Spec 12 S2: openid-client 6.8 (decision 0004). Discovery, PKCE + state + nonce, ID token
signatures checked, userinfo always read. Users connect their OpenID account from My account
and sign in either way; no accounts are made; a user without a password is matched once by
verified email. auth.mode mixed and oidc (admins keep a password way in), role from adminGroups,
allowedGroups, signing out at the provider. Settings -> Sign-in with Test. Tested against a fake
provider with RS256 and ES256 keys.
Spec 12 S1: migration 0012 (password_hash nullable, user_identities, sessions.method and the
OpenID token columns), auth.mode (only "password" until S2), GET /api/auth/methods, login body
{login, password} (email still accepted), signIn on users and /api/auth/me, a Signs in with
column on Users. The last active admin with a password can't be deleted, disabled or demoted.
Owner decisions 2026-10-25: a user connects their OpenID account and signs in with either it or
their password; the admin picks passwords only, OpenID only (admins keep a password way in), or
mixed. No automatic accounts. OIDC sessions re-checked every 15 min with the refresh token;
signing out also signs out at the provider; password sign-ins keep two-step. LDAP is one more way
to check a password. Decision 0004: openid-client and ldapts. Tasks S1-S5.
Spec 11 §6.2: a phone's directory is the extensions, its owner's own contacts (accepted shares
included), then the company's, in both modes. In Contrack mode, contacts from before Contrack
are offered for copying (contrack.toCopy, POST …/copy-to-contrack: one Contrack bulk import,
skipping what's already there; the builtin rows stay). Tested against a fake and a real
Contrack 2.0.2.
Spec 11 §4.4: the contrackLog job puts each connected person's calls with their Contrack
contacts on that contact's timeline once (contrack_call_log, migration 0011), from when they
connected. With summaries on (My settings → Contacts, needs call summaries), the escaped summary
and action items go in, and the first action item becomes a follow-up due the next day. Waits up
to 30 min for a summary; Contrack down is retried, refusals noted once. Tested against a fake and
a real Contrack 2.0.2.
Spec 11 §6.1: sp-sub-dial-user asks /api/internal/caller-name (func_curl, 2 s at most; this
machine only, with an APP_SECRET-derived key, not logged) for the ringing user's name for the
caller, and sets it on each ringing leg (pre-dial handler, CONNECTEDLINE), never on the call, so
forwards and other members don't see someone else's contacts. Ring group/queue/number prefixes
kept. Contrack lookups never wait (cache, contactsWarm every 5 min). Proven on real Asterisk
with baresip: apps/api/test/asterisk/cnam-test.ts.
Spec 11 §4: contacts.provider and contacts.contrack (local address only, company token
write-only), each person's own token (users.contrack_token_enc, migration 0010) in My settings,
books read from Contrack with a 5-minute cache, Add to contacts and accepted shares created in
Contrack, the company account on desk phones. Setup wizard step 8 Contacts. deploy/contrack
(2.0.2, pinned), Caddy/nginx blocks, decision 0003, admin guide Contacts. Tested against a fake
and a real Contrack 2.0.2 (contrack-live.test.ts).
C1 (spec 11): /api/me/contacts and /api/contacts (admin), our own vCard 2.1/3.0/4.0 and CSV
readers with a preview, contact_shares (migration 0009) for offers, and contact names in call
history, voicemail, Home, the softphone and the call report email. Admin → Contacts.
Owner decisions 2026-10-23: vCard import for admins and users; desk-phone caller name while
ringing is its own task (C4); the company is one Contrack account shown to everyone, each user
has their own; sharing offers a copy the colleague accepts or denies; imports skip duplicates.
My settings → Call report: each user's calls of the last hour, day (at a time) or week (on a
day at a time), company time zone: numbers, status, talk time, recorded, and optionally each
recorded call's summary (10 §3). No calls, no email. Job callReports every minute, 5 min after
the period ends; periods computed in PostgreSQL (DST-safe). Migration 0008_call_reports.
Owner asked for total minutes, call time and missed calls in one place, and the money left on
voip.ms. Reports now totals answered calls, talk time and outside minutes (in from arrival, out
from the answer: roughly what the provider bills), charts outside minutes per day, and shows each
voip.ms trunk's balance (getBalance; verify with a real account).
Owner: the first upload always played first. A class is now an Asterisk playlist in the order
set on Sounds & music (move up/down), new uploads last, with a Shuffle option (sort = random).
moh-test.ts plays both in a real Asterisk and reads the file order from its debug log.
Owner wanted the first user on 1. Numbering ranges now choose the Next free suggestion
(Settings → Numbering), the setup wizard takes a number per person, and a one-digit number
can't also be a menu key of a digital receptionist with direct dialing. Dialplan patterns that
take an internal number match one digit (_X!); golden case short-numbers proves it in Asterisk.
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).
Docker only, never bare metal (owner, 2026-10-21). A new version with pending migrations backs
up first (database, voicemail, sounds, MOH) and stops if it can't (SKIP_BACKUP_BEFORE_MIGRATE
to override). update.sh: checks, changelog, backup (node dist/backup-cli.js), log compaction,
build while the old version runs, restart, self-check, and rollback with restore. Container
logs capped in compose. Admin guide: Updating, for compose and Portainer.
- Asterisk doesn't load app_system, func_shell, app_disa, res_agi, app_externalivr.
- Turning two-step sign-in on asks for the password again.
- *97/*98: one voicemail login at a time per phone, 10 an hour (sp-sub-vmlimit).
- /prov: an unknown MAC gets the same 401 as a known one.
- The login limiter sweeps keys past every window.
Also the v1 checklist with the owner's results from 2026-10-21.
Its Contact holds pn-prid=...:voip&...:remote, and PJSIP_DIAL_CONTACTS passes that '&' through,
so Dial rejected the whole device list and the call dropped for every device of the user.
sp-sub-contacts walks the AOR's contacts, drops pn-* params and FILTERs the rest. call-test
registers a phone with such a Contact and checks it is rung.
.github/workflows/release.yml runs on Gitea and on the GitHub mirror (release: published only)
and pushes selfphoneng-api and selfphoneng-asterisk to that forge's registry: ghcr.io with the
job token, the Gitea registry with secrets REGISTRY_USER + REGISTRY_TOKEN (Gitea's job token
can't push packages). Tags: the version, plus <major>.<minor> and latest unless it's a
pre-release. It stops unless the tag is v + package.json's version, and checks the images
before pushing.
Run locally with act_runner exec against a throwaway registry: v0.0.1 pushed 0.0.1, 0.0 and
latest for both images (VERSION 0.0.1, OCI labels, source archive selfphoneng-0.0.1/);
v9.9.9 failed at the version check.
Version 0.0.1 in all package.json files, CHANGELOG.md started, release steps in 08 §3.1.
Task: none (owner request, 2026-10-20)
Owner's decision after the M11-T8 review: nothing capped how many calls a stolen password could
run at once. Settings → Security: calls.maxPerExtension (3) and calls.maxTotal (10), 0 = no
limit. Each sp-route-* counts the caller's extension (or whose forward it is: SP_LIMITEXT from
sp-sub-forward) with GROUP(spext) and every outside call with GROUP(spout); over a limit the
caller hears "all circuits are busy". 911 is never limited.
trunk-test.ts on a real Asterisk, limits 1 and 2: a 2nd call from 101's phone and a 3rd in total
are refused (with both 0 the same check fails). call-limits.test.ts covers the 0 combinations.
Also fixes the Security tab's hint: browser phones still work from anywhere.
Task: M11-T8
Under a full parallel run the event-log and webhook inserts took longer than the sleep, so
2-3 tests failed on most runs (also before M11-T8). They poll for the rows now.
Task: M11-T8
The 08 §1 threat table checked line by line against the code, plus a review by a separate agent
that hadn't written it. Findings were reproduced with a failing test before each fix:
- Critical: "011…,4" turned the route's Goto into Goto(sp-route-…,011…,4,1), starting past the
class check, so any extension could call abroad. FILTER on the dialed number (also stops "&"
adding a Dial target). trunk-test.ts proves it.
- High: -web endpoints (desk phone's password, any transport) skipped the phone ACL; now
127.0.0.1 only when it's on. call-test.ts: UDP refused, WebSocket from 127.0.0.1 registers.
- IVR direct dial no longer reaches paging groups or parked calls (sp-ivr-direct).
- Trunks identify_by = ip: a From of trunk-<id> from anywhere no longer rings phones.
- .dockerignore is an allowlist, so the public source archive can't pick up stray files.
- CSRF check on the matched route (/%61pi/… got around it); Linphone tokens masked in logs;
/api/live checks Origin and re-checks the session; *45/** filter digits; pg_dump password in
the environment; more webhook ranges blocked.
The review and what's left for the owner are in 08 §6.
Task: M11-T8
System page laid out after 3CX's admin dashboard: a System information panel (version,
host name, company, public IP, LAN IPs, ports, telephony, restart needed, copy buttons,
how to restart) and an Event log panel (system_events, 90 days: trunk down/up, Asterisk
down/up, restart needed, config failed, 911, public IP changed, backups, disk) with Load
more and a CSV download.
About (/about, no sign-in, linked from the login page, My phone and System): AGPL-3.0
notice, credits, the repository link, and "Code v<version>", which downloads
selfphone.tar.gz: the api image now tars its own build context (no .env, data,
node_modules, build output or .git) before installing anything.
Task: M11-T7
Nothing has shipped, so there is nothing to update (the script itself was removed in
a152508). The v1.0 definition of done now requires update.sh written, reviewed and
tested on a real server; the admin guide says there are no releases yet.
Task: M11-T7
System shows whether telephony runs and needs a restart, and how to restart it
(docker compose restart asterisk); the dashboard's restart warning, the Network hint and
the public IP email point there. The user drawer's Calling tab gets the Record calls
select the spec listed (recordMode, already in the dialplan). e2e m11-system.mjs.
Task: M11-T7
docs/admin-guide/: install and the setup wizard, voip.ms with screenshots, NAT and SIP
ALG, firewall (ufw, nftables limited to the voip.ms POP), fail2ban and CrowdSec, E911
duties, recording consent, troubleshooting (08 §3.3), updating, backup and restore,
monitoring and log rotation. scripts/update.sh as 08 §3.1 describes. The e2e checks
fail2ban's and CrowdSec's stock Asterisk rules against a real failed sign-in, takes the
screenshots and checks every link and anchor.
Task: M11-T7
iamdoubz
iamdoubz