Three separate faults, reported together as "the old web admin made
extensions correctly and this doesn't".
1. Every write through ea_docker_write left pjsip.conf owned by root. `tee`
runs as root inside the container while Asterisk runs as `asterisk` and
expects to own its own config — the vendored admin's add_device() chowns
it back immediately after writing, and services/asterisk.sh's device
migration does too. This was the one writer in the project that didn't,
and is the most likely reason an extension created here behaves
differently from one created in the vendored admin. Now chowned after
every write, with a scoped sudoers entry for it, and a warning logged if
the chown itself fails rather than passing silently.
2. The dashboard's update path rewrote the systemd unit and then restarted
the service without daemon-reload, so systemd kept running the cached
unit. Any Environment= line added since the last FRESH install was written
to disk and ignored — which is exactly how a box with DOMAIN_NAME and TURN
both set in .env still reported "no domain set" and "TURN not configured".
3. The connection panel rendered at the top of the card, so on any table long
enough to scroll, the answer appeared off-screen above the row that was
clicked. It is now a table row injected directly beneath its own
extension, toggled by the same button, and it survives the transport and
password actions by reopening after the reload they trigger.
Also: LAN devices now get ice_support=yes when a TURN server is configured,
matching the vendored admin (a device given TURN credentials but no ICE can't
use the relay); the panel reports the device type, so a Mobile extension can
be confirmed as such; and an unreadable .env now says which of "path not
set", "file missing", "permission denied" or "TURN_SERVER empty" applies
instead of the flat "not configured" that covered all four.
Verified: the three .env failure states each produce their own message; the
detail row lands directly after its anchor, only one is ever open, it clears
on re-render and reopens rather than sticking closed; no duplicate or missing
element IDs and no JS errors.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
services/sms-inbound.sh was rebuilt to deliver texts into Asterisk over AMI,
landing in the softphone's own thread, and ntfy was dropped from that path
entirely. Two pieces of prose written against the earlier design survived and
now contradict the working implementation:
- docs/anveo-direct-setup-guide.md still ended its "native Messages app"
section with "Codes arrive as ntfy push notifications instead". The point
of that section — no SIP client can write into Android Messages or iOS
Messages — is unchanged, but the place texts actually land is Sipnetic's
message thread.
- services/pstn-trunk.sh's generated README claimed inbound SMS "doesn't
touch Asterisk at all" and set up ntfy notifications. It is now the exact
opposite: sms-inbound reads this service's pstn-personal-dids.conf and
pstn-groups.conf to resolve DID ownership, the same files the inbound-voice
ring logic uses, which also makes install order matter — noted there now.
Documentation only; no behaviour change, and services/sms-inbound.sh is not
touched.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
Making the transport a visible, equally-weighted choice meant new users had
to know the answer before they could get one right — and the wrong answer
fails in the least diagnosable way available: a UDP-only endpoint doesn't
refuse a TLS registration, it ignores it, so the phone times out and nothing
is logged anywhere.
The add form now creates Remote/FQDN (TLS 5061) extensions without asking.
TLS is listed first in the markup so it is the default before any script
runs, and the JS no longer switches it to LAN on a box with no domain — that
box is not better served by UDP, it just needs the phone told to trust a
self-signed certificate, which the disclosure now says at the point of
choosing.
Transport and auto-answer moved behind an "Advanced…" toggle, leaving name,
extension and category as the whole form. The toggle resets on cancel and
after a successful add so it doesn't stay open across uses.
Verified in the browser on two fixtures — with and without DOMAIN_NAME set —
that the panel starts hidden, the transport reads fqdn untouched in both
cases, and each shows the right hint when opened.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
Adding an extension from the dashboard gave back a password and nothing else
— no server, no TURN credentials, and no indication of which transport the
endpoint had actually been written with. A correctly-created extension and
one that could never register looked identical.
Each row gets an info button opening the full set: SIP server, username,
password, transport and port, and the TURN server/user/password. The password
is read back from pjsip.conf rather than regenerated, so re-pairing a handset
no longer means deleting and recreating the extension. The same panel can
switch an extension between LAN (UDP 5060) and Remote/FQDN (TLS 5061) and
reset its password in place, keeping its category, room membership and PSTN
permissions.
That transport choice is the likeliest cause of a phone that looks right and
never registers: an endpoint written transport=transport-udp will not answer
a TLS registration and the phone just times out. The add form now defaults to
Remote/FQDN whenever DOMAIN_NAME is set, rather than always LAN.
Reading TURN details needs Asterisk's .env, which is chmod 600 — the
installer now grants the service user read on that one file via ACL and lists
it in the unit's ReadOnlyPaths, since ProtectSystem=strict would otherwise
hide it. The UI says so plainly if the file still isn't readable.
Also fixes a real gap: ea_add_device is a third, independent writer of
endpoint blocks alongside the two vendor paths that services/asterisk.sh
patches, and it was not emitting message_context=sip-messaging — so an
extension created from this dashboard silently had no internal SIP messaging
while one created from the vendor admin did.
Verified against a fixture: transport round-trips TLS↔LAN without
accumulating duplicate transport/media_encryption/ice_support keys, password
reset rewrites only the target extension's auth section, delete still works
after edits, and the browser shows the populated panel with the add form
defaulting to fqdn when a domain is configured.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
Inbound: a real text through the full path (Anveo webhook -> relay ->
AMI MessageSend -> Sipnetic) landed with a SIP 200 OK, confirmed live.
Removes the last "UNVERIFIED"/"believed correct" hedges from
sms-inbound.sh now that the AMI permission class and the
Destination-not-To fix are both proven, not just plausible.
Outbound over SIP: tested directly by sending a MESSAGE toward Anveo's
trunk (the mirror image of the inbound webhook). Anveo's SBC responded
501 Not Implemented -- a real, unambiguous rejection of the method
itself. Closes off this avenue for good, symmetric with inbound
SMS-over-SIP already being confirmed unavailable on this DID: neither
direction is offered on this account via SIP. Sending still has no
working path here until Anveo activates HTTP SMS-API access.
Also updates docs/anveo-direct-setup-guide.md's SMS section, which still
described the ntfy-based mechanism this session fully replaced with
AMI/Sipnetic delivery, and corrects its stale "not available on Direct"
sending claim with what's actually been confirmed: SIP MESSAGE outbound
is a dead end (501), the HTTP API is real but pending Anveo activation.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JDyKC6Kdg7tofmYSmRtgww
`manager show command MessageSend` (this box's own Asterisk, requested
live) documents Destination as the field that actually resolves an
outgoing message's endpoint/technology; To is documented as a
backward-compatible fallback for the destination when Destination is
omitted, and separately as just the outgoing SIP MESSAGE's To: header
content when Destination IS provided. Two live attempts using only To
(bare "pjsip:212", then domain-qualified "pjsip:212@domain") both
produced zero SIP wire traffic -- confirmed via `pjsip set logger on`
during a real delivery attempt against an actively-registered contact --
meaning that documented fallback path isn't actually wired up on this
Asterisk version regardless of what the docs promise.
Switched to Destination using the docs' own "endpoint" form: bare
"pjsip:<ext>", no domain, which resolves via the endpoint's default
aor/contact -- the same live, registered contact `pjsip show contacts`
already confirmed exists for this extension.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JDyKC6Kdg7tofmYSmRtgww
Live test: AMI MessageSend to a bare "pjsip:212" produced zero SIP wire
traffic (confirmed via `pjsip set logger on` during a real delivery
attempt with the target extension actively registered) -- Asterisk
never even tried reaching the registered contact, meaning the failure
was in URI resolution before anything got sent, not a rejection from
the softphone. From already carried a domain (SMS_DOMAIN); To didn't.
Testing whether that asymmetry was the actual cause.
Explicitly a live experiment, not a confirmed fix -- next test will
show whether this produces real SIP MESSAGE traffic in the logger.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JDyKC6Kdg7tofmYSmRtgww
Chased this as an ACL-ordering problem for the last several commits, and
those fixes were real and worth keeping, but none of them could ever
have fixed this: ProtectHome=true in the systemd unit doesn't just
restrict permissions, it mounts an empty, invisible filesystem over
/home, /root, and /run/user for the whole unit. ASTERISK_CONFIG_DIR
lives under /root/docker/... (or /home/<user>/docker/... on a non-root
install), so the relay process could never see it regardless of any ACL
grant on the real filesystem underneath -- from inside the sandboxed
unit it genuinely doesn't exist, while a plain unsandboxed shell
(confirmed live: `sudo -u smsrelay cat pstn-personal-dids.conf` outside
systemd) reads the exact same path fine.
Fix: ProtectHome=read-only instead of true. Still stops this service
from writing into /home or /root -- all it should ever need is read --
it just stops hiding them outright.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JDyKC6Kdg7tofmYSmRtgww
Reproduced live: pstn-personal-dids.conf was confirmed readable by
smsrelay right after a manual ACL grant, then unreadable again
("does not exist" in the relay's log -- os.path.isfile() swallows the
PermissionError and just returns False) immediately after the very next
fresh install. The only thing that ran in between was this same
install's own Asterisk container restart (needed to pick up the new AMI
secret).
The ACL grant was sequenced BEFORE that restart. CLAUDE.md documents the
container's entrypoint re-chowning its mounted config directory on every
restart and says chown alone can't touch ACL entries -- true, but
apparently this image's entrypoint also chmods, and chmod recomputes a
directory's ACL mask entry, which can silently weaken a named-user grant
made before it even though the grant's ACL entry itself is untouched.
Fix: do the grant last, after the restart-or-not branch, so nothing left
in this install run can undo it. ensure_docker_dir_ownership() (chown
only, confirmed in lib/common.sh, no chmod) stays where it was --
chow doesn't need this ordering fix, only the ACL grant does.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JDyKC6Kdg7tofmYSmRtgww
A token mismatch returned a bare 404 with no journal line at all, so
"nothing is happening" was indistinguishable from "no request ever
arrived" -- exactly what a stale provider URL looks like after a
RELAY_TOKEN rotation (every full reinstall generates a new one, which
invalidates whatever's still pasted into the DID's SMS tab until it's
updated). Now logs the request's source IP and path length -- never the
attempted path itself, since that's unauthenticated input from whoever
hit the port, no reason to trust or echo it into the journal.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JDyKC6Kdg7tofmYSmRtgww
The port-taken message on the last run ("Port 8093 was taken — the relay
will use 8094") was this service colliding with itself, not a real
conflict. Fresh-install never stopped the previous run before scanning
for a free port, so it always found its own earlier process still bound
to 8093, silently moved to 8094, and then `systemctl enable --now` was a
no-op against an already-active unit -- meaning the OLD process (holding
the OLD AMI secret and OLD relay token, from before this session's
settings.env fix) kept serving traffic while the freshly-written config
and port sat unused underneath it.
Fix: `systemctl stop sms-inbound` right before the port scan. Now the
scan only reports a real conflict from something else, and the box
should settle back on 8093 (or whatever's actually free) with the
process that's really running matching what was just configured.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JDyKC6Kdg7tofmYSmRtgww
Root cause of the "line 9: from: unbound variable" crash: SMS_FORWARD_URL
stores Anveo's own template placeholders literally ($[from]$, $[to]$,
$[message]$ -- text Anveo substitutes on its end, not ours). Written into
settings.env double-quoted, that landed in the file as
SMS_FORWARD_URL="...?from=$[from]$&...". Harmless to write, but the
update path `source`s this same file on every re-run, and bash reads
$[from] as legacy arithmetic expansion ($[...] == $((...))) even inside
double quotes -- a bare name in it means "look up variable from", which
is unset, and setup.sh runs under `set -uo pipefail`, so nounset kills
the whole installer before it gets anywhere near the AMI-diagnostics
code from the last two commits.
Fix: single-quote every value in the written settings.env. A source'd
single-quoted assignment never re-expands its contents, so this is safe
regardless of what SMS_FORWARD_URL (or anything else in that file) holds.
This only fixes future writes -- the box that hit this already has a
broken settings.env on disk from before this fix existed, and "update"
mode sources that file before it gets a chance to rewrite it, so it will
crash the same way one more time even after pulling this. Choosing
"full install" instead on the next run skips the source entirely and
regenerates the file correctly quoted.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JDyKC6Kdg7tofmYSmRtgww
First live test dropped an inbound text with only "no owner found for
this DID, dropped" in the journal -- not enough to tell a config-dir
path problem from an ACL/permission problem from a DID-normalization
mismatch from an actually-unassigned DID, without re-triggering a real
text each time. resolve_recipients() now prints which of those it hit:
config dir unset, file missing (which also silently covers "smsrelay
can't traverse/read config/asterisk" -- os.path.isfile() swallows
PermissionError and just returns False), normalized DID has no section,
section has no owner=, or a group owner has no current members.
Nothing added is more sensitive than what's already visible on the
dashboard itself (config dir path, a DID's own digits, the set of DIDs
that exist) -- never the message body.
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JDyKC6Kdg7tofmYSmRtgww
SMS-over-SIP is confirmed unavailable on this DID (the account's SMS tab
only offers "Forward to URL", no MESSAGE/INVITE option) so the diagnostic
dialplan from the previous commit is dead weight for delivery purposes,
though pstn-trunk.sh's message_context wiring stays since it's harmless
and costs nothing to leave in place.
Pivots the whole service: same proven HTTP-webhook front end (rate limit,
constant-time token check, message/from/to extraction), but the delivery
target changes from a push notification to a real SIP MESSAGE landing in
the extension's own softphone (Sipnetic), via Asterisk's Manager
Interface. "Direct mode" and all ntfy options are removed entirely per
instruction to fully replace ntfy, not keep both paths.
Recipient resolution reuses the exact on-disk formats
services/security-dashboard.sh already reads/writes: DID -> owner from
pstn-personal-dids.conf, and for a Ring-Group owner ("@Name"), that
group's current members from pstn-groups.conf -- so delivery always
reflects live DID ownership and live group membership, not a snapshot
from assignment time.
AMI access is scoped tightly: a dedicated "smsrelay" manager.conf user
with only the "message" permission class, bound to 127.0.0.1, secret
generated via generate_password. The service account also gets a POSIX
ACL grant (not chmod/group, which the container's entrypoint reverts on
every restart) for read-only access to the two config files above.
Flagged as UNVERIFIED in code comments and will need a live test: the
"message" AMI permission class name and MessageSend's To/From/Body
parameter names are believed correct from documentation but unconfirmed
against a real Asterisk instance. ami_deliver() logs every AMI response
in full specifically so the first real delivery attempt is
self-diagnosing if something here is wrong.
Verified before commit: bash -n on the full file, and py_compile on the
extracted relay.py (configparser/socket/hmac/http.server, all stdlib).
Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JDyKC6Kdg7tofmYSmRtgww
missing SMS-URI print on update
Two real bugs from the previous commit, both confirmed live on an
actual box where inbound calling had been working:
1. Interleaving the new #include between the two existing trunk
includes broke 'from-pstn-trunk' loading entirely (confirmed via
the box's own dialplan-reload verification check going from
passing to failing on the exact run that added the new include).
Root cause of the underlying quirk still isn't understood — same
as the original single-file-structure bug this file's own comments
already document — so the fix is defensive rather than diagnosed:
append the new include at the very end of the file instead of
between two already-working ones, minimizing disturbance to
whatever ordering/adjacency that loader is sensitive to.
2. When the same verification check fails, _pstn_ensure_live_includes
now automatically falls back to a full container restart and
re-checks, instead of just warning and leaving inbound calls
broken until the operator finds this on their own — confirmed live
this session that a restart (not just dialplan reload) reliably
fixes a context that's genuinely new to the running process.
3. The SMS-over-SIP URI printout was appended only to
install_pstn-trunk's fresh-install tail, which "update in place"
never reaches (it returns early). Factored into
_pstn_print_sms_over_sip_info and called from both paths, so it
actually shows up on the far more common update-in-place run.
Wires message_context=pstn-sms-inbound onto the trunk endpoint (same
pattern services/asterisk.sh already uses for internal SIP MESSAGE
texting — a dedicated context so an inbound MESSAGE doesn't fall back
to context=from-pstn-trunk, which already owns "_X." for INVITE-based
calls) and writes a new [pstn-sms-inbound] dialplan context.
That context deliberately only logs every plausible source for the
sender number, destination DID, and body (X-ANVEO-SMS-FROM/TO
headers, MESSAGE(from)/(to)/(body)) rather than attempting to parse
and route immediately — Anveo's own documentation for this feature
has already proven unreliable twice this session (a dead SMSbySMS
endpoint, and three mutually-contradicting docs pages for the general
API), so this waits for a real test message to confirm which fields
actually show up before building extraction/routing logic against
them.
The install summary now reads DOMAIN_NAME straight from Easy
Asterisk's own .env (the same FQDN already serving SIP for the voice
trunk) and prints the exact `smshandler@<domain>:5060` URI to paste
into the DID's SMS-over-SIP settings, plus how to watch for the first
test message in the Asterisk console.
The previous fix (trimming the Name comparison) was real but not the
actual culprit here. loadExtensions()'s fallback for a device with no
pstn-permissions.conf entry — the normal, default state for any
extension never granted PSTN access — set restrict: "none", which
isn't one of the dropdown's five valid values (full/restricted/
restricted-in/restricted-out/internal). The <select> silently fell
back to its first option ("full") while the underlying model kept
"none", so the mismatch reappeared identically on every load
regardless of caching, matching a persistent bug survives hard
refresh and a private window. Use "internal" instead, matching the
documented default and the value the dropdown actually supports.
rowEdits() compared the Name field's trimmed DOM value against the
untrimmed model value — any device whose stored name has stray
leading/trailing whitespace (e.g. from a manual pjsip.conf edit)
permanently failed that comparison on every load, showing the
unsaved-changes bar with nothing actually changed. Trim both sides.
Adds a Personal number column directly on the Ring Groups table (same
underlying write as the Personal Numbers card's owner picker, just
surfaced where it's actually needed instead of requiring a separate
trip) and closes the gap that group ownership previously left open:
a DID assigned to a group only rang members on inbound calls, with no
way for members to actually use it outbound.
_reconcile_group_cid_members() now makes a Ring Group's DID double as
every current member's outbound Caller-ID override in
pstn-permissions.conf, wired into every path that can change either
side of that relationship: assigning/reassigning/removing the group's
DID, joining/leaving the group, renaming it (repoints
pstn-personal-dids.conf's owner instead of leaving a stale reference,
members' Caller-ID untouched since neither the DID nor membership
changed), and deleting it (unassigns the DID entirely, clearing it
from every remaining member).
An individual extension's own personal_did always wins over a group's
- a member only ever inherits the group's DID into an empty
personal_did, and only ever loses an inherited one that still matches
the group's current DID, so a real individual assignment is never
clobbered. Verified end-to-end (create/assign/join/leave/rename/
delete, plus the individual-wins precedence case) against a local
harness before committing, given how much the outbound Caller-ID path
already burned this session.
Rooms' original flow was always create-empty-then-add-members-one-at-
a-time in the table afterward; the merge kept that flow and dropped
Groups' actual "check boxes, name it, done" creation UX, which is what
was actually asked for. Add-form now has a member picker so a Ring
Group can be built with its full membership in one step; existing
groups still use the per-row add/remove control for later changes.
Rooms already had the one thing Groups lacked — a real, dialable Easy
Asterisk extension with live ring/page dialplan logic — while Groups'
only unique capability was personal-DID ownership. Rebuilding that
ownership mechanism a second time under the Groups name would have
meant duplicating Rooms' dialplan-generation code; instead Rooms
(renamed "Ring Groups" for clarity) gains personal-DID ownership, and
the standalone Groups card/API/backend is removed entirely.
pstn-trunk.sh's existing group-owned-personal-DID machinery
(pstn-personal-group-ring.sh, reading pstn-groups.conf by name) is
untouched — a new sync_room_group_mirror() keeps that file in step
with a Ring Group's live membership automatically on every
create/rename/delete/member change, so pstn-trunk.sh never needs to
learn rooms.conf's format. Room names now validate against the same
pattern write_group() requires, so a room can never end up with a
name that would silently fail to sync once assigned a DID.
The PSTN-restart nudge now also fires on a Ring Group edit, but only
when that specific group currently owns a personal DID — most Ring
Groups never do, and prompting on every ordinary membership tweak
would just be noise.
"Auto-answer for everyone" is the existing Page (intercom) room type,
not a new control. Messaging stays exactly what it already was:
strictly per-extension, unrelated to Ring Group/personal-DID
membership.
Categories was a whole CRUD system (its own file, its own card, its
own API routes) for something that only ever had one functional
effect: tagging a device "mobile" to enable RTP NAT-keepalive tuning,
plus a per-category auto-answer default. Replaced with a single
"Mobile/cellular device" checkbox directly on the add-extension form
and each extension row, writing the same underlying category string
Easy Asterisk's device format already expects ("mobile" or
"standard") without a separate category registry to manage. Removes
ea_list_categories/ea_create_category/ea_delete_category/
ea_rename_category, their /api/ea-categories* routes, the Categories
card, and the now-unused categories.conf sudoers grant.
Rooms stays as-is — unlike Categories, it's the only actually-dialable
ring/page group in this dashboard (a real Easy Asterisk extension that
rings or pages members live), which the dashboard-only Groups feature
cannot replace.
The "Commit changes (restart Asterisk)" button was already wired
correctly, but lived in a passive banner that was easy to miss after
a save, letting an admin keep testing against Asterisk's stale
AST_CONFIG() read without realizing a restart was needed. Every
PSTN-affecting save now also prompts immediately ("Restart Asterisk
now?"), once per logical action (a single save, or one whole batch),
with the banner kept as a fallback for "not now".
Also:
- Extensions card is now collapsible like the other cards, open by
default.
- Concurrent-call caps card removed from the dashboard UI (the
underlying dialplan cap and pstn-limits.conf are untouched, just no
longer editable from this page).
- Categories/Rooms and Groups/Personal numbers are now visually
grouped under section headers, with a note clarifying that a Room
is a real dialable ring-group extension while a Group is a
dashboard-only bulk-action convenience, since both being "named
sets of extensions" invited exactly that confusion.
PJSIP's From-header username comes exclusively from the endpoint's
static from_user — it never falls back to the channel's CALLERID(num).
With from_user unset, Asterisk fell back to its own hardcoded default
("asterisk"), which the provider doesn't recognize as an account DID
and rejected outright: every outbound call went straight to busy with
the callee never ringing.
Restore the static from_user (required for the provider to accept the
call at all) and instead relay the per-call CALLERID(num) — personal_did
or the shared DID — via send_pai/send_rpid, which populate the
P-Asserted-Identity/Remote-Party-ID headers from the live channel
Caller-ID independent of from_user. This is the standard mechanism for
presenting a per-call Caller-ID over a trunk with one fixed From user.
The pjsip endpoint pinned a static from_user to the shared trunk DID,
which silently wins over CALLERID(num) for the SIP From header on
outbound calls. The dialplan's per-extension personal_did lookup
still set CALLERID(num) correctly, but every call still went out
showing the same DID regardless. Leaving from_user unset lets PJSIP
fall back to the channel's caller ID number, mirroring the inbound
side's existing no-static-callerid fix.
The behaviour was right but the vocabulary wasn't: the previous commit
replaced full/restricted/internal with a new none/open/out/in/both naming,
when the ask was to extend the existing tier dropdown rather than supplant it.
The dropdown now reads: full, restricted (both ways), restricted incoming,
restricted outgoing, internal. The stored values match — internal,
restricted, full, restricted-in, restricted-out — so the original three keep
their own names in the config file and migrating a legacy install is now
near-identity for them (tier=full becomes restrict=full, tier=restricted
becomes restrict=restricted, which is what that tier already meant).
restricted incoming = dials anywhere, only whitelisted numbers get through.
restricted outgoing = anyone can call in, may only dial the whitelist. Named
in parallel rather than as "full incoming", so the two sit next to each other
in the list without needing the parenthetical to tell them apart.
No behavioural change: the derived tier_out/allowed_out/tier_in/allowed_in
the dialplan reads are unchanged, so the dialplan itself is untouched.
Verified: legacy migration mapping the original tiers to their same-named
modes; both new modes round-tripping through the dashboard and compiling to
the right derived keys (restricted-in sets allowed_in only, restricted-out
sets allowed_out only); whitelist disabled on full and internal; and the
group-ring helper still ringing restricted-incoming only for a whitelisted
caller while restricted-outgoing rings for everyone.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
The previous commit gave each extension two independent lists — numbers it
may dial and caller IDs that may reach it. That was more than asked for: the
whitelist is one set of numbers per extension, and what varies is which
direction(s) it constrains.
pstn-permissions.conf now has an authored pair, 'restrict' and
'allowed_numbers', where restrict is one of:
none no PSTN at all open unrestricted both ways
out may only dial the list in may only be called by the list
both the list applies both ways
tier_out/allowed_out/tier_in/allowed_in are now derived from that pair rather
than authored directly, and remain what the dialplan reads — so the dialplan
is unchanged from the previous commit and stays tested. 'tier' still mirrors
tier_out for rollback. Keeping the compiled keys means the file has one place
a human edits and one place Asterisk reads, which is the same authored/
compiled split a named-number-list feature would need later.
The migration handles both prior shapes: a genuinely legacy single-tier file
(full becomes open, restricted becomes both — reproducing what the old
dialplan did), and the short-lived two-list shape from the previous commit
(inferred back to a mode, preferring the more restrictive reading). Still
idempotent, still backs up first.
The dashboard drops from two dropdowns and two fields to one of each, with
the whitelist greyed out for the modes that don't use one, and the PSTN
column sorting by how much reach a mode grants rather than alphabetically.
Verified: migration from both shapes; the dashboard round-trip writing
restrict=in with the list compiled to allowed_in only; and the group-ring
helper across all four modes — Restrict inbound rings only for a whitelisted
caller, Restrict outbound rings for everyone, Restrict both rings only for
its own list, No PSTN never rings.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01NAddJGE1G6eGaPzmScG5Vh
Claude