iamdoubz
57e0e3dac2
Feature codes per 03 §6.13: *78/*79 DND, *72<number>/*73 forward always, **<ext> directed pickup, *8 group pickup (also as a dialplan line, since pickupexten only catches a phone's own INVITE), *43 echo test, *65 say my extension; queue login and intercom stay for M8. SpToggle DND/CFU update user_runtime_state; a *72 target the user's calling permission doesn't allow is removed from AstDB, audited (forward_rejected) and sent to the user as a toast (new private live message). Forward targets are checked the same way everywhere (services/forwards.ts: an existing internal number, or an outside number some outbound rule allows them). GET/PATCH /api/me: DND, cfu/cfb/cfna, voicemail email options and PIN; PostgreSQL first, then AstDB; AstDB resync on connect now includes DND and forwards. Web: /me/settings (forwarding, DND, voicemail PIN and email options), a status card on Home, toasts. e2e m6-t7 with real phones covers every code, the refused forward toast, and the settings page reaching AstDB; it found the settings forms losing "Saved." on remount. Task: M6-T7
6.9 KiB
Browser checks
Dependency-free scripts that drive a headless Chromium (Edge/Chrome) over the DevTools protocol. They need an api serving the built web UI and a fresh database.
pnpm --filter @selfphoneng/web build && mkdir -p /tmp/sp && cp -r apps/web/dist /tmp/sp/public
# fresh DB, then from /tmp/sp:
DATABASE_URL=postgres://… INITIAL_ADMIN_EMAIL=admin@example.com INITIAL_ADMIN_PASSWORD=initial-admin-pw \
node --env-file=<repo>/.env --conditions=source <repo>/apps/api/src/index.ts
BROWSER=/usr/bin/chromium node scripts/e2e/m1-t4.mjs http://127.0.0.1:8080 admin@example.com initial-admin-pw /tmp/shots
node scripts/e2e/mobile.mjs http://127.0.0.1:8080 admin@example.com correct-horse-battery-staple /tmp/shots
m1-t4.mjs is the M1-T4 acceptance flow: forced password change, then create/edit/delete a user.
M2 live loop (any Docker host, no host networking)
m2-live.mjs runs the whole chain on a private Docker network: a test Asterisk with AMI published
on 127.0.0.1:15038, the api writing config into it, two baresip phones, and the dashboard in a
browser. It checks that a new extension is live in Asterisk within 2 s, both phones show online,
and the dots turn red during a call.
node scripts/e2e/m2-live.mjs up /tmp/sp-live-conf
# start the api with AMI_PORT=15038 AMI_SECRET=m2-live-secret ASTERISK_CONFIG_DIR=/tmp/sp-live-conf
node scripts/e2e/m1-t4.mjs http://127.0.0.1:8080 admin@example.com initial-admin-pw /tmp/shots
node scripts/e2e/m2-live.mjs check http://127.0.0.1:8080 admin@example.com correct-horse-battery-staple /tmp/shots
node scripts/e2e/m2-live.mjs down
m3-t1.mjs runs on the same setup (after m1-t4.mjs): it starts a fake voip.ms (a second Asterisk
that accepts one sub-account) on the m2-live network, creates the trunk in the UI and waits for
Registered, then presses Test. m3-t2.mjs (after m3-t1.mjs) adds a phone number routed to a user in the UI and
checks sp-inbound in Asterisk, and that the extension can no longer be deleted. m3-t3.mjs (after
both) checks the default outbound rules, the test box, a user's calling permission, and that
reordering changes the include order in Asterisk. m3-t4.mjs (after all three) runs a tiny SMTP sink on
127.0.0.1:2525 and a baresip phone as 101, then checks the dashboard warnings, Settings → Email and
Emergency, and the Test notifications button (email arrives, 101 rings, red banner). Nothing dials 911. The fake provider from m3-t1.mjs stays up until m2-live.mjs down: a container that took
over its IP would be identified as the trunk.
pnpm test:asterisk (golden-file load test, call-test.ts with 30 s two-way audio between two
baresip phones, and trunk-test.ts with registration against a fake provider) needs no api at all.
m3-t5.mjs is the M3-T5 acceptance and needs its own fresh database: it signs in with the
initial password, walks all ten wizard steps against a fake voip.ms, then proves an inbound call
to the DID rings the new user's softphone and an outbound call leaves with the DID as caller ID.
m4-t2.mjs is the M4-T2 acceptance (no browser): start m2-live with a voicemail dir
(m2-live.mjs up <confDir> <vmDir>) and the api with VOICEMAIL_DIR=<vmDir>. It runs the SMTP
sink (smtp-sink.mjs, shared with m3-t4.mjs), adds a user and extension 150 with voicemail,
leaves a message from a Local channel, and checks the email arrives within 10 s with a PCM 8 kHz
wav attached. The voicemail itself (MWI on a phone, *97) is covered by pnpm test:asterisk.
m4-t3.mjs <baseUrl> <adminEmail> <adminPassword> <vmDir> <shotsDir> is the M4-T3 acceptance on the
same setup: it adds a user with extension 160 and a baresip phone with the mwi module, leaves
messages, then as that user in the browser plays one (marks it read) and deletes another, each
time checking the phone's MWI goes off within 5 s. It records the "no answer" greeting with a
fake microphone and checks a caller hears it.
m5-t1.mjs <baseUrl> <adminEmail> <adminPassword> <shotsDir> is the M5-T1 check in Docker: start
the api with WEBRTC_WS_URL=ws://127.0.0.1:18088/ws. m2-live.mjs up publishes Asterisk's HTTP
server on 127.0.0.1:18088 and RTP 10000–10019/udp. The script sets LAN IP = the container and public
IP = 127.0.0.1 (so ICE offers 127.0.0.1), restarts telephony, then signs in as a new user, enables
the softphone with a fake microphone, and checks two-way audio to a baresip phone (171) in both
directions.
m5-t2.mjs (after m5-t1.mjs, same user) starts two baresip phones (172, 173) through
phones.mjs and checks the softphone features: the incoming call dialog, DTMF, hold (music on hold
at the other end), call waiting, blind and attended transfer, and "Use here" moving the softphone to
a second tab (cdp.mjs newTab()).
m5-t3.mjs (after m5-t2.mjs) is the M5-T3 acceptance. It watches People in one browser while 173
calls 172 and measures the time from Asterisk's hint changing to the dot changing (must be < 1 s).
Then a second browser, as a new user with desk phone 174 and the softphone off, clicks Call on
172: 174 must ring first (Asterisk channel order), then the two are connected.
m6-t1.mjs needs m2-live started with a sounds dir (m2-live.mjs up <confDir> <vmDir> <soundsDir>)
and the api with SOUNDS_DIR=<soundsDir>. It drops a generated wav on Sounds & music, uploads it,
checks that Asterisk can play custom/<id>, renames it, and deletes it, which is refused while it's
the recording notice. cdp.mjs fill()/click() take an optional selector to search within.
m6-t2.mjs (after m1-t4.mjs) creates ring group 500 in the UI with members 190 (never answers)
and 191 (auto-answer), and checks that a call from 192 rings both and is answered by 191. It then
switches the group to "one at a time" with 5 s each and checks that 190 rings alone before 191.
m6-t3.mjs (sounds dir like m6-t1.mjs) builds digital receptionist 600 in the UI. It uploads the
greeting through the form's SoundPicker, sends key 1 to 193 and key 2 to 194, and then a Local
channel sending DTMF presses 1, 2 and a wrong key (5): 193 or 194 rings, or "not a valid choice"
plays and then the greeting again.
m6-t4.mjs builds office hours in the UI. It paints Saturday 10–11 on the grid and checks that
Force closed / Force open route a call to 196 / 195 and set or clear AstDB SP/TCOVR/<id>. *321
and *331 from the admin's own phone (197) must reach the database and the page, *311 from a
non-admin phone must get "no service", and the 2026 US federal holiday import must reach the
dialplan.
m6-t7.mjs (after m1-t4.mjs) checks the feature codes from a real phone (198, belonging to user
fwd@example.com):
*78/*79set DND (the database, AstDB and the page follow, and calls meanwhile go to voicemail).*72199/*73forward everything to 199. A*72to an outside number this user may not call is refused, removed from AstDB, and shown as a toast./me/settingssaves a no-answer forward.*43echo test and*65say my extension.**200and*8pick up a call ringing at 200.