4 Commits

Include renames

Author	SHA1	Message	Date
iamdoubz	17db68a8c5	sounds: upload, convert, play, rename, delete with references ... /api/sounds (admin) takes the raw file, converts it with ffmpeg in one pass to <id>.wav (8 kHz) and <id>.g722 (HD for G.722 calls) in SOUNDS_DIR, measures it with ffprobe, and stores it as custom/<id>. Delete answers 409 with what still plays it (IVR greeting/invalid sound, announcements, the recording notice). Sounds & music page: drop or pick a file (name suggested from the file name), play, rename, delete. MOH classes stay in M8. The web client and greetings share one raw-upload helper; the api test audio helpers moved to test/audio.ts. e2e m6-t1: drag-and-drop upload in the browser, Asterisk plays custom/<id>, rename, delete refused while in use, then deleted. Task: M6-T1	2026-10-18 10:13:58 -05:00
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	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	f60dda0bb0	me: voicemail page, greetings, My Phone home ... /api/me/voicemail lists, streams (Range, download) and marks read or deletes through AMI VoicemailMove/VoicemailRemove, then rescans so MWI and the list agree. /api/me/greetings takes the raw recording or file and converts it with ffmpeg to 8 kHz PCM (media/audio.ts). My Phone gets a layout with Home (new voicemails) and Voicemail (play marks read, download, delete, record/upload greetings via MediaRecorder); the list refetches on the live voicemail.new event. e2e m4-t3 checks MWI goes off within 5 s after play and delete, and a browser-recorded greeting is what callers hear. Task: M4-T3	2026-10-17 22:56:10 -05:00