/api/ring-groups CRUD with members as ordered memberIds (unique, must
exist), number from the ring-group range, a no-answer destination
(never itself), and delete refused while something points at it. The
generator renders sp-rg-{id} (Dial through sp-rg-member: all at once with
the total ring time, or one at a time with the per-member time, music
instead of ringing on request), adds the number to sp-local-numbers, and
destApp sends ring_group destinations there. Golden case ringgroups.
The DestinationPicker is now generic over targets picked by id and gains
"Voicemail of a user" and "Ring group" (M6-T6 completes it). Caller-name
prefixes (DIDs and ring groups) get a space before the caller's name.
Ring groups page with an ordered member list.
e2e m6-t2 found the new-group form failing on an unset no-answer
destination (fixed: defaults to hang up). It checks ring-all (both ring,
191 answers) and hunt (190 alone for 5 s, then 191) with real calls. The
generator debounce test now waits for the apply instead of a fixed 150 ms.
Task: M6-T2
/api/sounds (admin) takes the raw file, converts it with ffmpeg in one
pass to <id>.wav (8 kHz) and <id>.g722 (HD for G.722 calls) in SOUNDS_DIR,
measures it with ffprobe, and stores it as custom/<id>. Delete answers 409
with what still plays it (IVR greeting/invalid sound, announcements, the
recording notice). Sounds & music page: drop or pick a file (name
suggested from the file name), play, rename, delete. MOH classes stay in
M8. The web client and greetings share one raw-upload helper; the
api test audio helpers moved to test/audio.ts.
e2e m6-t1: drag-and-drop upload in the browser, Asterisk plays
custom/<id>, rename, delete refused while in use, then deleted.
Task: M6-T1
People lists every extension with its status dot and a search box: a
right-hand column on wide screens, /me/people on small ones. Call uses the
browser softphone when it's on, else POST /api/me/click-to-call, which
originates to the user's desk phone first (409 if none is registered).
During a call: Transfer (blind), Talk first (consult), or drag the call
card onto a person; a held call card offers Complete transfer whenever
another call is active. useSoftphone moved to its own module.
e2e m5-t3: the dot changes ~30 ms after Asterisk's hint; click-to-call
rings 174 before 172. cdp click() takes a scope (People also has
Call/Transfer buttons). trunk-test ignores the RTCP-at-teardown EPERM
error seen once.
Task: M5-T3
The dock opens a panel with a 12-key dialpad (typing, paste normalization,
Enter) and microphone/speaker selection. Call cards have mute, hold,
keypad (DTMF) and transfer: "Transfer now" (blind) or "Talk first"
(attended, with Complete/Cancel). An incoming call opens a dialog when
idle, or a card during a call (call waiting holds the first call). A
synthesized WebAudio ringtone and desktop notifications, auto-answer for
Call-Info answer-after=0, and a poor-connection warning from getStats.
One tab registers: a Web Lock plus a BroadcastChannel "Use here".
Found by e2e: Chrome can report canInsertDTMF=false, so DTMF falls back to
SIP INFO; after a REFER's 2xx NOTIFY the browser sends BYE itself (RFC
5589), otherwise SIP.js keeps the call up. apps/web/TESTING.md is the
12-step manual run for Chrome and Firefox (still open, roadmap note);
e2e m5-t2 automates most of it in headless Edge.
Task: M5-T2
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
/api/me/voicemail lists, streams (Range, download) and marks read or
deletes through AMI VoicemailMove/VoicemailRemove, then rescans so MWI
and the list agree. /api/me/greetings takes the raw recording or file and
converts it with ffmpeg to 8 kHz PCM (media/audio.ts). My Phone gets a
layout with Home (new voicemails) and Voicemail (play marks read,
download, delete, record/upload greetings via MediaRecorder); the list
refetches on the live voicemail.new event. e2e m4-t3 checks MWI goes off
within 5 s after play and delete, and a browser-recorded greeting is what
callers hear.
Task: M4-T3
services/voicemail.ts rescans a mailbox on Asterisk's MessageWaiting event
(and every mailbox on AMI connect) instead of watching files, upserts
voicemail_messages by msg_id, marks vanished ones deleted, sends
voicemail.new to the owner only, and emails the wav. Delete-after-email
uses AMI VoicemailRemove, and only when the audio was attached. Asterisk
24 has VoicemailMove/VoicemailRemove, so 04 §5.6 drops the renumbering
plan. New VOICEMAIL_DIR setting; e2e m4-t2 checks the email arrives
within 10 s with a playable attachment (SMTP sink shared with m3-t4).
Task: M4-T2
Generate voicemail_selfphoneng.conf (03 §7) with one mailbox per extension
that has voicemail on, turn on ARG3 in sp-sub-user, and add *97/*98 to
sp-featurecodes (which replaces the pending placeholder). Endpoints with a
mailbox get mwi_subscribe_replaces_unsolicited so phones that SUBSCRIBE for
MWI aren't refused with a 500. A new test:asterisk check leaves a message,
sees MWI on a baresip phone, plays it with *97 and sees MWI turn off.
Task: M4-T1
/setup walks a fresh install to working calls: password (done by the
forced change), company, network (detect public IP, LAN IP from
/api/system/status, router checklist; restarts telephony because the
transports changed), email (skippable), voip.ms trunk (Next unlocks on
Registered), phone numbers, users (extension auto-numbered, temporary
passwords shown once), main number routing, emergency (default E911
caller ID required once a trunk exists), done. Steps 7-8 ship now
because the acceptance needs a user and a routed number; their M6
choices (everyone, IVR) come later. Admins land on /setup until it is
finished or skipped.
api: /api/setup/status and /complete, POST
/api/settings/network/detect-public-ip (ipify, then ifconfig.me),
GET /api/system/status (M3 subset), POST /api/system/asterisk/restart
(an AMI disconnect during a restart counts as success).
restartRequired clears on AMI connect once Asterisk's uptime shows it
started after the change. The first phone number becomes the default
outbound caller ID when none is set (found by the acceptance run:
calls otherwise left with the provider's caller ID).
Verified: scripts/e2e/m3-t5.mjs on a fresh database against the fake
voip.ms: wizard end to end, then an inbound call to the DID rings the
new user's softphone and an outbound call leaves as 13125550000 with
the DID in Remote-Party-ID. m1-t4 now passes through "Skip setup".
Task: M3-T5
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
Outbound routes CRUD, reorder and POST /test (mirrors the dialplan:
911 and internal numbers first, then the first matching enabled route,
class check, usable trunks, caller ID priority route > user > default).
The first trunk creates the default rules: toll-free first (because
1NXXNXXXXXX also matches 1800...), US/Canada, international disabled.
Generator: Asterisk orders patterns in one context by specificity, not
by file order, so each route gets its own sp-or-{id} context and
sp-outbound-routes includes them in position order (spec 03 §6.12
updated). sp-route-{id}: class check -> ss-noservice, caller ID
priority, per-trunk GROUP channel limit, failover on CHANUNAVAIL or
CONGESTION, all-circuits-busy-now when every trunk is full or off.
UI: Outbound rules page (↑/↓ reorder, pattern rows with help, ordered
trunk list, caller ID override, test box) and a Calling tab on users
(calling permission, outbound caller ID).
trunk-test.ts: 101 dials 3125550000 -> the fake provider receives
13125550000 with the DID in Remote-Party-ID (From user is the
sub-account), two-way audio; a caller without a class hears
ss-noservice. scripts/e2e/m3-t3.mjs covers the UI. With real voip.ms
still to do: whether per-call caller ID is honored (06 §4).
Task: M3-T3
DIDs CRUD (any input format, normalized; 409 on duplicates; a number
that is an emergency or default caller ID can't be removed). Shared
destination service: assertDestinationExists, describeDestination
("Ring Dan Doe (101)") and findReferences over every destination
column in 02 §1 plus the unmatched-number setting; deleting an
extension a destination points at is now a 409 naming the places.
Generator: destApp (user, external, hangup for now), sp-from-trunk-*
per enabled trunk, sp-did-exists (left out when empty) and sp-inbound
with caller-name prefix. Golden case voipms-basic. Phone numbers page
with a minimal DestinationPicker (user, outside number, hang up).
trunk-test.ts: the fake provider calls the DID with a 10-digit
Request-URI like voip.ms; baresip 101 answers, caller ID arrives
normalized, 500 packets each way on both legs. scripts/e2e/m3-t2.mjs
checks the UI → Asterisk path. Still to do with real voip.ms: the
mobile-to-DID call and recording which header carries the DID (06 §3).
Task: M3-T2
voip.ms-first trunk form with the portal checklist, POP server list
and Test button; passwords are write-only and encrypted. Registration
and IP-auth trunks render per 03 §5.3 (TCP/TLS add ;transport=). The
API refuses a server name that doesn't resolve and transports that are
off, because either makes Asterisk drop the trunk.
Live state: registration from PJSIPShowRegistrationsOutbound (re-read
on every Registry event: two sub-accounts on one server share
Username/Domain), reachability from PJSIPShowEndpoint and trunk
ContactStatus (static contacts aren't in PJSIPShowContacts). Trunk
state is admin-only. Dashboard gets a "Trunks up" card.
trunk-test.ts (in pnpm test:asterisk) runs a fake voip.ms (second
Asterisk): the good trunk registers, a wrong password is Rejected,
qualify reaches the provider; its recorded AMI drives the unit tests.
scripts/e2e/m3-t1.mjs does the same through the UI. The acceptance
check with a real voip.ms sub-account is still to do.
Tests: settle() now awaits an apply instead of a fixed sleep (flaked
under full-suite load).
Task: M3-T1
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
One namespace for internal numbers (range, reserved codes, parking, every
object table) with a next-free lookup. Creating an extension generates its
SIP password and voicemail PIN and a runtime-state row; every change and
password regeneration triggers a config apply. Credentials are revealed
only on request and each reveal is audited. Users get an Ext. column and a
drawer with General (number + next free), Phone and Credentials tabs.
scripts/e2e/m2-t5.mjs checks the flow in a browser and that the generated
PJSIP file follows; the files it produced load cleanly in Asterisk.
Task: M2-T5
Static sp-sub-* helpers and sp-from-ext from 03 §6.2 (with __SP_ variable
names), globals (default and E911 caller ID, per-extension class and admin
flags), and every user in sp-local-numbers with a PJSIP/<n>[&PJSIP/<n>-web]
hint. Voicemail stays off (ARG3=0) until M4. Contexts that later milestones
fill (emergency, feature codes, outbound routes) get a placeholder so
Asterisk doesn't warn about missing includes. Empty ccss.conf for the next
image build.
Task: M2-T4
Golden cases empty, one-user, webrtc-disabled and hostile-names; each also
renders identically from shuffled input. pnpm test:asterisk boots the
asterisk image with every golden case and fails on a missing endpoint or
context or any ERROR log line (checked against a deliberately broken case).
On 24.0.0-rc2: 101 and 101-web load with every set_var.
Task: M2-T3
render() is pure and gives every file the header (bodies arrive per task).
apply() runs one at a time, debounces requestApply, writes changed files
atomically (mode 0640), keeps .prev, sends each needed reload once, skips
pjsip reload and sets restartRequired when transports change, and records
generator_runs. AMI connects on startup and every reconnect forces a full
apply. Admin routes: config apply/rollback and a masked generated-files view.
Task: M2-T2
Parser tests replay a real Asterisk 24 session recorded byte for byte, in
every chunking from 1 byte up. Client tests use an in-process fake AMI.
Checked live against the container: login, command, hard kill rejects the
pending action, reconnect follows the 1/2/4/8/16/30 s backoff.
Task: M2-T1
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
Adds stub configs and noloads so a clean boot logs no ERROR lines,
ships stock indications.conf, seeds default MOH from the entrypoint.
Spec 01 synced with the real files.
Task: M0-T5
iamdoubz
iamdoubz