Add PSTN/DID/SMS end-to-end test checklist
docs/pstn-calling-voipms-plan.md (design log) and docs/anveo-direct-setup-guide.md (account + droplet setup) already cover getting a trunk/DID/SMS working from scratch, but neither is a quick top-to-bottom checklist for verifying an already-installed setup still works — registration, trunk reachability, tiers, outbound/inbound calls (shared DID and personal DID), the spend-cap kill-switch, international calling, internal SIP messaging, and SMS inbound, in order, with what to check when each step fails. Pulls known gotchas (Commit Changes required after dashboard tier edits, mobile vs geographic DIDs for verification codes, the SIP-based SMS path Anveo doesn't actually offer) from the existing docs so they're not missed mid-test. Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com> Claude-Session: https://claude.ai/code/session_01H4k6J1qXXyYxhGEgnJaMvn
This commit is contained in:
Claude
@@ -0,0 +1,242 @@
|
||||
# PSTN + DID + SMS — end-to-end test checklist
|
||||
|
||||
For verifying an **already-installed** Asterisk + PSTN trunk + SMS setup
|
||||
(droplet or home/LAN) actually works, top to bottom. Not a setup guide —
|
||||
see `docs/anveo-direct-setup-guide.md` for account/droplet setup from
|
||||
scratch, and `docs/pstn-calling-voipms-plan.md` for the design/architecture
|
||||
background. This doc assumes both are already done and just walks through
|
||||
proving it all still works, in order, with what to check when a step fails.
|
||||
|
||||
Run through this after any of: first go-live, a provider/DID change, an
|
||||
Asterisk or PSTN trunk reinstall, or just periodically to catch drift (a
|
||||
provider-side change, an expired international allow-list, a forgotten
|
||||
kill-switch trip).
|
||||
|
||||
## 0. Before you start
|
||||
|
||||
Confirm which layout this box uses — commands below need the right
|
||||
container name and directory:
|
||||
|
||||
```bash
|
||||
docker ps --format '{{.Names}}' | grep -i easy-asterisk
|
||||
# easy-asterisk -> ~/docker/asterisk
|
||||
# easy-asterisk-do -> ~/docker/asterisk-digital-ocean (pre-merge droplet)
|
||||
```
|
||||
|
||||
Set these once and reuse them in every command below:
|
||||
|
||||
```bash
|
||||
CONTAINER=easy-asterisk # or easy-asterisk-do
|
||||
EA_DIR=~/docker/asterisk # or ~/docker/asterisk-digital-ocean
|
||||
```
|
||||
|
||||
Check the three services this checklist covers are actually installed:
|
||||
|
||||
```bash
|
||||
sudo ./setup.sh --list | grep -E "asterisk|pstn-trunk|sms-inbound|security-dashboard"
|
||||
```
|
||||
|
||||
Read your box's own current settings before testing — this file has your
|
||||
real DID, tiers reference, ring-group, and rate, regenerated on every
|
||||
`pstn-trunk` install/update:
|
||||
|
||||
```bash
|
||||
cat $EA_DIR/README-pstn-trunk.md
|
||||
```
|
||||
|
||||
## 1. Registration sanity check
|
||||
|
||||
Every extension you're about to test with needs to actually be registered
|
||||
before anything else will work.
|
||||
|
||||
```bash
|
||||
docker exec -it $CONTAINER asterisk -rx "pjsip show endpoints"
|
||||
```
|
||||
|
||||
Each extension you plan to test should show `Avail` (registered), not
|
||||
`Unavail`. If a softphone shows registered in its own UI but Asterisk
|
||||
disagrees, the extension's transport (UDP vs TLS) is the usual culprit —
|
||||
check it via the Security Dashboard's Extensions tab **ⓘ** button on that
|
||||
row before going further, not by re-entering credentials.
|
||||
|
||||
## 2. Trunk itself
|
||||
|
||||
```bash
|
||||
docker exec -it $CONTAINER asterisk -rx "pjsip show endpoint pstn-trunk"
|
||||
```
|
||||
|
||||
Look for `Contact` state, not just that the endpoint exists. An
|
||||
IP-authenticated trunk has no registration to show — what matters here is
|
||||
that Asterisk considers the endpoint reachable, and that the provider's own
|
||||
portal doesn't report the account balance at $0 or the DID unassigned.
|
||||
|
||||
## 3. Permission tiers — set before testing calls
|
||||
|
||||
Every extension starts at **internal** (no PSTN at all) until granted a
|
||||
tier. Before testing outbound/inbound, set at least one extension to
|
||||
`full` via the Security Dashboard's Extensions tab (or `restricted` with an
|
||||
approved number, if that's what you want to test). **Press "Commit Changes
|
||||
(Restart Asterisk)"** after — tier/permission edits are written to disk
|
||||
immediately but the dialplan has been observed not picking them up until a
|
||||
full container restart, not just a reload (see "Dashboard changes may need
|
||||
Commit Changes" in `docs/anveo-direct-setup-guide.md`). Skipping this step
|
||||
is the single most common reason a "correctly configured" test call still
|
||||
fails.
|
||||
|
||||
## 4. Outbound call test
|
||||
|
||||
From a **full**-tier extension's softphone, dial a real US number as
|
||||
`1` + area code + number (11 digits — the dialplan matches the full NANP
|
||||
pattern, not a bare 10-digit number).
|
||||
|
||||
Watch the live console while dialing:
|
||||
|
||||
```bash
|
||||
docker exec -it $CONTAINER asterisk -rvvv
|
||||
```
|
||||
|
||||
Expected: the call connects, and a line appears in the call log:
|
||||
|
||||
```bash
|
||||
tail -f $EA_DIR/logs/pstn-trunk-calls.log
|
||||
```
|
||||
|
||||
**If it fails**, check in this order (matches the actual bugs this system
|
||||
has hit — see the "Bugs hit and fixed" section of
|
||||
`docs/anveo-direct-setup-guide.md`):
|
||||
- `restricted`/`internal` tier dialing a number not on its approved list —
|
||||
expected to fail with a busy signal, not a bug.
|
||||
- Concurrency cap already hit — check `pstn-limits.conf` or the dashboard's
|
||||
displayed caps.
|
||||
- Kill-switch tripped (see §7 below) — every PSTN call, in or out, blocked
|
||||
regardless of tier.
|
||||
- `dialplan show intercom` and count digits by hand against what you
|
||||
dialed — a pattern-length mismatch has bitten this exact system before.
|
||||
|
||||
## 5. Inbound call test — shared DID / ring-group
|
||||
|
||||
From any outside phone, dial the trunk's main DID (shown in
|
||||
`README-pstn-trunk.md`). Every `full`/`restricted`-tier member of the
|
||||
configured ring-group should ring simultaneously for ~20 seconds unless a
|
||||
personal DID routes it elsewhere (see §6).
|
||||
|
||||
```bash
|
||||
docker exec -it $CONTAINER asterisk -rx "dialplan show from-pstn-trunk"
|
||||
```
|
||||
|
||||
**If nothing rings**: confirm the calling number's actual Caller-ID is
|
||||
reaching Asterisk correctly — a bare `Hangup()`/no ring with no ntfy denial
|
||||
alert either usually means the inbound context isn't loading at all (check
|
||||
`asterisk -rx "dialplan show from-pstn-trunk"` returns content, not
|
||||
"No such context"). A `Busy(15)` + ntfy denial alert means it loaded fine
|
||||
and the caller's number just isn't on a `restricted`-tier member's approved
|
||||
list — expected behavior, not a bug.
|
||||
|
||||
## 6. Inbound call test — personal DID
|
||||
|
||||
If you've assigned a personal DID to a specific extension (Security
|
||||
Dashboard → PSTN Trunk tab → Personal numbers), dial that number instead of
|
||||
the shared trunk DID. Only the owning extension (or every current member,
|
||||
if assigned to a Group) should ring — no ring-group fallback. Confirm the
|
||||
owner's outbound Caller-ID also shows this DID instead of the shared trunk
|
||||
DID when *they* place a call out.
|
||||
|
||||
## 7. Spend-cap kill-switch (only if enabled)
|
||||
|
||||
Don't trigger this for real — it force-hangs-up active calls and blocks
|
||||
all PSTN traffic until manually cleared. To confirm it's wired up without
|
||||
tripping it:
|
||||
|
||||
```bash
|
||||
cat $EA_DIR/config/asterisk/pstn-trunk-killswitch.conf
|
||||
systemctl status pstn-trunk-usage.timer 2>/dev/null || crontab -l | grep pstn-trunk-usage
|
||||
```
|
||||
|
||||
The timer/cron entry should be active and recent (`journalctl -u
|
||||
pstn-trunk-usage.service -n 20` shows it actually ran within the last
|
||||
minute). If you do want to confirm the block itself works, lower
|
||||
`MAX_MONTHLY_SPEND` temporarily via `sudo ./setup.sh pstn-trunk` (update
|
||||
mode) to something you've already exceeded, confirm calling is blocked
|
||||
with the priority=urgent ntfy alert, then re-run and clear it (this is
|
||||
CLI-only by design — not on the dashboard).
|
||||
|
||||
## 8. International calling (only if you've allowed any countries)
|
||||
|
||||
```bash
|
||||
cat $EA_DIR/config/asterisk/pstn-intl-allowed.conf
|
||||
```
|
||||
|
||||
Only `full`-tier extensions can dial these regardless of the allow-list.
|
||||
Dial `011` + country code + number. If an expiry was set, confirm the
|
||||
allow-list actually clears itself after that date (the periodic
|
||||
usage-alert script does this, not a separate cron job) — worth checking
|
||||
once near the expiry date rather than assuming.
|
||||
|
||||
## 9. Internal SIP messaging (no PSTN/SMS involved)
|
||||
|
||||
From one extension enabled for `messaging=yes` (Security Dashboard →
|
||||
Extensions tab → Messaging column) to another, send a native SIP message
|
||||
in Sipnetic (or whichever softphone). Should land instantly, no carrier
|
||||
involved. Check the log if it doesn't arrive:
|
||||
|
||||
```bash
|
||||
tail -f $EA_DIR/logs/sip-messages.log
|
||||
```
|
||||
|
||||
An extension with `messaging=no` should have its message silently denied
|
||||
(fails closed) — confirm this too, not just the success path.
|
||||
|
||||
## 10. SMS inbound test
|
||||
|
||||
Prerequisite: `sudo ./setup.sh sms-inbound` already run, and the Anveo
|
||||
portal's DID → SMS tab → "Forward to URL" configured with the exact string
|
||||
the installer printed (see `docs/anveo-direct-setup-guide.md` §8 if this
|
||||
hasn't been done yet).
|
||||
|
||||
Text the DID's number from any outside phone. Expected: it lands in
|
||||
Sipnetic's message thread for whichever extension/group owns that DID
|
||||
within a few seconds — not a push notification, a real message in the
|
||||
softphone's own thread.
|
||||
|
||||
```bash
|
||||
journalctl -u sms-inbound -f
|
||||
```
|
||||
|
||||
Shows sender, recipient, message length, and the AMI exchange — never the
|
||||
message body itself. If nothing arrives:
|
||||
- Confirm the DID is a **mobile**-class number if you're testing a 2FA/verification
|
||||
code specifically — geographic DIDs get silently rejected by many
|
||||
services before a message is ever sent (see the setup guide's "Pick the
|
||||
right kind of number" section).
|
||||
- Confirm short-code SMS is actually enabled on this specific DID (not
|
||||
every number in the pool has it by default).
|
||||
- Check `pstn-personal-dids.conf`/`pstn-groups.conf` actually has an entry
|
||||
for the DID you texted — an unowned DID has nowhere to route to.
|
||||
|
||||
Outbound SMS from this box isn't a working path yet (see the setup guide's
|
||||
"What about sending?" section) — don't test for it.
|
||||
|
||||
## 11. Dashboard cross-check
|
||||
|
||||
Open the Security Dashboard's **Calls & Texts** tab and confirm the calls
|
||||
and messages from the tests above actually show up there — this is reading
|
||||
the same log files tested above, so an empty tab after a successful test
|
||||
call usually means the dashboard's `ASTERISK_LOG_DIR`/log paths have
|
||||
drifted from where Asterisk is actually writing, not that logging is
|
||||
broken. Also check the **Extensions** tab shows correct live status
|
||||
(online/offline) for whichever device you tested with.
|
||||
|
||||
## Quick reference — verification commands
|
||||
|
||||
```bash
|
||||
docker exec -it $CONTAINER asterisk -rx "pjsip show endpoints"
|
||||
docker exec -it $CONTAINER asterisk -rx "pjsip show endpoint pstn-trunk"
|
||||
docker exec -it $CONTAINER asterisk -rx "dialplan show intercom"
|
||||
docker exec -it $CONTAINER asterisk -rx "dialplan show from-pstn-trunk"
|
||||
tail -f $EA_DIR/logs/pstn-trunk-calls.log
|
||||
tail -f $EA_DIR/logs/sip-messages.log
|
||||
journalctl -u sms-inbound -f # SMS via Anveo's HTTP webhook — the path §10 actually tests
|
||||
tail -f $EA_DIR/logs/pstn-sms.log # SMS-over-SIP dialplan path — confirmed NOT offered by Anveo;
|
||||
# only relevant if a future provider delivers texts over SIP
|
||||
journalctl -u pstn-trunk-usage.service -n 20
|
||||
```
|
||||
Reference in New Issue
Block a user