Add a confirmed-working, step-by-step Anveo Direct + Easy Asterisk setup guide
Documents the actual end-to-end sequence that got a real DID working (account setup, DID ordering, both trunk objects, droplet configuration, permission setup, testing), including the real confirmed values (the $0.00388/min rate, the port-5060 SIP URI forwarding format, the two separate trunk-object types) and a summary of every real bug hit and fixed along the way. Linked from the existing planning-notes doc.
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Claude
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# Anveo Direct + Easy Asterisk — confirmed working setup guide
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This is the exact sequence that got a real Anveo Direct DID working end to
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end (both outbound and inbound) with `asterisk-digital-ocean.sh` +
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`pstn-trunk.sh`, confirmed live on a real droplet. Follow it in order for
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each additional number — steps 1–2 are one-time account setup; steps
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3–7 repeat per DID.
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## 0. Prerequisites
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- `asterisk-digital-ocean.sh` (or `asterisk.sh` for a LAN box) already
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installed and running, with at least one extension configured.
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- This box's public IP address (`curl -4 ifconfig.me`).
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## 1. Anveo Direct account (one-time)
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1. Create an account at `https://www.anveo.com/account.asp?account_type=direct`
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— this is what makes it a **Direct** account (trunk/DID product), not
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the hosted-PBX product.
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2. Fund the account balance directly (not a subscription plan). Bank
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transfers can take several hours to clear.
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3. **Known account-level caps as of this writing**, both separate from
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anything this installer enforces:
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- **Trial account limits**: 2 concurrent outbound calls (CPS: 2),
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$2/30-day spend limit, $2 minimum balance — shown on the Outbound
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Trunks page under "Account Limits." A "request higher limits" link
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sits next to the CallerID Policies section for asking Anveo to lift
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these once funded.
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- **Phone Numbers cap**: "You can manage up to 2 phone numbers. Please
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contact customer support to increase the limit" — a **separate** cap
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from the concurrent-call limit above, shown on the Phone Numbers
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summary page. Anveo indicated (not independently confirmed) that
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removing this requires providing a tax ID (EIN for a business, SSN
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for an individual) — decide for yourself whether that trade-off is
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worth it before planning on more than 2 DIDs.
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4. CallerID policy: USA/Canada/international destinations all require
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CallerID **from a verified/Anveo-owned number** — any DID ordered
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through Anveo satisfies this automatically.
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## 2. Order a DID
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1. Go to the DID ordering tool ("Phone Numbers around the World" /
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similar), pick a number, and select the **Per Minute** rate plan — not
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Prime.
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- **Per Minute**: $0.004/min incoming, 10 dedicated incoming channels
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bundled in, no separate trunk product needed for inbound.
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- **Prime**: free incoming, but requires a separately-purchased "Anveo
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Trunk" (a *different* product — flat monthly fee, $6.50–$17/mo by
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zone, confirmed via the "Anveo Direct Trunk Price List" page). Per
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Minute avoids this fee entirely, which is why it's the one to use.
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2. Confirm: $0.25 one-time setup, $0.15/month recurring (billed the 15th),
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3-month minimum term, pre-paid from account balance.
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## 3. Create the Outbound Service (Call Termination) Trunk
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This only needs to be done **once** — the same trunk carries outbound
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calls for every DID/extension. Skip this step for DIDs 2+.
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1. Outbound Trunks → **Add a new Call Termination Trunk**.
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2. Fill in:
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- **Title**: any label (e.g. `asterisk-do`)
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- **Dialing Prefix**: leave **blank** — confirmed not required (no red
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"required" asterisk on this field, unlike Authorized IP Addresses and
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Call Routing Method). The dialplan here dials the bare number, no
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prefix.
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- **Authorized IP Addresses**: this box's public IP, one per line. This
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is the IP-authentication Anveo uses instead of a SIP password.
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- **Rate Cap**: optional safety ceiling — e.g. `Yes, $1/min` filters out
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any route pricier than that. Domestic US termination runs far below
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this, so it doesn't interfere with normal calls, just blocks getting
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routed onto an absurdly expensive carrier by accident.
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- **Concurrent Calls Limit**: whatever you want as an Anveo-side cap
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(e.g. `Yes, 6`) — note the account's Trial "Outbound Channels: 2"
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limit (see step 1) overrides this until lifted; whichever number is
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lower wins.
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- **Call Routing Method**: leave the default **Custom Least Cost
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Routing Model** and all of its sub-fields (All Prime Routes / cost
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lowest-to-highest / 5 routes / cost-ordered failover / 10 sec
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timeout) exactly as shown — sensible defaults, nothing to tune here.
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3. Save.
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### Finding the real outbound rate
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The trunk's LCR pulls from **"All Prime Routes"** by default — meaning
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your real per-minute cost is whatever's in the **Anveo Direct Prime** rate
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card (Outbound Services/Call Termination page → "Download rates here"
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under the **Prime** column, not Value/Standard/All). Confirmed from that
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CSV: standard US-to-US domestic is a flat **`$0.00388/min`**, billed
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per-second (not rounded to the minute) — this is the generic `USA.`
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catch-all entry (prefix `1`); a few specific area codes have their own
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slightly different override rates, and Virgin Islands is billed
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separately/higher (~$0.02/min) despite looking like a normal US number.
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## 4. Create the SIP Trunk (inbound forwarding) — once per DID
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Anveo has **two unrelated "trunk" concepts** — don't confuse them:
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- The **Outbound Service Trunk** from step 3 handles calls *out*.
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- A separate **"SIP Trunk"** object (Account Options → wherever inbound
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routing objects live) handles calls *in* — this is what actually
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populates a DID's "Destination SIP Trunk" dropdown.
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For **each DID**, create one of these:
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1. **Trunk Name**: any label (e.g. `asterisk-do-inbound`) — can reuse the
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same one for multiple DIDs if you want them all forwarding the same
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way, or make one per DID for clarity.
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2. **Primary**: type **SIP URI**, value:
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```
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$[E164]$@<this box's public IP>:5060
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```
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(port 5060 — Easy Asterisk's default SIP port, confirmed from its own
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vendor source `DEFAULT_SIP_PORT="5060"`). The `$[E164]$` placeholder is
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replaced by Anveo with the DID's own number, digits only, country code
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included, no `+` — e.g. DID `+15551234567` becomes `15551234567@IP:5060`
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in the actual INVITE Anveo sends.
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3. **Failover**: leave blank.
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4. Save.
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5. Go to the DID's own **Call Options** tab → set **Destination SIP
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Trunk** to this new SIP Trunk object (not the Outbound Service Trunk)
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→ Save.
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6. **Optional, recommended once you're adding more DIDs**: Account
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Options → Service Defaults → set **Default Destination Trunk** to this
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SIP Trunk, so future DIDs auto-route here without repeating step 5.
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## 5. Configure the droplet
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```
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sudo ./setup.sh pstn-trunk
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```
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- Existing install → choose **update** (`r`) if you're just changing the
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DID/server, or the CLI will walk fresh prompts if none exists yet.
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- Provider quick-pick: **1) Anveo Direct** — pre-fills `sbc.anveo.com` and
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Anveo's 4 published signaling IPs (only one of which the hostname
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itself resolves to at any given moment; the others are entered as
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"additional known source IPs" so inbound is recognized regardless of
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which one a call actually arrives from).
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- **DID**: 10 digits, no leading `1` (e.g. `5551234567`).
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- **Outbound per-minute rate**: `0.00388` (see step 3's rate-lookup note)
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— pre-filled automatically when Anveo Direct is the selected provider.
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- Concurrent-call caps, ring-group extensions, ntfy topic, spend-cap
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kill-switch thresholds: set to taste (all live-editable later without
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reinstalling).
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- Who can call/be called (permission tiers, approved numbers), internal
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SIP messaging, and personal-number assignment are **not** prompted here
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— set them all in the Security Dashboard's **PSTN Trunk** tab instead
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(install it first if you haven't: `sudo ./setup.sh security-dashboard`).
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Every extension starts at `internal` (no PSTN access) until granted
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there.
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## 6. Grant permissions and (optionally) a personal number
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In the Security Dashboard's PSTN Trunk tab:
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1. Set the tier for each extension that should get PSTN access (`full` or
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`restricted` — `restricted` also needs at least one approved number).
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2. **Personal numbers** card: assign this DID to a specific extension if
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you want inbound calls to it to ring *only* that extension (instead of
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the whole shared ring-group) and its outbound Caller-ID to show this
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DID instead of the shared trunk DID. Enter the DID as **10 digits, no
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leading 1** (matches the same convention as the trunk DID above) —
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the dashboard's own input validates against exactly that format.
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## 7. Test
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- **Outbound**: from a full/restricted-tier extension in Sipnetic, dial
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`1` + area code + number (11 digits total — the dialplan matches the
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full NANP pattern, not a bare 10-digit number without the leading `1`).
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- **Inbound**: from any outside phone, dial the DID's real number. If a
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personal number was assigned, only that extension should ring; if not,
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the whole configured ring-group rings simultaneously until someone
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answers or 20 seconds pass.
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- Watch the live console while testing either direction:
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```
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docker exec -it easy-asterisk-do asterisk -rvvv
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```
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## Bugs hit and fixed along the way (informational — already fixed)
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These were all real, confirmed-live bugs in earlier versions of this
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installer, not configuration mistakes — listed here for context on what
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"just works" now that didn't before:
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- **NANP dial pattern was one digit short** (`_1NXXNXXXXX` / `_NXXNXXXXX`,
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10/9 characters instead of the correct 11/10) — Asterisk's exact-length
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pattern matching never matched a real number at all.
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- **`${CHANNEL(peername)}`** isn't a valid channel-function item on this
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Asterisk build — broke the caller-extension lookup used for every
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permission check.
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- **`${EXTEN}` corruption after `Goto` to a named extension** — every
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`Goto(somelabel,1)` to a *different* extension (not a same-extension
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priority label) resets `${EXTEN}` to that label's own name, silently
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breaking the approved-numbers check, the international country check,
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and the final `Dial()` itself. Fixed by capturing the real dialed
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number into its own variable before any such jump.
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- **Duplicate ring-group extensions** (the same extension listed twice in
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`RING_EXTS`) generated duplicate dialplan priority labels, causing an
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infinite loop that never reached the ring step at all. Now deduped
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automatically.
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- **Personal-DID lookup format mismatch** — DIDs are stored 10-digit, but
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Anveo delivers the called number as 11-digit E.164 in `${EXTEN}`; the
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lookup now normalizes to 10 digits first.
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- **Easy Asterisk only regenerates `pjsip.conf`/`extensions.conf` if they
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don't already exist** — a box that already has devices configured (the
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normal case) never picks up a vendor-script `#include` patch on a plain
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restart. Both the PSTN trunk and internal-messaging installers now also
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patch the live config files directly, not just the generator source.
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- **A stale/malformed `.pstn-trunk.env`** could crash the entire installer
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under `setup.sh`'s `set -u` on `source`. Now guarded and falls back to
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safe defaults, erroring clearly only if genuinely required fields are
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missing.
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## Still open
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- The 2-phone-number account cap (see step 1) blocks getting all 4
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planned numbers until either Anveo lifts it or a tax ID is provided —
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tabled for now.
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- The interactive CLI walkthrough (`pstn-trunk.sh` actually prompting
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through account setup step by step, not just this static doc) hasn't
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been built yet — this guide is the reference for building that once
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there's appetite for it.
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@@ -3,6 +3,12 @@
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Research and decisions from a design discussion, saved here so the work can
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be picked up in a fresh chat without re-deriving the background.
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**For the confirmed-working, step-by-step Anveo Direct account + droplet
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setup (real values, real bugs hit and fixed), see
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`docs/anveo-direct-setup-guide.md` instead of re-deriving it from this
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file** — this doc is the design/decision log; that one is the clean
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how-to.
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**Implemented** — see `services/pstn-trunk.sh` (run `sudo ./setup.sh
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pstn-trunk` after `asterisk-digital-ocean` **or** `asterisk` (home/LAN) is
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installed — both are supported, see the file for the static-IP caveat on the
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