4 Commits

Include renames

Author	SHA1	Message	Date
iamdoubz	c117d8bb1d	dids: phone numbers, destinations, inbound dialplan, Phone numbers page ... DIDs CRUD (any input format, normalized; 409 on duplicates; a number that is an emergency or default caller ID can't be removed). Shared destination service: assertDestinationExists, describeDestination ("Ring Dan Doe (101)") and findReferences over every destination column in 02 §1 plus the unmatched-number setting; deleting an extension a destination points at is now a 409 naming the places. Generator: destApp (user, external, hangup for now), sp-from-trunk-* per enabled trunk, sp-did-exists (left out when empty) and sp-inbound with caller-name prefix. Golden case voipms-basic. Phone numbers page with a minimal DestinationPicker (user, outside number, hang up). trunk-test.ts: the fake provider calls the DID with a 10-digit Request-URI like voip.ms; baresip 101 answers, caller ID arrives normalized, 500 packets each way on both legs. scripts/e2e/m3-t2.mjs checks the UI → Asterisk path. Still to do with real voip.ms: the mobile-to-DID call and recording which header carries the DID (06 §3). Task: M3-T2	2026-10-17 19:52:49 -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	06ac99b9f4	generator: PJSIP transports, extension endpoints and rtp config ... Golden cases empty, one-user, webrtc-disabled and hostile-names; each also renders identically from shuffled input. pnpm test:asterisk boots the asterisk image with every golden case and fails on a missing endpoint or context or any ERROR log line (checked against a deliberately broken case). On 24.0.0-rc2: 101 and 101-web load with every set_var. Task: M2-T3	2026-10-17 16:23:14 -05:00