Commit Graph
7 Commits
Author SHA1 Message Date
iamdoubz c7fd4df77c softphone: dock panel, incoming dialog, hold/mute/DTMF, transfers, one tab
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
2026-10-18 09:43:41 -05:00
iamdoubz ae3a723a3c softphone: SIP.js engine, /api/me/webrtc, minimal dock
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
2026-10-18 09:27:45 -05:00
iamdoubz 1e50d12c6e docs: track open real voip.ms acceptance checks for M3
Task: M3-T1..T3
2026-10-17 22:07:42 -05:00
iamdoubz c2b9fff126 setup: first-run wizard, telephony restart, system status
ci / check (push) Canceled after 0s
ci / images (push) Canceled after 0s
/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
2026-10-17 21:11:04 -05:00
iamdoubz c0c4036887 live: AMI-fed live state, /api/live WebSocket, dashboard and status dots
ci / check (push) Canceled after 0s
ci / images (push) Canceled after 0s
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
2026-10-17 16:49:25 -05:00
iamdoubz 909ac2d62d asterisk: 24.0.0-rc2 image built from source with static config
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
2026-10-17 14:31:05 -05:00
iamdoubz 547bb68a6f docs: initial SelfPhoneNG spec
Fix leftover __OP_* dialplan variables from the openphone rename (all
readers use SP_*). Record the reverse-proxy decision in 00 §10.
2026-10-17 14:11:09 -05:00