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Reference / Provider and Asterisk integration boundary
CONTRACT: LOCAL LAB · NOT LIVE TELEPHONY

Provider and Asterisk integration boundary

The 0.0.1 repository now contains a disabled-by-default Asterisk ARI connector in byot/, tested against a synthetic ARI server and network UDP. This is a local reference implementation, not a verified or deployed SIP/PSTN integration; real Asterisk interoperability, SIP.js registration, PSTN reachability, and settlement remain untested.

Gateway acceptance requirements

Before enabling live testing or any future private Asterisk/SBC adapter, independently verify the following against an isolated PBX with carrier egress disabled:

  1. Asterisk and TLS/WSS listeners authenticated with valid certificate and endpoint policy.
  2. SIP registration or configured-peer readiness and successful authenticated OPTIONS, without claiming this proves INVITE route or remote media.
  3. Mandatory SRTP/DTLS-SRTP where the endpoint and network permit it; reject downgrades. Confirm the negotiated media protection, not merely a checkbox.
  4. Actual INVITE, answer, bidirectional audio, RTP metrics, DTMF and release events.
  5. Independently enforce maximum session time at the provider/PBX, even if the AI process crashes.
  6. Reconcile unknown or partial states after process restart and network failure.
  7. Permit only explicitly approved test destinations before any UK geographic pilot.

Do not conflate products

The Windows Dial-Tone MicroSIP fork is a separate native application. Its security and readiness patterns have been reviewed, not ported into this Python service. The private telephony backend must remain separately reviewed and outside this public development release.

For the public self-hosted reference, see BYOT: Asterisk. The agent never supplies a trunk or arbitrary SIP endpoint.