iamdoubz
c7fd4df77c
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
4.7 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()).