Wire up internal SIP MESSAGE enforcement using a dedicated dialplan context
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
This commit is contained in:
Claude
@@ -412,32 +412,34 @@ generator output. Fixed by quoting every value in that heredoc.
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`full`-tier extensions can use it regardless of which countries are
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allowed.
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11. Internal SIP `MESSAGE` (native Asterisk texting, no carrier SMS/cost) —
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**partially done**. The permission layer is real and live-editable: a
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`messaging=yes` flag per extension in `pstn-permissions.conf`,
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independent of the PSTN calling tiers (an extension can be
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internal-tier for calling and still messaging-enabled, or vice versa),
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prompted at install time AND now a checkbox right in the Security
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Dashboard's PSTN Trunk permissions table (alongside tier/approved-
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numbers) — no need to re-run the CLI installer just to change who can
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message. Confirmed it correctly survives tier changes and personal-DID
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assignment/removal on the same extension (this is what surfaced the
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tier=internal section-wipe bug fixed above). **Not done**: the actual
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dialplan wiring that would make Asterisk *enforce* this flag on
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inbound `MESSAGE` requests — this flag currently does nothing at the
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Asterisk level yet, it's groundwork.
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Reasoned through but deliberately not shipped: Easy Asterisk dispatches
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messages through the same `[intercom]` context calls use (no
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`message_context` override), and whether a hand-written pattern there
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would take precedence over — or conflict with — Easy Asterisk's own
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generated per-device dial patterns in that same context isn't something
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that can be safely determined without a live install to test against.
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Shipping a guessed pattern risked either silently not working or, worse,
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interfering with call-routing precedence for the same extensions.
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Treat this the same way as the VoIP.ms live-account verification in
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item 6 above: a real gap, flagged rather than papered over, not a
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hypothetical. Next step for whoever picks this up: verify message
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routing behavior against a live Easy Asterisk container, then wire the
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dialplan gate using the existing flag.
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**done**. The permission layer: a `messaging=yes` flag per extension in
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`pstn-permissions.conf`, independent of the PSTN calling tiers (an
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extension can be internal-tier for calling and still messaging-enabled,
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or vice versa), prompted at install time AND a checkbox right in the
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Security Dashboard's PSTN Trunk permissions table (alongside
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tier/approved-numbers) — no need to re-run the CLI installer just to
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change who can message. Confirmed it correctly survives tier changes and
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personal-DID assignment/removal on the same extension (this is what
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surfaced the tier=internal section-wipe bug fixed above).
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The dialplan gap flagged here previously is now closed: a real install's
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`pjsip.conf`/`extensions.conf` were pulled (2026-07-23) and confirmed
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every endpoint sets `context=intercom` with `message_context` blank
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(falls back to `context`), and `[intercom]` gets one exact-match
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`exten => <ext>,1,...` per device, freshly regenerated by Easy
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Asterisk's own `rebuild_dialplan()` on every dialplan rebuild — exactly
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the collision this doc worried about. Solved by NOT sharing
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`[intercom]`: `services/asterisk-digital-ocean.sh` now explicitly sets
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`message_context=sip-messaging` on every endpoint (patched into both of
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Easy Asterisk's device-creation code paths — the CLI menu's bash
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heredoc and the web admin's Python `add_device()` — so new devices pick
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it up automatically, plus a one-time migration for devices that already
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existed) and routes messages to a dedicated `[sip-messaging]` context in
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`messaging-dialplan.conf`, gated on the sender's `messaging` flag via
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`AST_CONFIG()`. Zero overlap with `[intercom]`'s own call routing.
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One piece still flagged rather than papered over: the `MESSAGE(from)`
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sender-extraction (`CUT()`-based, written to tolerate a display-name
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prefix) hasn't been confirmed against real MESSAGE traffic yet — fails
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closed (denies) if it ever parses wrong, but worth a live test.
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12. Anveo Direct's real-time \$0-balance blocking, confirmed — support reply
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(MFonk, 7/22/2026): "all calls (incoming and outgoing) will be blocked"
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at \$0, in real time, not just via a recurring-fee grace period. This
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@@ -400,6 +400,150 @@ CRON
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fi
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}
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# ── Shared: internal SIP MESSAGE routing/enforcement ────────────────────────
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# Confirmed live against a real install's pjsip.conf/extensions.conf
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# (2026-07-23): every endpoint sets context=intercom and leaves
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# message_context blank, so PJSIP messaging falls back to context=intercom —
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# and [intercom] already owns an exact-match `exten => <ext>,1,...` per
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# device, freshly regenerated by the vendor's own rebuild_dialplan() on
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# every dialplan rebuild. A competing priority-1 declaration for the same
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# extension number in a #include'd file would race that (Asterisk doesn't
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# merge two independent priority-1 declarations for the same context+exten —
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# one silently wins) and risks breaking normal internal calling entirely.
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# So this uses its own dedicated [sip-messaging] context instead, reached by
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# explicitly setting message_context=sip-messaging on every endpoint, so
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# there is never any overlap with [intercom]'s own per-device call routing.
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#
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# The vendor's device-creation code has exactly two independent code paths
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# that write a fresh endpoint block (confirmed via grep — both contain the
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# literal line "context=intercom" exactly once): the CLI menu's bash heredoc,
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# and the web admin's Python add_device(). Patching the vendor's own
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# generator source (same technique as _pstn_patch_vendor_files) makes every
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# device added FROM NOW ON pick this up automatically, in either path.
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# Devices that already existed before this was installed need one one-time
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# migration pass over the live pjsip.conf (below) since they were written
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# before the patch existed.
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_asterisk_do_patch_messaging_vendor_files() {
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local EA_DIR="$1"
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local ENTRYPOINT="$EA_DIR/docker/entrypoint.sh"
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local EASY1="$EA_DIR/easy-asterisk.sh"
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local EASY2
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EASY2="$(find "$EA_DIR" -maxdepth 1 -name 'easy-asterisk-v*.sh' | head -1)"
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[[ -z "$EASY2" ]] && EASY2="$EA_DIR/easy-asterisk-v0.10.0.sh"
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local f
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for f in "$EASY1" "$EASY2"; do
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[[ -f "$f" ]] || { log_error "$f not found — is the base Asterisk install fully set up?"; return 1; }
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done
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# Device-creation templates: both occurrences of "context=intercom" in
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# these two files (identical vendor source, copied twice) are the CLI
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# and web-admin device-creation code paths — a single anchor on the bare
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# line patches both in one pass.
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for f in "$EASY1" "$EASY2"; do
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if ! grep -q '^message_context=sip-messaging$' "$f"; then
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if grep -q '^context=intercom$' "$f"; then
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sed -i '/^context=intercom$/a message_context=sip-messaging' "$f"
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else
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log_warning "$(basename "$f"): 'context=intercom' anchor not found — vendor template changed upstream."
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log_warning " Add 'message_context=sip-messaging' manually after every 'context=intercom' line in this file's device-creation code."
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fi
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fi
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done
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# extensions.conf: same [intercom] anchor _pstn_patch_vendor_files uses,
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# a SEPARATE #include so this coexists whether or not pstn-trunk is
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# installed — messaging is independent of the PSTN trunk entirely.
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for f in "$ENTRYPOINT" "$EASY1" "$EASY2"; do
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[[ -f "$f" ]] || continue
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if ! grep -q 'messaging-dialplan.conf' "$f"; then
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if grep -q '^\[intercom\]$' "$f"; then
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sed -i '/^\[intercom\]$/a #include messaging-dialplan.conf' "$f"
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else
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log_warning "$(basename "$f"): '[intercom]' anchor not found — vendor template changed upstream."
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log_warning " Add '#include messaging-dialplan.conf' manually after [intercom] in this file's extensions.conf heredoc."
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fi
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fi
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done
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log_success "Vendor generator functions patched for internal SIP messaging."
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}
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# One-time migration for devices that already existed before the patch above
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# — new devices pick up message_context=sip-messaging automatically from now
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# on, but anything already in pjsip.conf was written before that existed.
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# Idempotent: buffers the file and only inserts where the very next line
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# isn't already the exact value, so reruns (every "update") never duplicate it.
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_asterisk_do_migrate_existing_devices_message_context() {
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local PJSIP_FILE="$1"
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[[ -f "$PJSIP_FILE" ]] || return 0
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grep -q '^context=intercom$' "$PJSIP_FILE" || return 0
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local TMP_FILE
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TMP_FILE="$(mktemp)"
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awk '
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{ lines[NR] = $0 }
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END {
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for (i = 1; i <= NR; i++) {
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print lines[i]
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if (lines[i] == "context=intercom" && lines[i+1] != "message_context=sip-messaging") {
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print "message_context=sip-messaging"
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}
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}
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}
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' "$PJSIP_FILE" > "$TMP_FILE"
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if ! diff -q "$PJSIP_FILE" "$TMP_FILE" >/dev/null 2>&1; then
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cp "$PJSIP_FILE" "$PJSIP_FILE.backup.$(date +%Y%m%d-%H%M%S)"
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mv "$TMP_FILE" "$PJSIP_FILE"
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chown asterisk:asterisk "$PJSIP_FILE" 2>/dev/null || true
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log_success "Existing devices migrated to message_context=sip-messaging (backup saved alongside pjsip.conf)."
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else
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rm -f "$TMP_FILE"
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fi
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}
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# The actual enforcement — gated on the SENDER's own "messaging" flag in
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# pstn-permissions.conf (the exact file/flag the Security Dashboard's
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# "Internal SIP messaging" checkbox writes, independent of whether the PSTN
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# trunk is installed), read live via AST_CONFIG() on every message, same
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# mechanism pstn-trunk.sh's own dialplan already relies on for permission
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# tiers — no restart needed to take effect. Off by default: an extension
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# with no entry, or messaging=no, is denied. UNVERIFIED: MESSAGE(from)'s
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# exact format hasn't been confirmed on a live install — the CUT()-based
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# extraction below is written to tolerate a display name (e.g. this
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# project's "name0" <999> callerid format) but if it ever fails to parse,
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# FROM_EXT ends up empty/wrong and the AST_CONFIG() lookup simply finds no
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# match, which denies by default (same fail-closed behavior as an
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# unlisted extension) rather than silently allowing anything through.
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_asterisk_do_write_messaging_dialplan() {
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local FILE="$1"
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cat > "$FILE" << 'EOF'
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; Internal SIP MESSAGE routing/enforcement — services/asterisk-digital-ocean.sh.
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; Regenerated on every install/update; edit there, not here directly.
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;
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; Reached via each endpoint's message_context=sip-messaging (patched into
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; Easy Asterisk's own device-creation code — see
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; _asterisk_do_patch_messaging_vendor_files) instead of falling back to
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; [intercom], which already owns an exact-match "exten => <ext>,1,..." per
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; device for CALLS, regenerated fresh on every dialplan rebuild — a
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; competing priority-1 declaration for the same extension number here would
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; race that and risk breaking normal internal calling. This context ONLY
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; ever receives MESSAGE requests, never calls.
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[sip-messaging]
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exten => _X.,1,NoOp(SIP MESSAGE to ${EXTEN})
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same => n,Set(FROM_URI=${MESSAGE(from)})
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same => n,Set(FROM_PART=${CUT(FROM_URI,@,1)})
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same => n,Set(FROM_EXT=${CUT(FROM_PART,:,2)})
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same => n,Set(SENDER_OK=${AST_CONFIG(pstn-permissions.conf,${FROM_EXT},messaging)})
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same => n,GotoIf($["${SENDER_OK}" = "yes"]?deliver:deny)
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same => n(deliver),MessageSend(pjsip:${EXTEN},${FROM_URI})
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same => n,Hangup()
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same => n(deny),NoOp(Denied — extension ${FROM_EXT} is not messaging-enabled)
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same => n,Hangup()
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EOF
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}
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_asterisk_do_remove_presence_timer() {
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systemctl disable --now asterisk-presence-alert.timer 2>/dev/null || true
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rm -f /etc/systemd/system/asterisk-presence-alert.timer /etc/systemd/system/asterisk-presence-alert.service
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@@ -567,6 +711,11 @@ install_asterisk-digital-ocean() {
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echo "[DRY-RUN] Would offer optional ntfy alerts on extension registration going offline/online"
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echo "[DRY-RUN] (checked every 2 minutes via systemd timer, cron.d fallback; always asked,"
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echo "[DRY-RUN] update mode included)"
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echo "[DRY-RUN] Would patch vendor device-creation code + extensions.conf generator to route"
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echo "[DRY-RUN] internal SIP MESSAGE through a dedicated [sip-messaging] dialplan context,"
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echo "[DRY-RUN] gated live on each sender's 'messaging' flag in pstn-permissions.conf (the"
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echo "[DRY-RUN] same file/flag the Security Dashboard's checkbox writes) — independent of"
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echo "[DRY-RUN] whether the PSTN trunk is installed; migrates any already-existing devices too"
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return 0
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fi
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@@ -590,6 +739,11 @@ install_asterisk-digital-ocean() {
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_asterisk_do_refresh_vendor_files
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_asterisk_do_write_compose
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_asterisk_do_write_logrotate "$EA_DIR"
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_asterisk_do_patch_messaging_vendor_files "$EA_DIR"
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_asterisk_do_write_messaging_dialplan "$EA_DIR/config/asterisk/messaging-dialplan.conf"
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_asterisk_do_migrate_existing_devices_message_context "$EA_DIR/config/asterisk/pjsip.conf"
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ensure_docker_dir_ownership "$EA_DIR/config/asterisk"
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chmod 644 "$EA_DIR/config/asterisk/messaging-dialplan.conf"
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log_info "Rebuilding and restarting containers..."
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if docker compose up -d --build --force-recreate; then
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@@ -661,6 +815,10 @@ install_asterisk-digital-ocean() {
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_asterisk_do_refresh_vendor_files
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_asterisk_do_write_logrotate "$EA_DIR"
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_asterisk_do_patch_messaging_vendor_files "$EA_DIR"
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_asterisk_do_write_messaging_dialplan "$EA_DIR/config/asterisk/messaging-dialplan.conf"
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ensure_docker_dir_ownership "$EA_DIR/config/asterisk"
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chmod 644 "$EA_DIR/config/asterisk/messaging-dialplan.conf"
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# ── DigitalOcean droplet detection ────────────────────────────────────────
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# A droplet's own public IP/ID are readable, unauthenticated, from the
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@@ -1113,6 +1271,32 @@ plan for the admin panel.
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| 10000–20000 | UDP | RTP media streams |
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| 49152–49252 | UDP | TURN relay media ports |
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## Internal SIP messaging (no PSTN trunk needed)
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Every extension can send/receive Asterisk's native SIP MESSAGE (no carrier
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SMS, no PSTN, no cost) once its "messaging" flag is set to yes in
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\`pstn-permissions.conf\` — via the Security Dashboard's "Internal SIP
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messaging" card, or by hand. This works independent of \`pstn-trunk.sh\`
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entirely. Under the hood: every device endpoint gets
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\`message_context=sip-messaging\`, routing messages to a dedicated
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\`config/asterisk/messaging-dialplan.conf\` context instead of \`[intercom]\`
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(which already owns per-device call routing) — this install/update patches
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both the device-creation code (so new extensions pick it up automatically)
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and any devices that already existed. Confirmed against a live install's
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\`pjsip.conf\`/\`extensions.conf\` on 2026-07-23 (message_context falls back to
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context=intercom, one exact-match dialplan entry per device) — the MESSAGE
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sender-extraction logic itself is still unconfirmed against real traffic;
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if messages silently don't arrive, check
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\`docker exec easy-asterisk-do asterisk -rx "core set verbose 3"\` while
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sending one.
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## Extension presence (online/offline) alerts
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Optional ntfy alert when an extension's SIP registration changes state —
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offered on both fresh install and "update in place". Checked every 2
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minutes (systemd timer, cron.d fallback); fires only on a change, never on
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every check.
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## Other services (installed separately, not by this script)
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This installer only sets up Asterisk + coturn. Everything else — Caddy,
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@@ -1553,7 +1553,7 @@ INDEX_HTML = """<!doctype html>
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<div class="card">
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<h3 style="margin-top:0">Internal SIP messaging</h3>
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<p class="muted">
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Asterisk's native SIP texting between extensions — no carrier SMS, no PSTN, no cost, and no dependency on a PSTN trunk being installed at all. Independent of the calling permissions below. Note: this flag is live-editable here, but whether Asterisk actually delivers/gates messages using it depends on dialplan wiring not yet verified against a live install.
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Asterisk's native SIP texting between extensions — no carrier SMS, no PSTN, no cost, and no dependency on a PSTN trunk being installed at all. Independent of the calling permissions below. Enforced live by a dedicated dialplan context (see services/asterisk-digital-ocean.sh's README) — install/rerun that service to pick up the dialplan wiring if this box predates it.
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</p>
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<table id="msg-table"><thead><tr><th>Ext</th><th>Name</th><th>Enabled</th><th></th></tr></thead><tbody></tbody></table>
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<div id="msg-msg" class="muted" style="margin-top:0.5rem"></div>
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@@ -1595,7 +1595,7 @@ INDEX_HTML = """<!doctype html>
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Changes apply live, on the next call — no Asterisk restart needed.
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</p>
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<p class="muted">
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<b>Messaging</b> — Asterisk's native internal SIP texting (no carrier SMS, no PSTN, no cost), independent of the calling tier. Note: this flag is live-editable here, but whether Asterisk actually delivers/gates messages using it depends on dialplan wiring not yet verified against a live install — see this service's README.
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<b>Messaging</b> — Asterisk's native internal SIP texting (no carrier SMS, no PSTN, no cost), independent of the calling tier. Enforced live by a dedicated dialplan context — see services/asterisk-digital-ocean.sh's README for how, and its caveat on the sender-extraction logic still needing real-traffic confirmation.
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</p>
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<table id="pstn-table"><thead><tr><th>Ext</th><th>Name</th><th>Tier</th><th>Approved numbers (restricted only)</th><th>Messaging</th><th></th></tr></thead><tbody></tbody></table>
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<div id="pstn-msg" class="muted" style="margin-top:0.5rem"></div>
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Reference in New Issue
Block a user