3 Commits

Include renames

Author	SHA1	Message	Date
iamdoubz	609965bba5	softphone: People panel with presence, click-to-call ... 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	2026-10-18 10:06:28 -05:00
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