' → Check

== Testing ==
: Reboot the ATA after applying settings.
: Check '''Status tab''' → Registration should be '''Yes'''.
: Dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">4443</code> for a Voice Echo Test.

== External links ==
* [https://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht801v2 Official product page]
* [https://www.grandstream.com/hubfs/Product_Documentation/HT801%20V2/Datasheet_HT801_V2_English.pdf Product datasheet and specifications] (PDF)
* [https://documentation.grandstream.com/article-categories/ht8xx-series Grandstream Documentation Center]
** [https://documentation.grandstream.com/knowledge-base/ht801-v2-quick-installation-guide/ HT801v2 Quick Installation Guide]
** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-administration-guide/ HT801v2 Administration Guide]
** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-user-guide/ HT801v2 User Guide]
* [https://www.grandstream.com/support/firmware Firmware download site]

[[Category:Analog Telephone Adapters]]

File:Grandstream HT80v2 FXS Port1 Call Freatures Settings.png

2026-06-18T16:00:44Z

Scomtois:

File:Grandstream HT801v2 FXS Port1 Advanced Settings.png

2026-06-18T15:59:10Z

Scomtois:

File:HT801v2 FXS1 General Settings NetworkSettings.png

2026-06-18T15:57:26Z

Scomtois:

Mitel 6900 Series SIP Phones

2026-06-13T02:26:15Z

Scomtois: /* Basic SIP Authentication Settings */

[[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>

Mitel 6900 Series SIP Phones

2026-06-13T02:23:43Z

Scomtois: /* Basic SIP Network Settings */

[[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 max! Some of regular Canadian providers will not show more than 15 characters. We suggest shrinking or adapt your caller ID.'''
:'''Spaces are allowed in a caller id name.'''
</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>

Mitel 6900 Series SIP Phones

2026-06-13T02:23:16Z

Scomtois: /* Basic SIP Network Settings */

[[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 max! Some of regular Canadian providers will not show more than 15 characters. We suggest shrinking or adapt your caller ID.'''
:'''Spaces are allowed in a caller id name.'''
</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>''
::* Use the same server as the Proxy Server.
:'''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>

Mitel 6900 Series SIP Phones

2026-06-13T02:22:14Z

Scomtois: /* Basic SIP Authentication Settings */

[[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 max! Some of regular Canadian providers will not show more than 15 characters. We suggest shrinking or adapt your caller ID.'''
:'''Spaces are allowed in a caller id name.'''
</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>''
::* Use the same server as the Proxy Server.
:'''Registrar Port:''' <code>5060</code> ''(This is the default SIP port.)''

:'''Registration Period:''' <code>120</code> ''secondes''
:: Recommended registration expiry value.

== 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>

Mitel 6900 Series SIP Phones

2026-06-13T02:21:54Z

Scomtois: /* Basic SIP Authentication Settings */

[[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|300px]]
:'''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 max! Some of regular Canadian providers will not show more than 15 characters. We suggest shrinking or adapt your caller ID.'''
:'''Spaces are allowed in a caller id name.'''
</p></div>

:: [[File:Mitel_6900_Line1_Basic_SIP_Auth.png|none|300px]]

== 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>''
::* Use the same server as the Proxy Server.
:'''Registrar Port:''' <code>5060</code> ''(This is the default SIP port.)''

:'''Registration Period:''' <code>120</code> ''secondes''
:: Recommended registration expiry value.

== 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>

File:Mitel 6900 Line1 Basic SIP Network.png

2026-06-13T02:21:41Z

Scomtois:

Mitel 6900 Series SIP Phones

2026-06-13T02:20:31Z

Scomtois: Created page with "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..."

[[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|300px]]
:'''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 max! Some of regular Canadian providers will not show more than 15 characters. We suggest shrinking or adapt your caller ID.'''
:'''Spaces are allowed in a caller id name.'''
</p></div>

::: [[File:Mitel_6900_Line1_Basic_SIP_Auth.png|none|300px]]

== 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>''
::* Use the same server as the Proxy Server.
:'''Registrar Port:''' <code>5060</code> ''(This is the default SIP port.)''

:'''Registration Period:''' <code>120</code> ''secondes''
:: Recommended registration expiry value.

== 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>

File:Mitel 6900 Line1 Basic SIP Auth.png

2026-06-13T01:14:01Z

Scomtois:

File:Mitel 6900 ScreenName.png

2026-06-12T22:06:20Z

Scomtois:

File:Mitel 6900 Series.png

2026-06-12T21:06:00Z

Scomtois:

Ringotel

2026-05-14T01:30:53Z

Scomtois: /* Connect to VoIP.ms (PBX) */

[[File:VoIP.ms_Ringotel_Logo.png|right|300px]]
'''Ringotel''' is a cloud-based softphone and unified communications platform designed to work with standard SIP-based VoIP systems.
It provides features such as voice calling, messaging, presence, and video across desktop and mobile devices, while simplifying deployment and management for service providers and enterprises. Ringotel operates as a secure VoIP tunnel that routes SIP traffic between users and an existing PBX or VoIP system, reducing infrastructure complexity and improving reliability.

Ringotel supports centralized user provisioning and management through an admin portal or API, allowing automated onboarding, configuration, and device setup. It also offers integration options with CRM systems, messaging services, and business tools, and includes features such as push notifications, call recording, visual voicemail, BLF (busy lamp field), and rich messaging.

The platform is compatible with most standard SIP PBXs and telephony systems, enabling VoIP providers and organizations to extend their existing voice infrastructure to softphone users without deploying additional VPNs or session border controllers.

----

== Pre-Requisites ==
Before configuring Ringotel Shell, ensure the following requirements are met:

* You must have a Ringotel account. After account creation, you will be redirected to the softphone provisioning portal.
* Prepare Organization details - [https://help.ringotel.com/en/articles/11187474-softphone-provisioning Ringotel KB]:
** Organization Name
** Domain (Ringotel subdomain)
** Region closest to your users.
* Prepare connection details:
** '''VoIP.ms''' — Create a Sub‑Account with User/Password authentication.

== VoIP.ms Sub-Account Specification ==
:: [[File:VoIP.ms_Ringotel_SubAccount-Creation.png|500px|none]]

----
<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 Notes'''
: 🧱 NAT / Firewall Notes
: If your PBX is behind a NAT firewall, you may need to allow incoming connections from Ringotel server IP addresses in your firewall or security appliance settings. [https://help.ringotel.com/en/articles/11190222-configure-firewall-to-work-with-ringotel See related firewall documentation for details].

</div>
----

== Create an Organization ==
: 1. Go to '''Organizations''' in the Ringotel Shell portal.
: 2. Click '''Create Organization'''.
: 3. Fill required fields:
::* Organization Name
::* Domain (unique subdomain)
::* Region (closest to user base)
: 4. (Optional) Configure onboarding language, tags, and additional settings.
: 5. Click '''Next''' to complete the organization setup.

== Create a Connection ==
A Connection defines how Ringotel communicates with your PBX or SIP trunk provider.

=== Connect to VoIP.ms (SIP Trunk Provider) ===
:1. Within the organization, click '''Create Connection'''.
:2. Select '''Connect to a SIP Trunk Provider'''.
:3. Enter the VoIP.ms details:
::* POP server address
::* Sub‑Account Username
::* Sub‑Account Password
::* Check '''Registration'''
::* Use the same Sub‑Account username for: Number, Username, and Authname fields
:4. Click '''Next''' to finish.
:: [[File:VoIPms_Ringotel_CreateConnectionSIPTrunk.png|500px|none]]

=== Connect to VoIP.ms (PBX) ===
:1. Select '''Connect to a PBX server'''.
:2. Complete the following field:
::* '''Connection name''': Define a connection name for your reference.
::* '''Protocol''': Select UDP or TCP. ''(To use SIP traffic encryption, select SIP-TLS)''
::: *If you have selected SIP-TLS for SIP traffic encryption, you need to enable SIP traffic encryption on your Sub Account)
::* '''Domain or IP address''': indicate your preferred pop server.
::* ''Country: Optional. ℹ️ To use this feature, your DID needs have "Dial Mode" set to "E164". ''
:::[[File:VoIPms_Ringotel_CreateConnectionSIPPBX.png|550px]]

:3 Click '''Advanced settings'''
::* '''Audio codecs''' Check the following codecs '''G.711 Ulaw'''
::* '''Miscellanous''' '''Display Name''': Specify your Outbound CallerID Name
::* '''Registration TTL''': indicate '''120''' secondes
::: [[File:VoIPms_Ringotel_CreateConnectionSIPPBX_Advanced_AudioCodecs.png|350px]][[File:VoIPms_Ringotel_CreateConnectionSIPPBX_Advanced_Misc.png|350px]]
:4. Save the connection.

----

== 4. Add Users ==
:1. Navigate to '''Users''' inside your organization.
:2. Click '''Add User'''.
:3. Provide:
::* Name
::* Email (for onboarding instructions)
::* Assigned Connection (if multiple connections are created)
::* SIP credentials (auto-filled when PBX integration is active)
:4. Save the user to automatically generate onboarding links.

----

= VoIP.ms SMS/MMS Integration with Ringotel =
:This section describes how you can enable the '''SMS and MMS messaging''' for your DID numbers using '''Ringotel'''.
:Once configured, SMS and MMS messages sent to your VoIP.ms DID can be sent and received directly from Ringotel applications.

== Overview ==
VoIP.ms supports SMS and MMS on compatible DID numbers. By integrating VoIP.ms with Ringotel, you can:
:* Send and receive SMS/MMS using VoIP.ms DID numbers
:* Assign numbers to individual users or teams
:* Manage conversations from desktop and mobile applications

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 149, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 144, 45, 0.08);
width:70vw;">

🔐 '''IMPORTANT''' ⚠️
: : '''Note:''' SMS via integrations requires the '''Ringotel Pro''' plan.
</div>

== Prerequisites ==
Before starting, ensure you have:
* An active '''VoIP.ms account'''
* One or more '''SMS-capable DID numbers'''
* An active '''Ringotel organization''' with Pro features enabled
* Administrator access to both VoIP.ms and Ringotel

== Enable SMS Integration in Ringotel ==

# Log in to the '''Ringotel Admin Portal'''
# Open your organization
# Navigate to '''Features'''
# Enable '''SMS via Integrations'''
# Save changes
# Open the '''Integrations''' tab
# Select '''VoIP.ms''' and click '''Set up Integration'''
# Copy the '''Webhook URL''' (required for VoIP.ms configuration)

== Configure the DID Number in VoIP.ms ==

=== Enable SMS/MMS on the DID ===

# Log in to the '''VoIP.ms customer portal'''
# Go to '''DID Numbers → Manage DID(s)'''
# Select the DID number and click '''Edit'''
# Under '''Message Service (SMS/MMS)''':
## Enable '''SMS/MMS'''
# Enable '''SMS/MMS URL Callback'''
# Paste the Ringotel Webhook URL and append the following parameters:

<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58%;">

: <nowiki>https://shell.ringotel.co/services/Voipms/aXXXXXXXXXXXXXXXXXXXX?id={ID}&to={TO}&from={FROM}&message={MESSAGE}&files={MEDIA}</nowiki>

: The final URL should look similar to the following:
: <nowiki>https://shell.ringotel.co/services/Voipms/a15945585810414401989?id={ID}&to={TO}&from={FROM}&message={MESSAGE}&files={MEDIA}</nowiki>

</div>

: 7. Save the DID settings

== Enable VoIP.ms API Access ==
Ringotel communicates with VoIP.ms using the VoIP.ms API.

:1. In the VoIP.ms portal, navigate to '''Main Menu → SOAP and REST/JSON API'''
:: [[File:VOIPMS_Main_Menu_RestJsonAPI.png|500px|none]]

:2. Set an '''API Password'''
:3. Enable the API if it is not already enabled
:4. Under '''Enable IP Addresses''', add the following Ringotel IP addresses:

<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58%;">

: 54.145.196.146, 18.130.161.112, 3.0.222.77

</div>
:5. Save the API settings

:: [[File:VOIPMS_API_Ringotel-IPs.png|500px|none]]

== Complete the Integration in Ringotel ==
:After setting up your API password and the IP addresses in your VoIP.ms portal, return to your Ringotel Admin portal to complete the connection.
:## Your '''VoIP.ms login email'''
:## Your '''VoIP.ms API password'''
:# Click '''Save & Continue'''

:: [[File:VoIPms_Ringotel_CreateConnectionSIPTrunk.png|500px|none]]

== Add SMS Numbers and Assign Users ==
# Add your VoIP.ms DID numbers with SMS enabled
# Assign users who can send and receive messages
# Enter phone numbers in '''national format''' (for example, <code>5033889028</code> for US/Canada)
# Configure optional settings:
## '''Session stickiness''' (keeps conversations assigned to the same user)
## '''Number reformatting''' (prevents duplicate conversation threads)
## '''Sender name in messages'''
## '''Auto-replies''' and opt-out messages
# Save the configuration

Users may need to '''log out and log back in''' to access SMS features.

== Notes and Limitations ==

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

⚠️ '''LIMITATIONS'''
: ➜ SMS and MMS availability depends on the DID type and country
: ➜ MMS support may vary by carrier
: ➜ Short codes and toll-free numbers may require additional configuration

</div>

Ringotel

2026-05-14T01:05:32Z

Scomtois: /* Connect to VoIP.ms (PBX) */

[[File:VoIP.ms_Ringotel_Logo.png|right|300px]]
'''Ringotel''' is a cloud-based softphone and unified communications platform designed to work with standard SIP-based VoIP systems.
It provides features such as voice calling, messaging, presence, and video across desktop and mobile devices, while simplifying deployment and management for service providers and enterprises. Ringotel operates as a secure VoIP tunnel that routes SIP traffic between users and an existing PBX or VoIP system, reducing infrastructure complexity and improving reliability.

Ringotel supports centralized user provisioning and management through an admin portal or API, allowing automated onboarding, configuration, and device setup. It also offers integration options with CRM systems, messaging services, and business tools, and includes features such as push notifications, call recording, visual voicemail, BLF (busy lamp field), and rich messaging.

The platform is compatible with most standard SIP PBXs and telephony systems, enabling VoIP providers and organizations to extend their existing voice infrastructure to softphone users without deploying additional VPNs or session border controllers.

----

== Pre-Requisites ==
Before configuring Ringotel Shell, ensure the following requirements are met:

* You must have a Ringotel account. After account creation, you will be redirected to the softphone provisioning portal.
* Prepare Organization details - [https://help.ringotel.com/en/articles/11187474-softphone-provisioning Ringotel KB]:
** Organization Name
** Domain (Ringotel subdomain)
** Region closest to your users.
* Prepare connection details:
** '''VoIP.ms''' — Create a Sub‑Account with User/Password authentication.

== VoIP.ms Sub-Account Specification ==
:: [[File:VoIP.ms_Ringotel_SubAccount-Creation.png|500px|none]]

----
<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 Notes'''
: 🧱 NAT / Firewall Notes
: If your PBX is behind a NAT firewall, you may need to allow incoming connections from Ringotel server IP addresses in your firewall or security appliance settings. [https://help.ringotel.com/en/articles/11190222-configure-firewall-to-work-with-ringotel See related firewall documentation for details].

</div>
----

== Create an Organization ==
: 1. Go to '''Organizations''' in the Ringotel Shell portal.
: 2. Click '''Create Organization'''.
: 3. Fill required fields:
::* Organization Name
::* Domain (unique subdomain)
::* Region (closest to user base)
: 4. (Optional) Configure onboarding language, tags, and additional settings.
: 5. Click '''Next''' to complete the organization setup.

== Create a Connection ==
A Connection defines how Ringotel communicates with your PBX or SIP trunk provider.

=== Connect to VoIP.ms (SIP Trunk Provider) ===
:1. Within the organization, click '''Create Connection'''.
:2. Select '''Connect to a SIP Trunk Provider'''.
:3. Enter the VoIP.ms details:
::* POP server address
::* Sub‑Account Username
::* Sub‑Account Password
::* Check '''Registration'''
::* Use the same Sub‑Account username for: Number, Username, and Authname fields
:4. Click '''Next''' to finish.
:: [[File:VoIPms_Ringotel_CreateConnectionSIPTrunk.png|500px|none]]

=== Connect to VoIP.ms (PBX) ===
:1. Select '''Connect to a PBX server'''.
:2. Complete the following field:
::* '''Connection name''':
::* '''Protocol''':
::* '''Domain or IP address''':
::* '''Country''':
::* '''Reformat inbound numbers''':
::: [[File:VoIPms_Ringotel_CreateConnectionSIPPBX.png|500px|none]]

:3 Now, click '''Advanced settings'''
::* '''Audio codecs''' Check the following codecs '''G.711 Ulaw'''
::: [[File:VoIPms_Ringotel_CreateConnectionSIPPBX_Advanced_AudioCodecs.png|500px|none]]
::* '''Miscellanous''' '''Display Name''': Specify your Outbound CallerID Name
::* '''Registration TTL''': indicate '''120''' secondes
::: [[File:VoIPms_Ringotel_CreateConnectionSIPPBX_Advanced_Misc.png|500px|none]]

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 NOTE '''
: Please note, because Ringotel servers connect to the VoIP infrastructure on behalf of the users, consider white-listing Ringotel server IPs in your security appliance settings, like Firewall, NAT, Fail2Ban, etc.
: You can find the complete list of Ringotel IPs in the Ringotel [https://help.ringotel.com/en/articles/11190222-configure-firewall-to-work-with-ringotel help centre].
</div>
:3. Ringotel usually doesn’t require any changes to existing PBX configuration; additional settings are needed only if PBX is behind NAT.
:4. Save the connection.

----

== 4. Add Users ==
:1. Navigate to '''Users''' inside your organization.
:2. Click '''Add User'''.
:3. Provide:
::* Name
::* Email (for onboarding instructions)
::* Assigned Connection (if multiple connections are created)
::* SIP credentials (auto-filled when PBX integration is active)
:4. Save the user to automatically generate onboarding links.

----

= VoIP.ms SMS/MMS Integration with Ringotel =
:This section describes how you can enable the '''SMS and MMS messaging''' for your DID numbers using '''Ringotel'''.
:Once configured, SMS and MMS messages sent to your VoIP.ms DID can be sent and received directly from Ringotel applications.

== Overview ==
VoIP.ms supports SMS and MMS on compatible DID numbers. By integrating VoIP.ms with Ringotel, you can:
:* Send and receive SMS/MMS using VoIP.ms DID numbers
:* Assign numbers to individual users or teams
:* Manage conversations from desktop and mobile applications

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 149, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 144, 45, 0.08);
width:70vw;">

🔐 '''IMPORTANT''' ⚠️
: : '''Note:''' SMS via integrations requires the '''Ringotel Pro''' plan.
</div>

== Prerequisites ==
Before starting, ensure you have:
* An active '''VoIP.ms account'''
* One or more '''SMS-capable DID numbers'''
* An active '''Ringotel organization''' with Pro features enabled
* Administrator access to both VoIP.ms and Ringotel

== Enable SMS Integration in Ringotel ==

# Log in to the '''Ringotel Admin Portal'''
# Open your organization
# Navigate to '''Features'''
# Enable '''SMS via Integrations'''
# Save changes
# Open the '''Integrations''' tab
# Select '''VoIP.ms''' and click '''Set up Integration'''
# Copy the '''Webhook URL''' (required for VoIP.ms configuration)

== Configure the DID Number in VoIP.ms ==

=== Enable SMS/MMS on the DID ===

# Log in to the '''VoIP.ms customer portal'''
# Go to '''DID Numbers → Manage DID(s)'''
# Select the DID number and click '''Edit'''
# Under '''Message Service (SMS/MMS)''':
## Enable '''SMS/MMS'''
# Enable '''SMS/MMS URL Callback'''
# Paste the Ringotel Webhook URL and append the following parameters:

<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58%;">

: <nowiki>https://shell.ringotel.co/services/Voipms/aXXXXXXXXXXXXXXXXXXXX?id={ID}&to={TO}&from={FROM}&message={MESSAGE}&files={MEDIA}</nowiki>

: The final URL should look similar to the following:
: <nowiki>https://shell.ringotel.co/services/Voipms/a15945585810414401989?id={ID}&to={TO}&from={FROM}&message={MESSAGE}&files={MEDIA}</nowiki>

</div>

: 7. Save the DID settings

== Enable VoIP.ms API Access ==
Ringotel communicates with VoIP.ms using the VoIP.ms API.

:1. In the VoIP.ms portal, navigate to '''Main Menu → SOAP and REST/JSON API'''
:: [[File:VOIPMS_Main_Menu_RestJsonAPI.png|500px|none]]

:2. Set an '''API Password'''
:3. Enable the API if it is not already enabled
:4. Under '''Enable IP Addresses''', add the following Ringotel IP addresses:

<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58%;">

: 54.145.196.146, 18.130.161.112, 3.0.222.77

</div>
:5. Save the API settings

:: [[File:VOIPMS_API_Ringotel-IPs.png|500px|none]]

== Complete the Integration in Ringotel ==
:After setting up your API password and the IP addresses in your VoIP.ms portal, return to your Ringotel Admin portal to complete the connection.
:## Your '''VoIP.ms login email'''
:## Your '''VoIP.ms API password'''
:# Click '''Save & Continue'''

:: [[File:VoIPms_Ringotel_CreateConnectionSIPTrunk.png|500px|none]]

== Add SMS Numbers and Assign Users ==
# Add your VoIP.ms DID numbers with SMS enabled
# Assign users who can send and receive messages
# Enter phone numbers in '''national format''' (for example, <code>5033889028</code> for US/Canada)
# Configure optional settings:
## '''Session stickiness''' (keeps conversations assigned to the same user)
## '''Number reformatting''' (prevents duplicate conversation threads)
## '''Sender name in messages'''
## '''Auto-replies''' and opt-out messages
# Save the configuration

Users may need to '''log out and log back in''' to access SMS features.

== Notes and Limitations ==

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

⚠️ '''LIMITATIONS'''
: ➜ SMS and MMS availability depends on the DID type and country
: ➜ MMS support may vary by carrier
: ➜ Short codes and toll-free numbers may require additional configuration

</div>

File:VoIPms Ringotel CreateConnectionSIPPBX Advanced Misc.png

2026-05-14T01:05:29Z

Scomtois:

File:VoIPms Ringotel CreateConnectionSIPPBX Advanced AudioCodecs.png

2026-05-14T01:05:22Z

Scomtois:

Ringotel

2026-05-14T01:02:05Z

Scomtois: /* Connect to VoIP.ms (PBX System) */

[[File:VoIP.ms_Ringotel_Logo.png|right|300px]]
'''Ringotel''' is a cloud-based softphone and unified communications platform designed to work with standard SIP-based VoIP systems.
It provides features such as voice calling, messaging, presence, and video across desktop and mobile devices, while simplifying deployment and management for service providers and enterprises. Ringotel operates as a secure VoIP tunnel that routes SIP traffic between users and an existing PBX or VoIP system, reducing infrastructure complexity and improving reliability.

Ringotel supports centralized user provisioning and management through an admin portal or API, allowing automated onboarding, configuration, and device setup. It also offers integration options with CRM systems, messaging services, and business tools, and includes features such as push notifications, call recording, visual voicemail, BLF (busy lamp field), and rich messaging.

The platform is compatible with most standard SIP PBXs and telephony systems, enabling VoIP providers and organizations to extend their existing voice infrastructure to softphone users without deploying additional VPNs or session border controllers.

----

== Pre-Requisites ==
Before configuring Ringotel Shell, ensure the following requirements are met:

* You must have a Ringotel account. After account creation, you will be redirected to the softphone provisioning portal.
* Prepare Organization details - [https://help.ringotel.com/en/articles/11187474-softphone-provisioning Ringotel KB]:
** Organization Name
** Domain (Ringotel subdomain)
** Region closest to your users.
* Prepare connection details:
** '''VoIP.ms''' — Create a Sub‑Account with User/Password authentication.

== VoIP.ms Sub-Account Specification ==
:: [[File:VoIP.ms_Ringotel_SubAccount-Creation.png|500px|none]]

----
<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 Notes'''
: 🧱 NAT / Firewall Notes
: If your PBX is behind a NAT firewall, you may need to allow incoming connections from Ringotel server IP addresses in your firewall or security appliance settings. [https://help.ringotel.com/en/articles/11190222-configure-firewall-to-work-with-ringotel See related firewall documentation for details].

</div>
----

== Create an Organization ==
: 1. Go to '''Organizations''' in the Ringotel Shell portal.
: 2. Click '''Create Organization'''.
: 3. Fill required fields:
::* Organization Name
::* Domain (unique subdomain)
::* Region (closest to user base)
: 4. (Optional) Configure onboarding language, tags, and additional settings.
: 5. Click '''Next''' to complete the organization setup.

== Create a Connection ==
A Connection defines how Ringotel communicates with your PBX or SIP trunk provider.

=== Connect to VoIP.ms (SIP Trunk Provider) ===
:1. Within the organization, click '''Create Connection'''.
:2. Select '''Connect to a SIP Trunk Provider'''.
:3. Enter the VoIP.ms details:
::* POP server address
::* Sub‑Account Username
::* Sub‑Account Password
::* Check '''Registration'''
::* Use the same Sub‑Account username for: Number, Username, and Authname fields
:4. Click '''Next''' to finish.
:: [[File:VoIPms_Ringotel_CreateConnectionSIPTrunk.png|500px|none]]

=== Connect to VoIP.ms (PBX) ===
:1. Select '''Connect to a PBX server'''.
:2. Complete the following field:
::* '''Connection name''':
::* '''Protocol''':
::* '''Domain or IP address''':
::* '''Country''':
::* '''Reformat inbound numbers''':
:3 Click '''Advanced settings'''
::* '''Audio codecs''' Check the following codecs '''G.711 Ulaw'''
::* '''Miscellanous''' '''Display Name''': Specify your Outbound CallerID Name
::* '''Registration TTL''': indicate '''120''' secondes

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 NOTE '''
: Please note, because Ringotel servers connect to the VoIP infrastructure on behalf of the users, consider white-listing Ringotel server IPs in your security appliance settings, like Firewall, NAT, Fail2Ban, etc.
: You can find the complete list of Ringotel IPs in the Ringotel [https://help.ringotel.com/en/articles/11190222-configure-firewall-to-work-with-ringotel help centre].
</div>
:3. Ringotel usually doesn’t require any changes to existing PBX configuration; additional settings are needed only if PBX is behind NAT.
:4. Save the connection.

:: [[File:VoIPms_Ringotel_CreateConnectionSIPPBX.png|500px|none]]

----

== 4. Add Users ==
:1. Navigate to '''Users''' inside your organization.
:2. Click '''Add User'''.
:3. Provide:
::* Name
::* Email (for onboarding instructions)
::* Assigned Connection (if multiple connections are created)
::* SIP credentials (auto-filled when PBX integration is active)
:4. Save the user to automatically generate onboarding links.

----

= VoIP.ms SMS/MMS Integration with Ringotel =
:This section describes how you can enable the '''SMS and MMS messaging''' for your DID numbers using '''Ringotel'''.
:Once configured, SMS and MMS messages sent to your VoIP.ms DID can be sent and received directly from Ringotel applications.

== Overview ==
VoIP.ms supports SMS and MMS on compatible DID numbers. By integrating VoIP.ms with Ringotel, you can:
:* Send and receive SMS/MMS using VoIP.ms DID numbers
:* Assign numbers to individual users or teams
:* Manage conversations from desktop and mobile applications

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 149, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 144, 45, 0.08);
width:70vw;">

🔐 '''IMPORTANT''' ⚠️
: : '''Note:''' SMS via integrations requires the '''Ringotel Pro''' plan.
</div>

== Prerequisites ==
Before starting, ensure you have:
* An active '''VoIP.ms account'''
* One or more '''SMS-capable DID numbers'''
* An active '''Ringotel organization''' with Pro features enabled
* Administrator access to both VoIP.ms and Ringotel

== Enable SMS Integration in Ringotel ==

# Log in to the '''Ringotel Admin Portal'''
# Open your organization
# Navigate to '''Features'''
# Enable '''SMS via Integrations'''
# Save changes
# Open the '''Integrations''' tab
# Select '''VoIP.ms''' and click '''Set up Integration'''
# Copy the '''Webhook URL''' (required for VoIP.ms configuration)

== Configure the DID Number in VoIP.ms ==

=== Enable SMS/MMS on the DID ===

# Log in to the '''VoIP.ms customer portal'''
# Go to '''DID Numbers → Manage DID(s)'''
# Select the DID number and click '''Edit'''
# Under '''Message Service (SMS/MMS)''':
## Enable '''SMS/MMS'''
# Enable '''SMS/MMS URL Callback'''
# Paste the Ringotel Webhook URL and append the following parameters:

<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58%;">

: <nowiki>https://shell.ringotel.co/services/Voipms/aXXXXXXXXXXXXXXXXXXXX?id={ID}&to={TO}&from={FROM}&message={MESSAGE}&files={MEDIA}</nowiki>

: The final URL should look similar to the following:
: <nowiki>https://shell.ringotel.co/services/Voipms/a15945585810414401989?id={ID}&to={TO}&from={FROM}&message={MESSAGE}&files={MEDIA}</nowiki>

</div>

: 7. Save the DID settings

== Enable VoIP.ms API Access ==
Ringotel communicates with VoIP.ms using the VoIP.ms API.

:1. In the VoIP.ms portal, navigate to '''Main Menu → SOAP and REST/JSON API'''
:: [[File:VOIPMS_Main_Menu_RestJsonAPI.png|500px|none]]

:2. Set an '''API Password'''
:3. Enable the API if it is not already enabled
:4. Under '''Enable IP Addresses''', add the following Ringotel IP addresses:

<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58%;">

: 54.145.196.146, 18.130.161.112, 3.0.222.77

</div>
:5. Save the API settings

:: [[File:VOIPMS_API_Ringotel-IPs.png|500px|none]]

== Complete the Integration in Ringotel ==
:After setting up your API password and the IP addresses in your VoIP.ms portal, return to your Ringotel Admin portal to complete the connection.
:## Your '''VoIP.ms login email'''
:## Your '''VoIP.ms API password'''
:# Click '''Save & Continue'''

:: [[File:VoIPms_Ringotel_CreateConnectionSIPTrunk.png|500px|none]]

== Add SMS Numbers and Assign Users ==
# Add your VoIP.ms DID numbers with SMS enabled
# Assign users who can send and receive messages
# Enter phone numbers in '''national format''' (for example, <code>5033889028</code> for US/Canada)
# Configure optional settings:
## '''Session stickiness''' (keeps conversations assigned to the same user)
## '''Number reformatting''' (prevents duplicate conversation threads)
## '''Sender name in messages'''
## '''Auto-replies''' and opt-out messages
# Save the configuration

Users may need to '''log out and log back in''' to access SMS features.

== Notes and Limitations ==

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

⚠️ '''LIMITATIONS'''
: ➜ SMS and MMS availability depends on the DID type and country
: ➜ MMS support may vary by carrier
: ➜ Short codes and toll-free numbers may require additional configuration

</div>

File:VoIPms Ringotel CreateConnectionSIPPBX.png

2026-05-14T00:50:48Z

Scomtois:

Ringotel

2026-05-14T00:42:01Z

Scomtois:

[[File:VoIP.ms_Ringotel_Logo.png|right|300px]]
'''Ringotel''' is a cloud-based softphone and unified communications platform designed to work with standard SIP-based VoIP systems.
It provides features such as voice calling, messaging, presence, and video across desktop and mobile devices, while simplifying deployment and management for service providers and enterprises. Ringotel operates as a secure VoIP tunnel that routes SIP traffic between users and an existing PBX or VoIP system, reducing infrastructure complexity and improving reliability.

Ringotel supports centralized user provisioning and management through an admin portal or API, allowing automated onboarding, configuration, and device setup. It also offers integration options with CRM systems, messaging services, and business tools, and includes features such as push notifications, call recording, visual voicemail, BLF (busy lamp field), and rich messaging.

The platform is compatible with most standard SIP PBXs and telephony systems, enabling VoIP providers and organizations to extend their existing voice infrastructure to softphone users without deploying additional VPNs or session border controllers.

----

== Pre-Requisites ==
Before configuring Ringotel Shell, ensure the following requirements are met:

* You must have a Ringotel account. After account creation, you will be redirected to the softphone provisioning portal.
* Prepare Organization details - [https://help.ringotel.com/en/articles/11187474-softphone-provisioning Ringotel KB]:
** Organization Name
** Domain (Ringotel subdomain)
** Region closest to your users.
* Prepare connection details:
** '''VoIP.ms''' — Create a Sub‑Account with User/Password authentication.

== VoIP.ms Sub-Account Specification ==
:: [[File:VoIP.ms_Ringotel_SubAccount-Creation.png|500px|none]]

----
<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 Notes'''
: 🧱 NAT / Firewall Notes
: If your PBX is behind a NAT firewall, you may need to allow incoming connections from Ringotel server IP addresses in your firewall or security appliance settings. [https://help.ringotel.com/en/articles/11190222-configure-firewall-to-work-with-ringotel See related firewall documentation for details].

</div>
----

== Create an Organization ==
: 1. Go to '''Organizations''' in the Ringotel Shell portal.
: 2. Click '''Create Organization'''.
: 3. Fill required fields:
::* Organization Name
::* Domain (unique subdomain)
::* Region (closest to user base)
: 4. (Optional) Configure onboarding language, tags, and additional settings.
: 5. Click '''Next''' to complete the organization setup.

== Create a Connection ==
A Connection defines how Ringotel communicates with your PBX or SIP trunk provider.

=== Connect to VoIP.ms (SIP Trunk Provider) ===
:1. Within the organization, click '''Create Connection'''.
:2. Select '''Connect to a SIP Trunk Provider'''.
:3. Enter the VoIP.ms details:
::* POP server address
::* Sub‑Account Username
::* Sub‑Account Password
::* Check '''Registration'''
::* Use the same Sub‑Account username for: Number, Username, and Authname fields
:4. Click '''Next''' to finish.
:: [[File:VoIPms_Ringotel_CreateConnectionSIPTrunk.png|500px|none]]

=== Connect to VoIP.ms (PBX System) ===
:1. Select '''Connect to a PBX System'''.
:2. In the connection settings, specify connection name, transport protocol, and a PBX server IP address or domain name.

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 NOTE '''
: Please note, because Ringotel servers connect to the VoIP infrastructure on behalf of the users, consider white-listing Ringotel server IPs in your security appliance settings, like Firewall, NAT, Fail2Ban, etc.
: You can find the complete list of Ringotel IPs in the Ringotel [https://help.ringotel.com/en/articles/11190222-configure-firewall-to-work-with-ringotel help centre].
</div>
:3. Ringotel usually doesn’t require any changes to existing PBX configuration; additional settings are needed only if PBX is behind NAT.
:4. Save the connection.

:: [[File:VoIPms_Ringotel_CreateConnectionSIPPBX.png|500px|none]]

----

== 4. Add Users ==
:1. Navigate to '''Users''' inside your organization.
:2. Click '''Add User'''.
:3. Provide:
::* Name
::* Email (for onboarding instructions)
::* Assigned Connection (if multiple connections are created)
::* SIP credentials (auto-filled when PBX integration is active)
:4. Save the user to automatically generate onboarding links.

----

= VoIP.ms SMS/MMS Integration with Ringotel =
:This section describes how you can enable the '''SMS and MMS messaging''' for your DID numbers using '''Ringotel'''.
:Once configured, SMS and MMS messages sent to your VoIP.ms DID can be sent and received directly from Ringotel applications.

== Overview ==
VoIP.ms supports SMS and MMS on compatible DID numbers. By integrating VoIP.ms with Ringotel, you can:
:* Send and receive SMS/MMS using VoIP.ms DID numbers
:* Assign numbers to individual users or teams
:* Manage conversations from desktop and mobile applications

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 149, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 144, 45, 0.08);
width:70vw;">

🔐 '''IMPORTANT''' ⚠️
: : '''Note:''' SMS via integrations requires the '''Ringotel Pro''' plan.
</div>

== Prerequisites ==
Before starting, ensure you have:
* An active '''VoIP.ms account'''
* One or more '''SMS-capable DID numbers'''
* An active '''Ringotel organization''' with Pro features enabled
* Administrator access to both VoIP.ms and Ringotel

== Enable SMS Integration in Ringotel ==

# Log in to the '''Ringotel Admin Portal'''
# Open your organization
# Navigate to '''Features'''
# Enable '''SMS via Integrations'''
# Save changes
# Open the '''Integrations''' tab
# Select '''VoIP.ms''' and click '''Set up Integration'''
# Copy the '''Webhook URL''' (required for VoIP.ms configuration)

== Configure the DID Number in VoIP.ms ==

=== Enable SMS/MMS on the DID ===

# Log in to the '''VoIP.ms customer portal'''
# Go to '''DID Numbers → Manage DID(s)'''
# Select the DID number and click '''Edit'''
# Under '''Message Service (SMS/MMS)''':
## Enable '''SMS/MMS'''
# Enable '''SMS/MMS URL Callback'''
# Paste the Ringotel Webhook URL and append the following parameters:

<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58%;">

: <nowiki>https://shell.ringotel.co/services/Voipms/aXXXXXXXXXXXXXXXXXXXX?id={ID}&to={TO}&from={FROM}&message={MESSAGE}&files={MEDIA}</nowiki>

: The final URL should look similar to the following:
: <nowiki>https://shell.ringotel.co/services/Voipms/a15945585810414401989?id={ID}&to={TO}&from={FROM}&message={MESSAGE}&files={MEDIA}</nowiki>

</div>

: 7. Save the DID settings

== Enable VoIP.ms API Access ==
Ringotel communicates with VoIP.ms using the VoIP.ms API.

:1. In the VoIP.ms portal, navigate to '''Main Menu → SOAP and REST/JSON API'''
:: [[File:VOIPMS_Main_Menu_RestJsonAPI.png|500px|none]]

:2. Set an '''API Password'''
:3. Enable the API if it is not already enabled
:4. Under '''Enable IP Addresses''', add the following Ringotel IP addresses:

<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58%;">

: 54.145.196.146, 18.130.161.112, 3.0.222.77

</div>
:5. Save the API settings

:: [[File:VOIPMS_API_Ringotel-IPs.png|500px|none]]

== Complete the Integration in Ringotel ==
:After setting up your API password and the IP addresses in your VoIP.ms portal, return to your Ringotel Admin portal to complete the connection.
:## Your '''VoIP.ms login email'''
:## Your '''VoIP.ms API password'''
:# Click '''Save & Continue'''

:: [[File:VoIPms_Ringotel_CreateConnectionSIPTrunk.png|500px|none]]

== Add SMS Numbers and Assign Users ==
# Add your VoIP.ms DID numbers with SMS enabled
# Assign users who can send and receive messages
# Enter phone numbers in '''national format''' (for example, <code>5033889028</code> for US/Canada)
# Configure optional settings:
## '''Session stickiness''' (keeps conversations assigned to the same user)
## '''Number reformatting''' (prevents duplicate conversation threads)
## '''Sender name in messages'''
## '''Auto-replies''' and opt-out messages
# Save the configuration

Users may need to '''log out and log back in''' to access SMS features.

== Notes and Limitations ==

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

⚠️ '''LIMITATIONS'''
: ➜ SMS and MMS availability depends on the DID type and country
: ➜ MMS support may vary by carrier
: ➜ Short codes and toll-free numbers may require additional configuration

</div>

Agents IA

2026-04-25T00:47:22Z

Scomtois:

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #f8fafc 0%, #f1f5f9 100%); border: 1px solid #e2e8f0; border-radius: 8px; padding: 35px; margin-bottom: 25px; box-shadow: 0 1px 3px rgba(0,0,0,0.05);"
|-
| style="text-align: center;" |

<div style="font-size: 2.8em; font-weight: 200; color: #1e293b; margin-bottom: 15px; letter-spacing: -2px;">Agents IA (Bêta)</div>

<div style="font-size: 1rem; color: #64748b; margin-bottom: 15px;">
Configurez un agent vocal pour mieux gérer vos appels entrants.
</div>

<div style="font-size: 0.9rem; color: #94a3b8;">
'''[https://wiki.voip.ms/article/AI_agents 🇺🇸 English]'''  •  '''[https://wiki.voip.ms/ 🇲🇽 Español (Próximamente)]'''
</div>

|}

Ce guide explique comment configurer l’agent vocal IA de VoIP.ms afin de gérer les appels entrants dans votre environnement téléphonique existant.

L’agent vocal IA agit comme un premier point de contact virtuel, capable de répondre aux appels, comprendre l’intention de l’appelant et répondre selon le comportement que vous définissez lors de la configuration. Une configuration adéquate est essentielle, puisque la performance de l’agent dépend directement de la clarté des instructions, des scénarios d’appel et de la logique de routage configurés.

'''Important :''' L’agent IA n’apprend pas automatiquement votre entreprise. Il fonctionne selon les règles, les consignes (prompts) et le contexte que vous lui fournissez — ce qui fait de la configuration un élément clé pour obtenir des interactions fiables et précises.

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #fef2f2 0%, #fee2e2 100%); border: 1px solid #fca5a5; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(248, 113, 113, 0.1);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 500; color: #b91c1c; margin-bottom: 15px;">'''Tarification'''</div>

Durant la période bêta, le tarif est de '''10 $ par agent, par mois'''. Par la suite, le tarif standard sera de 25 $ par agent, par mois. De plus, la '''tarification à la minute est de 0,25 $/min'''. Cela inclut les capacités STT, TTD et LLM, sans frais cachés.

|}

__TOC__

= Pour commencer =

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |

Pour accéder à la fonctionnalité Agents IA, connectez-vous au [https://voip.ms portail client VoIP.ms] et cliquez sur '''Agents IA (BETA)''' dans le menu de navigation.

[[File:AI_Agent_-_Dashboard.png|thumb|none|600px]]

|}

= Créer un agent =

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |

Cliquez sur '''Créer un nouvel agent IA'''.

[[File:IA agents 2.png|thumb|none|600px]]

Vous verrez deux options :

* '''Assistant vocal simple''' — Configuration guidée étape par étape, idéale pour la majorité des cas d’usage. Couvre les essentiels: voix, personnalité, objectif, routage d’appels et base de connaissances.
* '''Agent vocal personnalisé''' — Offre un contrôle complet sur le modèle LLM, le modèle TTS, les serveurs MCP, les outils webhook et les paramètres personnalisés pour des intégrations avancées.

[[File:IA agents 3.png|thumb|none|600px]]

Choisissez l’option qui convient le mieux à vos besoins. Les sections ci-dessous couvrent les deux approches en détail.

|}

== Assistant vocal simple ==

L’'''Assistant vocal simple''' vous guide à travers une série d’étapes ciblées. Complétez chaque étape et cliquez sur '''Suivant''' pour poursuivre.

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |

=== Étape 1 : Configuration de base ===

Dans cette section, vous définirez les éléments de base de votre agent :

* '''Nom de l’agent''' — Attribuez un nom pour identifier facilement l’agent
* '''Description''' — Ajoutez une courte description de son rôle (ex. Agent de support niveau 1)
* '''Voix''' — Choisissez parmi une vaste liste de voix disponibles

[[File:AI_Agent_-_Basic_Configuration.png|thumb|none|600px]]

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #fef2f2 0%, #fee2e2 100%); border: 1px solid #fca5a5; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(248, 113, 113, 0.1);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 500; color: #b91c1c; margin-bottom: 15px;">'''Note'''</div>

Certaines voix n’offrent pas encore de prévisualisation. Cette fonctionnalité sera élargie dans de futures mises à jour.

|}

Une fois terminé, cliquez sur '''Suivant''' pour continuer.

|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |

=== Étape 2 : Personnalité et ton ===

Cette section détermine la manière dont votre agent communique. Selon vos réponses, VoIP.ms commence à générer un prompt en arrière-plan.

* '''Langue de votre agent''' — Sélectionnez la langue principale de l’agent (anglais, espagnol ou français)
* '''Ton et style d’interaction''' — Définissez comment l’agent doit interagir avec les appelants. Par défaut, '''Court et concis''' est recommandé.
* '''Personnalité de l'agent''' — Utilisez ce champ pour fournir des consignes et des informations contextuelles supplémentaires telles que: le nom de l'agent, le nom de l'entreprise, les directives comportementales ou les préférences en matière de style de communication. Ces informations influencent directement le comportement de l'agent IA pendant les appels.

[[File:AI_Agent_-_Personality_and_Tone.png|thumb|none|600px]]

|}

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=== Étape 3 : Rôle et communication ===

Ici, vous devez définir le rôle de l’agent et comment il interagit au début d’un appel :

* '''Rôle de l’agent''' — Choisissez son rôle, par exemple résolution de problèmes courants (par défaut) ou collecte d’information.
* '''Salutation de l'agent''' — Définissez le message initial que l’agent prononcera lorsqu’il répondra à un appel. Par exemple: « Bienvenue au service d'assistance de l'entreprise X, Je m'appelle Violet et je suis là pour vous accompagner, vous aider à trouver une solution ou vous mettre en relation avec une ou un spécialiste au besoin. Comment puis-je vous aider aujourd'hui? »

[[File:AI_Agent_-_Purpose_and_Communication.png|thumb|none|600px]] >>> '''À venir.'''

|}

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=== Étape 4 : Acheminement des appels ===

Cette section vous permet de déterminer si l'agent IA peut transférer des appels. Si vous ne souhaitez pas de transferts, cliquez simplement sur '''Suivant'''.

Pour '''activer les transferts''', cochez '''Autoriser l'agent IA à transférer les appels vers des destinataires spécifiques''' puis ajoutez une '''étiquette''' pour décrire le scénario de transfert (ex. "Équipe des ventes", "Support technique") ou le nom d’une personne (ex. "John Doe").

Ajoutez ensuite autant de destinations que souhaité parmi:

* [[Sub_Accounts|Extensions internes]]
* Numéros DID
* [[Phone_book|Entrées de composition rapide du répertoire téléphonique]]

[[File:IA agents 11.png|thumb|none|600px]]

|}

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=== Étape 5 : Base de connaissances ===

Dans cette section, vous définissez les sources d’information que l’agent utilisera pour répondre aux appelants.

Vous pouvez ajouter des liens vers :
* votre site web
* une base de connaissances
* une FAQ
* un Wiki

'''Assurez-vous que tous les liens soient accessibles publiquement et lisibles.'''

'''Optionnel :''' Activez '''Autoriser la recherche sur Internet pour obtenir des informations supplémentaires''' pour permettre à l’agent de compléter ses réponses avec des résultats web en direct.

Une fois toutes les sections complétées, cliquez sur '''Créer un agent'''.

[[File:IA agents 12.png|thumb|none|600px]]

|}

== Agent vocal avancé ==

L’'''Agent vocal avancé''' offre un contrôle complet sur les modèles sous-jacents, les intégrations ('''à venir!''') et le comportement de l’agent. Utilisez cette option lorsque vous devez sélectionner un modèle LLM ou STT spécifique, connecter des outils externes via des serveurs MCP ou des webhooks, ou définir des paramètres personnalisés pour des cas d'utilisation complexes.

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=== Paramètres généraux ===

* '''Nom de l’agent''' — Attribuez un nom pour identifier l'agent
* '''Modèle TTS''' — Sélectionnez le modèle de synthèse vocale pour la génération de voix
* '''Voix''' — Choisissez la voix utilisée lors des appels. Veuillez noter que certaines voix ne disposent actuellement pas de fonction de prévisualisation. Cela sera étendu dans de futures mises à jour.
* '''Modèle LLM''' — Sélectionnez le modèle de langage qui alimente le raisonnement et les réponses de l’agent
* '''Audio en arrière-plan''' Ajoutez un fond sonore pendant les appels. Optionnel
* '''Langue''' — Définir la langue principale de l’agent (anglais, espagnol ou français)
* '''Description''' — Résumez le rôle de l’agent

[[File:IA advanced 1.png|thumb|none|600px]]

|}

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=== Routage et transferts d’appels ===

Pour '''activer les transferts''', cochez '''Autoriser l'agent IA à transférer les appels vers des destinataires spécifiques'''. (ex. "Équipe des ventes", "Support technique") ou le nom d’une personne (ex. "John Doe").

Ajoutez ensuite des destinations :

* [[Sub_Accounts|Extensions internes]]
* Numéros de téléphone (DID)
* [[Phone_book|Entrées de composition rapide du répertoire téléphonique]]

[[File:IA advanced 2.png|thumb|none|600px]]

|}

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=== Base de connaissances ===

Dans cette section, vous définissez les sources d’information que l’agent utilisera pour répondre aux appelants. Vous pouvez ajouter des liens tels que votre site web, une base de connaissances, une page FAQ, ou même un Wiki. Assurez-vous que tous les liens sont accessibles au public et lisibles.

'''Optionnel :''' Activer '''Autoriser la recherche sur internet pour obtenir des informations supplémentaires''' lorsque cela est nécessaire afin de compléter les réponses avec des résultats web en direct.

[[File:AI_Agent_-_Advanced_-_Knowledge_Base.png|thumb|none|600px]]

|}

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=== Outils avancés et intégrations (À venir!) ===

* Configurer des serveurs MCP
* Ajouter des outils et webhooks
* Définir des paramètres personnalisés

[[File:AI_Agent_-_Advanced_-_Integrations.png|thumb|none|600px]]

|}

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=== Paramètres de communication ===

* '''Salutations''' — Message initial joué à la réponse
* '''Message du système (Règles et instructions)''' — Instructions principales guidant le comportement de l’agent ([[AI_Prompts|voir des exemples ici]])

[[File:IA advanced 5.1.png|thumb|none|600px]]

|}

= Associer votre agent à un DID =

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Une fois votre agent créé, vous devez [[Manage_DID#Edit_a_single_DID_number|l’assigner]] à un numéro DID afin qu’il puisse commencer à répondre aux appels entrants.

# Allez dans '''Numéros DID''' → '''Gestion des DID'''
# Cliquez sur '''Modifier''' sur le DID que vous souhaitez diriger vers l'agent
# Dans la destination de routage, sélectionnez '''AI Voice Agent''' dans le menu déroulant
# Choisissez l’agent créé dans la liste
# Cliquez sur '''Appliquer'''

Votre agent IA répondra maintenant aux appels entrants sur ce DID.

[[File:AI_Agent_-_Manage_DIDs.png|thumb|none|600px]]

|}

= Questions fréquentes =

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'''Q : Est-ce que l’agent IA peut faire des appels sortants?'''
: R : Non — les agents IA sont conçus uniquement pour les appels entrants. Ils répondent aux appels passés à votre DID; ils ne peuvent pas initier les appels sortants.

'''Q : Quelles langues l'agent prend-il en charge?'''
: R : Anglais, espagnol et français.

'''Q : Quelle est la différence entre le mode de base et avancée?'''
: R : Le mode simple vous guide à travers un ensemble précis d’étapes et couvre tout ce dont la plupart des utilisateurs ont besoin: voix, personnalité, rôle, routage des appels et une base de connaissances. Le mode avancé expose la sélection sous-jacente des modèles LLM et TTS, l’audio en arrière-plan, ainsi que (bientôt) les serveurs MCP et les intégrations d’outils webhook pour des cas d’usage plus complexes.

'''Q : Puis-je avoir plusieurs agents?'''
: R : Oui - Vous pouvez créer autant d’agents que nécessaire et assigner chacun à un DID différent.

'''Q : Puis-je modifier un agent après sa création?'''
: R : Oui — depuis le tableau de bord des agents IA, vous pouvez modifier, dupliquer ou supprimer n’importe quel agent à tout moment.

'''Q : Que se passe-t-il si l’agent ne connaît pas la réponse?'''
: R : L’agent informera l’appelant qu’il ne possède pas cette information. Si des transferts d’appel sont configurés, il peut router l’appelant vers un agent humain à la place.

'''Q : L’agent répond-il 24/7?'''
: R : Oui - une fois affecté à un DID, l’agent répond à tous les appels entrants à n’importe quelle heure.

'''Q : Fonctionne-t-il avec mon DID existant?'''
: R : Oui — tout DID admissible peut être configuré avec un agent IA depuis les '''DID Numbers''' → Gestion des DID.

'''Q : Que signifie "(Bêta)"?'''
: R : La fonctionnalité est en développement actif. Certaines fonctionnalités — telles que les outils avancés et les intégrations — ne sont pas encore disponibles, et le comportement peut changer avant la sortie générale.

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Nous continuerons d’ajouter des questions à cette FAQ au fil du temps. '''Merci d’avance pour vos commentaires, votre soutien et vos questions!'''

|}

|}

= Besoin d’aide ou des commentaires? =

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<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Ressources de soutien</div>

* '''Courriel''' — [mailto:support@voip.ms soutien@voip.ms]

'''Lorsque vous signalez un problème avec un agent IA, incluez:'''

* Le nom de l’agent et le DID assigné
* La date et l’heure de l’appel, incluant le UniqueID dans les CDR
* Une description du comportement observé versus celui attendu

|}

[[Category:Guides]]

Agents IA

2026-04-25T00:46:53Z

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<div style="font-size: 2.8em; font-weight: 200; color: #1e293b; margin-bottom: 15px; letter-spacing: -2px;">Agents IA (Bêta)</div>

<div style="font-size: 1rem; color: #64748b; margin-bottom: 15px;">
Configurez un agent vocal pour mieux gérer vos appels entrants.
</div>

<div style="font-size: 0.9rem; color: #94a3b8;">
'''[https://wiki.voip.ms/AI_agents 🇺🇸 English]'''  •  '''[https://wiki.voip.ms/ 🇲🇽 Español (Próximamente)]'''
</div>

|}

Ce guide explique comment configurer l’agent vocal IA de VoIP.ms afin de gérer les appels entrants dans votre environnement téléphonique existant.

L’agent vocal IA agit comme un premier point de contact virtuel, capable de répondre aux appels, comprendre l’intention de l’appelant et répondre selon le comportement que vous définissez lors de la configuration. Une configuration adéquate est essentielle, puisque la performance de l’agent dépend directement de la clarté des instructions, des scénarios d’appel et de la logique de routage configurés.

'''Important :''' L’agent IA n’apprend pas automatiquement votre entreprise. Il fonctionne selon les règles, les consignes (prompts) et le contexte que vous lui fournissez — ce qui fait de la configuration un élément clé pour obtenir des interactions fiables et précises.

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<div style="font-size: 1.1em; font-weight: 500; color: #b91c1c; margin-bottom: 15px;">'''Tarification'''</div>

Durant la période bêta, le tarif est de '''10 $ par agent, par mois'''. Par la suite, le tarif standard sera de 25 $ par agent, par mois. De plus, la '''tarification à la minute est de 0,25 $/min'''. Cela inclut les capacités STT, TTD et LLM, sans frais cachés.

|}

__TOC__

= Pour commencer =

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Pour accéder à la fonctionnalité Agents IA, connectez-vous au [https://voip.ms portail client VoIP.ms] et cliquez sur '''Agents IA (BETA)''' dans le menu de navigation.

[[File:AI_Agent_-_Dashboard.png|thumb|none|600px]]

|}

= Créer un agent =

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Cliquez sur '''Créer un nouvel agent IA'''.

[[File:IA agents 2.png|thumb|none|600px]]

Vous verrez deux options :

* '''Assistant vocal simple''' — Configuration guidée étape par étape, idéale pour la majorité des cas d’usage. Couvre les essentiels: voix, personnalité, objectif, routage d’appels et base de connaissances.
* '''Agent vocal personnalisé''' — Offre un contrôle complet sur le modèle LLM, le modèle TTS, les serveurs MCP, les outils webhook et les paramètres personnalisés pour des intégrations avancées.

[[File:IA agents 3.png|thumb|none|600px]]

Choisissez l’option qui convient le mieux à vos besoins. Les sections ci-dessous couvrent les deux approches en détail.

|}

== Assistant vocal simple ==

L’'''Assistant vocal simple''' vous guide à travers une série d’étapes ciblées. Complétez chaque étape et cliquez sur '''Suivant''' pour poursuivre.

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=== Étape 1 : Configuration de base ===

Dans cette section, vous définirez les éléments de base de votre agent :

* '''Nom de l’agent''' — Attribuez un nom pour identifier facilement l’agent
* '''Description''' — Ajoutez une courte description de son rôle (ex. Agent de support niveau 1)
* '''Voix''' — Choisissez parmi une vaste liste de voix disponibles

[[File:AI_Agent_-_Basic_Configuration.png|thumb|none|600px]]

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<div style="font-size: 1.1em; font-weight: 500; color: #b91c1c; margin-bottom: 15px;">'''Note'''</div>

Certaines voix n’offrent pas encore de prévisualisation. Cette fonctionnalité sera élargie dans de futures mises à jour.

|}

Une fois terminé, cliquez sur '''Suivant''' pour continuer.

|}

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=== Étape 2 : Personnalité et ton ===

Cette section détermine la manière dont votre agent communique. Selon vos réponses, VoIP.ms commence à générer un prompt en arrière-plan.

* '''Langue de votre agent''' — Sélectionnez la langue principale de l’agent (anglais, espagnol ou français)
* '''Ton et style d’interaction''' — Définissez comment l’agent doit interagir avec les appelants. Par défaut, '''Court et concis''' est recommandé.
* '''Personnalité de l'agent''' — Utilisez ce champ pour fournir des consignes et des informations contextuelles supplémentaires telles que: le nom de l'agent, le nom de l'entreprise, les directives comportementales ou les préférences en matière de style de communication. Ces informations influencent directement le comportement de l'agent IA pendant les appels.

[[File:AI_Agent_-_Personality_and_Tone.png|thumb|none|600px]]

|}

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|-
| style="vertical-align: top;" |

=== Étape 3 : Rôle et communication ===

Ici, vous devez définir le rôle de l’agent et comment il interagit au début d’un appel :

* '''Rôle de l’agent''' — Choisissez son rôle, par exemple résolution de problèmes courants (par défaut) ou collecte d’information.
* '''Salutation de l'agent''' — Définissez le message initial que l’agent prononcera lorsqu’il répondra à un appel. Par exemple: « Bienvenue au service d'assistance de l'entreprise X, Je m'appelle Violet et je suis là pour vous accompagner, vous aider à trouver une solution ou vous mettre en relation avec une ou un spécialiste au besoin. Comment puis-je vous aider aujourd'hui? »

[[File:AI_Agent_-_Purpose_and_Communication.png|thumb|none|600px]] >>> '''À venir.'''

|}

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|-
| style="vertical-align: top;" |

=== Étape 4 : Acheminement des appels ===

Cette section vous permet de déterminer si l'agent IA peut transférer des appels. Si vous ne souhaitez pas de transferts, cliquez simplement sur '''Suivant'''.

Pour '''activer les transferts''', cochez '''Autoriser l'agent IA à transférer les appels vers des destinataires spécifiques''' puis ajoutez une '''étiquette''' pour décrire le scénario de transfert (ex. "Équipe des ventes", "Support technique") ou le nom d’une personne (ex. "John Doe").

Ajoutez ensuite autant de destinations que souhaité parmi:

* [[Sub_Accounts|Extensions internes]]
* Numéros DID
* [[Phone_book|Entrées de composition rapide du répertoire téléphonique]]

[[File:IA agents 11.png|thumb|none|600px]]

|}

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|-
| style="vertical-align: top;" |

=== Étape 5 : Base de connaissances ===

Dans cette section, vous définissez les sources d’information que l’agent utilisera pour répondre aux appelants.

Vous pouvez ajouter des liens vers :
* votre site web
* une base de connaissances
* une FAQ
* un Wiki

'''Assurez-vous que tous les liens soient accessibles publiquement et lisibles.'''

'''Optionnel :''' Activez '''Autoriser la recherche sur Internet pour obtenir des informations supplémentaires''' pour permettre à l’agent de compléter ses réponses avec des résultats web en direct.

Une fois toutes les sections complétées, cliquez sur '''Créer un agent'''.

[[File:IA agents 12.png|thumb|none|600px]]

|}

== Agent vocal avancé ==

L’'''Agent vocal avancé''' offre un contrôle complet sur les modèles sous-jacents, les intégrations ('''à venir!''') et le comportement de l’agent. Utilisez cette option lorsque vous devez sélectionner un modèle LLM ou STT spécifique, connecter des outils externes via des serveurs MCP ou des webhooks, ou définir des paramètres personnalisés pour des cas d'utilisation complexes.

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |

=== Paramètres généraux ===

* '''Nom de l’agent''' — Attribuez un nom pour identifier l'agent
* '''Modèle TTS''' — Sélectionnez le modèle de synthèse vocale pour la génération de voix
* '''Voix''' — Choisissez la voix utilisée lors des appels. Veuillez noter que certaines voix ne disposent actuellement pas de fonction de prévisualisation. Cela sera étendu dans de futures mises à jour.
* '''Modèle LLM''' — Sélectionnez le modèle de langage qui alimente le raisonnement et les réponses de l’agent
* '''Audio en arrière-plan''' Ajoutez un fond sonore pendant les appels. Optionnel
* '''Langue''' — Définir la langue principale de l’agent (anglais, espagnol ou français)
* '''Description''' — Résumez le rôle de l’agent

[[File:IA advanced 1.png|thumb|none|600px]]

|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
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=== Routage et transferts d’appels ===

Pour '''activer les transferts''', cochez '''Autoriser l'agent IA à transférer les appels vers des destinataires spécifiques'''. (ex. "Équipe des ventes", "Support technique") ou le nom d’une personne (ex. "John Doe").

Ajoutez ensuite des destinations :

* [[Sub_Accounts|Extensions internes]]
* Numéros de téléphone (DID)
* [[Phone_book|Entrées de composition rapide du répertoire téléphonique]]

[[File:IA advanced 2.png|thumb|none|600px]]

|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |

=== Base de connaissances ===

Dans cette section, vous définissez les sources d’information que l’agent utilisera pour répondre aux appelants. Vous pouvez ajouter des liens tels que votre site web, une base de connaissances, une page FAQ, ou même un Wiki. Assurez-vous que tous les liens sont accessibles au public et lisibles.

'''Optionnel :''' Activer '''Autoriser la recherche sur internet pour obtenir des informations supplémentaires''' lorsque cela est nécessaire afin de compléter les réponses avec des résultats web en direct.

[[File:AI_Agent_-_Advanced_-_Knowledge_Base.png|thumb|none|600px]]

|}

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=== Outils avancés et intégrations (À venir!) ===

* Configurer des serveurs MCP
* Ajouter des outils et webhooks
* Définir des paramètres personnalisés

[[File:AI_Agent_-_Advanced_-_Integrations.png|thumb|none|600px]]

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=== Paramètres de communication ===

* '''Salutations''' — Message initial joué à la réponse
* '''Message du système (Règles et instructions)''' — Instructions principales guidant le comportement de l’agent ([[AI_Prompts|voir des exemples ici]])

[[File:IA advanced 5.1.png|thumb|none|600px]]

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= Associer votre agent à un DID =

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Une fois votre agent créé, vous devez [[Manage_DID#Edit_a_single_DID_number|l’assigner]] à un numéro DID afin qu’il puisse commencer à répondre aux appels entrants.

# Allez dans '''Numéros DID''' → '''Gestion des DID'''
# Cliquez sur '''Modifier''' sur le DID que vous souhaitez diriger vers l'agent
# Dans la destination de routage, sélectionnez '''AI Voice Agent''' dans le menu déroulant
# Choisissez l’agent créé dans la liste
# Cliquez sur '''Appliquer'''

Votre agent IA répondra maintenant aux appels entrants sur ce DID.

[[File:AI_Agent_-_Manage_DIDs.png|thumb|none|600px]]

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= Questions fréquentes =

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'''Q : Est-ce que l’agent IA peut faire des appels sortants?'''
: R : Non — les agents IA sont conçus uniquement pour les appels entrants. Ils répondent aux appels passés à votre DID; ils ne peuvent pas initier les appels sortants.

'''Q : Quelles langues l'agent prend-il en charge?'''
: R : Anglais, espagnol et français.

'''Q : Quelle est la différence entre le mode de base et avancée?'''
: R : Le mode simple vous guide à travers un ensemble précis d’étapes et couvre tout ce dont la plupart des utilisateurs ont besoin: voix, personnalité, rôle, routage des appels et une base de connaissances. Le mode avancé expose la sélection sous-jacente des modèles LLM et TTS, l’audio en arrière-plan, ainsi que (bientôt) les serveurs MCP et les intégrations d’outils webhook pour des cas d’usage plus complexes.

'''Q : Puis-je avoir plusieurs agents?'''
: R : Oui - Vous pouvez créer autant d’agents que nécessaire et assigner chacun à un DID différent.

'''Q : Puis-je modifier un agent après sa création?'''
: R : Oui — depuis le tableau de bord des agents IA, vous pouvez modifier, dupliquer ou supprimer n’importe quel agent à tout moment.

'''Q : Que se passe-t-il si l’agent ne connaît pas la réponse?'''
: R : L’agent informera l’appelant qu’il ne possède pas cette information. Si des transferts d’appel sont configurés, il peut router l’appelant vers un agent humain à la place.

'''Q : L’agent répond-il 24/7?'''
: R : Oui - une fois affecté à un DID, l’agent répond à tous les appels entrants à n’importe quelle heure.

'''Q : Fonctionne-t-il avec mon DID existant?'''
: R : Oui — tout DID admissible peut être configuré avec un agent IA depuis les '''DID Numbers''' → Gestion des DID.

'''Q : Que signifie "(Bêta)"?'''
: R : La fonctionnalité est en développement actif. Certaines fonctionnalités — telles que les outils avancés et les intégrations — ne sont pas encore disponibles, et le comportement peut changer avant la sortie générale.

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Nous continuerons d’ajouter des questions à cette FAQ au fil du temps. '''Merci d’avance pour vos commentaires, votre soutien et vos questions!'''

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= Besoin d’aide ou des commentaires? =

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<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Ressources de soutien</div>

* '''Courriel''' — [mailto:support@voip.ms soutien@voip.ms]

'''Lorsque vous signalez un problème avec un agent IA, incluez:'''

* Le nom de l’agent et le DID assigné
* La date et l’heure de l’appel, incluant le UniqueID dans les CDR
* Une description du comportement observé versus celui attendu

|}

[[Category:Guides]]

Caller ID

2026-03-06T13:19:26Z

Scomtois:

{| class="wikitable"
|+
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! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/ID_de_l%27appelant Français] ||
[https://wiki.voip.ms/article/Numero_Identificador_(Caller_ID) Español]
|}

== Blog Article ==

<div style="position: relative; width: 90%; height: 20px;
background: linear-gradient(to right, #f8f9fa, rgba(255, 0, 0, 0));
border: 1px solid transparent;
border-image: linear-gradient(to right, #dee2e6, rgba(255, 0, 0, 0));
border-radius: 8px; padding: 12px; overflow: hidden; display: flex; align-items: left; justify-content: left; text-align: left; font-family: sans-serif; color: #333;">
<div style="position: absolute; font-size: 100px; opacity: 0.15; z-index: 1; pointer-events: none; user-select: none; transform: rotate(-10deg);">📰</div>
<p style="position: relative; z-index: 2; margin: 0; line-height: 1.5; background: none !important;">

📃 [https://wiki.voip.ms/article/What_Is_CNAM_and_How_to_Leverage_It_for_Your_Business%3F What Is CNAM and How to Leverage It for Your Business?]

</p>
</div>


= Caller ID =
: Caller ID is a telephone service that transmits the calling party´s number to the called party´s telephone . When available, the Caller ID number can be complemented with Caller ID name (e.g., John Smith).
: If you are placing outgoing calls, you will need to pass a Caller ID to ensure proper termination of your calls.
: There are two types of caller ID and it is important to differentiate them: Caller ID Number (CID) and Caller ID Name (CNAM).
: Please note that the Caller ID is only guaranteed when using Premium routes, and only for US48 and Canadian calls. In Canada you may find Caller ID working on some Value routes.
: Incoming Caller ID (from people calling you) is addressed below.

__TOC__

= Caller ID Rules =
: In line with all the rulings in favor of protecting and empowering consumers, VoIP.ms enforces valid Caller IDs as per the NANPA and ITU-T E.164 standards.
: Therefore, calls displaying the following Caller ID information will be rejected:

: * International calls with Caller ID containing '''less than 7 digits''' and '''greater than 15 digits''';
: * North America calls with Caller ID not containing exactly '''10 digits''' ''(exception will be made for 7 digits for 310-xxx numbers)'';
: * North America calls with Caller ID containing '''10 digits''', but with an unassigned NPA (first 3 digits of the number);
: * North America calls with Caller ID containing '''10 digits''', but where an unassignable NXX is used (i.e. the second block of 3 digits where the first digit is either zero (0) or one (1)).

== Outgoing Caller ID Name ==
: The Caller ID name is an additional information you can pass along with your Caller ID number. This will be received on the callee's end and it could be your given name or the name of your business.
:: For example: ''"John Smith" <9145551234>''

: The sample above is a Caller ID that includes both Caller ID name and Caller ID number, commonly abbreviated as CID and CNAM among other variations.

: It is not possible to set a Caller ID name from the VoIP.ms portal.

: If you plan to make calls to Canadian numbers, you can simply pass the Caller ID name from your device or system as most of them support this.'''*''' <br/>
: You will need to check for a field on the interface from the device to enter this setting, and in case you are using a more advanced system, get assistance to set the outgoing Caller ID name set up.

: The Caller ID name on US calls works differently, this is controlled by a national CNAM database with records of numbers and names matching each number.
: When you make a call to a US number, you will send a Caller ID number and the system will check the CNAM database for a name matching that same Caller ID number in order to display both name and number to the final phone.

: CNAM is only available for some USA numbers. In order to update your Caller ID'''*''' on the CNAM database for your US calls, there is a process to follow which has a cost of $10 USD (one time only fee).
: Please contact the VoIP.ms support staff to get further details on what information you need to submit and to confirm if your local US DID is available for a CNAM update.

:: '''CNAM update is only available for some Local US DIDs. Toll frees cannot have their Caller ID name updated'''.

<div style="border-left: 6px solid blue; background-color: lightgrey; width: 85%;">
'''IMPORTANT''':
- 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''' max! Some of regular Canadian providers will not show more than '''15 characters'''. We suggest shrinking or adapt your caller ID.
- Spaces are allowed in a caller id name.
</div>

== Outgoing Caller ID Number ==
: Caller ID number is the most common Caller ID type passed. If a more complex system capable of passing its own Caller ID is being used, such as a [[Welcome#PBX|PBX]], the Caller ID field is likely set from the trunk, or one of its extensions.
: If you are using devices like [[Welcome#Devices|Analogue Telephone Adapters]], [[Welcome#Devices|IP phones]], or [[Welcome#Softphones|softphones]], the Caller ID number is available to be set from your VoIP.ms account via the customer portal.

== Outbound Caller ID - Main account ==
: To set the Caller ID number for your Main account, access:

:: '''1'''. Form your VoIP.ms Portal, go to "[[Account Settings]]" from '''Main Menu''' -> '''Account settings'''
::: [[File:AccountSettingsMenu.png|thumb|none|400px|Account settings]]
:: '''2'''. Go to '''General''' Tab and there set the '''CallerID number'''
::: [[File:VoIPms_AccountSettings_General_CallerID_Number.png|thumb|none|400px|General]]

== Outbound Caller ID - Sub Accounts ==
:The Caller ID number for a sub account can be set during the creation process, or later by clicking Edit on the [[Sub Accounts|subaccount]].
::[[File:CallerIDSubAcc.png|thumb|none|700px|CallerID field in a sub account]]

: * '''For calls to US and Canada, a 10-digit Caller ID number is required to ensure proper call termination. The portal's Caller ID field only supports numerical digits.'''
: * '''Toll free Caller IDs are not recommended, especially when Calling Toll free numbers, to reduce potential connection issues.'''

: If you are using an analog telephone adapter (ATA), IP phone, or softphone, this is where you need to adjust the settings.
: It is important to ensure that you transmit a valid caller ID to ensure proper termination.

: Here you will have the option of having one of your incomings DID numbers displayed as an outgoing call identification number.
: You can also use a personalized number that does not appear in your lists of DID numbers.

: However, you must respect the standards in force by the regulations to this effect.
: '''Always use a number that belongs to you''' or to that '''you have permission to use''' as the outgoing call display number.

: '''I use a system of capable of passing its own CallerID Number''': If you have a PBX that manages the outgoing call identification number, you will have an option for this.

:: ''Note: Using a Toll free outbound caller id is not recommended, especially when calling Toll-Free numbers.''

=Different methods of passing your Caller ID Number=
'''CallerID Number''': You can set the [[Caller ID|callerID Number]] or your DID Number or proceed with a verification process to pass the callerID of a number that is not within VoIP.ms network. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], make sure that the callerID matches a DID or a verified number from within your account to successfully make outgoing calls.

*'''Use one of my DIDs''': If you want to pass one of your DID Numbers as the callerID Number, select this option and choose the DID, if more than one, that will display on the called party.

* '''Use a Verified CallerID''': This is a quick verification process to ensure that the callerID you will pass is validated and authorized to be used. To start the process:

: '''1'''. Simply click on '''Click here to get started''' above the CallerID Number setting
::: [[File:VoIPms_AccountSettings_General_Number_Verification_Section.png|none|650px]]
: '''2'''. Follow the CallerID Verification Process
:: '''1'''. Select the country of your DID and Enter the number you want to verify.
:: '''2'''. You can also add a reference/note for your own use.
:: '''3'''. Choose the verification to be done by receiving a call or by receiving a SMS (only available for Canada/United States).
::: [[File:VoIPms_Non-VoIPms_Number_Verification.png|none|650px]]
: '''3'''. Once you press '''Send verification Code''', you will receive a phone call / SMS message with a validation code.
:: Enter that code in the '''Validation Code''' field and press '''Verify'''.
::: [[File:VoIPms_Non-VoIPms_Number_Verification_Code.png|none|650px]]

:: > If for any reasons you have not received the code, you can ask the system to send it again or press cancel to change the method you would like to receive the code.

: * '''I use a system capable of passing its own CallerID''': If you are using a PBX, you will need to pass the callerID directly from your system.
:: Our system will verify if the callerID you are passing matches one of your DID or one of your verified callerID.
:: If you are passing an invalid callerID and attempt to make an outgoing call, our system will send you an email advising that a call was attempted with an invalid callerID.

==Outbound caller ID number using Call Forwarding==
: When you chose to forward all your incoming calls from your VoIP.ms number to an external phone number, the original's caller ID number will be transmitted to this external number.
: However, if the incoming caller is hiding their phone number this call will fail due to the new regulations for CID, on where sending a valid caller ID number is mandatory.

: A workaround is to create a filter for '''"Anonymous"''' calls along with another call forwarding entry but this one using the '''"Caller ID number override"'''.
: This way, incoming calls using a valid CID will be transmitted to your external number showing the original's CID and incoming calls hiding their number will use the filter and the forward with caller ID Override, showing the CID override number and completing the call.

: '''To receive "Anonymous" calls when using call forwarding, follow these steps:'''
:: '''1'''. Go to '''DID Numbers>> Call forwarding''' section and create a new call forwarding entry to the same number you're forwarding to, but this new entry will contain the '''"Caller ID override"'''. For the caller ID override, you can type your VoIP.ms number or any valid number you want to see when you receive an anonymous call. You might want to fill the '''"Description field"''' so you can identify this entry on the next step.
:: '''2'''. Thereafter, go to '''DID Numbers>> CallerID filtering''' section and create a new filter.
::: * For step 1, mark '''"Anonymous CallerID number"''' checkbox
::: * For step 2, this is optional you can choose to apply this filter to all your numbers or just to a specific one.
::: * For Step 3, mark the '''"Call forwarding"''' checkbox and make sure to use the call forwarding entry with the caller ID override number.
::: * Save your filter.

== Incoming Caller ID number and name ==

: You will receive the Caller ID number and Caller ID name that the VoIP.ms server receives from the caller, this is exactly what will be sent to you on Incoming calls.
: You can always check what Caller ID number VoIP.ms receives by going into your [[Call Detail Records]] to check the incoming calls.

: The incoming Caller ID name works almost the same way, except that this is an optional setting that you need to enable per DID number on the [[Manage DID|DID settings]] page. This option is called "CallerID Name Lookup". : When enabled, the system will perform a query on the LIBD/CNAM Database, for callers with Canadian or US CID number, in order to find a name matching that CID number. The system then will display the result of this query in the Caller ID name portion of the '''Caller ID''', leading to a "Caller ID name"<5551231234> when people call your number.

: If the calling number is already in your [[phone book]], the name will be taken from there instead of doing a CNAM lookup on an external database.

:: '''IMPORTANT NOTE FOR CANADIAN DIDs'''
:: The majority of the Canadian DID numbers support CNAM Pass-Through.
:: This means that for your incoming calls the system '''won't do a CNAM query''' (and also not charge you) if the incoming call already has a Caller ID name, even if the DID receiving the call has the CNAM queries enabled in your VoIP.ms customer portal.
----

== Listen to the current account's Caller ID Number ==
: To listen to the caller ID set in the account you're dialing from, please dial '''822''' and the Caller ID number will be read back.
: Notice that it will depend on your device's dial plan or dial pattern for this to work.
: These test calls won't have any cost for you and will appear in your [[Call_Detail_Records | CDR]] as "CallerID Testing".

==Configuring your CallerID with the Reseller Portal==
: Once within the reseller portal, head into Services, My Services to see the callerID options.
: The options available are the same as your main account or sub accounts.
:: [[File: CallerIDverif_Reseller.png |thumb|none|600px]]

=Notes=

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<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;">

:'''Outgoing Caller ID is not guaranteed on calls to Canadian cellular numbers, even when using the Premium route.'''
:'''This is due to the way Canadian carriers work - they sometimes pass a random Caller ID that they have on record, changing the original.'''
:'''This is out of our control as it is the way Canadian carriers handle calls to cellular numbers.'''
:'''There are also issues with incorrect Caller ID being sent on outbound [[toll-free number]] calls from Bell Mobility (and its resellers);'''
:'''the display shows a disconnected Bell number in an area code corresponding to the caller's location instead of the caller's mobile number.'''
</p></div>

[[category:guides]]