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

scripts/e2e/README.md

@@ -32,7 +32,9 @@ 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. `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.
+checks `sp-inbound` in Asterisk, and that the extension can no longer be deleted. `m3-t3.mjs` (after
+both) checks the default outbound rules, the test box, a user's calling permission, and that
+reordering changes the include order in Asterisk.
 
 `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.