Claude
8e2552ae73
The user found ToS language stating the account "may run on a negative balance" and describing a 30-day-notice-then-suspend / 30-consecutive- days-then-close process - worth reading against the wiki's "balance must be over $0 to call" claim the whole toll-fraud design leans on. Reconciled: these describe two different things, not a contradiction. New call attempts should still be blocked in real time at $0 (the wiki's claim, and the core assumption this design needs). The ToS's negative- balance language most plausibly covers recurring fees (DID/E911) landing when balance is already near zero, and in-progress-call settlement edge cases - not a window where fraud keeps dialing while negative. Still not verified against a live account either way.
270 lines
17 KiB
Markdown
270 lines
17 KiB
Markdown
# PSTN Calling via VoIP.ms — Planning Notes
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Research and decisions from a design discussion, saved here so the work can
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be picked up in a fresh chat without re-deriving the background.
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**Implemented** — see `services/pstn-trunk.sh` (run `sudo ./setup.sh
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pstn-trunk` after `asterisk-digital-ocean` **or** `asterisk` (home/LAN) is
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installed — both are supported, see the file for the static-IP caveat on the
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LAN variant). Generic SIP trunk add-on that defaults to VoIP.ms but isn't
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hardcoded to it — any provider supporting IP authentication works. Covers:
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- IP-authenticated trunk, US/NANP-only outbound dialplan, no catch-all.
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- **Two independent concurrent-call caps**, one per direction (default 10
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outbound / 10 inbound — bumped up from an initial default of 3 once roles
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existed to gate who can even reach the trunk; "ability creep is real," so
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the two caps stay the hard backstop regardless). Global per direction, not
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per-extension.
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- **Three-tier per-extension permission model**: `internal` (default — no
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PSTN at all, but can always call/receive other extensions and internal
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ring groups), `restricted` (also only pre-approved US numbers, both
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directions), `full` (also any US number). Internal extension-to-extension
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dialing is *never* gated by any tier — deliberately, even though VoIP.ms
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itself offers free SIP-to-SIP calling, to avoid routing purely-internal
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calls through an extra external hop for no benefit.
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- **Permissions AND concurrency caps are both live, not baked into the
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dialplan.** Stored in `pstn-permissions.conf` / `pstn-limits.conf`, read
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by the dialplan via Asterisk's `AST_CONFIG()` on every call — editing
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either file takes effect on the next call, no restart, no reinstall.
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`services/pstn-trunk.sh`'s "update in place" mode deliberately never
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touches either (same protection this repo's update-mode convention
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already gives `.env`/firewall/Caddy config
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elsewhere) — only a "fresh" reinstall (with confirmation) or the web UI
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below change it.
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- A configurable **inbound ring-group** (one extension or several), each
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member's live tier/approved-numbers checked per inbound call via an
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unrolled per-member dialplan block (no AGI needed).
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- **`services/security-dashboard.sh` integration** — a "PSTN Trunk" tab
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shows both concurrency caps and every extension (parsed from
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`pjsip.conf`) with its live tier and approved numbers, all editable with
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no restart. This is what makes the tier model and caps actually
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manageable day-to-day instead of needing a reinstall for every change.
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- **ntfy alerts** on denied/rejected calls (immediate — permission denied,
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number not approved, or either concurrency cap hit) and spend/volume
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thresholds (hourly check: once/month on a spend threshold, every hour on
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a call-burst threshold).
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- Structural settings (server, DID, ring-group *membership*, ntfy,
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rate/thresholds) persist to `.pstn-trunk.env` so "update in place"
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reapplies them without re-prompting — the concurrency cap *numbers*
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themselves are not structural, they live in `pstn-limits.conf` instead
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(see above).
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`services/pstn-trunk.sh`'s own header comment explains how the trunk/dialplan
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config survives Easy Asterisk's regeneration, and why permissions are a
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separate live file rather than baked in — both architectural wrinkles
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discovered while implementing this, worth reading before touching either
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file.
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## Decision so far
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- **Provider: VoIP.ms.** Chosen for its prepaid-balance model: turn off
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auto-recharge in the account's Finances settings and outbound calls simply
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fail once the balance hits $0 — that's the toll-fraud backstop if the
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droplet's Asterisk (`asterisk-digital-ocean`) is ever compromised.
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**Update — read VoIP.ms's actual ToS (not just the wiki) on this.** The
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wiki says plainly "only accounts with a balance over $0 are able to send
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and receive calls" — new call attempts should be blocked in real time at
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$0, and that's still the core assumption this design leans on. But the
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ToS separately says the account "may run on a negative balance," that any
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negative balance is "immediately due and payable," that VoIP.ms may
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suspend an account below a $5 minimum balance (30-day notice first), and
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may permanently close it after 30 *consecutive* days negative. Read
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together, not a contradiction — two different things:
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- **Can new calls start** — real-time balance check, blocked at $0. Core
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assumption holds.
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- **Can the balance ever read negative** — yes, most plausibly from
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recurring fees (DID monthly, E911) landing when the balance is already
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near zero, or edge-case settlement of an in-progress call ticking
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slightly negative before teardown. Neither is a runaway toll-fraud
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scenario; both mean liability isn't cleanly capped at the funded amount
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to the exact penny, and the account needs topping up within the 30-day
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windows or it gets suspended/closed (an account-status consequence, not
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"30 free days of unblocked calling while negative").
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- Still not verified against an actual live account — this is a read of
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their published wiki + ToS text, not a test. Watch the real balance for
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the first month or two after go-live, and don't panic at a small
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negative reading — check whether it's a recurring fee or an actual call
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spike before assuming the block failed.
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- **Scope: US calling only, for now.** No international, no premium-rate
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destinations. Enforce this twice — once via whatever dial-plan/prefix
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VoIP.ms requires for US routing, and again independently in Asterisk's own
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dialplan (see below), so a compromised extension can't reach anything
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outside the US even if the trunk itself would technically allow more later.
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- **Inbound: wanted.** A DID is in scope, not outbound-only. Decide pay-per-minute
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vs. unlimited DID plan based on expected inbound volume (see cost estimate
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below), and decide E911 deliberately rather than skipping it by default —
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VoIP.ms doesn't require it, but without it 911 dialed from the line either
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fails or doesn't carry accurate address/location info.
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## Cost estimate (100 min/month each direction, US-only)
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Verified against VoIP.ms's public wiki/rate pages, not a live account — confirm
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at sign-up since rates can change.
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| Item | Rate | Monthly | Annual |
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|---|---|---|---|
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| DID (phone number), pay-per-minute plan | $0.85/mo flat | — | $10.20 |
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| Inbound usage | $0.009/min | $0.90 | $10.80 |
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| Outbound usage | $0.01/min | $1.00 | $12.00 |
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| **Total** | | **~$2.75** | **~$33** |
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- Skipping the DID (outbound-only) drops this to ~$12/year.
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- Adding E911 adds a $1.50 one-time fee plus **$1.50/month** regulatory fee
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(~$18/year) — pushes the total above to ~$51/year.
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- **Funding minimum:** VoIP.ms requires a **$15 minimum deposit** to activate
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calling — a one-time balance top-up, not a recurring charge. At ~$2.75/month
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usage that balance lasts ~5 months before a refill is needed (longer at
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lower volume). Leave auto-recharge **off** per the toll-fraud design above.
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## Why this matters (toll fraud)
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A compromised Asterisk box can dial premium-rate or international numbers
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that cost real money fast (some destinations run several $/min) before
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anyone notices. Two independent layers matter more than either alone:
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1. **Trunk-side cap** — prepaid balance, auto-recharge off. Limits total
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possible loss to whatever the balance is topped up to, but a live
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compromise could still burn through that balance in minutes if nothing
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else restricts what can be dialed.
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2. **Dialplan-side restriction** — Asterisk should refuse to route calls
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outside the US/NANP pattern at all, regardless of what the trunk allows.
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This is the first line of defense and should exist independent of the
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trunk's own capabilities.
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**Important nuance: these two layers bound different things, and neither
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alone bounds both.** NANP-only restriction bounds *cost-per-minute* (a
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compromised box can only ever reach $0.01/min US numbers, never $2–5/min
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international/premium destinations) — that risk is fully closed. It does
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**not** bound *how fast* the prepaid balance gets burned: nothing stops a
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compromised box from opening many concurrent US-destination calls in
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parallel and draining the whole balance (e.g. $15 balance ÷ $0.01/min =
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1,500 minutes total, which 20 concurrent legs could burn through in under
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an hour). The prepaid-balance-off-auto-recharge layer bounds the *dollar*
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ceiling; only a concurrent-call cap bounds the *speed* of a breach. Treat
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the concurrent-call cap and spend/volume alert below as required before
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funding a live trunk, not optional hardening.
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**Implemented:** the concurrent-call caps in `services/pstn-trunk.sh` are
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*global* per direction (default 10 outbound / 10 inbound, each tracked via
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its own `GROUP()`/`GROUP_COUNT()` in the dialplan, shared across all
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extensions) — not per-extension. Inbound didn't have a cap at all until
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this was pointed out as a gap (outbound's cap doesn't protect against an
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inbound call-flood, which also costs money per-minute on VoIP.ms) — both
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directions are covered symmetrically now. The spend/volume alert is also
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implemented: an hourly cron script reads a
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call log the dialplan appends to directly (not Asterisk's CDR — see the
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service file's own comments for why) and alerts via ntfy once per month when
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estimated spend crosses a threshold, and every hour that call volume in the
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last hour looks like a burst. Denied/rejected calls alert immediately,
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separately from that hourly check.
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## What it takes technically (asterisk-digital-ocean)
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- A PJSIP trunk: `endpoint` / `aor` / `identify` sections in the pjsip
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config. **Implemented with IP authentication** (no `auth` section, no SIP
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password stored anywhere) — see `services/pstn-trunk.sh`. Provider name,
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server hostname, and DID are all prompted at install time (VoIP.ms is only
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the suggested default), so any provider supporting IP auth works.
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- An outbound dialplan route matching US numbers only — **implemented**:
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`_1NXXNXXXXX` (11-digit NANP with leading 1) and `_NXXNXXXXX` (10-digit,
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auto-prefixed with 1), both routed to the trunk. No catch-all `_X.`
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pattern.
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- **Three-tier permission model — implemented**, superseding an earlier flat
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allow-list design. `internal` / `restricted` / `full` per extension, read
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live from `pstn-permissions.conf` via `AST_CONFIG()` rather than baked
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into the dialplan text — no changes needed to Easy Asterisk's own
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per-device pjsip.conf sections, since the gate lives entirely in files
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this repo already owns. Internal extension-to-extension dialing is never
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gated by any tier — only the two NANP patterns (outbound) and the
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ring-group (inbound) are. Numbers are stored pipe-separated specifically
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because they're used as a `REGEX()` alternation pattern in the dialplan —
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see the security note below on why the untrusted call-time value must
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never be interpolated into the *pattern* side of that check.
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- **Inbound ring-group — implemented.** A space-separated list of
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extensions to ring for inbound calls (one, or several for a ring group via
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`Dial(PJSIP/a&PJSIP/b,20)`), prompted at install. Each member's tier is
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checked live per inbound call (an unrolled dialplan block per member —
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full always rings, restricted only rings if the caller's number is on
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that member's approved list, internal never rings).
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- **Security note on the REGEX() checks**: an inbound Caller-ID (or an
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outbound dialed number) is attacker-influenced data and must never be
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interpolated into the *pattern* argument of `REGEX()` — only ever the
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string being tested. Doing it backwards would let a crafted Caller-ID
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(e.g. containing regex metacharacters) forge a match against an unrelated
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approved-numbers entry. Both checks in `services/pstn-trunk.sh` put the
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admin-controlled approved-list in the pattern position and the live call
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data in the tested-string position — worth keeping that direction if this
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is ever refactored.
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- **Web UI — implemented.** `services/security-dashboard.sh`'s "PSTN Trunk"
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tab lists every extension (parsed from `pjsip.conf`, the same marker
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format Easy Asterisk's own `rebuild_dialplan()` uses) with a tier dropdown
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and approved-numbers field, saving straight to `pstn-permissions.conf`.
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Tested end-to-end with a real running instance of the (stdlib-only)
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Python app: extension parsing, tier changes, number normalization/
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validation, and persistence all verified with actual HTTP requests against
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a live server in a sandboxed test — not just read through.
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- Provider-specific setup that isn't scriptable (user does this manually):
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create the account, order a DID, decide pay-per-minute vs. unlimited DID
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plan and whether to add E911, pick a server/POP, fund the prepaid balance
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($15 minimum for VoIP.ms), turn off auto-recharge. `services/pstn-trunk.sh`
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prompts for the server hostname, DID, allowed extensions, ring extensions,
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both concurrency caps, ntfy topic, and spend-alert settings at install
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time (the cap *numbers* are then live/web-editable afterward — see above).
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- Defense-in-depth alongside the trunk:
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- **Implemented:** independent outbound/inbound concurrent-call caps in
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the dialplan (`GROUP()`/`GROUP_COUNT()`, default 10/10) so an
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unauthorized or compromised extension can't open dozens of simultaneous
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legs in either direction. Global per direction, not per-extension —
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see the note above.
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- **Implemented:** an outbound call-count/spend alert via ntfy — a
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self-contained call log (not Asterisk's CDR) plus an hourly cron script.
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Denied/rejected calls also alert immediately. See
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`services/pstn-trunk.sh`'s "Spend/volume alerts" README section for the
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exact mechanics and why CDR wasn't used.
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- Worth being explicit that CrowdSec's existing `asterisk_bf` /
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`asterisk_user_enum` scenarios (see `services/crowdsec.sh`) cover
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registration brute-force, which is a *different* threat model from a
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legitimately-registered extension being used for toll fraud — nobody
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should assume CrowdSec alone already covers this.
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## Provider landscape (for reference — not chosen)
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- **SIP.US** — also prepaid, flat per-channel rate, built-in fraud
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detection. Considered, not chosen.
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- Several providers (Nextiva, IDT Express) advertise AI/ML-based fraud
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monitoring as a second layer on top of normal billing — an extra net, not
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a substitute for a hard prepaid ceiling.
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- Most providers don't market an explicit "spending cap" feature — the
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prepaid-balance + auto-recharge-off pattern is the de facto mechanism
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across the space, VoIP.ms included.
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## Open items for whoever picks this up next
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1. ~~Decide: new `services/pstn-trunk.sh`...~~ Done — separate service file,
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generalized to any IP-auth SIP provider (VoIP.ms is just the default).
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2. ~~IP auth vs. registration~~ Done — IP authentication, no password stored.
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3. ~~Exact NANP dial pattern(s)~~ Done — `_1NXXNXXXXX` / `_NXXNXXXXX`.
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4. ~~Inbound~~ Done — rings a configurable list of extensions (ring-group
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supported), each checked live per-call against its own tier. ~~Permission
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model~~ Done — superseded the original flat allow-list with a 3-tier
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model (internal/restricted/full) managed live via
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`pstn-permissions.conf` + the Security Dashboard web UI, no reinstall
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needed to change. ~~Generic Asterisk target~~ Done —
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`services/pstn-trunk.sh` now supports either `asterisk-digital-ocean` or
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the home/LAN `asterisk` install (the latter with a static-IP caveat for
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the provider's IP authentication). Still unresolved: pick pay-per-minute
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vs. unlimited DID plan on VoIP.ms's side based on real expected volume,
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and decide on E911 (see cost estimate).
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5. ~~Concurrent-call cap~~ Done — both directions now (inbound was a real
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gap, since it also costs money per-minute and outbound's cap doesn't
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cover it), default 10/10, global not per-extension, live-editable via
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`pstn-limits.conf`/web UI. ~~Spend/volume alert~~ Done — ntfy, hourly
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threshold + burst check, plus immediate alerts on denied/rejected calls.
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6. Verify against a live VoIP.ms account (still not done — only their wiki
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+ ToS text has been read, see "Decision so far" above for what that
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turned up): confirm new outbound calls actually get blocked at $0
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balance as documented; watch whether/when the balance goes slightly
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negative in normal operation (expected from recurring fees, not
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necessarily a sign of a problem) and top up within the 30-day windows
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the ToS describes so the account/DID doesn't get suspended or closed.
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Also confirm the chosen POP server's actual source IP for inbound calls
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matches what `services/pstn-trunk.sh` resolved via DNS at install time
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(VoIP.ms's docs mention some redundancy/failover between servers — if
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inbound calls ever stop matching the `identify` section, this is the
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first thing to check).
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