Merge pull request #243 from outis1one/claude/asterisk-caller-id-number-y1b3ls
Add SMS-over-SIP inbound (Anveo MESSAGE method), diagnostic-first
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Outis
+100
-1
@@ -191,6 +191,22 @@ _pstn_ensure_live_includes() {
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fi
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fi
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# SMS-over-SIP's own context — same reasoning as above (own #include, not
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# sharing a file), anchored on the voice-inbound include so all three
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# trunk includes sit together.
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if [[ -f "$EXT_LIVE" ]] && ! grep -q 'pstn-sms-inbound-dialplan.conf' "$EXT_LIVE"; then
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if grep -q '^#include pstn-trunk-inbound-dialplan.conf$' "$EXT_LIVE"; then
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sed -i '/^#include pstn-trunk-inbound-dialplan.conf$/a #include pstn-sms-inbound-dialplan.conf' "$EXT_LIVE"
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log_success "Patched the SMS-over-SIP #include directly into the live extensions.conf."
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elif grep -q '^\[intercom\]$' "$EXT_LIVE"; then
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sed -i '/^\[intercom\]$/a #include pstn-sms-inbound-dialplan.conf' "$EXT_LIVE"
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log_success "Patched the SMS-over-SIP #include directly into the live extensions.conf."
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else
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log_warning "Couldn't find an anchor for the SMS-over-SIP include in the live extensions.conf — add"
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log_warning "'#include pstn-sms-inbound-dialplan.conf' manually, then reload."
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fi
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fi
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docker exec "$CONTAINER_NAME" asterisk -rx "module reload res_pjsip.so" &>/dev/null || true
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docker exec "$CONTAINER_NAME" asterisk -rx "dialplan reload" &>/dev/null || true
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@@ -268,6 +284,17 @@ _pstn_write_pjsip_include() {
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; personal_did extension's outbound calls still show the shared trunk DID
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; after this, the provider needs the CID delivered a different way (e.g.
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; a per-DID sub-account or registration) and this will need revisiting.
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;
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; message_context=pstn-sms-inbound below is the SMS-over-SIP counterpart to
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; services/asterisk.sh's message_context=sip-messaging for internal
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; texting — same reason: without a dedicated context, an inbound SIP
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; MESSAGE would fall back to this endpoint's own context=from-pstn-trunk,
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; which already owns "_X." for INVITE-based inbound calls, and a competing
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; declaration there risks breaking that. Only meaningful if
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; _pstn_write_sms_inbound_dialplan has actually written that context —
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; harmless (message_context pointing at a context with no matching
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; extension just gets Asterisk's own default handling) if SMS-over-SIP
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; was never configured on this box.
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[pstn-trunk]
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type=aor
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@@ -296,10 +323,57 @@ trust_id_inbound=yes
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send_pai=yes
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send_rpid=yes
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direct_media=no
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message_context=pstn-sms-inbound
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EOF
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sed -i "s/__PSTN_SERVER__/${SERVER}/g; s/__PSTN_DID__/${DID}/g" "$FILE"
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}
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# ── Shared: SMS-over-SIP inbound (Anveo "SMS over SIP", MESSAGE method) ────
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# DELIBERATELY diagnostic-first, not routing-first. Anveo's own docs for
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# this feature are thin (confirmed live, same session: their SMSbySMS HTTP
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# docs described a dead endpoint, and their general-API docs disagreed with
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# each other on version number, host path AND auth model — see
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# docs/anveo-direct-setup-guide.md's SMS section history). Given that track
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# record, this does NOT assume X-ANVEO-SMS-FROM/X-ANVEO-SMS-TO behave
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# exactly as documented — it logs every plausible source for the sender
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# number, destination DID, and body, and stops (Hangup, same terminator the
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# existing internal [sip-messaging] context uses), so the FIRST real test
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# text shows exactly what Anveo actually sends before anything tries to
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# parse and route it automatically. Once a real message has been observed
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# in the log, the extraction/routing logic (owner lookup via
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# pstn-personal-dids.conf, ring-group fan-out) gets built against confirmed
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# fields instead of the docs' word for it.
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#
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# message_context=pstn-sms-inbound on the trunk endpoint (see
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# _pstn_write_pjsip_include) is what routes an inbound MESSAGE here instead
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# of falling back to context=from-pstn-trunk, which already owns "_X." for
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# INVITE-based inbound calls — same reasoning as
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# services/asterisk.sh's message_context=sip-messaging for internal texting,
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# just the external-provider counterpart of it.
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_pstn_write_sms_inbound_dialplan() {
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local FILE="$1"
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cat > "$FILE" << 'EOF'
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; SMS-over-SIP inbound (Anveo MESSAGE method) — services/pstn-trunk.sh.
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; Regenerated on every install/update; edit there, not here directly.
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;
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; Diagnostic-first on purpose — see _pstn_write_sms_inbound_dialplan's own
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; comment in pstn-trunk.sh. This ONLY logs what arrived; it does not yet
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; deliver anything into a mailbox/extension. Send a real test text, then:
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; docker exec <container> asterisk -rx "core set verbose 5"
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; (send the text)
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; check the console/full log for the "SMS-over-SIP MESSAGE received" line
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[pstn-sms-inbound]
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exten => _X.,1,NoOp(SMS-over-SIP MESSAGE received — dialplan EXTEN=${EXTEN})
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same => n,Set(SMS_HDR_FROM=${PJSIP_HEADER(read,X-ANVEO-SMS-FROM)})
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same => n,Set(SMS_HDR_TO=${PJSIP_HEADER(read,X-ANVEO-SMS-TO)})
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same => n,Set(SMS_MSG_FROM=${MESSAGE(from)})
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same => n,Set(SMS_MSG_TO=${MESSAGE(to)})
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same => n,Set(SMS_BODY=${MESSAGE(body)})
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same => n,NoOp(DIAGNOSTIC (not yet routed) -- X-ANVEO-SMS-FROM=[${SMS_HDR_FROM}] X-ANVEO-SMS-TO=[${SMS_HDR_TO}] MESSAGE(from)=[${SMS_MSG_FROM}] MESSAGE(to)=[${SMS_MSG_TO}] EXTEN=[${EXTEN}] body=[${SMS_BODY}])
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same => n,Hangup()
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EOF
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}
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# ── Shared: one inbound ring-group member's live permission check ─────────
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# Emits a block that only adds this extension to PSTN_RING_LIST if it's
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# "full" INBOUND tier, or "restricted" inbound tier AND the caller ID is on
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@@ -1616,6 +1690,7 @@ _pstn_apply_settings() {
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_pstn_write_pjsip_include "$ASTERISK_DIR/pstn-trunk-pjsip.conf" "$SERVER" "$SERVER_IPS" "$DID"
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_pstn_write_dialplan_include "$ASTERISK_DIR/pstn-trunk-dialplan.conf" "$DID" "$NTFY_URL"
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_pstn_write_inbound_dialplan_include "$ASTERISK_DIR/pstn-trunk-inbound-dialplan.conf" "$RING_EXTS" "$NTFY_URL"
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_pstn_write_sms_inbound_dialplan "$ASTERISK_DIR/pstn-sms-inbound-dialplan.conf"
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_pstn_write_personal_group_ring_script "$ASTERISK_DIR/pstn-personal-group-ring.sh"
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# Must run alongside the dialplan write, not after a reload: the dialplan
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# above reads tier_out/tier_in, so an un-migrated permissions file would
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@@ -1625,7 +1700,7 @@ _pstn_apply_settings() {
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"$RATE" "$MONTH_THRESHOLD" "$BURST_THRESHOLD" "$MAX_MONTHLY_SPEND" "$NTFY_URL" "$CONTAINER_NAME"
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ensure_docker_dir_ownership "$ASTERISK_DIR"
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chmod 644 "$ASTERISK_DIR/pstn-trunk-pjsip.conf" "$ASTERISK_DIR/pstn-trunk-dialplan.conf" \
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"$ASTERISK_DIR/pstn-trunk-inbound-dialplan.conf"
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"$ASTERISK_DIR/pstn-trunk-inbound-dialplan.conf" "$ASTERISK_DIR/pstn-sms-inbound-dialplan.conf"
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chmod 755 "$ASTERISK_DIR/pstn-personal-group-ring.sh"
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# Values are double-quoted: this file is `source`d back in on "update"
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@@ -2490,4 +2565,28 @@ MD
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log_info "via the Security Dashboard's PSTN Trunk tab — sudo ./setup.sh security-dashboard"
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log_info "if it isn't installed yet."
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echo ""
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# SMS-over-SIP (Anveo, MESSAGE method) — reuses whatever FQDN
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# services/asterisk.sh already set up for SIP (the same one the voice
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# trunk answers on), read straight from its own .env rather than
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# prompting again for something already configured.
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local _SMS_DOMAIN=""
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[[ -f "$EA_DIR/.env" ]] && _SMS_DOMAIN="$(grep -E '^DOMAIN_NAME=' "$EA_DIR/.env" | cut -d= -f2-)"
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if [[ -n "$_SMS_DOMAIN" ]]; then
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log_info "SMS over SIP (optional, Anveo-specific — if your provider supports it):"
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log_info " In the DID's SMS tab, choose the MESSAGE method and set the SIP URI to:"
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echo " smshandler@${_SMS_DOMAIN}:5060"
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log_info " That's the same FQDN already set up for this box's SIP (Anveo requires a"
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log_info " literal host:port here — no DNS SRV support — and ignores registration"
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log_info " entirely for this delivery method)."
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log_info " This only logs what arrives for now, on purpose — send one real test text"
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log_info " after configuring it, then check:"
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echo " docker exec $CONTAINER_NAME asterisk -rx \"core set verbose 5\""
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log_info " and look for the 'SMS-over-SIP MESSAGE received' line to confirm which"
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log_info " headers Anveo actually sends before anything tries to route it."
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else
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log_warning "SMS over SIP: no DOMAIN_NAME found in $EA_DIR/.env — this box has no FQDN"
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log_warning "set up for SIP, so there's no domain to give Anveo for smshandler@...:5060."
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fi
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echo ""
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}
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