Each release attaches selfphoneng-probe-linux-x64 and -arm64 (Node's single executable
applications: esbuild bundle + postject into the official Node binary, checked against
nodejs.org's SHASUMS256.txt), the bundle as selfphoneng-probe.cjs (Node 22+) and SHA256SUMS.
Built in binary.Dockerfile with BuildKit's local output, so Gitea's runner can build it too.
deploy/probe/selfphoneng-probe.service runs it under systemd, sandboxed.
The server checks itself (public IP, DNS, local networks, phone ACL, SIP transports, free audio
ports) and says how to fix each problem. A quick or complete network test asks the probe
(apps/probe, own image) to test this server from outside: HTTP/HTTPS, SIP over UDP/TCP/TLS, the
browser phone's /ws, sampled audio ports. The probe only tests the address that asks.
The audio port range comes from RTP_PORT_START/END (default 10000-20000).
GET /api/extensions/:id/softphone-qr (admin) and /api/me/softphone-qr
return {url, qr, expiresAt}: the link Linphone fetches and its QR as a
PNG data URL (qrcode), valid 24 h. Both are audited like revealing the
SIP password. JSON rather than a bare PNG, so the page can show the
link for desktop Linphone too.
The token is stateless per 07 §5: base64url(extensionId.expiry.hmac),
HMAC-SHA256 with APP_SECRET, checked timing-safe. /prov/linphone/
<token>.xml matches it before the other file names are lower-cased
(the token is case-sensitive); a bad or expired token is a 404 and a
counted failure. auth_info keys checked against liblinphone's config
reader; proxy keys stay (verify).
Web: a "Set up a phone app" section on My settings and in the admin
Credentials tab: Linphone with the QR, MicroSIP/Baresip by hand, and
the plain warning about mobile apps without push. e2e m10-t3: as a
user, the QR's link gives the XML and a baresip phone registers with
only what it holds.
Task: M10-T3
GET /api/me/webrtc returns the <ext>-web identity, WebSocket URL
(WEBRTC_WS_URL overrides wss://PUBLIC_HOSTNAME/ws) and STUN/TURN servers.
SoftphoneEngine wraps SIP.js 0.21.2: register with backoff reconnect,
call, answer, decline, hang up, remote audio. A minimal dock on /me
enables it (asks for the microphone on the click) and dials/answers;
M5-T2 grows it into the full panel.
Generator: ice_host_candidates keeps the LAN address
(include_local_address), so LAN browsers don't need hairpin NAT, and -web
endpoints set from_domain = selfphoneng.invalid because SIP.js drops
INVITEs whose From host starts with a digit (Docker/VPS hostnames).
e2e m5-t1 (Docker): browser registers as 170-web and has two-way audio
with a baresip phone both ways. The LAN and hotspot check on the owner's
server is still open (roadmap note).
Task: M5-T1
Dialplan: sp-emergency-dial (911, 9911) first in sp-from-ext;
sp-emergency has no class, channel-limit or DND check, dials every
enabled trunk in name order with the E911 caller ID, and raises
UserEvent SpEmergency. Static sp-emergency-alert (_X! so "0" matches)
for the ring-out.
api: SpEmergency -> emergency_events row, live emergency.call (admins
and notify users only), email via nodemailer (SMTP from settings) and
an Originate to every notify extension, each outcome recorded. Test
button runs the same with TEST in front, records and dials nothing.
GET/PATCH /api/settings (smtp password write-only, references
checked, read-only system keys), SMTP test, /api/system/warnings.
Fix: clearing a nullable setting stored SQL NULL (now JSON null).
UI: Settings page (Email, Emergency with the 06 §6 notice), user
Emergency tab, dashboard warnings banner, red 911 banner.
Verified: generator/emergency.test.ts; trunk-test.ts dials 911 from a
caller with no class to the fake provider (E911 caller ID in RPID,
SpEmergency on AMI, alert plays on baresip 101); scripts/e2e/m3-t4.mjs
(SMTP sink receives the test email, 101 rings, banner shows).
nodemailer pinned to 10.0.11 (newest release over two weeks old).
Task: M3-T4
LiveState loads extension states and registered devices from Asterisk on
every AMI connect and follows ExtensionStatus/ContactStatus; the hub sends
a snapshot then deltas batched every 250 ms (devices for admins only),
pings every 30 s and closes unauthenticated sockets with 4401. The web UI
keeps one reconnecting socket in a zustand store, shows a Reconnecting
banner, a Dashboard (online, on a call, telephony engine) and status dots.
Fix: Asterisk 24 has no 'pjsip reload'; reload with
'module reload res_pjsip.so'. AMI errors now carry the CLI output.
Milestone check, automated: call-test.ts registers two baresip phones and
holds a 30 s call with RTP both ways (pnpm test:asterisk); the recorded
session drives the live-state tests. scripts/e2e/m2-live.mjs runs api +
Asterisk + phones + browser on a private Docker network: a new extension
is live in Asterisk in ~1.3 s, the dashboard shows 2/2 online and red dots
during the call.
Task: M2-T6
Login, forced password change, admin layout with every 05 §2 page in the
sidebar (unbuilt ones name their milestone), Users table/cards + drawer
(General tab), one-time temporary passwords, reset password. The last
active admin can't be demoted, deactivated or deleted. Audit diffs only
mask secret strings. scripts/e2e has a dependency-free CDP browser check
for this task's acceptance flow.
Task: M1-T4
argon2id passwords with a min length and a common-password list (SecLists,
MIT). Login failures are limited per IP and per email; must_change_password
blocks everything outside /api/auth/*. The first admin is created from env
on startup. pnpm dev derives DATABASE_URL from .env.
Task: M1-T3
Migration 0000 enables citext; 0001 creates all 39 tables. DB tests use a
throwaway database per file (TEST_DATABASE_URL, derived from .env in dev,
a postgres service in CI). Decision 0002: pg driver and @fastify/cookie.
Task: M1-T1
iamdoubz