Owner: the first upload always played first. A class is now an Asterisk playlist in the order
set on Sounds & music (move up/down), new uploads last, with a Shuffle option (sort = random).
moh-test.ts plays both in a real Asterisk and reads the file order from its debug log.
.github/workflows/release.yml runs on Gitea and on the GitHub mirror (release: published only)
and pushes selfphoneng-api and selfphoneng-asterisk to that forge's registry: ghcr.io with the
job token, the Gitea registry with secrets REGISTRY_USER + REGISTRY_TOKEN (Gitea's job token
can't push packages). Tags: the version, plus <major>.<minor> and latest unless it's a
pre-release. It stops unless the tag is v + package.json's version, and checks the images
before pushing.
Run locally with act_runner exec against a throwaway registry: v0.0.1 pushed 0.0.1, 0.0 and
latest for both images (VERSION 0.0.1, OCI labels, source archive selfphoneng-0.0.1/);
v9.9.9 failed at the version check.
Version 0.0.1 in all package.json files, CHANGELOG.md started, release steps in 08 §3.1.
Task: none (owner request, 2026-10-20)
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
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
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
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
iamdoubz