5 Commits

Include renames

Author	SHA1	Message	Date
iamdoubz	0deff2a637	dial plan: made from routes and number lengths; Yealink dial-now, the ATA plan on the Credentials tab	2026-10-21 17:14:13 -05:00
iamdoubz	aa6681da03	numbers: any free internal number of 1 to 6 digits; ranges only suggest ... Owner wanted the first user on 1. Numbering ranges now choose the Next free suggestion (Settings → Numbering), the setup wizard takes a number per person, and a one-digit number can't also be a menu key of a digital receptionist with direct dialing. Dialplan patterns that take an internal number match one digit (_X!); golden case short-numbers proves it in Asterisk.	2026-10-21 13:17:25 -05:00
iamdoubz	1b7f60b5f2	trunks: CRUD API, PJSIP trunk config, live registration status, Trunks page ... voip.ms-first trunk form with the portal checklist, POP server list and Test button; passwords are write-only and encrypted. Registration and IP-auth trunks render per 03 §5.3 (TCP/TLS add ;transport=). The API refuses a server name that doesn't resolve and transports that are off, because either makes Asterisk drop the trunk. Live state: registration from PJSIPShowRegistrationsOutbound (re-read on every Registry event: two sub-accounts on one server share Username/Domain), reachability from PJSIPShowEndpoint and trunk ContactStatus (static contacts aren't in PJSIPShowContacts). Trunk state is admin-only. Dashboard gets a "Trunks up" card. trunk-test.ts (in pnpm test:asterisk) runs a fake voip.ms (second Asterisk): the good trunk registers, a wrong password is Rejected, qualify reaches the provider; its recorded AMI drives the unit tests. scripts/e2e/m3-t1.mjs does the same through the UI. The acceptance check with a real voip.ms sub-account is still to do. Tests: settle() now awaits an apply instead of a fixed sleep (flaked under full-suite load). Task: M3-T1	2026-10-17 19:42:14 -05:00
iamdoubz	c0c4036887	live: AMI-fed live state, /api/live WebSocket, dashboard and status dots ... 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	1cfc13ead3	extensions: numbering service, CRUD API, credentials, Users page tabs ... One namespace for internal numbers (range, reserved codes, parking, every object table) with a next-free lookup. Creating an extension generates its SIP password and voicemail PIN and a runtime-state row; every change and password regeneration triggers a config apply. Credentials are revealed only on request and each reveal is audited. Users get an Ext. column and a drawer with General (number + next free), Phone and Credentials tabs. scripts/e2e/m2-t5.mjs checks the flow in a browser and that the generated PJSIP file follows; the files it produced load cleanly in Asterisk. Task: M2-T5	2026-10-17 16:35:07 -05:00