Fold asterisk-digital-ocean into asterisk with droplet auto-detection
services/asterisk-digital-ocean.sh was a near-verbatim copy of services/asterisk.sh — same vendor refresh, compose template, messaging dialplan, presence alerts, UFW rules and dashboard/trunk chaining, with the helper functions renamed _asterisk_do_*. Two copies meant every fix had to land twice, and several never did. There is now one `asterisk` service. It reads the DigitalOcean metadata service and asks either way (so a droplet with metadata blocked, or another provider's public VM, can still opt in), then gates the genuinely droplet-specific behaviour on that one answer: swapfile for low-RAM plans, public-FQDN-only setup with no LAN/VLAN prompts, a Caddy site block pinned to that FQDN, the remote-Authelia option, and the doctl Cloud Firewall. Two things that were droplet-only for no real reason now apply everywhere: the entrypoint patch that writes security-level events to logs/full, and the logrotate config for that file. Without them the Security Dashboard's Security Log tab and CrowdSec's Asterisk acquisition were silently empty on every home/LAN install; crowdsec.sh now detects either install directory. Existing droplets are left alone: an install at ~/docker/asterisk-digital-ocean keeps its directory and easy-asterisk-do container names, since its Caddyfile block, UFW rules, Cloud Firewall, CrowdSec acquisition and PSTN trunk all name those exactly. New installs use ~/docker/asterisk / easy-asterisk. `sudo ./setup.sh asterisk-digital-ocean` still works via a new SERVICE_ALIAS map in setup.sh, without a second menu entry. Co-Authored-By: Claude Opus 5 <noreply@anthropic.com> Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
This commit is contained in:
Claude
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@@ -1,6 +1,6 @@
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#!/bin/bash
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# services/pstn-trunk.sh — SIP PSTN trunk add-on for asterisk-digital-ocean
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# (or the home/LAN asterisk install): US-only outbound (NANP dialplan
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# services/pstn-trunk.sh — SIP PSTN trunk add-on for services/asterisk.sh:
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# US-only outbound (NANP dialplan
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# restriction), independent outbound/inbound concurrent-call caps, a 3-tier
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# permission model per extension (internal-only / restricted to pre-approved
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# numbers / full US calling), a configurable inbound ring-group,
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@@ -15,19 +15,19 @@
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# works the same way. VoIP.ms and Anveo Direct are both confirmed working;
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# see docs/pstn-calling-voipms-plan.md for the design/cost background.
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#
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# Requires an existing services/asterisk-digital-ocean.sh OR services/asterisk.sh
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# install — this adds a PSTN trunk on top of one of them and does not stand
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# alone. Permission tiers AND concurrency caps are managed live (no restart
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# Requires an existing services/asterisk.sh install (either directory layout —
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# ~/docker/asterisk, or ~/docker/asterisk-digital-ocean on a box set up before
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# the droplet edition was merged back in) — this adds a PSTN trunk on top and
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# does not stand alone. Permission tiers AND concurrency caps are managed live (no restart
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# needed) via pstn-permissions.conf / pstn-limits.conf — editable by hand, or
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# from services/security-dashboard.sh's "PSTN Trunk" tab if that's installed.
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# from services/security-dashboard.sh's Extensions tab if that's installed.
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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 pstn-trunk homelab "SIP PSTN trunk for asterisk-digital-ocean/asterisk — US-only, per-extension permission tiers, spend/volume alerts (any IP-authenticated provider — VoIP.ms and Anveo Direct both confirmed)"
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register_service pstn-trunk homelab "SIP PSTN trunk for asterisk — US-only, per-extension permission tiers, spend/volume alerts (any IP-authenticated provider — VoIP.ms and Anveo Direct both confirmed)"
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# ── Surviving Easy Asterisk's regeneration ──────────────────────────────────
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# Easy Asterisk (the vendor project asterisk-digital-ocean.sh/asterisk.sh
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# build on) fully OVERWRITES both pjsip.conf and extensions.conf from its own
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# Easy Asterisk (the vendor project services/asterisk.sh builds on) fully OVERWRITES both pjsip.conf and extensions.conf from its own
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# 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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@@ -41,9 +41,9 @@ register_service pstn-trunk homelab "SIP PSTN trunk for asterisk-digital-ocean/a
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# the #include itself survive regeneration too, _pstn_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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# in _asterisk_refresh_vendor_files (see services/asterisk.sh).
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#
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# Caveat: if the base asterisk-digital-ocean/asterisk install is later
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# Caveat: if the base asterisk install is later
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# refreshed ("update in place", which re-copies fresh vendor files)
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# independently of this service, the patch is wiped along with it and needs
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# reapplying — run this service again (fresh or update mode both reapply it)
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@@ -459,7 +459,7 @@ EOF
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# zero of the outbound NANP patterns either, and `dialplan show
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# from-pstn-trunk` reported the context didn't exist at all, with no
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# warning or error anywhere (config log, full log, or the reload command's
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# own output) pointing at why. Meanwhile services/asterisk-digital-ocean.sh's
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# own output) pointing at why. Meanwhile services/asterisk.sh's
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# messaging-dialplan.conf — #include'd via the exact same mechanism, right
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# after [intercom] in the same extensions.conf — loaded fine every time.
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# The one structural difference: messaging-dialplan.conf's first real line
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@@ -1608,7 +1608,7 @@ install_pstn-trunk() {
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[[ "$ASTERISK_KIND" == "asterisk-digital-ocean" ]] && CONTAINER_NAME="easy-asterisk-do"
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if [ "$DRY_RUN" = true ]; then
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echo "[DRY-RUN] Would require an existing asterisk-digital-ocean OR asterisk (LAN) install"
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echo "[DRY-RUN] Would require an existing asterisk install (droplet or home/LAN)"
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echo "[DRY-RUN] Would prompt for: known-provider quick-pick (Anveo Direct runs a full 5-step"
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echo "[DRY-RUN] interactive portal walkthrough — account/funding, DID ordering, both trunk"
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echo "[DRY-RUN] objects, confirmed rate — pausing for Enter between each; VoIP.ms pre-fills known"
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@@ -1640,10 +1640,12 @@ install_pstn-trunk() {
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fi
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if [[ -z "$EA_DIR" ]]; then
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log_error "Neither asterisk-digital-ocean nor asterisk (LAN) is installed — install one first:"
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log_error " sudo ./setup.sh asterisk-digital-ocean (recommended — public droplet, static IP)"
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log_error " sudo ./setup.sh asterisk (home/LAN — see the static-IP caveat below)"
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log_error "This service adds a PSTN trunk on top of one of them; it doesn't stand alone."
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log_error "Asterisk is not installed — install it first:"
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log_error " sudo ./setup.sh asterisk"
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log_error "A public droplet (which that installer detects and tunes for) is the"
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log_error "recommended host, since IP authentication wants a static IP — see the"
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log_error "caveat below for what that means on a home/LAN box."
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log_error "This service adds a PSTN trunk on top of it; it doesn't stand alone."
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return 1
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fi
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@@ -1652,7 +1654,7 @@ install_pstn-trunk() {
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log_warning "Using the home/LAN asterisk install. IP authentication needs a STABLE public IP —"
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log_warning "if this box is behind a dynamic home IP, your provider's IP allow-list goes stale"
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log_warning "whenever your ISP rotates it, breaking calls until you update it there yourself."
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log_warning "A static IP from your ISP avoids that; asterisk-digital-ocean sidesteps it entirely."
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log_warning "A static IP from your ISP avoids that; a cloud droplet sidesteps it entirely."
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fi
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log_info "Configuring a SIP PSTN trunk for $ASTERISK_KIND (any IP-authenticated provider —"
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@@ -2213,12 +2215,11 @@ Asterisk's native SIP \`MESSAGE\` support (extension-to-extension texting —
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no carrier SMS, no PSTN, no cost) is gated by a \`messaging=yes\` flag per
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extension in \`pstn-permissions.conf\`, independent of the PSTN calling
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tiers above — off by default, same "opt in" posture. Live-editable any
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time via the Security Dashboard's "PSTN Trunk" tab, in its own
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always-available "Internal SIP messaging" card — no dependency on this
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trunk (or any PSTN trunk at all) being installed.
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time via the Security Dashboard's Extensions tab, in the Messaging column of
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its always-available extensions table — no dependency on this trunk (or any
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PSTN trunk at all) being installed.
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Actually enforced, not just a flag — \`services/asterisk-digital-ocean.sh\`
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(and \`services/asterisk.sh\` for the LAN edition) routes messages through a
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Actually enforced, not just a flag — \`services/asterisk.sh\` routes messages through a
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dedicated \`[sip-messaging]\` dialplan context (separate from \`[intercom]\`'s
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own per-device call routing, so there's no collision risk) and checks this
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same flag via \`AST_CONFIG()\` before delivering. One caveat still flagged
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