# PSTN Calling via VoIP.ms — Planning Notes

Research and decisions from a design discussion, saved here so the work can
be picked up in a fresh chat without re-deriving the background.

**Implemented** — see `services/pstn-trunk.sh` (run `sudo ./setup.sh
pstn-trunk` after `asterisk-digital-ocean` **or** `asterisk` (home/LAN) is
installed — both are supported, see the file for the static-IP caveat on the
LAN variant). Generic SIP trunk add-on that defaults to VoIP.ms but isn't
hardcoded to it — any provider supporting IP authentication works. Covers:

- IP-authenticated trunk, US/NANP-only outbound dialplan, no catch-all.
- A configurable concurrent-call cap (default 3, global not per-extension).
- **Three-tier per-extension permission model**: `internal` (default — no
  PSTN at all, but can always call/receive other extensions and internal
  ring groups), `restricted` (also only pre-approved US numbers, both
  directions), `full` (also any US number). Internal extension-to-extension
  dialing is *never* gated by any tier.
- **Permissions are live, not baked into the dialplan.** Stored in
  `pstn-permissions.conf`, read by the dialplan via Asterisk's
  `AST_CONFIG()` on every call — editing that file takes effect on the next
  call, no restart, no reinstall. `services/pstn-trunk.sh`'s "update in
  place" mode deliberately never touches it (same protection this repo's
  update-mode convention already gives `.env`/firewall/Caddy config
  elsewhere) — only a "fresh" reinstall (with confirmation) or the web UI
  below change it.
- A configurable **inbound ring-group** (one extension or several), each
  member's live tier/approved-numbers checked per inbound call via an
  unrolled per-member dialplan block (no AGI needed).
- **`services/security-dashboard.sh` integration** — a "PSTN Trunk" tab
  lists every extension (parsed from `pjsip.conf`) with its live tier and
  approved numbers, editable with no restart. This is what makes the tier
  model actually manageable day-to-day instead of needing a reinstall for
  every roster change.
- **ntfy alerts** on denied/rejected calls (immediate — permission denied,
  number not approved, or concurrency cap hit) and spend/volume thresholds
  (hourly check: once/month on a spend threshold, every hour on a call-burst
  threshold).
- Structural settings (server, DID, ring-group membership, cap, ntfy,
  rate/thresholds) persist to `.pstn-trunk.env` so "update in place"
  reapplies them without re-prompting.

`services/pstn-trunk.sh`'s own header comment explains how the trunk/dialplan
config survives Easy Asterisk's regeneration, and why permissions are a
separate live file rather than baked in — both architectural wrinkles
discovered while implementing this, worth reading before touching either
file.

## Decision so far
- **Provider: VoIP.ms.** Chosen for its prepaid-balance model: turn off
  auto-recharge in the account's Finances settings and outbound calls simply
  fail once the balance hits $0 — that's the toll-fraud backstop if the
  droplet's Asterisk (`asterisk-digital-ocean`) is ever compromised. This
  behavior wasn't verified against a live account — confirm the
  auto-recharge toggle still works this way at sign-up time, since billing
  UX can change.
- **Scope: US calling only, for now.** No international, no premium-rate
  destinations. Enforce this twice — once via whatever dial-plan/prefix
  VoIP.ms requires for US routing, and again independently in Asterisk's own
  dialplan (see below), so a compromised extension can't reach anything
  outside the US even if the trunk itself would technically allow more later.
- **Inbound: wanted.** A DID is in scope, not outbound-only. Decide pay-per-minute
  vs. unlimited DID plan based on expected inbound volume (see cost estimate
  below), and decide E911 deliberately rather than skipping it by default —
  VoIP.ms doesn't require it, but without it 911 dialed from the line either
  fails or doesn't carry accurate address/location info.

## Cost estimate (100 min/month each direction, US-only)

Verified against VoIP.ms's public wiki/rate pages, not a live account — confirm
at sign-up since rates can change.

| Item | Rate | Monthly | Annual |
|---|---|---|---|
| DID (phone number), pay-per-minute plan | $0.85/mo flat | — | $10.20 |
| Inbound usage | $0.009/min | $0.90 | $10.80 |
| Outbound usage | $0.01/min | $1.00 | $12.00 |
| **Total** | | **~$2.75** | **~$33** |

- Skipping the DID (outbound-only) drops this to ~$12/year.
- Adding E911 adds a $1.50 one-time fee plus **$1.50/month** regulatory fee
  (~$18/year) — pushes the total above to ~$51/year.
- **Funding minimum:** VoIP.ms requires a **$15 minimum deposit** to activate
  calling — a one-time balance top-up, not a recurring charge. At ~$2.75/month
  usage that balance lasts ~5 months before a refill is needed (longer at
  lower volume). Leave auto-recharge **off** per the toll-fraud design above.

## Why this matters (toll fraud)
A compromised Asterisk box can dial premium-rate or international numbers
that cost real money fast (some destinations run several $/min) before
anyone notices. Two independent layers matter more than either alone:
1. **Trunk-side cap** — prepaid balance, auto-recharge off. Limits total
   possible loss to whatever the balance is topped up to, but a live
   compromise could still burn through that balance in minutes if nothing
   else restricts what can be dialed.
2. **Dialplan-side restriction** — Asterisk should refuse to route calls
   outside the US/NANP pattern at all, regardless of what the trunk allows.
   This is the first line of defense and should exist independent of the
   trunk's own capabilities.

**Important nuance: these two layers bound different things, and neither
alone bounds both.** NANP-only restriction bounds *cost-per-minute* (a
compromised box can only ever reach $0.01/min US numbers, never $2–5/min
international/premium destinations) — that risk is fully closed. It does
**not** bound *how fast* the prepaid balance gets burned: nothing stops a
compromised box from opening many concurrent US-destination calls in
parallel and draining the whole balance (e.g. $15 balance ÷ $0.01/min =
1,500 minutes total, which 20 concurrent legs could burn through in under
an hour). The prepaid-balance-off-auto-recharge layer bounds the *dollar*
ceiling; only a concurrent-call cap bounds the *speed* of a breach. Treat
the concurrent-call cap and spend/volume alert below as required before
funding a live trunk, not optional hardening.

**Implemented:** the concurrent-call cap in `services/pstn-trunk.sh` is a
*global* cap (configurable, default max 3 outbound legs total via the trunk,
via `GROUP()`/`GROUP_COUNT()` in the dialplan, shared across all extensions)
— not per-extension. That was the explicit ask when this got built. The
spend/volume alert is also implemented now: an hourly cron script reads a
call log the dialplan appends to directly (not Asterisk's CDR — see the
service file's own comments for why) and alerts via ntfy once per month when
estimated spend crosses a threshold, and every hour that call volume in the
last hour looks like a burst. Denied/rejected calls alert immediately,
separately from that hourly check.

## What it takes technically (asterisk-digital-ocean)
- A PJSIP trunk: `endpoint` / `aor` / `identify` sections in the pjsip
  config. **Implemented with IP authentication** (no `auth` section, no SIP
  password stored anywhere) — see `services/pstn-trunk.sh`. Provider name,
  server hostname, and DID are all prompted at install time (VoIP.ms is only
  the suggested default), so any provider supporting IP auth works.
- An outbound dialplan route matching US numbers only — **implemented**:
  `_1NXXNXXXXX` (11-digit NANP with leading 1) and `_NXXNXXXXX` (10-digit,
  auto-prefixed with 1), both routed to the trunk. No catch-all `_X.`
  pattern.
- **Three-tier permission model — implemented**, superseding an earlier flat
  allow-list design. `internal` / `restricted` / `full` per extension, read
  live from `pstn-permissions.conf` via `AST_CONFIG()` rather than baked
  into the dialplan text — no changes needed to Easy Asterisk's own
  per-device pjsip.conf sections, since the gate lives entirely in files
  this repo already owns. Internal extension-to-extension dialing is never
  gated by any tier — only the two NANP patterns (outbound) and the
  ring-group (inbound) are. Numbers are stored pipe-separated specifically
  because they're used as a `REGEX()` alternation pattern in the dialplan —
  see the security note below on why the untrusted call-time value must
  never be interpolated into the *pattern* side of that check.
- **Inbound ring-group — implemented.** A space-separated list of
  extensions to ring for inbound calls (one, or several for a ring group via
  `Dial(PJSIP/a&PJSIP/b,20)`), prompted at install. Each member's tier is
  checked live per inbound call (an unrolled dialplan block per member —
  full always rings, restricted only rings if the caller's number is on
  that member's approved list, internal never rings).
- **Security note on the REGEX() checks**: an inbound Caller-ID (or an
  outbound dialed number) is attacker-influenced data and must never be
  interpolated into the *pattern* argument of `REGEX()` — only ever the
  string being tested. Doing it backwards would let a crafted Caller-ID
  (e.g. containing regex metacharacters) forge a match against an unrelated
  approved-numbers entry. Both checks in `services/pstn-trunk.sh` put the
  admin-controlled approved-list in the pattern position and the live call
  data in the tested-string position — worth keeping that direction if this
  is ever refactored.
- **Web UI — implemented.** `services/security-dashboard.sh`'s "PSTN Trunk"
  tab lists every extension (parsed from `pjsip.conf`, the same marker
  format Easy Asterisk's own `rebuild_dialplan()` uses) with a tier dropdown
  and approved-numbers field, saving straight to `pstn-permissions.conf`.
  Tested end-to-end with a real running instance of the (stdlib-only)
  Python app: extension parsing, tier changes, number normalization/
  validation, and persistence all verified with actual HTTP requests against
  a live server in a sandboxed test — not just read through.
- Provider-specific setup that isn't scriptable (user does this manually):
  create the account, order a DID, decide pay-per-minute vs. unlimited DID
  plan and whether to add E911, pick a server/POP, fund the prepaid balance
  ($15 minimum for VoIP.ms), turn off auto-recharge. `services/pstn-trunk.sh`
  prompts for the server hostname, DID, allowed extensions, ring extensions,
  concurrency cap, ntfy topic, and spend-alert settings at install time.
- Defense-in-depth alongside the trunk:
  - **Implemented:** a global concurrent-call cap in the dialplan
    (`GROUP()`/`GROUP_COUNT()`, configurable, default max 3 outbound legs via
    the trunk at once) so an unauthorized or compromised extension can't
    open dozens of simultaneous outbound legs. Global, not per-extension —
    see the note above.
  - **Implemented:** an outbound call-count/spend alert via ntfy — a
    self-contained call log (not Asterisk's CDR) plus an hourly cron script.
    Denied/rejected calls also alert immediately. See
    `services/pstn-trunk.sh`'s "Spend/volume alerts" README section for the
    exact mechanics and why CDR wasn't used.
  - Worth being explicit that CrowdSec's existing `asterisk_bf` /
    `asterisk_user_enum` scenarios (see `services/crowdsec.sh`) cover
    registration brute-force, which is a *different* threat model from a
    legitimately-registered extension being used for toll fraud — nobody
    should assume CrowdSec alone already covers this.

## Provider landscape (for reference — not chosen)
- **SIP.US** — also prepaid, flat per-channel rate, built-in fraud
  detection. Considered, not chosen.
- Several providers (Nextiva, IDT Express) advertise AI/ML-based fraud
  monitoring as a second layer on top of normal billing — an extra net, not
  a substitute for a hard prepaid ceiling.
- Most providers don't market an explicit "spending cap" feature — the
  prepaid-balance + auto-recharge-off pattern is the de facto mechanism
  across the space, VoIP.ms included.

## Open items for whoever picks this up next
1. ~~Decide: new `services/pstn-trunk.sh`...~~ Done — separate service file,
   generalized to any IP-auth SIP provider (VoIP.ms is just the default).
2. ~~IP auth vs. registration~~ Done — IP authentication, no password stored.
3. ~~Exact NANP dial pattern(s)~~ Done — `_1NXXNXXXXX` / `_NXXNXXXXX`.
4. ~~Inbound~~ Done — rings a configurable list of extensions (ring-group
   supported), each checked live per-call against its own tier. ~~Permission
   model~~ Done — superseded the original flat allow-list with a 3-tier
   model (internal/restricted/full) managed live via
   `pstn-permissions.conf` + the Security Dashboard web UI, no reinstall
   needed to change. ~~Generic Asterisk target~~ Done —
   `services/pstn-trunk.sh` now supports either `asterisk-digital-ocean` or
   the home/LAN `asterisk` install (the latter with a static-IP caveat for
   the provider's IP authentication). Still unresolved: pick pay-per-minute
   vs. unlimited DID plan on VoIP.ms's side based on real expected volume,
   and decide on E911 (see cost estimate).
5. ~~Concurrent-call cap~~ Done — configurable, default 3, global not
   per-extension. ~~Spend/volume alert~~ Done — ntfy, hourly threshold +
   burst check, plus immediate alerts on denied/rejected calls.
6. Verify against a live VoIP.ms account: auto-recharge-off behavior at
   sign-up, and that the chosen POP server's actual source IP for inbound
   calls matches what `services/pstn-trunk.sh` resolved via DNS at install
   time (VoIP.ms's docs mention some redundancy/failover between servers —
   if inbound calls ever stop matching the `identify` section, this is the
   first thing to check).