ubuntu-post-install/docs/anveo-direct-setup-guide.md

Anveo Direct + Easy Asterisk — confirmed working setup guide

This is the exact sequence that got a real Anveo Direct DID working end to end (both outbound and inbound) with asterisk-digital-ocean.sh + pstn-trunk.sh, confirmed live on a real droplet. Follow it in order for each additional number — steps 1–2 are one-time account setup; steps 3–7 repeat per DID.

0. Prerequisites

• asterisk-digital-ocean.sh (or asterisk.sh for a LAN box) already installed and running, with at least one extension configured.
• This box's public IP address (curl -4 ifconfig.me).

1. Anveo Direct account (one-time)

1. Create an account at https://www.anveo.com/account.asp?account_type=direct — this is what makes it a Direct account (trunk/DID product), not the hosted-PBX product.
2. Fund the account balance directly (not a subscription plan). Bank transfers can take several hours to clear.
3. Known account-level caps as of this writing, both separate from anything this installer enforces:
   • Trial account limits: 2 concurrent outbound calls (CPS: 2), $2/30-day spend limit, $2 minimum balance — shown on the Outbound Trunks page under "Account Limits." A "request higher limits" link sits next to the CallerID Policies section for asking Anveo to lift these once funded.
   • Phone Numbers cap: "You can manage up to 2 phone numbers. Please contact customer support to increase the limit" — a separate cap from the concurrent-call limit above, shown on the Phone Numbers summary page. Anveo indicated (not independently confirmed) that removing this requires providing a tax ID (EIN for a business, SSN for an individual) — decide for yourself whether that trade-off is worth it before planning on more than 2 DIDs.
4. CallerID policy: USA/Canada/international destinations all require CallerID from a verified/Anveo-owned number — any DID ordered through Anveo satisfies this automatically.

2. Order a DID

1. Go to the DID ordering tool ("Phone Numbers around the World" / similar), pick a number, and select the Per Minute rate plan — not Prime.
   • Per Minute: $0.004/min incoming, 10 dedicated incoming channels bundled in, no separate trunk product needed for inbound.
   • Prime: free incoming, but requires a separately-purchased "Anveo Trunk" (a different product — flat monthly fee, $6.50–$17/mo by zone, confirmed via the "Anveo Direct Trunk Price List" page). Per Minute avoids this fee entirely, which is why it's the one to use.
2. Confirm: $0.25 one-time setup, $0.15/month recurring (billed the 15th), 3-month minimum term, pre-paid from account balance.

3. Create the Outbound Service (Call Termination) Trunk

This only needs to be done once — the same trunk carries outbound calls for every DID/extension. Skip this step for DIDs 2+.

1. Outbound Trunks → Add a new Call Termination Trunk.
2. Fill in:
   • Title: any label (e.g. asterisk-do)
   • Dialing Prefix: leave blank — confirmed not required (no red "required" asterisk on this field, unlike Authorized IP Addresses and Call Routing Method). The dialplan here dials the bare number, no prefix.
   • Authorized IP Addresses: this box's public IP, one per line. This is the IP-authentication Anveo uses instead of a SIP password.
   • Rate Cap: optional safety ceiling — e.g. Yes, $1/min filters out any route pricier than that. Domestic US termination runs far below this, so it doesn't interfere with normal calls, just blocks getting routed onto an absurdly expensive carrier by accident.
   • Concurrent Calls Limit: whatever you want as an Anveo-side cap (e.g. Yes, 6) — note the account's Trial "Outbound Channels: 2" limit (see step 1) overrides this until lifted; whichever number is lower wins.
   • Call Routing Method: leave the default Custom Least Cost Routing Model and all of its sub-fields (All Prime Routes / cost lowest-to-highest / 5 routes / cost-ordered failover / 10 sec timeout) exactly as shown — sensible defaults, nothing to tune here.
3. Save.

Finding the real outbound rate

The trunk's LCR pulls from "All Prime Routes" by default — meaning your real per-minute cost is whatever's in the Anveo Direct Prime rate card (Outbound Services/Call Termination page → "Download rates here" under the Prime column, not Value/Standard/All). Confirmed from that CSV: standard US-to-US domestic is a flat $0.00388/min, billed per-second (not rounded to the minute) — this is the generic USA. catch-all entry (prefix 1); a few specific area codes have their own slightly different override rates, and Virgin Islands is billed separately/higher (~$0.02/min) despite looking like a normal US number.

4. Create the SIP Trunk (inbound forwarding) — once per DID

Anveo has two unrelated "trunk" concepts — don't confuse them:

• The Outbound Service Trunk from step 3 handles calls out.
• A separate "SIP Trunk" object (Account Options → wherever inbound routing objects live) handles calls in — this is what actually populates a DID's "Destination SIP Trunk" dropdown.

For each DID, create one of these:

1. Trunk Name: any label (e.g. asterisk-do-inbound) — can reuse the same one for multiple DIDs if you want them all forwarding the same way, or make one per DID for clarity.
2. Primary: type SIP URI, value:

   $[E164]$@<this box's public IP>:5060
   

   (port 5060 — Easy Asterisk's default SIP port, confirmed from its own vendor source DEFAULT_SIP_PORT="5060"). The $[E164]$ placeholder is replaced by Anveo with the DID's own number, digits only, country code included, no + — e.g. DID +15551234567 becomes 15551234567@IP:5060 in the actual INVITE Anveo sends.
3. Failover: leave blank.
4. Save.
5. Go to the DID's own Call Options tab → set Destination SIP Trunk to this new SIP Trunk object (not the Outbound Service Trunk) → Save.
6. Optional, recommended once you're adding more DIDs: Account Options → Service Defaults → set Default Destination Trunk to this SIP Trunk, so future DIDs auto-route here without repeating step 5.

5. Configure the droplet

sudo ./setup.sh pstn-trunk

• Existing install → choose update (r) if you're just changing the DID/server, or the CLI will walk fresh prompts if none exists yet.
• Provider quick-pick: 1) Anveo Direct — pre-fills sbc.anveo.com and Anveo's 4 published signaling IPs (only one of which the hostname itself resolves to at any given moment; the others are entered as "additional known source IPs" so inbound is recognized regardless of which one a call actually arrives from).
• DID: 10 digits, no leading 1 (e.g. 5551234567).
• Outbound per-minute rate: 0.00388 (see step 3's rate-lookup note) — pre-filled automatically when Anveo Direct is the selected provider.
• Concurrent-call caps, ring-group extensions, ntfy topic, spend-cap kill-switch thresholds: set to taste (all live-editable later without reinstalling).
• Who can call/be called (permission tiers, approved numbers), internal SIP messaging, and personal-number assignment are not prompted here — set them all in the Security Dashboard's PSTN Trunk tab instead (install it first if you haven't: sudo ./setup.sh security-dashboard). Every extension starts at internal (no PSTN access) until granted there.

6. Grant permissions and (optionally) a personal number

In the Security Dashboard's PSTN Trunk tab:

1. Set the tier for each extension that should get PSTN access (full or restricted — restricted also needs at least one approved number).
2. Personal numbers card: assign this DID to a specific extension if you want inbound calls to it to ring only that extension (instead of the whole shared ring-group) and its outbound Caller-ID to show this DID instead of the shared trunk DID. Enter the DID as 10 digits, no leading 1 (matches the same convention as the trunk DID above) — the dashboard's own input validates against exactly that format.

7. Test

• Outbound: from a full/restricted-tier extension in Sipnetic, dial 1 + area code + number (11 digits total — the dialplan matches the full NANP pattern, not a bare 10-digit number without the leading 1).
• Inbound: from any outside phone, dial the DID's real number. If a personal number was assigned, only that extension should ring; if not, the whole configured ring-group rings simultaneously until someone answers or 20 seconds pass.
• Watch the live console while testing either direction:

  docker exec -it easy-asterisk-do asterisk -rvvv
  

Bugs hit and fixed along the way (informational — already fixed)

These were all real, confirmed-live bugs in earlier versions of this installer, not configuration mistakes — listed here for context on what "just works" now that didn't before:

• NANP dial pattern was one digit short (_1NXXNXXXXX / _NXXNXXXXX, 10/9 characters instead of the correct 11/10) — Asterisk's exact-length pattern matching never matched a real number at all.
• ${CHANNEL(peername)} isn't a valid channel-function item on this Asterisk build — broke the caller-extension lookup used for every permission check.
• ${EXTEN} corruption after Goto to a named extension — every Goto(somelabel,1) to a different extension (not a same-extension priority label) resets ${EXTEN} to that label's own name, silently breaking the approved-numbers check, the international country check, and the final Dial() itself. Fixed by capturing the real dialed number into its own variable before any such jump.
• Duplicate ring-group extensions (the same extension listed twice in RING_EXTS) generated duplicate dialplan priority labels, causing an infinite loop that never reached the ring step at all. Now deduped automatically.
• Personal-DID lookup format mismatch — DIDs are stored 10-digit, but Anveo delivers the called number as 11-digit E.164 in ${EXTEN}; the lookup now normalizes to 10 digits first.
• Easy Asterisk only regenerates pjsip.conf/extensions.conf if they don't already exist — a box that already has devices configured (the normal case) never picks up a vendor-script #include patch on a plain restart. Both the PSTN trunk and internal-messaging installers now also patch the live config files directly, not just the generator source.
• A stale/malformed .pstn-trunk.env could crash the entire installer under setup.sh's set -u on source. Now guarded and falls back to safe defaults, erroring clearly only if genuinely required fields are missing.

Still open

• The 2-phone-number account cap (see step 1) blocks getting all 4 planned numbers until either Anveo lifts it or a tax ID is provided — tabled for now.
• The interactive CLI walkthrough (pstn-trunk.sh actually prompting through account setup step by step, not just this static doc) hasn't been built yet — this guide is the reference for building that once there's appetite for it.