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

scripts/e2e/README.md

@@ -31,7 +31,8 @@ node scripts/e2e/m2-live.mjs down
 
 `m3-t1.mjs` runs on the same setup (after `m1-t4.mjs`): it starts a fake voip.ms (a second Asterisk
 that accepts one sub-account) on the m2-live network, creates the trunk in the UI and waits for
-**Registered**, then presses Test.
+**Registered**, then presses Test. `m3-t2.mjs` (after `m3-t1.mjs`) adds a phone number routed to a user in the UI and
+checks `sp-inbound` in Asterisk, and that the extension can no longer be deleted.
 
 `pnpm test:asterisk` (golden-file load test, `call-test.ts` with 30 s two-way audio between two
 baresip phones, and `trunk-test.ts` with registration against a fake provider) needs no api at all.