Merge pull request #306 from outis1one/claude/ionos-script-integration-x32ofw
Claude/ionos script integration x32ofw
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# PSTN + DID + SMS — end-to-end test checklist
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For verifying an **already-installed** Asterisk + PSTN trunk + SMS setup
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(droplet or home/LAN) actually works, top to bottom. Not a setup guide —
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see `docs/anveo-direct-setup-guide.md` for account/droplet setup from
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scratch, and `docs/pstn-calling-voipms-plan.md` for the design/architecture
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background. This doc assumes both are already done and just walks through
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proving it all still works, in order, with what to check when a step fails.
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Run through this after any of: first go-live, a provider/DID change, an
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Asterisk or PSTN trunk reinstall, or just periodically to catch drift (a
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provider-side change, an expired international allow-list, a forgotten
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kill-switch trip).
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## 0. Before you start
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`$CONTAINER`/`$EA_DIR` only live in the shell session where you set them —
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a new terminal, a reboot, or just enough time passing between test steps
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leaves them empty again, and an empty `$CONTAINER` silently turns
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`docker exec -it $CONTAINER asterisk -rx ...` into `docker exec -it
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asterisk -rx ...`, which fails with "No such container: asterisk" (there's
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no container literally named that — it's `easy-asterisk` or
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`easy-asterisk-do`). Auto-detect instead of typing it by hand, so a stale
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or forgotten variable can't silently break every command below:
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```bash
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CONTAINER="$(docker ps --format '{{.Names}}' | grep -m1 -E '^easy-asterisk(-do)?$')"
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EA_DIR=~/docker/asterisk; [ -d ~/docker/asterisk-digital-ocean ] && EA_DIR=~/docker/asterisk-digital-ocean
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echo "Container: $CONTAINER Dir: $EA_DIR"
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```
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If `$CONTAINER` prints empty, the container isn't running at all — check
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`docker compose -f $EA_DIR/docker-compose.yml ps` before anything else.
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Re-run this block at the start of every new terminal session, not just
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once — it's cheap and removes the whole class of failure above.
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Check the three services this checklist covers are actually installed:
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```bash
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sudo ./setup.sh --list | grep -E "asterisk|pstn-trunk|sms-inbound|security-dashboard"
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```
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Read your box's own current settings before testing — this file has your
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real DID, tiers reference, ring-group, and rate, regenerated on every
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`pstn-trunk` install/update:
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```bash
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cat $EA_DIR/README-pstn-trunk.md
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```
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## 1. Registration sanity check
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Every extension you're about to test with needs to actually be registered
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before anything else will work.
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```bash
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docker exec -it $CONTAINER asterisk -rx "pjsip show endpoints"
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```
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Each extension you plan to test should show `Avail` (registered), not
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`Unavail`. If a softphone shows registered in its own UI but Asterisk
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disagrees, the extension's transport (UDP vs TLS) is the usual culprit —
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check it via the Security Dashboard's Extensions tab **ⓘ** button on that
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row before going further, not by re-entering credentials.
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## 2. Trunk itself
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```bash
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docker exec -it $CONTAINER asterisk -rx "pjsip show endpoint pstn-trunk"
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```
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Look for `Contact` state, not just that the endpoint exists. An
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IP-authenticated trunk has no registration to show — what matters here is
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that Asterisk considers the endpoint reachable, and that the provider's own
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portal doesn't report the account balance at $0 or the DID unassigned.
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## 3. Permission tiers — set before testing calls
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Every extension starts at **internal** (no PSTN at all) until granted a
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tier. Before testing outbound/inbound, set at least one extension to
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`full` via the Security Dashboard's Extensions tab (or `restricted` with an
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approved number, if that's what you want to test). **Press "Commit Changes
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(Restart Asterisk)"** after — tier/permission edits are written to disk
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immediately but the dialplan has been observed not picking them up until a
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full container restart, not just a reload (see "Dashboard changes may need
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Commit Changes" in `docs/anveo-direct-setup-guide.md`). Skipping this step
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is the single most common reason a "correctly configured" test call still
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fails.
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## 4. Outbound call test
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From a **full**-tier extension's softphone, dial a real US number as
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`1` + area code + number (11 digits — the dialplan matches the full NANP
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pattern, not a bare 10-digit number).
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Watch the live console while dialing:
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```bash
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docker exec -it $CONTAINER asterisk -rvvv
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```
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Expected: the call connects, and a line appears in the call log:
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```bash
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tail -f $EA_DIR/logs/pstn-trunk-calls.log
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```
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**If it fails**, check in this order (matches the actual bugs this system
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has hit — see the "Bugs hit and fixed" section of
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`docs/anveo-direct-setup-guide.md`):
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- `restricted`/`internal` tier dialing a number not on its approved list —
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expected to fail with a busy signal, not a bug.
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- Concurrency cap already hit — check `pstn-limits.conf` or the dashboard's
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displayed caps.
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- Kill-switch tripped (see §7 below) — every PSTN call, in or out, blocked
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regardless of tier.
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- `dialplan show intercom` and count digits by hand against what you
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dialed — a pattern-length mismatch has bitten this exact system before.
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## 5. Inbound call test — shared DID / ring-group
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From any outside phone, dial the trunk's main DID (shown in
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`README-pstn-trunk.md`). Every `full`/`restricted`-tier member of the
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configured ring-group should ring simultaneously for ~20 seconds unless a
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personal DID routes it elsewhere (see §6).
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```bash
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docker exec -it $CONTAINER asterisk -rx "dialplan show from-pstn-trunk"
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```
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**If nothing rings**: confirm the calling number's actual Caller-ID is
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reaching Asterisk correctly — a bare `Hangup()`/no ring with no ntfy denial
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alert either usually means the inbound context isn't loading at all (check
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`asterisk -rx "dialplan show from-pstn-trunk"` returns content, not
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"No such context"). A `Busy(15)` + ntfy denial alert means it loaded fine
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and the caller's number just isn't on a `restricted`-tier member's approved
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list — expected behavior, not a bug.
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## 6. Inbound call test — personal DID
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If you've assigned a personal DID to a specific extension (Security
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Dashboard → PSTN Trunk tab → Personal numbers), dial that number instead of
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the shared trunk DID. Only the owning extension (or every current member,
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if assigned to a Group) should ring — no ring-group fallback. Confirm the
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owner's outbound Caller-ID also shows this DID instead of the shared trunk
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DID when *they* place a call out.
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## 7. Spend-cap kill-switch (only if enabled)
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Don't trigger this for real — it force-hangs-up active calls and blocks
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all PSTN traffic until manually cleared. To confirm it's wired up without
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tripping it:
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```bash
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cat $EA_DIR/config/asterisk/pstn-trunk-killswitch.conf
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systemctl status pstn-trunk-usage.timer 2>/dev/null || crontab -l | grep pstn-trunk-usage
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```
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The timer/cron entry should be active and recent (`journalctl -u
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pstn-trunk-usage.service -n 20` shows it actually ran within the last
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minute). If you do want to confirm the block itself works, lower
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`MAX_MONTHLY_SPEND` temporarily via `sudo ./setup.sh pstn-trunk` (update
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mode) to something you've already exceeded, confirm calling is blocked
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with the priority=urgent ntfy alert, then re-run and clear it (this is
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CLI-only by design — not on the dashboard).
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## 8. International calling (only if you've allowed any countries)
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```bash
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cat $EA_DIR/config/asterisk/pstn-intl-allowed.conf
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```
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Only `full`-tier extensions can dial these regardless of the allow-list.
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Dial `011` + country code + number. If an expiry was set, confirm the
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allow-list actually clears itself after that date (the periodic
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usage-alert script does this, not a separate cron job) — worth checking
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once near the expiry date rather than assuming.
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## 9. Internal SIP messaging (no PSTN/SMS involved)
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From one extension enabled for `messaging=yes` (Security Dashboard →
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Extensions tab → Messaging column) to another, send a native SIP message
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in Sipnetic (or whichever softphone). Should land instantly, no carrier
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involved. Check the log if it doesn't arrive:
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```bash
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tail -f $EA_DIR/logs/sip-messages.log
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```
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An extension with `messaging=no` should have its message silently denied
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(fails closed) — confirm this too, not just the success path.
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## 10. SMS inbound test
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Prerequisite: `sudo ./setup.sh sms-inbound` already run, and the Anveo
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portal's DID → SMS tab → "Forward to URL" configured with the exact string
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the installer printed (see `docs/anveo-direct-setup-guide.md` §8 if this
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hasn't been done yet).
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Text the DID's number from any outside phone. Expected: it lands in
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Sipnetic's message thread for whichever extension/group owns that DID
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within a few seconds — not a push notification, a real message in the
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softphone's own thread.
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```bash
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journalctl -u sms-inbound -f
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```
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Shows sender, recipient, message length, and the AMI exchange — never the
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message body itself. If nothing arrives:
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- Confirm the DID is a **mobile**-class number if you're testing a 2FA/verification
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code specifically — geographic DIDs get silently rejected by many
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services before a message is ever sent (see the setup guide's "Pick the
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right kind of number" section).
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- Confirm short-code SMS is actually enabled on this specific DID (not
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every number in the pool has it by default).
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- Check `pstn-personal-dids.conf`/`pstn-groups.conf` actually has an entry
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for the DID you texted — an unowned DID has nowhere to route to.
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Outbound SMS from this box isn't a working path yet (see the setup guide's
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"What about sending?" section) — don't test for it.
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## 11. Dashboard cross-check
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Open the Security Dashboard's **Calls & Texts** tab and confirm the calls
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and messages from the tests above actually show up there — this is reading
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the same log files tested above, so an empty tab after a successful test
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call usually means the dashboard's `ASTERISK_LOG_DIR`/log paths have
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drifted from where Asterisk is actually writing, not that logging is
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broken. Also check the **Extensions** tab shows correct live status
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(online/offline) for whichever device you tested with.
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## Quick reference — verification commands
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```bash
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docker exec -it $CONTAINER asterisk -rx "pjsip show endpoints"
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docker exec -it $CONTAINER asterisk -rx "pjsip show endpoint pstn-trunk"
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docker exec -it $CONTAINER asterisk -rx "dialplan show intercom"
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docker exec -it $CONTAINER asterisk -rx "dialplan show from-pstn-trunk"
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tail -f $EA_DIR/logs/pstn-trunk-calls.log
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tail -f $EA_DIR/logs/sip-messages.log
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journalctl -u sms-inbound -f # SMS via Anveo's HTTP webhook — the path §10 actually tests
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tail -f $EA_DIR/logs/pstn-sms.log # SMS-over-SIP dialplan path — confirmed NOT offered by Anveo;
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# only relevant if a future provider delivers texts over SIP
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journalctl -u pstn-trunk-usage.service -n 20
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```
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