Add voipms-trunk service: US-only outbound PSTN, max 3 concurrent calls
Adds a VoIP.ms SIP trunk on top of asterisk-digital-ocean: IP-authenticated trunk (no password stored), NANP-only outbound dialplan, a global 3-call concurrent cap via GROUP()/GROUP_COUNT(), and inbound routing to one extension. Config lives in its own include files rather than being appended directly to pjsip.conf/extensions.conf, since Easy Asterisk fully regenerates both from its own internal state — the includes are patched into the vendor's generator functions so they survive that regeneration. Wires the new service into setup.sh's is_installed() and README's services table, and updates docs/pstn-calling-voipms-plan.md to reflect what's now implemented vs. still open (spend/volume alerting, live-account verification).
This commit is contained in:
Claude
@@ -67,7 +67,7 @@ a ready-to-copy Caddy config snippet to `~/docker/caddy-snippets/`.
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| Group | Services |
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|-------|---------|
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| `base` | `net-tools`, `ncdu`, `git`, `curl`, `wget`, `htop`, `tree`, `zip`/`unzip`, `ca-certificates`, `gnupg`, `jq`, `rsync`; `glow` (terminal markdown reader, Charm apt repo); Docker CE + Compose plugin; `openssh-server` with GitHub/Launchpad SSH key import, optional password-auth lockdown, and SSH Host aliases; optional NetBird overlay network |
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| `homelab` | `caddy`, `crowdsec`, `authelia`, `homeassistant`, `asterisk`, `asterisk-digital-ocean`, `security-dashboard`, `sunshine` |
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| `homelab` | `caddy`, `crowdsec`, `authelia`, `homeassistant`, `asterisk`, `asterisk-digital-ocean`, `voipms-trunk`, `security-dashboard`, `sunshine` |
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| `utilities` | `actualbudget`, `ai-gpu`, `ai-stack`, `archivebox`, `changedetection`, `ddclient`, `filebrowser`, `fmd`, `gatus`, `homebox`, `iopaint`, `joplin`, `koha`, `magicmirror`, `mail-archiver`, `mattermost`, `mealie`, `meshcentral`, `n8n`, `nextcloud`, `ntfy`, `onlyoffice`, `paintplus`, `portainer`, `rustdesk`, `stirling-pdf`, `syncthing`, `traccar`, `unifi`, `uptimekuma`, `vaultwarden`, `watchyourlan`, `watchtower`, `wg-easy` |
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| `media` | `arm`, `audiobookshelf`, `calibre-web`, `emby`, `immich`, `jellyfin`, `lyrion` |
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| `cameras` | `frigate`, `frigate-audio`, `frigate-notify`, `sky-cam` |
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@@ -92,6 +92,7 @@ homelab
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homeassistant
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asterisk
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asterisk-digital-ocean
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voipms-trunk
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security-dashboard
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sunshine
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@@ -1,9 +1,15 @@
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# 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. **Nothing
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has been implemented yet** — this is prep for a future `services/*.sh`
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addition on top of `asterisk-digital-ocean`.
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be picked up in a fresh chat without re-deriving the background.
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**Implemented** — see `services/voipms-trunk.sh` (run `sudo ./setup.sh
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voipms-trunk` after `asterisk-digital-ocean` is installed). IP-authenticated
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trunk, US/NANP-only outbound dialplan, 3-concurrent-call cap, inbound to one
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extension. That file's own header comment explains how it survives Easy
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Asterisk's config regeneration (an architectural wrinkle discovered while
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implementing this — worth reading before touching either file). The
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outbound spend/volume alert mentioned below is still not implemented.
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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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@@ -67,33 +73,38 @@ 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 per-extension concurrent-call cap and spend/volume alert below as
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required before funding a live trunk, not optional hardening.
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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/voipms-trunk.sh` is a
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*global* cap (max 3 outbound legs total via the trunk, via
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`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; a
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per-extension cap layered on top is still a possible future refinement, not
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done. The spend/volume alert is still not implemented.
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## What it takes technically (asterisk-digital-ocean)
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- A PJSIP trunk to VoIP.ms: `endpoint` / `aor` / `auth` / `identify`
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sections in the pjsip config, using either IP authentication or SIP
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registration — VoIP.ms supports both. IP auth is simpler for a droplet
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(it has a static IP already) and avoids storing a SIP password in the
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config at all — worth confirming with VoIP.ms which they actually
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recommend before choosing.
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- An outbound dialplan route matching US numbers only, e.g. `_1NXXNXXXXX`
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(11-digit NANP with leading 1) or `_NXXNXXXXX`, depending on how numbers
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get dialed from the existing extensions, routed to the VoIP.ms trunk. No
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catch-all `_X.` pattern — an explicit NANP pattern is itself a hard block
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on non-US destinations at the dialplan level.
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- A PJSIP trunk to VoIP.ms: `endpoint` / `aor` / `identify` sections in the
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pjsip config. **Implemented with IP authentication** (no `auth` section,
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no SIP password stored anywhere) — see `services/voipms-trunk.sh`.
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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 VoIP.ms trunk. No catch-all
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`_X.` pattern.
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- VoIP.ms-specific setup that isn't scriptable (user does this manually):
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create the account, order a DID (inbound is wanted — see above), decide
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pay-per-minute vs. unlimited DID plan and whether to add E911, pick a
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VoIP.ms POP/server (affects the trunk hostname), fund the prepaid balance
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($15 minimum), turn off auto-recharge.
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- Defense-in-depth to design alongside the trunk (not yet designed):
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- Per-extension concurrent-call cap in the dialplan (`GROUP()` /
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`GROUP_COUNT()`) so one compromised extension can't open dozens of
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simultaneous outbound legs at once.
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- A simple outbound call-count/spend alert — could live in the existing
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`security-dashboard` service (see `services/security-dashboard.sh`) or
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as a separate CDR-based check. Not designed yet.
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($15 minimum), turn off auto-recharge. `services/voipms-trunk.sh` prompts
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for the POP hostname, DID, and a ring extension for inbound 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()`, max 3 outbound legs via the trunk at once)
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so a compromised extension can't open dozens of simultaneous outbound
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legs. Global, not per-extension — see the note above.
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- **Not implemented:** a simple outbound call-count/spend alert — could
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live in the existing `security-dashboard` service (see
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`services/security-dashboard.sh`) or as a separate CDR-based check.
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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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@@ -111,16 +122,21 @@ required before funding a live trunk, not optional hardening.
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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/voipms-trunk.sh`, or an optional trunk section
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added directly to `services/asterisk-digital-ocean.sh`? Leaning toward a
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separate service file so trunk config isn't forced on installs that
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don't want PSTN calling, matching this repo's one-feature-per-file
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convention (see CLAUDE.md).
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2. IP auth vs. registration — confirm which VoIP.ms recommends for a single
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fixed-IP droplet.
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3. Exact NANP dial pattern(s) and any prefix-stripping VoIP.ms requires.
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4. Inbound is decided (wanted) — still need to pick pay-per-minute vs.
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unlimited DID plan based on real expected volume, and decide on E911.
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5. Design the concurrent-call cap and any spend/volume alerting mentioned
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above — treat as required before funding a live trunk, not optional
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(see toll-fraud nuance above: NANP-only bounds cost/min, not burn speed).
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1. ~~Decide: new `services/voipms-trunk.sh`...~~ Done — separate service file.
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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 one extension, prompted at install time. Still
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unresolved: pick pay-per-minute vs. unlimited DID plan on VoIP.ms's side
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based on real expected volume, and decide on E911 (see cost estimate).
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5. ~~Concurrent-call cap~~ Done — global 3-call cap. Still not implemented:
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the spend/volume alert (security-dashboard integration or CDR-based
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check) — treat as still-required before fully trusting this against a
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sustained breach, not optional (see toll-fraud nuance above: the NANP
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restriction + call cap bound cost/min and burn speed, but nothing here
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yet notices a breach in progress or alerts on unusual volume).
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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/voipms-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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@@ -0,0 +1,389 @@
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#!/bin/bash
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# services/voipms-trunk.sh — VoIP.ms SIP trunk add-on for asterisk-digital-ocean:
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# US-only outbound (NANP dialplan restriction), capped at 3 concurrent
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# outbound calls, IP-authenticated trunk (no SIP password stored), plus
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# inbound routing to one extension.
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#
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# Requires an existing services/asterisk-digital-ocean.sh install — this adds
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# a PSTN trunk on top of it and does not stand alone. See
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# docs/pstn-calling-voipms-plan.md for the design/cost background this is
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# built from.
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#
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# Part of the modular post-install system (sourced by setup.sh).
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register_service voipms-trunk homelab "VoIP.ms SIP trunk for asterisk-digital-ocean — US-only outbound, max 3 concurrent calls"
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# ── Surviving Easy Asterisk's regeneration ──────────────────────────────────
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# Easy Asterisk (the vendor project asterisk-digital-ocean.sh builds on) fully
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# OVERWRITES both pjsip.conf and extensions.conf from its own internal state:
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# - extensions.conf: rebuilt by rebuild_dialplan() on every container start,
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# and whenever a device/room is added or removed via the web admin.
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# - pjsip.conf: rewritten by generate_pjsip_conf() whenever VLAN/domain/TLS
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# settings are changed via the CLI menu (docker exec ... easy-asterisk).
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# It restores only its own "; === Device:"-marked sections from backup —
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# a hand-appended trunk section would be silently wiped the next time
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# that runs.
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# So the trunk/dialplan content below lives in its own files and is
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# #include'd from the generated files instead of appended directly. To make
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# the #include itself survive regeneration too, _voipms_patch_vendor_files
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# (below) patches it into the vendor's *generator functions* — the same
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# technique this repo already uses for the logger.conf security-logging fix
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# in _asterisk_do_refresh_vendor_files (see services/asterisk-digital-ocean.sh).
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#
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# Caveat: if the base asterisk-digital-ocean install is later refreshed
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# ("update in place", which re-copies fresh vendor files) independently of
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# this service, the patch is wiped along with it and needs reapplying — run
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# this service again (fresh or update mode both reapply it) after any
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# asterisk-digital-ocean update.
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# ── Shared: patch vendor generator functions to #include our config ────────
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# Anchors on "user_agent=EasyAsterisk" (pjsip.conf's [global] section) and
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# "[intercom]" (extensions.conf) — each confirmed to appear exactly once per
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# file in the vendor source, so this is safe regardless of what else changes
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# around it upstream. Idempotent: skips files that already have the include.
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_voipms_patch_vendor_files() {
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local EA_DIR="$1"
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local ENTRYPOINT="$EA_DIR/docker/entrypoint.sh"
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local EASY1="$EA_DIR/easy-asterisk.sh"
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local EASY2="$EA_DIR/easy-asterisk-v0.10.0.sh"
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local f
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for f in "$ENTRYPOINT" "$EASY1" "$EASY2"; do
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[[ -f "$f" ]] || { log_error "$f not found — is asterisk-digital-ocean fully installed?"; return 1; }
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done
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for f in "$ENTRYPOINT" "$EASY1" "$EASY2"; do
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if ! grep -q 'voipms-trunk-pjsip.conf' "$f"; then
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if grep -q '^user_agent=EasyAsterisk$' "$f"; then
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sed -i '/^user_agent=EasyAsterisk$/a #include voipms-trunk-pjsip.conf' "$f"
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else
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log_warning "$(basename "$f"): 'user_agent=EasyAsterisk' anchor not found — vendor template changed upstream."
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log_warning " Add '#include voipms-trunk-pjsip.conf' manually after [global] in this file's pjsip.conf heredoc."
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fi
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fi
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done
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for f in "$ENTRYPOINT" "$EASY1" "$EASY2"; do
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if ! grep -q 'voipms-trunk-dialplan.conf' "$f"; then
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if grep -q '^\[intercom\]$' "$f"; then
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sed -i '/^\[intercom\]$/a #include voipms-trunk-dialplan.conf' "$f"
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else
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log_warning "$(basename "$f"): '[intercom]' anchor not found — vendor template changed upstream."
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log_warning " Add '#include voipms-trunk-dialplan.conf' manually after [intercom] in this file's extensions.conf heredoc."
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fi
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fi
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done
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log_success "Vendor generator functions patched to include the VoIP.ms trunk config."
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}
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# ── Shared: pjsip trunk config (aor/identify/endpoint, IP-authenticated) ───
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_voipms_write_pjsip_include() {
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local FILE="$1" SERVER="$2" SERVER_IP="$3" DID="$4"
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cat > "$FILE" << 'EOF'
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; VoIP.ms trunk — IP authentication, no password stored (see
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; docs/pstn-calling-voipms-plan.md). Regenerated by services/voipms-trunk.sh —
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; edit there, not here directly, or a reinstall/update will overwrite this.
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;
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; match= below is the resolved IP of the server hostname at install time.
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; VoIP.ms's servers are fixed per-POP, but if calls stop matching after a
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; provider-side change, re-run this service to re-resolve and rewrite it.
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[voipms-trunk]
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type=aor
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contact=sip:__VOIPMS_SERVER__
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qualify_frequency=60
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[voipms-trunk]
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type=identify
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endpoint=voipms-trunk
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match=__VOIPMS_SERVER_IP__
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[voipms-trunk]
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type=endpoint
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context=from-voipms
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disallow=all
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allow=ulaw,alaw
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aors=voipms-trunk
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from_user=__VOIPMS_DID__
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from_domain=__VOIPMS_SERVER__
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callerid=__VOIPMS_DID__
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direct_media=no
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EOF
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sed -i "s/__VOIPMS_SERVER_IP__/${SERVER_IP}/g; s/__VOIPMS_SERVER__/${SERVER}/g; s/__VOIPMS_DID__/${DID}/g" "$FILE"
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}
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# ── Shared: outbound/inbound dialplan (NANP-only, 3-concurrent cap) ────────
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# Continues in the [intercom] context established just above this include
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# (rebuild_dialplan() writes "[intercom]" then this #include right after it),
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# so existing extensions can dial out through it directly. [from-voipms]
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# below is a separate context, for calls arriving from the trunk.
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_voipms_write_dialplan_include() {
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local FILE="$1" DID="$2" RING_EXT="$3"
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cat > "$FILE" << 'EOF'
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; VoIP.ms outbound/inbound — US-only (NANP), max 3 concurrent outbound calls.
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; Regenerated by services/voipms-trunk.sh — edit there, not here directly.
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;
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; No catch-all pattern here on purpose: only these two NANP patterns route
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; to the trunk, so a compromised extension can't reach anything else even if
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; the trunk itself would technically allow more. See
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; docs/pstn-calling-voipms-plan.md for the full toll-fraud reasoning.
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exten => _1NXXNXXXXX,1,NoOp(VoIP.ms outbound call to ${EXTEN})
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same => n,GotoIf($[${GROUP_COUNT(voipms-out)} >= 3]?voipms_busy,1)
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same => n,Set(GROUP()=voipms-out)
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same => n,Set(CALLERID(num)=__VOIPMS_DID__)
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same => n,Dial(PJSIP/${EXTEN}@voipms-trunk,60)
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same => n,Hangup()
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exten => _NXXNXXXXX,1,NoOp(Assuming NANP, adding leading 1)
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same => n,Goto(1${EXTEN},1)
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exten => voipms_busy,1,NoOp(VoIP.ms: 3 concurrent outbound calls already active — rejecting)
|
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same => n,Busy(15)
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same => n,Hangup()
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|
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[from-voipms]
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exten => _X.,1,NoOp(Inbound PSTN call from ${CALLERID(num)})
|
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same => n,Dial(PJSIP/__VOIPMS_RING_EXT__,20)
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same => n,Hangup()
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EOF
|
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sed -i "s/__VOIPMS_DID__/${DID}/g; s/__VOIPMS_RING_EXT__/${RING_EXT}/g" "$FILE"
|
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}
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install_voipms-trunk() {
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require_docker || return 1
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local EA_DIR="$DOCKER_DIR/asterisk-digital-ocean"
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local ASTERISK_DIR="$EA_DIR/config/asterisk"
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local PJSIP_INCLUDE="$ASTERISK_DIR/voipms-trunk-pjsip.conf"
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local DIALPLAN_INCLUDE="$ASTERISK_DIR/voipms-trunk-dialplan.conf"
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if [ "$DRY_RUN" = true ]; then
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echo "[DRY-RUN] Would require an existing asterisk-digital-ocean install at $EA_DIR"
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echo "[DRY-RUN] Would prompt for VoIP.ms server/POP hostname, DID, and ring extension"
|
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echo "[DRY-RUN] Would resolve the server hostname to an IP for inbound call matching"
|
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echo "[DRY-RUN] Would patch vendor generator functions to #include the trunk config"
|
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echo "[DRY-RUN] Would write $PJSIP_INCLUDE and $DIALPLAN_INCLUDE"
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echo "[DRY-RUN] Would offer 'update in place' instead of a fresh install if already configured"
|
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echo "[DRY-RUN] Would restart the asterisk container to apply"
|
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return 0
|
||||
fi
|
||||
|
||||
if [[ ! -f "$EA_DIR/docker-compose.yml" ]]; then
|
||||
log_error "asterisk-digital-ocean isn't installed at $EA_DIR — install it first:"
|
||||
log_error " sudo ./setup.sh asterisk-digital-ocean"
|
||||
log_error "This service adds a VoIP.ms PSTN trunk on top of it; it doesn't stand alone."
|
||||
return 1
|
||||
fi
|
||||
|
||||
log_info "Configuring VoIP.ms SIP trunk for asterisk-digital-ocean..."
|
||||
log_info "US-only outbound (NANP dialplan), max 3 concurrent outbound calls, IP-authenticated trunk."
|
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echo ""
|
||||
log_warning "Before continuing, on VoIP.ms's side you should already have: created an"
|
||||
log_warning "account, funded the \$15 minimum prepaid balance, turned OFF auto-recharge"
|
||||
log_warning "(Client Area -> Balance Management), ordered a DID with IP authentication"
|
||||
log_warning "pointed at this droplet's public IP, and picked a server/POP. Also restrict"
|
||||
log_warning "outbound routing to US/NANP on VoIP.ms's own side if it offers that — this"
|
||||
log_warning "dialplan is the second, independent layer, not a substitute for the first."
|
||||
log_warning "See docs/pstn-calling-voipms-plan.md for the full background."
|
||||
echo ""
|
||||
|
||||
# ── Existing install? Offer update-in-place instead of a full reinstall ──
|
||||
if [[ -f "$PJSIP_INCLUDE" && -f "$DIALPLAN_INCLUDE" ]]; then
|
||||
log_info "Existing VoIP.ms trunk config found."
|
||||
local REINSTALL_MODE=""
|
||||
prompt_reinstall_mode REINSTALL_MODE
|
||||
case "$REINSTALL_MODE" in
|
||||
update)
|
||||
local _SERVER _SERVER_IP _DID _RING_EXT
|
||||
_SERVER="$(grep -m1 '^from_domain=' "$PJSIP_INCLUDE" | cut -d= -f2-)"
|
||||
_SERVER_IP="$(grep -m1 '^match=' "$PJSIP_INCLUDE" | cut -d= -f2-)"
|
||||
_DID="$(grep -m1 '^from_user=' "$PJSIP_INCLUDE" | cut -d= -f2-)"
|
||||
# Match only the inbound line (Dial(PJSIP/<digits>,20)) — the
|
||||
# outbound line matches Dial(PJSIP/${EXTEN}@voipms-trunk,60),
|
||||
# which starts with "$" right after PJSIP/, not a digit.
|
||||
_RING_EXT="$(grep -m1 -E 'Dial\(PJSIP/[0-9]+,20\)' "$DIALPLAN_INCLUDE" | sed -E 's/.*Dial\(PJSIP\/([0-9]+),20\).*/\1/')"
|
||||
if [[ -z "$_SERVER" || -z "$_DID" || -z "$_RING_EXT" ]]; then
|
||||
log_warning "Couldn't parse existing settings back out — falling back to a fresh install (every prompt below)."
|
||||
else
|
||||
_voipms_patch_vendor_files "$EA_DIR" || return 1
|
||||
_voipms_write_pjsip_include "$PJSIP_INCLUDE" "$_SERVER" "$_SERVER_IP" "$_DID"
|
||||
_voipms_write_dialplan_include "$DIALPLAN_INCLUDE" "$_DID" "$_RING_EXT"
|
||||
ensure_docker_dir_ownership "$ASTERISK_DIR"
|
||||
( cd "$EA_DIR" && docker compose restart asterisk ) \
|
||||
&& log_success "Updated — settings unchanged (server $_SERVER, DID $_DID, ring ext $_RING_EXT)." \
|
||||
|| log_warning "Restart failed — check: docker compose -f $EA_DIR/docker-compose.yml logs asterisk"
|
||||
return 0
|
||||
fi
|
||||
;;
|
||||
cancel)
|
||||
log_info "Leaving the existing VoIP.ms trunk config as-is."
|
||||
return 0
|
||||
;;
|
||||
fresh)
|
||||
log_info "Proceeding with a full fresh reinstall — every prompt below runs from scratch."
|
||||
;;
|
||||
esac
|
||||
fi
|
||||
|
||||
# ── Prompts — VoIP.ms account details aren't scriptable, user sets these
|
||||
# up manually on VoIP.ms's own site first (see warning above) ───────────
|
||||
local VOIPMS_SERVER=""
|
||||
prompt_text "VoIP.ms server/POP hostname (e.g. atlanta2.voip.ms — pick the one closest to this droplet from VoIP.ms's server list):" "" VOIPMS_SERVER
|
||||
if [[ -z "$VOIPMS_SERVER" ]]; then
|
||||
log_error "A VoIP.ms server hostname is required — aborting."
|
||||
return 1
|
||||
fi
|
||||
|
||||
local VOIPMS_SERVER_IP=""
|
||||
VOIPMS_SERVER_IP="$(getent ahostsv4 "$VOIPMS_SERVER" 2>/dev/null | awk '{print $1}' | head -1)"
|
||||
if [[ -z "$VOIPMS_SERVER_IP" ]]; then
|
||||
log_warning "Couldn't resolve $VOIPMS_SERVER — the identify section needs an IP to match inbound calls against."
|
||||
prompt_text "Enter its IP manually (check VoIP.ms's server list page):" "" VOIPMS_SERVER_IP
|
||||
if [[ -z "$VOIPMS_SERVER_IP" ]]; then
|
||||
log_error "No IP available — aborting."
|
||||
return 1
|
||||
fi
|
||||
else
|
||||
log_success "Resolved $VOIPMS_SERVER -> $VOIPMS_SERVER_IP"
|
||||
fi
|
||||
|
||||
local VOIPMS_DID=""
|
||||
prompt_text "VoIP.ms DID (the 10-digit US phone number assigned to this trunk, digits only):" "" VOIPMS_DID
|
||||
if [[ ! "$VOIPMS_DID" =~ ^[0-9]{10}$ ]]; then
|
||||
log_error "That doesn't look like a 10-digit US number — aborting."
|
||||
return 1
|
||||
fi
|
||||
|
||||
local RING_EXT=""
|
||||
prompt_text "Local extension to ring for inbound calls (must already exist in Easy Asterisk):" "" RING_EXT
|
||||
if [[ -z "$RING_EXT" ]]; then
|
||||
log_error "An extension number is required for inbound routing — aborting."
|
||||
return 1
|
||||
fi
|
||||
|
||||
# ── Patch vendor generator functions, then write the include files ───────
|
||||
_voipms_patch_vendor_files "$EA_DIR" || return 1
|
||||
|
||||
mkdir -p "$ASTERISK_DIR"
|
||||
_voipms_write_pjsip_include "$PJSIP_INCLUDE" "$VOIPMS_SERVER" "$VOIPMS_SERVER_IP" "$VOIPMS_DID"
|
||||
_voipms_write_dialplan_include "$DIALPLAN_INCLUDE" "$VOIPMS_DID" "$RING_EXT"
|
||||
ensure_docker_dir_ownership "$ASTERISK_DIR"
|
||||
chmod 644 "$PJSIP_INCLUDE" "$DIALPLAN_INCLUDE"
|
||||
|
||||
# No new firewall rules: asterisk-digital-ocean.sh already opens SIP
|
||||
# (5060/5061) and RTP (10000-20000) to the internet, and VoIP.ms's source
|
||||
# IPs vary by POP/redundancy, so there's no single IP to scope this to
|
||||
# even if narrowing it were otherwise worthwhile.
|
||||
|
||||
# ── Docs (separate file — asterisk-digital-ocean already owns README.md
|
||||
# in this same directory via write_readme, so don't overwrite it) ───────
|
||||
local DOC_FILE="$EA_DIR/README-voipms-trunk.md"
|
||||
cat > "$DOC_FILE" << MD
|
||||
# VoIP.ms SIP trunk (add-on to asterisk-digital-ocean)
|
||||
|
||||
US-only outbound PSTN calling over a VoIP.ms trunk, capped at 3 concurrent
|
||||
outbound calls, plus inbound routing to one extension. See
|
||||
\`docs/pstn-calling-voipms-plan.md\` in the repo for the full design
|
||||
background, cost estimate, and toll-fraud reasoning.
|
||||
|
||||
## Current settings
|
||||
|
||||
| Setting | Value |
|
||||
|---|---|
|
||||
| VoIP.ms server/POP | ${VOIPMS_SERVER} (${VOIPMS_SERVER_IP}) |
|
||||
| DID | ${VOIPMS_DID} |
|
||||
| Inbound rings extension | ${RING_EXT} |
|
||||
| Outbound scope | US/NANP only — \`_1NXXNXXXXX\` / \`_NXXNXXXXX\` patterns, no catch-all |
|
||||
| Concurrency cap | 3 simultaneous outbound calls (GROUP()/GROUP_COUNT() in the dialplan); a 4th gets a busy signal |
|
||||
|
||||
## How this survives Easy Asterisk's own regeneration
|
||||
|
||||
Easy Asterisk rewrites \`pjsip.conf\` and \`extensions.conf\` from its own
|
||||
internal state (device list, network settings) rather than treating them as
|
||||
hand-edited files. Trunk/dialplan config here lives in two files of its own,
|
||||
\`#include\`'d from the generated files:
|
||||
|
||||
- \`config/asterisk/voipms-trunk-pjsip.conf\` — the trunk's \`aor\`/\`identify\`/
|
||||
\`endpoint\` sections (IP-authenticated, no password stored).
|
||||
- \`config/asterisk/voipms-trunk-dialplan.conf\` — the NANP-only outbound
|
||||
routing + concurrency cap, and the \`[from-voipms]\` inbound context.
|
||||
|
||||
The \`#include\` lines themselves are patched into Easy Asterisk's *generator
|
||||
functions* (\`docker/entrypoint.sh\`, \`easy-asterisk.sh\`,
|
||||
\`easy-asterisk-v0.10.0.sh\`) so they get re-emitted every time those functions
|
||||
regenerate the config, instead of being wiped.
|
||||
|
||||
**Caveat:** if the base \`asterisk-digital-ocean\` service is ever updated
|
||||
independently (\`sudo ./setup.sh asterisk-digital-ocean\`, choosing "update in
|
||||
place" — that path re-copies fresh vendor files), this patch is wiped along
|
||||
with it. Re-run \`sudo ./setup.sh voipms-trunk\` afterward (update mode
|
||||
reapplies the patch and rewrites both include files with your existing
|
||||
settings, no re-prompting).
|
||||
|
||||
## Changing settings
|
||||
|
||||
Re-run \`sudo ./setup.sh voipms-trunk\` and choose "reinstall in place" —
|
||||
current settings are read back out of the include files above and reapplied.
|
||||
Choose "full install" instead to re-prompt for everything (new server, DID,
|
||||
or ring extension).
|
||||
|
||||
## Manual edits
|
||||
|
||||
Don't hand-edit \`voipms-trunk-pjsip.conf\` / \`voipms-trunk-dialplan.conf\`
|
||||
directly if you plan to re-run this installer later — it overwrites both
|
||||
files unconditionally. If you do need one-off manual changes (e.g. testing a
|
||||
second POP as a failover \`match=\` line), edit them directly and just don't
|
||||
re-run the installer until you're ready to lose those changes, or restart
|
||||
the container instead of running the installer to test them:
|
||||
|
||||
\`\`\`bash
|
||||
docker compose -f $EA_DIR/docker-compose.yml restart asterisk
|
||||
\`\`\`
|
||||
|
||||
## What the 3-call cap does and doesn't cover
|
||||
|
||||
It bounds *how many* outbound calls can be active at once — the first line
|
||||
of defense against a compromised extension opening dozens of parallel legs.
|
||||
It does not bound total spend by itself; that's still the prepaid-balance +
|
||||
auto-recharge-off layer on VoIP.ms's side. Both matter — see the toll-fraud
|
||||
section of \`docs/pstn-calling-voipms-plan.md\`.
|
||||
|
||||
## Verifying it's working
|
||||
|
||||
\`\`\`bash
|
||||
docker exec -it easy-asterisk-do asterisk -rx "pjsip show endpoint voipms-trunk"
|
||||
docker exec -it easy-asterisk-do asterisk -rx "dialplan show intercom"
|
||||
docker exec -it easy-asterisk-do asterisk -rx "dialplan show from-voipms"
|
||||
\`\`\`
|
||||
|
||||
A registered device should be able to dial a 10-digit or 11-digit US number
|
||||
and reach the VoIP.ms trunk; a call to \`${VOIPMS_DID}\` from outside should
|
||||
ring extension \`${RING_EXT}\`.
|
||||
MD
|
||||
chown "$ACTUAL_USER:$ACTUAL_USER" "$DOC_FILE" 2>/dev/null || true
|
||||
|
||||
# ── Apply ──────────────────────────────────────────────────────────────
|
||||
echo ""
|
||||
local RESTART_NOW=""
|
||||
prompt_yn "Restart the asterisk container now to apply the trunk config? (y/n):" "y" RESTART_NOW
|
||||
if [[ "$RESTART_NOW" =~ ^[Yy]$ ]]; then
|
||||
if ( cd "$EA_DIR" && docker compose restart asterisk ); then
|
||||
log_success "Asterisk restarted — trunk config applied."
|
||||
else
|
||||
log_warning "Restart failed — check: docker compose -f $EA_DIR/docker-compose.yml logs asterisk"
|
||||
fi
|
||||
else
|
||||
log_info "Apply later with: docker compose -f $EA_DIR/docker-compose.yml restart asterisk"
|
||||
fi
|
||||
|
||||
echo ""
|
||||
log_success "VoIP.ms trunk configured."
|
||||
echo " Server: $VOIPMS_SERVER ($VOIPMS_SERVER_IP)"
|
||||
echo " DID: $VOIPMS_DID"
|
||||
echo " Outbound: US/NANP only, max 3 concurrent calls (dialplan-enforced)"
|
||||
echo " Inbound rings: extension $RING_EXT"
|
||||
echo " Docs: $DOC_FILE"
|
||||
echo ""
|
||||
}
|
||||
@@ -89,6 +89,7 @@ is_installed() {
|
||||
sync-cc) [ -f "$ACTUAL_HOME/sync-cc/sync_cc.py" ] ;;
|
||||
sky-cam) [ -d "$ACTUAL_HOME/sky-cam/.git" ] ;;
|
||||
sky-cam-frigate) [ -d "$ACTUAL_HOME/sky-cam/.git" ] && [ -f "$ACTUAL_HOME/sky-cam/frigate-retime.sh" ] ;;
|
||||
voipms-trunk) [ -f "$DOCKER_DIR/asterisk-digital-ocean/config/asterisk/voipms-trunk-pjsip.conf" ] ;;
|
||||
ssh-config) false ;; # repeatable management tool, never shows [installed]
|
||||
*) [ -e "$DOCKER_DIR/$1" ] ;;
|
||||
esac
|
||||
|
||||
Reference in New Issue
Block a user