Commit Graph
8 Commits
Author SHA1 Message Date
iamdoubz 0d29131d18 voicemail: mailbox config, voicemail on no-answer/busy, *97/*98, MWI
Generate voicemail_selfphoneng.conf (03 §7) with one mailbox per extension
that has voicemail on, turn on ARG3 in sp-sub-user, and add *97/*98 to
sp-featurecodes (which replaces the pending placeholder). Endpoints with a
mailbox get mwi_subscribe_replaces_unsolicited so phones that SUBSCRIBE for
MWI aren't refused with a 500. A new test:asterisk check leaves a message,
sees MWI on a baresip phone, plays it with *97 and sees MWI turn off.

Task: M4-T1
2026-10-17 22:21:53 -05:00
iamdoubz 409724bc55 emergency: 911 dialplan, notifications, test button, settings, warnings
Dialplan: sp-emergency-dial (911, 9911) first in sp-from-ext;
sp-emergency has no class, channel-limit or DND check, dials every
enabled trunk in name order with the E911 caller ID, and raises
UserEvent SpEmergency. Static sp-emergency-alert (_X! so "0" matches)
for the ring-out.

api: SpEmergency -> emergency_events row, live emergency.call (admins
and notify users only), email via nodemailer (SMTP from settings) and
an Originate to every notify extension, each outcome recorded. Test
button runs the same with TEST in front, records and dials nothing.
GET/PATCH /api/settings (smtp password write-only, references
checked, read-only system keys), SMTP test, /api/system/warnings.
Fix: clearing a nullable setting stored SQL NULL (now JSON null).

UI: Settings page (Email, Emergency with the 06 §6 notice), user
Emergency tab, dashboard warnings banner, red 911 banner.

Verified: generator/emergency.test.ts; trunk-test.ts dials 911 from a
caller with no class to the fake provider (E911 caller ID in RPID,
SpEmergency on AMI, alert plays on baresip 101); scripts/e2e/m3-t4.mjs
(SMTP sink receives the test email, 101 rings, banner shows).
nodemailer pinned to 10.0.11 (newest release over two weeks old).

Task: M3-T4
2026-10-17 20:52:50 -05:00
iamdoubz 4496f6b89a outbound: routes with failover and caller ID, test box, Outbound rules page
Outbound routes CRUD, reorder and POST /test (mirrors the dialplan:
911 and internal numbers first, then the first matching enabled route,
class check, usable trunks, caller ID priority route > user > default).
The first trunk creates the default rules: toll-free first (because
1NXXNXXXXXX also matches 1800...), US/Canada, international disabled.

Generator: Asterisk orders patterns in one context by specificity, not
by file order, so each route gets its own sp-or-{id} context and
sp-outbound-routes includes them in position order (spec 03 §6.12
updated). sp-route-{id}: class check -> ss-noservice, caller ID
priority, per-trunk GROUP channel limit, failover on CHANUNAVAIL or
CONGESTION, all-circuits-busy-now when every trunk is full or off.

UI: Outbound rules page (↑/↓ reorder, pattern rows with help, ordered
trunk list, caller ID override, test box) and a Calling tab on users
(calling permission, outbound caller ID).

trunk-test.ts: 101 dials 3125550000 -> the fake provider receives
13125550000 with the DID in Remote-Party-ID (From user is the
sub-account), two-way audio; a caller without a class hears
ss-noservice. scripts/e2e/m3-t3.mjs covers the UI. With real voip.ms
still to do: whether per-call caller ID is honored (06 §4).

Task: M3-T3
2026-10-17 20:04:35 -05:00
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
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 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
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