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
GET /api/reports/summary and /api/reports/queues (company-time-zone days, last 30 by
default, a year at most) and the Reports page: totals, stacked bars per day, missed per
day, a weekday x hour heatmap and the queue table. Plain divs, no chart library.
Task: M11-T6
API tokens (Settings -> API tokens): Authorization: Bearer sp_<40 base62>,
stored as sha256, shown once. A token acts as its admin with their current
role and stops when they're disabled; Bearer requests skip the CSRF header
and never fall back to the cookie. A token can't manage tokens.
Webhooks (Settings -> Webhooks): call.started/answered/ended,
voicemail.new, emergency.call, trunk/extension registration changes, from
the live state; queued per webhook, sent every 10 s with
X-SelfPhoneNG-Signature (HMAC of timestamp.body), retried after 1 m, 5 m,
30 m, 2 h, 12 h, then failed. Private/loopback targets are refused in the
connection's own DNS lookup (no rebinding), unless allowed. Send test and
a delivery log per webhook.
The e2e found voicemail.new missing for mailboxes without a user: the
voicemail scan now queues it for every mailbox. vmpin.test.ts gets 180 s:
Git Bash forks make it ~85 s on Windows (the earlier flaky gate run).
Task: M11-T5
apps/api/src/jobs.ts takes over the once-a-minute timer from index.ts
(no croner: every schedule in 04 §8 is whole minutes). Each job has a
running flag; errors are logged.
- Retention (at start and when the date changes, once AMI is up):
recordings, voicemail past voicemail.retentionDays (VoicemailRemove, so
Asterisk renumbers and updates MWI), expired sessions, webhook
deliveries older than 30 days.
- publicIpWatch (every 5 min, Settings -> Network): a changed IP is saved,
audited, applied (restart required) and emailed to admins.
- trunkWatch: a registering trunk rejected/unregistered for 5 minutes ->
one email per outage.
- diskWatch (hourly): under 10% free -> one email a day.
- A failed daily backup emails admins too.
Settings gets a Network tab (public IP, Detect, re-check) and voicemail
retention on the Recording tab.
Task: M11-T4
A backup is data/backups/selfphoneng-YYYYMMDD-HHMM.tar.gz (mode 0600):
pg_dump, voicemail, custom sounds, music on hold, recordings if asked,
and a manifest. No generated Asterisk config: it holds plaintext
passwords and comes back from the database. The api makes one a day after
backup.time (once-a-minute check, one try a day) and the dashboard warns
when today's is missing. Settings -> Backups sets the schedule, lists them
for download and has Back up now.
scripts/restore.sh <file> refuses a backup made under another APP_SECRET
(the manifest carries a fingerprint, not the secret), asks for yes, then
replaces the database and the folders the backup has.
Tested on two throwaway servers (docker:dind running the real images):
back up on one, restore on the other. That found the api image crashing
on start since M1: tsc doesn't copy auth/common-passwords.txt into dist.
The Dockerfile copies it now and CI loads the app inside the image.
Task: M11-T3
The certificate lives in data/tls (fullchain.pem + privkey.pem), mounted
read-only into asterisk and the api. Asterisk quietly drops transport-tls
when the files are missing, so the api won't turn TLS on unless they load
and match, shows the certificate on Settings -> Security, and warns on the
dashboard when it expires within 14 days. scripts/tls-cert.sh copies it
from the bundled Caddy (or given paths) and reloads res_pjsip: Asterisk 24
re-reads a changed certificate on reload, calls stay up.
transport-tls checks the provider's certificate (verify_server). Asterisk
24 checks a qualify's certificate against the IP instead of the name, which
fails on voip.ms, so TLS trunks don't qualify; REGISTER and INVITE verify
the name fine.
Task: M11-T2
auth/totp.ts per 08 §2.4: RFC 6238 (HMAC-SHA1, 30 s, 6 digits, ±1
step) with node:crypto only, checked against the RFC's test vectors.
Migration 0003: users.totp_recovery_hashes and totp_last_step.
Admins turn it on from My account (the header's "Account"): scan the
QR or type the key, confirm a code, get 10 recovery codes once. Login
asks for the code only after the password is right (401 totp_required,
no session); a wrong code is a failed login (rate limit, audit). A
step signs in once (totp_last_step updated atomically) and a recovery
code once (array_remove guarded by any()). Off and new codes need the
password; another admin can turn it off for a lost phone, which signs
that user out. Users list shows it (twoStep).
Also, found by the full e2e chain on one shared database:
- Linphone tokens: only the exact token issued is accepted. The last
base64 character can carry ignored bits, so another spelling decoded
to the same signed token.
- Test scripts no longer assume a fresh system: m3-t3 picks Dan in the
test box, m8-check dials whatever slot the park took (findslot =
next), m10-t4 uses the admin's existing extension, m11-t1 clears the
public IP on its own first (local nets are in the transports while
one is set, and transports change only on a restart).
e2e m11-t1-totp; the full chain M1-M11 (33 scripts) passes.
Task: M11-T1
Headers (08 §2.4) on every response: CSP (connect-src also allows
WEBRTC_WS_URL's origin when set), nosniff, Referrer-Policy,
Permissions-Policy. The dark-mode bootstrap moved from an inline
<script> to /theme.js so script-src stays 'self'.
Phone ACL (08 §2.2): with sip.restrictPhonesToLocalNets, every desk
extension endpoint gets deny 0.0.0.0/0 and permit for each local and
extra net. The spec's snippet was wrong: contact_acl names acl.conf
sections, a pjsip type=acl is the global ACL (it would filter trunks),
and Contact-header rules let a NATed remote phone through. Browser
endpoints come through the reverse proxy and aren't filtered. Golden
case phone-acl. Settings → Security holds it, with the provisioning
switches.
PIN rules for PINs changed from the phone's voicemail menu, which went
around the api: externpasscheck = sp-vm-pincheck (in the Asterisk
image, same rules as isTrivialPin, a unit test keeps them in step) and
minpassword = 4. Accepted PINs reach the api through AMI
VoicemailPasswordChange and are stored; before, the next config write
silently put the old PIN back.
e2e m11-t1: no CSP violations across admin pages; Asterisk refuses a
REGISTER ("Not match Endpoint ACL") until the network is listed; from
the phone, 1111 is refused by the check script and 7391 is stored and
survives a later config write.
Task: M11-T1
POST /api/live/calls/:linkedid/spy {mode} per 04 §7.3: rings the
admin's desk phone (409 if it isn't connected, 400 without an
extension) and, when they answer, runs ChanSpy on the call's first
user leg (never a trunk or Local channel, never the admin's own),
which for an inbound or queue call is the agent. Options qE / qEw /
qEB: no beep or name, the spy leg ends with the call, whisper to that
user only, or barge (both sides hear). Audited as call_spy.
Dashboard: Listen / Whisper / Barge on talking calls with a user on
them, and "your desk phone is ringing" once sent. The spy leg shows as
its own call; its Hang up stops listening.
e2e m10-t4: 380 calls 381, the admin's 382 runs ChanSpy with each
mode's options, and the barge leg ends with the call. The roadmap
records that M10's milestone check (real Yealink/Grandstream, Linphone
scanning the QR) waits for hardware the owner doesn't have yet.
Task: M10-T4
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
Yealink: BLF keys from line key 2, named after the extension (or the
number, e.g. a parking slot), with **N pickup; keys up to 33 past the
last BLF are cleared so removed BLFs leave the phone. Both vendors get
the time zone and NTP, and a phonebook URL that carries the phone's
credentials (phones fetch it like the config).
Time zones: packages/shared/src/phone-timezones.ts maps the US/Canada
zones (with aliases), UTC and London to Yealink's offset and name and
a POSIX string (Grandstream P64). Anything else is UTC, and the Phones
page warns.
/prov/phonebook/<mac>.xml (and <mac>/phonebook.xml, the name
Grandstream asks for under its path): every extension, then the
company contacts that have numbers, in the phone's vendor's format.
Same Basic auth as the config.
The config preview now masks both secrets wherever they appear (the
phonebook URL carries the provisioning password) and says when the
time zone falls back.
Golden files for both templates and both phonebooks in
src/provisioning/__golden__. e2e m10-t2: BLF keys, Central time, the
phonebook fetched from the URL in the config, hints for the BLF
targets, and the time zone warning. Vendor key names stay (verify per
model) until a real phone is checked.
Task: M10-T2
/api/devices CRUD (MAC in any format → 12 lowercase hex; a 20-char
provisioning password, stored encrypted and shown to admins, who type
it into the phone) and GET /:id/config-preview with passwords masked.
GET /prov/... per 07 §2: <mac>.cfg (Yealink), cfg<mac>.xml
(Grandstream), the empty Yealink common file. Basic auth with the MAC
and the device's password (timing-safe), 404 for unknown MACs or the
other vendor's file, 403 outside the local nets when LAN-only, 404 when
provisioning is off, Cache-Control no-store, last provisioned time and
IP recorded. Limits: 20 requests / 10 min per IP outside the local nets
(an office's phones rebooting together would pass it), and 10 failures
block the IP for an hour. A first request without credentials isn't a
failure: phones ask that way and retry after the 401.
Templates (pure functions of a ProvisionContext) hold the account,
voicemail, pickup codes and the provisioning settings; BLF, time zone
and phonebook come with M10-T2.
Web: Phones page (table, drawer with what to type into the phone,
config preview; manual settings for generic phones). e2e m10-t1: phone
added in the UI, 401 then the file, and a baresip phone registers with
only the account from it.
Also: the loadSnapshot test made its database inside the test (20 s);
under a full parallel run that timed out now and then. It now does it
in a hook like the others.
Task: M10-T1
The owner's voip.ms account has no SMS. M9-T1 (API client, number
import, server list) stays: it doesn't need SMS. Its check against a
real account (API enabled, this server's IP allowed) is still open.
Nothing in later milestones depends on SMS.
Task: M9-T1
integrations/voipms/client.ts: voipmsApi(url, creds, method, params)
per 06 §5.1 (GET rest.php, 15 s timeout, status "success" or an error
code). The API password is only in the query string: never logged,
never in an error. Known codes get a message for people
(invalid_credentials, ip_not_enabled, ...). getServers and getDids
parse forgivingly; no_did is an empty list.
Routes: GET /api/trunks/:id/voipms-servers, GET and POST
/api/trunks/:id/import-dids. Import adds only numbers voip.ms lists
and that aren't added yet, label from the description, destination
hangup. 400 without API credentials, 502 provider_error with voip.ms's
reason. VOIPMS_API_URL overrides the endpoint for tests.
Web: the trunk form's server list comes from voip.ms once the trunk
has API access, falling back to the built-in list with the reason.
"Import numbers from voip.ms" sits in the trunk form's API section and
the setup wizard's numbers step. Saving a new API password asks
voip.ms again (found in e2e: the old error stayed).
The response shapes are from the 06 spec, not a real account: voip.ms's
API docs need a login. Marked TODO(spec) and (verify) in 06. Tests use
a fake rest.php; e2e m9-t1 too (trunk created off, so nothing registers
to the real voip.ms).
Task: M9-T1
withinSlaPercent was "answered within sla_seconds / answered", so a
queue where a caller timed out still showed 100%. It is now the usual
service level: answered within sla_seconds out of every caller the
queue is done with, leaving out those who hung up before sla_seconds
(nobody could have answered them in time). Timed out and turned away
count as missed; callers still waiting aren't counted. Written into
04; the page label says "Callers answered within N s".
Real calls: m8-t1 now shows 50% (1 answered in time, 1 timed out, the
early hang-up left out) and the milestone check 67% (2 of 3), and
asserts it. The check moves to queue 552 (*45552) so it doesn't clash
with m8-t1's 550 on a shared database.
Task: M8-T1
scripts/e2e/m8-check.mjs, the scenario from 09 M8 with real calls from
a DID: queue 550 with agents 340 and 341, max wait 30 s, then 342. 340
pauses with *45550 and 341 is paused by an admin, so three callers
wait. 341 comes back and gets the longest waiting caller, parks them
on 701 (blind transfer to 700) and 343 picks up by dialing 701; 341
takes the next caller and the last one rings 342 after 30 s. Stats:
3 offered, 2 answered, 1 timed out.
Roadmap marks the M8 milestone check done; the e2e README describes
m8-t4, m8-t5 and m8-check.
Task: M8-check
/api/moh-classes: create (slug, not "default"), delete (409 while an
extension or queue uses it), add/play/remove files. Files are 8 kHz
wav in MOH_DIR/<class>/ and sound_files rows of kind moh. Extensions
and queues may only pick default or an existing class.
Two things found with real calls, written back into 03 and 04:
- An empty class plays silence on Asterisk 24. Only classes with
music are rendered, so a channel asking for an empty one gets
default.
- The generator reloads only changed files, and a file upload changes
no config (except a class's first/last). The route sends moh reload
after every file change.
Web: a Music on hold section on Sounds & music, and a select of sets
in the user drawer's Phone tab and the queue form. load-golden makes
the class folders like the api does. Golden case queue gains two
classes. e2e m8-t5: set made in the UI, upload listed by Asterisk, a
queue caller hears it, a user's moh_suggest, delete refused while
used, an emptied set falls back to default.
Task: M8-T5
/api/paging-groups CRUD (members as memberIds). The generator renders
sp-page-{id} per 03 §6.11 (desk phones only, auto-answer headers via
sp-sub-autoanswer) plus the number in sp-local-numbers. Added the `s`
option so phones already on a call aren't paged, which also keeps a
pager who is a member from being paged themselves. A group with no
members just hangs up.
*80<ext> intercom checks the target first: digits only and an endpoint
in sp-from-ext, else ss-noservice (the spec's one-liner passed
${EXTEN:3} straight into Page()). Both written back into 03.
Web: Paging page (list, drawer form with member checkboxes). Golden
case paging. e2e m8-t4 with baresip phones that honor the headers
(sip_autoanswer): an ordinary call rings, a page is answered by itself
and muted, the busy pager is skipped, *80 is two-way, *80999 refused.
The check on a real Yealink/Grandstream stays open (09).
Task: M8-T4
iamdoubz