From fcb01a4d9343bc958912dbf2b5bf06996172a97a Mon Sep 17 00:00:00 2001
From: Claude
Date: Thu, 23 Jul 2026 05:03:13 +0000
Subject: [PATCH 1/4] Add personal/per-extension DID assignment ("their own
numbers")
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
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
---
docs/pstn-calling-voipms-plan.md | 29 ++++
services/pstn-trunk.sh | 205 ++++++++++++++++++++----
services/security-dashboard.sh | 260 ++++++++++++++++++++++++++++---
3 files changed, 442 insertions(+), 52 deletions(-)
diff --git a/docs/pstn-calling-voipms-plan.md b/docs/pstn-calling-voipms-plan.md
index 99c52af..e82ad8b 100644
--- a/docs/pstn-calling-voipms-plan.md
+++ b/docs/pstn-calling-voipms-plan.md
@@ -475,3 +475,32 @@ generator output. Fixed by quoting every value in that heredoc.
`README-pstn-trunk.md` for the full honest breakdown of what's actually
hard (the provider's own $0-balance block) vs. estimate-based (this
kill-switch).
+15. ~~Multiple DIDs, one per extension ("personal numbers")~~ Done —
+ additive to the existing shared trunk DID/ring-group, not a
+ replacement. Anveo Direct's DID pricing (~$0.15/mo + $0.25 setup on the
+ Per Minute plan) makes "everyone gets their own number" genuinely cheap
+ at personal-use volumes, and multiple DIDs sharing one trunk/account is
+ exactly what that plan is built for (10 dedicated incoming channels
+ bundled in). New `pstn-personal-dids.conf` (DID -> owner extension,
+ read live by the dialplan for inbound routing) plus a `personal_did=`
+ field per extension in `pstn-permissions.conf` (the outbound Caller-ID
+ override) — both kept in sync automatically by a single write path
+ (CLI prompt at install/update, or the Security Dashboard's "PSTN
+ Trunk" tab), never requiring the admin to hand-edit both files
+ consistently. Inbound: 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 a personal DID isn't the shared line.
+ Outbound: `pstn_check_busy` (the one shared exit point for both
+ domestic and international dialing) looks up the calling extension's
+ `personal_did` and uses it as `CALLERID(num)` instead of the shared
+ trunk DID when one is assigned. A personal DID assigned to an
+ internal-tier extension is accepted but silently never rings anyone
+ until that extension is also granted full/restricted tier — both the
+ CLI and the dashboard warn about this at assignment time rather than
+ blocking it, matching this repo's general permissive-with-warnings
+ style. Caught and fixed a real bug while building this: the
+ dashboard's `write_permission()` did `cp.remove_section(ext)` when
+ tier was set to "internal", which silently discarded any
+ `messaging=yes` or `personal_did=` already on that extension — fixed
+ to remove only the tier/allowed_numbers keys, dropping the section
+ only once nothing else is left in it.
diff --git a/services/pstn-trunk.sh b/services/pstn-trunk.sh
index 2b29cab..6188bbb 100644
--- a/services/pstn-trunk.sh
+++ b/services/pstn-trunk.sh
@@ -11,9 +11,9 @@
# never gated by any of the above, regardless of tier — the trunk is purely
# an additional path out to/in from the real phone network.
#
-# Defaults to VoIP.ms (see docs/pstn-calling-voipms-plan.md for the design/
-# cost background this is built from) but isn't hardcoded to it — any SIP
-# trunk provider that supports IP authentication works the same way.
+# Provider-agnostic — any SIP trunk provider that supports IP authentication
+# works the same way. VoIP.ms and Anveo Direct are both confirmed working;
+# see docs/pstn-calling-voipms-plan.md for the design/cost background.
#
# Requires an existing services/asterisk-digital-ocean.sh OR services/asterisk.sh
# install — this adds a PSTN trunk on top of one of them and does not stand
@@ -23,7 +23,7 @@
#
# Part of the modular post-install system (sourced by setup.sh).
-register_service pstn-trunk homelab "SIP PSTN trunk for asterisk-digital-ocean/asterisk — US-only, per-extension permission tiers, spend/volume alerts (defaults to VoIP.ms)"
+register_service pstn-trunk homelab "SIP PSTN trunk for asterisk-digital-ocean/asterisk — US-only, per-extension permission tiers, spend/volume alerts (any IP-authenticated provider — VoIP.ms and Anveo Direct both confirmed)"
# ── Surviving Easy Asterisk's regeneration ──────────────────────────────────
# Easy Asterisk (the vendor project asterisk-digital-ocean.sh/asterisk.sh
@@ -306,7 +306,8 @@ exten => pstn_check_busy,1,Set(PSTN_MAX_OUT=${AST_CONFIG(pstn-limits.conf,limits
same => n,Set(PSTN_MAX_OUT=${IF($["${PSTN_MAX_OUT}" = ""]?10:${PSTN_MAX_OUT})})
same => n,GotoIf($[${GROUP_COUNT(pstn-out)} >= ${PSTN_MAX_OUT}]?pstn_busy,1)
same => n,Set(GROUP()=pstn-out)
- same => n,Set(CALLERID(num)=__PSTN_DID__)
+ same => n,Set(PSTN_PERSONAL_CID=${AST_CONFIG(pstn-permissions.conf,${PSTN_CALLER},personal_did)})
+ same => n,Set(CALLERID(num)=${IF($["${PSTN_PERSONAL_CID}" = ""]?__PSTN_DID__:${PSTN_PERSONAL_CID})})
same => n,Set(PSTN_START=${EPOCH})
same => n,Dial(PJSIP/${EXTEN}@pstn-trunk,60)
same => n,Set(PSTN_DUR=$[${EPOCH} - ${PSTN_START}])
@@ -340,9 +341,11 @@ EOF
cat >> "$FILE" << 'EOF'
[from-pstn-trunk]
-exten => _X.,1,NoOp(Inbound PSTN call from ${CALLERID(num)})
+exten => _X.,1,NoOp(Inbound PSTN call from ${CALLERID(num)} to ${EXTEN})
same => n,Set(PSTN_KILLED=${AST_CONFIG(pstn-trunk-killswitch.conf,state,tripped)})
same => n,GotoIf($["${PSTN_KILLED}" = "1"]?pstn_in_killed,1)
+ same => n,Set(PSTN_PERSONAL_OWNER=${AST_CONFIG(pstn-personal-dids.conf,${EXTEN},owner)})
+ same => n,GotoIf($["${PSTN_PERSONAL_OWNER}" != ""]?pstn_personal_inbound,1)
same => n,Set(PSTN_RING_LIST=)
same => n,Set(PSTN_RING_SEP=)
EOF
@@ -376,6 +379,29 @@ __ALERT_BUSY_IN_LINE__
exten => pstn_in_killed,1,NoOp(PSTN trunk - spend-cap kill-switch is tripped, rejecting inbound call)
__ALERT_KILLED_IN_LINE__
same => n,Hangup()
+
+; Personal DID inbound routing — rings ONLY the assigned owner, gated by
+; that owner's own tier/approved-numbers (same check every ring-group
+; member gets, just for a single specific target instead of a list, and
+; with no fallback to the shared ring-group if the owner can't take this
+; call — it's their own number, not the shared line).
+exten => pstn_personal_inbound,1,Set(PSTN_OWNER_TIER=${AST_CONFIG(pstn-permissions.conf,${PSTN_PERSONAL_OWNER},tier)})
+ same => n,GotoIf($["${PSTN_OWNER_TIER}" = "full"]?pstn_personal_ring,1)
+ same => n,Set(PSTN_OWNER_ALLOWED=${AST_CONFIG(pstn-permissions.conf,${PSTN_PERSONAL_OWNER},allowed_numbers)})
+ same => n,GotoIf($["${PSTN_OWNER_TIER}" = "restricted" & ${REGEX("^(${PSTN_OWNER_ALLOWED})$" ${CALLERID(num)})}=1]?pstn_personal_ring,1)
+ same => n,NoOp(Denied - personal DID ${EXTEN}'s owner ${PSTN_PERSONAL_OWNER} not authorized for this caller)
+__ALERT_DENY_PERSONAL_LINE__
+ same => n,Hangup()
+
+exten => pstn_personal_ring,1,Set(PSTN_MAX_IN=${AST_CONFIG(pstn-limits.conf,limits,max_inbound)})
+ same => n,Set(PSTN_MAX_IN=${IF($["${PSTN_MAX_IN}" = ""]?10:${PSTN_MAX_IN})})
+ same => n,GotoIf($[${GROUP_COUNT(pstn-in)} >= ${PSTN_MAX_IN}]?pstn_in_busy,1)
+ same => n,Set(GROUP()=pstn-in)
+ same => n,Set(PSTN_START=${EPOCH})
+ same => n,Dial(PJSIP/${PSTN_PERSONAL_OWNER},20)
+ same => n,Set(PSTN_DUR=$[${EPOCH} - ${PSTN_START}])
+ same => n,System(printf '%s|in|%s|%s|%s\n' "${PSTN_START}" "${CALLERID(num)}" "${EXTEN}" "${PSTN_DUR}" >> /var/log/asterisk/pstn-trunk-calls.log)
+ same => n,Hangup()
EOF
if [[ -n "$NTFY_URL" ]]; then
@@ -383,8 +409,9 @@ EOF
sed -i "s#__ALERT_DENY_INBOUND_LINE__# same => n,System(curl -m 5 -s -d 'PSTN trunk: inbound call rejected - caller not approved for any ring target.' '${_esc_url2}' >/dev/null 2>\\&1 \\&)#" "$FILE"
sed -i "s#__ALERT_BUSY_IN_LINE__# same => n,System(curl -m 5 -s -d 'PSTN trunk: inbound concurrent-call cap reached - a call was rejected.' '${_esc_url2}' >/dev/null 2>\\&1 \\&)#" "$FILE"
sed -i "s#__ALERT_KILLED_IN_LINE__# same => n,System(curl -m 5 -s -H 'Priority: urgent' -d 'PSTN trunk: inbound call rejected - spend-cap kill-switch is tripped.' '${_esc_url2}' >/dev/null 2>\\&1 \\&)#" "$FILE"
+ sed -i "s#__ALERT_DENY_PERSONAL_LINE__# same => n,System(curl -m 5 -s -d 'PSTN trunk: inbound call to a personal DID rejected - owner not authorized for this caller.' '${_esc_url2}' >/dev/null 2>\\&1 \\&)#" "$FILE"
else
- sed -i "/__ALERT_DENY_INBOUND_LINE__/d; /__ALERT_BUSY_IN_LINE__/d; /__ALERT_KILLED_IN_LINE__/d" "$FILE"
+ sed -i "/__ALERT_DENY_INBOUND_LINE__/d; /__ALERT_BUSY_IN_LINE__/d; /__ALERT_KILLED_IN_LINE__/d; /__ALERT_DENY_PERSONAL_LINE__/d" "$FILE"
fi
}
@@ -411,18 +438,30 @@ _pstn_write_limits_file() {
# ── Shared: initial permission tiers (fresh install / explicit reset only —
# "update in place" never calls this, matching how .env/firewall/Caddy config
# are protected elsewhere in this repo; see file-level comment above) ──────
-# Args: FILE, space-separated FULL_EXTS, space-separated MESSAGING_EXTS, then
-# "ext" "pipe|separated|numbers" pairs for each restricted extension.
+# Args: FILE, space-separated FULL_EXTS, space-separated MESSAGING_EXTS,
+# space-separated "ext=did" PERSONAL_DID_ASSIGNMENTS, then "ext"
+# "pipe|separated|numbers" pairs for each restricted extension.
_pstn_write_permissions_file() {
- local FILE="$1" FULL_EXTS="$2" MESSAGING_EXTS="$3"
- shift 3
+ local FILE="$1" FULL_EXTS="$2" MESSAGING_EXTS="$3" PERSONAL_DID_ASSIGNMENTS="$4"
+ shift 4
+ local -A _personal_did_map=()
+ local _pd_token
+ for _pd_token in $PERSONAL_DID_ASSIGNMENTS; do
+ _personal_did_map["${_pd_token%%=*}"]="${_pd_token#*=}"
+ done
local _written_exts=""
{
- echo "; PSTN permission tiers — internal / restricted / full — PLUS an independent"
- echo "; 'messaging' flag for Asterisk's native internal SIP MESSAGE texting (no"
- echo "; carrier SMS, no PSTN, no cost — a separate axis from PSTN calling, since"
- echo "; the risk profile is different: an extension can be internal-tier for"
- echo "; calling and still messaging-enabled, or vice versa)."
+ echo "; PSTN permission tiers — internal / restricted / full — PLUS two independent"
+ echo "; axes per extension:"
+ echo "; - 'messaging' for Asterisk's native internal SIP MESSAGE texting (no carrier"
+ echo "; SMS, no PSTN, no cost — a separate axis from PSTN calling, since the risk"
+ echo "; profile is different: an extension can be internal-tier for calling and"
+ echo "; still messaging-enabled, or vice versa)."
+ echo "; - 'personal_did' assigns this extension its own DID (see"
+ echo "; pstn-personal-dids.conf, which the dialplan reads for inbound routing —"
+ echo "; this key here is only the OUTBOUND Caller-ID override). A personal DID"
+ echo "; only actually rings anyone if its owner is also full or restricted tier —"
+ echo "; internal tier means no PSTN either way, personal DID or not."
echo "; Read LIVE by the dialplan on every call (AST_CONFIG()) — no Asterisk"
echo "; restart needed when this changes. Edit here directly, via the Security"
echo "; Dashboard web UI's \"PSTN Trunk\" tab (if installed), or by re-running"
@@ -438,6 +477,7 @@ _pstn_write_permissions_file() {
echo "[$_ext]"
echo "tier=full"
[[ " $MESSAGING_EXTS " == *" $_ext "* ]] && echo "messaging=yes"
+ [[ -n "${_personal_did_map[$_ext]:-}" ]] && echo "personal_did=${_personal_did_map[$_ext]}"
echo ""
_written_exts="$_written_exts $_ext"
done
@@ -448,6 +488,7 @@ _pstn_write_permissions_file() {
echo "tier=restricted"
echo "allowed_numbers=${_nums}"
[[ " $MESSAGING_EXTS " == *" $_ext "* ]] && echo "messaging=yes"
+ [[ -n "${_personal_did_map[$_ext]:-}" ]] && echo "personal_did=${_personal_did_map[$_ext]}"
echo ""
_written_exts="$_written_exts $_ext"
done
@@ -455,6 +496,15 @@ _pstn_write_permissions_file() {
if [[ " $_written_exts " != *" $_ext "* ]]; then
echo "[$_ext]"
echo "messaging=yes"
+ [[ -n "${_personal_did_map[$_ext]:-}" ]] && echo "personal_did=${_personal_did_map[$_ext]}"
+ echo ""
+ _written_exts="$_written_exts $_ext"
+ fi
+ done
+ for _ext in "${!_personal_did_map[@]}"; do
+ if [[ " $_written_exts " != *" $_ext "* ]]; then
+ echo "[$_ext]"
+ echo "personal_did=${_personal_did_map[$_ext]}"
echo ""
fi
done
@@ -462,6 +512,36 @@ _pstn_write_permissions_file() {
chmod 664 "$FILE"
}
+# ── Shared: personal DID -> owner-extension mapping (fresh install / explicit
+# reset only — same "update never touches it" protection as
+# pstn-permissions.conf). Args: FILE, then "did" "owner" pairs.
+_pstn_write_personal_dids_file() {
+ local FILE="$1"
+ shift
+ {
+ echo "; Personal DID -> owner-extension mapping. Read LIVE by the dialplan"
+ echo "; (AST_CONFIG()) on every inbound call — no restart needed. An inbound call"
+ echo "; to a DID listed here routes directly to its owner, checked against the"
+ echo "; owner's OWN tier/approved-numbers in pstn-permissions.conf — no ring-group"
+ echo "; fallback, since this is that extension's own number, not the shared line."
+ echo "; The matching outbound Caller-ID override lives in pstn-permissions.conf"
+ echo "; ('personal_did=' per extension) — kept in sync automatically whenever a"
+ echo "; DID is assigned/removed via the CLI installer or the Security Dashboard's"
+ echo "; PSTN Trunk tab, rather than hand-editing both files separately."
+ echo "; The shared trunk DID keeps working as the main/ring-group line regardless"
+ echo "; of anything assigned here."
+ echo ""
+ while [[ $# -gt 0 ]]; do
+ local _did="$1" _owner="$2"
+ shift 2
+ echo "[$_did]"
+ echo "owner=$_owner"
+ echo ""
+ done
+ } > "$FILE"
+ chmod 664 "$FILE"
+}
+
# ── Shared: kill-switch state (fresh install / explicit reset only — same
# "update never touches it" protection as pstn-permissions.conf/
# pstn-limits.conf) ─────────────────────────────────────────────────────────
@@ -1090,13 +1170,14 @@ install_pstn-trunk() {
local PERMISSIONS_FILE="$ASTERISK_DIR/pstn-permissions.conf"
local LIMITS_FILE="$ASTERISK_DIR/pstn-limits.conf"
local KILLSWITCH_FILE="$ASTERISK_DIR/pstn-trunk-killswitch.conf"
+ local PERSONAL_DIDS_FILE="$ASTERISK_DIR/pstn-personal-dids.conf"
local SETTINGS_FILE="$EA_DIR/.pstn-trunk.env"
local CONTAINER_NAME="easy-asterisk"
[[ "$ASTERISK_KIND" == "asterisk-digital-ocean" ]] && CONTAINER_NAME="easy-asterisk-do"
if [ "$DRY_RUN" = true ]; then
echo "[DRY-RUN] Would require an existing asterisk-digital-ocean OR asterisk (LAN) install"
- echo "[DRY-RUN] Would prompt for: SIP provider name (default VoIP.ms), server/POP hostname, DID,"
+ echo "[DRY-RUN] Would prompt for: SIP provider name (any IP-authenticated provider), server/POP hostname, DID,"
echo "[DRY-RUN] full-PSTN extensions, restricted-PSTN extensions + their approved numbers,"
echo "[DRY-RUN] internal SIP messaging extensions (separate from PSTN calling permission),"
echo "[DRY-RUN] max concurrent outbound/inbound calls (default 10/10), inbound ring-group extensions,"
@@ -1133,15 +1214,17 @@ install_pstn-trunk() {
log_warning "A static IP from your ISP avoids that; asterisk-digital-ocean sidesteps it entirely."
fi
- log_info "Configuring a SIP PSTN trunk for $ASTERISK_KIND (defaults to VoIP.ms)."
+ log_info "Configuring a SIP PSTN trunk for $ASTERISK_KIND (any IP-authenticated provider —"
+ log_info "VoIP.ms and Anveo Direct are both confirmed working; see docs/pstn-calling-voipms-plan.md)."
log_info "US-only outbound (NANP dialplan), a concurrent-call cap, per-extension permission"
log_info "tiers, an inbound ring-group, and ntfy alerts on denied/rejected calls plus"
log_info "spend/volume checks."
echo ""
log_warning "Before continuing, on your provider's side you should already have: created an"
log_warning "account, funded and set up prepaid billing with auto-recharge OFF (VoIP.ms: Client"
- log_warning "Area -> Balance Management), ordered a DID with IP authentication pointed at this"
- log_warning "box's public IP, and picked a server/POP. Also restrict outbound routing to"
+ log_warning "Area -> Balance Management; Anveo Direct: fund the account balance directly),"
+ log_warning "ordered a DID with IP authentication pointed at this box's public IP, and picked"
+ log_warning "a server/POP. Also restrict outbound routing to"
log_warning "US/NANP on the provider's own side if it offers that — this dialplan is the second,"
log_warning "independent layer, not a substitute for the first."
log_warning "See docs/pstn-calling-voipms-plan.md for the full background."
@@ -1200,7 +1283,7 @@ install_pstn-trunk() {
# ── Prompts — provider account details aren't scriptable, set up manually
# on the provider's own site first (see warning above) ───────────────────
local PROVIDER_NAME=""
- prompt_text "SIP trunk provider name (for your reference/docs only):" "VoIP.ms" PROVIDER_NAME
+ prompt_text "SIP trunk provider name (for your reference/docs only — e.g. VoIP.ms, Anveo Direct):" "" PROVIDER_NAME
local TRUNK_SERVER=""
prompt_text "Server/POP hostname (e.g. atlanta2.voip.ms for VoIP.ms — pick the one closest to this box from your provider's server list):" "" TRUNK_SERVER
@@ -1317,6 +1400,48 @@ install_pstn-trunk() {
local MESSAGING_EXTS=""
prompt_text "Extensions allowed to use internal SIP messaging (space-separated, blank = none):" "" MESSAGING_EXTS
+ # ── Personal numbers — optional, additive to the shared trunk DID ──────
+ # Multiple DIDs can share this one trunk/account. Assigning one to a
+ # specific extension makes inbound calls to it ring ONLY that extension
+ # (still gated by that extension's own tier/approved-numbers — a
+ # personal DID doesn't bypass PSTN permission, it just narrows routing
+ # from "the shared ring group" to "this one owner"), and makes that
+ # extension's outbound calls show its own DID as Caller-ID instead of
+ # the shared one. The shared DID/ring-group above is unaffected either
+ # way — this is purely additive.
+ echo ""
+ echo " Personal numbers (optional): assign specific DIDs to specific extensions."
+ echo " Inbound calls to that DID ring only its owner; outbound calls from that"
+ echo " extension show its own DID as Caller-ID. Requires the owner to also be"
+ echo " full or restricted tier to actually receive anything on it."
+ local WANT_PERSONAL_DIDS=""
+ prompt_yn "Assign any personal DIDs now? (y/n):" "n" WANT_PERSONAL_DIDS
+ local PERSONAL_DID_PAIRS=() PERSONAL_DID_ASSIGNMENTS=""
+ if [[ "$WANT_PERSONAL_DIDS" =~ ^[Yy]$ ]]; then
+ local _pd_more="y"
+ while [[ "$_pd_more" =~ ^[Yy]$ ]]; do
+ local _pd_did="" _pd_owner=""
+ prompt_text " DID (10-digit US number, digits only):" "" _pd_did
+ if [[ "$_pd_did" =~ ^[0-9]{10}$ ]]; then
+ prompt_text " Owner extension for $_pd_did:" "" _pd_owner
+ if [[ "$_pd_owner" =~ ^[0-9]+$ ]]; then
+ PERSONAL_DID_PAIRS+=("$_pd_did" "$_pd_owner")
+ PERSONAL_DID_ASSIGNMENTS="${PERSONAL_DID_ASSIGNMENTS} ${_pd_owner}=${_pd_did}"
+ if [[ " $FULL_EXTS $RESTRICTED_EXTS " != *" $_pd_owner "* ]]; then
+ log_warning "Extension $_pd_owner isn't full/restricted tier yet — it won't actually"
+ log_warning "receive calls on $_pd_did until you also grant it one of those tiers."
+ fi
+ log_success "Will assign $_pd_did to extension $_pd_owner."
+ else
+ log_warning "Not a valid extension — skipped."
+ fi
+ else
+ log_warning "Not a valid 10-digit DID — skipped."
+ fi
+ prompt_yn " Assign another? (y/n):" "n" _pd_more
+ done
+ fi
+
echo ""
local WANT_NTFY=""
prompt_yn "Send an ntfy alert when a call is denied (permission tier/approved-number check failed) or rejected (concurrency cap hit)? (y/n):" "y" WANT_NTFY
@@ -1348,7 +1473,7 @@ install_pstn-trunk() {
log_info "Spend/volume alert settings (used only to estimate cost and flag unusual usage —"
log_info "not billing-accurate, just a safety net)."
local RATE_PER_MIN=""
- prompt_text " Outbound per-minute rate in USD (VoIP.ms US rate is 0.01):" "0.01" RATE_PER_MIN
+ prompt_text " Outbound per-minute rate in USD (check your provider's published rate — e.g. VoIP.ms US is ~0.01, Anveo Direct US is ~0.001):" "0.01" RATE_PER_MIN
local MONTH_THRESHOLD=""
prompt_text " Alert once when estimated spend this month reaches (USD):" "10" MONTH_THRESHOLD
local BURST_THRESHOLD=""
@@ -1383,9 +1508,10 @@ install_pstn-trunk() {
"$RING_EXTS" "$NTFY_URL" "$RATE_PER_MIN" \
"$MONTH_THRESHOLD" "$BURST_THRESHOLD" "$PROVIDER_NAME" "$MAX_MONTHLY_SPEND" "$CONTAINER_NAME" || return 1
- _pstn_write_permissions_file "$PERMISSIONS_FILE" "$FULL_EXTS" "$MESSAGING_EXTS" "${RESTRICTED_ARGS[@]}"
+ _pstn_write_permissions_file "$PERMISSIONS_FILE" "$FULL_EXTS" "$MESSAGING_EXTS" "$PERSONAL_DID_ASSIGNMENTS" "${RESTRICTED_ARGS[@]}"
_pstn_write_limits_file "$LIMITS_FILE" "$MAX_OUTBOUND" "$MAX_INBOUND"
_pstn_write_killswitch_file "$KILLSWITCH_FILE"
+ _pstn_write_personal_dids_file "$PERSONAL_DIDS_FILE" "${PERSONAL_DID_PAIRS[@]}"
ensure_docker_dir_ownership "$ASTERISK_DIR"
# No new firewall rules: the base install already opens SIP (5060/5061)
@@ -1461,9 +1587,10 @@ a strong safety net, not an absolute guarantee against any overage. See
cat > "$DOC_FILE" << MD
# SIP PSTN trunk (add-on to $ASTERISK_KIND)
-US-only outbound PSTN calling over a SIP trunk (defaults to VoIP.ms, works
-with any IP-authenticated provider), per-extension permission tiers, a
-configurable concurrent-call cap, and an inbound ring-group. See
+US-only outbound PSTN calling over a SIP trunk (any IP-authenticated
+provider — VoIP.ms and Anveo Direct both confirmed working), per-extension
+permission tiers, a configurable concurrent-call cap, and an inbound
+ring-group. See
\`docs/pstn-calling-voipms-plan.md\` in the repo for the full design
background, cost estimate, and toll-fraud reasoning.
@@ -1656,13 +1783,31 @@ call-routing precedence in the same \`[intercom]\` context. Treat the
permission flag as ready for a dashboard/CLI-managed allow-list once that
routing is confirmed, not as fully wired yet.
+## Personal numbers
+
+Multiple DIDs can share this one trunk/account — assign one to a specific
+extension and inbound calls to it route straight to that owner, still
+gated by the owner's own tier/approved-numbers (no ring-group fallback,
+since it's that extension's own line, not the shared one), while that
+extension's outbound calls show its own DID as Caller-ID instead of the
+shared trunk DID above. Entirely additive: the shared DID/ring-group keeps
+working for everyone regardless of what's assigned here.
+
+$([ "${#PERSONAL_DID_PAIRS[@]}" -gt 0 ] && { local _i; for ((_i=0; _i<${#PERSONAL_DID_PAIRS[@]}; _i+=2)); do echo "- \`${PERSONAL_DID_PAIRS[$_i]}\` -> extension ${PERSONAL_DID_PAIRS[$_i+1]}"; done; } || echo "None assigned yet.")
+
+Stored in \`config/asterisk/pstn-personal-dids.conf\` (DID -> owner, read live
+by the dialplan for inbound routing) and a \`personal_did=\` field per
+extension in \`pstn-permissions.conf\` (the outbound Caller-ID override) —
+both kept in sync automatically by the CLI installer and the Security
+Dashboard's "PSTN Trunk" tab, live, no restart needed.
+
## Managing this from a web UI
If \`services/security-dashboard.sh\` is installed, its "PSTN Trunk" tab
-shows both the per-extension permission tiers and the outbound/inbound
-concurrency caps, all editable live — no restart, no reinstall. Install/
-update it any time with \`sudo ./setup.sh security-dashboard\`; it
-auto-detects this install.
+shows the per-extension permission tiers, the outbound/inbound concurrency
+caps, and personal-number assignments, all editable live — no restart, no
+reinstall. Install/update it any time with \`sudo ./setup.sh
+security-dashboard\`; it auto-detects this install.
## Manual edits
diff --git a/services/security-dashboard.sh b/services/security-dashboard.sh
index d25bef8..6bb851a 100644
--- a/services/security-dashboard.sh
+++ b/services/security-dashboard.sh
@@ -925,6 +925,51 @@ def list_extensions():
return extensions
+def _write_ini_cp(path, header, cp):
+ """Shared temp-write-then-rename for every live-editable PSTN conf file —
+ one copy of the atomic-write/error-handling logic instead of repeating
+ it per file. Returns (ok, error_message_or_None)."""
+ if not path:
+ return False, "No Asterisk install detected on this box"
+ tmp_path = path + ".tmp"
+ try:
+ with open(tmp_path, "w") as f:
+ f.write(header)
+ cp.write(f)
+ os.replace(tmp_path, path)
+ except OSError as e:
+ try:
+ os.remove(tmp_path)
+ except OSError:
+ pass
+ return False, "Failed writing %s: %s" % (path, e)
+ return True, None
+
+
+PERMISSIONS_HEADER = (
+ "; PSTN permission tiers - internal / restricted / full - PLUS two\n"
+ "; independent per-extension axes: messaging (internal SIP MESSAGE\n"
+ "; texting) and personal_did (outbound Caller-ID override; inbound\n"
+ "; routing for personal DIDs lives in pstn-personal-dids.conf).\n"
+ "; Read LIVE by the dialplan on every call (AST_CONFIG()) - no\n"
+ "; Asterisk restart needed. Managed here (Security Dashboard); also\n"
+ "; safe to edit by hand. 'sudo ./setup.sh pstn-trunk' update mode\n"
+ "; never touches this file, only a fresh reinstall does.\n"
+ "; Any extension not listed here is internal-only (no PSTN) by default.\n\n"
+)
+
+PERSONAL_DIDS_HEADER = (
+ "; Personal DID -> owner-extension mapping. Read LIVE by the dialplan\n"
+ "; (AST_CONFIG()) on every inbound call - no restart needed. Managed here\n"
+ "; (Security Dashboard); also safe to edit by hand. Kept in sync with\n"
+ "; pstn-permissions.conf's personal_did= field automatically by\n"
+ "; write_personal_did()/remove_personal_did() below - editing this file\n"
+ "; by hand also requires updating that field yourself to match.\n"
+ "; 'sudo ./setup.sh pstn-trunk' update mode never touches this file, only\n"
+ "; a fresh reinstall does.\n\n"
+)
+
+
def _permissions_path():
return os.path.join(ASTERISK_CONFIG_DIR, "pstn-permissions.conf") if ASTERISK_CONFIG_DIR else None
@@ -980,8 +1025,20 @@ def write_permission(ext, tier, numbers_raw):
cp = _read_permissions_cp()
if tier == "internal":
+ # Only drop the tier/allowed_numbers keys, NOT the whole section —
+ # an extension can independently have messaging=yes and/or a
+ # personal_did assigned, and those must survive a tier change back
+ # to internal. Confirmed live as a real bug: cp.remove_section(ext)
+ # here used to silently discard both whenever tier was set to
+ # internal. Only remove the section itself once nothing else is
+ # left in it.
if cp.has_section(ext):
- cp.remove_section(ext)
+ if cp.has_option(ext, "tier"):
+ cp.remove_option(ext, "tier")
+ if cp.has_option(ext, "allowed_numbers"):
+ cp.remove_option(ext, "allowed_numbers")
+ if not cp.options(ext):
+ cp.remove_section(ext)
else:
if not cp.has_section(ext):
cp.add_section(ext)
@@ -991,26 +1048,9 @@ def write_permission(ext, tier, numbers_raw):
elif cp.has_option(ext, "allowed_numbers"):
cp.remove_option(ext, "allowed_numbers")
- path = _permissions_path()
- tmp_path = path + ".tmp"
- try:
- with open(tmp_path, "w") as f:
- f.write(
- "; PSTN permission tiers - internal / restricted / full.\n"
- "; Read LIVE by the dialplan on every call (AST_CONFIG()) - no\n"
- "; Asterisk restart needed. Managed here (Security Dashboard); also\n"
- "; safe to edit by hand. 'sudo ./setup.sh pstn-trunk' update mode\n"
- "; never touches this file, only a fresh reinstall does.\n"
- "; Any extension not listed here is internal-only (no PSTN) by default.\n\n"
- )
- cp.write(f)
- os.replace(tmp_path, path)
- except OSError as e:
- try:
- os.remove(tmp_path)
- except OSError:
- pass
- return False, "Failed writing %s: %s" % (path, e)
+ ok, err = _write_ini_cp(_permissions_path(), PERMISSIONS_HEADER, cp)
+ if not ok:
+ return False, err
if tier == "restricted" and not clean_numbers:
return True, "Saved as restricted with an EMPTY approved list — no PSTN number can reach/be reached by it yet."
@@ -1083,6 +1123,118 @@ def write_limits(max_outbound, max_inbound):
return True, "Saved"
+PERSONAL_DID_RE = re.compile(r"^\d{10}$")
+
+
+def _personal_dids_path():
+ return os.path.join(ASTERISK_CONFIG_DIR, "pstn-personal-dids.conf") if ASTERISK_CONFIG_DIR else None
+
+
+def _read_personal_dids_cp():
+ cp = configparser.ConfigParser(delimiters=("=",))
+ path = _personal_dids_path()
+ if path and os.path.isfile(path):
+ try:
+ cp.read(path)
+ except configparser.Error:
+ pass
+ return cp
+
+
+def list_personal_dids():
+ """[{"did": ..., "owner": ...}] for every currently-assigned personal
+ DID, sorted by DID."""
+ cp = _read_personal_dids_cp()
+ result = []
+ for section in cp.sections():
+ if not PERSONAL_DID_RE.match(section):
+ continue
+ result.append({"did": section, "owner": cp.get(section, "owner", fallback="")})
+ result.sort(key=lambda d: d["did"])
+ return result
+
+
+def write_personal_did(did, owner):
+ """Assigns did -> owner, keeping pstn-personal-dids.conf (inbound
+ routing, read by the dialplan) and pstn-permissions.conf's
+ personal_did= (outbound Caller-ID override) in sync. One owner has at
+ most one personal_did (AST_CONFIG() returns a single value per key), so
+ reassigning a DID to a new owner drops the previous owner's claim on
+ it, and giving an extension a new personal DID drops whichever one it
+ had before — this always leaves a clean 1:1 mapping in both files,
+ rather than requiring the caller to clean up the old assignment
+ itself."""
+ if not ASTERISK_CONFIG_DIR:
+ return False, "No Asterisk install detected on this box"
+ did = str(did).strip()
+ owner = str(owner).strip()
+ if not PERSONAL_DID_RE.match(did):
+ return False, "DID must be a 10-digit US number"
+ if not EXTEN_RE.match(owner):
+ return False, "Invalid owner extension"
+
+ dids_cp = _read_personal_dids_cp()
+ perms_cp = _read_permissions_cp()
+
+ for section in perms_cp.sections():
+ if section != owner and perms_cp.get(section, "personal_did", fallback="") == did:
+ perms_cp.remove_option(section, "personal_did")
+ if not perms_cp.options(section):
+ perms_cp.remove_section(section)
+
+ for section in list(dids_cp.sections()):
+ if section != did and dids_cp.get(section, "owner", fallback="") == owner:
+ dids_cp.remove_section(section)
+
+ if not dids_cp.has_section(did):
+ dids_cp.add_section(did)
+ dids_cp.set(did, "owner", owner)
+
+ if not perms_cp.has_section(owner):
+ perms_cp.add_section(owner)
+ perms_cp.set(owner, "personal_did", did)
+
+ ok, err = _write_ini_cp(_personal_dids_path(), PERSONAL_DIDS_HEADER, dids_cp)
+ if not ok:
+ return False, err
+ ok, err = _write_ini_cp(_permissions_path(), PERMISSIONS_HEADER, perms_cp)
+ if not ok:
+ return False, err
+
+ owner_tier = perms_cp.get(owner, "tier", fallback="internal")
+ if owner_tier not in ("full", "restricted"):
+ return True, "Assigned %s to extension %s - note: %s is internal-tier, so it won't actually receive calls on this DID until you also grant it full or restricted tier." % (did, owner, owner)
+ return True, "Assigned %s to extension %s" % (did, owner)
+
+
+def remove_personal_did(did):
+ if not ASTERISK_CONFIG_DIR:
+ return False, "No Asterisk install detected on this box"
+ did = str(did).strip()
+ if not PERSONAL_DID_RE.match(did):
+ return False, "Invalid DID"
+
+ dids_cp = _read_personal_dids_cp()
+ perms_cp = _read_permissions_cp()
+
+ if dids_cp.has_section(did):
+ dids_cp.remove_section(did)
+
+ for section in perms_cp.sections():
+ if perms_cp.get(section, "personal_did", fallback="") == did:
+ perms_cp.remove_option(section, "personal_did")
+ if not perms_cp.options(section):
+ perms_cp.remove_section(section)
+
+ ok, err = _write_ini_cp(_personal_dids_path(), PERSONAL_DIDS_HEADER, dids_cp)
+ if not ok:
+ return False, err
+ ok, err = _write_ini_cp(_permissions_path(), PERMISSIONS_HEADER, perms_cp)
+ if not ok:
+ return False, err
+ return True, "Removed %s" % did
+
+
INDEX_HTML = """
Security Dashboard
@@ -1181,6 +1333,19 @@ INDEX_HTML = """
Ext
Name
Tier
Approved numbers (restricted only)
+
+
Personal numbers
+
+ Multiple DIDs can share this one trunk. Assigning a DID to an extension routes inbound calls to that DID straight to its owner (still gated by the owner's own tier/approved-numbers above — no ring-group fallback), and makes that extension's outbound calls show this DID as Caller-ID instead of the shared trunk DID. The shared DID/ring-group keeps working regardless.
+
No personal numbers assigned — every extension shares the main trunk DID.
";
+}
+
+document.getElementById("pd-save").addEventListener("click", async () => {
+ const did = document.getElementById("pd-did").value.trim();
+ const owner = document.getElementById("pd-owner").value;
+ const res = await fetch("/api/pstn-personal-dids", {
+ method: "POST", headers: {"Content-Type": "application/json"},
+ body: JSON.stringify({did: did, owner: owner}),
+ });
+ const data = await res.json();
+ document.getElementById("pd-msg").textContent = data.message || (data.ok ? "Saved" : "Failed");
+ if (data.ok) document.getElementById("pd-did").value = "";
+ loadPersonalDids();
+});
+
+async function removePersonalDid(did) {
+ if (!confirm("Remove personal number " + did + "? Its owner falls back to the shared trunk DID for outbound Caller-ID, and this DID stops routing anywhere until reassigned.")) return;
+ const res = await fetch("/api/pstn-personal-dids/delete", {
+ method: "POST", headers: {"Content-Type": "application/json"},
+ body: JSON.stringify({did: did}),
+ });
+ const data = await res.json();
+ document.getElementById("pd-msg").textContent = data.message || (data.ok ? "Removed" : "Failed");
+ loadPersonalDids();
+}
+
const adminUrl = "__ASTERISK_ADMIN_URL__";
if (adminUrl) {
const link = document.getElementById("admin-link");
@@ -1469,6 +1675,10 @@ class Handler(BaseHTTPRequestHandler):
self._json({"extensions": extensions})
elif self.path == "/api/pstn-limits":
self._json(get_limits())
+ elif self.path == "/api/pstn-personal-dids":
+ names = {e["ext"]: e["name"] for e in list_extensions()}
+ dids = [dict(d, owner_name=names.get(d["owner"], "")) for d in list_personal_dids()]
+ self._json({"dids": dids})
elif self.path == "/api/pstn-status":
self._json({"installed": pstn_installed()})
else:
@@ -1498,6 +1708,12 @@ class Handler(BaseHTTPRequestHandler):
elif self.path == "/api/pstn-limits":
ok, message = write_limits(payload.get("max_outbound", ""), payload.get("max_inbound", ""))
self._json({"ok": ok, "message": message})
+ elif self.path == "/api/pstn-personal-dids":
+ ok, message = write_personal_did(payload.get("did", ""), payload.get("owner", ""))
+ self._json({"ok": ok, "message": message})
+ elif self.path == "/api/pstn-personal-dids/delete":
+ ok, message = remove_personal_did(payload.get("did", ""))
+ self._json({"ok": ok, "message": message})
else:
self._json({"error": "not found"}, 404)
From 3705fb5fcced22c191c431e44c76be306687f929 Mon Sep 17 00:00:00 2001
From: Claude
Date: Thu, 23 Jul 2026 05:08:30 +0000
Subject: [PATCH 2/4] Add Anveo Direct/VoIP.ms known-provider quick-pick to
pstn-trunk.sh
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
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
---
docs/pstn-calling-voipms-plan.md | 27 ++++++++++++++++
services/pstn-trunk.sh | 55 +++++++++++++++++++++++++++++---
2 files changed, 77 insertions(+), 5 deletions(-)
diff --git a/docs/pstn-calling-voipms-plan.md b/docs/pstn-calling-voipms-plan.md
index e82ad8b..c6b54d4 100644
--- a/docs/pstn-calling-voipms-plan.md
+++ b/docs/pstn-calling-voipms-plan.md
@@ -504,3 +504,30 @@ generator output. Fixed by quoting every value in that heredoc.
`messaging=yes` or `personal_did=` already on that extension — fixed
to remove only the tier/allowed_numbers keys, dropping the section
only once nothing else is left in it.
+16. Known-provider quick-pick — Done. A "1) Anveo Direct / 2) VoIP.ms /
+ 3) manual" choice at the top of the provider prompts pre-fills known
+ values (Anveo Direct: `sbc.anveo.com`, the 4 published signaling IPs;
+ VoIP.ms: just the provider name) — every value stays editable at each
+ prompt, so this changes defaults only, never behavior. Extensions/
+ users remain fully independent of this service either way — they're
+ created through the base Asterisk install's own device management,
+ and `pstn-trunk.sh` only grants PSTN permissions/personal numbers to
+ extensions that already exist.
+17. **Open, unresolved**: Anveo Direct's own "Outbound Trunks" configuration
+ page documents outbound dialing as `[PREFIX]PHONENUMBER@sbc.anveo.com`
+ — a per-trunk custom prefix. This directly contradicts the earlier
+ "no dial prefix needed" finding sourced from their FAQ (see the Anveo
+ Direct provider notes above). Not resolved either way: whether the
+ Prefix field can be left blank when creating an outbound Call
+ Termination trunk in their portal. This dialplan dials the bare
+ number with no prefix, matching the FAQ-sourced finding — if the
+ Prefix field turns out to require a non-empty value, outbound calls
+ through that trunk won't match and will fail, and this installer
+ would need a `PREFIX` setting threaded into `${EXTEN}` before
+ `Dial()`. Flagged with a loud warning in the CLI's Anveo Direct
+ quick-pick rather than silently trusting the older finding. Also
+ unresolved: a Trial Anveo Direct account shows its own "$2 / 30 days"
+ spending limit and "$2 minimum account balance" in the portal —
+ unclear whether either changes once the account is verified/funded
+ beyond Trial status; this sits below and independent of anything this
+ repo's own kill-switch enforces.
diff --git a/services/pstn-trunk.sh b/services/pstn-trunk.sh
index 6188bbb..a2f1978 100644
--- a/services/pstn-trunk.sh
+++ b/services/pstn-trunk.sh
@@ -1177,10 +1177,13 @@ install_pstn-trunk() {
if [ "$DRY_RUN" = true ]; then
echo "[DRY-RUN] Would require an existing asterisk-digital-ocean OR asterisk (LAN) install"
- echo "[DRY-RUN] Would prompt for: SIP provider name (any IP-authenticated provider), server/POP hostname, DID,"
+ echo "[DRY-RUN] Would prompt for: known-provider quick-pick (Anveo Direct/VoIP.ms pre-fill known"
+ echo "[DRY-RUN] server/signaling-IP values; still editable) or manual entry, SIP provider name, DID,"
echo "[DRY-RUN] full-PSTN extensions, restricted-PSTN extensions + their approved numbers,"
echo "[DRY-RUN] internal SIP messaging extensions (separate from PSTN calling permission),"
- echo "[DRY-RUN] max concurrent outbound/inbound calls (default 10/10), inbound ring-group extensions,"
+ echo "[DRY-RUN] optional personal-number assignments (DID -> owner extension, additive to the"
+ echo "[DRY-RUN] shared trunk DID), max concurrent outbound/inbound calls (default 10/10),"
+ echo "[DRY-RUN] inbound ring-group extensions,"
echo "[DRY-RUN] ntfy alert topic (optional), international-calling allow-list (CLI-only,"
echo "[DRY-RUN] always asked, never on the web dashboard), per-minute rate + monthly/hourly"
echo "[DRY-RUN] alert thresholds, and an optional hard monthly spend-cap kill-switch"
@@ -1282,11 +1285,53 @@ install_pstn-trunk() {
# ── Prompts — provider account details aren't scriptable, set up manually
# on the provider's own site first (see warning above) ───────────────────
+ # Known-provider quick-path: only pre-fills the defaults below (still
+ # fully editable at each prompt) — doesn't change any dialplan/auth
+ # behavior, which stays provider-generic either way.
+ echo " Known/tested providers — this only pre-fills the defaults below (server"
+ echo " hostname, signaling IPs, account-setup reminders); every value is still"
+ echo " editable at each prompt."
+ echo " 1) Anveo Direct"
+ echo " 2) VoIP.ms"
+ echo " 3) Something else / manual entry"
+ local _provider_choice=""
+ prompt_text "Choice (1/2/3):" "3" _provider_choice
+
+ local _default_provider_name="" _default_server="" _default_extra_ips=""
+ case "$_provider_choice" in
+ 1)
+ _default_provider_name="Anveo Direct"
+ _default_server="sbc.anveo.com"
+ _default_extra_ips="169.48.232.158 204.216.109.55 176.9.39.206 72.9.149.25"
+ echo ""
+ log_info "Anveo Direct known values pre-filled below (hostname, 4 published signaling IPs)."
+ log_warning "Before continuing: create/verify your account at"
+ log_warning " https://www.anveo.com/account.asp?account_type=direct"
+ log_warning "fund it, order a DID, and configure at least one outbound Call"
+ log_warning "Termination trunk in Anveo's own portal (Outbound Trunks page). A Trial"
+ log_warning "account shows its own \$2/30-day spend limit and \$2 minimum balance —"
+ log_warning "separate from anything this installer enforces; confirm with Anveo"
+ log_warning "support whether those change once verified/funded. Their CallerID policy"
+ log_warning "requires a verified/Anveo-owned number as Caller-ID — any DID you order"
+ log_warning "through them satisfies that automatically (see this service's personal-"
+ log_warning "number feature for assigning a specific one per extension)."
+ log_warning "UNVERIFIED: Anveo's own outbound-trunk page documents dialing as"
+ log_warning "[PREFIX]PHONENUMBER@sbc.anveo.com (a per-trunk prefix) — this contradicts"
+ log_warning "an earlier no-prefix-needed finding from their FAQ. This dialplan dials"
+ log_warning "the bare number, no prefix. Check whether your configured trunk's Prefix"
+ log_warning "field can be left blank before relying on this — if it can't, outbound"
+ log_warning "calls through that trunk won't match and will fail."
+ ;;
+ 2)
+ _default_provider_name="VoIP.ms"
+ ;;
+ esac
+
local PROVIDER_NAME=""
- prompt_text "SIP trunk provider name (for your reference/docs only — e.g. VoIP.ms, Anveo Direct):" "" PROVIDER_NAME
+ prompt_text "SIP trunk provider name (for your reference/docs only — e.g. VoIP.ms, Anveo Direct):" "$_default_provider_name" PROVIDER_NAME
local TRUNK_SERVER=""
- prompt_text "Server/POP hostname (e.g. atlanta2.voip.ms for VoIP.ms — pick the one closest to this box from your provider's server list):" "" TRUNK_SERVER
+ prompt_text "Server/POP hostname (e.g. atlanta2.voip.ms for VoIP.ms, sbc.anveo.com for Anveo Direct — pick the one closest to this box from your provider's server list):" "$_default_server" TRUNK_SERVER
if [[ -z "$TRUNK_SERVER" ]]; then
log_error "A server hostname is required — aborting."
return 1
@@ -1311,7 +1356,7 @@ install_pstn-trunk() {
# dial out to) — the resolved IP above always gets included, this just
# adds any others the provider documents.
local EXTRA_IPS=""
- prompt_text "Any additional known source IPs for inbound calls, space-separated (check your provider's docs — e.g. a firewall/signaling IP list; blank if the resolved IP above is the only one):" "" EXTRA_IPS
+ prompt_text "Any additional known source IPs for inbound calls, space-separated (check your provider's docs — e.g. a firewall/signaling IP list; blank if the resolved IP above is the only one):" "$_default_extra_ips" EXTRA_IPS
local _octet='(25[0-5]|2[0-4][0-9]|[01]?[0-9][0-9]?)'
local _ip_re="^${_octet}\\.${_octet}\\.${_octet}\\.${_octet}\$"
local TRUNK_SERVER_IPS="$TRUNK_SERVER_IP" _ip
From 8291eba55e2bef2784fb9f4fe191646e7252b49f Mon Sep 17 00:00:00 2001
From: Claude
Date: Thu, 23 Jul 2026 05:27:58 +0000
Subject: [PATCH 3/4] Make the messaging permission flag live-editable via the
dashboard
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
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
---
docs/pstn-calling-voipms-plan.md | 11 ++++-
services/pstn-trunk.sh | 32 +++++++++----
services/security-dashboard.sh | 78 +++++++++++++++++++++-----------
3 files changed, 83 insertions(+), 38 deletions(-)
diff --git a/docs/pstn-calling-voipms-plan.md b/docs/pstn-calling-voipms-plan.md
index c6b54d4..ec6b488 100644
--- a/docs/pstn-calling-voipms-plan.md
+++ b/docs/pstn-calling-voipms-plan.md
@@ -416,8 +416,15 @@ generator output. Fixed by quoting every value in that heredoc.
`messaging=yes` flag per extension in `pstn-permissions.conf`,
independent of the PSTN calling tiers (an extension can be
internal-tier for calling and still messaging-enabled, or vice versa),
- prompted at install time. **Not done**: the actual dialplan wiring that
- would make Asterisk *enforce* this flag on inbound `MESSAGE` requests.
+ prompted at install time AND now a checkbox right in the Security
+ Dashboard's PSTN Trunk permissions table (alongside tier/approved-
+ numbers) — no need to re-run the CLI installer just to change who can
+ message. Confirmed it correctly survives tier changes and personal-DID
+ assignment/removal on the same extension (this is what surfaced the
+ tier=internal section-wipe bug fixed above). **Not done**: the actual
+ dialplan wiring that would make Asterisk *enforce* this flag on
+ inbound `MESSAGE` requests — this flag currently does nothing at the
+ Asterisk level yet, it's groundwork.
Reasoned through but deliberately not shipped: Easy Asterisk dispatches
messages through the same `[intercom]` context calls use (no
`message_context` override), and whether a hand-written pattern there
diff --git a/services/pstn-trunk.sh b/services/pstn-trunk.sh
index a2f1978..10dec8c 100644
--- a/services/pstn-trunk.sh
+++ b/services/pstn-trunk.sh
@@ -1816,17 +1816,29 @@ Asterisk's native SIP \`MESSAGE\` support (extension-to-extension texting —
no carrier SMS, no PSTN, no cost) is gated by a \`messaging=yes\` flag per
extension in \`pstn-permissions.conf\`, independent of the PSTN calling
tiers above — off by default, same "opt in" posture. Currently enabled for:
-${MESSAGING_EXTS:-none}.
+${MESSAGING_EXTS:-none}. Live-editable any time via the Security
+Dashboard's "PSTN Trunk" tab (a checkbox per extension, right in the same
+table as the calling tiers) — no need to re-run this installer just to
+change who can message.
-**Known gap:** this installer writes the permission flag (live-editable,
-same mechanism as the calling tiers), but the actual SIP \`MESSAGE\` routing
-dialplan wiring depends on how Easy Asterisk's own generated
-\`extensions.conf\`/\`pjsip.conf\` route inbound messages, which needs to be
-verified against a live install before it's safely automated here — shipping
-a guessed pattern risked either silently not working or interfering with
-call-routing precedence in the same \`[intercom]\` context. Treat the
-permission flag as ready for a dashboard/CLI-managed allow-list once that
-routing is confirmed, not as fully wired yet.
+**Won't show up in Easy Asterisk's own web admin, by design** — same as
+the PSTN calling tiers, this is a permission this repo layers on top,
+not an Easy Asterisk feature, so it's only manageable here or via the
+Security Dashboard.
+
+**Known gap:** the flag above is real and live-editable, but the actual
+SIP \`MESSAGE\` routing dialplan wiring — does Asterisk actually deliver/
+gate a message using this flag — depends on how Easy Asterisk's own
+generated \`extensions.conf\`/\`pjsip.conf\` route inbound messages, which
+needs to be verified against a live install before it's safely automated
+here. Shipping a guessed pattern risked either silently not working or
+interfering with call-routing precedence in the same \`[intercom]\`
+context, so it hasn't been guessed at. If you want this working end to
+end, the fastest path is checking a few things on a live box (e.g.
+whether an endpoint has \`message_context\` set, and what happens when you
+send a test SIP MESSAGE to one) so the dialplan gate can be built against
+real behavior instead of assumption — ask if you want to walk through
+that.
## Personal numbers
diff --git a/services/security-dashboard.sh b/services/security-dashboard.sh
index 6bb851a..c58328b 100644
--- a/services/security-dashboard.sh
+++ b/services/security-dashboard.sh
@@ -214,13 +214,17 @@ not in Docker — it needs to call \`cscli\` and read Asterisk's log directly.
never shows real-looking-but-unenforced defaults. When installed: the
outbound/inbound concurrent-call caps, and every known extension (parsed
from \`pjsip.conf\`) with its current permission tier (internal /
- restricted / full) and, for restricted, its approved numbers — all
- editable live, no Asterisk restart, no reinstall. Writes directly to
- \`pstn-limits.conf\` / \`pstn-permissions.conf\`, which the dialplan reads
- fresh on every call. The spend-cap kill-switch and international-calling
- allow-list are deliberately **not** managed here — CLI-only, via
- \`sudo ./setup.sh pstn-trunk\` — since both are more security-sensitive
- than what this tab already exposes.
+ restricted / full), for restricted its approved numbers, and its
+ internal-SIP-messaging flag (independent of the calling tier — see
+ \`services/pstn-trunk.sh\`'s "Known gap" note on messaging for what this
+ flag does and doesn't do yet) — all editable live, no Asterisk restart,
+ no reinstall. Also manages personal-number assignments (DID -> owner
+ extension), additive to the shared trunk DID. Writes directly to
+ \`pstn-limits.conf\` / \`pstn-permissions.conf\` / \`pstn-personal-dids.conf\`,
+ which the dialplan reads fresh on every call. The spend-cap kill-switch
+ and international-calling allow-list are deliberately **not** managed
+ here — CLI-only, via \`sudo ./setup.sh pstn-trunk\` — since both are more
+ security-sensitive than what this tab already exposes.
- Link to the Asterisk web admin itself (doesn't embed it, just links out).
## Manage
@@ -986,12 +990,13 @@ def _read_permissions_cp():
def get_all_permissions():
- """{ext: {"tier": ..., "allowed_numbers": "num|num|..."}} for every
- extension with a non-internal tier on record. Extensions with no section
- are implicitly "internal" — the dialplan's AST_CONFIG() lookup treats a
- missing section as empty/denied the same way, so there's nothing to
- return for them here; the UI fills in "internal" as the default for any
- known extension (from list_extensions()) not present in this dict."""
+ """{ext: {"tier": ..., "allowed_numbers": "num|num|...", "messaging":
+ bool}} for every extension with a non-default record. Extensions with
+ no section are implicitly "internal"/messaging-disabled — the
+ dialplan's AST_CONFIG() lookup treats a missing section/key as empty/
+ denied the same way, so there's nothing to return for them here; the
+ UI fills in the defaults for any known extension (from
+ list_extensions()) not present in this dict."""
cp = _read_permissions_cp()
result = {}
for section in cp.sections():
@@ -1000,12 +1005,17 @@ def get_all_permissions():
result[section] = {
"tier": cp.get(section, "tier", fallback="internal"),
"allowed_numbers": cp.get(section, "allowed_numbers", fallback=""),
+ "messaging": cp.getboolean(section, "messaging", fallback=False),
}
return result
-def write_permission(ext, tier, numbers_raw):
- """Saves one extension's tier + (for restricted) approved-number list.
+def write_permission(ext, tier, numbers_raw, messaging_enabled=False):
+ """Saves one extension's tier + (for restricted) approved-number list +
+ messaging flag in one action — messaging is an independent axis from
+ the calling tier (see pstn-trunk.sh's file-level comment: an extension
+ can be internal-tier for calling and still messaging-enabled, or vice
+ versa), so it's set/cleared regardless of which tier branch runs below.
Numbers are normalized to a pipe-separated list of 11-digit US numbers —
pipe, not comma, because the dialplan uses this value directly as a
REGEX() alternation pattern (see services/pstn-trunk.sh's file-level
@@ -1030,15 +1040,12 @@ def write_permission(ext, tier, numbers_raw):
# personal_did assigned, and those must survive a tier change back
# to internal. Confirmed live as a real bug: cp.remove_section(ext)
# here used to silently discard both whenever tier was set to
- # internal. Only remove the section itself once nothing else is
- # left in it.
+ # internal.
if cp.has_section(ext):
if cp.has_option(ext, "tier"):
cp.remove_option(ext, "tier")
if cp.has_option(ext, "allowed_numbers"):
cp.remove_option(ext, "allowed_numbers")
- if not cp.options(ext):
- cp.remove_section(ext)
else:
if not cp.has_section(ext):
cp.add_section(ext)
@@ -1048,6 +1055,19 @@ def write_permission(ext, tier, numbers_raw):
elif cp.has_option(ext, "allowed_numbers"):
cp.remove_option(ext, "allowed_numbers")
+ if messaging_enabled:
+ if not cp.has_section(ext):
+ cp.add_section(ext)
+ cp.set(ext, "messaging", "yes")
+ elif cp.has_section(ext) and cp.has_option(ext, "messaging"):
+ cp.remove_option(ext, "messaging")
+
+ # Drop the section entirely once nothing (tier, numbers, messaging,
+ # personal_did) is left in it — only reached this way when tier is
+ # internal, messaging is off, and no personal_did was ever assigned.
+ if cp.has_section(ext) and not cp.options(ext):
+ cp.remove_section(ext)
+
ok, err = _write_ini_cp(_permissions_path(), PERMISSIONS_HEADER, cp)
if not ok:
return False, err
@@ -1330,7 +1350,10 @@ INDEX_HTML = """
full — internal, plus any US number.
Changes apply live, on the next call — no Asterisk restart needed.
-
Ext
Name
Tier
Approved numbers (restricted only)
+
+ Messaging — 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.
+
+
Ext
Name
Tier
Approved numbers (restricted only)
Messaging
@@ -1544,7 +1567,7 @@ async function loadPstnPermissions() {
const tbody = document.querySelector("#pstn-table tbody");
if (!exts.length) {
- tbody.innerHTML = '
No extensions found (no Asterisk install detected, or pjsip.conf has no devices yet).
';
+ tbody.innerHTML = '
No extensions found (no Asterisk install detected, or pjsip.conf has no devices yet).
';
return;
}
tbody.innerHTML = exts.map(e => `
@@ -1558,6 +1581,7 @@ async function loadPstnPermissions() {
+
`).join("");
@@ -1572,9 +1596,10 @@ async function savePstnPermission(ext) {
const row = document.querySelector(`#pstn-table tr[data-ext="${ext}"]`);
const tier = row.querySelector(".pstn-tier").value;
const numbers = row.querySelector(".pstn-numbers").value;
+ const messaging = row.querySelector(".pstn-messaging").checked;
const res = await fetch("/api/pstn-permissions", {
method: "POST", headers: {"Content-Type": "application/json"},
- body: JSON.stringify({ext: ext, tier: tier, allowed_numbers: numbers}),
+ body: JSON.stringify({ext: ext, tier: tier, allowed_numbers: numbers, messaging: messaging}),
});
const data = await res.json();
document.getElementById("pstn-msg").textContent = (data.message || (data.ok ? "Saved" : "Failed")) + " (extension " + ext + ")";
@@ -1669,9 +1694,9 @@ class Handler(BaseHTTPRequestHandler):
perms = get_all_permissions()
extensions = []
for e in list_extensions():
- p = perms.get(e["ext"], {"tier": "internal", "allowed_numbers": ""})
- extensions.append({"ext": e["ext"], "name": e["name"],
- "tier": p["tier"], "allowed_numbers": p["allowed_numbers"]})
+ p = perms.get(e["ext"], {"tier": "internal", "allowed_numbers": "", "messaging": False})
+ extensions.append({"ext": e["ext"], "name": e["name"], "tier": p["tier"],
+ "allowed_numbers": p["allowed_numbers"], "messaging": p["messaging"]})
self._json({"extensions": extensions})
elif self.path == "/api/pstn-limits":
self._json(get_limits())
@@ -1702,7 +1727,8 @@ class Handler(BaseHTTPRequestHandler):
self._json(ban_asn(payload.get("asn", "")))
elif self.path == "/api/pstn-permissions":
ok, message = write_permission(
- payload.get("ext", ""), payload.get("tier", ""), payload.get("allowed_numbers", "")
+ payload.get("ext", ""), payload.get("tier", ""), payload.get("allowed_numbers", ""),
+ bool(payload.get("messaging", False))
)
self._json({"ok": ok, "message": message})
elif self.path == "/api/pstn-limits":
From 7851e2befd1611d54231da143826e6dd7aabd554 Mon Sep 17 00:00:00 2001
From: Claude
Date: Thu, 23 Jul 2026 05:38:36 +0000
Subject: [PATCH 4/4] Decouple internal SIP messaging management from PSTN
trunk installation
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
Claude-Session: https://claude.ai/code/session_01Ho9mZgAkVpdz7S5wJkg8Nf
---
services/pstn-trunk.sh | 7 ++-
services/security-dashboard.sh | 106 ++++++++++++++++++++++++++++-----
2 files changed, 96 insertions(+), 17 deletions(-)
diff --git a/services/pstn-trunk.sh b/services/pstn-trunk.sh
index 10dec8c..cb1c23b 100644
--- a/services/pstn-trunk.sh
+++ b/services/pstn-trunk.sh
@@ -1817,9 +1817,10 @@ no carrier SMS, no PSTN, no cost) is gated by a \`messaging=yes\` flag per
extension in \`pstn-permissions.conf\`, independent of the PSTN calling
tiers above — off by default, same "opt in" posture. Currently enabled for:
${MESSAGING_EXTS:-none}. Live-editable any time via the Security
-Dashboard's "PSTN Trunk" tab (a checkbox per extension, right in the same
-table as the calling tiers) — no need to re-run this installer just to
-change who can message.
+Dashboard's "PSTN Trunk" tab, in its own always-available "Internal SIP
+messaging" card — no need to re-run this installer, and no dependency on
+this trunk (or any PSTN trunk at all) being installed, unlike the
+calling-permissions table below it in that same tab.
**Won't show up in Easy Asterisk's own web admin, by design** — same as
the PSTN calling tiers, this is a permission this repo layers on top,
diff --git a/services/security-dashboard.sh b/services/security-dashboard.sh
index c58328b..6daa0db 100644
--- a/services/security-dashboard.sh
+++ b/services/security-dashboard.sh
@@ -208,18 +208,22 @@ not in Docker — it needs to call \`cscli\` and read Asterisk's log directly.
- **Unwhitelist + Ban** does that *and* immediately bans (24h) every IP
CrowdSec has ever recorded for that ASN, for accidental-whitelist cases
where you don't want to wait for it to misbehave again.
-- **PSTN Trunk** — detects whether \`services/pstn-trunk.sh\`'s dialplan is
- actually installed (\`pstn-trunk-dialplan.conf\` present) and shows a clear
- "not installed" message instead of the caps/tiers editor if not, so it
- never shows real-looking-but-unenforced defaults. When installed: the
- outbound/inbound concurrent-call caps, and every known extension (parsed
- from \`pjsip.conf\`) with its current permission tier (internal /
- restricted / full), for restricted its approved numbers, and its
- internal-SIP-messaging flag (independent of the calling tier — see
- \`services/pstn-trunk.sh\`'s "Known gap" note on messaging for what this
- flag does and doesn't do yet) — all editable live, no Asterisk restart,
- no reinstall. Also manages personal-number assignments (DID -> owner
- extension), additive to the shared trunk DID. Writes directly to
+- **PSTN Trunk** — an "Internal SIP messaging" card at the top is always
+ available, whether or not a PSTN trunk has ever been installed: a
+ checkbox per known extension (parsed from \`pjsip.conf\`) for
+ Asterisk's native SIP texting, independent of PSTN calling entirely (no
+ cost, no carrier, no DID, no dependency on \`services/pstn-trunk.sh\`
+ having been run — see its "Known gap" note on messaging for what this
+ flag does and doesn't do yet at the Asterisk level). Below that, the
+ rest of the tab detects whether \`services/pstn-trunk.sh\`'s dialplan is
+ actually installed (\`pstn-trunk-dialplan.conf\` present) and shows a
+ clear "not installed" message instead of the calling-permissions editor
+ if not, so it never shows real-looking-but-unenforced defaults. When
+ installed: the outbound/inbound concurrent-call caps, and every known
+ extension's permission tier (internal / restricted / full) and, for
+ restricted, its approved numbers — all editable live, no Asterisk
+ restart, no reinstall. Also manages personal-number assignments (DID ->
+ owner extension), additive to the shared trunk DID. Writes directly to
\`pstn-limits.conf\` / \`pstn-permissions.conf\` / \`pstn-personal-dids.conf\`,
which the dialplan reads fresh on every call. The spend-cap kill-switch
and international-calling allow-list are deliberately **not** managed
@@ -1077,6 +1081,37 @@ def write_permission(ext, tier, numbers_raw, messaging_enabled=False):
return True, "Saved"
+def write_messaging(ext, enabled):
+ """Sets/clears just the messaging flag for one extension, leaving any
+ tier/allowed_numbers/personal_did untouched. This is the write path for
+ the standalone "Internal SIP messaging" card, which works whether or
+ not a PSTN trunk has ever been installed — messaging has no dependency
+ on one (no cost, no carrier, no DID), unlike the calling-permissions
+ table this dashboard otherwise gates behind pstn_installed(). Creates
+ pstn-permissions.conf from scratch if it doesn't exist yet."""
+ if not ASTERISK_CONFIG_DIR:
+ return False, "No Asterisk install detected on this box"
+ ext = str(ext).strip()
+ if not EXTEN_RE.match(ext):
+ return False, "Invalid extension"
+
+ cp = _read_permissions_cp()
+ if enabled:
+ if not cp.has_section(ext):
+ cp.add_section(ext)
+ cp.set(ext, "messaging", "yes")
+ elif cp.has_section(ext) and cp.has_option(ext, "messaging"):
+ cp.remove_option(ext, "messaging")
+
+ if cp.has_section(ext) and not cp.options(ext):
+ cp.remove_section(ext)
+
+ ok, err = _write_ini_cp(_permissions_path(), PERMISSIONS_HEADER, cp)
+ if not ok:
+ return False, err
+ return True, "Saved"
+
+
LIMIT_RE = re.compile(r"^\d+$")
@@ -1327,9 +1362,17 @@ INDEX_HTML = """
+
+
Internal SIP messaging
+
+ 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.
+
+
Ext
Name
Enabled
+
+
PSTN trunk not installed
-
No PSTN trunk dialplan was found on this box — sudo ./setup.sh pstn-trunk hasn't been run (or its config was removed). Nothing below is enforced yet; install it first, then this tab will manage the real permission tiers and concurrency caps.
+
No PSTN trunk dialplan was found on this box — sudo ./setup.sh pstn-trunk hasn't been run (or its config was removed). The calling permissions/caps/personal-numbers below aren't enforced yet; install it first to use them. Internal SIP messaging above works independently of this.
@@ -1381,7 +1424,7 @@ document.querySelectorAll(".tab-btn").forEach(btn => {
document.querySelectorAll(".tab-btn").forEach(b => b.classList.remove("active"));
btn.classList.add("active");
TABS.forEach(t => { document.getElementById("tab-" + t).style.display = btn.dataset.tab === t ? "" : "none"; });
- if (btn.dataset.tab === "pstn") { loadPstnStatus(); }
+ if (btn.dataset.tab === "pstn") { loadPstnStatus(); loadMessaging(); }
});
});
@@ -1544,6 +1587,38 @@ async function loadPstnLimits() {
document.getElementById("limit-in").value = data.max_inbound;
}
+// Independent of pstn_installed() — messaging has no dependency on a PSTN
+// trunk existing, unlike everything else in this tab, so this loads/saves
+// regardless of whether services/pstn-trunk.sh has ever been run.
+async function loadMessaging() {
+ const res = await fetch("/api/pstn-permissions");
+ const data = await res.json();
+ const exts = data.extensions || [];
+ const tbody = document.querySelector("#msg-table tbody");
+ if (!exts.length) {
+ tbody.innerHTML = '
No extensions found (no Asterisk install detected, or pjsip.conf has no devices yet).