The certificate lives in data/tls (fullchain.pem + privkey.pem), mounted
read-only into asterisk and the api. Asterisk quietly drops transport-tls
when the files are missing, so the api won't turn TLS on unless they load
and match, shows the certificate on Settings -> Security, and warns on the
dashboard when it expires within 14 days. scripts/tls-cert.sh copies it
from the bundled Caddy (or given paths) and reloads res_pjsip: Asterisk 24
re-reads a changed certificate on reload, calls stay up.
transport-tls checks the provider's certificate (verify_server). Asterisk
24 checks a qualify's certificate against the IP instead of the name, which
fails on voip.ms, so TLS trunks don't qualify; REGISTER and INVITE verify
the name fine.
Task: M11-T2
/api/moh-classes: create (slug, not "default"), delete (409 while an
extension or queue uses it), add/play/remove files. Files are 8 kHz
wav in MOH_DIR/<class>/ and sound_files rows of kind moh. Extensions
and queues may only pick default or an existing class.
Two things found with real calls, written back into 03 and 04:
- An empty class plays silence on Asterisk 24. Only classes with
music are rendered, so a channel asking for an empty one gets
default.
- The generator reloads only changed files, and a file upload changes
no config (except a class's first/last). The route sends moh reload
after every file change.
Web: a Music on hold section on Sounds & music, and a select of sets
in the user drawer's Phone tab and the queue form. load-golden makes
the class folders like the api does. Golden case queue gains two
classes. e2e m8-t5: set made in the UI, upload listed by Asterisk, a
queue caller hears it, a user's moh_suggest, delete refused while
used, an emptied set falls back to default.
Task: M8-T5
sp-sub-record now also sets CDR(SP_REC); phone numbers can record every
call (record_inbound), with the notice from Settings → Recording played
first (SP_REC_ANNOUNCE). Aggregation links the MixMonitor file as a
recordings row (size, ffprobe length). /api/recordings lists, plays
(Range; admins and the people on the call) and deletes; recordings past
recordings.retentionDays are deleted daily.
Web: Recordings page (play, download, delete), a player in Call history
and My calls, a Recording settings tab with the consent notice (08 §5),
and "Record every call" on phone numbers.
e2e m7-t3: the notice plays before a recorded inbound call and the
recording plays from Call history (the M7 acceptance). The AMI
Originate helper moved to scripts/e2e/ami.mjs.
Task: M7-T3
/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
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
services/voicemail.ts rescans a mailbox on Asterisk's MessageWaiting event
(and every mailbox on AMI connect) instead of watching files, upserts
voicemail_messages by msg_id, marks vanished ones deleted, sends
voicemail.new to the owner only, and emails the wav. Delete-after-email
uses AMI VoicemailRemove, and only when the audio was attached. Asterisk
24 has VoicemailMove/VoicemailRemove, so 04 §5.6 drops the renumbering
plan. New VOICEMAIL_DIR setting; e2e m4-t2 checks the email arrives
within 10 s with a playable attachment (SMTP sink shared with m3-t4).
Task: M4-T2
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
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
iamdoubz