• A VoIP.ms account with a funded balance
• A DID (phone number) ordered on your account
• A SIP subaccount configured with the settings below
• The hostname of your VoIP.ms POP server (must match your DID's Point of Presence)
• Issabel PBX installed and accessible via its web interface

Required subaccount settings in VoIP.ms:

• Authentication Type: User/Password
• Device Type: Asterisk, IP PBX, Gateway or VoIP Switch
• CallerID Number: System will send its own CallerID
• Allowed Codecs: G.722 (recommended), ulaw (fallback)

See Subaccount Setup and Choosing a Server if you need help with these steps.

In the Issabel web interface, navigate to:

PBX → PBX Configuration → Trunks

Click Add SIP Trunk.

In the Outgoing Settings section, enter the following in the PEER Details field. Replace the example values with your actual subaccount credentials and server.

type=peer
username=Your VoIP.ms sub account (eg. 123456_issabel)
secret=Your VoIP.ms sub account password
host=Enter the VoIP.ms server that matches your DID. See [[Choosing_Server|Choosing a Server]] for the full list.

montreal2.voip.ms
fromuser=123456_issabel
trustrpid=yes
sendrpid=yes
qualify=yes
nat=yes
insecure=port,invite
disallow=all
allow=g722,ulaw
context=from-trunk
canreinvite=nonat

In the Incoming Settings section, leave the USER Details field empty. VoIP.ms delivers inbound calls via registration — no incoming peer configuration is required.

In the Registration field, enter your registration string using this format:

SubAccountUsername:[email protected]/SubAccountUsername

Example:

123456_issabel:[email protected]/123456_issabel

Click Save and then Apply Config to save the trunk.

Navigate to:

PBX → PBX Configuration → Inbound Routes

Click Add Incoming Route and fill in the following:

Click Save and then Apply Config.

Navigate to:

PBX → PBX Configuration → Outbound Routes

Click Add Route and configure:

Dial Patterns

Add the following dial patterns to match the calls you want to route through VoIP.ms:

Click Save and then Apply Config.

Navigate to:

PBX → Tools → Asterisk CLI

Run the following command:

sip show registry

You should see your trunk listed with a status of Registered. If the status shows Unregistered, double-check your sub account username, password, and server hostname in the PEER Details and registration string.

You can also verify registration status in the VoIP.ms customer portal — log in, go to Portal Home, and scroll to the bottom to see the registration state of your subaccounts.

Test Your Setup

• Dial 4443 from an internal extension — this is the VoIP.ms echo test. Speak into the phone and you will hear your voice played back. A successful echo test confirms the trunk is registered and audio is working in both directions.
• Dial 822 to hear the caller ID your system is sending.
• Call your VoIP.ms DID from an external phone to confirm inbound routing works.