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.
Ports the internal SIP MESSAGE dialplan enforcement, extension presence
(online/offline) ntfy alerts, and the live-config-include patching from
asterisk-digital-ocean.sh, for feature parity between the two Asterisk
flavors. The pstn-trunk.sh live-include fix already applied to both, since
that's shared code parametrized by which flavor is installed.
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').
Confirmed live: pressing Enter on any "(y/n): y"-style prompt set the
variable to an empty string instead of the stated default, since prompt_yn
had no ${response:-$default} fallback (prompt_text already had one). Every
downstream [[ "$VAR" =~ ^[Yy]$ ]] check treated "just press Enter" as no.
Also shows the default value in the prompt text itself for both helpers,
since neither displayed it before.
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.
Confirmed against a live install's pjsip.conf/extensions.conf that every
endpoint falls back to context=intercom for messaging (message_context
blank), and that [intercom] owns one exact-match per-device dial pattern
regenerated on every dialplan rebuild. Rather than risk racing that, every
endpoint now gets message_context=sip-messaging (patched into both of Easy
Asterisk's device-creation code paths, plus a one-time migration for
existing devices), routing messages to their own [sip-messaging] context
gated on the existing pstn-permissions.conf messaging flag.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
- ntfy.sh: auth-default-access was deny-all, silently blocking both publish
and anonymous subscribe on every topic once an auth-file is present.
Default to read-write (same model as public ntfy.sh) so alerts from
crowdsec.sh/pstn-trunk.sh actually get delivered.
- crowdsec.sh: add update/fresh/cancel reinstall gating — reruns previously
re-asked every optional prompt (ASN exempt, geo-allowlist, ntfy, remote
LAPI) unconditionally with no way to just refresh in place.
- asterisk-digital-ocean.sh: add optional ntfy alerts on extension
registration going offline/online, checked every 2 minutes via systemd
timer (cron.d fallback), offered on both fresh install and update.
- security-dashboard.sh: replace the outbound-only Asterisk Web Admin link
with an embedded, lazy-loaded iframe tab, with a best-effort Caddy
frame-ancestors patch and an always-available "open in new tab" fallback.
Groups: a new always-available "Groups" card in the PSTN tab (same place
as Internal SIP messaging, no dependency on a PSTN trunk being installed)
lets you name a set of extensions and bulk-enable/disable messaging for
all of them at once. Purely a management-layer convenience - pstn-groups.conf
is never read by the dialplan, which only ever looks at per-extension keys
in pstn-permissions.conf. Applying a group action just calls the same
write_messaging() each individual checkbox uses, once per current member.
Editing membership never retroactively changes anything already applied,
and deleting a group never touches members' own settings - confirmed with
tests covering create/apply/edit-membership/re-apply/delete.
Security Log: added a per-column filter row (Time/Event/Account/Remote/
Severity), live as you type, filtering client-side against the
already-fetched events rather than re-querying - persists correctly across
the existing 30-second auto-refresh.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
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
parse_security_log() did f.readlines() on ASTERISK_LOG before slicing the
last 5000 lines - that reads the ENTIRE file into memory first. That log
is Asterisk's unrotated console/security output, and this tab polls it
every 30 seconds from the browser. Confirmed live: on a 1GB-RAM droplet
with a 1.4GB log file, this ballooned the dashboard (explicitly meant to
be a lightweight stdlib-only process) to 677MB RSS / 1.8GB peak swap,
which left CrowdSec unable to even start (boot timeout) and directly
contributed to the droplet becoming unresponsive.
Fixed by reading only a bounded ~2MB tail from the end of the file
(seek + fixed-size read) instead of the whole thing - memory use is now
constant regardless of how large the log grows. Tested against a 180MB
synthetic log: memory delta dropped from being proportional to file size
to ~7MB, in 0.05s.
Also added log rotation (services/asterisk-digital-ocean.sh) for that same
file, which had no rotation at all and reached 1.4GB in about 3 days -
copytruncate avoids needing to signal the containerized Asterisk process
to reopen its log handle.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
"cscli scenarios remove crowdsecurity/asterisk_bf crowdsecurity/asterisk_user_enum"
was failing silently (stderr suppressed, "|| true" swallowed the non-zero
exit) because these scenarios are members of the crowdsecurity/asterisk
collection, and cscli refuses to remove/disable a collection member
without --force. The un-exempted hub originals kept running side-by-side
with the ASN-exempt local forks the entire time, independently banning the
same traffic with no ASN awareness at all - confirmed live by a ban
showing scenario "crowdsecurity/asterisk_bf" (the hub original) rather
than "local/asterisk_bf" (the fork) for an ASN that had just been exempted.
Fixed in two places:
- crowdsec.sh's initial setup now passes --force and surfaces a clear
warning (with the manual fix command) if it still fails, instead of
silently continuing.
- security-dashboard's set-asn-exempt.sh helper now re-asserts the forced
disable on every ASN-list save, so an install already affected by this
(from before this fix, or where the step failed for any other reason)
self-heals the next time the ASN list is edited via the dashboard,
without needing to re-run the full CrowdSec installer.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
set_asn_exempt() was writing /etc/crowdsec/scenarios/local-asterisk_*.yaml
directly from the dashboard's own process, but those files are root:root
mode 644 and the secdash service user (--shell /usr/sbin/nologin) has no
write grant to them - confirmed by the reported "[Errno 13] Permission
denied" error. Every whitelist/unban attempt through the dashboard was
silently a no-op as far as CrowdSec was concerned, which is why a
carrier's IP kept getting re-banned even after repeatedly trying to
exempt its ASN.
Fixed by routing the edit through a dedicated root-owned helper script
(set-asn-exempt.sh, installed alongside app.py, root:root mode 700) that's
the one thing added to the sudoers whitelist - same pattern every other
CrowdSec-touching action here already uses (cscli via sudo), rather than
loosening the scenario files' own permissions.
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
Support reply (MFonk, 7/22/2026) confirms both open questions: all calls
(incoming and outgoing) are blocked in real time at $0 balance, not just
via the recurring-fee grace-period mechanism, and the 4 signaling IPs
already hardcoded into this doc's research notes are still current.
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).
The user found ToS language stating the account "may run on a negative
balance" and describing a 30-day-notice-then-suspend / 30-consecutive-
days-then-close process - worth reading against the wiki's "balance must
be over $0 to call" claim the whole toll-fraud design leans on.
Reconciled: these describe two different things, not a contradiction.
New call attempts should still be blocked in real time at $0 (the wiki's
claim, and the core assumption this design needs). The ToS's negative-
balance language most plausibly covers recurring fees (DID/E911) landing
when balance is already near zero, and in-progress-call settlement edge
cases - not a window where fraud keeps dialing while negative. Still not
verified against a live account either way.
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").
Adds a VoIP.ms SIP trunk on top of asterisk-digital-ocean: IP-authenticated
trunk (no password stored), NANP-only outbound dialplan, a global 3-call
concurrent cap via GROUP()/GROUP_COUNT(), and inbound routing to one
extension. Config lives in its own include files rather than being
appended directly to pjsip.conf/extensions.conf, since Easy Asterisk fully
regenerates both from its own internal state — the includes are patched
into the vendor's generator functions so they survive that regeneration.
Wires the new service into setup.sh's is_installed() and README's services
table, and updates docs/pstn-calling-voipms-plan.md to reflect what's now
implemented vs. still open (spend/volume alerting, live-account
verification).
Inbound (DID) is now decided as wanted, not outbound-only. Adds a cost
estimate table for 100 min/month each direction, and clarifies that
NANP-only restriction bounds cost-per-minute but not burn speed, so the
concurrent-call cap and spend alert are required before funding a live
trunk, not optional hardening.
Captures the toll-fraud / prepaid-cap research and decisions from a design
discussion (provider: VoIP.ms, US-only calling for now) so implementation
can be picked up in a future session without re-deriving the background.
Nothing implemented yet.
- CrowdSec tab: per-ASN "Unwhitelist" (drop from the exempt filter, future
traffic evaluated normally) and "Unwhitelist + Ban" (also immediately
bans every IP CrowdSec has on record for that ASN, for accidental-
whitelist cases) buttons. set_asn_exempt now allows clearing the list
down to zero ASNs, needed to unwhitelist the last remaining entry.
- New sudoers permission (cscli decisions add --ip * --duration * --type
ban --reason *) scoped narrowly, list-form subprocess args only.
- Caddy/Authelia config factored into _secdash_configure_caddy() and a new
_secdash_remove_caddy_block() (whole-block delete-and-regenerate, not
in-place patching) so "update" mode can now offer to reconfigure it.
- Installer offers an independent HTTP Basic Auth layer in front of
Authelia (Caddy basicauth, generated via `caddy hash-password`) so a
future Authelia bug/misconfig alone isn't enough to expose a page that
can delete active security bans.
Currently-exempt ASNs with no active ban (e.g. T-Mobile once its bans
stop firing) had no carrier name to show, since the name lookup only
looked at cscli decisions list (active bans only). Add a
cscli alerts list-based fallback (includes expired/resolved alerts)
and merge it into the name lookup used by /api/asn-exempt.
Confirmed the real cscli decisions list -o json structure live rather
than guessing again: AS number/name and country live on each alert's
"source" object (source.as_number, source.as_name, source.cn), not on
the individual decision. Surfaces this as a "Network / Carrier" and
"Country" column in the bans table, adds a per-row "Exempt ASN" button
that appends straight to the Asterisk brute-force ASN exemption list,
and labels the exempt list's own entries with carrier names (pulled
from current ban data where available) instead of showing bare numbers.
Outis