Replaces the static wall of warnings with a real 5-step guided walkthrough
(account/funding, DID ordering, both trunk objects, confirmed rate),
pausing for confirmation between each step and showing this box's actual
public IP inline for direct copy-paste into Anveo's Authorized IP
Addresses and SIP Trunk forwarding fields. Built directly from
docs/anveo-direct-setup-guide.md now that the whole flow is confirmed
working end-to-end, so a fresh run shouldn't need the trial-and-error
session that guide was written from.
Confirmed live: PSTN_PERSONAL_OWNER came back empty despite a real
assignment existing, because the Security Dashboard stores personal DIDs
as 10 digits (PERSONAL_DID_RE = ^\d{10}$) while Anveo's inbound INVITE
delivers the called number as 11-digit E.164 (with leading 1) in ${EXTEN}
- confirmed via PSTN_DID_CALLED=15557776655 in the live test. AST_CONFIG()
looking up an 11-digit key never found the 10-digit section.
Normalizes to 10 digits before the lookup (strips a leading digit only
when the string is actually 11 characters, so this doesn't misfire against
a provider that already sends 10). Also fixed the dashboard's own DID
input placeholder, which contradicted its own 10-digit validation error.
Confirmed live: a ring-group list with a repeated extension ("201 301 313
312 313") generated duplicate ring<ext>/skip<ext> priority labels in the
same [from-pstn-trunk] extension. Asterisk doesn't error on this - it
silently resolves Goto() to the wrong occurrence and loops between the two
blocks forever, so the call never reaches the actual Dial() and nothing
ever rings, with no error to point at the cause. Dedupes (first-occurrence
order preserved) regardless of how the duplicate got in there.
Confirmed live: Dial() actually placed the call to "PJSIP/pstn_check_busy@
pstn-trunk" instead of the dialed number. Root cause is systemic, not one
line — Goto(label,1) to a *different named extension* (as opposed to a
same-extension priority label, which _pstn_ring_member_block already knew
to avoid) resets ${EXTEN} to that label's own name. Every helper extension
downstream of a Goto (pstn_check_allow_out, pstn_check_busy,
pstn_intl_check_country, pstn_intl_country_denied, pstn_personal_inbound)
was reading ${EXTEN} expecting the originally-dialed number and getting
its own extension name instead.
This wasn't just cosmetic: pstn_check_allow_out's REGEX approved-number
check compared against the label name too, meaning restricted-tier
extensions could never successfully place a call regardless of whether
the number was actually approved.
Fixed by capturing the real dialed number into PSTN_DIALED (outbound) /
PSTN_DID_CALLED (inbound) at each entry point, before any Goto, and using
that instead of ${EXTEN} in every downstream extension.
Confirmed live: "Unknown or unavailable item requested: 'peername'" left
PSTN_CALLER empty, which made AST_CONFIG() error ("requires a category")
and PSTN_TIER come back empty too — every call fell through to "no PSTN
permission" and got busy-signaled regardless of the extension's actual
tier. Replaced with CUT(CHANNEL,/,2) then CUT(...,-,1), extracting the
endpoint name from the channel name itself (e.g. "PJSIP/201-00000006" ->
"201") — a plain string operation with no dependency on which channel
function items this particular Asterisk build supports.
_1NXXNXXXXX / _NXXNXXXXX were one digit short (10/9 characters instead of
the correct 11/10 for a full NANP number: 1 + area code(3) + exchange(3) +
subscriber(4)). Asterisk's exact-length pattern matching never matched a
real dialed number as a result — confirmed live via "extension not found
in context 'intercom'" on every outbound test call, traced by dumping
`dialplan show intercom` and counting characters against the actual
pattern. This was the real root cause underneath everything else fixed
along the way (stale settings file, the #include not reaching the live
config) — those were real bugs too, but this one was still there
underneath all of them.
setup.sh runs with set -uo pipefail, and source-ing a settings file with
any stray unexpanded/unset variable reference (confirmed live on an older
deployed copy) was fatal to the entire script, not just that one prompt.
Wraps the source in set +u/-u and falls back to sane defaults for anything
missing, but still hard-errors with a clear message if TRUNK_SERVER/
TRUNK_DID themselves are missing, rather than silently writing a broken
trunk config.
Removed the full/restricted-extension, approved-numbers pool/whiptail,
messaging-extensions, and personal-DID-assignment prompts from the
installer. All four are already live-editable, no-restart-needed settings
the Security Dashboard's PSTN Trunk tab manages end to end — the CLI wall
of prompts (that then needed a full reinstall to change) just duplicated
that with more friction. Every extension now starts at internal tier (no
PSTN, no messaging) until granted via the dashboard. Kept: provider/DID
setup, concurrency caps, and the inbound ring-group (basic trunk wiring,
not a permission). Also updated the generated README and CLI summary to
match, and refreshed the stale "known gap" messaging section that predated
the dialplan enforcement built earlier this session.
Confirmed live via a real failed test call ("extension not found in context
'intercom'"): Easy Asterisk's entrypoint only regenerates pjsip.conf/
extensions.conf if they don't already exist. Patching the vendor generator
functions (as both pstn-trunk.sh and the earlier SIP MESSAGE work did) only
takes effect on a future full regeneration — a box that already has devices
configured, the normal case, never triggers one on a plain restart, so the
#include lines never actually reached the live files despite the generator
patch succeeding.
Both installers now also patch the live pjsip.conf/extensions.conf directly
(same anchors, idempotent) and force a reload, so the trunk/messaging
dialplan actually takes effect immediately regardless of whether Easy
Asterisk ever regenerates its config on its own.
0.00388/min for standard US-to-US domestic, confirmed against Anveo's own
Prime rate card CSV (the route set the outbound trunk's LCR config actually
pulls from) rather than the earlier ~0.001 ballpark guess. Defaults to it
automatically when the Anveo Direct quick-pick was chosen.
Restricted extensions previously required retyping the full 11-digit
number list from scratch at each extension's own prompt. Now offers
entering a pool once, then a whiptail checklist per extension to pick a
subset (falls back to typing numbers directly, or 'all' for the whole
pool, when whiptail isn't available or the run is unattended).
User feedback: reading the tier list (internal/restricted/full) right before
"Full-PSTN extensions:" read as ambiguous about whether to type the tier
word or extension numbers. Reworded both prompts to say "extension NUMBERS"
explicitly and added inline examples (e.g. '999 213').
The dialing-prefix ambiguity is resolved: Anveo's own trunk edit screen has
no required-field marker on Dialing Prefix, confirming it's optional. Also
documents the separate Outbound Service Trunk vs. inbound SIP Trunk
distinction discovered while walking through a real account setup.
Messaging has no dependency on a PSTN trunk existing (no cost, no
carrier, no DID), but the whole PSTN Trunk tab - including the messaging
checkbox added last commit - was hidden behind pstn_installed(), which
only becomes true once services/pstn-trunk.sh's dialplan is actually
wired in. That meant enabling messaging required going through full SIP
trunk/provider setup first for no real reason.
Added a standalone "Internal SIP messaging" card that's always visible in
the tab regardless of trunk status, backed by a new write_messaging()
that only touches the messaging key (leaving tier/allowed_numbers/
personal_did untouched) and creates pstn-permissions.conf from scratch if
it doesn't exist yet. Confirmed the systemd unit's ReadWritePaths and the
group/chmod grants already covered this - both are set up whenever a base
Asterisk install is detected, independent of pstn-trunk - so no
permission-layer changes were needed, only the dashboard's own artificial
UI gate.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
get_all_permissions()/write_permission() now handle messaging alongside
tier/allowed_numbers as one save action, and the Security Dashboard's PSTN
Trunk table gets a Messaging checkbox column - no more needing to re-run
pstn-trunk.sh's CLI just to change who can use internal SIP texting,
matching how tier/allowed_numbers/personal_did already worked.
Tested that messaging correctly survives tier changes and personal-DID
assignment/removal on the same extension (independent axes, as intended).
Still explicitly not done, and said so in both READMEs rather than
implying otherwise now that there's a nice UI for it: the actual SIP
MESSAGE dialplan wiring that would make Asterisk enforce this flag. That
gap hasn't changed - only the permission storage/UI layer around it has.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
Prompts now start with a 1/2/3 choice (Anveo Direct / VoIP.ms / manual)
that pre-fills already-known values - Anveo Direct's sbc.anveo.com
hostname and its 4 published signaling IPs, so they don't need retyping
from memory - plus provider-specific account-setup reminders. Every
value stays editable at each subsequent prompt; this only changes
defaults, not behavior, so existing manual-entry users see no change.
Flagged rather than silently trusted: Anveo Direct's own Outbound Trunks
page documents dialing as [PREFIX]PHONENUMBER@sbc.anveo.com, which
contradicts an earlier no-prefix-needed finding from their FAQ. The
dialplan still dials the bare number (matching the older finding) - the
quick-pick prints a loud warning to verify the Prefix field can be left
blank before relying on this, rather than guessing and hardcoding prefix
handling without live confirmation either way.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
Multiple DIDs can now share one trunk/account, each assigned to a specific
extension - additive to the existing shared trunk DID/ring-group, which
keeps working unchanged for everyone regardless of what's assigned here.
- New pstn-personal-dids.conf (DID -> owner extension), read live by the
dialplan for inbound routing: a call to a personal DID routes straight
to its owner, checked against the owner's own tier/approved-numbers, no
ring-group fallback since it's that extension's own line.
- New personal_did= field per extension in pstn-permissions.conf: the
outbound Caller-ID override, used by pstn_check_busy (the shared exit
point for both domestic and international dialing) instead of the
shared trunk DID when the calling extension has one assigned.
- Both files kept in sync automatically by one write path - CLI prompt at
install/update, or a new "Personal numbers" card in the Security
Dashboard's PSTN Trunk tab - rather than requiring hand-editing both
consistently. Reassigning a DID or giving an extension a new one cleanly
drops the stale side of the old mapping.
- Assigning a DID to an internal-tier extension is accepted but warned
about (won't ring anyone until also granted full/restricted tier),
matching this repo's permissive-with-warnings style rather than
blocking the action outright.
Also fixed a real bug found while building this: security-dashboard.sh's
write_permission() did cp.remove_section(ext) whenever tier was set to
"internal", silently discarding any messaging=yes or personal_did=
already set on that extension. Now removes only the tier/allowed_numbers
keys, dropping the section only once nothing else remains in it.
Separately, de-anchored pstn-trunk.sh's install prompts from VoIP.ms as
the implicit default (provider name default, server-hostname example
wording, rate-prompt wording) now that Anveo Direct is an equally
confirmed, tested provider option.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
- CLI country menu now asks for a per-minute rate when a country is added,
stored alongside its code/name in pstn-intl-allowed.conf. The usage-alert
script buckets each month's outbound calls into domestic vs. per-country
international (longest-code-prefix match on the dialed digits) and prices
each bucket at its own rate instead of one blended flat rate, which badly
under/over-estimated international cost.
- While the kill-switch is tripped, the periodic script now also force-hangs
up any PSTN call already in progress (docker exec + Asterisk's "channel
request hangup" against the trunk's active channels), not just new calls
- closing the gap where an in-progress call just kept running until it
ended naturally. Runs every check while tripped, not only the run that
trips it, to close a start-of-call race.
- Fixed a bug caught by testing: the hangup sweep was nested inside the
"does the call log exist" guard, so it silently never fired if the log
was missing/rotated while tripped. Moved it outside that guard.
- Documented an honest "how bulletproof is this?" breakdown in the
generated README: the provider's own real-time $0-balance block is the
actual hard ceiling (confirmed for Anveo Direct), the kill-switch here is
an estimate-based second layer, not a replacement for funding the prepaid
account at your actual risk tolerance.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
- Replace the hourly cron.d spend/volume check with a per-minute systemd
timer (cron.d fallback), since it's now also the kill-switch's
enforcement point and the interval is the exposure window.
- Add a hard monthly spend-cap kill-switch: trips pstn-trunk-killswitch.conf
(read live by the dialplan, both call directions, internal calling
unaffected) once estimated spend reaches an admin-set cap, plus a loud
ntfy warning at 80% of the cap before it trips. Does not auto-reset;
clearing it is a CLI-only action (update mode) so a compromised/careless
web session can't quietly re-enable spend after a trip.
- Add a CLI-only (never web-dashboard) international-calling allow-list:
continent/country menu, always asked every run with the resulting
allow-list printed exactly once right after, optional auto-expiry with
day-of and at-expiry ntfy notices, and active re-blocking once expired.
Dialplan gates the US "011" prefix pattern on it for full-tier extensions.
- Add an independent messaging_enabled flag per extension in
pstn-permissions.conf for Asterisk's native SIP MESSAGE texting, since
the risk profile (no cost) differs from PSTN calling permissions. The
dialplan wiring to enforce it is flagged as a known gap pending live
verification against Easy Asterisk's message routing, rather than
shipping a guessed pattern.
- security-dashboard.sh: detect whether pstn-trunk is actually installed
(not just base Asterisk) and show a clear "not installed" state instead
of misleading default caps/extensions; add clickable column-header
sorting to the CrowdSec active-bans table.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
Anveo Direct's "Call Security" rate-cap feature (found while evaluating
their account portal) prompted a closer look at our own NANP dialplan
pattern, which turned out to have a real gap: NANP covers Caribbean/
Atlantic nations and several US territories, not just the US - these dial
exactly like a normal 10-digit US number but bill at international/
premium rates, a well-known toll-fraud/"one-ring scam" vector precisely
because the format looks domestic. The existing _1NXXNXXXXX/_NXXNXXXXX
patterns matched all of them equally, meaning any extension (including
"full" tier) could reach premium Caribbean/territory numbers despite the
whole point of the design being "US only, no premium-rate destinations."
Verified the area code list against multiple sources rather than relying
on training-data recall alone, then added an explicit block on all 27
(Bahamas, Barbados, Anguilla, Antigua & Barbuda, British Virgin Islands,
US Virgin Islands, Cayman Islands, Bermuda, Grenada, Turks & Caicos,
Jamaica, Montserrat, Northern Mariana Islands, Guam, American Samoa, Sint
Maarten, Saint Lucia, Dominica, Saint Vincent, Puerto Rico, Dominican
Republic, Trinidad & Tobago, Saint Kitts & Nevis) - checked via REGEX()
against the extracted area code, before permission-tier checks, so it
applies regardless of tier. Tested the extraction/matching logic against
both blocked and legitimate US area codes.
Updated docs/pstn-calling-voipms-plan.md's toll-fraud reasoning, which had
previously (incorrectly) claimed NANP-only restriction alone fully closed
the cost-per-minute risk - noted as a real, live gap rather than glossing
over it. Also noted Anveo Direct's provider-side Call Security setting
(rate cap + max call duration) as a complementary backstop worth
configuring on their account portal directly, independent of this repo's
code.
VoIP.ms's one-POP-one-hostname model meant a single resolved IP was enough
for the trunk's identify match=, but Anveo Direct (researched as an
alternative after finding VoIP.ms's ToS selectively requires government ID
at signup) sends inbound signaling from a fixed set of published IPs
instead. Generalizes the pjsip include to accept multiple match= lines in
one identify object (PJSIP allows repeating it to build one match set - no
separate config objects needed per IP), with a new install prompt for any
additional known source IPs beyond the auto-resolved one.
No dial-prefix support was added despite earlier research suggesting Anveo
Direct needed one - verified against their current official FAQ (fetched
directly rather than trusting a stale search-result claim) that plain
number dialing works with no prefix.
Fixed a real bug caught while testing the update-mode round trip:
.pstn-trunk.env was written with unquoted values, which breaks `source`
entirely once any value contains a space - a multi-IP list or a
multi-word provider name (e.g. "Anveo Direct") would make bash try to run
the second word as a command. Predates this change (a multi-extension
ring group has the same shape); only surfaced by actually re-sourcing the
file and reapplying, not by inspecting generator output. Fixed by quoting
every value.
Updated docs/pstn-calling-voipms-plan.md with the Anveo Direct/DIDLogic/
SIP.US research findings and what's now implemented vs. still open
(real-time balance enforcement at Anveo specifically is unconfirmed).
Adds an inbound concurrent-call cap mirroring the existing outbound one -
outbound alone didn't protect against an inbound call-flood, which also
costs money per-minute on VoIP.ms. Both defaults bumped from 3 to 10.
Moves the cap numbers themselves out of static dialplan text and into a new
pstn-limits.conf, read live via AST_CONFIG() the same way permission tiers
already are - changing either cap takes effect on the next call, no
Asterisk restart, no reinstall. "update in place" never touches this file,
matching the existing protection for pstn-permissions.conf/.env/firewall/
Caddy config.
Adds a concurrency-caps card to security-dashboard.sh's "PSTN Trunk" tab,
above the existing permissions table, so both caps are visible and editable
from the same web page. Tested against a real running instance of the
Python app: default fallback when the file doesn't exist yet, save/persist,
invalid-input rejection, and a bash-to-Python round trip on the generated
file format.
Inbound dialplan ordering mirrors outbound's existing pattern: permission
check (is any ring-group member authorized for this caller) before the
concurrency check, consistent with outbound's tier-check-then-busy-check
order.
Reworks the outbound permission model from a flat allow-list into three
per-extension tiers (internal / restricted / full), addressing the ask for
extensions that can only reach pre-approved numbers plus extensions with
full US calling, while internal extension-to-extension dialing and ring
groups stay ungated for everyone regardless of tier.
Permissions now live in pstn-permissions.conf, read by the dialplan via
Asterisk's AST_CONFIG() on every call instead of being baked into static
dialplan text - editing that file takes effect on the next call, no
Asterisk restart and no re-running the installer. "update in place" mode
never touches this file (same protection this repo's update-mode
convention already gives .env/firewall/Caddy config); only a "fresh"
reinstall (with confirmation) or the web UI change it.
Adds a "PSTN Trunk" tab to services/security-dashboard.sh: lists every
extension (parsed from pjsip.conf) with its live tier and approved numbers,
editable with no restart - this is what makes the tier model actually
manageable day to day. Extracted the dashboard's systemd-unit writing into
its own function so "update" mode refreshes it too (previously only fresh
installs did), and generalized both the dashboard and the trunk service to
detect either asterisk-digital-ocean or the home/LAN asterisk install.
Inbound ring-group membership now checks each member's tier live per call
via an unrolled per-member dialplan block (full always rings, restricted
only if the caller's number is approved, internal never rings) rather than
a single static Dial() string.
Caught and fixed two real bugs during testing against a sandboxed vendor
copy and a live instance of the (stdlib-only) Python dashboard app:
- Asterisk Goto/GotoIf argument parsing: ring<ext>/skip<ext> are named
priorities within the same extension (declared via "same => n(label),..."),
not separate exten => entries, so jumping to them needs the single-argument
Goto(label) form - the two-argument Goto(label,1) form used initially
addresses a different, nonexistent extension named "label" instead.
- A security-relevant REGEX() direction issue: the inbound Caller-ID check
initially interpolated attacker-influenced call data into the PATTERN side
of a REGEX() match rather than the tested-string side, which would let a
crafted Caller-ID forge a match against an unrelated approved-numbers
entry. Fixed by keeping the admin-controlled approved-list as the pattern
and the live call data as the string being tested, consistently on both
the outbound and inbound checks.
Verified end-to-end: dialplan/pjsip generation and vendor-file patching
(idempotent, syntax-checked) as before, plus the new permission-file
round-trip between bash and Python, and the dashboard's new API endpoints
exercised against a real running Python server (extension parsing, tier
changes, number normalization, invalid-input rejection, atomic file writes).
Generalizes services/pstn-trunk.sh (renamed from voipms-trunk.sh in the
prior commit) away from VoIP.ms specifics - any IP-authenticated SIP
provider works, VoIP.ms is just the suggested default. Adds:
- Role-based outbound permission: a configurable allow-list of extensions
that may dial PSTN numbers (regex-gated on CHANNEL(peername)), separate
from internal extension-to-extension dialing which stays open to everyone
regardless. Blank list preserves the original "everyone can dial out"
behavior.
- Inbound ring-group: rings a configurable list of extensions instead of a
single hardcoded one.
- ntfy alerts: immediate on denied (unauthorized extension) or rejected
(concurrency cap hit) calls, plus an hourly cron-driven check that alerts
once per month when estimated spend crosses a threshold and every hour
call volume looks like a burst. Uses a self-contained pipe-delimited call
log rather than Asterisk's CDR, to avoid depending on CDR module
availability and CSV comma-quoting.
- Settings persisted to .pstn-trunk.env so "update in place" reapplies
everything from that file instead of fragile re-parsing out of generated
Asterisk config (which had a real bug: update mode was extracting the
wrong Dial(PJSIP/...) line).
Tested end-to-end against a sandboxed copy of the real vendor files:
permission-gate regex, ring-group dial-string construction, ntfy line
injection/removal, and the usage-alert script's threshold/burst/monthly-
dedup logic all verified with synthetic data. Caught and fixed a sed `&`
escaping bug in the ring-group substitution before it shipped (RING_DIAL
contains literal `&` join characters, which sed's replacement syntax
otherwise treats as "insert the match").
Renames services/voipms-trunk.sh to services/pstn-trunk.sh and generalizes
it away from VoIP.ms specifics - any IP-authenticated SIP provider works,
VoIP.ms is just the suggested default. Adds:
- Role-based outbound permission: a configurable allow-list of extensions
that may dial PSTN numbers (regex-gated on CHANNEL(peername)), separate
from internal extension-to-extension dialing which stays open to everyone
regardless. Blank list preserves the original "everyone can dial out"
behavior.
- Inbound ring-group: rings a configurable list of extensions instead of a
single hardcoded one.
- ntfy alerts: immediate on denied (unauthorized extension) or rejected
(concurrency cap hit) calls, plus an hourly cron-driven check that alerts
once per month when estimated spend crosses a threshold and every hour
call volume looks like a burst. Uses a self-contained pipe-delimited call
log rather than Asterisk's CDR, to avoid depending on CDR module
availability and CSV comma-quoting.
- Settings persisted to .pstn-trunk.env so "update in place" reapplies
everything from that file instead of fragile re-parsing out of generated
Asterisk config (which had a real bug: update mode was extracting the
wrong Dial(PJSIP/...) line).
Tested end-to-end against a sandboxed copy of the real vendor files:
permission-gate regex, ring-group dial-string construction, ntfy line
injection/removal, and the usage-alert script's threshold/burst/monthly-
dedup logic all verified with synthetic data. Caught and fixed a sed `&`
escaping bug in the ring-group substitution before it shipped (RING_DIAL
contains literal `&` join characters, which sed's replacement syntax
otherwise treats as "insert the match").
Claude