Spec 11 §6.1: sp-sub-dial-user asks /api/internal/caller-name (func_curl, 2 s at most; this
machine only, with an APP_SECRET-derived key, not logged) for the ringing user's name for the
caller, and sets it on each ringing leg (pre-dial handler, CONNECTEDLINE), never on the call, so
forwards and other members don't see someone else's contacts. Ring group/queue/number prefixes
kept. Contrack lookups never wait (cache, contactsWarm every 5 min). Proven on real Asterisk
with baresip: apps/api/test/asterisk/cnam-test.ts.
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.
- Asterisk doesn't load app_system, func_shell, app_disa, res_agi, app_externalivr.
- Turning two-step sign-in on asks for the password again.
- *97/*98: one voicemail login at a time per phone, 10 an hour (sp-sub-vmlimit).
- /prov: an unknown MAC gets the same 401 as a known one.
- The login limiter sweeps keys past every window.
Also the v1 checklist with the owner's results from 2026-10-21.
Its Contact holds pn-prid=...:voip&...:remote, and PJSIP_DIAL_CONTACTS passes that '&' through,
so Dial rejected the whole device list and the call dropped for every device of the user.
sp-sub-contacts walks the AOR's contacts, drops pn-* params and FILTERs the rest. call-test
registers a phone with such a Contact and checks it is rung.
Owner's decision after the M11-T8 review: nothing capped how many calls a stolen password could
run at once. Settings → Security: calls.maxPerExtension (3) and calls.maxTotal (10), 0 = no
limit. Each sp-route-* counts the caller's extension (or whose forward it is: SP_LIMITEXT from
sp-sub-forward) with GROUP(spext) and every outside call with GROUP(spout); over a limit the
caller hears "all circuits are busy". 911 is never limited.
trunk-test.ts on a real Asterisk, limits 1 and 2: a 2nd call from 101's phone and a 3rd in total
are refused (with both 0 the same check fails). call-limits.test.ts covers the 0 combinations.
Also fixes the Security tab's hint: browser phones still work from anywhere.
Task: M11-T8
The 08 §1 threat table checked line by line against the code, plus a review by a separate agent
that hadn't written it. Findings were reproduced with a failing test before each fix:
- Critical: "011…,4" turned the route's Goto into Goto(sp-route-…,011…,4,1), starting past the
class check, so any extension could call abroad. FILTER on the dialed number (also stops "&"
adding a Dial target). trunk-test.ts proves it.
- High: -web endpoints (desk phone's password, any transport) skipped the phone ACL; now
127.0.0.1 only when it's on. call-test.ts: UDP refused, WebSocket from 127.0.0.1 registers.
- IVR direct dial no longer reaches paging groups or parked calls (sp-ivr-direct).
- Trunks identify_by = ip: a From of trunk-<id> from anywhere no longer rings phones.
- .dockerignore is an allowlist, so the public source archive can't pick up stray files.
- CSRF check on the matched route (/%61pi/… got around it); Linphone tokens masked in logs;
/api/live checks Origin and re-checks the session; *45/** filter digits; pg_dump password in
the environment; more webhook ranges blocked.
The review and what's left for the owner are in 08 §6.
Task: M11-T8
/api/paging-groups CRUD (members as memberIds). The generator renders
sp-page-{id} per 03 §6.11 (desk phones only, auto-answer headers via
sp-sub-autoanswer) plus the number in sp-local-numbers. Added the `s`
option so phones already on a call aren't paged, which also keeps a
pager who is a member from being paged themselves. A group with no
members just hangs up.
*80<ext> intercom checks the target first: digits only and an endpoint
in sp-from-ext, else ss-noservice (the spec's one-liner passed
${EXTEN:3} straight into Page()). Both written back into 03.
Web: Paging page (list, drawer form with member checkboxes). Golden
case paging. e2e m8-t4 with baresip phones that honor the headers
(sip_autoanswer): an ordinary call rings, a page is answered by itself
and muted, the busy pager is skipped, *80 is two-way, *80999 refused.
The check on a real Yealink/Grandstream stays open (09).
Task: M8-T4
Queues per 03 §6.8: queues_selfphoneng.conf (members as Local agents
following their hint), sp-queue-{id} (answer, prefix, record, Queue with
max wait, timeout destination), the number in sp-local-numbers and
destApp queue. Golden case queue; the Asterisk load test now checks
`queue show` lists each queue's members.
*45<queue> toggles login and checks PQMSTATUS/UPQMSTATUS, so a code for
a queue the caller isn't an agent of plays ss-noservice and stores
nothing (spec snippet fixed). Pauses live in queue_member_state, AstDB
SP/QPAUSE (resynced on connect) and app_queue; QueuePause is re-sent
after every `queue reload all` and rollback, which unpause everyone.
/api/queues CRUD with members [{extensionId, penalty}], admin
pause/unpause, GET /:id/stats; /api/me/queues/:id/pause|unpause for
Home. Stats come from queue_calls (new table), filled from app_queue's
AMI events plus UserEvent SpQueueExit in the fail branch: a real call
showed that when no agent rings, the Queue part of a call vanishes into
the next leg's CDR, and Asterisk 24 sends QueueCallerAbandon on
timeouts too.
Live: QueueTracker (QueueStatus on connect and after applies, queue
events); admins see every queue, users the queues they're agents of.
Web: Queues page (list, form with every column, detail page with
waiting callers, agents and pause buttons, today's stats), queue tiles
on the dashboard, queue login switches on Home, Queue in the
DestinationPicker.
CDR fixtures for queue answered/abandoned/timeout calls. e2e m8-t1 with
real phones: 3 callers wait with 2 agents, one paused by *45550 and one
in the UI, unpause connects the longest waiting caller, one times out
to the overflow phone, stats count it all. The whole e2e chain passes.
Task: M8-T1
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
Feature codes per 03 §6.13: *78/*79 DND, *72<number>/*73 forward always,
**<ext> directed pickup, *8 group pickup (also as a dialplan line, since
pickupexten only catches a phone's own INVITE), *43 echo test, *65 say
my extension; queue login and intercom stay for M8. SpToggle DND/CFU
update user_runtime_state; a *72 target the user's calling permission
doesn't allow is removed from AstDB, audited (forward_rejected) and sent
to the user as a toast (new private live message). Forward targets are
checked the same way everywhere (services/forwards.ts: an existing
internal number, or an outside number some outbound rule allows them).
GET/PATCH /api/me: DND, cfu/cfb/cfna, voicemail email options and PIN;
PostgreSQL first, then AstDB; AstDB resync on connect now includes DND
and forwards. Web: /me/settings (forwarding, DND, voicemail PIN and
email options), a status card on Home, toasts.
e2e m6-t7 with real phones covers every code, the refused forward toast,
and the settings page reaching AstDB; it found the settings forms losing
"Saved." on remount.
Task: M6-T7
/api/time-conditions (company time zone by default, ranges validated,
unique override digit, never pointing at itself) with a live status
("open/closed/holiday now", same logic as the dialplan, shared), and
POST /:id/override, which writes PostgreSQL then AstDB SP/TCOVR.
/api/holidays CRUD plus import-us-federal (observed dates, computed in
code, skips days already listed). services/astdb.ts resyncs AstDB on
every AMI connect (PostgreSQL wins); services/toggles.ts applies
SpToggle TCOVR from *31N/*32N/*33N.
Generator: sp-tc-{id} per 03 §6.10 (override, holidays within the next
year, ranges rendered a minute short because GotoIfTime's end is
inclusive), sp-tc-override with a no-service fallback, admin-only
*3[123]X in sp-featurecodes, destApp time_condition. The date changing
in the company time zone triggers a forced apply so holidays roll over.
Golden cases holidays and ivr-business-hours.
Office hours page: click-and-drag half-hour grid plus an exact range
list, destinations, holiday destination, override key, status chip with
Force open / Force closed / Automatic; Holidays tab with the US federal
import. e2e m6-t4 found two bugs (pointer capture release throwing,
drawer overflow) and checks routing, AstDB, *321/*331 from an admin
phone, refusal for others, and the import reaching the dialplan.
Task: M6-T4
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
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
Static sp-sub-* helpers and sp-from-ext from 03 §6.2 (with __SP_ variable
names), globals (default and E911 caller ID, per-extension class and admin
flags), and every user in sp-local-numbers with a PJSIP/<n>[&PJSIP/<n>-web]
hint. Voicemail stays off (ARG3=0) until M4. Contexts that later milestones
fill (emergency, feature codes, outbound routes) get a placeholder so
Asterisk doesn't warn about missing includes. Empty ccss.conf for the next
image build.
Task: M2-T4
iamdoubz