Andrew Mercer
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Instead of registering the HT812 directly with a provider, it can register as an extension on your own PBX. The PBX then connects to a provider through a SIP trunk.

rotary phone → HT812 (extension 101) → Asterisk/FreePBX → SIP trunk → VoIP.ms → PSTN

What this adds:

  • Internal extensions: Phone 1 and Phone 2 on the HT812, softphones, and other ATAs can call each other for free.
  • Voicemail-to-email, IVR menus, ring groups, time-of-day routing, and call recording.
  • Custom dial plan codes. Because a rotary phone can't send * or #, give features all-digit codes (for example, dial 8000 for voicemail) so a rotary phone can reach them.

See PBX software for Asterisk, FreePBX and alternatives. Complete Base configuration on the HT812 first.

On the PBX

  1. Create a SIP (PJSIP) extension for each HT812 port, for example 101 and 102, with a long random secret.
  2. Restrict the extension to your LAN (allowed IP range / permit list) so the credentials can't be used from the internet.
  3. Add a SIP trunk to your provider. With VoIP.ms, register the PBX with a dedicated sub-account and route your DID to it.
  4. Inbound route: send the DID to extension 101 (or a ring group).
  5. Outbound route: match 10-digit, 1+10-digit, and 911 numbers and send them out the trunk. Make sure the trunk sends your DID as caller ID, and that 911 calls always go out with the number that has a registered 911 address.

On the HT812

Profile 1

Field Value
Primary SIP Server PBX's LAN IP or hostname
SIP Transport UDP (or TCP/TLS if configured on the PBX)
NAT Traversal No (the PBX is on the same LAN)
Register Expiration 5 (minutes)
DTMF RFC2833
Dial Plan Base plan plus rules for internal extensions, for example 1xx (100–199) and 8xxx (feature codes)

FXS Ports

Field Port 1 Port 2
Profile ID Profile 1 Profile 1
SIP User ID 101 102
Authenticate ID 101 102
Authenticate Password extension 101 secret extension 102 secret
Pulse dialing Enabled (rotary phone) As needed

Security

Self-hosted PBXs are a frequent target for toll fraud. FreePBX in particular had serious, actively exploited vulnerabilities in 2025 affecting internet-exposed admin panels.

  • Never expose the PBX admin web UI to the internet. Use a VPN for remote access.
  • Don't port-forward SIP (5060) to the PBX unless you need remote phones, and use a firewall allow-list or VPN if you do.
  • Keep the PBX patched, use long random extension secrets, and set provider-side spending limits and international-calling restrictions.

Sources