The SMS tab has one control — a "Forward to URL" checkbox and field, with
SAVE/RETURN buttons where RETURN discards. The instructions described a
destination dropdown that isn't there. Also flags that the generated URLs are
long (~90 chars relay, ~150+ direct) and worth re-opening the tab to confirm
they saved whole.
Adds a provider-risk section: the realistic way to lose the number is account
action or a lapsed balance rather than the company folding, so keep the
balance small, save a recent invoice offline (porting out needs a signed LOA
plus the latest bill, which you can't download once an account is closed),
and note that Anveo states it does not block port-outs.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
The UI is inlined in app.py, so an upgrade changes the HTML behind a URL that
never changes and carried no cache headers. A browser holding the previous
page after an update is indistinguishable from the update having failed.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
New service for one narrow job — getting SMS verification codes sent to a
VoIP number onto a phone with no SIM. Deliberately not a texting app: no
outbound path (Anveo Direct has none; that needs an Anveo Retail account, and
a free texting app covers sending), and messages arrive as push notifications
rather than being routed into Asterisk as SIP MESSAGE, since a code you read
and type is better served by a notification than a softphone chat thread.
Two modes, both driven entirely from the provider's "forward SMS to URL" box:
- direct — the provider calls ntfy itself; nothing installed here. ntfy
accepts GET publishing at /{topic}/(publish|send|trigger) with message and
title as query params, and auth via ?auth= holding base64url (unpadded) of
the literal "Bearer <token>" — confirmed against ntfy's server.go and
server_auth.go rather than its docs.
- relay — a stdlib systemd service, Caddy-fronted on its own domain with no
Authelia (the provider can't log in; a random 32-char token in the path is
the secret). Buys two things direct mode can't have: an unescaped "&" in a
message body survives intact, because the relay takes everything after the
last message= verbatim instead of parse_qs — which is why the generated URL
always puts the message placeholder last — and no ntfy credentials sit in a
third party's web portal.
Verification codes are bearer credentials, so: a 24-char random topic name
(the repo's ntfy defaults to auth-default-access: read-write, making the topic
name the read credential), constant-time token compare, a 60/min rate limit,
and the relay logs sender/recipient/length but never the message body.
The Anveo guide gains a section covering the two things that actually decide
whether codes arrive: short-code support (Anveo has it, unusually — VoIP.ms
does not except for Google) and Anveo's carrier-sourced *mobile* DIDs, which
are classified as mobile in the lookups that reject VoIP numbers at signup.
Also documents MMS and group texts being out of reach, and why the native
Messages app never sees any of this.
Verified against a stub ntfy: plain OTP, encoded "&", unencoded "&", "+" as
space, wrong token (404), missing message (400) and the rate limit (57x204
then 429) all behave; both installer modes were run end to end in a sandbox
and their generated URLs, settings files and READMEs checked.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
The tab consolidation fixed what the page showed but not how it read. With
two extensions and nothing else configured it rendered ~1900px tall: six
cards all expanded, three paragraphs of prose before the first control, eight
Save buttons with no indication of which rows had been touched, browser
prompt() dialogs for renames, and a nine-column table with no scroll
container.
Interaction:
- Name and Category are edited in place and feed one batched save. Edited
rows get a highlight and a left rail; a sticky bar reports the count and
Save changes commits only those rows, routing each field to the endpoint it
needs (rename/category through the container, tier/numbers/messaging
through the permissions file — or the messaging-only endpoint with no
trunk). Discard reverts; leaving with edits pending warns. Delete keeps its
own per-row control since it is destructive.
- Categories, Rooms, Groups, Caps and Personal numbers became <details>
sections with item counts, so the tab opens on the extensions table.
Explanations moved behind "what this means" disclosures.
- Add-extension is behind a button; the one-time device password gets a
persistent dismissible callout rather than a line that scrolls away.
Everything else reports through toasts, replacing six inline message divs.
Presentation: a token-based stylesheet (spacing/radius/colour scale), sticky
translucent header, sticky table headers, status as a colour-coded pill,
focus-visible outlines, primary/secondary/danger button hierarchy, and a
.table-wrap that owns horizontal overflow. Security Log and CrowdSec cards
picked up the same card-body padding and scroll wrappers.
Verified in Chromium against a fixture Asterisk config at 1280px and 390px:
full layout opens at ~700px tall with five collapsed sections, editing two
rows marks both and shows "2 extensions edited", saving persists and clears
the dirty state, per-extension failures are reported individually rather than
swallowed, the bare no-container/no-trunk layout still collapses to
Ext/Name/Messaging with two cards and saves via the messaging-only endpoint,
and at 390px the page does not scroll horizontally while the table does.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
Three tabs listed the same extensions three different ways: Asterisk Admin
as devices with category/status/transport, Extensions as a row of messaging
checkboxes, PSTN Trunk as permission tiers with a duplicate Messaging column
that wrote the same flag. Changing one extension meant knowing which of the
three owned the setting you wanted.
There is now one Extensions tab with one extensions table, merged from
pjsip.conf (via /api/pstn-permissions, which always works) and /api/ea-devices
where the Easy Asterisk container is reachable, keyed by extension so a row
known to only one source still shows. Capabilities add columns rather than
nav buttons: Category/Status/Transport are .ea-only, Tier/Approved-numbers
are .pstn-only, and both classes start on <body> so nothing flashes before
/api/ea-status and /api/pstn-status answer. Categories, Rooms, Groups,
Concurrent-call caps and Personal numbers are cards under the same tab, gated
the same way.
Per-row Save picks its write path: tier + approved numbers + messaging via
/api/pstn-permissions with a trunk installed, messaging alone via
/api/pstn-messaging without one — which is what that endpoint has always
been for. No backend changes; the standalone messaging-chips card and the
duplicate Messaging column are both gone.
Verified in Chromium against a fixture Asterisk config: full layout renders
nine columns and six cards, the bare layout collapses to Ext/Name/Messaging
with two cards, and both save paths write pstn-permissions.conf correctly
(messaging-only leaves tier and allowed_numbers untouched).
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
services/asterisk-digital-ocean.sh was a near-verbatim copy of
services/asterisk.sh — same vendor refresh, compose template, messaging
dialplan, presence alerts, UFW rules and dashboard/trunk chaining, with the
helper functions renamed _asterisk_do_*. Two copies meant every fix had to
land twice, and several never did.
There is now one `asterisk` service. It reads the DigitalOcean metadata
service and asks either way (so a droplet with metadata blocked, or another
provider's public VM, can still opt in), then gates the genuinely
droplet-specific behaviour on that one answer: swapfile for low-RAM plans,
public-FQDN-only setup with no LAN/VLAN prompts, a Caddy site block pinned
to that FQDN, the remote-Authelia option, and the doctl Cloud Firewall.
Two things that were droplet-only for no real reason now apply everywhere:
the entrypoint patch that writes security-level events to logs/full, and
the logrotate config for that file. Without them the Security Dashboard's
Security Log tab and CrowdSec's Asterisk acquisition were silently empty on
every home/LAN install; crowdsec.sh now detects either install directory.
Existing droplets are left alone: an install at ~/docker/asterisk-digital-ocean
keeps its directory and easy-asterisk-do container names, since its Caddyfile
block, UFW rules, Cloud Firewall, CrowdSec acquisition and PSTN trunk all
name those exactly. New installs use ~/docker/asterisk / easy-asterisk.
`sudo ./setup.sh asterisk-digital-ocean` still works via a new SERVICE_ALIAS
map in setup.sh, without a second menu entry.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
A group-owned personal DID is now working end to end against a real
inbound call: it rings every authorized group member simultaneously, first
to answer wins, and members whose own tier/allowed_numbers don't authorize
the caller are correctly skipped.
Updates the code comments that were written before that confirmation
existed (the Busy(15) block's "NOT yet confirmed live", and the group-ring
script's unverified-SHELL() note), and documents all four fixes plus the
"Commit Changes" caveat in the Anveo setup guide's bugs-fixed section so
the next person hitting one-ring-then-busy has the trail.
Also clears the guide's "Still open" section: the 2-number account cap no
longer applies, and the interactive walkthrough it described as unbuilt
was built (_pstn_anveo_walkthrough).
Confirmed live this session: writes to pstn-permissions.conf,
pstn-groups.conf, and pstn-personal-dids.conf land on disk immediately, but
AST_CONFIG() in the dialplan sometimes kept returning a stale value (e.g.
a personal DID's old owner after reassigning it to a group) until the
Asterisk container was fully restarted - not just dialplan/module reload.
This contradicts the "live, no restart needed" premise the rest of this
tab's copy relies on, so surfacing it explicitly beats letting admins
discover it by trial and error.
Adds a "Commit Changes (Restart Asterisk)" button and banner to the PSTN
Trunk tab, shown after any save on that tab (permissions, limits, groups,
personal DIDs) and warning via beforeunload if left uncommitted. Backend
restart_asterisk_container() runs `docker restart` on the Easy Asterisk
container through the same scoped-sudoers/run_sudo mechanism the native
Easy Asterisk Admin tab already uses for its own docker exec calls, so this
needed one new sudoers line, not a new privilege model.
Live trace confirmed: once trust_id_inbound=yes started surfacing real
caller identity (previous commit), Anveo delivers it "+E.164" style (e.g.
"+15551234567") instead of the bare digits the old callerid-fallback path
produced. PSTN_CALLERID_NORM's existing "add a leading 1 if length is 10"
check never fires for a 12-character "+"-prefixed value and never strips
the "+" either, so it can never match an 11-digit, digits-only
allowed_numbers entry no matter how correctly the number is whitelisted -
a correctly-configured restricted-tier extension or group member would
still always get busy.
Fixed in three places: the inbound dialplan's PSTN_CALLERID_NORM
computation (new PSTN_CID_RAW step strips a leading "+" first), the
group-ring shell script's own defensive re-normalization, and the
dashboard's admin-input-side normalizers (_normalize_nanp_number,
_normalize_personal_did_input) so pasting a number straight out of a
phone's call log (which naturally includes the "+") works too instead of
being silently dropped.
Live trace confirmed every inbound call's CALLERID(num) came back identical
to the DID that was dialed, regardless of who actually called - meaning
every restricted-tier allowed_numbers check was structurally impossible to
satisfy, since the "caller ID" Asterisk saw was never the real one.
The [pstn-trunk] endpoint had a static `callerid=<DID>` default (redundant
for outbound - the dialplan already sets CALLERID(num) on the calling
channel before Dial(), which is what actually gets presented) with no
trust_id_inbound setting. Removed the static default and added
trust_id_inbound=yes so real caller identity the provider sends
(P-Asserted-Identity/Remote-Party-ID, in addition to the From header) is
honored for inbound calls instead of falling back to the DID.
Root cause found via live debugging: pstn-trunk-dialplan.conf mixed two
contexts in one file - outbound patterns continuing [intercom], then a
[from-pstn-trunk] header appearing partway through the same file. On the
reporting box this caused the ENTIRE file to silently fail to load: `dialplan
show intercom` showed zero of the outbound NANP patterns, and `dialplan show
from-pstn-trunk` reported the context didn't exist, with no warning anywhere
(config log, full log, or the reload command's own output).
messaging-dialplan.conf, #include'd the same way right after [intercom],
loaded fine - the working structural difference is that its first real line
IS its own context header, never trying to continue [intercom] first.
Split pstn-trunk-dialplan.conf into that same working shape:
- pstn-trunk-dialplan.conf keeps only the outbound content (continues
[intercom], unchanged from before).
- pstn-trunk-inbound-dialplan.conf is new, and starts with [from-pstn-trunk]
as its first non-comment line - nothing before it.
Both _pstn_patch_vendor_files (generator patch) and _pstn_ensure_live_includes
(direct live-file patch) now add a second #include line for the new file.
_pstn_ensure_live_includes also now verifies after reload that
'from-pstn-trunk' actually has a matching extension, and warns loudly if not,
instead of unconditionally reporting success regardless of whether the
reload actually worked.
pstn_in_denied, the personal-DID owner denial, and pstn_personal_denied_group
fell straight to a bare Hangup() on an unanswered channel instead of
Busy(15) like every other denial path in this file (outbound tier/number/
intl/killswitch). A bare Hangup() doesn't map to "486 Busy Here" the way
Busy() does, so a denied inbound call (wrong tier, number not on the
approved list, group membership mismatch) could sound like a generic
call-failed/fast-busy rather than a clean busy signal - easy to mistake
for a broken permission check when the real issue is a config mismatch.
Also added a NoOp() logging the resolved group name, caller ID, and
PSTN_RING_LIST right after the group-ring SHELL() call, so the next test
of a group-assigned personal number shows directly (via the Asterisk
console/full log) whether the ring list came back empty - and if so, which
of group membership, tier, or allowed_numbers to check next - instead of
requiring another guess-and-recheck cycle.
Full audit of every phone-number comparison/storage point across
pstn-trunk.sh, security-dashboard.sh, asterisk.sh, asterisk-digital-ocean.sh,
and the vendored easy-asterisk base script, prompted by the inbound
Caller-ID normalization fix — the same digit-count mismatch was also
possible on the admin-input side, just silent instead of loud:
- security-dashboard.sh's write_permission(): a 10-digit whitelist entry
was silently DROPPED (NUMBER_RE required exactly 11 digits), with no
warning unless every entry in the field was invalid — a mixed
10-digit + 11-digit list saved "successfully" while quietly losing the
10-digit one. Now normalizes any bare 10-digit token to 11-digit instead
of discarding it (_normalize_nanp_number).
- write_personal_did(): required exactly 10 digits, rejecting an
11-digit entry outright with a clear (but avoidable) error. Now accepts
either and normalizes to the canonical 10-digit storage form
(_normalize_personal_did_input).
- pstn-trunk.sh's TRUNK_DID install prompt: same fix, strips a leading
"1" instead of aborting the install over it.
Everything else checked out clean: outbound dialed-number matching
already normalizes via the existing _NXXNXXXXXX pattern (adds "1" before
any tier check), the area-code/country-code international gates aren't
phone numbers so digit-count doesn't apply, and asterisk.sh/
asterisk-digital-ocean.sh/the vendored base script have no phone-number
comparison logic at all — this class of bug only lives in the PSTN
permission/whitelist layer this project added on top.
allowed_numbers is always stored 11-digit (dashboard's NUMBER_RE requires
exactly 11 digits), but nothing normalized the inbound CALLERID(num) to
match. Confirmed live: Anveo delivered a bare 10-digit caller ID
(5557654321) for a call to a group-owned personal DID whose members'
allowed_numbers were correctly stored as 15557654321 — every restricted-
tier REGEX comparison failed on a plain digit-count mismatch, not because
the number was actually unauthorized. Outbound already guards against the
mirror-image case (_NXXNXXXXXX prepends "1" before any tier check); this
adds the equivalent for inbound via a new PSTN_CALLERID_NORM dialplan
variable, used everywhere CALLERID(num) previously fed a restricted-tier
comparison: the shared ring-group's per-member block, the single-owner
personal DID check, and the group-owned personal DID's ring script
invocation (which also gets a defensive normalize of its own, since it's
useful to invoke by hand for testing).
Verified by regenerating the actual group-ring script and running it
against the exact config dump and caller ID from the live failing call —
now correctly resolves all three group members instead of an empty list.
install_pstn-trunk's "update" branch unconditionally restarted the
asterisk container, then immediately called _pstn_ensure_live_includes,
which already patches the live pjsip.conf/extensions.conf #include lines
and reloads res_pjsip/dialplan — the restart accomplished nothing the
reload doesn't, while dropping any calls already in progress. Confirmed
live as a real double-restart when this runs chained from
asterisk-digital-ocean.sh, which had just rebuilt/restarted the same
container moments earlier.
Fresh installs keep their existing (still restart-first) apply step
unchanged — only "update in place" loses the redundant restart.
_ini_get() in _pstn_write_personal_group_ring_script split on '=' and
compared $1 == key without trimming whitespace, but configparser.write()
(used to write pstn-groups.conf and pstn-permissions.conf) pads '=' with
spaces by default. Every lookup — including the group's own members= line —
silently returned empty, so group-owned personal DID ring-groups never
actually rang anyone regardless of member tier/whitelist config. Trim
whitespace around both the extracted key and value before comparing.
Also switch the script's final echo to printf to remove any ambiguity in
the dialplan's SHELL()-empty-string check.
Confirmed live on a real droplet: secdash could read pjsip.conf fine but
/api/pstn-status kept returning false even though pstn-trunk-dialplan.conf
genuinely existed. Root cause: the Asterisk container's own entrypoint runs
`chown -R asterisk:asterisk /etc/asterisk` on every container start, and
the numeric UID/GID it resolves to inside the container coincidentally
collided with unrelated host system accounts (config/asterisk ended up
owned by messagebus:uuidd on this box) - silently reverting whatever group
membership _secdash_grant_asterisk_access had granted secdash at install
time. This wasn't a one-time misconfiguration; it would have silently
broken again on every future container restart.
Switched the grant mechanism from chmod + usermod group membership to
POSIX ACLs (setfacl) - chown doesn't touch ACL entries (only chmod
recalculates the ACL mask, and nothing in this flow chmods after install),
so the grant survives the container's own maintenance chown. Default ACLs
(-d) also make newly-created files (a regenerated dialplan, a fresh
personal-DID entry) inherit the same access automatically.
Also added _secdash_grant_ancestor_traversal, which grants execute-only ACL
traversal up the directory tree - needed on any box where DOCKER_DIR sits
under a restrictive parent (e.g. /root on some cloud images defaults to
700, blocking a non-root secdash from ever reaching deeper directories no
matter what those directories themselves grant). Falls back to the old
chmod/group approach with a warning if the 'acl' package is somehow
unavailable (it's installed automatically otherwise).
Verified with a real non-root system test user against a fixture
reproducing the exact failure (a leaf directory with no "other" access,
owned by an unrelated user/group): confirmed blocked before the fix,
confirmed read+write access after, and confirmed access survives a
simulated `chown -R` (the container restart scenario) plus correct
inheritance onto a freshly-created file afterward - all without re-running
the grant.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
asterisk.sh and asterisk-digital-ocean.sh now offer, at the end of both
their fresh-install and update-mode paths, to also set up the Security
Dashboard and configure a real PSTN trunk in the same run - one script
walks through the whole stack instead of needing to separately remember
and run `sudo ./setup.sh security-dashboard` / `sudo ./setup.sh pstn-trunk`
afterward.
Both target services keep their own register_service call and stay fully
independently invocable - this is purely an additive convenience layer
(_asterisk_offer_dashboard_and_trunk / _asterisk_do_offer_dashboard_and_trunk),
not a replacement. An already-installed piece is silently refreshed
(install_security-dashboard/install_pstn-trunk each already have their own
update/fresh/cancel reinstall-mode gate, so calling them again just does
the right thing); a not-yet-installed piece gets one y/n instead of every
detailed prompt firing.
Guarded with declare -F so a standalone `sudo bash asterisk.sh` copy (no
sibling services/*.sh files sourced) skips both cleanly with an
explanatory message instead of erroring on an undefined function.
Verified: full sourcing simulation resolves all four install_* functions
correctly, `setup.sh --dry-run --unattended asterisk` and
`asterisk-digital-ocean` both complete cleanly end-to-end, and
`setup.sh --list` still shows all four services as independently
selectable.
Documented the pattern in CLAUDE.md under a new "Chaining into another
service from within your own" section for future contributors.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
Supersedes the previous commit's reverse-proxy approach entirely: instead
of Caddy routing to Easy Asterisk's own separate vendored web admin
process, the dashboard now reimplements that admin's functionality
natively - one process, one page, real tab-switching, no separate app to
proxy, patch, or embed. Reverts asterisk.sh/asterisk-digital-ocean.sh's
WEBADMIN_BASE_PATH vendor patching and Caddy-skip logic back to their
pre-proxy state (confirmed identical via diff) since neither is needed
anymore.
security-dashboard.sh additions:
- ea_* functions covering full device/category/room parity with
vendor/easy-asterisk/easy-asterisk-v0.10.0.sh's own web admin: list/add/
delete/rename/change-category for devices, list/create/delete/rename for
categories, list/create/delete/rename/add-member/remove-member for
rooms, plus live registered/unregistered status. Reads go straight
through the host-side bind-mounted config files (same as the existing
list_extensions() already does for pjsip.conf); writes go through
`docker exec -i <container> tee <path>` instead of a direct host-side
write, since Easy Asterisk's container writes these files as its own
internal user and a host-side write would just be fighting that
ownership again on the next container restart.
- Found and fixed a real bug (inherited from the vendored admin's own
template, not introduced here): a plain non-mobile LAN device leaves
both the keepalive and ice template lines empty, producing two
consecutive blank lines inside the endpoint's pjsip.conf stanza instead
of one - which broke the delete/rename/category-change parsers' "blank
line ends this device's block" boundary detection, leaving an orphaned
tail of config behind on delete. Fixed by building the endpoint block
from a filtered line list instead of positional template blanks.
Confirmed via a full synthetic add/rename/category-change/delete cycle
against realistic pjsip.conf/categories.conf/rooms.conf fixtures (with
docker exec mocked to a local file) - round-trips back to the original
fixture correctly.
- New plumbing: _secdash_grant_asterisk_access grants read-only access to
categories.conf/rooms.conf's directory (separate from pjsip.conf's,
confirmed against the vendored source - /etc/easy-asterisk/*, not
/etc/asterisk/*); _secdash_write_sudoers adds six exact (no wildcards)
docker-exec sudoers entries scoped to the one Asterisk container
actually installed, validated live with visudo -c; _secdash_write_systemd_unit
passes the new ASTERISK_EA_CONFIG_DIR/ASTERISK_EA_CONTAINER env vars and
adds the config dir to ReadOnlyPaths, validated live with
systemd-analyze verify.
- New UI: Asterisk Admin tab with Devices/Categories/Rooms cards, sortable
tables matching the existing style, inline category-reassignment
dropdowns, and per-room member chips with an inline add-member picker.
Nav button visibility now checks live container reachability
(/api/ea-status) instead of just Asterisk-install detection.
Still unverified: the actual `docker exec` calls (module reload, dialplan
rebuild, live status) against a real running Easy Asterisk container -
only the file-parsing/transformation logic itself has been exercised, via
mocked writes, not the real container plumbing.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
No more cross-origin iframe: the Security Dashboard now reverse-proxies
the real Asterisk web admin natively at /asterisk-admin/ on its own
domain via Caddy's handle_path, instead of embedding a separate site in
a frame. One domain, one login wall, for both.
- services/asterisk.sh / services/asterisk-digital-ocean.sh: patch the
vendored web admin's one hardcoded absolute API path
(`const API_BASE = '/api'`, confirmed via the real vendored source to
be the only absolute-path reference anywhere in its HTML/JS - no other
hrefs, no login-page redirect, plain HTTP Basic Auth instead) so it
resolves correctly when mounted under a sub-path, via a new
WEBADMIN_BASE_PATH env var threaded through entrypoint.sh. Verified
against the real vendored file: patched output is
'/asterisk-admin/api' with the env var set, unchanged '/api' without
it. Skip each service's own dedicated admin Caddy domain when the
Security Dashboard is already installed, since it fronts the admin
instead.
- services/security-dashboard.sh: _secdash_configure_caddy now accepts
the admin's port and Asterisk's own directory/domain, path-routes
/asterisk-admin/* alongside the dashboard's own handle{} block, and
writes WEB_ADMIN_BASE_PATH into Asterisk's .env + restarts that
container once proxying is confirmed live. Defaults the dashboard's
own domain prompt to the droplet's DOMAIN_NAME when detected, since
Caddy's SIP-TLS cert sync already depends on serving that exact
domain. Removed the old CSP frame-ancestors patching and the iframe
itself; the nav is now a plain link, shown only once the proxy is
confirmed wired up.
- Fixed _secdash_remove_caddy_block's marker match to tolerate the
dashboard's reverse_proxy line now living one indent level deeper
(inside its own handle{} block) - verified against a synthetic
Caddyfile that it still finds and removes exactly the right block.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
Restructure the nav so it reflects what's actually installed on this box,
letting one dashboard URL cover everything from a bare LAN Asterisk box up
to a full droplet with a trunk and CrowdSec:
- Security Log and a new Extensions tab (Groups + Internal SIP messaging,
split out of the old "PSTN Trunk" tab) are always available - they only
need Asterisk itself, not a trunk or CrowdSec.
- Asterisk Admin, PSTN Trunk, and CrowdSec each check their own live
install state on every page load and hide their own nav button entirely
when not present, instead of showing an empty/placeholder tab.
- Add crowdsec_installed() (checks for /usr/bin/cscli) and a
/api/crowdsec-status endpoint, mirroring the existing pstn_installed()/
/api/pstn-status pattern.
This fixes the earlier design where messaging/groups management lived
inside the PSTN Trunk tab even though both work with plain Asterisk and no
trunk at all - hiding that tab would have taken them down with it.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
- Security Log tab: replace the per-column text filter row with clickable
sortable column headers (same pattern as the CrowdSec bans table).
- PSTN Trunk tab: add sortable headers to the permissions, personal-DID
(numeric on DID), and groups tables.
- Move the "Internal SIP messaging" card to the bottom of the PSTN tab and
replace its per-row table+Save-button layout with a checkbox chip row
and a single Save-changes button.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
The dashboard's Personal Numbers card now accepts a group (stored as
"@GroupName", unambiguous against a same-named numeric extension) as well
as a plain extension. A group-owned DID rings every CURRENT member whose
own tier/approved-numbers authorize the caller, computed fresh on every
call by a generated pstn-personal-group-ring.sh (invoked via the
dialplan's SHELL() function) rather than unrolled at install time, since
group membership can change any time via the dashboard with no reinstall
- unlike the shared ring-group, which is fixed at install/update time.
Applies the identical per-member permission check the shared ring-group
already bakes into the dialplan, just computed in a plain shell loop
against the same two config files - group ownership doesn't bypass the
tier/approved-numbers model. Tested standalone against mock config data
(full/restricted/internal mix, matching/non-matching caller, empty and
nonexistent groups) - all four cases behaved correctly. The dialplan's own
SHELL() invocation is still unverified against a real call.
Group ownership never touches pstn-permissions.conf's personal_did
(outbound Caller-ID override) field, since there's no single extension to
hang that on for a group.
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.
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').
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
- 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
Claude