Claude
2ac2982e38
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).
390 lines
19 KiB
Bash
390 lines
19 KiB
Bash
#!/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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[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 $DIALPLAN_INCLUDE"
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echo "[DRY-RUN] Would offer 'update in place' instead of a fresh install if already configured"
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echo "[DRY-RUN] Would restart the asterisk container to apply"
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return 0
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fi
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if [[ ! -f "$EA_DIR/docker-compose.yml" ]]; then
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log_error "asterisk-digital-ocean isn't installed at $EA_DIR — install it first:"
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log_error " sudo ./setup.sh asterisk-digital-ocean"
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log_error "This service adds a VoIP.ms PSTN trunk on top of it; it doesn't stand alone."
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return 1
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fi
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log_info "Configuring VoIP.ms SIP trunk for asterisk-digital-ocean..."
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log_info "US-only outbound (NANP dialplan), max 3 concurrent outbound calls, IP-authenticated trunk."
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echo ""
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log_warning "Before continuing, on VoIP.ms's side you should already have: created an"
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log_warning "account, funded the \$15 minimum prepaid balance, turned OFF auto-recharge"
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log_warning "(Client Area -> Balance Management), ordered a DID with IP authentication"
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log_warning "pointed at this droplet's public IP, and picked a server/POP. Also restrict"
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log_warning "outbound routing to US/NANP on VoIP.ms's own side if it offers that — this"
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log_warning "dialplan is the second, independent layer, not a substitute for the first."
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log_warning "See docs/pstn-calling-voipms-plan.md for the full background."
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echo ""
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# ── Existing install? Offer update-in-place instead of a full reinstall ──
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if [[ -f "$PJSIP_INCLUDE" && -f "$DIALPLAN_INCLUDE" ]]; then
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log_info "Existing VoIP.ms trunk config found."
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local REINSTALL_MODE=""
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prompt_reinstall_mode REINSTALL_MODE
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case "$REINSTALL_MODE" in
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update)
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local _SERVER _SERVER_IP _DID _RING_EXT
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_SERVER="$(grep -m1 '^from_domain=' "$PJSIP_INCLUDE" | cut -d= -f2-)"
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_SERVER_IP="$(grep -m1 '^match=' "$PJSIP_INCLUDE" | cut -d= -f2-)"
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_DID="$(grep -m1 '^from_user=' "$PJSIP_INCLUDE" | cut -d= -f2-)"
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# Match only the inbound line (Dial(PJSIP/<digits>,20)) — the
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# outbound line matches Dial(PJSIP/${EXTEN}@voipms-trunk,60),
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# which starts with "$" right after PJSIP/, not a digit.
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_RING_EXT="$(grep -m1 -E 'Dial\(PJSIP/[0-9]+,20\)' "$DIALPLAN_INCLUDE" | sed -E 's/.*Dial\(PJSIP\/([0-9]+),20\).*/\1/')"
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if [[ -z "$_SERVER" || -z "$_DID" || -z "$_RING_EXT" ]]; then
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log_warning "Couldn't parse existing settings back out — falling back to a fresh install (every prompt below)."
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else
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_voipms_patch_vendor_files "$EA_DIR" || return 1
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_voipms_write_pjsip_include "$PJSIP_INCLUDE" "$_SERVER" "$_SERVER_IP" "$_DID"
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_voipms_write_dialplan_include "$DIALPLAN_INCLUDE" "$_DID" "$_RING_EXT"
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ensure_docker_dir_ownership "$ASTERISK_DIR"
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( cd "$EA_DIR" && docker compose restart asterisk ) \
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&& log_success "Updated — settings unchanged (server $_SERVER, DID $_DID, ring ext $_RING_EXT)." \
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|| log_warning "Restart failed — check: docker compose -f $EA_DIR/docker-compose.yml logs asterisk"
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return 0
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fi
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;;
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cancel)
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log_info "Leaving the existing VoIP.ms trunk config as-is."
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return 0
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;;
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fresh)
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log_info "Proceeding with a full fresh reinstall — every prompt below runs from scratch."
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;;
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esac
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fi
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# ── Prompts — VoIP.ms account details aren't scriptable, user sets these
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# up manually on VoIP.ms's own site first (see warning above) ───────────
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local VOIPMS_SERVER=""
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prompt_text "VoIP.ms server/POP hostname (e.g. atlanta2.voip.ms — pick the one closest to this droplet from VoIP.ms's server list):" "" VOIPMS_SERVER
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if [[ -z "$VOIPMS_SERVER" ]]; then
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log_error "A VoIP.ms server hostname is required — aborting."
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return 1
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fi
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local VOIPMS_SERVER_IP=""
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VOIPMS_SERVER_IP="$(getent ahostsv4 "$VOIPMS_SERVER" 2>/dev/null | awk '{print $1}' | head -1)"
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if [[ -z "$VOIPMS_SERVER_IP" ]]; then
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log_warning "Couldn't resolve $VOIPMS_SERVER — the identify section needs an IP to match inbound calls against."
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prompt_text "Enter its IP manually (check VoIP.ms's server list page):" "" VOIPMS_SERVER_IP
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if [[ -z "$VOIPMS_SERVER_IP" ]]; then
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log_error "No IP available — aborting."
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return 1
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fi
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else
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log_success "Resolved $VOIPMS_SERVER -> $VOIPMS_SERVER_IP"
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fi
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local VOIPMS_DID=""
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prompt_text "VoIP.ms DID (the 10-digit US phone number assigned to this trunk, digits only):" "" VOIPMS_DID
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if [[ ! "$VOIPMS_DID" =~ ^[0-9]{10}$ ]]; then
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log_error "That doesn't look like a 10-digit US number — aborting."
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return 1
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fi
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local RING_EXT=""
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prompt_text "Local extension to ring for inbound calls (must already exist in Easy Asterisk):" "" RING_EXT
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if [[ -z "$RING_EXT" ]]; then
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log_error "An extension number is required for inbound routing — aborting."
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return 1
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fi
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# ── Patch vendor generator functions, then write the include files ───────
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_voipms_patch_vendor_files "$EA_DIR" || return 1
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mkdir -p "$ASTERISK_DIR"
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_voipms_write_pjsip_include "$PJSIP_INCLUDE" "$VOIPMS_SERVER" "$VOIPMS_SERVER_IP" "$VOIPMS_DID"
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_voipms_write_dialplan_include "$DIALPLAN_INCLUDE" "$VOIPMS_DID" "$RING_EXT"
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ensure_docker_dir_ownership "$ASTERISK_DIR"
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chmod 644 "$PJSIP_INCLUDE" "$DIALPLAN_INCLUDE"
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# No new firewall rules: asterisk-digital-ocean.sh already opens SIP
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# (5060/5061) and RTP (10000-20000) to the internet, and VoIP.ms's source
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# IPs vary by POP/redundancy, so there's no single IP to scope this to
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# even if narrowing it were otherwise worthwhile.
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# ── Docs (separate file — asterisk-digital-ocean already owns README.md
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# in this same directory via write_readme, so don't overwrite it) ───────
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local DOC_FILE="$EA_DIR/README-voipms-trunk.md"
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cat > "$DOC_FILE" << MD
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# VoIP.ms SIP trunk (add-on to asterisk-digital-ocean)
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US-only outbound PSTN calling over a VoIP.ms trunk, capped at 3 concurrent
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outbound calls, plus inbound routing to one extension. See
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\`docs/pstn-calling-voipms-plan.md\` in the repo for the full design
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background, cost estimate, and toll-fraud reasoning.
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## Current settings
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| Setting | Value |
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|---|---|
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| VoIP.ms server/POP | ${VOIPMS_SERVER} (${VOIPMS_SERVER_IP}) |
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| DID | ${VOIPMS_DID} |
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| Inbound rings extension | ${RING_EXT} |
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| Outbound scope | US/NANP only — \`_1NXXNXXXXX\` / \`_NXXNXXXXX\` patterns, no catch-all |
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| Concurrency cap | 3 simultaneous outbound calls (GROUP()/GROUP_COUNT() in the dialplan); a 4th gets a busy signal |
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## How this survives Easy Asterisk's own regeneration
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Easy Asterisk rewrites \`pjsip.conf\` and \`extensions.conf\` from its own
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internal state (device list, network settings) rather than treating them as
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hand-edited files. Trunk/dialplan config here lives in two files of its own,
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\`#include\`'d from the generated files:
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- \`config/asterisk/voipms-trunk-pjsip.conf\` — the trunk's \`aor\`/\`identify\`/
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\`endpoint\` sections (IP-authenticated, no password stored).
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- \`config/asterisk/voipms-trunk-dialplan.conf\` — the NANP-only outbound
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routing + concurrency cap, and the \`[from-voipms]\` inbound context.
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The \`#include\` lines themselves are patched into Easy Asterisk's *generator
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functions* (\`docker/entrypoint.sh\`, \`easy-asterisk.sh\`,
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\`easy-asterisk-v0.10.0.sh\`) so they get re-emitted every time those functions
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regenerate the config, instead of being wiped.
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**Caveat:** if the base \`asterisk-digital-ocean\` service is ever updated
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independently (\`sudo ./setup.sh asterisk-digital-ocean\`, choosing "update in
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place" — that path re-copies fresh vendor files), this patch is wiped along
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with it. Re-run \`sudo ./setup.sh voipms-trunk\` afterward (update mode
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reapplies the patch and rewrites both include files with your existing
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settings, no re-prompting).
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## Changing settings
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Re-run \`sudo ./setup.sh voipms-trunk\` and choose "reinstall in place" —
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current settings are read back out of the include files above and reapplied.
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Choose "full install" instead to re-prompt for everything (new server, DID,
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or ring extension).
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## Manual edits
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Don't hand-edit \`voipms-trunk-pjsip.conf\` / \`voipms-trunk-dialplan.conf\`
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directly if you plan to re-run this installer later — it overwrites both
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files unconditionally. If you do need one-off manual changes (e.g. testing a
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second POP as a failover \`match=\` line), edit them directly and just don't
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re-run the installer until you're ready to lose those changes, or restart
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the container instead of running the installer to test them:
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\`\`\`bash
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docker compose -f $EA_DIR/docker-compose.yml restart asterisk
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\`\`\`
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## What the 3-call cap does and doesn't cover
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It bounds *how many* outbound calls can be active at once — the first line
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of defense against a compromised extension opening dozens of parallel legs.
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It does not bound total spend by itself; that's still the prepaid-balance +
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auto-recharge-off layer on VoIP.ms's side. Both matter — see the toll-fraud
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section of \`docs/pstn-calling-voipms-plan.md\`.
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## Verifying it's working
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\`\`\`bash
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docker exec -it easy-asterisk-do asterisk -rx "pjsip show endpoint voipms-trunk"
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docker exec -it easy-asterisk-do asterisk -rx "dialplan show intercom"
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docker exec -it easy-asterisk-do asterisk -rx "dialplan show from-voipms"
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\`\`\`
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A registered device should be able to dial a 10-digit or 11-digit US number
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and reach the VoIP.ms trunk; a call to \`${VOIPMS_DID}\` from outside should
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ring extension \`${RING_EXT}\`.
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MD
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chown "$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 ""
|
|
}
|