# 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. ```sh 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=/.env --conditions=source /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. ```sh 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 `) and the api with `VOICEMAIL_DIR=`. 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 ` 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 ` 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()`).