Claude
1e2a3743ab
Generalizes services/pstn-trunk.sh (renamed from voipms-trunk.sh in the prior commit) away from VoIP.ms specifics - any IP-authenticated SIP provider works, VoIP.ms is just the suggested default. Adds: - Role-based outbound permission: a configurable allow-list of extensions that may dial PSTN numbers (regex-gated on CHANNEL(peername)), separate from internal extension-to-extension dialing which stays open to everyone regardless. Blank list preserves the original "everyone can dial out" behavior. - Inbound ring-group: rings a configurable list of extensions instead of a single hardcoded one. - ntfy alerts: immediate on denied (unauthorized extension) or rejected (concurrency cap hit) calls, plus an hourly cron-driven check that alerts once per month when estimated spend crosses a threshold and every hour call volume looks like a burst. Uses a self-contained pipe-delimited call log rather than Asterisk's CDR, to avoid depending on CDR module availability and CSV comma-quoting. - Settings persisted to .pstn-trunk.env so "update in place" reapplies everything from that file instead of fragile re-parsing out of generated Asterisk config (which had a real bug: update mode was extracting the wrong Dial(PJSIP/...) line). Tested end-to-end against a sandboxed copy of the real vendor files: permission-gate regex, ring-group dial-string construction, ntfy line injection/removal, and the usage-alert script's threshold/burst/monthly- dedup logic all verified with synthetic data. Caught and fixed a sed `&` escaping bug in the ring-group substitution before it shipped (RING_DIAL contains literal `&` join characters, which sed's replacement syntax otherwise treats as "insert the match").
170 lines
10 KiB
Markdown
170 lines
10 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` is installed). Generic SIP trunk
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add-on that defaults to VoIP.ms but isn't hardcoded to it — any provider
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supporting IP authentication works. Covers: IP-authenticated trunk,
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US/NANP-only outbound dialplan, a configurable concurrent-call cap (default
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3), **role-based outbound permission** (some extensions internal-only, some
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PSTN-enabled — internal intercom dialing is never gated either way), a
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configurable **inbound ring-group** (one extension or several), **ntfy
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alerts** on denied/rejected calls (immediate) and spend/volume thresholds
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(hourly check), and settings persisted to `.pstn-trunk.env` so "update in
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place" reapplies everything without re-prompting. That file's own header
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comment explains how it survives Easy Asterisk's config regeneration (an
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architectural wrinkle discovered while implementing this — worth reading
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before touching either 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. This
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behavior wasn't verified against a live account — confirm the
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auto-recharge toggle still works this way at sign-up time, since billing
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UX can change.
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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 cap in `services/pstn-trunk.sh` is a
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*global* cap (configurable, default max 3 outbound legs total via the trunk,
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via `GROUP()`/`GROUP_COUNT()` in the dialplan, shared across all extensions)
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— not per-extension. That was the explicit ask when this got built. The
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spend/volume alert is also implemented now: 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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- **Role-based outbound permission — implemented.** A space-separated list
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of extensions allowed to dial PSTN, prompted at install (blank = every
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extension, the original default before roles existed). Baked into the
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dialplan as a `REGEX()` check against `${CHANNEL(peername)}` — no changes
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needed to Easy Asterisk's own per-device pjsip.conf sections, since the
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gate lives entirely in code this repo already owns. Internal
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extension-to-extension dialing is never gated by this, regardless of PSTN
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permission — only the two NANP patterns above are.
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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.
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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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concurrency cap, ntfy topic, and spend-alert settings at install time.
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- Defense-in-depth alongside the trunk:
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- **Implemented:** a global concurrent-call cap in the dialplan
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(`GROUP()`/`GROUP_COUNT()`, configurable, default max 3 outbound legs via
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the trunk at once) so an unauthorized or compromised extension can't
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open dozens of simultaneous outbound legs. Global, 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), prompted at install time. ~~Role-based outbound permission~~
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Done — space-separated allow-list, blank = everyone. Still unresolved:
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pick pay-per-minute vs. unlimited DID plan on VoIP.ms's side based on
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real expected volume, and decide on E911 (see cost estimate).
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5. ~~Concurrent-call cap~~ Done — configurable, default 3, global not
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per-extension. ~~Spend/volume alert~~ Done — ntfy, hourly threshold +
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burst check, plus immediate alerts on denied/rejected calls.
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6. Verify against a live VoIP.ms account: auto-recharge-off behavior at
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sign-up, and that the chosen POP server's actual source IP for inbound
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calls matches what `services/pstn-trunk.sh` resolved via DNS at install
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time (VoIP.ms's docs mention some redundancy/failover between servers —
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if inbound calls ever stop matching the `identify` section, this is the
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first thing to check).
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