Andrew Mercer
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VoIP.ms is a Montreal-based pay-as-you-go SIP provider and the most common choice for an ATA in Canada. It gives full SIP credentials, inexpensive local numbers (DIDs), and per-minute or flat-rate calling. See also SIP providers.

Complete Base configuration first.

Part 1: VoIP.ms portal

1. Account and funds

Create an account and add funds. VoIP.ms is prepaid.

2. Create a sub-account for the ATA

Don't register the ATA with your main account credentials. Create a sub-account (Sub Accounts → Create Sub Account) for each line:

  • Protocol: SIP
  • Authentication type: user/password
  • Device type: ATA, IP phone, or softphone
  • Password: a long, random password. SIP accounts are a constant target for toll-fraud scanners.
  • Restrict international calling on the sub-account unless you need it.

The sub-account's SIP username is your account number, an underscore, and the sub-account name (for example 123456_rotary). Copy the exact value from the portal.

3. Buy a DID and route it to the sub-account

Order a number (DID), then edit its settings:

  • Routing: SIP/IAX → the sub-account, not "[main account]". Routing to the main account is the most common reason an ATA can make calls but never rings.
  • Point of Presence (POP): pick the server nearest you (a Toronto or Montreal server for Ontario). Note the full server hostname shown on the DID's settings page.

4. Caller ID

On the sub-account, set the outgoing CallerID number to your DID so outbound calls show your number.

5. Register 911

Enable 911 service on the DID and register the street address where the phone is physically located. VoIP 911 is not tied to a physical line the way a landline is; without a registered address, emergency calls may not reach the right dispatch centre with your location. There is a monthly 911 fee; check the current price in the portal. Update the address if the ATA ever moves.

Part 2: HT812 settings

Profile 1 (server settings)

Field Value
Primary SIP Server The DID's POP server hostname, exactly as shown in the portal
Outbound Proxy Leave blank
SIP Transport UDP
SIP Registration Yes
Unregister On Reboot Yes
Register Expiration 5 (minutes)
NAT Traversal Keep-Alive
DTMF RFC2833
Preferred Vocoder PCMU (G.711u) first, then PCMA
Dial Plan See Base configuration

The Primary SIP Server must match the DID's POP. If they differ, outbound calls may work while incoming calls never arrive.

FXS Port 1 (line settings)

Field Value
Profile ID Profile 1
SIP User ID Sub-account username (for example 123456_rotary)
Authenticate ID Same as SIP User ID
Authenticate Password Sub-account password
Name Display name (optional)
Pulse dialing Enabled (for a rotary phone)

Click Apply, then check the Status page. The port should show Registered.

Second line (optional)

Create a second sub-account and DID. Put its credentials on FXS Port 2, also using Profile 1.

Testing

  1. Get dial tone on Phone 1.
  2. Call a cell phone, and check the caller ID shows your DID.
  3. Call your DID from the cell phone to test ringing.
  4. Leave a call connected for a few minutes and confirm audio is heard both ways.

Troubleshooting

Symptom Likely cause
Can call out, phone never rings DID routed to the main account instead of the sub-account, or ATA server doesn't match the DID's POP
Registered, then stops receiving calls after a while NAT mapping expiring: lower Register Expiration, enable keep-alive
One-way or no audio SIP ALG on the home router (turn it off), or double NAT
Rotary digits wrong Dial needs cleaning/adjustment, or pulse dialing not enabled on that port
Can't reach voicemail menus Rotary phones can't send * or #; use a pulse-to-tone converter

Sources