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").
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.
Both scripts now detect and install msitools if missing, supporting
apt-get (Debian/Ubuntu), dnf (Fedora), and pacman (Arch). Falls back
to a clear error if the package manager is not recognised.
Remove msitools from manual prerequisites in the doc.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01D8ckUJQtj1pH8jtAddBDZs
Both scripts now download and install GE-Proton10-34 if not already present:
- setup-swbf2-linux.sh: installs to ~/.steam/root/compatibilitytools.d/
- setup-swbf2-wolf.sh: installs into the WolfSteam container via docker cp + tar
Both pause after install and prompt the user to set GE-Proton in Steam
(Compatibility → Force) before continuing, since that step requires the GUI.
Update docs to reflect GE-Proton is now handled automatically.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01D8ckUJQtj1pH8jtAddBDZs
- Remove editorial content (EA server speculation, 2005 alternative, opinions)
- Add native Linux Steam section covering all six setup steps
- Add separate key paths tables for Wolf and native
- Add native Linux troubleshooting commands alongside Wolf equivalents
- Rename title to reflect both platforms
- Fresh install table rewritten as factual survival matrix
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01D8ckUJQtj1pH8jtAddBDZs
Long-form document tracing the actual discovery process: the wrong turns,
the aha moments, and the reasoning behind each step. Covers:
- Why EA is involved in a Steam game at all (link2ea:// architecture)
- JunoConfigureRegistry failure and why DISABLEROLLBACK was a dead end
- msiextract bypass as the actual solution
- Why direct system.reg edits don't survive reboots (wineserver ownership)
- The launch wrapper approach for registry persistence
- RPC_S_SERVER_UNAVAILABLE and EALocalHostSvc
- Wolf-specific bwrap capabilities mismatch
- localconfig.vdf atomic rename problem and chmod 555 fix
Written for sharing — no identifying information, explains the "why"
for each fix so readers can adapt if EA changes their MSI structure.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01D8ckUJQtj1pH8jtAddBDZs
Documents the EA account requirement, first-time login flow, credential
caching for machine transfers, community server options if EA shuts down,
and the SWBF2 2005 alternative for a launcher-free experience.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01D8ckUJQtj1pH8jtAddBDZs
Revised approach: skip link2ea:// and EA App account requirement entirely.
Launch starwarsbattlefrontii.exe directly via the wrapper, spoof Origin
registry entries so the game's built-in launcher check passes. Documents
what doesn't work and why (msiexec/bwrap/EA account issues).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01D8ckUJQtj1pH8jtAddBDZs
Documents the EA App installation workaround for Star Wars Battlefront II
(2017) running under Wolf/WolfSteam with GE-Proton10-34. Covers the bwrap
capability problem, Steam launch wrapper trick, msiextract bypass for the
JunoConfigureRegistry Wine incompatibility, and correct EA Desktop file layout.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01D8ckUJQtj1pH8jtAddBDZs
- Swaps image from filebrowser/filebrowser to gtstef/filebrowser:stable
- Uses ./data:/home/filebrowser/data volume layout (Quantum convention)
- Generates data/config.yaml instead of settings.json + database.db
- Mounts primary path as /files with defaultEnabled: true
- Deploys fbq-add-source.sh for adding extra directories post-install
- Updates README with Quantum-specific instructions
docs: save standalone bootstrap as docs/standalone-template.sh
Preserves the full standalone pattern from filebrowser.sh as the
reference template for adding standalone support to other services.
https://claude.ai/code/session_014CCYqVwW6d6f5dw1qRokYt
Claude