v1.25.0

feat(proxy-engine): add live TTS streaming interactions and incoming number range support
v1.24.0
2026-04-14 18:52:13 +00:00 · 2026-04-14 18:52:13 +00:00 · 2026-04-14 16:35:54 +00:00 · 2026-04-14 16:35:54 +00:00 · 2026-04-14 10:45:59 +00:00 · 2026-04-14 10:45:59 +00:00
53 changed files with 4946 additions and 2262 deletions
--- a/CLAUDE.md
+++ b/CLAUDE.md
@@ -1,41 +1,103 @@
 # Project Notes

-## Architecture: Hub Model (Call as Centerpiece)
+## Architecture: Hub Model in Rust (Call as Centerpiece)

-All call logic lives in `ts/call/`. The Call is the central entity with N legs.
+The call hub lives in the Rust proxy-engine (`rust/crates/proxy-engine/`). TypeScript is the **control plane only** — it configures the engine, sends high-level commands (`hangup`, `make_call`, `webrtc_offer`, etc.), and receives events (`incoming_call`, `call_answered`, `device_registered`, `webrtc_audio_rx`, …). No raw SIP/RTP ever touches TypeScript.

-### Key Files
- `ts/call/call-manager.ts` — singleton registry, factory methods, SIP routing
- `ts/call/call.ts` — the hub: owns legs, media forwarding
- `ts/call/sip-leg.ts` — SIP device/provider connection (wraps SipDialog)
- `ts/call/webrtc-leg.ts` — browser WebRTC connection (wraps werift PeerConnection)
- `ts/call/rtp-port-pool.ts` — unified RTP port pool
- `ts/sipproxy.ts` — thin bootstrap wiring everything together
- `ts/webrtcbridge.ts` — browser device registration (signaling only)
+The `Call` is still the central entity: it owns N legs and a central mixer task that provides mix-minus audio to all participants. Legs can be `SipProvider`, `SipDevice`, `WebRtc`, or `Tool` (recording/transcription observer).

-### WebRTC Browser Call Flow (Critical)
+### Key Rust files (`rust/crates/proxy-engine/src/`)

-The browser call flow has a specific signaling order that MUST be followed:
+- `call_manager.rs` — singleton registry, call factory methods, SIP routing (inbound/outbound/passthrough), B2BUA state machine, inbound route resolution
+- `call.rs` — the `Call` hub + `LegInfo` struct, owns legs and the mixer task
+- `sip_leg.rs` — full SIP dialog management for B2BUA legs (INVITE, 407 auth retry, BYE, CANCEL, early media)
+- `rtp.rs` — RTP port pool (uses `Weak<UdpSocket>` so calls auto-release ports on drop) + RTP header helpers
+- `mixer.rs` — 20 ms-tick mix-minus engine (48 kHz f32 internal, per-leg transcoding via `codec-lib`, per-leg denoising)
+- `jitter_buffer.rs` — per-leg reordering/packet-loss compensation
+- `leg_io.rs` — spawns inbound/outbound RTP I/O tasks per SIP leg
+- `webrtc_engine.rs` — browser WebRTC sessions (werift-rs based), ICE/DTLS/SRTP
+- `provider.rs` — SIP trunk registrations, public-IP detection via Via `received=`
+- `registrar.rs` — accepts REGISTER from SIP phones, tracks contacts (push-based device status)
+- `config.rs` — `AppConfig` deserialized from TS, route resolvers (`resolve_outbound_route`, `resolve_inbound_route`)
+- `main.rs` — IPC command dispatcher (`handle_command`), event emitter, top-level SIP packet router
+- `sip_transport.rs` — owning wrapper around the main SIP UDP socket
+- `voicemail.rs` / `recorder.rs` / `audio_player.rs` / `tts.rs` — media subsystems
+- `tool_leg.rs` — per-source observer audio for recording/transcription tools
+- `ipc.rs` — event-emission helper used throughout

-1. `POST /api/call` with browser deviceId → CallManager creates Call, saves pending state, notifies browser via `webrtc-incoming`
-2. Browser sends `webrtc-offer` (with its own `sessionId`) → CallManager creates a **standalone** WebRtcLeg (NOT attached to any call yet)
-3. Browser sends `webrtc-accept` (with `callId` + `sessionId`) → CallManager links the standalone WebRtcLeg to the Call, then starts the SIP provider leg
+### Key TS files (control plane)

-**The WebRtcLeg CANNOT be created at call creation time** because the browser's session ID is unknown until the `webrtc-offer` arrives.
+- `ts/sipproxy.ts` — entrypoint, wires the proxy engine bridge + web UI + WebRTC signaling
+- `ts/proxybridge.ts` — `@push.rocks/smartrust` bridge to the Rust binary, typed `TProxyCommands` map
+- `ts/config.ts` — JSON config loader (`IAppConfig`, `IProviderConfig`, etc.), sent to Rust via `configure`
+- `ts/voicebox.ts` — voicemail metadata persistence (WAV files live in `.nogit/voicemail/{boxId}/`)
+- `ts/webrtcbridge.ts` — browser WebSocket signaling, browser device registry (`deviceIdToWs`)
+- `ts/call/prompt-cache.ts` — the only remaining file under `ts/call/` (IVR prompt caching)

-### WebRTC Audio Return Channel (Critical)
+### Rust SIP protocol library

-The SIP→browser audio path works through the Call hub:
+`rust/crates/sip-proto/` is a zero-dependency SIP data library (parse/build/mutate/serialize messages, dialog management, SDP helpers, digest auth). Do not add transport or timer logic there — it's purely data-level.

-1. Provider sends RTP to SipLeg's socket
-2. SipLeg's `onRtpReceived` fires → Call hub's `forwardRtp`
-3. Call hub calls `webrtcLeg.sendRtp(data)` → which calls `forwardToBrowser()`
-4. `forwardToBrowser` transcodes (G.722→Opus) and sends via `sender.sendRtp()` (WebRTC PeerConnection)
+## Event-push architecture for device status

-**`WebRtcLeg.sendRtp()` MUST feed into `forwardToBrowser()`** (the WebRTC PeerConnection path), NOT send to a UDP address. This was a bug that caused one-way audio.
+Device status flows **via push events**, not pull-based IPC queries:

-The browser→SIP direction works independently: `ontrack.onReceiveRtp` → `forwardToSip()` → transcodes → sends directly to provider's media endpoint via UDP.
+1. Rust emits `device_registered` when a phone REGISTERs
+2. TS `sipproxy.ts` maintains a `deviceStatuses` Map, updated from the event
+3. Map snapshot goes into the WebSocket `status` broadcast
+4. Web UI (`ts_web/elements/sipproxy-devices.ts`) reads it from the push stream

-### SIP Protocol Library
+There used to be a `get_status` pull IPC for this, but it was never called from TS and has been removed. If a new dashboard ever needs a pull-based snapshot, the push Map is the right source to read from.

-`ts/sip/` is a zero-dependency SIP protocol library. Do not add transport or timer logic there — it's purely data-level (parse/build/mutate/serialize).
+## Inbound routing (wired in Commit 4 of the cleanup PR)
+
+Inbound route resolution goes through `config.resolve_inbound_route(provider_id, called_number, caller_number)` inside `create_inbound_call` (call_manager.rs). The result carries a `ring_browsers` flag that propagates to the `incoming_call` event; `ts/sipproxy.ts` gates the `webrtc-incoming` browser fan-out behind that flag.
+
+**Known limitations / TODOs** (documented in code at `create_inbound_call`):
+- Multi-target inbound fork is not yet implemented — only the first registered device from `route.device_ids` is rung.
+- `ring_browsers` is **informational only**: browsers see a toast but do not race the SIP device to answer. True first-to-answer-wins requires a multi-leg fork + per-leg CANCEL, which is not built yet.
+- `voicemail_box`, `ivr_menu_id`, `no_answer_timeout` are resolved but not yet honored downstream.
+
+## WebRTC Browser Call Flow (Critical)
+
+The browser call signaling order is strict:
+
+1. Browser initiates outbound via a TS API (e.g. `POST /api/call`) — TS creates a pending call in the Rust engine via `make_call` and notifies the browser with a `webrtc-incoming` push.
+2. Browser sends `webrtc-offer` (with its own `sessionId`) → Rust `handle_webrtc_offer` creates a **standalone** WebRTC session (NOT attached to any call yet).
+3. Browser sends `webrtc_link` (with `callId` + `sessionId`) → Rust links the standalone session to the Call and wires the WebRTC leg through the mixer.
+
+**The WebRTC leg cannot be fully attached at call-creation time** because the browser's session ID is unknown until the `webrtc-offer` arrives.
+
+### WebRTC audio return channel (Critical)
+
+The SIP→browser audio path goes through the mixer, not a direct RTP relay:
+
+1. Provider sends RTP → received on the provider leg's UDP socket (`leg_io::spawn_sip_inbound`)
+2. Packet flows through `jitter_buffer` → mixer's inbound mpsc channel
+3. Mixer decodes/resamples/denoises, computes mix-minus per leg
+4. WebRTC leg receives its mix-minus frame, encodes to Opus, and pushes via the WebRTC engine's peer connection sender
+
+Browser→SIP works symmetrically: `ontrack.onReceiveRtp` → WebRTC leg's outbound mpsc → mixer → other legs' inbound channels.
+
+## SDP/Record-Route NAT (fixed in Commit 3 of the cleanup PR)
+
+The proxy tracks a `public_ip: Option<String>` on every `LegInfo` (populated from provider-leg construction sites). When `route_passthrough_message` rewrites SDP (`c=` line) or emits a `Record-Route`, it picks `advertise_ip` based on the destination leg's kind:
+
+- `SipProvider` → `other.public_ip.unwrap_or(lan_ip)` (provider reaches us via public IP)
+- `SipDevice` / `WebRtc` / `Tool` / `Media` → `lan_ip` (everything else is LAN or proxy-internal)
+
+This fixed a real NAT-traversal bug where the proxy advertised its RFC1918 LAN IP to the provider in SDP, causing one-way or no audio for device-originated inbound traffic behind NAT.
+
+## Build & development
+
+- **Build:** `pnpm run buildRust` (never `cargo build` directly — tsrust cross-compiles for both `x86_64-unknown-linux-gnu` and `aarch64-unknown-linux-gnu`)
+- **Cross-compile setup:** the aarch64 target requires `gcc-aarch64-linux-gnu` + `libstdc++6-arm64-cross` (Debian/Ubuntu). See `rust/.cargo/config.toml` for the linker wiring. A committed symlink at `rust/.cargo/crosslibs/aarch64/libstdc++.so` → `/usr/aarch64-linux-gnu/lib/libstdc++.so.6` avoids needing the `libstdc++-13-dev-arm64-cross` package.
+- **Bundle web UI:** `pnpm run bundle` (esbuild, output: `dist_ts_web/bundle.js`)
+- **Full build:** `pnpm run build` (= `buildRust && bundle`)
+- **Start server:** `pnpm run start` (runs `tsx ts/sipproxy.ts`)
+
+## Persistent files
+
+- `.nogit/config.json` — app config (providers, devices, routes, voiceboxes, IVR menus)
+- `.nogit/voicemail/{boxId}/` — voicemail WAV files + `messages.json` index
+- `.nogit/prompts/` — cached TTS prompts for IVR menus
--- a/changelog.md
+++ b/changelog.md
@@ -1,5 +1,66 @@
 # Changelog

+## 2026-04-14 - 1.25.0 - feat(proxy-engine)
+add live TTS streaming interactions and incoming number range support
+
+- add a new start_tts_interaction command and bridge API to begin IVR or leg interactions before full TTS rendering completes
+- stream synthesized TTS chunks into the mixer with cancellation handling so prompts can stop cleanly on digit match, leg removal, or shutdown
+- extract PCM-to-mixer frame conversion for reusable live prompt processing
+- extend routing pattern matching to support numeric number ranges like start..end, including + prefixed values
+- add incomingNumbers config typing and frontend config update support for single, range, and regex number modes
+
+## 2026-04-14 - 1.24.0 - feat(routing)
+require explicit inbound DID routes and normalize SIP identities for provider-based number matching
+
+- Inbound route resolution now returns no match unless a configured inbound route explicitly matches the provider and called number.
+- Normalized routing identities were added for SIP/TEL URIs so inbound DIDs and outbound dialed numbers match consistently across provider-specific formats.
+- Call handling and incoming call events now use normalized numbers, improving routing accuracy for shared trunk providers.
+- Route configuration docs and the web route editor were updated to support explicit inbound DID ownership, voicemail fallback, and IVR selection.
+- Mixer RTP handling was enhanced to better support variable packet durations, timestamp-based gap fill, and non-blocking output drop reporting.
+
+## 2026-04-14 - 1.23.0 - feat(runtime)
+refactor runtime state and proxy event handling for typed WebRTC linking and shared status models
+
+- extract proxy event handling into dedicated runtime modules for status tracking and WebRTC session-to-call linking
+- introduce shared typed proxy event and status interfaces used by both backend and web UI
+- update web UI server initialization to use structured options and await async config save hooks
+- simplify browser signaling by routing WebRTC offer/ICE handling through frontend-to-Rust integration
+- align device status rendering with the new address/port fields in dashboard views
+
+## 2026-04-12 - 1.22.0 - feat(proxy-engine)
+add on-demand TTS caching for voicemail and IVR prompts
+
+- Route inbound calls directly to configured IVR menus and track them with a dedicated IVR call state
+- Generate voicemail greetings and IVR menu prompts inside the Rust proxy engine on demand instead of precomputing prompts in TypeScript
+- Add cacheable TTS output with sidecar metadata and enable Kokoro CMUdict support for improved prompt generation
+- Extend proxy configuration to include voiceboxes and IVR menus, and update documentation to reflect Kokoro-only prompt generation
+
+## 2026-04-11 - 1.21.0 - feat(providers)
+replace provider creation modal with a guided multi-step setup flow
+
+- Adds a stepper-based provider creation flow with provider type selection, connection, credentials, advanced settings, and review steps.
+- Applies built-in templates for Sipgate and O2/Alice from the selected provider type instead of separate add actions.
+- Adds a final review step with generated provider ID preview and duplicate ID collision handling before saving.
+
+## 2026-04-11 - 1.20.5 - fix(readme)
+improve architecture and call flow documentation with Mermaid diagrams
+
+- Replace ASCII architecture and audio pipeline diagrams with Mermaid diagrams for better readability
+- Document the WebRTC browser call setup sequence, including offer handling and session-to-call linking
+
+## 2026-04-11 - 1.20.4 - fix(deps)
+bump @design.estate/dees-catalog to ^3.71.1
+
+- Updates the @design.estate/dees-catalog dependency from ^3.70.0 to ^3.71.1 in package.json.
+
+## 2026-04-11 - 1.20.3 - fix(ts-config,proxybridge,voicebox)
+align voicebox config types and add missing proxy bridge command definitions
+
+- Reuses the canonical IVoiceboxConfig type from voicebox.ts in config.ts to eliminate duplicated type definitions and optionality mismatches.
+- Makes voicemail timing and limits optional in voicebox config so defaults can be applied consistently during initialization.
+- Adds VoiceboxManager.addMessage and updates recording handling to use it directly for persisted voicemail metadata.
+- Extends proxy bridge command typings with add_leg, remove_leg, and WebRTC signaling commands, and tightens sendCommand typing.
+
 ## 2026-04-11 - 1.20.2 - fix(proxy-engine)
 fix inbound route browser ringing and provider-facing SDP advertisement while preventing RTP port exhaustion

--- a/package.json
+++ b/package.json
@@ -1,6 +1,6 @@
 {
  "name": "siprouter",
-  "version": "1.20.2",
+  "version": "1.25.0",
  "private": true,
  "type": "module",
  "scripts": {
@@ -13,7 +13,7 @@
    "restartBackground": "pnpm run buildRust && pnpm run bundle; test -f .server.pid && kill $(cat .server.pid) 2>/dev/null; sleep 1; rm -f sip_trace.log proxy.out && nohup tsx ts/sipproxy.ts > proxy.out 2>&1 & echo $! > .server.pid; sleep 2; cat proxy.out"
  },
  "dependencies": {
-    "@design.estate/dees-catalog": "^3.70.0",
+    "@design.estate/dees-catalog": "^3.77.0",
    "@design.estate/dees-element": "^2.2.4",
    "@push.rocks/smartrust": "^1.3.2",
    "@push.rocks/smartstate": "^2.3.0",
--- a/pnpm-lock.yaml
+++ b/pnpm-lock.yaml
@@ -9,8 +9,8 @@ importers:
  .:
    dependencies:
      '@design.estate/dees-catalog':
-        specifier: ^3.70.0
-        version: 3.70.0(@tiptap/pm@2.27.2)
+        specifier: ^3.77.0
+        version: 3.77.0(@tiptap/pm@2.27.2)
      '@design.estate/dees-element':
        specifier: ^2.2.4
        version: 2.2.4
@@ -81,8 +81,8 @@ packages:
  '@configvault.io/interfaces@1.0.17':
    resolution: {integrity: sha512-bEcCUR2VBDJsTin8HQh8Uw/mlYl2v8A3jMIaQ+MTB9Hrqd6CZL2dL7iJdWyFl/3EIX+LDxWFR+Oq7liIq7w+1Q==}

-  '@design.estate/dees-catalog@3.70.0':
-    resolution: {integrity: sha512-bNqOxxl83FNCCV+7QoUj6oeRC0VTExWOClrLrHNMoLIU0TCtzhcmQqiuJhdWrcCwZ5RBhXHGMSFsR62d2RcWpw==}
+  '@design.estate/dees-catalog@3.77.0':
+    resolution: {integrity: sha512-2IfvH390WXCF733XcmEcUP9skqogTz9xlqQw5PUJZy0u2Hf6+hJTyQOi4mcKmhpTE/kCpaD51uw21Lr4ncW6cg==}

  '@design.estate/dees-comms@1.0.30':
    resolution: {integrity: sha512-KchMlklJfKAjQiJiR0xmofXtQ27VgZtBIxcMwPE9d+h3jJRv+lPZxzBQVOM0eyM0uS44S5vJMZ11IeV4uDXSHg==}
@@ -93,8 +93,8 @@ packages:
  '@design.estate/dees-element@2.2.4':
    resolution: {integrity: sha512-O9cA6flBMMd+pBwMQrZXwAWel9yVxgokolb+Em6gvkXxPJ0P/B5UDn4Vc2d4ts3ta55PTBm+l2dPeDVGx/bl7Q==}

-  '@design.estate/dees-wcctools@3.8.0':
-    resolution: {integrity: sha512-CC14iVKUrguzD9jIrdPBd9fZ4egVJEZMxl5y8iy0l7WLumeoYvGsoXj5INVkRPLRVLqziIdi4Je1hXqHt2NU+g==}
+  '@design.estate/dees-wcctools@3.8.4':
+    resolution: {integrity: sha512-KpFK/azK+a/Xpq33pXKcho+tdFKVHhKZM5ArvHqo9QMwTczgp5DZZgowTDUuqAofjZwnuVfCPHK/Pw9e64N46A==}

  '@emnapi/core@1.9.2':
    resolution: {integrity: sha512-UC+ZhH3XtczQYfOlu3lNEkdW/p4dsJ1r/bP7H8+rhao3TTTMO1ATq/4DdIi23XuGoFY+Cz0JmCbdVl0hz9jZcA==}
@@ -1566,8 +1566,8 @@ packages:
  humanize-ms@1.2.1:
    resolution: {integrity: sha1-xG4xWaKT9riW2ikxbYtv6Lt5u+0=}

-  ibantools@4.5.2:
-    resolution: {integrity: sha512-is+8TgZcKS/AMv/z9nW1zz0bhjhoyjpA1p0nc3A6GkW/InOdcQiUZpkufADzh/aO/LY/TOD/P3oPWncNRn5QMA==}
+  ibantools@4.5.4:
+    resolution: {integrity: sha512-6jX1gh4aH6XH+o0ey+wtkMTzkcvsEta7DakIOZSng9voZYpMw3U+gK1+tZChk3aRcPcloEt0NOzksjaRZiqXbw==}

  iconv-lite@0.4.24:
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@@ -1694,8 +1694,8 @@ packages:
    resolution: {integrity: sha512-JvNw9Y81y33E+BEYPr0U7omo+U9AySnsMsEiXgwT6yqd31VQWTLNQqmT4ou5eqPFUrTfIDFta2wKhB1hyohtAQ==}
    engines: {node: 20 || >=22}

-  lucide@0.577.0:
-    resolution: {integrity: sha512-PpC/m5eOItp/WU/GlQPFBXDOhq6HibL73KzYP37OX3LM7VmzWQF8voEj8QRWUFvy9FIKfeDQkWYoyS1D/MdWFA==}
+  lucide@1.8.0:
+    resolution: {integrity: sha512-JjV/QnadgFLj1Pyu9IKl0lknrolFEzo04B64QcYLLeRzZl/iEHpdbSrRRKbyXcv45SZNv+WGjIUCT33e7xHO6Q==}

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@@ -2462,7 +2462,7 @@ snapshots:
      '@api.global/typedrequest-interfaces': 3.0.19
      '@api.global/typedsocket': 4.1.2(@push.rocks/smartserve@2.0.3)
      '@cloudflare/workers-types': 4.20260409.1
-      '@design.estate/dees-catalog': 3.70.0(@tiptap/pm@2.27.2)
+      '@design.estate/dees-catalog': 3.77.0(@tiptap/pm@2.27.2)
      '@design.estate/dees-comms': 1.0.30
      '@push.rocks/lik': 6.4.0
      '@push.rocks/smartdelay': 3.0.5
@@ -2529,11 +2529,11 @@ snapshots:
    dependencies:
      '@api.global/typedrequest-interfaces': 3.0.19

-  '@design.estate/dees-catalog@3.70.0(@tiptap/pm@2.27.2)':
+  '@design.estate/dees-catalog@3.77.0(@tiptap/pm@2.27.2)':
    dependencies:
      '@design.estate/dees-domtools': 2.5.4
      '@design.estate/dees-element': 2.2.4
-      '@design.estate/dees-wcctools': 3.8.0
+      '@design.estate/dees-wcctools': 3.8.4
      '@fortawesome/fontawesome-svg-core': 7.2.0
      '@fortawesome/free-brands-svg-icons': 7.2.0
      '@fortawesome/free-regular-svg-icons': 7.2.0
@@ -2551,9 +2551,9 @@ snapshots:
      '@tsclass/tsclass': 9.5.0
      echarts: 5.6.0
      highlight.js: 11.11.1
-      ibantools: 4.5.2
+      ibantools: 4.5.4
      lightweight-charts: 5.1.0
-      lucide: 0.577.0
+      lucide: 1.8.0
      monaco-editor: 0.55.1
      pdfjs-dist: 4.10.38
      xterm: 5.3.0
@@ -2610,7 +2610,7 @@ snapshots:
      - supports-color
      - vue

-  '@design.estate/dees-wcctools@3.8.0':
+  '@design.estate/dees-wcctools@3.8.4':
    dependencies:
      '@design.estate/dees-domtools': 2.5.4
      '@design.estate/dees-element': 2.2.4
@@ -4369,7 +4369,7 @@ snapshots:
    dependencies:
      ms: 2.1.3

-  ibantools@4.5.2: {}
+  ibantools@4.5.4: {}

  iconv-lite@0.4.24:
    dependencies:
@@ -4487,7 +4487,7 @@ snapshots:

  lru-cache@11.3.3: {}

-  lucide@0.577.0: {}
+  lucide@1.8.0: {}

  make-dir@3.1.0:
    dependencies:
--- a/readme.md
+++ b/readme.md
@@ -20,7 +20,7 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware
 - 🎯 **Adaptive Jitter Buffer** — Per-leg jitter buffering with sequence-based reordering, adaptive depth (60–120ms), Opus PLC for lost packets, and hold/resume detection
 - 📧 **Voicemail** — Configurable voicemail boxes with TTS greetings, recording, and web playback
 - 🔢 **IVR Menus** — DTMF-navigable interactive voice response with nested menus, routing actions, and custom prompts
- 🗣️ **Neural TTS** — Kokoro-powered announcements and greetings with 25+ voice presets, backed by espeak-ng fallback
+- 🗣️ **Neural TTS** — Kokoro-powered greetings and IVR prompts with 25+ voice presets
 - 🎙️ **Call Recording** — Per-source separated WAV recording at 48kHz via tool legs
 - 🖥️ **Web Dashboard** — Real-time SPA with 9 views: live calls, browser phone, routing, voicemail, IVR, contacts, providers, and streaming logs

@@ -28,39 +28,26 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware

 ## 🏗️ Architecture

-```
-┌─────────────────────────────────────┐
-│           Browser Softphone          │
-│     (WebRTC via WebSocket signaling) │
-└──────────────┬──────────────────────┘
-               │ Opus/WebRTC
-               ▼
-┌──────────────────────────────────────┐
-│            siprouter                  │
-│                                      │
-│  TypeScript Control Plane            │
-│  ┌────────────────────────────────┐  │
-│  │ Config · WebRTC Signaling      │  │
-│  │ REST API · Web Dashboard       │  │
-│  │ Voicebox Manager · TTS Cache   │  │
-│  └────────────┬───────────────────┘  │
-│          JSON-over-stdio IPC         │
-│  ┌────────────┴───────────────────┐  │
-│  │ Rust proxy-engine (data plane) │  │
-│  │                                │  │
-│  │ SIP Stack · Dialog SM · Auth   │  │
-│  │ Call Manager · N-Leg Mixer     │  │
-│  │ 48kHz f32 Bus · Jitter Buffer  │  │
-│  │ Codec Engine · RTP Port Pool   │  │
-│  │ WebRTC Engine · Kokoro TTS     │  │
-│  │ Voicemail · IVR · Recording    │  │
-│  └────┬──────────────────┬────────┘  │
-└───────┤──────────────────┤───────────┘
-        │                  │
- ┌──────┴──────┐    ┌──────┴──────┐
- │ SIP Devices │    │ SIP Trunk   │
- │ (HT801 etc) │    │ Providers   │
- └─────────────┘    └─────────────┘
+```mermaid
+flowchart TB
+    Browser["🌐 Browser Softphone<br/>(WebRTC via WebSocket signaling)"]
+    Devices["📞 SIP Devices<br/>(HT801, desk phones, ATAs)"]
+    Trunks["☎️ SIP Trunk Providers<br/>(sipgate, easybell, …)"]
+
+    subgraph Router["siprouter"]
+        direction TB
+        subgraph TS["TypeScript Control Plane"]
+            TSBits["Config · WebRTC Signaling<br/>REST API · Web Dashboard<br/>Voicebox Manager · TTS Cache"]
+        end
+        subgraph Rust["Rust proxy-engine (data plane)"]
+            RustBits["SIP Stack · Dialog SM · Auth<br/>Call Manager · N-Leg Mixer<br/>48kHz f32 Bus · Jitter Buffer<br/>Codec Engine · RTP Port Pool<br/>WebRTC Engine · Kokoro TTS<br/>Voicemail · IVR · Recording"]
+        end
+        TS <-->|"JSON-over-stdio IPC"| Rust
+    end
+
+    Browser <-->|"Opus / WebRTC"| TS
+    Rust <-->|"SIP / RTP"| Devices
+    Rust <-->|"SIP / RTP"| Trunks
 ```

 ### 🧠 Key Design Decisions
@@ -71,6 +58,37 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware
 - **Per-Session Codec Isolation** — Each call leg gets its own encoder/decoder/resampler/denoiser state — no cross-call corruption.
 - **SDP Codec Negotiation** — Outbound encoding uses the codec actually negotiated in SDP answers, not just the first offered codec.

+### 📲 WebRTC Browser Call Flow
+
+Browser calls are set up in a strict three-step dance — the WebRTC leg cannot be attached at call-creation time because the browser's session ID is only known once the SDP offer arrives:
+
+```mermaid
+sequenceDiagram
+    participant B as Browser
+    participant TS as TypeScript (sipproxy.ts)
+    participant R as Rust proxy-engine
+    participant P as SIP Provider
+
+    B->>TS: POST /api/call
+    TS->>R: make_call (pending call, no WebRTC leg yet)
+    R-->>TS: call_created
+    TS-->>B: webrtc-incoming (callId)
+
+    B->>TS: webrtc-offer (sessionId, SDP)
+    TS->>R: handle_webrtc_offer
+    R-->>TS: webrtc-answer (SDP)
+    TS-->>B: webrtc-answer
+    Note over R: Standalone WebRTC session<br/>(not yet attached to call)
+
+    B->>TS: webrtc_link (callId + sessionId)
+    TS->>R: link session → call
+    R->>R: wire WebRTC leg through mixer
+    R->>P: SIP INVITE
+    P-->>R: 200 OK + SDP
+    R-->>TS: call_answered
+    Note over B,P: Bidirectional Opus ↔ codec-transcoded<br/>audio flows through the mixer
+```
+
 ---

 ## 🚀 Getting Started
@@ -80,7 +98,6 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware
 - **Node.js** ≥ 20 with `tsx` globally available
 - **pnpm** for package management
 - **Rust** toolchain (for building the proxy engine)
- **espeak-ng** (optional, for TTS fallback)

 ### Install & Build

@@ -131,24 +148,41 @@ Create `.nogit/config.json`:
  "routing": {
    "routes": [
      {
-        "id": "inbound-default",
-        "name": "Ring all devices",
-        "priority": 100,
-        "direction": "inbound",
-        "match": {},
+        "id": "inbound-main-did",
+        "name": "Main DID",
+        "priority": 200,
+        "enabled": true,
+        "match": {
+          "direction": "inbound",
+          "sourceProvider": "my-trunk",
+          "numberPattern": "+49421219694"
+        },
        "action": {
          "targets": ["desk-phone"],
          "ringBrowsers": true,
-          "voicemailBox": "main",
-          "noAnswerTimeout": 25
+          "voicemailBox": "main"
+        }
+      },
+      {
+        "id": "inbound-support-did",
+        "name": "Support DID",
+        "priority": 190,
+        "enabled": true,
+        "match": {
+          "direction": "inbound",
+          "sourceProvider": "my-trunk",
+          "numberPattern": "+49421219695"
+        },
+        "action": {
+          "ivrMenuId": "support-menu"
        }
      },
      {
        "id": "outbound-default",
        "name": "Route via trunk",
        "priority": 100,
-        "direction": "outbound",
-        "match": {},
+        "enabled": true,
+        "match": { "direction": "outbound" },
        "action": { "provider": "my-trunk" }
      }
    ]
@@ -170,9 +204,11 @@ Create `.nogit/config.json`:
 }
 ```

+Inbound number ownership is explicit: add one inbound route per DID (or DID prefix) and scope it with `sourceProvider` when a provider delivers multiple external numbers.
+
 ### TTS Setup (Optional)

-For neural announcements and voicemail greetings, download the Kokoro TTS model:
+For neural voicemail greetings and IVR prompts, download the Kokoro TTS model:

 ```bash
 mkdir -p .nogit/tts
@@ -182,7 +218,7 @@ curl -L -o .nogit/tts/voices.bin \
  https://github.com/mzdk100/kokoro/releases/download/V1.0/voices.bin
 ```

-Without the model files, TTS falls back to `espeak-ng`. Without either, announcements are skipped — everything else works fine.
+Without the model files, TTS prompts (IVR menus, voicemail greetings) are skipped — everything else works fine.

 ### Run

@@ -209,7 +245,6 @@ siprouter/
 │   ├── frontend.ts                # Web dashboard HTTP/WS server + REST API
 │   ├── webrtcbridge.ts            # WebRTC signaling layer
 │   ├── registrar.ts               # Browser softphone registration
-│   ├── announcement.ts            # TTS announcement generator (espeak-ng / Kokoro)
 │   ├── voicebox.ts                # Voicemail box management
 │   └── call/
 │       └── prompt-cache.ts        # Named audio prompt WAV management
@@ -246,9 +281,17 @@ The `proxy-engine` binary handles all real-time audio processing with a **48kHz

 ### Audio Pipeline

-```
-Inbound:   Wire RTP → Jitter Buffer → Decode → Resample to 48kHz → Denoise (RNNoise) → Mix Bus
-Outbound:  Mix Bus → Mix-Minus → Resample to codec rate → Encode → Wire RTP
+```mermaid
+flowchart LR
+    subgraph Inbound["Inbound path (per leg)"]
+        direction LR
+        IN_RTP["Wire RTP"] --> IN_JB["Jitter Buffer"] --> IN_DEC["Decode"] --> IN_RS["Resample → 48 kHz"] --> IN_DN["Denoise (RNNoise)"] --> IN_BUS["Mix Bus"]
+    end
+
+    subgraph Outbound["Outbound path (per leg)"]
+        direction LR
+        OUT_BUS["Mix Bus"] --> OUT_MM["Mix-Minus"] --> OUT_RS["Resample → codec rate"] --> OUT_ENC["Encode"] --> OUT_RTP["Wire RTP"]
+    end
 ```

 - **Adaptive jitter buffer** — per-leg `BTreeMap`-based buffer keyed by RTP sequence number. Delivers exactly one frame per 20ms mixer tick in sequence order. Adaptive target depth starts at 3 frames (60ms) and adjusts between 2–6 frames based on observed network jitter. Handles hold/resume by detecting large forward sequence jumps and resetting cleanly.
@@ -262,13 +305,12 @@ Outbound:  Mix Bus → Mix-Minus → Resample to codec rate → Encode → Wire

 ## 🗣️ Neural TTS

-Announcements and voicemail greetings are synthesized using [Kokoro TTS](https://github.com/mzdk100/kokoro) — an 82M parameter neural model running via ONNX Runtime directly in the Rust process:
+Voicemail greetings and IVR prompts are synthesized using [Kokoro TTS](https://github.com/mzdk100/kokoro) — an 82M parameter neural model running via ONNX Runtime directly in the Rust process:

 - **24 kHz, 16-bit mono** output
 - **25+ voice presets** — American/British, male/female (e.g., `af_bella`, `am_adam`, `bf_emma`, `bm_george`)
 - **~800ms** synthesis time for a 3-second phrase
 - Lazy-loaded on first use — no startup cost if TTS is unused
- Falls back to `espeak-ng` if the ONNX model is not available

 ---

--- a/rust/.cargo/config.toml
+++ b/rust/.cargo/config.toml
@@ -0,0 +1,30 @@
+# Cross-compile configuration for the proxy-engine crate.
+#
+# tsrust builds for both x86_64-unknown-linux-gnu and aarch64-unknown-linux-gnu
+# from an x86_64 host. Without this config, cargo invokes the host `cc` to
+# link aarch64 objects and fails with
+#   rust-lld: error: <obj.o> is incompatible with elf64-x86-64
+#
+# Required Debian/Ubuntu packages for the aarch64 target to work:
+#   sudo apt install gcc-aarch64-linux-gnu g++-aarch64-linux-gnu \
+#                    libc6-dev-arm64-cross libstdc++6-arm64-cross
+#
+# The `libstdc++.so` dev symlink (needed by the -lstdc++ flag that the
+# kokoro-tts/ort build scripts emit) is provided by this repo at
+# ./crosslibs/aarch64/libstdc++.so, pointing at the versioned shared
+# library installed by `libstdc++6-arm64-cross`. This avoids requiring
+# the `libstdc++-13-dev-arm64-cross` package, which is not always
+# installed alongside the runtime.
+
+[target.aarch64-unknown-linux-gnu]
+linker = "aarch64-linux-gnu-gcc"
+rustflags = ["-C", "link-arg=-L.cargo/crosslibs/aarch64"]
+
+# Tell cc-rs-based build scripts (ring, zstd-sys, audiopus_sys, ort-sys) to
+# use the aarch64 cross toolchain when compiling C sources for the aarch64
+# target. Without these, they'd default to the host `cc` and produce x86_64
+# objects that the aarch64 linker then rejects.
+[env]
+CC_aarch64_unknown_linux_gnu = "aarch64-linux-gnu-gcc"
+CXX_aarch64_unknown_linux_gnu = "aarch64-linux-gnu-g++"
+AR_aarch64_unknown_linux_gnu = "aarch64-linux-gnu-ar"
--- a/rust/.cargo/crosslibs/aarch64/libstdc++.so
+++ b/rust/.cargo/crosslibs/aarch64/libstdc++.so
@@ -0,0 +1 @@
+/usr/aarch64-linux-gnu/lib/libstdc++.so.6
--- a/rust/Cargo.lock
+++ b/rust/Cargo.lock
@@ -532,6 +532,15 @@ dependencies = [
 "cc",
 ]

+[[package]]
+name = "cmudict-fast"
+version = "0.8.0"
+source = "registry+https://github.com/rust-lang/crates.io-index"
+checksum = "2c9f73004e928ed46c3e7fd7406d2b12c8674153295f08af084b49860276dc02"
+dependencies = [
+ "thiserror",
+]
+
 [[package]]
 name = "codec-lib"
 version = "0.1.0"
@@ -1730,6 +1739,7 @@ dependencies = [
 "bincode 2.0.1",
 "cc",
 "chinese-number",
+ "cmudict-fast",
 "futures",
 "jieba-rs",
 "log",
--- a/rust/crates/codec-lib/src/lib.rs
+++ b/rust/crates/codec-lib/src/lib.rs
@@ -115,9 +115,8 @@ pub struct TranscodeState {
 impl TranscodeState {
    /// Create a new transcoding session with fresh codec state.
    pub fn new() -> Result<Self, String> {
-        let mut opus_enc =
-            OpusEncoder::new(SampleRate::Hz48000, Channels::Mono, Application::Voip)
-                .map_err(|e| format!("opus encoder: {e}"))?;
+        let mut opus_enc = OpusEncoder::new(SampleRate::Hz48000, Channels::Mono, Application::Voip)
+            .map_err(|e| format!("opus encoder: {e}"))?;
        opus_enc
            .set_complexity(5)
            .map_err(|e| format!("opus set_complexity: {e}"))?;
@@ -160,14 +159,9 @@ impl TranscodeState {
        let key = (from_rate, to_rate, canonical_chunk);

        if !self.resamplers.contains_key(&key) {
-            let r = FftFixedIn::<f64>::new(
-                from_rate as usize,
-                to_rate as usize,
-                canonical_chunk,
-                1,
-                1,
-            )
-            .map_err(|e| format!("resampler {from_rate}->{to_rate}: {e}"))?;
+            let r =
+                FftFixedIn::<f64>::new(from_rate as usize, to_rate as usize, canonical_chunk, 1, 1)
+                    .map_err(|e| format!("resampler {from_rate}->{to_rate}: {e}"))?;
            self.resamplers.insert(key, r);
        }
        let resampler = self.resamplers.get_mut(&key).unwrap();
@@ -284,8 +278,7 @@ impl TranscodeState {
        match pt {
            PT_OPUS => {
                let mut pcm = vec![0i16; 5760]; // up to 120ms at 48kHz
-                let packet =
-                    OpusPacket::try_from(data).map_err(|e| format!("opus packet: {e}"))?;
+                let packet = OpusPacket::try_from(data).map_err(|e| format!("opus packet: {e}"))?;
                let out =
                    MutSignals::try_from(&mut pcm[..]).map_err(|e| format!("opus signals: {e}"))?;
                let n: usize = self
@@ -343,8 +336,7 @@ impl TranscodeState {
        match pt {
            PT_OPUS => {
                let mut pcm = vec![0.0f32; 5760]; // up to 120ms at 48kHz
-                let packet =
-                    OpusPacket::try_from(data).map_err(|e| format!("opus packet: {e}"))?;
+                let packet = OpusPacket::try_from(data).map_err(|e| format!("opus packet: {e}"))?;
                let out =
                    MutSignals::try_from(&mut pcm[..]).map_err(|e| format!("opus signals: {e}"))?;
                let n: usize = self
@@ -368,8 +360,8 @@ impl TranscodeState {
    /// Returns f32 PCM at 48kHz. `frame_size` should be 960 for 20ms.
    pub fn opus_plc(&mut self, frame_size: usize) -> Result<Vec<f32>, String> {
        let mut pcm = vec![0.0f32; frame_size];
-        let out = MutSignals::try_from(&mut pcm[..])
-            .map_err(|e| format!("opus plc signals: {e}"))?;
+        let out =
+            MutSignals::try_from(&mut pcm[..]).map_err(|e| format!("opus plc signals: {e}"))?;
        let n: usize = self
            .opus_dec
            .decode_float(None::<OpusPacket<'_>>, out, false)
@@ -425,14 +417,9 @@ impl TranscodeState {
        let key = (from_rate, to_rate, canonical_chunk);

        if !self.resamplers_f32.contains_key(&key) {
-            let r = FftFixedIn::<f32>::new(
-                from_rate as usize,
-                to_rate as usize,
-                canonical_chunk,
-                1,
-                1,
-            )
-            .map_err(|e| format!("resampler f32 {from_rate}->{to_rate}: {e}"))?;
+            let r =
+                FftFixedIn::<f32>::new(from_rate as usize, to_rate as usize, canonical_chunk, 1, 1)
+                    .map_err(|e| format!("resampler f32 {from_rate}->{to_rate}: {e}"))?;
            self.resamplers_f32.insert(key, r);
        }
        let resampler = self.resamplers_f32.get_mut(&key).unwrap();
@@ -508,8 +495,10 @@ mod tests {
            let encoded = mulaw_encode(sample);
            let decoded = mulaw_decode(encoded);
            // µ-law is lossy; verify the decoded value is close.
-            assert!((sample as i32 - decoded as i32).abs() < 1000,
-                "µ-law roundtrip failed for {sample}: got {decoded}");
+            assert!(
+                (sample as i32 - decoded as i32).abs() < 1000,
+                "µ-law roundtrip failed for {sample}: got {decoded}"
+            );
        }
    }

@@ -518,8 +507,10 @@ mod tests {
        for sample in [-32768i16, -1000, -1, 0, 1, 1000, 32767] {
            let encoded = alaw_encode(sample);
            let decoded = alaw_decode(encoded);
-            assert!((sample as i32 - decoded as i32).abs() < 1000,
-                "A-law roundtrip failed for {sample}: got {decoded}");
+            assert!(
+                (sample as i32 - decoded as i32).abs() < 1000,
+                "A-law roundtrip failed for {sample}: got {decoded}"
+            );
        }
    }

@@ -543,7 +534,9 @@ mod tests {
    fn pcmu_to_pcma_roundtrip() {
        let mut st = TranscodeState::new().unwrap();
        // 160 bytes = 20ms of PCMU at 8kHz
-        let pcmu_data: Vec<u8> = (0..160).map(|i| mulaw_encode((i as i16 * 200) - 16000)).collect();
+        let pcmu_data: Vec<u8> = (0..160)
+            .map(|i| mulaw_encode((i as i16 * 200) - 16000))
+            .collect();
        let pcma = st.transcode(&pcmu_data, PT_PCMU, PT_PCMA, None).unwrap();
        assert_eq!(pcma.len(), 160); // Same frame size
        let back = st.transcode(&pcma, PT_PCMA, PT_PCMU, None).unwrap();
--- a/rust/crates/proxy-engine/Cargo.toml
+++ b/rust/crates/proxy-engine/Cargo.toml
@@ -19,7 +19,7 @@ regex-lite = "0.1"
 webrtc = "0.8"
 rand = "0.8"
 hound = "3.5"
-kokoro-tts = { version = "0.3", default-features = false }
+kokoro-tts = { version = "0.3", default-features = false, features = ["use-cmudict"] }
 ort = { version = "=2.0.0-rc.11", default-features = false, features = [
    "std", "download-binaries", "copy-dylibs", "ndarray",
    "tls-native-vendored"
--- a/rust/crates/proxy-engine/src/audio_player.rs
+++ b/rust/crates/proxy-engine/src/audio_player.rs
@@ -36,10 +36,7 @@ pub async fn play_wav_file(

    // Read all samples as i16.
    let samples: Vec<i16> = if spec.bits_per_sample == 16 {
-        reader
-            .samples::<i16>()
-            .filter_map(|s| s.ok())
-            .collect()
+        reader.samples::<i16>().filter_map(|s| s.ok()).collect()
    } else if spec.bits_per_sample == 32 && spec.sample_format == hound::SampleFormat::Float {
        reader
            .samples::<f32>()
@@ -199,10 +196,7 @@ pub fn load_prompt_pcm_frames(wav_path: &str) -> Result<Vec<Vec<f32>>, String> {
            .map(|s| s as f32 / 32768.0)
            .collect()
    } else if spec.bits_per_sample == 32 && spec.sample_format == hound::SampleFormat::Float {
-        reader
-            .samples::<f32>()
-            .filter_map(|s| s.ok())
-            .collect()
+        reader.samples::<f32>().filter_map(|s| s.ok()).collect()
    } else {
        return Err(format!(
            "unsupported WAV format: {}bit {:?}",
@@ -214,14 +208,23 @@ pub fn load_prompt_pcm_frames(wav_path: &str) -> Result<Vec<Vec<f32>>, String> {
        return Ok(vec![]);
    }

+    pcm_to_mix_frames(&samples, wav_rate)
+}
+
+/// Convert PCM samples at an arbitrary rate into 48kHz 20ms mixer frames.
+pub fn pcm_to_mix_frames(samples: &[f32], sample_rate: u32) -> Result<Vec<Vec<f32>>, String> {
+    if samples.is_empty() {
+        return Ok(vec![]);
+    }
+
    // Resample to MIX_RATE (48kHz) if needed.
-    let resampled = if wav_rate != MIX_RATE {
+    let resampled = if sample_rate != MIX_RATE {
        let mut transcoder = TranscodeState::new().map_err(|e| format!("codec init: {e}"))?;
        transcoder
-            .resample_f32(&samples, wav_rate, MIX_RATE)
+            .resample_f32(samples, sample_rate, MIX_RATE)
            .map_err(|e| format!("resample: {e}"))?
    } else {
-        samples
+        samples.to_vec()
    };

    // Split into MIX_FRAME_SIZE (960) sample frames.
--- a/rust/crates/proxy-engine/src/call.rs
+++ b/rust/crates/proxy-engine/src/call.rs
@@ -23,6 +23,7 @@ pub enum CallState {
    Ringing,
    Connected,
    Voicemail,
+    Ivr,
    Terminated,
 }

@@ -37,6 +38,7 @@ impl CallState {
            Self::Ringing => "ringing",
            Self::Connected => "connected",
            Self::Voicemail => "voicemail",
+            Self::Ivr => "ivr",
            Self::Terminated => "terminated",
        }
    }
--- a/rust/crates/proxy-engine/src/call_manager.rs
+++ b/rust/crates/proxy-engine/src/call_manager.rs
@@ -5,24 +5,29 @@
 //! The mixer provides mix-minus audio to all participants.

 use crate::call::{Call, CallDirection, CallState, LegId, LegInfo, LegKind, LegState};
-use crate::config::{AppConfig, ProviderConfig};
+use crate::config::{normalize_routing_identity, AppConfig, ProviderConfig};
 use crate::ipc::{emit_event, OutTx};
 use crate::leg_io::{create_leg_channels, spawn_sip_inbound, spawn_sip_outbound};
 use crate::mixer::spawn_mixer;
 use crate::registrar::Registrar;
 use crate::rtp::RtpPortPool;
 use crate::sip_leg::{SipLeg, SipLegAction, SipLegConfig};
-use sip_proto::helpers::{build_sdp, generate_call_id, generate_tag, parse_sdp_endpoint, SdpOptions};
+use crate::tts::TtsEngine;
+use sip_proto::helpers::{
+    build_sdp, generate_call_id, generate_tag, parse_sdp_endpoint, SdpOptions,
+};
 use sip_proto::message::{ResponseOptions, SipMessage};
 use sip_proto::rewrite::{rewrite_sdp, rewrite_sip_uri};
 use std::collections::HashMap;
 use std::net::SocketAddr;
+use std::path::Path;
 use std::sync::Arc;
 use tokio::net::UdpSocket;
+use tokio::sync::Mutex;

 /// Result of creating an inbound call — carries both the call id and
 /// whether browsers should be notified (flows from the matched inbound
-/// route's `ring_browsers` flag, or the fallback default).
+/// route's `ring_browsers` flag).
 pub struct InboundCallCreated {
    pub call_id: String,
    pub ring_browsers: bool,
@@ -211,15 +216,27 @@ impl CallManager {
                        let device_leg = call.legs.values().find(|l| l.kind == LegKind::SipDevice);
                        if let Some(dev) = device_leg {
                            if let Some(dev_addr) = dev.signaling_addr {
-                                let ringing = SipMessage::create_response(180, "Ringing", device_invite, None);
+                                let ringing = SipMessage::create_response(
+                                    180,
+                                    "Ringing",
+                                    device_invite,
+                                    None,
+                                );
                                let _ = socket.send_to(&ringing.serialize(), dev_addr).await;
                            }
                        }
                    }
                }
-                emit_event(&self.out_tx, "call_ringing", serde_json::json!({ "call_id": call_id }));
-                emit_event(&self.out_tx, "leg_state_changed",
-                    serde_json::json!({ "call_id": call_id, "leg_id": leg_id, "state": "ringing" }));
+                emit_event(
+                    &self.out_tx,
+                    "call_ringing",
+                    serde_json::json!({ "call_id": call_id }),
+                );
+                emit_event(
+                    &self.out_tx,
+                    "leg_state_changed",
+                    serde_json::json!({ "call_id": call_id, "leg_id": leg_id, "state": "ringing" }),
+                );
            }
            SipLegAction::ConnectedWithAck(ack_buf) => {
                let _ = socket.send_to(&ack_buf, target).await;
@@ -242,16 +259,30 @@ impl CallManager {
                    spawn_sip_inbound(rtp_socket.clone(), channels.inbound_tx);
                    spawn_sip_outbound(rtp_socket, remote_addr, channels.outbound_rx);
                    if let Some(call) = self.calls.get(call_id) {
-                        call.add_leg_to_mixer(leg_id, sip_pt, channels.inbound_rx, channels.outbound_tx)
-                            .await;
+                        call.add_leg_to_mixer(
+                            leg_id,
+                            sip_pt,
+                            channels.inbound_rx,
+                            channels.outbound_tx,
+                        )
+                        .await;
                    }
                }

                // For device-originated calls: send 200 OK to device and wire device leg.
                if let Some(call) = self.calls.get(call_id) {
                    if let Some(device_invite) = call.device_invite.clone() {
-                        let device_leg_info: Option<(SocketAddr, u16, Arc<UdpSocket>, Option<SocketAddr>, String)> =
-                            call.legs.values().find(|l| l.kind == LegKind::SipDevice).and_then(|dev| {
+                        let device_leg_info: Option<(
+                            SocketAddr,
+                            u16,
+                            Arc<UdpSocket>,
+                            Option<SocketAddr>,
+                            String,
+                        )> = call
+                            .legs
+                            .values()
+                            .find(|l| l.kind == LegKind::SipDevice)
+                            .and_then(|dev| {
                                Some((
                                    dev.signaling_addr?,
                                    dev.rtp_port,
@@ -261,11 +292,21 @@ impl CallManager {
                                ))
                            });

-                        if let Some((dev_addr, dev_rtp_port, dev_rtp_socket, dev_remote, dev_leg_id)) = device_leg_info {
+                        if let Some((
+                            dev_addr,
+                            dev_rtp_port,
+                            dev_rtp_socket,
+                            dev_remote,
+                            dev_leg_id,
+                        )) = device_leg_info
+                        {
                            // Build SDP pointing device to our device_rtp port.
                            // Use LAN IP for the device (it's on the local network).
                            let call_ref = self.calls.get(call_id).unwrap();
-                            let prov_leg = call_ref.legs.values().find(|l| l.kind == LegKind::SipProvider);
+                            let prov_leg = call_ref
+                                .legs
+                                .values()
+                                .find(|l| l.kind == LegKind::SipProvider);
                            let lan_ip_str = prov_leg
                                .and_then(|l| l.sip_leg.as_ref())
                                .map(|sl| sl.config.lan_ip.clone())
@@ -277,13 +318,18 @@ impl CallManager {
                                ..Default::default()
                            });

-                            let ok = SipMessage::create_response(200, "OK", &device_invite, Some(ResponseOptions {
-                                to_tag: Some(generate_tag()),
-                                contact: Some(format!("<sip:{}:{}>", lan_ip_str, 5060)),
-                                body: Some(sdp),
-                                content_type: Some("application/sdp".to_string()),
-                                extra_headers: None,
-                            }));
+                            let ok = SipMessage::create_response(
+                                200,
+                                "OK",
+                                &device_invite,
+                                Some(ResponseOptions {
+                                    to_tag: Some(generate_tag()),
+                                    contact: Some(format!("<sip:{}:{}>", lan_ip_str, 5060)),
+                                    body: Some(sdp),
+                                    content_type: Some("application/sdp".to_string()),
+                                    extra_headers: None,
+                                }),
+                            );
                            let _ = socket.send_to(&ok.serialize(), dev_addr).await;

                            // Update device leg state.
@@ -310,27 +356,47 @@ impl CallManager {
                            if let Some(dev_remote_addr) = dev_remote {
                                let dev_channels = create_leg_channels();
                                spawn_sip_inbound(dev_rtp_socket.clone(), dev_channels.inbound_tx);
-                                spawn_sip_outbound(dev_rtp_socket, dev_remote_addr, dev_channels.outbound_rx);
+                                spawn_sip_outbound(
+                                    dev_rtp_socket,
+                                    dev_remote_addr,
+                                    dev_channels.outbound_rx,
+                                );
                                if let Some(call) = self.calls.get(call_id) {
-                                    call.add_leg_to_mixer(&dev_leg_id, dev_pt, dev_channels.inbound_rx, dev_channels.outbound_tx)
-                                        .await;
+                                    call.add_leg_to_mixer(
+                                        &dev_leg_id,
+                                        dev_pt,
+                                        dev_channels.inbound_rx,
+                                        dev_channels.outbound_tx,
+                                    )
+                                    .await;
                                }
                            }
                        }
                    }
                }

-                emit_event(&self.out_tx, "call_answered", serde_json::json!({
-                    "call_id": call_id,
-                    "provider_media_addr": remote.map(|a| a.ip().to_string()),
-                    "provider_media_port": remote.map(|a| a.port()),
-                    "sip_pt": sip_pt,
-                }));
-                emit_event(&self.out_tx, "leg_state_changed",
-                    serde_json::json!({ "call_id": call_id, "leg_id": leg_id, "state": "connected" }));
+                emit_event(
+                    &self.out_tx,
+                    "call_answered",
+                    serde_json::json!({
+                        "call_id": call_id,
+                        "provider_media_addr": remote.map(|a| a.ip().to_string()),
+                        "provider_media_port": remote.map(|a| a.port()),
+                        "sip_pt": sip_pt,
+                    }),
+                );
+                emit_event(
+                    &self.out_tx,
+                    "leg_state_changed",
+                    serde_json::json!({ "call_id": call_id, "leg_id": leg_id, "state": "connected" }),
+                );
            }
            SipLegAction::Terminated(reason) => {
-                let duration = self.calls.get(call_id).map(|c| c.duration_secs()).unwrap_or(0);
+                let duration = self
+                    .calls
+                    .get(call_id)
+                    .map(|c| c.duration_secs())
+                    .unwrap_or(0);

                // Notify device if this is a device-originated outbound call.
                if let Some(call) = self.calls.get(call_id) {
@@ -340,7 +406,8 @@ impl CallManager {
                            if let Some(dev_addr) = dev.signaling_addr {
                                // Map reason to SIP response code.
                                let code: u16 = if reason.starts_with("rejected_") {
-                                    reason.strip_prefix("rejected_")
+                                    reason
+                                        .strip_prefix("rejected_")
                                        .and_then(|s| s.parse().ok())
                                        .unwrap_or(503)
                                } else if reason == "bye" {
@@ -351,7 +418,12 @@ impl CallManager {
                                    503
                                };
                                if code > 0 && dev.state != LegState::Connected {
-                                    let resp = SipMessage::create_response(code, "Service Unavailable", device_invite, None);
+                                    let resp = SipMessage::create_response(
+                                        code,
+                                        "Service Unavailable",
+                                        device_invite,
+                                        None,
+                                    );
                                    let _ = socket.send_to(&resp.serialize(), dev_addr).await;
                                }
                            }
@@ -364,22 +436,38 @@ impl CallManager {
                        leg.state = LegState::Terminated;
                    }
                }
-                emit_event(&self.out_tx, "leg_state_changed",
-                    serde_json::json!({ "call_id": call_id, "leg_id": leg_id, "state": "terminated" }));
-                emit_event(&self.out_tx, "call_ended",
-                    serde_json::json!({ "call_id": call_id, "reason": reason, "duration": duration }));
+                emit_event(
+                    &self.out_tx,
+                    "leg_state_changed",
+                    serde_json::json!({ "call_id": call_id, "leg_id": leg_id, "state": "terminated" }),
+                );
+                emit_event(
+                    &self.out_tx,
+                    "call_ended",
+                    serde_json::json!({ "call_id": call_id, "reason": reason, "duration": duration }),
+                );
                self.terminate_call(call_id).await;
                return true;
            }
            SipLegAction::SendAndTerminate(buf, reason) => {
                let _ = socket.send_to(&buf, from_addr).await;
-                let duration = self.calls.get(call_id).map(|c| c.duration_secs()).unwrap_or(0);
-                emit_event(&self.out_tx, "call_ended",
-                    serde_json::json!({ "call_id": call_id, "reason": reason, "duration": duration }));
+                let duration = self
+                    .calls
+                    .get(call_id)
+                    .map(|c| c.duration_secs())
+                    .unwrap_or(0);
+                emit_event(
+                    &self.out_tx,
+                    "call_ended",
+                    serde_json::json!({ "call_id": call_id, "reason": reason, "duration": duration }),
+                );
                self.terminate_call(call_id).await;
                return true;
            }
-            SipLegAction::AuthRetry { ack_407, invite_with_auth } => {
+            SipLegAction::AuthRetry {
+                ack_407,
+                invite_with_auth,
+            } => {
                if let Some(ack) = ack_407 {
                    let _ = socket.send_to(&ack, target).await;
                }
@@ -413,11 +501,21 @@ impl CallManager {
        let this_kind = this_leg.map(|l| l.kind).unwrap_or(LegKind::SipProvider);

        // Find the counterpart leg.
-        let other_leg = call.legs.values().find(|l| l.id != this_leg_id && l.state != LegState::Terminated);
-        let (other_addr, other_rtp_port, other_leg_id, other_kind, other_public_ip) = match other_leg {
-            Some(l) => (l.signaling_addr, l.rtp_port, l.id.clone(), l.kind, l.public_ip.clone()),
-            None => return false,
-        };
+        let other_leg = call
+            .legs
+            .values()
+            .find(|l| l.id != this_leg_id && l.state != LegState::Terminated);
+        let (other_addr, other_rtp_port, other_leg_id, other_kind, other_public_ip) =
+            match other_leg {
+                Some(l) => (
+                    l.signaling_addr,
+                    l.rtp_port,
+                    l.id.clone(),
+                    l.kind,
+                    l.public_ip.clone(),
+                ),
+                None => return false,
+            };
        let forward_to = match other_addr {
            Some(a) => a,
            None => return false,
@@ -436,7 +534,9 @@ impl CallManager {
        };

        // Check if the other leg is a B2BUA leg (has SipLeg for proper dialog mgmt).
-        let other_has_sip_leg = call.legs.get(&other_leg_id)
+        let other_has_sip_leg = call
+            .legs
+            .get(&other_leg_id)
            .map(|l| l.sip_leg.is_some())
            .unwrap_or(false);

@@ -470,7 +570,8 @@ impl CallManager {
                    if let Some(other) = call.legs.get_mut(&other_leg_id) {
                        if let Some(sip_leg) = &mut other.sip_leg {
                            if let Some(hangup_buf) = sip_leg.build_hangup() {
-                                let _ = socket.send_to(&hangup_buf, sip_leg.config.sip_target).await;
+                                let _ =
+                                    socket.send_to(&hangup_buf, sip_leg.config.sip_target).await;
                            }
                        }
                    }
@@ -501,7 +602,8 @@ impl CallManager {
                    if let Some(other) = call.legs.get_mut(&other_leg_id) {
                        if let Some(sip_leg) = &mut other.sip_leg {
                            if let Some(hangup_buf) = sip_leg.build_hangup() {
-                                let _ = socket.send_to(&hangup_buf, sip_leg.config.sip_target).await;
+                                let _ =
+                                    socket.send_to(&hangup_buf, sip_leg.config.sip_target).await;
                            }
                        }
                    }
@@ -538,13 +640,17 @@ impl CallManager {
            if this_kind == LegKind::SipProvider {
                // From provider → forward to device: rewrite request URI.
                if let Some(ruri) = fwd.request_uri().map(|s| s.to_string()) {
-                    let new_ruri = rewrite_sip_uri(&ruri, &forward_to.ip().to_string(), forward_to.port());
+                    let new_ruri =
+                        rewrite_sip_uri(&ruri, &forward_to.ip().to_string(), forward_to.port());
                    fwd.set_request_uri(&new_ruri);
                }
            }
            if fwd.is_dialog_establishing() {
                // Record-Route must also be routable from the destination leg.
-                fwd.prepend_header("Record-Route", &format!("<sip:{advertise_ip}:{lan_port};lr>"));
+                fwd.prepend_header(
+                    "Record-Route",
+                    &format!("<sip:{advertise_ip}:{lan_port};lr>"),
+                );
            }
            let _ = socket.send_to(&fwd.serialize(), forward_to).await;
            return true;
@@ -569,13 +675,20 @@ impl CallManager {
                if code == 180 || code == 183 {
                    if call.state == CallState::SettingUp {
                        call.state = CallState::Ringing;
-                        emit_event(&self.out_tx, "call_ringing", serde_json::json!({ "call_id": call_id }));
+                        emit_event(
+                            &self.out_tx,
+                            "call_ringing",
+                            serde_json::json!({ "call_id": call_id }),
+                        );
                    }
                    if let Some(leg) = call.legs.get_mut(this_leg_id) {
                        leg.state = LegState::Ringing;
                    }
-                    emit_event(&self.out_tx, "leg_state_changed",
-                        serde_json::json!({ "call_id": call_id, "leg_id": this_leg_id, "state": "ringing" }));
+                    emit_event(
+                        &self.out_tx,
+                        "leg_state_changed",
+                        serde_json::json!({ "call_id": call_id, "leg_id": this_leg_id, "state": "ringing" }),
+                    );
                } else if code >= 200 && code < 300 {
                    let mut needs_wiring = false;
                    if let Some(leg) = call.legs.get_mut(this_leg_id) {
@@ -595,12 +708,19 @@ impl CallManager {
                        needs_wiring = true;
                    }

-                    emit_event(&self.out_tx, "leg_state_changed",
-                        serde_json::json!({ "call_id": call_id, "leg_id": this_leg_id, "state": "connected" }));
+                    emit_event(
+                        &self.out_tx,
+                        "leg_state_changed",
+                        serde_json::json!({ "call_id": call_id, "leg_id": this_leg_id, "state": "connected" }),
+                    );

                    if call.state != CallState::Connected {
                        call.state = CallState::Connected;
-                        emit_event(&self.out_tx, "call_answered", serde_json::json!({ "call_id": call_id }));
+                        emit_event(
+                            &self.out_tx,
+                            "call_answered",
+                            serde_json::json!({ "call_id": call_id }),
+                        );
                    }

                    // Forward the response before wiring (drop call borrow).
@@ -681,6 +801,7 @@ impl CallManager {
        rtp_pool: &mut RtpPortPool,
        socket: &UdpSocket,
        public_ip: Option<&str>,
+        tts_engine: Arc<Mutex<TtsEngine>>,
    ) -> Option<InboundCallCreated> {
        let call_id = self.next_call_id();
        let lan_ip = &config.proxy.lan_ip;
@@ -689,31 +810,59 @@ impl CallManager {

        // Extract caller/callee info.
        let from_header = invite.get_header("From").unwrap_or("");
-        let caller_number = SipMessage::extract_uri(from_header)
-            .unwrap_or("Unknown")
-            .to_string();
-        let called_number = invite
-            .request_uri()
-            .and_then(|uri| SipMessage::extract_uri(uri))
-            .unwrap_or("")
-            .to_string();
+        let caller_number = normalize_routing_identity(from_header);
+        let called_number = normalize_routing_identity(invite.request_uri().unwrap_or(""));

-        // Resolve via the configured inbound routing table. This honors
-        // user-defined routes from the UI (numberPattern, callerPattern,
-        // sourceProvider, targets, ringBrowsers). If no route matches, the
-        // fallback returns an empty `device_ids` and `ring_browsers: true`,
-        // which preserves pre-routing behavior via the `resolve_first_device`
-        // fallback below.
+        // Resolve via the configured inbound routing table. The matched route
+        // is the source of truth for which external numbers this provider is
+        // allowed to deliver to us.
        //
        // TODO: Multi-target inbound fork is not yet implemented.
        //   - `route.device_ids` beyond the first registered target are ignored.
        //   - `ring_browsers` is informational only — browsers see a toast but
        //     do not race the SIP device. First-to-answer-wins requires a
        //     multi-leg fork + per-leg CANCEL, which is not built yet.
-        //   - `voicemail_box`, `ivr_menu_id`, `no_answer_timeout` are not honored.
-        let route = config.resolve_inbound_route(provider_id, &called_number, &caller_number);
+        let route = match config.resolve_inbound_route(provider_id, &called_number, &caller_number)
+        {
+            Some(route) => route,
+            None => {
+                let resp = SipMessage::create_response(404, "Not Found", invite, None);
+                let _ = socket.send_to(&resp.serialize(), from_addr).await;
+                return None;
+            }
+        };
        let ring_browsers = route.ring_browsers;

+        // IVR routing: if the route targets an IVR menu, go there directly.
+        if let Some(ref ivr_menu_id) = route.ivr_menu_id {
+            if let Some(ivr) = &config.ivr {
+                if ivr.enabled {
+                    if let Some(menu) = ivr.menus.iter().find(|m| m.id == *ivr_menu_id) {
+                        let call_id = self
+                            .route_to_ivr(
+                                &call_id,
+                                invite,
+                                from_addr,
+                                &caller_number,
+                                provider_id,
+                                provider_config,
+                                config,
+                                rtp_pool,
+                                socket,
+                                public_ip,
+                                menu,
+                                &tts_engine,
+                            )
+                            .await?;
+                        return Some(InboundCallCreated {
+                            call_id,
+                            ring_browsers,
+                        });
+                    }
+                }
+            }
+        }
+
        // Pick the first registered device from the matched targets, or fall
        // back to any-registered-device if the route has no resolved targets.
        let device_addr = route
@@ -726,13 +875,28 @@ impl CallManager {
            Some(addr) => addr,
            None => {
                // No device registered → voicemail.
+                // Resolve greeting WAV on-demand (may trigger TTS generation).
+                let greeting_wav =
+                    resolve_greeting_wav(config, route.voicemail_box.as_deref(), &tts_engine).await;
                let call_id = self
                    .route_to_voicemail(
-                        &call_id, invite, from_addr, &caller_number,
-                        provider_id, provider_config, config, rtp_pool, socket, public_ip,
+                        &call_id,
+                        invite,
+                        from_addr,
+                        &caller_number,
+                        provider_id,
+                        provider_config,
+                        config,
+                        rtp_pool,
+                        socket,
+                        public_ip,
+                        greeting_wav,
                    )
                    .await?;
-                return Some(InboundCallCreated { call_id, ring_browsers });
+                return Some(InboundCallCreated {
+                    call_id,
+                    ring_browsers,
+                });
            }
        };

@@ -827,8 +991,10 @@ impl CallManager {
        // Register SIP Call-ID → both legs (provider leg handles provider messages).
        // For passthrough, both legs share the same SIP Call-ID.
        // We route based on source address in route_passthrough_message.
-        self.sip_index
-            .insert(sip_call_id.clone(), (call_id.clone(), provider_leg_id.clone()));
+        self.sip_index.insert(
+            sip_call_id.clone(),
+            (call_id.clone(), provider_leg_id.clone()),
+        );

        // Rewrite and forward INVITE to device.
        let mut fwd_invite = invite.clone();
@@ -861,7 +1027,10 @@ impl CallManager {
            }
        }

-        Some(InboundCallCreated { call_id, ring_browsers })
+        Some(InboundCallCreated {
+            call_id,
+            ring_browsers,
+        })
    }

    /// Initiate an outbound B2BUA call from the dashboard.
@@ -910,7 +1079,9 @@ impl CallManager {

        // Send INVITE.
        let to_uri = format!("sip:{number}@{}", provider_config.domain);
-        sip_leg.send_invite(registered_aor, &to_uri, &sip_call_id, socket).await;
+        sip_leg
+            .send_invite(registered_aor, &to_uri, &sip_call_id, socket)
+            .await;

        // Create call with mixer.
        let (mixer_cmd_tx, mixer_task) = spawn_mixer(call_id.clone(), self.out_tx.clone());
@@ -1081,7 +1252,9 @@ impl CallManager {

        // Build proper To URI and send INVITE.
        let to_uri = format!("sip:{}@{}", dialed_number, provider_config.domain);
-        sip_leg.send_invite(registered_aor, &to_uri, &provider_sip_call_id, socket).await;
+        sip_leg
+            .send_invite(registered_aor, &to_uri, &provider_sip_call_id, socket)
+            .await;

        call.legs.insert(
            provider_leg_id.clone(),
@@ -1157,7 +1330,9 @@ impl CallManager {

        let mut sip_leg = SipLeg::new(leg_id.clone(), leg_config);
        let to_uri = format!("sip:{number}@{}", provider_config.domain);
-        sip_leg.send_invite(registered_aor, &to_uri, &sip_call_id, socket).await;
+        sip_leg
+            .send_invite(registered_aor, &to_uri, &sip_call_id, socket)
+            .await;

        let codec_pt = provider_config.codecs.first().copied().unwrap_or(9);

@@ -1228,9 +1403,16 @@ impl CallManager {
        };

        let mut sip_leg = SipLeg::new(leg_id.clone(), leg_config);
-        let to_uri = format!("sip:{}@{}:{}", device_id, device_addr.ip(), device_addr.port());
+        let to_uri = format!(
+            "sip:{}@{}:{}",
+            device_id,
+            device_addr.ip(),
+            device_addr.port()
+        );
        let from_uri = format!("sip:sipproxy@{lan_ip}:{lan_port}");
-        sip_leg.send_invite(&from_uri, &to_uri, &sip_call_id, socket).await;
+        sip_leg
+            .send_invite(&from_uri, &to_uri, &sip_call_id, socket)
+            .await;

        let leg_info = LegInfo {
            id: leg_id.clone(),
@@ -1264,12 +1446,7 @@ impl CallManager {
    }

    /// Remove a leg from a call.
-    pub async fn remove_leg(
-        &mut self,
-        call_id: &str,
-        leg_id: &str,
-        socket: &UdpSocket,
-    ) -> bool {
+    pub async fn remove_leg(&mut self, call_id: &str, leg_id: &str, socket: &UdpSocket) -> bool {
        let call = match self.calls.get_mut(call_id) {
            Some(c) => c,
            None => return false,
@@ -1282,7 +1459,9 @@ impl CallManager {
        if let Some(leg) = call.legs.get_mut(leg_id) {
            if let Some(sip_leg) = &mut leg.sip_leg {
                if let Some(hangup_bytes) = sip_leg.build_hangup() {
-                    let _ = socket.send_to(&hangup_bytes, sip_leg.config.sip_target).await;
+                    let _ = socket
+                        .send_to(&hangup_bytes, sip_leg.config.sip_target)
+                        .await;
                }
            }
            leg.state = LegState::Terminated;
@@ -1328,9 +1507,7 @@ impl CallManager {
        target_call_id: &str,
    ) -> bool {
        // Validate both calls exist and the leg is in the source call.
-        if !self.calls.contains_key(source_call_id)
-            || !self.calls.contains_key(target_call_id)
-        {
+        if !self.calls.contains_key(source_call_id) || !self.calls.contains_key(target_call_id) {
            return false;
        }

@@ -1475,7 +1652,9 @@ impl CallManager {
            }
            if let Some(sip_leg) = &mut leg.sip_leg {
                if let Some(hangup_bytes) = sip_leg.build_hangup() {
-                    let _ = socket.send_to(&hangup_bytes, sip_leg.config.sip_target).await;
+                    let _ = socket
+                        .send_to(&hangup_bytes, sip_leg.config.sip_target)
+                        .await;
                }
            } else if let Some(addr) = leg.signaling_addr {
                // Passthrough leg — send a simple BYE.
@@ -1536,6 +1715,7 @@ impl CallManager {
        rtp_pool: &mut RtpPortPool,
        socket: &UdpSocket,
        public_ip: Option<&str>,
+        greeting_wav: Option<String>,
    ) -> Option<String> {
        let lan_ip = &config.proxy.lan_ip;
        let pub_ip = public_ip.unwrap_or(lan_ip.as_str());
@@ -1559,7 +1739,9 @@ impl CallManager {
        });

        let response = SipMessage::create_response(
-            200, "OK", invite,
+            200,
+            "OK",
+            invite,
            Some(sip_proto::message::ResponseOptions {
                to_tag: Some(sip_proto::helpers::generate_tag()),
                contact: Some(format!("<sip:{}:{}>", lan_ip, config.proxy.lan_port)),
@@ -1630,17 +1812,21 @@ impl CallManager {
            .as_millis();
        let recording_dir = "nogit/voicemail/default".to_string();
        let recording_path = format!("{recording_dir}/msg-{timestamp}.wav");
-        let greeting_wav = find_greeting_wav();
-
        let out_tx = self.out_tx.clone();
        let call_id_owned = call_id.to_string();
        let caller_owned = caller_number.to_string();
        let rtp_socket = rtp_alloc.socket;
        tokio::spawn(async move {
            crate::voicemail::run_voicemail_session(
-                rtp_socket, provider_media, codec_pt,
-                greeting_wav, recording_path, 120_000,
-                call_id_owned, caller_owned, out_tx,
+                rtp_socket,
+                provider_media,
+                codec_pt,
+                greeting_wav,
+                recording_path,
+                120_000,
+                call_id_owned,
+                caller_owned,
+                out_tx,
            )
            .await;
        });
@@ -1648,11 +1834,218 @@ impl CallManager {
        Some(call_id.to_string())
    }

+    // -----------------------------------------------------------------------
+    // IVR routing
+    // -----------------------------------------------------------------------
+
+    #[allow(clippy::too_many_arguments)]
+    async fn route_to_ivr(
+        &mut self,
+        call_id: &str,
+        invite: &SipMessage,
+        from_addr: SocketAddr,
+        caller_number: &str,
+        provider_id: &str,
+        provider_config: &ProviderConfig,
+        config: &AppConfig,
+        rtp_pool: &mut RtpPortPool,
+        socket: &UdpSocket,
+        public_ip: Option<&str>,
+        menu: &crate::config::IvrMenuConfig,
+        tts_engine: &Arc<Mutex<TtsEngine>>,
+    ) -> Option<String> {
+        let lan_ip = &config.proxy.lan_ip;
+
+        let rtp_alloc = match rtp_pool.allocate().await {
+            Some(a) => a,
+            None => {
+                let resp = SipMessage::create_response(503, "Service Unavailable", invite, None);
+                let _ = socket.send_to(&resp.serialize(), from_addr).await;
+                return None;
+            }
+        };
+
+        let codec_pt = provider_config.codecs.first().copied().unwrap_or(9);
+        let pub_ip = public_ip.unwrap_or(lan_ip.as_str());
+
+        let sdp = sip_proto::helpers::build_sdp(&sip_proto::helpers::SdpOptions {
+            ip: pub_ip,
+            port: rtp_alloc.port,
+            payload_types: &provider_config.codecs,
+            ..Default::default()
+        });
+
+        let response = SipMessage::create_response(
+            200,
+            "OK",
+            invite,
+            Some(sip_proto::message::ResponseOptions {
+                to_tag: Some(sip_proto::helpers::generate_tag()),
+                contact: Some(format!("<sip:{}:{}>", lan_ip, config.proxy.lan_port)),
+                body: Some(sdp),
+                content_type: Some("application/sdp".to_string()),
+                ..Default::default()
+            }),
+        );
+        let _ = socket.send_to(&response.serialize(), from_addr).await;
+
+        let provider_media = if invite.has_sdp_body() {
+            parse_sdp_endpoint(&invite.body)
+                .and_then(|ep| format!("{}:{}", ep.address, ep.port).parse().ok())
+        } else {
+            Some(from_addr)
+        };
+        let provider_media = provider_media.unwrap_or(from_addr);
+
+        // Create call with IVR state.
+        let (mixer_cmd_tx, mixer_task) = spawn_mixer(call_id.to_string(), self.out_tx.clone());
+        let mut call = Call::new(
+            call_id.to_string(),
+            CallDirection::Inbound,
+            provider_id.to_string(),
+            mixer_cmd_tx.clone(),
+            mixer_task,
+        );
+        call.state = CallState::Ivr;
+        call.caller_number = Some(caller_number.to_string());
+
+        let provider_leg_id = format!("{call_id}-prov");
+        call.legs.insert(
+            provider_leg_id.clone(),
+            LegInfo {
+                id: provider_leg_id.clone(),
+                kind: LegKind::SipProvider,
+                state: LegState::Connected,
+                codec_pt,
+                sip_leg: None,
+                sip_call_id: Some(invite.call_id().to_string()),
+                webrtc_session_id: None,
+                rtp_socket: Some(rtp_alloc.socket.clone()),
+                rtp_port: rtp_alloc.port,
+                public_ip: public_ip.map(|s| s.to_string()),
+                remote_media: Some(provider_media),
+                signaling_addr: Some(from_addr),
+                metadata: HashMap::new(),
+            },
+        );
+
+        self.sip_index.insert(
+            invite.call_id().to_string(),
+            (call_id.to_string(), provider_leg_id.clone()),
+        );
+        self.calls.insert(call_id.to_string(), call);
+
+        // Emit leg_added for the provider leg.
+        if let Some(call) = self.calls.get(call_id) {
+            for leg in call.legs.values() {
+                emit_leg_added_event(&self.out_tx, call_id, leg);
+            }
+        }
+
+        // Generate the IVR prompt as a live chunked TTS stream so playback can
+        // start after the first chunk instead of waiting for a full WAV render.
+        let voice = menu.prompt_voice.as_deref().unwrap_or("af_bella");
+        let live_prompt = {
+            let mut tts = tts_engine.lock().await;
+            match tts
+                .start_live_prompt(crate::tts::TtsPromptRequest {
+                    model_path: crate::tts::DEFAULT_MODEL_PATH.to_string(),
+                    voices_path: crate::tts::DEFAULT_VOICES_PATH.to_string(),
+                    voice_name: voice.to_string(),
+                    text: menu.prompt_text.clone(),
+                })
+                .await
+            {
+                Ok(prompt) => Some(prompt),
+                Err(e) => {
+                    eprintln!("[ivr] live TTS setup failed: {e}");
+                    None
+                }
+            }
+        };
+
+        // Load prompt and run interaction via the mixer.
+        let out_tx = self.out_tx.clone();
+        let call_id_owned = call_id.to_string();
+        let expected_digits: Vec<char> = menu
+            .entries
+            .iter()
+            .filter_map(|e| e.digit.chars().next())
+            .collect();
+        let timeout_ms = menu.timeout_sec.unwrap_or(5) * 1000;
+
+        tokio::spawn(async move {
+            if let Some(prompt) = live_prompt {
+                let (result_tx, result_rx) = tokio::sync::oneshot::channel();
+                let _ = mixer_cmd_tx
+                    .send(crate::mixer::MixerCommand::StartInteraction {
+                        leg_id: provider_leg_id.clone(),
+                        prompt_pcm_frames: prompt.initial_frames,
+                        prompt_stream_rx: Some(prompt.stream_rx),
+                        prompt_cancel_tx: Some(prompt.cancel_tx),
+                        expected_digits: expected_digits.clone(),
+                        timeout_ms,
+                        result_tx,
+                    })
+                    .await;
+
+                // Wait for digit or timeout.
+                let safety = tokio::time::Duration::from_millis(timeout_ms as u64 + 30000);
+                let result = match tokio::time::timeout(safety, result_rx).await {
+                    Ok(Ok(r)) => r,
+                    Ok(Err(_)) => crate::mixer::InteractionResult::Cancelled,
+                    Err(_) => crate::mixer::InteractionResult::Timeout,
+                };
+
+                match &result {
+                    crate::mixer::InteractionResult::Digit(d) => {
+                        eprintln!("[ivr] caller pressed '{d}' on call {call_id_owned}");
+                        emit_event(
+                            &out_tx,
+                            "ivr_digit",
+                            serde_json::json!({
+                                "call_id": call_id_owned,
+                                "digit": d.to_string(),
+                            }),
+                        );
+                    }
+                    crate::mixer::InteractionResult::Timeout => {
+                        eprintln!("[ivr] timeout on call {call_id_owned}");
+                        emit_event(
+                            &out_tx,
+                            "ivr_timeout",
+                            serde_json::json!({ "call_id": call_id_owned }),
+                        );
+                    }
+                    crate::mixer::InteractionResult::Cancelled => {
+                        eprintln!("[ivr] cancelled on call {call_id_owned}");
+                    }
+                }
+            } else {
+                eprintln!("[ivr] no prompt available for call {call_id_owned}, ending");
+                emit_event(
+                    &out_tx,
+                    "ivr_error",
+                    serde_json::json!({
+                        "call_id": call_id_owned,
+                        "error": "no prompt available",
+                    }),
+                );
+            }
+        });
+
+        Some(call_id.to_string())
+    }
+
    // -----------------------------------------------------------------------
    // Internal helpers
    // -----------------------------------------------------------------------

-    fn resolve_first_device(&self, config: &AppConfig, registrar: &Registrar) -> Option<SocketAddr> {
+    fn resolve_first_device(
+        &self,
+        config: &AppConfig,
+        registrar: &Registrar,
+    ) -> Option<SocketAddr> {
        for device in &config.devices {
            if let Some(addr) = registrar.get_device_contact(&device.id) {
                return Some(addr);
@@ -1662,13 +2055,55 @@ impl CallManager {
    }
 }

-fn find_greeting_wav() -> Option<String> {
-    let candidates = [
+/// Resolve the greeting WAV for a voicemail box.
+///
+/// Priority:
+/// 1. Pre-recorded WAV from voicebox config (`greetingWavPath`)
+/// 2. On-demand TTS generation from greeting text (cached via `cacheable: true`)
+/// 3. Legacy hardcoded paths (`.nogit/voicemail/default/greeting.wav`, etc.)
+/// 4. None — voicemail session plays beep only
+async fn resolve_greeting_wav(
+    config: &AppConfig,
+    voicebox_id: Option<&str>,
+    tts_engine: &Arc<Mutex<TtsEngine>>,
+) -> Option<String> {
+    // 1. Look up voicebox config.
+    let vb = voicebox_id.and_then(|id| config.voiceboxes.iter().find(|v| v.id == id && v.enabled));
+
+    if let Some(vb) = vb {
+        // 2. Pre-recorded WAV takes priority.
+        if let Some(ref wav) = vb.greeting_wav_path {
+            if Path::new(wav).exists() {
+                return Some(wav.clone());
+            }
+        }
+        // 3. TTS on-demand with caching.
+        let text = vb.greeting_text.as_deref().unwrap_or(
+            "The person you are trying to reach is not available. Please leave a message after the tone.",
+        );
+        let voice = vb.greeting_voice.as_deref().unwrap_or("af_bella");
+        let output = format!(".nogit/tts/voicemail-greeting-{}.wav", vb.id);
+
+        let params = serde_json::json!({
+            "model": ".nogit/tts/kokoro-v1.0.onnx",
+            "voices": ".nogit/tts/voices.bin",
+            "voice": voice,
+            "text": text,
+            "output": &output,
+            "cacheable": true,
+        });
+        let mut tts = tts_engine.lock().await;
+        if tts.generate(&params).await.is_ok() {
+            return Some(output);
+        }
+    }
+
+    // 4. Fallback: legacy hardcoded paths.
+    for path in &[
        ".nogit/voicemail/default/greeting.wav",
        ".nogit/voicemail/greeting.wav",
-    ];
-    for path in &candidates {
-        if std::path::Path::new(path).exists() {
+    ] {
+        if Path::new(path).exists() {
            return Some(path.to_string());
        }
    }
--- a/rust/crates/proxy-engine/src/config.rs
+++ b/rust/crates/proxy-engine/src/config.rs
@@ -4,6 +4,7 @@
 //! proxy engine via the `configure` command. These types mirror the TS interfaces.

 use serde::Deserialize;
+use sip_proto::message::SipMessage;
 use std::net::SocketAddr;

 /// Network endpoint.
@@ -159,6 +160,10 @@ pub struct AppConfig {
    pub providers: Vec<ProviderConfig>,
    pub devices: Vec<DeviceConfig>,
    pub routing: RoutingConfig,
+    #[serde(default)]
+    pub voiceboxes: Vec<VoiceboxConfig>,
+    #[serde(default)]
+    pub ivr: Option<IvrConfig>,
 }

 #[derive(Debug, Clone, Deserialize)]
@@ -166,12 +171,158 @@ pub struct RoutingConfig {
    pub routes: Vec<Route>,
 }

+// ---------------------------------------------------------------------------
+// Voicebox config
+// ---------------------------------------------------------------------------
+
+#[allow(dead_code)]
+#[derive(Debug, Clone, Deserialize)]
+pub struct VoiceboxConfig {
+    pub id: String,
+    #[serde(default)]
+    pub enabled: bool,
+    #[serde(rename = "greetingText")]
+    pub greeting_text: Option<String>,
+    #[serde(rename = "greetingVoice")]
+    pub greeting_voice: Option<String>,
+    #[serde(rename = "greetingWavPath")]
+    pub greeting_wav_path: Option<String>,
+    #[serde(rename = "maxRecordingSec")]
+    pub max_recording_sec: Option<u32>,
+}
+
+// ---------------------------------------------------------------------------
+// IVR config
+// ---------------------------------------------------------------------------
+
+#[allow(dead_code)]
+#[derive(Debug, Clone, Deserialize)]
+pub struct IvrConfig {
+    pub enabled: bool,
+    pub menus: Vec<IvrMenuConfig>,
+    #[serde(rename = "entryMenuId")]
+    pub entry_menu_id: String,
+}
+
+#[derive(Debug, Clone, Deserialize)]
+pub struct IvrMenuConfig {
+    pub id: String,
+    #[serde(rename = "promptText")]
+    pub prompt_text: String,
+    #[serde(rename = "promptVoice")]
+    pub prompt_voice: Option<String>,
+    pub entries: Vec<IvrMenuEntry>,
+    #[serde(rename = "timeoutSec")]
+    pub timeout_sec: Option<u32>,
+}
+
+#[allow(dead_code)]
+#[derive(Debug, Clone, Deserialize)]
+pub struct IvrMenuEntry {
+    pub digit: String,
+    pub action: String,
+    pub target: Option<String>,
+}
+
 // ---------------------------------------------------------------------------
 // Pattern matching (ported from ts/config.ts)
 // ---------------------------------------------------------------------------

+/// Extract the URI user part and normalize phone-like identities for routing.
+///
+/// This keeps inbound route matching stable across provider-specific URI shapes,
+/// e.g. `sip:+49 421 219694@trunk.example` and `sip:0049421219694@trunk.example`
+/// both normalize to `+49421219694`.
+pub fn normalize_routing_identity(value: &str) -> String {
+    let extracted = SipMessage::extract_uri_user(value).unwrap_or(value).trim();
+    if extracted.is_empty() {
+        return String::new();
+    }
+
+    let mut digits = String::new();
+    let mut saw_plus = false;
+
+    for (idx, ch) in extracted.chars().enumerate() {
+        if ch.is_ascii_digit() {
+            digits.push(ch);
+            continue;
+        }
+
+        if ch == '+' && idx == 0 {
+            saw_plus = true;
+            continue;
+        }
+
+        if matches!(ch, ' ' | '\t' | '-' | '.' | '/' | '(' | ')') {
+            continue;
+        }
+
+        return extracted.to_string();
+    }
+
+    if digits.is_empty() {
+        return extracted.to_string();
+    }
+    if saw_plus {
+        return format!("+{digits}");
+    }
+    if digits.starts_with("00") && digits.len() > 2 {
+        return format!("+{}", &digits[2..]);
+    }
+
+    digits
+}
+
+fn parse_numeric_range_value(value: &str) -> Option<(bool, &str)> {
+    let trimmed = value.trim();
+    if trimmed.is_empty() {
+        return None;
+    }
+
+    let (has_plus, digits) = if let Some(rest) = trimmed.strip_prefix('+') {
+        (true, rest)
+    } else {
+        (false, trimmed)
+    };
+
+    if digits.is_empty() || !digits.chars().all(|c| c.is_ascii_digit()) {
+        return None;
+    }
+
+    Some((has_plus, digits))
+}
+
+fn matches_numeric_range_pattern(pattern: &str, value: &str) -> bool {
+    let Some((start, end)) = pattern.split_once("..") else {
+        return false;
+    };
+
+    let Some((start_plus, start_digits)) = parse_numeric_range_value(start) else {
+        return false;
+    };
+    let Some((end_plus, end_digits)) = parse_numeric_range_value(end) else {
+        return false;
+    };
+    let Some((value_plus, value_digits)) = parse_numeric_range_value(value) else {
+        return false;
+    };
+
+    if start_plus != end_plus || value_plus != start_plus {
+        return false;
+    }
+    if start_digits.len() != end_digits.len() || value_digits.len() != start_digits.len() {
+        return false;
+    }
+    if start_digits > end_digits {
+        return false;
+    }
+
+    value_digits >= start_digits && value_digits <= end_digits
+}
+
 /// Test a value against a pattern string.
 /// - None/empty: matches everything (wildcard)
+/// - `start..end`: numeric range match
 /// - Trailing '*': prefix match
 /// - Starts with '/': regex match
 /// - Otherwise: exact match
@@ -187,6 +338,10 @@ pub fn matches_pattern(pattern: Option<&str>, value: &str) -> bool {
        return value.starts_with(&pattern[..pattern.len() - 1]);
    }

+    if matches_numeric_range_pattern(pattern, value) {
+        return true;
+    }
+
    // Regex match: "/^\\+49/" or "/pattern/i"
    if pattern.starts_with('/') {
        if let Some(last_slash) = pattern[1..].rfind('/') {
@@ -306,7 +461,7 @@ impl AppConfig {
        provider_id: &str,
        called_number: &str,
        caller_number: &str,
-    ) -> InboundRouteResult {
+    ) -> Option<InboundRouteResult> {
        let mut routes: Vec<&Route> = self
            .routing
            .routes
@@ -330,22 +485,170 @@ impl AppConfig {
                continue;
            }

-            return InboundRouteResult {
+            return Some(InboundRouteResult {
                device_ids: route.action.targets.clone().unwrap_or_default(),
                ring_browsers: route.action.ring_browsers.unwrap_or(false),
                voicemail_box: route.action.voicemail_box.clone(),
                ivr_menu_id: route.action.ivr_menu_id.clone(),
                no_answer_timeout: route.action.no_answer_timeout,
-            };
+            });
        }

-        // Fallback: ring all devices + browsers.
-        InboundRouteResult {
-            device_ids: vec![],
-            ring_browsers: true,
-            voicemail_box: None,
-            ivr_menu_id: None,
-            no_answer_timeout: None,
-        }
+        None
+    }
+}
+
+#[cfg(test)]
+mod tests {
+    use super::*;
+
+    fn test_app_config(routes: Vec<Route>) -> AppConfig {
+        AppConfig {
+            proxy: ProxyConfig {
+                lan_ip: "127.0.0.1".to_string(),
+                lan_port: 5070,
+                public_ip_seed: None,
+                rtp_port_range: RtpPortRange {
+                    min: 20_000,
+                    max: 20_100,
+                },
+            },
+            providers: vec![ProviderConfig {
+                id: "provider-a".to_string(),
+                display_name: "Provider A".to_string(),
+                domain: "example.com".to_string(),
+                outbound_proxy: Endpoint {
+                    address: "example.com".to_string(),
+                    port: 5060,
+                },
+                username: "user".to_string(),
+                password: "pass".to_string(),
+                register_interval_sec: 300,
+                codecs: vec![9],
+                quirks: Quirks {
+                    early_media_silence: false,
+                    silence_payload_type: None,
+                    silence_max_packets: None,
+                },
+            }],
+            devices: vec![DeviceConfig {
+                id: "desk".to_string(),
+                display_name: "Desk".to_string(),
+                expected_address: "127.0.0.1".to_string(),
+                extension: "100".to_string(),
+            }],
+            routing: RoutingConfig { routes },
+            voiceboxes: vec![],
+            ivr: None,
+        }
+    }
+
+    #[test]
+    fn normalize_routing_identity_extracts_uri_user_and_phone_number() {
+        assert_eq!(
+            normalize_routing_identity("sip:0049 421 219694@voip.easybell.de"),
+            "+49421219694"
+        );
+        assert_eq!(
+            normalize_routing_identity("<tel:+49 (421) 219694>"),
+            "+49421219694"
+        );
+        assert_eq!(normalize_routing_identity("sip:100@pbx.local"), "100");
+        assert_eq!(normalize_routing_identity("sip:alice@pbx.local"), "alice");
+    }
+
+    #[test]
+    fn resolve_inbound_route_requires_explicit_match() {
+        let cfg = test_app_config(vec![]);
+        assert!(cfg
+            .resolve_inbound_route("provider-a", "+49421219694", "+491701234567")
+            .is_none());
+    }
+
+    #[test]
+    fn resolve_inbound_route_matches_per_number_on_shared_provider() {
+        let cfg = test_app_config(vec![
+            Route {
+                id: "main".to_string(),
+                name: "Main DID".to_string(),
+                priority: 200,
+                enabled: true,
+                match_criteria: RouteMatch {
+                    direction: "inbound".to_string(),
+                    number_pattern: Some("+49421219694".to_string()),
+                    caller_pattern: None,
+                    source_provider: Some("provider-a".to_string()),
+                    source_device: None,
+                },
+                action: RouteAction {
+                    targets: Some(vec!["desk".to_string()]),
+                    ring_browsers: Some(true),
+                    voicemail_box: None,
+                    ivr_menu_id: None,
+                    no_answer_timeout: None,
+                    provider: None,
+                    failover_providers: None,
+                    strip_prefix: None,
+                    prepend_prefix: None,
+                },
+            },
+            Route {
+                id: "support".to_string(),
+                name: "Support DID".to_string(),
+                priority: 100,
+                enabled: true,
+                match_criteria: RouteMatch {
+                    direction: "inbound".to_string(),
+                    number_pattern: Some("+49421219695".to_string()),
+                    caller_pattern: None,
+                    source_provider: Some("provider-a".to_string()),
+                    source_device: None,
+                },
+                action: RouteAction {
+                    targets: None,
+                    ring_browsers: Some(false),
+                    voicemail_box: Some("support-box".to_string()),
+                    ivr_menu_id: None,
+                    no_answer_timeout: Some(20),
+                    provider: None,
+                    failover_providers: None,
+                    strip_prefix: None,
+                    prepend_prefix: None,
+                },
+            },
+        ]);
+
+        let main = cfg
+            .resolve_inbound_route("provider-a", "+49421219694", "+491701234567")
+            .expect("main DID should match");
+        assert_eq!(main.device_ids, vec!["desk".to_string()]);
+        assert!(main.ring_browsers);
+
+        let support = cfg
+            .resolve_inbound_route("provider-a", "+49421219695", "+491701234567")
+            .expect("support DID should match");
+        assert_eq!(support.voicemail_box.as_deref(), Some("support-box"));
+        assert_eq!(support.no_answer_timeout, Some(20));
+        assert!(!support.ring_browsers);
+    }
+
+    #[test]
+    fn matches_pattern_supports_numeric_ranges() {
+        assert!(matches_pattern(
+            Some("042116767546..042116767548"),
+            "042116767547"
+        ));
+        assert!(!matches_pattern(
+            Some("042116767546..042116767548"),
+            "042116767549"
+        ));
+        assert!(matches_pattern(
+            Some("+4942116767546..+4942116767548"),
+            "+4942116767547"
+        ));
+        assert!(!matches_pattern(
+            Some("+4942116767546..+4942116767548"),
+            "042116767547"
+        ));
    }
 }
--- a/rust/crates/proxy-engine/src/ipc.rs
+++ b/rust/crates/proxy-engine/src/ipc.rs
@@ -19,7 +19,13 @@ pub struct Command {
 }

 /// Send a response to a command.
-pub fn respond(tx: &OutTx, id: &str, success: bool, result: Option<serde_json::Value>, error: Option<&str>) {
+pub fn respond(
+    tx: &OutTx,
+    id: &str,
+    success: bool,
+    result: Option<serde_json::Value>,
+    error: Option<&str>,
+) {
    let mut resp = serde_json::json!({ "id": id, "success": success });
    if let Some(r) = result {
        resp["result"] = r;
--- a/rust/crates/proxy-engine/src/leg_io.rs
+++ b/rust/crates/proxy-engine/src/leg_io.rs
@@ -63,7 +63,8 @@ pub fn spawn_sip_inbound(
                        if offset + 4 > n {
                            continue; // Malformed: extension header truncated.
                        }
-                        let ext_len = u16::from_be_bytes([buf[offset + 2], buf[offset + 3]]) as usize;
+                        let ext_len =
+                            u16::from_be_bytes([buf[offset + 2], buf[offset + 3]]) as usize;
                        offset += 4 + ext_len * 4;
                    }
                    if offset >= n {
@@ -74,7 +75,17 @@ pub fn spawn_sip_inbound(
                    if payload.is_empty() {
                        continue;
                    }
-                    if inbound_tx.send(RtpPacket { payload, payload_type: pt, marker, seq, timestamp }).await.is_err() {
+                    if inbound_tx
+                        .send(RtpPacket {
+                            payload,
+                            payload_type: pt,
+                            marker,
+                            seq,
+                            timestamp,
+                        })
+                        .await
+                        .is_err()
+                    {
                        break; // Channel closed — leg removed.
                    }
                }
--- a/rust/crates/proxy-engine/src/main.rs
+++ b/rust/crates/proxy-engine/src/main.rs
--- a/rust/crates/proxy-engine/src/mixer.rs
+++ b/rust/crates/proxy-engine/src/mixer.rs
@@ -7,7 +7,8 @@
 //! All encoding/decoding happens at leg boundaries. Per-leg inbound denoising at 48kHz.
 //!
 //! The mixer runs a 20ms tick loop:
-//! 1. Drain inbound channels, decode to f32, resample to 48kHz, denoise per-leg
+//! 1. Drain inbound channels, reorder RTP, decode variable-duration packets to 48kHz,
+//!    and queue them in per-leg PCM buffers
 //! 2. Compute total mix (sum of all **participant** legs' f32 PCM as f64)
 //! 3. For each participant leg: mix-minus = total - own, resample to leg codec rate, encode, send
 //! 4. For each isolated leg: play prompt frame or silence, check DTMF
@@ -16,11 +17,12 @@

 use crate::ipc::{emit_event, OutTx};
 use crate::jitter_buffer::{JitterBuffer, JitterResult};
-use crate::rtp::{build_rtp_header, rtp_clock_increment};
+use crate::rtp::{build_rtp_header, rtp_clock_increment, rtp_clock_rate};
+use crate::tts::TtsStreamMessage;
 use codec_lib::{codec_sample_rate, new_denoiser, TranscodeState};
 use nnnoiseless::DenoiseState;
 use std::collections::{HashMap, VecDeque};
-use tokio::sync::{mpsc, oneshot};
+use tokio::sync::{mpsc, oneshot, watch};
 use tokio::task::JoinHandle;
 use tokio::time::{self, Duration, MissedTickBehavior};

@@ -29,6 +31,12 @@ use tokio::time::{self, Duration, MissedTickBehavior};
 const MIX_RATE: u32 = 48000;
 /// Samples per 20ms frame at the mixing rate.
 const MIX_FRAME_SIZE: usize = 960; // 48000 * 0.020
+/// Safety cap for how much timestamp-derived gap fill we synthesize at once.
+const MAX_GAP_FILL_SAMPLES: usize = MIX_FRAME_SIZE * 6; // 120ms
+/// Bound how many decode / concealment steps a leg can consume in one tick.
+const MAX_PACKET_STEPS_PER_TICK: usize = 24;
+/// Report the first output drop immediately, then every N drops.
+const DROP_REPORT_INTERVAL: u64 = 50;

 /// A raw RTP payload received from a leg (no RTP header).
 pub struct RtpPacket {
@@ -39,10 +47,6 @@ pub struct RtpPacket {
    /// RTP sequence number for reordering.
    pub seq: u16,
    /// RTP timestamp from the original packet header.
-    ///
-    /// Set on inbound RTP but not yet consumed downstream — reserved for
-    /// future jitter/sync work in the mixer.
-    #[allow(dead_code)]
    pub timestamp: u32,
 }

@@ -61,6 +65,12 @@ enum LegRole {
 struct IsolationState {
    /// PCM frames at MIX_RATE (960 samples each, 48kHz f32) queued for playback.
    prompt_frames: VecDeque<Vec<f32>>,
+    /// Live TTS frames arrive here while playback is already in progress.
+    prompt_stream_rx: Option<mpsc::Receiver<TtsStreamMessage>>,
+    /// Cancels the background TTS producer when the interaction ends early.
+    prompt_cancel_tx: Option<watch::Sender<bool>>,
+    /// Whether the live prompt stream has ended.
+    prompt_stream_finished: bool,
    /// Digits that complete the interaction (e.g., ['1', '2']).
    expected_digits: Vec<char>,
    /// Ticks remaining before timeout (decremented each tick after prompt ends).
@@ -109,6 +119,7 @@ struct ToolLegSlot {
    #[allow(dead_code)]
    tool_type: ToolType,
    audio_tx: mpsc::Sender<ToolAudioBatch>,
+    dropped_batches: u64,
 }

 // ---------------------------------------------------------------------------
@@ -136,6 +147,10 @@ pub enum MixerCommand {
        leg_id: String,
        /// PCM frames at MIX_RATE (48kHz f32), each 960 samples.
        prompt_pcm_frames: Vec<Vec<f32>>,
+        /// Optional live prompt stream. Frames are appended as they are synthesized.
+        prompt_stream_rx: Option<mpsc::Receiver<TtsStreamMessage>>,
+        /// Optional cancellation handle for the live prompt stream.
+        prompt_cancel_tx: Option<watch::Sender<bool>>,
        expected_digits: Vec<char>,
        timeout_ms: u32,
        result_tx: oneshot::Sender<InteractionResult>,
@@ -163,8 +178,15 @@ struct MixerLegSlot {
    denoiser: Box<DenoiseState<'static>>,
    inbound_rx: mpsc::Receiver<RtpPacket>,
    outbound_tx: mpsc::Sender<Vec<u8>>,
+    /// Decoded PCM waiting for playout. Variable-duration RTP packets are
+    /// decoded into this FIFO; the mixer consumes exactly one 20ms frame per tick.
+    pcm_buffer: VecDeque<f32>,
    /// Last decoded+denoised PCM frame at MIX_RATE (960 samples, 48kHz f32).
    last_pcm_frame: Vec<f32>,
+    /// Next RTP timestamp expected from the inbound stream.
+    expected_rtp_timestamp: Option<u32>,
+    /// Best-effort estimate of packet duration in RTP clock units.
+    estimated_packet_ts: u32,
    /// Number of consecutive ticks with no inbound packet.
    silent_ticks: u32,
    /// Per-leg jitter buffer for packet reordering and timing.
@@ -173,15 +195,302 @@ struct MixerLegSlot {
    rtp_seq: u16,
    rtp_ts: u32,
    rtp_ssrc: u32,
+    /// Dropped outbound frames for this leg (queue full / closed).
+    outbound_drops: u64,
    /// Current role of this leg in the mixer.
    role: LegRole,
 }

+fn mix_samples_to_rtp_ts(codec_pt: u8, mix_samples: usize) -> u32 {
+    let clock_rate = rtp_clock_rate(codec_pt).max(1) as u64;
+    (((mix_samples as u64 * clock_rate) + (MIX_RATE as u64 / 2)) / MIX_RATE as u64) as u32
+}
+
+fn rtp_ts_to_mix_samples(codec_pt: u8, rtp_ts: u32) -> usize {
+    let clock_rate = rtp_clock_rate(codec_pt).max(1) as u64;
+    (((rtp_ts as u64 * MIX_RATE as u64) + (clock_rate / 2)) / clock_rate) as usize
+}
+
+fn is_forward_rtp_delta(delta: u32) -> bool {
+    delta > 0 && delta < 0x8000_0000
+}
+
+fn should_emit_drop_event(total_drops: u64) -> bool {
+    total_drops == 1 || total_drops % DROP_REPORT_INTERVAL == 0
+}
+
+fn emit_output_drop_event(
+    out_tx: &OutTx,
+    call_id: &str,
+    leg_id: Option<&str>,
+    tool_leg_id: Option<&str>,
+    stream: &str,
+    reason: &str,
+    total_drops: u64,
+) {
+    if !should_emit_drop_event(total_drops) {
+        return;
+    }
+
+    emit_event(
+        out_tx,
+        "mixer_output_drop",
+        serde_json::json!({
+            "call_id": call_id,
+            "leg_id": leg_id,
+            "tool_leg_id": tool_leg_id,
+            "stream": stream,
+            "reason": reason,
+            "total_drops": total_drops,
+        }),
+    );
+}
+
+fn fade_concealment_from_last_frame(slot: &mut MixerLegSlot, samples: usize, decay: f32) {
+    let mut template = if slot.last_pcm_frame.is_empty() {
+        vec![0.0f32; MIX_FRAME_SIZE]
+    } else {
+        slot.last_pcm_frame.clone()
+    };
+
+    let mut remaining = samples;
+    while remaining > 0 {
+        for sample in &mut template {
+            *sample *= decay;
+        }
+        let take = remaining.min(template.len());
+        slot.pcm_buffer.extend(template.iter().take(take).copied());
+        remaining -= take;
+    }
+}
+
+fn append_packet_loss_concealment(slot: &mut MixerLegSlot, samples: usize) {
+    let mut remaining = samples.max(1);
+    while remaining > 0 {
+        let chunk = remaining.min(MIX_FRAME_SIZE);
+        if slot.codec_pt == codec_lib::PT_OPUS {
+            match slot.transcoder.opus_plc(chunk) {
+                Ok(mut pcm) => {
+                    pcm.resize(chunk, 0.0);
+                    slot.pcm_buffer.extend(pcm);
+                }
+                Err(_) => fade_concealment_from_last_frame(slot, chunk, 0.8),
+            }
+        } else {
+            fade_concealment_from_last_frame(slot, chunk, 0.85);
+        }
+        remaining -= chunk;
+    }
+}
+
+fn decode_packet_to_mix_pcm(slot: &mut MixerLegSlot, pkt: &RtpPacket) -> Option<Vec<f32>> {
+    let (pcm, rate) = slot
+        .transcoder
+        .decode_to_f32(&pkt.payload, pkt.payload_type)
+        .ok()?;
+
+    let pcm_48k = if rate == MIX_RATE {
+        pcm
+    } else {
+        slot.transcoder
+            .resample_f32(&pcm, rate, MIX_RATE)
+            .unwrap_or_else(|_| vec![0.0f32; MIX_FRAME_SIZE])
+    };
+
+    let processed = if slot.codec_pt != codec_lib::PT_OPUS {
+        TranscodeState::denoise_f32(&mut slot.denoiser, &pcm_48k)
+    } else {
+        pcm_48k
+    };
+
+    Some(processed)
+}
+
+fn queue_inbound_packet(slot: &mut MixerLegSlot, pkt: RtpPacket) {
+    if let Some(pcm_48k) = decode_packet_to_mix_pcm(slot, &pkt) {
+        if pcm_48k.is_empty() {
+            return;
+        }
+
+        if let Some(expected_ts) = slot.expected_rtp_timestamp {
+            let gap_ts = pkt.timestamp.wrapping_sub(expected_ts);
+            if is_forward_rtp_delta(gap_ts) {
+                let gap_samples = rtp_ts_to_mix_samples(slot.codec_pt, gap_ts);
+                if gap_samples <= MAX_GAP_FILL_SAMPLES {
+                    append_packet_loss_concealment(slot, gap_samples);
+                } else {
+                    slot.pcm_buffer.clear();
+                }
+            }
+        }
+
+        let packet_ts = mix_samples_to_rtp_ts(slot.codec_pt, pcm_48k.len());
+        if packet_ts > 0 {
+            slot.estimated_packet_ts = packet_ts;
+            slot.expected_rtp_timestamp = Some(pkt.timestamp.wrapping_add(packet_ts));
+        }
+        slot.pcm_buffer.extend(pcm_48k);
+    }
+}
+
+fn fill_leg_playout_buffer(slot: &mut MixerLegSlot) {
+    let mut steps = 0usize;
+    while slot.pcm_buffer.len() < MIX_FRAME_SIZE && steps < MAX_PACKET_STEPS_PER_TICK {
+        steps += 1;
+        match slot.jitter.consume() {
+            JitterResult::Packet(pkt) => queue_inbound_packet(slot, pkt),
+            JitterResult::Missing => {
+                let conceal_ts = slot
+                    .estimated_packet_ts
+                    .max(rtp_clock_increment(slot.codec_pt));
+                let conceal_samples =
+                    rtp_ts_to_mix_samples(slot.codec_pt, conceal_ts).clamp(1, MAX_GAP_FILL_SAMPLES);
+                append_packet_loss_concealment(slot, conceal_samples);
+                if let Some(expected_ts) = slot.expected_rtp_timestamp {
+                    slot.expected_rtp_timestamp = Some(expected_ts.wrapping_add(conceal_ts));
+                }
+            }
+            JitterResult::Filling => break,
+        }
+    }
+}
+
+fn take_mix_frame(slot: &mut MixerLegSlot) -> Vec<f32> {
+    let mut frame = Vec::with_capacity(MIX_FRAME_SIZE);
+    while frame.len() < MIX_FRAME_SIZE {
+        if let Some(sample) = slot.pcm_buffer.pop_front() {
+            frame.push(sample);
+        } else {
+            frame.push(0.0);
+        }
+    }
+    frame
+}
+
+fn soft_limit_sample(sample: f32) -> f32 {
+    const KNEE: f32 = 0.85;
+
+    let abs = sample.abs();
+    if abs <= KNEE {
+        sample
+    } else {
+        let excess = abs - KNEE;
+        let compressed = KNEE + (excess / (1.0 + (excess / (1.0 - KNEE))));
+        sample.signum() * compressed.min(1.0)
+    }
+}
+
+fn try_send_leg_output(
+    out_tx: &OutTx,
+    call_id: &str,
+    leg_id: &str,
+    slot: &mut MixerLegSlot,
+    rtp: Vec<u8>,
+    stream: &str,
+) {
+    let reason = match slot.outbound_tx.try_send(rtp) {
+        Ok(()) => return,
+        Err(mpsc::error::TrySendError::Full(_)) => "full",
+        Err(mpsc::error::TrySendError::Closed(_)) => "closed",
+    };
+
+    slot.outbound_drops += 1;
+    emit_output_drop_event(
+        out_tx,
+        call_id,
+        Some(leg_id),
+        None,
+        stream,
+        reason,
+        slot.outbound_drops,
+    );
+}
+
+fn try_send_tool_output(
+    out_tx: &OutTx,
+    call_id: &str,
+    tool_leg_id: &str,
+    tool: &mut ToolLegSlot,
+    batch: ToolAudioBatch,
+) {
+    let reason = match tool.audio_tx.try_send(batch) {
+        Ok(()) => return,
+        Err(mpsc::error::TrySendError::Full(_)) => "full",
+        Err(mpsc::error::TrySendError::Closed(_)) => "closed",
+    };
+
+    tool.dropped_batches += 1;
+    emit_output_drop_event(
+        out_tx,
+        call_id,
+        None,
+        Some(tool_leg_id),
+        "tool-batch",
+        reason,
+        tool.dropped_batches,
+    );
+}
+
+fn cancel_prompt_producer(state: &mut IsolationState) {
+    if let Some(cancel_tx) = state.prompt_cancel_tx.take() {
+        let _ = cancel_tx.send(true);
+    }
+}
+
+fn cancel_isolated_interaction(state: &mut IsolationState) {
+    cancel_prompt_producer(state);
+    if let Some(tx) = state.result_tx.take() {
+        let _ = tx.send(InteractionResult::Cancelled);
+    }
+}
+
+fn drain_prompt_stream(
+    out_tx: &OutTx,
+    call_id: &str,
+    leg_id: &str,
+    state: &mut IsolationState,
+) {
+    loop {
+        let Some(mut stream_rx) = state.prompt_stream_rx.take() else {
+            return;
+        };
+
+        match stream_rx.try_recv() {
+            Ok(TtsStreamMessage::Frames(frames)) => {
+                state.prompt_frames.extend(frames);
+                state.prompt_stream_rx = Some(stream_rx);
+            }
+            Ok(TtsStreamMessage::Finished) => {
+                state.prompt_stream_finished = true;
+                return;
+            }
+            Ok(TtsStreamMessage::Failed(error)) => {
+                emit_event(
+                    out_tx,
+                    "mixer_error",
+                    serde_json::json!({
+                        "call_id": call_id,
+                        "leg_id": leg_id,
+                        "error": format!("tts stream failed: {error}"),
+                    }),
+                );
+                state.prompt_stream_finished = true;
+                return;
+            }
+            Err(mpsc::error::TryRecvError::Empty) => {
+                state.prompt_stream_rx = Some(stream_rx);
+                return;
+            }
+            Err(mpsc::error::TryRecvError::Disconnected) => {
+                state.prompt_stream_finished = true;
+                return;
+            }
+        }
+    }
+}
+
 /// Spawn the mixer task for a call. Returns the command sender and task handle.
-pub fn spawn_mixer(
-    call_id: String,
-    out_tx: OutTx,
-) -> (mpsc::Sender<MixerCommand>, JoinHandle<()>) {
+pub fn spawn_mixer(call_id: String, out_tx: OutTx) -> (mpsc::Sender<MixerCommand>, JoinHandle<()>) {
    let (cmd_tx, cmd_rx) = mpsc::channel::<MixerCommand>(32);

    let handle = tokio::spawn(async move {
@@ -192,11 +501,7 @@ pub fn spawn_mixer(
 }

 /// The 20ms mixing loop.
-async fn mixer_loop(
-    call_id: String,
-    mut cmd_rx: mpsc::Receiver<MixerCommand>,
-    out_tx: OutTx,
-) {
+async fn mixer_loop(call_id: String, mut cmd_rx: mpsc::Receiver<MixerCommand>, out_tx: OutTx) {
    let mut legs: HashMap<String, MixerLegSlot> = HashMap::new();
    let mut tool_legs: HashMap<String, ToolLegSlot> = HashMap::new();
    let mut interval = time::interval(Duration::from_millis(20));
@@ -237,11 +542,15 @@ async fn mixer_loop(
                            denoiser: new_denoiser(),
                            inbound_rx,
                            outbound_tx,
+                            pcm_buffer: VecDeque::new(),
                            last_pcm_frame: vec![0.0f32; MIX_FRAME_SIZE],
+                            expected_rtp_timestamp: None,
+                            estimated_packet_ts: rtp_clock_increment(codec_pt),
                            silent_ticks: 0,
                            rtp_seq: 0,
                            rtp_ts: 0,
                            rtp_ssrc: rand::random(),
+                            outbound_drops: 0,
                            role: LegRole::Participant,
                            jitter: JitterBuffer::new(),
                        },
@@ -251,9 +560,7 @@ async fn mixer_loop(
                    // If the leg is isolated, send Cancelled before dropping.
                    if let Some(slot) = legs.get_mut(&leg_id) {
                        if let LegRole::Isolated(ref mut state) = slot.role {
-                            if let Some(tx) = state.result_tx.take() {
-                                let _ = tx.send(InteractionResult::Cancelled);
-                            }
+                            cancel_isolated_interaction(state);
                        }
                    }
                    legs.remove(&leg_id);
@@ -263,9 +570,7 @@ async fn mixer_loop(
                    // Cancel all outstanding interactions before shutting down.
                    for slot in legs.values_mut() {
                        if let LegRole::Isolated(ref mut state) = slot.role {
-                            if let Some(tx) = state.result_tx.take() {
-                                let _ = tx.send(InteractionResult::Cancelled);
-                            }
+                            cancel_isolated_interaction(state);
                        }
                    }
                    return;
@@ -273,6 +578,8 @@ async fn mixer_loop(
                Ok(MixerCommand::StartInteraction {
                    leg_id,
                    prompt_pcm_frames,
+                    prompt_stream_rx,
+                    prompt_cancel_tx,
                    expected_digits,
                    timeout_ms,
                    result_tx,
@@ -280,13 +587,14 @@ async fn mixer_loop(
                    if let Some(slot) = legs.get_mut(&leg_id) {
                        // Cancel any existing interaction first.
                        if let LegRole::Isolated(ref mut old_state) = slot.role {
-                            if let Some(tx) = old_state.result_tx.take() {
-                                let _ = tx.send(InteractionResult::Cancelled);
-                            }
+                            cancel_isolated_interaction(old_state);
                        }
                        let timeout_ticks = timeout_ms / 20;
                        slot.role = LegRole::Isolated(IsolationState {
                            prompt_frames: VecDeque::from(prompt_pcm_frames),
+                            prompt_stream_rx,
+                            prompt_cancel_tx,
+                            prompt_stream_finished: false,
                            expected_digits,
                            timeout_ticks_remaining: timeout_ticks,
                            prompt_done: false,
@@ -294,6 +602,9 @@ async fn mixer_loop(
                        });
                    } else {
                        // Leg not found — immediately cancel.
+                        if let Some(cancel_tx) = prompt_cancel_tx {
+                            let _ = cancel_tx.send(true);
+                        }
                        let _ = result_tx.send(InteractionResult::Cancelled);
                    }
                }
@@ -302,7 +613,14 @@ async fn mixer_loop(
                    tool_type,
                    audio_tx,
                }) => {
-                    tool_legs.insert(leg_id, ToolLegSlot { tool_type, audio_tx });
+                    tool_legs.insert(
+                        leg_id,
+                        ToolLegSlot {
+                            tool_type,
+                            audio_tx,
+                            dropped_batches: 0,
+                        },
+                    );
                }
                Ok(MixerCommand::RemoveToolLeg { leg_id }) => {
                    tool_legs.remove(&leg_id);
@@ -343,54 +661,11 @@ async fn mixer_loop(
                }
            }

-            // Step 2b: Consume exactly one frame from the jitter buffer.
-            match slot.jitter.consume() {
-                JitterResult::Packet(pkt) => {
-                    match slot.transcoder.decode_to_f32(&pkt.payload, pkt.payload_type) {
-                        Ok((pcm, rate)) => {
-                            let pcm_48k = if rate == MIX_RATE {
-                                pcm
-                            } else {
-                                slot.transcoder
-                                    .resample_f32(&pcm, rate, MIX_RATE)
-                                    .unwrap_or_else(|_| vec![0.0f32; MIX_FRAME_SIZE])
-                            };
-                            let processed = if slot.codec_pt != codec_lib::PT_OPUS {
-                                TranscodeState::denoise_f32(&mut slot.denoiser, &pcm_48k)
-                            } else {
-                                pcm_48k
-                            };
-                            let mut frame = processed;
-                            frame.resize(MIX_FRAME_SIZE, 0.0);
-                            slot.last_pcm_frame = frame;
-                        }
-                        Err(_) => {}
-                    }
-                }
-                JitterResult::Missing => {
-                    // Invoke Opus PLC or fade for non-Opus codecs.
-                    if slot.codec_pt == codec_lib::PT_OPUS {
-                        match slot.transcoder.opus_plc(MIX_FRAME_SIZE) {
-                            Ok(pcm) => {
-                                slot.last_pcm_frame = pcm;
-                            }
-                            Err(_) => {
-                                for s in slot.last_pcm_frame.iter_mut() {
-                                    *s *= 0.8;
-                                }
-                            }
-                        }
-                    } else {
-                        // Non-Opus: fade last frame toward silence.
-                        for s in slot.last_pcm_frame.iter_mut() {
-                            *s *= 0.85;
-                        }
-                    }
-                }
-                JitterResult::Filling => {
-                    slot.last_pcm_frame = vec![0.0f32; MIX_FRAME_SIZE];
-                }
-            }
+            // Step 2b: Decode enough RTP to cover one 20ms playout frame.
+            // Variable-duration packets (10ms, 20ms, 60ms, ...) accumulate in
+            // the per-leg PCM FIFO; we pop exactly one 20ms frame below.
+            fill_leg_playout_buffer(slot);
+            slot.last_pcm_frame = take_mix_frame(slot);

            // Run jitter adaptation + prune stale packets.
            slot.jitter.adapt();
@@ -404,6 +679,9 @@ async fn mixer_loop(
            }
            if slot.silent_ticks > 150 {
                slot.last_pcm_frame = vec![0.0f32; MIX_FRAME_SIZE];
+                slot.pcm_buffer.clear();
+                slot.expected_rtp_timestamp = None;
+                slot.estimated_packet_ts = rtp_clock_increment(slot.codec_pt);
            }
        }

@@ -426,12 +704,12 @@ async fn mixer_loop(
        for (lid, slot) in legs.iter_mut() {
            match &mut slot.role {
                LegRole::Participant => {
-                    // Mix-minus: total minus this leg's own contribution, clamped to [-1.0, 1.0].
+                    // Mix-minus: total minus this leg's own contribution.
+                    // Apply a light soft limiter instead of hard clipping the sum.
                    let mut mix_minus = Vec::with_capacity(MIX_FRAME_SIZE);
                    for i in 0..MIX_FRAME_SIZE {
-                        let sample =
-                            (total_mix[i] - slot.last_pcm_frame[i] as f64) as f32;
-                        mix_minus.push(sample.clamp(-1.0, 1.0));
+                        let sample = (total_mix[i] - slot.last_pcm_frame[i] as f64) as f32;
+                        mix_minus.push(soft_limit_sample(sample));
                    }

                    // Resample from 48kHz to the leg's codec native rate.
@@ -445,11 +723,10 @@ async fn mixer_loop(
                    };

                    // Encode to the leg's codec (f32 → i16 → codec inside encode_from_f32).
-                    let encoded =
-                        match slot.transcoder.encode_from_f32(&resampled, slot.codec_pt) {
-                            Ok(e) if !e.is_empty() => e,
-                            _ => continue,
-                        };
+                    let encoded = match slot.transcoder.encode_from_f32(&resampled, slot.codec_pt) {
+                        Ok(e) if !e.is_empty() => e,
+                        _ => continue,
+                    };

                    // Build RTP packet with header.
                    let header =
@@ -460,10 +737,11 @@ async fn mixer_loop(
                    slot.rtp_seq = slot.rtp_seq.wrapping_add(1);
                    slot.rtp_ts = slot.rtp_ts.wrapping_add(rtp_clock_increment(slot.codec_pt));

-                    // Non-blocking send — drop frame if channel is full.
-                    let _ = slot.outbound_tx.try_send(rtp);
+                    try_send_leg_output(&out_tx, &call_id, lid, slot, rtp, "participant-audio");
                }
                LegRole::Isolated(state) => {
+                    drain_prompt_stream(&out_tx, &call_id, lid, state);
+
                    // Check for DTMF digit from this leg.
                    let mut matched_digit: Option<char> = None;
                    for (src_lid, dtmf_pkt) in &dtmf_forward {
@@ -487,12 +765,14 @@ async fn mixer_loop(

                    if let Some(digit) = matched_digit {
                        // Interaction complete — digit matched.
-                        completed_interactions
-                            .push((lid.clone(), InteractionResult::Digit(digit)));
+                        completed_interactions.push((lid.clone(), InteractionResult::Digit(digit)));
                    } else {
-                        // Play prompt frame or silence.
+                        // Play prompt frame, wait for live TTS, or move to timeout once the
+                        // prompt stream has fully drained.
                        let pcm_frame = if let Some(frame) = state.prompt_frames.pop_front() {
                            frame
+                        } else if !state.prompt_stream_finished {
+                            vec![0.0f32; MIX_FRAME_SIZE]
                        } else {
                            state.prompt_done = true;
                            vec![0.0f32; MIX_FRAME_SIZE]
@@ -508,6 +788,7 @@ async fn mixer_loop(
                                .unwrap_or_default()
                        };

+                        let mut prompt_rtp: Option<Vec<u8>> = None;
                        if let Ok(encoded) =
                            slot.transcoder.encode_from_f32(&resampled, slot.codec_pt)
                        {
@@ -521,10 +802,9 @@ async fn mixer_loop(
                                let mut rtp = header.to_vec();
                                rtp.extend_from_slice(&encoded);
                                slot.rtp_seq = slot.rtp_seq.wrapping_add(1);
-                                slot.rtp_ts = slot
-                                    .rtp_ts
-                                    .wrapping_add(rtp_clock_increment(slot.codec_pt));
-                                let _ = slot.outbound_tx.try_send(rtp);
+                                slot.rtp_ts =
+                                    slot.rtp_ts.wrapping_add(rtp_clock_increment(slot.codec_pt));
+                                prompt_rtp = Some(rtp);
                            }
                        }

@@ -537,6 +817,17 @@ async fn mixer_loop(
                                state.timeout_ticks_remaining -= 1;
                            }
                        }
+
+                        if let Some(rtp) = prompt_rtp {
+                            try_send_leg_output(
+                                &out_tx,
+                                &call_id,
+                                lid,
+                                slot,
+                                rtp,
+                                "isolated-prompt",
+                            );
+                        }
                    }
                }
            }
@@ -546,6 +837,7 @@ async fn mixer_loop(
        for (lid, result) in completed_interactions {
            if let Some(slot) = legs.get_mut(&lid) {
                if let LegRole::Isolated(ref mut state) = slot.role {
+                    cancel_prompt_producer(state);
                    if let Some(tx) = state.result_tx.take() {
                        let _ = tx.send(result);
                    }
@@ -566,7 +858,7 @@ async fn mixer_loop(
                })
                .collect();

-            for tool in tool_legs.values() {
+            for (tool_leg_id, tool) in tool_legs.iter_mut() {
                let batch = ToolAudioBatch {
                    sources: sources
                        .iter()
@@ -576,8 +868,7 @@ async fn mixer_loop(
                        })
                        .collect(),
                };
-                // Non-blocking send — drop batch if tool can't keep up.
-                let _ = tool.audio_tx.try_send(batch);
+                try_send_tool_output(&out_tx, &call_id, tool_leg_id, tool, batch);
            }
        }

@@ -610,7 +901,7 @@ async fn mixer_loop(
                rtp_out.extend_from_slice(&dtmf_pkt.payload);
                target_slot.rtp_seq = target_slot.rtp_seq.wrapping_add(1);
                // Don't increment rtp_ts for DTMF — it shares timestamp context with audio.
-                let _ = target_slot.outbound_tx.try_send(rtp_out);
+                try_send_leg_output(&out_tx, &call_id, target_lid, target_slot, rtp_out, "dtmf");
            }
        }
    }
--- a/rust/crates/proxy-engine/src/provider.rs
+++ b/rust/crates/proxy-engine/src/provider.rs
@@ -267,11 +267,7 @@ impl ProviderManager {

    /// Try to handle a SIP response as a provider registration response.
    /// Returns true if consumed.
-    pub async fn handle_response(
-        &self,
-        msg: &SipMessage,
-        socket: &UdpSocket,
-    ) -> bool {
+    pub async fn handle_response(&self, msg: &SipMessage, socket: &UdpSocket) -> bool {
        for ps_arc in &self.providers {
            let mut ps = ps_arc.lock().await;
            let was_registered = ps.is_registered;
@@ -322,7 +318,10 @@ impl ProviderManager {
    }

    /// Find a provider by its config ID (e.g. "easybell").
-    pub async fn find_by_provider_id(&self, provider_id: &str) -> Option<Arc<Mutex<ProviderState>>> {
+    pub async fn find_by_provider_id(
+        &self,
+        provider_id: &str,
+    ) -> Option<Arc<Mutex<ProviderState>>> {
        for ps_arc in &self.providers {
            let ps = ps_arc.lock().await;
            if ps.config.id == provider_id {
--- a/rust/crates/proxy-engine/src/recorder.rs
+++ b/rust/crates/proxy-engine/src/recorder.rs
@@ -25,8 +25,7 @@ impl Recorder {
    ) -> Result<Self, String> {
        // Ensure parent directory exists.
        if let Some(parent) = Path::new(file_path).parent() {
-            std::fs::create_dir_all(parent)
-                .map_err(|e| format!("create dir: {e}"))?;
+            std::fs::create_dir_all(parent).map_err(|e| format!("create dir: {e}"))?;
        }

        let sample_rate = 8000u32; // Record at 8kHz (standard telephony)
@@ -57,10 +56,13 @@ impl Recorder {

    /// Create a recorder that writes raw PCM at a given sample rate.
    /// Used by tool legs that already have decoded PCM (no RTP processing needed).
-    pub fn new_pcm(file_path: &str, sample_rate: u32, max_duration_ms: Option<u64>) -> Result<Self, String> {
+    pub fn new_pcm(
+        file_path: &str,
+        sample_rate: u32,
+        max_duration_ms: Option<u64>,
+    ) -> Result<Self, String> {
        if let Some(parent) = Path::new(file_path).parent() {
-            std::fs::create_dir_all(parent)
-                .map_err(|e| format!("create dir: {e}"))?;
+            std::fs::create_dir_all(parent).map_err(|e| format!("create dir: {e}"))?;
        }

        let spec = hound::WavSpec {
--- a/rust/crates/proxy-engine/src/registrar.rs
+++ b/rust/crates/proxy-engine/src/registrar.rs
@@ -60,18 +60,17 @@ impl Registrar {

    /// Try to handle a SIP REGISTER from a device.
    /// Returns Some(response_bytes) if handled, None if not a known device.
-    pub fn handle_register(
-        &mut self,
-        msg: &SipMessage,
-        from_addr: SocketAddr,
-    ) -> Option<Vec<u8>> {
+    pub fn handle_register(&mut self, msg: &SipMessage, from_addr: SocketAddr) -> Option<Vec<u8>> {
        if msg.method() != Some("REGISTER") {
            return None;
        }

        // Find the device by matching the source IP against expectedAddress.
        let from_ip = from_addr.ip().to_string();
-        let device = self.devices.iter().find(|d| d.expected_address == from_ip)?;
+        let device = self
+            .devices
+            .iter()
+            .find(|d| d.expected_address == from_ip)?;

        let from_header = msg.get_header("From").unwrap_or("");
        let aor = SipMessage::extract_uri(from_header)
@@ -79,9 +78,7 @@ impl Registrar {
            .unwrap_or_else(|| format!("sip:{}@{}", device.extension, from_ip));

        let expires_header = msg.get_header("Expires");
-        let requested: u32 = expires_header
-            .and_then(|s| s.parse().ok())
-            .unwrap_or(3600);
+        let requested: u32 = expires_header.and_then(|s| s.parse().ok()).unwrap_or(3600);
        let expires = requested.min(MAX_EXPIRES);

        let entry = RegisteredDevice {
@@ -122,10 +119,7 @@ impl Registrar {
            Some(ResponseOptions {
                to_tag: Some(generate_tag()),
                contact: Some(contact),
-                extra_headers: Some(vec![(
-                    "Expires".to_string(),
-                    expires.to_string(),
-                )]),
+                extra_headers: Some(vec![("Expires".to_string(), expires.to_string())]),
                ..Default::default()
            }),
        );
@@ -145,8 +139,8 @@ impl Registrar {
    /// Find a registered device by its source IP address.
    pub fn find_by_address(&self, addr: &SocketAddr) -> Option<&RegisteredDevice> {
        let ip = addr.ip().to_string();
-        self.registered.values().find(|e| {
-            e.contact_addr.ip().to_string() == ip && Instant::now() <= e.expires_at
-        })
+        self.registered
+            .values()
+            .find(|e| e.contact_addr.ip().to_string() == ip && Instant::now() <= e.expires_at)
    }
 }
--- a/rust/crates/proxy-engine/src/rtp.rs
+++ b/rust/crates/proxy-engine/src/rtp.rs
@@ -82,10 +82,15 @@ pub fn build_rtp_header(pt: u8, seq: u16, timestamp: u32, ssrc: u32) -> [u8; 12]

 /// Get the RTP clock increment per 20ms frame for a payload type.
 pub fn rtp_clock_increment(pt: u8) -> u32 {
+    rtp_clock_rate(pt) / 50
+}
+
+/// Get the RTP clock rate for a payload type.
+pub fn rtp_clock_rate(pt: u8) -> u32 {
    match pt {
-        9 => 160,   // G.722: 8000 Hz clock rate (despite 16kHz audio) × 0.02s
-        0 | 8 => 160, // PCMU/PCMA: 8000 × 0.02
-        111 => 960, // Opus: 48000 × 0.02
-        _ => 160,
+        9 => 8000,     // G.722 uses an 8kHz RTP clock despite 16kHz audio.
+        0 | 8 => 8000, // PCMU/PCMA
+        111 => 48000,  // Opus
+        _ => 8000,
    }
 }
--- a/rust/crates/proxy-engine/src/sip_leg.rs
+++ b/rust/crates/proxy-engine/src/sip_leg.rs
@@ -128,17 +128,24 @@ impl SipLeg {
                max_forwards: Some(70),
                body: Some(sdp),
                content_type: Some("application/sdp".to_string()),
-                extra_headers: Some(vec![
-                    ("User-Agent".to_string(), "SipRouter/1.0".to_string()),
-                ]),
+                extra_headers: Some(vec![(
+                    "User-Agent".to_string(),
+                    "SipRouter/1.0".to_string(),
+                )]),
            },
        );

-        self.dialog = Some(SipDialog::from_uac_invite(&invite, ip, self.config.lan_port));
+        self.dialog = Some(SipDialog::from_uac_invite(
+            &invite,
+            ip,
+            self.config.lan_port,
+        ));
        self.invite = Some(invite.clone());
        self.state = LegState::Inviting;

-        let _ = socket.send_to(&invite.serialize(), self.config.sip_target).await;
+        let _ = socket
+            .send_to(&invite.serialize(), self.config.sip_target)
+            .await;
    }

    /// Handle an incoming SIP message routed to this leg.
@@ -443,10 +450,7 @@ pub enum SipLegAction {
 /// Build an ACK for a non-2xx response (same transaction as the INVITE).
 fn build_non_2xx_ack(original_invite: &SipMessage, response: &SipMessage) -> SipMessage {
    let via = original_invite.get_header("Via").unwrap_or("").to_string();
-    let from = original_invite
-        .get_header("From")
-        .unwrap_or("")
-        .to_string();
+    let from = original_invite.get_header("From").unwrap_or("").to_string();
    let to = response.get_header("To").unwrap_or("").to_string();
    let call_id = original_invite.call_id().to_string();
    let cseq_num: u32 = original_invite
--- a/rust/crates/proxy-engine/src/sip_transport.rs
+++ b/rust/crates/proxy-engine/src/sip_transport.rs
@@ -28,10 +28,8 @@ impl SipTransport {
    }

    /// Spawn the UDP receive loop. Calls the handler for every received packet.
-    pub fn spawn_receiver<F>(
-        &self,
-        handler: F,
-    ) where
+    pub fn spawn_receiver<F>(&self, handler: F)
+    where
        F: Fn(&[u8], SocketAddr) + Send + 'static,
    {
        let socket = self.socket.clone();
--- a/rust/crates/proxy-engine/src/tool_leg.rs
+++ b/rust/crates/proxy-engine/src/tool_leg.rs
@@ -51,7 +51,8 @@ pub fn spawn_recording_tool(
                });

                // Convert f32 [-1.0, 1.0] to i16 for WAV writing.
-                let pcm_i16: Vec<i16> = source.pcm_48k
+                let pcm_i16: Vec<i16> = source
+                    .pcm_48k
                    .iter()
                    .map(|&s| (s * 32767.0).round().clamp(-32768.0, 32767.0) as i16)
                    .collect();
--- a/rust/crates/proxy-engine/src/tts.rs
+++ b/rust/crates/proxy-engine/src/tts.rs
@@ -1,15 +1,49 @@
 //! Text-to-speech engine — synthesizes text to WAV files using Kokoro neural TTS.
 //!
 //! The model is loaded lazily on first use. If the model/voices files are not
-//! present, the generate command returns an error and the TS side falls back
-//! to espeak-ng.
+//! present, the generate command returns an error and the caller skips the prompt.
+//!
+//! Caching is handled internally via a `.meta` sidecar file next to each WAV.
+//! When `cacheable` is true, the engine checks whether the existing WAV was
+//! generated from the same text+voice; if so it returns immediately (cache hit).
+//! Callers never need to check for cached files — that is entirely this module's
+//! responsibility.

+use crate::audio_player::pcm_to_mix_frames;
 use kokoro_tts::{KokoroTts, Voice};
 use std::path::Path;
+use std::sync::Arc;
+use tokio::sync::{mpsc, watch};
+
+pub const DEFAULT_MODEL_PATH: &str = ".nogit/tts/kokoro-v1.0.onnx";
+pub const DEFAULT_VOICES_PATH: &str = ".nogit/tts/voices.bin";
+const TTS_OUTPUT_RATE: u32 = 24000;
+const MAX_CHUNK_CHARS: usize = 220;
+const MIN_CHUNK_CHARS: usize = 80;
+
+pub enum TtsStreamMessage {
+    Frames(Vec<Vec<f32>>),
+    Finished,
+    Failed(String),
+}
+
+pub struct TtsLivePrompt {
+    pub initial_frames: Vec<Vec<f32>>,
+    pub stream_rx: mpsc::Receiver<TtsStreamMessage>,
+    pub cancel_tx: watch::Sender<bool>,
+}
+
+#[derive(Clone)]
+pub struct TtsPromptRequest {
+    pub model_path: String,
+    pub voices_path: String,
+    pub voice_name: String,
+    pub text: String,
+}

 /// Wraps the Kokoro TTS engine with lazy model loading.
 pub struct TtsEngine {
-    tts: Option<KokoroTts>,
+    tts: Option<Arc<KokoroTts>>,
    /// Path that was used to load the current model (for cache invalidation).
    loaded_model_path: String,
    loaded_voices_path: String,
@@ -24,6 +58,69 @@ impl TtsEngine {
        }
    }

+    async fn ensure_loaded(
+        &mut self,
+        model_path: &str,
+        voices_path: &str,
+    ) -> Result<Arc<KokoroTts>, String> {
+        if !Path::new(model_path).exists() {
+            return Err(format!("model not found: {model_path}"));
+        }
+        if !Path::new(voices_path).exists() {
+            return Err(format!("voices not found: {voices_path}"));
+        }
+
+        if self.tts.is_none()
+            || self.loaded_model_path != model_path
+            || self.loaded_voices_path != voices_path
+        {
+            eprintln!("[tts] loading model: {model_path}");
+            let tts = Arc::new(
+                KokoroTts::new(model_path, voices_path)
+                    .await
+                    .map_err(|e| format!("model load failed: {e:?}"))?,
+            );
+            self.tts = Some(tts);
+            self.loaded_model_path = model_path.to_string();
+            self.loaded_voices_path = voices_path.to_string();
+        }
+
+        Ok(self.tts.as_ref().unwrap().clone())
+    }
+
+    pub async fn start_live_prompt(
+        &mut self,
+        request: TtsPromptRequest,
+    ) -> Result<TtsLivePrompt, String> {
+        if request.text.trim().is_empty() {
+            return Err("empty text".into());
+        }
+
+        let tts = self
+            .ensure_loaded(&request.model_path, &request.voices_path)
+            .await?;
+        let voice = select_voice(&request.voice_name);
+        let chunks = chunk_text(&request.text);
+        if chunks.is_empty() {
+            return Err("empty text".into());
+        }
+
+        let initial_frames = synth_text_to_mix_frames(&tts, chunks[0].as_str(), voice).await?;
+        let remaining_chunks: Vec<String> = chunks.into_iter().skip(1).collect();
+        let (stream_tx, stream_rx) = mpsc::channel(8);
+        let (cancel_tx, cancel_rx) = watch::channel(false);
+
+        tokio::spawn(async move {
+            stream_live_prompt_chunks(tts, voice, remaining_chunks, stream_tx, cancel_rx).await;
+        });
+
+        Ok(TtsLivePrompt {
+            initial_frames,
+            stream_rx,
+            cancel_tx,
+        })
+    }
+
    /// Generate a WAV file from text.
    ///
    /// Params (from IPC JSON):
@@ -32,52 +129,64 @@ impl TtsEngine {
    ///   - `voice`: voice name (e.g. "af_bella")
    ///   - `text`: text to synthesize
    ///   - `output`: output WAV file path
-    pub async fn generate(&mut self, params: &serde_json::Value) -> Result<serde_json::Value, String> {
-        let model_path = params.get("model").and_then(|v| v.as_str())
+    ///   - `cacheable`: if true, skip synthesis when the output WAV already
+    ///     matches the same text+voice (checked via a `.meta` sidecar file)
+    pub async fn generate(
+        &mut self,
+        params: &serde_json::Value,
+    ) -> Result<serde_json::Value, String> {
+        let model_path = params
+            .get("model")
+            .and_then(|v| v.as_str())
            .ok_or("missing 'model' param")?;
-        let voices_path = params.get("voices").and_then(|v| v.as_str())
+        let voices_path = params
+            .get("voices")
+            .and_then(|v| v.as_str())
            .ok_or("missing 'voices' param")?;
-        let voice_name = params.get("voice").and_then(|v| v.as_str())
+        let voice_name = params
+            .get("voice")
+            .and_then(|v| v.as_str())
            .unwrap_or("af_bella");
-        let text = params.get("text").and_then(|v| v.as_str())
+        let text = params
+            .get("text")
+            .and_then(|v| v.as_str())
            .ok_or("missing 'text' param")?;
-        let output_path = params.get("output").and_then(|v| v.as_str())
+        let output_path = params
+            .get("output")
+            .and_then(|v| v.as_str())
            .ok_or("missing 'output' param")?;
+        let cacheable = params
+            .get("cacheable")
+            .and_then(|v| v.as_bool())
+            .unwrap_or(false);

        if text.is_empty() {
            return Err("empty text".into());
        }

-        // Check that model/voices files exist.
-        if !Path::new(model_path).exists() {
-            return Err(format!("model not found: {model_path}"));
-        }
-        if !Path::new(voices_path).exists() {
-            return Err(format!("voices not found: {voices_path}"));
+        // Cache check: if cacheable and the sidecar matches, return immediately.
+        if cacheable && self.is_cache_hit(output_path, text, voice_name) {
+            eprintln!("[tts] cache hit: {output_path}");
+            return Ok(serde_json::json!({ "output": output_path }));
        }

-        // Lazy-load or reload if paths changed.
-        if self.tts.is_none()
-            || self.loaded_model_path != model_path
-            || self.loaded_voices_path != voices_path
-        {
-            eprintln!("[tts] loading model: {model_path}");
-            let tts = KokoroTts::new(model_path, voices_path)
-                .await
-                .map_err(|e| format!("model load failed: {e:?}"))?;
-            self.tts = Some(tts);
-            self.loaded_model_path = model_path.to_string();
-            self.loaded_voices_path = voices_path.to_string();
+        // Ensure parent directory exists.
+        if let Some(parent) = Path::new(output_path).parent() {
+            let _ = std::fs::create_dir_all(parent);
        }

-        let tts = self.tts.as_ref().unwrap();
+        let tts = self.ensure_loaded(model_path, voices_path).await?;
        let voice = select_voice(voice_name);

-        eprintln!("[tts] synthesizing voice '{voice_name}': \"{text}\"");
-        let (samples, duration) = tts.synth(text, voice)
+        eprintln!("[tts] synthesizing WAV voice '{voice_name}' to {output_path}");
+        let (samples, duration) = tts
+            .synth(text, voice)
            .await
            .map_err(|e| format!("synthesis failed: {e:?}"))?;
-        eprintln!("[tts] synthesized {} samples in {duration:?}", samples.len());
+        eprintln!(
+            "[tts] synthesized {} samples in {duration:?}",
+            samples.len()
+        );

        // Write 24kHz 16-bit mono WAV.
        let spec = hound::WavSpec {
@@ -91,13 +200,149 @@ impl TtsEngine {
            .map_err(|e| format!("WAV create failed: {e}"))?;
        for &sample in &samples {
            let s16 = (sample * 32767.0).round().clamp(-32768.0, 32767.0) as i16;
-            writer.write_sample(s16).map_err(|e| format!("WAV write: {e}"))?;
+            writer
+                .write_sample(s16)
+                .map_err(|e| format!("WAV write: {e}"))?;
+        }
+        writer
+            .finalize()
+            .map_err(|e| format!("WAV finalize: {e}"))?;
+
+        // Write sidecar for future cache checks.
+        if cacheable {
+            self.write_cache_meta(output_path, text, voice_name);
        }
-        writer.finalize().map_err(|e| format!("WAV finalize: {e}"))?;

        eprintln!("[tts] wrote {output_path}");
        Ok(serde_json::json!({ "output": output_path }))
    }
+
+    // -----------------------------------------------------------------------
+    // Cache helpers
+    // -----------------------------------------------------------------------
+
+    /// Check if the WAV + sidecar on disk match the given text+voice.
+    fn is_cache_hit(&self, output_path: &str, text: &str, voice: &str) -> bool {
+        let meta_path = format!("{output_path}.meta");
+        if !Path::new(output_path).exists() || !Path::new(&meta_path).exists() {
+            return false;
+        }
+        match std::fs::read_to_string(&meta_path) {
+            Ok(contents) => contents == Self::cache_key(text, voice),
+            Err(_) => false,
+        }
+    }
+
+    /// Write the sidecar `.meta` file next to the WAV.
+    fn write_cache_meta(&self, output_path: &str, text: &str, voice: &str) {
+        let meta_path = format!("{output_path}.meta");
+        let _ = std::fs::write(&meta_path, Self::cache_key(text, voice));
+    }
+
+    /// Build the cache key from text + voice.
+    fn cache_key(text: &str, voice: &str) -> String {
+        format!("{}\0{}", text, voice)
+    }
+}
+
+async fn synth_text_to_mix_frames(
+    tts: &Arc<KokoroTts>,
+    text: &str,
+    voice: Voice,
+) -> Result<Vec<Vec<f32>>, String> {
+    let (samples, duration) = tts
+        .synth(text, voice)
+        .await
+        .map_err(|e| format!("synthesis failed: {e:?}"))?;
+    eprintln!(
+        "[tts] synthesized chunk ({} chars, {} samples) in {duration:?}",
+        text.chars().count(),
+        samples.len()
+    );
+    pcm_to_mix_frames(&samples, TTS_OUTPUT_RATE)
+}
+
+async fn stream_live_prompt_chunks(
+    tts: Arc<KokoroTts>,
+    voice: Voice,
+    chunks: Vec<String>,
+    stream_tx: mpsc::Sender<TtsStreamMessage>,
+    mut cancel_rx: watch::Receiver<bool>,
+) {
+    for chunk in chunks {
+        if *cancel_rx.borrow() {
+            break;
+        }
+
+        match synth_text_to_mix_frames(&tts, &chunk, voice).await {
+            Ok(frames) => {
+                if *cancel_rx.borrow() {
+                    break;
+                }
+                if stream_tx.send(TtsStreamMessage::Frames(frames)).await.is_err() {
+                    return;
+                }
+            }
+            Err(error) => {
+                let _ = stream_tx.send(TtsStreamMessage::Failed(error)).await;
+                return;
+            }
+        }
+
+        if cancel_rx.has_changed().unwrap_or(false) && *cancel_rx.borrow_and_update() {
+            break;
+        }
+    }
+
+    let _ = stream_tx.send(TtsStreamMessage::Finished).await;
+}
+
+fn chunk_text(text: &str) -> Vec<String> {
+    let mut chunks = Vec::new();
+    let mut current = String::new();
+
+    for ch in text.chars() {
+        current.push(ch);
+
+        let len = current.chars().count();
+        let hard_split = len >= MAX_CHUNK_CHARS && (ch.is_whitespace() || is_soft_boundary(ch));
+        let natural_split = len >= MIN_CHUNK_CHARS && is_sentence_boundary(ch);
+
+        if natural_split || hard_split {
+            push_chunk(&mut chunks, &mut current);
+        }
+    }
+
+    push_chunk(&mut chunks, &mut current);
+
+    if chunks.len() >= 2 {
+        let last_len = chunks.last().unwrap().chars().count();
+        if last_len < (MIN_CHUNK_CHARS / 2) {
+            let tail = chunks.pop().unwrap();
+            if let Some(prev) = chunks.last_mut() {
+                prev.push(' ');
+                prev.push_str(tail.trim());
+            }
+        }
+    }
+
+    chunks
+}
+
+fn push_chunk(chunks: &mut Vec<String>, current: &mut String) {
+    let trimmed = current.trim();
+    if !trimmed.is_empty() {
+        chunks.push(trimmed.to_string());
+    }
+    current.clear();
+}
+
+fn is_sentence_boundary(ch: char) -> bool {
+    matches!(ch, '.' | '!' | '?' | '\n' | ';' | ':')
+}
+
+fn is_soft_boundary(ch: char) -> bool {
+    matches!(ch, ',' | ';' | ':' | ')' | ']' | '\n')
 }

 /// Map voice name string to Kokoro Voice enum variant.
--- a/rust/crates/proxy-engine/src/voicemail.rs
+++ b/rust/crates/proxy-engine/src/voicemail.rs
@@ -128,8 +128,8 @@ async fn record_from_socket(
                    break; // Max duration reached.
                }
            }
-            Ok(Err(_)) => break,     // Socket error (closed).
-            Err(_) => break,          // Timeout (max duration + grace).
+            Ok(Err(_)) => break, // Socket error (closed).
+            Err(_) => break,     // Timeout (max duration + grace).
        }
    }

--- a/rust/crates/proxy-engine/src/webrtc_engine.rs
+++ b/rust/crates/proxy-engine/src/webrtc_engine.rs
@@ -58,9 +58,7 @@ impl WebRtcEngine {
            .register_default_codecs()
            .map_err(|e| format!("register codecs: {e}"))?;

-        let api = APIBuilder::new()
-            .with_media_engine(media_engine)
-            .build();
+        let api = APIBuilder::new().with_media_engine(media_engine).build();

        let config = RTCConfiguration {
            ice_servers: vec![],
@@ -91,8 +89,7 @@ impl WebRtcEngine {
            .map_err(|e| format!("add track: {e}"))?;

        // Shared mixer channel sender (populated when linked to a call).
-        let mixer_tx: Arc<Mutex<Option<mpsc::Sender<RtpPacket>>>> =
-            Arc::new(Mutex::new(None));
+        let mixer_tx: Arc<Mutex<Option<mpsc::Sender<RtpPacket>>>> = Arc::new(Mutex::new(None));

        // ICE candidate handler.
        let out_tx_ice = self.out_tx.clone();
@@ -256,7 +253,11 @@ impl WebRtcEngine {

    pub async fn close_session(&mut self, session_id: &str) -> Result<(), String> {
        if let Some(session) = self.sessions.remove(session_id) {
-            session.pc.close().await.map_err(|e| format!("close: {e}"))?;
+            session
+                .pc
+                .close()
+                .await
+                .map_err(|e| format!("close: {e}"))?;
        }
        Ok(())
    }
--- a/rust/crates/sip-proto/src/dialog.rs
+++ b/rust/crates/sip-proto/src/dialog.rs
@@ -51,9 +51,7 @@ impl SipDialog {
                .map(|s| s.to_string())
                .unwrap_or_else(generate_tag),
            remote_tag: None,
-            local_uri: SipMessage::extract_uri(from)
-                .unwrap_or("")
-                .to_string(),
+            local_uri: SipMessage::extract_uri(from).unwrap_or("").to_string(),
            remote_uri: SipMessage::extract_uri(to).unwrap_or("").to_string(),
            local_cseq,
            remote_cseq: 0,
@@ -181,10 +179,7 @@ impl SipDialog {
                format!("<{}>{remote_tag_str}", self.remote_uri),
            ),
            ("Call-ID".to_string(), self.call_id.clone()),
-            (
-                "CSeq".to_string(),
-                format!("{} {method}", self.local_cseq),
-            ),
+            ("CSeq".to_string(), format!("{} {method}", self.local_cseq)),
            ("Max-Forwards".to_string(), "70".to_string()),
        ];

@@ -243,10 +238,7 @@ impl SipDialog {
                format!("<{}>{remote_tag_str}", self.remote_uri),
            ),
            ("Call-ID".to_string(), self.call_id.clone()),
-            (
-                "CSeq".to_string(),
-                format!("{} ACK", self.local_cseq),
-            ),
+            ("CSeq".to_string(), format!("{} ACK", self.local_cseq)),
            ("Max-Forwards".to_string(), "70".to_string()),
        ];

@@ -271,10 +263,7 @@ impl SipDialog {
            ("From".to_string(), from),
            ("To".to_string(), to),
            ("Call-ID".to_string(), self.call_id.clone()),
-            (
-                "CSeq".to_string(),
-                format!("{} CANCEL", self.local_cseq),
-            ),
+            ("CSeq".to_string(), format!("{} CANCEL", self.local_cseq)),
            ("Max-Forwards".to_string(), "70".to_string()),
            ("Content-Length".to_string(), "0".to_string()),
        ];
@@ -284,11 +273,7 @@ impl SipDialog {
            .unwrap_or(&self.remote_target)
            .to_string();

-        SipMessage::new(
-            format!("CANCEL {ruri} SIP/2.0"),
-            headers,
-            String::new(),
-        )
+        SipMessage::new(format!("CANCEL {ruri} SIP/2.0"), headers, String::new())
    }

    /// Transition the dialog to terminated state.
--- a/rust/crates/sip-proto/src/helpers.rs
+++ b/rust/crates/sip-proto/src/helpers.rs
@@ -27,7 +27,9 @@ pub fn generate_branch() -> String {

 fn random_hex(bytes: usize) -> String {
    let mut rng = rand::thread_rng();
-    (0..bytes).map(|_| format!("{:02x}", rng.gen::<u8>())).collect()
+    (0..bytes)
+        .map(|_| format!("{:02x}", rng.gen::<u8>()))
+        .collect()
 }

 // ---- Codec registry --------------------------------------------------------
@@ -142,7 +144,9 @@ pub fn parse_digest_challenge(header: &str) -> Option<DigestChallenge> {
                return Some(after[1..1 + end].to_string());
            }
            // Unquoted value.
-            let end = after.find(|c: char| c == ',' || c.is_whitespace()).unwrap_or(after.len());
+            let end = after
+                .find(|c: char| c == ',' || c.is_whitespace())
+                .unwrap_or(after.len());
            return Some(after[..end].to_string());
        }
        None
@@ -241,11 +245,7 @@ pub struct MwiResult {
    pub extra_headers: Vec<(String, String)>,
 }

-pub fn build_mwi_body(
-    new_messages: u32,
-    old_messages: u32,
-    account_uri: &str,
-) -> MwiResult {
+pub fn build_mwi_body(new_messages: u32, old_messages: u32, account_uri: &str) -> MwiResult {
    let waiting = if new_messages > 0 { "yes" } else { "no" };
    let body = format!(
        "Messages-Waiting: {waiting}\r\n\
--- a/rust/crates/sip-proto/src/lib.rs
+++ b/rust/crates/sip-proto/src/lib.rs
@@ -4,9 +4,9 @@
 //! SDP handling, Digest authentication, and URI rewriting.
 //! Ported from the TypeScript `ts/sip/` library.

-pub mod message;
 pub mod dialog;
 pub mod helpers;
+pub mod message;
 pub mod rewrite;

 /// Network endpoint (address + port + optional negotiated codec).
--- a/rust/crates/sip-proto/src/message.rs
+++ b/rust/crates/sip-proto/src/message.rs
@@ -14,7 +14,11 @@ pub struct SipMessage {

 impl SipMessage {
    pub fn new(start_line: String, headers: Vec<(String, String)>, body: String) -> Self {
-        Self { start_line, headers, body }
+        Self {
+            start_line,
+            headers,
+            body,
+        }
    }

    // ---- Parsing -----------------------------------------------------------
@@ -175,7 +179,8 @@ impl SipMessage {

    /// Inserts a header at the top of the header list.
    pub fn prepend_header(&mut self, name: &str, value: &str) -> &mut Self {
-        self.headers.insert(0, (name.to_string(), value.to_string()));
+        self.headers
+            .insert(0, (name.to_string(), value.to_string()));
        self
    }

@@ -233,10 +238,7 @@ impl SipMessage {
            .to_display_name
            .map(|d| format!("\"{d}\" "))
            .unwrap_or_default();
-        let to_tag_str = opts
-            .to_tag
-            .map(|t| format!(";tag={t}"))
-            .unwrap_or_default();
+        let to_tag_str = opts.to_tag.map(|t| format!(";tag={t}")).unwrap_or_default();

        let mut headers = vec![
            (
@@ -364,7 +366,43 @@ impl SipMessage {
                .find(|c: char| c == ';' || c == '>')
                .unwrap_or(trimmed.len());
            let result = &trimmed[..end];
-            if result.is_empty() { None } else { Some(result) }
+            if result.is_empty() {
+                None
+            } else {
+                Some(result)
+            }
+        }
+    }
+
+    /// Extract the user part from a SIP/TEL URI or header value.
+    pub fn extract_uri_user(uri_or_header_value: &str) -> Option<&str> {
+        let raw = Self::extract_uri(uri_or_header_value).unwrap_or(uri_or_header_value);
+        let raw = raw.trim();
+        if raw.is_empty() {
+            return None;
+        }
+
+        let user_part = if raw
+            .get(..5)
+            .is_some_and(|prefix| prefix.eq_ignore_ascii_case("sips:"))
+        {
+            &raw[5..]
+        } else if raw.get(..4).is_some_and(|prefix| {
+            prefix.eq_ignore_ascii_case("sip:") || prefix.eq_ignore_ascii_case("tel:")
+        }) {
+            &raw[4..]
+        } else {
+            raw
+        };
+
+        let end = user_part
+            .find(|c: char| matches!(c, '@' | ';' | '?' | '>'))
+            .unwrap_or(user_part.len());
+        let result = &user_part[..end];
+        if result.is_empty() {
+            None
+        } else {
+            Some(result)
        }
    }
 }
@@ -506,6 +544,19 @@ mod tests {
            SipMessage::extract_uri("\"Name\" <sip:user@host>;tag=abc"),
            Some("sip:user@host")
        );
+        assert_eq!(
+            SipMessage::extract_uri_user("\"Name\" <sip:+49 421 219694@host>;tag=abc"),
+            Some("+49 421 219694")
+        );
+        assert_eq!(
+            SipMessage::extract_uri_user("sip:0049421219694@voip.easybell.de"),
+            Some("0049421219694")
+        );
+        assert_eq!(
+            SipMessage::extract_uri_user("tel:+49421219694;phone-context=example.com"),
+            Some("+49421219694")
+        );
+        assert_eq!(SipMessage::extract_uri_user("SIP:user@host"), Some("user"));
    }

    #[test]
@@ -535,7 +586,10 @@ mod tests {
        );
        assert_eq!(invite.method(), Some("INVITE"));
        assert_eq!(invite.call_id(), "test-123");
-        assert!(invite.get_header("Via").unwrap().contains("192.168.1.1:5070"));
+        assert!(invite
+            .get_header("Via")
+            .unwrap()
+            .contains("192.168.1.1:5070"));

        let response = SipMessage::create_response(
            200,
--- a/rust/crates/sip-proto/src/rewrite.rs
+++ b/rust/crates/sip-proto/src/rewrite.rs
@@ -92,7 +92,11 @@ pub fn rewrite_sdp(body: &str, ip: &str, port: u16) -> (String, Option<Endpoint>
        .collect();

    let original = match (orig_addr, orig_port) {
-        (Some(a), Some(p)) => Some(Endpoint { address: a, port: p, codec_pt: None }),
+        (Some(a), Some(p)) => Some(Endpoint {
+            address: a,
+            port: p,
+            codec_pt: None,
+        }),
        _ => None,
    };

--- a/ts/00_commitinfo_data.ts
+++ b/ts/00_commitinfo_data.ts
@@ -3,6 +3,6 @@
 */
 export const commitinfo = {
  name: 'siprouter',
-  version: '1.20.2',
+  version: '1.25.0',
  description: 'undefined'
 }
--- a/ts/announcement.ts
+++ b/ts/announcement.ts
@@ -1,137 +0,0 @@
-/**
- * TTS announcement module — generates announcement WAV files at startup.
- *
- * Engine priority: espeak-ng (formant TTS, fast) → Kokoro neural TTS via
- * proxy-engine → disabled.
- *
- * The generated WAV is left on disk for Rust's audio_player / start_interaction
- * to play during calls. No encoding or RTP playback happens in TypeScript.
- */
-
-import { execSync } from 'node:child_process';
-import fs from 'node:fs';
-import path from 'node:path';
-import { sendProxyCommand, isProxyReady } from './proxybridge.ts';
-
-// ---------------------------------------------------------------------------
-// State
-// ---------------------------------------------------------------------------
-
-const TTS_DIR = path.join(process.cwd(), '.nogit', 'tts');
-const ANNOUNCEMENT_TEXT = "Hello. I'm connecting your call now.";
-const CACHE_WAV = path.join(TTS_DIR, 'announcement.wav');
-
-// Kokoro fallback constants.
-const KOKORO_MODEL = 'kokoro-v1.0.onnx';
-const KOKORO_VOICES = 'voices.bin';
-const KOKORO_VOICE = 'af_bella';
-
-// ---------------------------------------------------------------------------
-// TTS generators
-// ---------------------------------------------------------------------------
-
-/** Check if espeak-ng is available on the system. */
-function isEspeakAvailable(): boolean {
-  try {
-    execSync('which espeak-ng', { stdio: 'pipe' });
-    return true;
-  } catch {
-    return false;
-  }
-}
-
-/** Generate announcement WAV via espeak-ng (primary engine). */
-function generateViaEspeak(wavPath: string, text: string, log: (msg: string) => void): boolean {
-  log('[tts] generating announcement audio via espeak-ng...');
-  try {
-    execSync(
-      `espeak-ng -v en-us -s 150 -w "${wavPath}" "${text}"`,
-      { timeout: 10000, stdio: 'pipe' },
-    );
-    log('[tts] espeak-ng WAV generated');
-    return true;
-  } catch (e: any) {
-    log(`[tts] espeak-ng failed: ${e.message}`);
-    return false;
-  }
-}
-
-/** Generate announcement WAV via Kokoro TTS (fallback, runs inside proxy-engine). */
-async function generateViaKokoro(wavPath: string, text: string, log: (msg: string) => void): Promise<boolean> {
-  const modelPath = path.join(TTS_DIR, KOKORO_MODEL);
-  const voicesPath = path.join(TTS_DIR, KOKORO_VOICES);
-
-  if (!fs.existsSync(modelPath) || !fs.existsSync(voicesPath)) {
-    log('[tts] Kokoro model/voices not found — Kokoro fallback unavailable');
-    return false;
-  }
-
-  if (!isProxyReady()) {
-    log('[tts] proxy-engine not ready — Kokoro fallback unavailable');
-    return false;
-  }
-
-  log('[tts] generating announcement audio via Kokoro TTS (fallback)...');
-  try {
-    await sendProxyCommand('generate_tts', {
-      model: modelPath,
-      voices: voicesPath,
-      voice: KOKORO_VOICE,
-      text,
-      output: wavPath,
-    });
-    log('[tts] Kokoro WAV generated (via proxy-engine)');
-    return true;
-  } catch (e: any) {
-    log(`[tts] Kokoro failed: ${e.message}`);
-    return false;
-  }
-}
-
-// ---------------------------------------------------------------------------
-// Initialization
-// ---------------------------------------------------------------------------
-
-/**
- * Pre-generate the announcement WAV file.
- * Must be called after the proxy engine is initialized.
- *
- * Engine priority: espeak-ng → Kokoro → disabled.
- */
-export async function initAnnouncement(log: (msg: string) => void): Promise<boolean> {
-  fs.mkdirSync(TTS_DIR, { recursive: true });
-
-  try {
-    if (!fs.existsSync(CACHE_WAV)) {
-      let generated = false;
-
-      // Try espeak-ng first.
-      if (isEspeakAvailable()) {
-        generated = generateViaEspeak(CACHE_WAV, ANNOUNCEMENT_TEXT, log);
-      } else {
-        log('[tts] espeak-ng not installed — trying Kokoro fallback');
-      }
-
-      // Fall back to Kokoro (via proxy-engine).
-      if (!generated) {
-        generated = await generateViaKokoro(CACHE_WAV, ANNOUNCEMENT_TEXT, log);
-      }
-
-      if (!generated) {
-        log('[tts] no TTS engine available — announcements disabled');
-        return false;
-      }
-    }
-
-    log('[tts] announcement WAV ready');
-    return true;
-  } catch (e: any) {
-    log(`[tts] init error: ${e.message}`);
-    return false;
-  }
-}
-
-/** Get the path to the cached announcement WAV, or null if not generated. */
-export function getAnnouncementWavPath(): string | null {
-  return fs.existsSync(CACHE_WAV) ? CACHE_WAV : null;
-}
--- a/ts/call/prompt-cache.ts
+++ b/ts/call/prompt-cache.ts
@@ -1,275 +0,0 @@
-/**
- * PromptCache — manages named audio prompt WAV files for IVR and voicemail.
- *
- * Generates WAV files via espeak-ng (primary) or Kokoro TTS through the
- * proxy-engine (fallback). Also supports loading pre-existing WAV files
- * and programmatic tone generation.
- *
- * All audio playback happens in Rust (audio_player / start_interaction).
- * This module only manages WAV files on disk.
- */
-
-import { execSync } from 'node:child_process';
-import fs from 'node:fs';
-import path from 'node:path';
-import { Buffer } from 'node:buffer';
-import { sendProxyCommand, isProxyReady } from '../proxybridge.ts';
-
-// ---------------------------------------------------------------------------
-// Types
-// ---------------------------------------------------------------------------
-
-/** A cached prompt — just a WAV file path and metadata. */
-export interface ICachedPrompt {
-  /** Unique prompt identifier. */
-  id: string;
-  /** Path to the WAV file on disk. */
-  wavPath: string;
-  /** Total duration in milliseconds (approximate, from WAV header). */
-  durationMs: number;
-}
-
-// ---------------------------------------------------------------------------
-// TTS helpers
-// ---------------------------------------------------------------------------
-
-const TTS_DIR = path.join(process.cwd(), '.nogit', 'tts');
-
-/** Check if espeak-ng is available. */
-function isEspeakAvailable(): boolean {
-  try {
-    execSync('which espeak-ng', { stdio: 'pipe' });
-    return true;
-  } catch {
-    return false;
-  }
-}
-
-/** Generate WAV via espeak-ng. */
-function generateViaEspeak(wavPath: string, text: string): boolean {
-  try {
-    execSync(
-      `espeak-ng -v en-us -s 150 -w "${wavPath}" "${text}"`,
-      { timeout: 10000, stdio: 'pipe' },
-    );
-    return true;
-  } catch {
-    return false;
-  }
-}
-
-/** Generate WAV via Kokoro TTS (runs inside proxy-engine). */
-async function generateViaKokoro(wavPath: string, text: string, voice: string): Promise<boolean> {
-  const modelPath = path.join(TTS_DIR, 'kokoro-v1.0.onnx');
-  const voicesPath = path.join(TTS_DIR, 'voices.bin');
-  if (!fs.existsSync(modelPath) || !fs.existsSync(voicesPath)) return false;
-  if (!isProxyReady()) return false;
-
-  try {
-    await sendProxyCommand('generate_tts', {
-      model: modelPath,
-      voices: voicesPath,
-      voice,
-      text,
-      output: wavPath,
-    });
-    return true;
-  } catch {
-    return false;
-  }
-}
-
-/** Read a WAV file's duration from its header. */
-function getWavDurationMs(wavPath: string): number {
-  try {
-    const wav = fs.readFileSync(wavPath);
-    if (wav.length < 44) return 0;
-    if (wav.toString('ascii', 0, 4) !== 'RIFF') return 0;
-
-    let sampleRate = 16000;
-    let dataSize = 0;
-    let bitsPerSample = 16;
-    let channels = 1;
-    let offset = 12;
-
-    while (offset < wav.length - 8) {
-      const chunkId = wav.toString('ascii', offset, offset + 4);
-      const chunkSize = wav.readUInt32LE(offset + 4);
-      if (chunkId === 'fmt ') {
-        channels = wav.readUInt16LE(offset + 10);
-        sampleRate = wav.readUInt32LE(offset + 12);
-        bitsPerSample = wav.readUInt16LE(offset + 22);
-      }
-      if (chunkId === 'data') {
-        dataSize = chunkSize;
-      }
-      offset += 8 + chunkSize;
-      if (offset % 2 !== 0) offset++;
-    }
-
-    const bytesPerSample = (bitsPerSample / 8) * channels;
-    const totalSamples = bytesPerSample > 0 ? dataSize / bytesPerSample : 0;
-    return sampleRate > 0 ? Math.round((totalSamples / sampleRate) * 1000) : 0;
-  } catch {
-    return 0;
-  }
-}
-
-// ---------------------------------------------------------------------------
-// PromptCache
-// ---------------------------------------------------------------------------
-
-export class PromptCache {
-  private prompts = new Map<string, ICachedPrompt>();
-  private log: (msg: string) => void;
-  private espeakAvailable: boolean | null = null;
-
-  constructor(log: (msg: string) => void) {
-    this.log = log;
-  }
-
-  // -------------------------------------------------------------------------
-  // Public API
-  // -------------------------------------------------------------------------
-
-  /** Get a cached prompt by ID. */
-  get(id: string): ICachedPrompt | null {
-    return this.prompts.get(id) ?? null;
-  }
-
-  /** Check if a prompt is cached. */
-  has(id: string): boolean {
-    return this.prompts.has(id);
-  }
-
-  /** List all cached prompt IDs. */
-  listIds(): string[] {
-    return [...this.prompts.keys()];
-  }
-
-  /**
-   * Generate a TTS prompt WAV and cache its path.
-   * Uses espeak-ng (primary) or Kokoro (fallback).
-   */
-  async generatePrompt(id: string, text: string, voice = 'af_bella'): Promise<ICachedPrompt | null> {
-    fs.mkdirSync(TTS_DIR, { recursive: true });
-    const wavPath = path.join(TTS_DIR, `prompt-${id}.wav`);
-
-    // Check espeak availability once.
-    if (this.espeakAvailable === null) {
-      this.espeakAvailable = isEspeakAvailable();
-    }
-
-    // Generate WAV if not already on disk.
-    if (!fs.existsSync(wavPath)) {
-      let generated = false;
-      if (this.espeakAvailable) {
-        generated = generateViaEspeak(wavPath, text);
-      }
-      if (!generated) {
-        generated = await generateViaKokoro(wavPath, text, voice);
-      }
-      if (!generated) {
-        this.log(`[prompt-cache] failed to generate TTS for "${id}"`);
-        return null;
-      }
-      this.log(`[prompt-cache] generated WAV for "${id}"`);
-    }
-
-    return this.registerWav(id, wavPath);
-  }
-
-  /**
-   * Load a pre-existing WAV file as a prompt.
-   */
-  async loadWavPrompt(id: string, wavPath: string): Promise<ICachedPrompt | null> {
-    if (!fs.existsSync(wavPath)) {
-      this.log(`[prompt-cache] WAV not found: ${wavPath}`);
-      return null;
-    }
-    return this.registerWav(id, wavPath);
-  }
-
-  /**
-   * Generate a beep tone WAV and cache it.
-   */
-  async generateBeep(
-    id: string,
-    freqHz = 1000,
-    durationMs = 500,
-    amplitude = 8000,
-  ): Promise<ICachedPrompt | null> {
-    fs.mkdirSync(TTS_DIR, { recursive: true });
-    const wavPath = path.join(TTS_DIR, `prompt-${id}.wav`);
-
-    if (!fs.existsSync(wavPath)) {
-      // Generate 16kHz 16-bit mono sine wave WAV.
-      const sampleRate = 16000;
-      const totalSamples = Math.floor((sampleRate * durationMs) / 1000);
-      const pcm = Buffer.alloc(totalSamples * 2);
-
-      for (let i = 0; i < totalSamples; i++) {
-        const t = i / sampleRate;
-        const fadeLen = Math.floor(sampleRate * 0.01); // 10ms fade
-        let envelope = 1.0;
-        if (i < fadeLen) envelope = i / fadeLen;
-        else if (i > totalSamples - fadeLen) envelope = (totalSamples - i) / fadeLen;
-
-        const sample = Math.round(Math.sin(2 * Math.PI * freqHz * t) * amplitude * envelope);
-        pcm.writeInt16LE(Math.max(-32768, Math.min(32767, sample)), i * 2);
-      }
-
-      // Write WAV file.
-      const headerSize = 44;
-      const dataSize = pcm.length;
-      const wav = Buffer.alloc(headerSize + dataSize);
-
-      // RIFF header
-      wav.write('RIFF', 0);
-      wav.writeUInt32LE(36 + dataSize, 4);
-      wav.write('WAVE', 8);
-
-      // fmt chunk
-      wav.write('fmt ', 12);
-      wav.writeUInt32LE(16, 16);        // chunk size
-      wav.writeUInt16LE(1, 20);         // PCM format
-      wav.writeUInt16LE(1, 22);         // mono
-      wav.writeUInt32LE(sampleRate, 24);
-      wav.writeUInt32LE(sampleRate * 2, 28); // byte rate
-      wav.writeUInt16LE(2, 32);         // block align
-      wav.writeUInt16LE(16, 34);        // bits per sample
-
-      // data chunk
-      wav.write('data', 36);
-      wav.writeUInt32LE(dataSize, 40);
-      pcm.copy(wav, 44);
-
-      fs.writeFileSync(wavPath, wav);
-      this.log(`[prompt-cache] beep WAV generated for "${id}"`);
-    }
-
-    return this.registerWav(id, wavPath);
-  }
-
-  /** Remove a prompt from the cache. */
-  remove(id: string): void {
-    this.prompts.delete(id);
-  }
-
-  /** Clear all cached prompts. */
-  clear(): void {
-    this.prompts.clear();
-  }
-
-  // -------------------------------------------------------------------------
-  // Internal
-  // -------------------------------------------------------------------------
-
-  private registerWav(id: string, wavPath: string): ICachedPrompt {
-    const durationMs = getWavDurationMs(wavPath);
-    const prompt: ICachedPrompt = { id, wavPath, durationMs };
-    this.prompts.set(id, prompt);
-    this.log(`[prompt-cache] cached "${id}": ${wavPath} (${(durationMs / 1000).toFixed(1)}s)`);
-    return prompt;
-  }
-}
--- a/ts/config.ts
+++ b/ts/config.ts
@@ -8,6 +8,7 @@

 import fs from 'node:fs';
 import path from 'node:path';
+import type { IVoiceboxConfig } from './voicebox.js';

 // ---------------------------------------------------------------------------
 // Shared types (previously in ts/sip/types.ts, now inlined)
@@ -47,6 +48,24 @@ export interface IDeviceConfig {
  extension: string;
 }

+export type TIncomingNumberMode = 'single' | 'range' | 'regex';
+
+export interface IIncomingNumberConfig {
+  id: string;
+  label: string;
+  providerId?: string;
+  mode: TIncomingNumberMode;
+  countryCode?: string;
+  areaCode?: string;
+  localNumber?: string;
+  rangeEnd?: string;
+  pattern?: string;
+
+  // Legacy persisted fields kept for migration compatibility.
+  number?: string;
+  rangeStart?: string;
+}
+
 // ---------------------------------------------------------------------------
 // Match/Action routing model
 // ---------------------------------------------------------------------------
@@ -61,8 +80,11 @@ export interface ISipRouteMatch {
  direction: 'inbound' | 'outbound';

  /**
-   * Match the dialed/called number (To/Request-URI for inbound DID, dialed digits for outbound).
-   * Supports: exact string, prefix with trailing '*' (e.g. "+4930*"), or regex ("/^\\+49/").
+   * Match the normalized called number.
+   *
+   * Inbound: matches the provider-delivered DID / Request-URI user part.
+   * Outbound: matches the normalized dialed digits.
+   * Supports: exact string, numeric range `start..end`, prefix with trailing '*' (e.g. "+4930*"), or regex ("/^\\+49/").
   */
  numberPattern?: string;

@@ -88,13 +110,13 @@ export interface ISipRouteAction {

  // --- Inbound actions (IVR / voicemail) ---

-  /** Route directly to a voicemail box (skip ringing devices). */
+  /** Voicemail fallback for matched inbound routes. */
  voicemailBox?: string;

  /** Route to an IVR menu by menu ID (skip ringing devices). */
  ivrMenuId?: string;

-  /** Override no-answer timeout (seconds) before routing to voicemail. */
+  /** Reserved for future no-answer handling. */
  noAnswerTimeout?: number;

  // --- Outbound actions (provider selection) ---
@@ -160,24 +182,13 @@ export interface IContact {
 // Voicebox configuration
 // ---------------------------------------------------------------------------

-export interface IVoiceboxConfig {
-  /** Unique ID — typically matches device ID or extension. */
-  id: string;
-  /** Whether this voicebox is active. */
-  enabled: boolean;
-  /** Custom TTS greeting text. */
-  greetingText?: string;
-  /** TTS voice ID (default 'af_bella'). */
-  greetingVoice?: string;
-  /** Path to uploaded WAV greeting (overrides TTS). */
-  greetingWavPath?: string;
-  /** Seconds to wait before routing to voicemail (default 25). */
-  noAnswerTimeoutSec?: number;
-  /** Maximum recording duration in seconds (default 120). */
-  maxRecordingSec?: number;
-  /** Maximum stored messages per box (default 50). */
-  maxMessages?: number;
-}
+// Canonical definition lives in voicebox.ts (imported at the top of this
+// file) — re-exported here so consumers can import everything from a
+// single config module without pulling in the voicebox implementation.
+// This used to be a duplicated interface and caused
+// "number | undefined is not assignable to number" type errors when
+// passing config.voiceboxes into VoiceboxManager.init().
+export type { IVoiceboxConfig };

 // ---------------------------------------------------------------------------
 // IVR configuration
@@ -241,6 +252,7 @@ export interface IAppConfig {
  proxy: IProxyConfig;
  providers: IProviderConfig[];
  devices: IDeviceConfig[];
+  incomingNumbers?: IIncomingNumberConfig[];
  routing: IRoutingConfig;
  contacts: IContact[];
  voiceboxes?: IVoiceboxConfig[];
@@ -295,6 +307,14 @@ export function loadConfig(): IAppConfig {
    d.extension ??= '100';
  }

+  cfg.incomingNumbers ??= [];
+  for (const incoming of cfg.incomingNumbers) {
+    if (!incoming.id) incoming.id = `incoming-${Date.now()}`;
+    incoming.label ??= incoming.id;
+    incoming.mode ??= incoming.pattern ? 'regex' : incoming.rangeStart || incoming.rangeEnd ? 'range' : 'single';
+    incoming.countryCode ??= incoming.mode === 'regex' ? undefined : '+49';
+  }
+
  cfg.routing ??= { routes: [] };
  cfg.routing.routes ??= [];

--- a/ts/frontend.ts
+++ b/ts/frontend.ts
@@ -14,12 +14,36 @@ import { WebSocketServer, WebSocket } from 'ws';
 import { handleWebRtcSignaling } from './webrtcbridge.ts';
 import type { VoiceboxManager } from './voicebox.ts';

-// CallManager was previously used for WebRTC call handling. Now replaced by Rust proxy-engine.
-// Kept as `any` type for backward compat with the function signature until full WebRTC port.
-type CallManager = any;
-
 const CONFIG_PATH = path.join(process.cwd(), '.nogit', 'config.json');

+interface IHandleRequestContext {
+  getStatus: () => unknown;
+  log: (msg: string) => void;
+  onStartCall: (number: string, deviceId?: string, providerId?: string) => { id: string } | null;
+  onHangupCall: (callId: string) => boolean;
+  onConfigSaved?: () => void | Promise<void>;
+  voiceboxManager?: VoiceboxManager;
+}
+
+interface IWebUiOptions extends IHandleRequestContext {
+  port: number;
+  onWebRtcOffer?: (sessionId: string, sdp: string, ws: WebSocket) => Promise<void>;
+  onWebRtcIce?: (sessionId: string, candidate: unknown) => Promise<void>;
+  onWebRtcClose?: (sessionId: string) => Promise<void>;
+  onWebRtcAccept?: (callId: string, sessionId: string) => void;
+}
+
+interface IWebRtcSocketMessage {
+  type?: string;
+  sessionId?: string;
+  callId?: string;
+  sdp?: string;
+  candidate?: unknown;
+  userAgent?: string;
+  _remoteIp?: string | null;
+  [key: string]: unknown;
+}
+
 // ---------------------------------------------------------------------------
 // WebSocket broadcast
 // ---------------------------------------------------------------------------
@@ -82,14 +106,9 @@ function loadStaticFiles(): void {
 async function handleRequest(
  req: http.IncomingMessage,
  res: http.ServerResponse,
-  getStatus: () => unknown,
-  log: (msg: string) => void,
-  onStartCall: (number: string, deviceId?: string, providerId?: string) => { id: string } | null,
-  onHangupCall: (callId: string) => boolean,
-  onConfigSaved?: () => void,
-  callManager?: CallManager,
-  voiceboxManager?: VoiceboxManager,
+  context: IHandleRequestContext,
 ): Promise<void> {
+  const { getStatus, log, onStartCall, onHangupCall, onConfigSaved, voiceboxManager } = context;
  const url = new URL(req.url || '/', `http://${req.headers.host || 'localhost'}`);
  const method = req.method || 'GET';

@@ -247,6 +266,7 @@ async function handleRequest(
          if (existing && ud.displayName !== undefined) existing.displayName = ud.displayName;
        }
      }
+      if (updates.incomingNumbers !== undefined) cfg.incomingNumbers = updates.incomingNumbers;
      if (updates.routing) {
        if (updates.routing.routes) {
          cfg.routing.routes = updates.routing.routes;
@@ -258,7 +278,7 @@ async function handleRequest(

      fs.writeFileSync(CONFIG_PATH, JSON.stringify(cfg, null, 2) + '\n');
      log('[config] updated config.json');
-      onConfigSaved?.();
+      await onConfigSaved?.();
      return sendJson(res, { ok: true });
    } catch (e: any) {
      return sendJson(res, { ok: false, error: e.message }, 400);
@@ -339,21 +359,21 @@ async function handleRequest(
 // ---------------------------------------------------------------------------

 export function initWebUi(
-  getStatus: () => unknown,
-  log: (msg: string) => void,
-  onStartCall: (number: string, deviceId?: string, providerId?: string) => { id: string } | null,
-  onHangupCall: (callId: string) => boolean,
-  onConfigSaved?: () => void,
-  callManager?: CallManager,
-  voiceboxManager?: VoiceboxManager,
-  /** WebRTC signaling handlers — forwarded to Rust proxy-engine. */
-  onWebRtcOffer?: (sessionId: string, sdp: string, ws: WebSocket) => Promise<void>,
-  onWebRtcIce?: (sessionId: string, candidate: any) => Promise<void>,
-  onWebRtcClose?: (sessionId: string) => Promise<void>,
-  /** Called when browser sends webrtc-accept (callId + sessionId linking). */
-  onWebRtcAccept?: (callId: string, sessionId: string) => void,
+  options: IWebUiOptions,
 ): void {
-  const WEB_PORT = 3060;
+  const {
+    port,
+    getStatus,
+    log,
+    onStartCall,
+    onHangupCall,
+    onConfigSaved,
+    voiceboxManager,
+    onWebRtcOffer,
+    onWebRtcIce,
+    onWebRtcClose,
+    onWebRtcAccept,
+  } = options;

  loadStaticFiles();

@@ -367,12 +387,12 @@ export function initWebUi(
    const cert = fs.readFileSync(certPath, 'utf8');
    const key = fs.readFileSync(keyPath, 'utf8');
    server = https.createServer({ cert, key }, (req, res) =>
-      handleRequest(req, res, getStatus, log, onStartCall, onHangupCall, onConfigSaved, callManager, voiceboxManager).catch(() => { res.writeHead(500); res.end(); }),
+      handleRequest(req, res, { getStatus, log, onStartCall, onHangupCall, onConfigSaved, voiceboxManager }).catch(() => { res.writeHead(500); res.end(); }),
    );
    useTls = true;
  } catch {
    server = http.createServer((req, res) =>
-      handleRequest(req, res, getStatus, log, onStartCall, onHangupCall, onConfigSaved, callManager, voiceboxManager).catch(() => { res.writeHead(500); res.end(); }),
+      handleRequest(req, res, { getStatus, log, onStartCall, onHangupCall, onConfigSaved, voiceboxManager }).catch(() => { res.writeHead(500); res.end(); }),
    );
  }

@@ -386,12 +406,12 @@ export function initWebUi(

    socket.on('message', (raw) => {
      try {
-        const msg = JSON.parse(raw.toString());
+        const msg = JSON.parse(raw.toString()) as IWebRtcSocketMessage;
        if (msg.type === 'webrtc-offer' && msg.sessionId) {
          // Forward to Rust proxy-engine for WebRTC handling.
-          if (onWebRtcOffer) {
+          if (onWebRtcOffer && typeof msg.sdp === 'string') {
            log(`[webrtc-ws] offer msg keys: ${Object.keys(msg).join(',')}, sdp type: ${typeof msg.sdp}, sdp len: ${msg.sdp?.length || 0}`);
-            onWebRtcOffer(msg.sessionId, msg.sdp, socket as any).catch((e: any) =>
+            onWebRtcOffer(msg.sessionId, msg.sdp, socket).catch((e: any) =>
              log(`[webrtc] offer error: ${e.message}`));
          }
        } else if (msg.type === 'webrtc-ice' && msg.sessionId) {
@@ -409,7 +429,7 @@ export function initWebUi(
          }
        } else if (msg.type?.startsWith('webrtc-')) {
          msg._remoteIp = remoteIp;
-          handleWebRtcSignaling(socket as any, msg);
+          handleWebRtcSignaling(socket, msg);
        }
      } catch { /* ignore */ }
    });
@@ -418,8 +438,8 @@ export function initWebUi(
    socket.on('error', () => wsClients.delete(socket));
  });

-  server.listen(WEB_PORT, '0.0.0.0', () => {
-    log(`web ui listening on ${useTls ? 'https' : 'http'}://0.0.0.0:${WEB_PORT}`);
+  server.listen(port, '0.0.0.0', () => {
+    log(`web ui listening on ${useTls ? 'https' : 'http'}://0.0.0.0:${port}`);
  });

  setInterval(() => broadcastWs('status', getStatus()), 1000);
--- a/ts/proxybridge.ts
+++ b/ts/proxybridge.ts
@@ -4,13 +4,36 @@
 * The proxy-engine handles ALL SIP protocol mechanics. TypeScript only:
 * - Sends configuration
 * - Receives high-level events (incoming_call, call_ended, etc.)
- * - Sends high-level commands (hangup, make_call, play_audio)
+ * - Sends high-level commands (hangup, make_call, add_leg, webrtc_offer)
 *
 * No raw SIP ever touches TypeScript.
 */

 import path from 'node:path';
 import { RustBridge } from '@push.rocks/smartrust';
+import type { TProxyEventMap } from './shared/proxy-events.ts';
+export type {
+  ICallAnsweredEvent,
+  ICallEndedEvent,
+  ICallRingingEvent,
+  IDeviceRegisteredEvent,
+  IIncomingCallEvent,
+  ILegAddedEvent,
+  ILegRemovedEvent,
+  ILegStateChangedEvent,
+  IOutboundCallEvent,
+  IOutboundCallStartedEvent,
+  IProviderRegisteredEvent,
+  IRecordingDoneEvent,
+  ISipUnhandledEvent,
+  IVoicemailErrorEvent,
+  IVoicemailStartedEvent,
+  IWebRtcAudioRxEvent,
+  IWebRtcIceCandidateEvent,
+  IWebRtcStateEvent,
+  IWebRtcTrackEvent,
+  TProxyEventMap,
+} from './shared/proxy-events.ts';

 // ---------------------------------------------------------------------------
 // Command type map for smartrust
@@ -29,18 +52,14 @@ type TProxyCommands = {
    params: { number: string; device_id?: string; provider_id?: string };
    result: { call_id: string };
  };
-  play_audio: {
-    params: { call_id: string; leg_id?: string; file_path: string; codec?: number };
-    result: Record<string, never>;
+  add_leg: {
+    params: { call_id: string; number: string; provider_id?: string };
+    result: { leg_id: string };
  };
-  start_recording: {
-    params: { call_id: string; file_path: string; max_duration_ms?: number };
+  remove_leg: {
+    params: { call_id: string; leg_id: string };
    result: Record<string, never>;
  };
-  stop_recording: {
-    params: { call_id: string };
-    result: { file_path: string; duration_ms: number };
-  };
  add_device_leg: {
    params: { call_id: string; device_id: string };
    result: { leg_id: string };
@@ -63,6 +82,19 @@ type TProxyCommands = {
    };
    result: { result: 'digit' | 'timeout' | 'cancelled'; digit?: string };
  };
+  start_tts_interaction: {
+    params: {
+      call_id: string;
+      leg_id: string;
+      text: string;
+      voice?: string;
+      model?: string;
+      voices?: string;
+      expected_digits: string;
+      timeout_ms: number;
+    };
+    result: { result: 'digit' | 'timeout' | 'cancelled'; digit?: string };
+  };
  add_tool_leg: {
    params: {
      call_id: string;
@@ -80,55 +112,39 @@ type TProxyCommands = {
    result: Record<string, never>;
  };
  generate_tts: {
-    params: { model: string; voices: string; voice: string; text: string; output: string };
+    params: { model: string; voices: string; voice: string; text: string; output: string; cacheable?: boolean };
    result: { output: string };
  };
+  // WebRTC signaling — bridged from the browser via the TS control plane.
+  webrtc_offer: {
+    params: { session_id: string; sdp: string };
+    result: { sdp: string };
+  };
+  webrtc_ice: {
+    params: {
+      session_id: string;
+      candidate: string;
+      sdp_mid?: string;
+      sdp_mline_index?: number;
+    };
+    result: Record<string, never>;
+  };
+  webrtc_link: {
+    params: {
+      session_id: string;
+      call_id: string;
+      provider_media_addr: string;
+      provider_media_port: number;
+      sip_pt?: number;
+    };
+    result: Record<string, never>;
+  };
+  webrtc_close: {
+    params: { session_id: string };
+    result: Record<string, never>;
+  };
 };

-// ---------------------------------------------------------------------------
-// Event types from Rust
-// ---------------------------------------------------------------------------
-
-export interface IIncomingCallEvent {
-  call_id: string;
-  from_uri: string;
-  to_number: string;
-  provider_id: string;
-  /** Whether registered browsers should see a `webrtc-incoming` toast for
-   *  this call. Set by the Rust engine from the matched inbound route's
-   *  `ringBrowsers` flag (defaults to `true` when no route matches, so
-   *  deployments without explicit routes preserve pre-routing behavior). */
-  ring_browsers?: boolean;
-}
-
-export interface IOutboundCallEvent {
-  call_id: string;
-  from_device: string | null;
-  to_number: string;
-}
-
-export interface ICallEndedEvent {
-  call_id: string;
-  reason: string;
-  duration: number;
-  from_side?: string;
-}
-
-export interface IProviderRegisteredEvent {
-  provider_id: string;
-  registered: boolean;
-  public_ip: string | null;
-}
-
-export interface IDeviceRegisteredEvent {
-  device_id: string;
-  display_name: string;
-  address: string;
-  port: number;
-  aor: string;
-  expires: number;
-}
-
 // ---------------------------------------------------------------------------
 // Bridge singleton
 // ---------------------------------------------------------------------------
@@ -137,6 +153,16 @@ let bridge: RustBridge<TProxyCommands> | null = null;
 let initialized = false;
 let logFn: ((msg: string) => void) | undefined;

+type TWebRtcIceCandidate = {
+  candidate?: string;
+  sdpMid?: string;
+  sdpMLineIndex?: number;
+} | string;
+
+function errorMessage(error: unknown): string {
+  return error instanceof Error ? error.message : String(error);
+}
+
 function buildLocalPaths(): string[] {
  const root = process.cwd();
  // Map Node's process.arch to tsrust's friendly target name.
@@ -195,8 +221,8 @@ export async function initProxyEngine(log?: (msg: string) => void): Promise<bool
    initialized = true;
    log?.('[proxy-engine] spawned and ready');
    return true;
-  } catch (e: any) {
-    log?.(`[proxy-engine] init error: ${e.message}`);
+  } catch (error: unknown) {
+    log?.(`[proxy-engine] init error: ${errorMessage(error)}`);
    bridge = null;
    return false;
  }
@@ -206,14 +232,14 @@ export async function initProxyEngine(log?: (msg: string) => void): Promise<bool
 * Send the full app config to the proxy engine.
 * This binds the SIP socket, starts provider registrations, etc.
 */
-export async function configureProxyEngine(config: Record<string, unknown>): Promise<boolean> {
+export async function configureProxyEngine(config: TProxyCommands['configure']['params']): Promise<boolean> {
  if (!bridge || !initialized) return false;
  try {
-    const result = await bridge.sendCommand('configure', config as any);
-    logFn?.(`[proxy-engine] configured, SIP bound on ${(result as any)?.bound || '?'}`);
+    const result = await sendProxyCommand('configure', config);
+    logFn?.(`[proxy-engine] configured, SIP bound on ${result.bound || '?'}`);
    return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] configure error: ${e.message}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] configure error: ${errorMessage(error)}`);
    return false;
  }
 }
@@ -224,14 +250,14 @@ export async function configureProxyEngine(config: Record<string, unknown>): Pro
 export async function makeCall(number: string, deviceId?: string, providerId?: string): Promise<string | null> {
  if (!bridge || !initialized) return null;
  try {
-    const result = await bridge.sendCommand('make_call', {
+    const result = await sendProxyCommand('make_call', {
      number,
      device_id: deviceId,
      provider_id: providerId,
-    } as any);
-    return (result as any)?.call_id || null;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] make_call error: ${e?.message || e}`);
+    });
+    return result.call_id || null;
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] make_call error: ${errorMessage(error)}`);
    return null;
  }
 }
@@ -242,7 +268,7 @@ export async function makeCall(number: string, deviceId?: string, providerId?: s
 export async function hangupCall(callId: string): Promise<boolean> {
  if (!bridge || !initialized) return false;
  try {
-    await bridge.sendCommand('hangup', { call_id: callId } as any);
+    await sendProxyCommand('hangup', { call_id: callId });
    return true;
  } catch {
    return false;
@@ -255,10 +281,9 @@ export async function hangupCall(callId: string): Promise<boolean> {
 export async function webrtcOffer(sessionId: string, sdp: string): Promise<{ sdp: string } | null> {
  if (!bridge || !initialized) return null;
  try {
-    const result = await bridge.sendCommand('webrtc_offer', { session_id: sessionId, sdp } as any);
-    return result as any;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] webrtc_offer error: ${e?.message || e}`);
+    return await sendProxyCommand('webrtc_offer', { session_id: sessionId, sdp });
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] webrtc_offer error: ${errorMessage(error)}`);
    return null;
  }
 }
@@ -266,15 +291,15 @@ export async function webrtcOffer(sessionId: string, sdp: string): Promise<{ sdp
 /**
 * Forward an ICE candidate to the proxy engine.
 */
-export async function webrtcIce(sessionId: string, candidate: any): Promise<void> {
+export async function webrtcIce(sessionId: string, candidate: TWebRtcIceCandidate): Promise<void> {
  if (!bridge || !initialized) return;
  try {
-    await bridge.sendCommand('webrtc_ice', {
+    await sendProxyCommand('webrtc_ice', {
      session_id: sessionId,
-      candidate: candidate?.candidate || candidate,
-      sdp_mid: candidate?.sdpMid,
-      sdp_mline_index: candidate?.sdpMLineIndex,
-    } as any);
+      candidate: typeof candidate === 'string' ? candidate : candidate.candidate || '',
+      sdp_mid: typeof candidate === 'string' ? undefined : candidate.sdpMid,
+      sdp_mline_index: typeof candidate === 'string' ? undefined : candidate.sdpMLineIndex,
+    });
  } catch { /* ignore */ }
 }

@@ -285,16 +310,16 @@ export async function webrtcIce(sessionId: string, candidate: any): Promise<void
 export async function webrtcLink(sessionId: string, callId: string, providerMediaAddr: string, providerMediaPort: number, sipPt: number = 9): Promise<boolean> {
  if (!bridge || !initialized) return false;
  try {
-    await bridge.sendCommand('webrtc_link', {
+    await sendProxyCommand('webrtc_link', {
      session_id: sessionId,
      call_id: callId,
      provider_media_addr: providerMediaAddr,
      provider_media_port: providerMediaPort,
      sip_pt: sipPt,
-    } as any);
+    });
    return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] webrtc_link error: ${e?.message || e}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] webrtc_link error: ${errorMessage(error)}`);
    return false;
  }
 }
@@ -305,14 +330,14 @@ export async function webrtcLink(sessionId: string, callId: string, providerMedi
 export async function addLeg(callId: string, number: string, providerId?: string): Promise<string | null> {
  if (!bridge || !initialized) return null;
  try {
-    const result = await bridge.sendCommand('add_leg', {
+    const result = await sendProxyCommand('add_leg', {
      call_id: callId,
      number,
      provider_id: providerId,
-    } as any);
-    return (result as any)?.leg_id || null;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] add_leg error: ${e?.message || e}`);
+    });
+    return result.leg_id || null;
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] add_leg error: ${errorMessage(error)}`);
    return null;
  }
 }
@@ -323,10 +348,10 @@ export async function addLeg(callId: string, number: string, providerId?: string
 export async function removeLeg(callId: string, legId: string): Promise<boolean> {
  if (!bridge || !initialized) return false;
  try {
-    await bridge.sendCommand('remove_leg', { call_id: callId, leg_id: legId } as any);
+    await sendProxyCommand('remove_leg', { call_id: callId, leg_id: legId });
    return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] remove_leg error: ${e?.message || e}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] remove_leg error: ${errorMessage(error)}`);
    return false;
  }
 }
@@ -337,7 +362,7 @@ export async function removeLeg(callId: string, legId: string): Promise<boolean>
 export async function webrtcClose(sessionId: string): Promise<void> {
  if (!bridge || !initialized) return;
  try {
-    await bridge.sendCommand('webrtc_close', { session_id: sessionId } as any);
+    await sendProxyCommand('webrtc_close', { session_id: sessionId });
  } catch { /* ignore */ }
 }

@@ -351,13 +376,13 @@ export async function webrtcClose(sessionId: string): Promise<void> {
 export async function addDeviceLeg(callId: string, deviceId: string): Promise<string | null> {
  if (!bridge || !initialized) return null;
  try {
-    const result = await bridge.sendCommand('add_device_leg', {
+    const result = await sendProxyCommand('add_device_leg', {
      call_id: callId,
      device_id: deviceId,
-    } as any);
-    return (result as any)?.leg_id || null;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] add_device_leg error: ${e?.message || e}`);
+    });
+    return result.leg_id || null;
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] add_device_leg error: ${errorMessage(error)}`);
    return null;
  }
 }
@@ -372,14 +397,14 @@ export async function transferLeg(
 ): Promise<boolean> {
  if (!bridge || !initialized) return false;
  try {
-    await bridge.sendCommand('transfer_leg', {
+    await sendProxyCommand('transfer_leg', {
      source_call_id: sourceCallId,
      leg_id: legId,
      target_call_id: targetCallId,
-    } as any);
+    });
    return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] transfer_leg error: ${e?.message || e}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] transfer_leg error: ${errorMessage(error)}`);
    return false;
  }
 }
@@ -395,15 +420,15 @@ export async function replaceLeg(
 ): Promise<string | null> {
  if (!bridge || !initialized) return null;
  try {
-    const result = await bridge.sendCommand('replace_leg', {
+    const result = await sendProxyCommand('replace_leg', {
      call_id: callId,
      old_leg_id: oldLegId,
      number,
      provider_id: providerId,
-    } as any);
-    return (result as any)?.new_leg_id || null;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] replace_leg error: ${e?.message || e}`);
+    });
+    return result.new_leg_id || null;
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] replace_leg error: ${errorMessage(error)}`);
    return null;
  }
 }
@@ -421,16 +446,49 @@ export async function startInteraction(
 ): Promise<{ result: 'digit' | 'timeout' | 'cancelled'; digit?: string } | null> {
  if (!bridge || !initialized) return null;
  try {
-    const result = await bridge.sendCommand('start_interaction', {
+    return await sendProxyCommand('start_interaction', {
      call_id: callId,
      leg_id: legId,
      prompt_wav: promptWav,
      expected_digits: expectedDigits,
      timeout_ms: timeoutMs,
-    } as any);
-    return result as any;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] start_interaction error: ${e?.message || e}`);
+    });
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] start_interaction error: ${errorMessage(error)}`);
+    return null;
+  }
+}
+
+/**
+ * Start a live TTS interaction on a specific leg. The first chunk is rendered
+ * up front and the rest streams into the mixer while playback is already live.
+ */
+export async function startTtsInteraction(
+  callId: string,
+  legId: string,
+  text: string,
+  expectedDigits: string,
+  timeoutMs: number,
+  options?: {
+    voice?: string;
+    model?: string;
+    voices?: string;
+  },
+): Promise<{ result: 'digit' | 'timeout' | 'cancelled'; digit?: string } | null> {
+  if (!bridge || !initialized) return null;
+  try {
+    return await sendProxyCommand('start_tts_interaction', {
+      call_id: callId,
+      leg_id: legId,
+      text,
+      expected_digits: expectedDigits,
+      timeout_ms: timeoutMs,
+      voice: options?.voice,
+      model: options?.model,
+      voices: options?.voices,
+    });
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] start_tts_interaction error: ${errorMessage(error)}`);
    return null;
  }
 }
@@ -446,14 +504,14 @@ export async function addToolLeg(
 ): Promise<string | null> {
  if (!bridge || !initialized) return null;
  try {
-    const result = await bridge.sendCommand('add_tool_leg', {
+    const result = await sendProxyCommand('add_tool_leg', {
      call_id: callId,
      tool_type: toolType,
      config,
-    } as any);
-    return (result as any)?.tool_leg_id || null;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] add_tool_leg error: ${e?.message || e}`);
+    });
+    return result.tool_leg_id || null;
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] add_tool_leg error: ${errorMessage(error)}`);
    return null;
  }
 }
@@ -464,13 +522,13 @@ export async function addToolLeg(
 export async function removeToolLeg(callId: string, toolLegId: string): Promise<boolean> {
  if (!bridge || !initialized) return false;
  try {
-    await bridge.sendCommand('remove_tool_leg', {
+    await sendProxyCommand('remove_tool_leg', {
      call_id: callId,
      tool_leg_id: toolLegId,
-    } as any);
+    });
    return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] remove_tool_leg error: ${e?.message || e}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] remove_tool_leg error: ${errorMessage(error)}`);
    return false;
  }
 }
@@ -486,15 +544,15 @@ export async function setLegMetadata(
 ): Promise<boolean> {
  if (!bridge || !initialized) return false;
  try {
-    await bridge.sendCommand('set_leg_metadata', {
+    await sendProxyCommand('set_leg_metadata', {
      call_id: callId,
      leg_id: legId,
      key,
      value,
-    } as any);
+    });
    return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] set_leg_metadata error: ${e?.message || e}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] set_leg_metadata error: ${errorMessage(error)}`);
    return false;
  }
 }
@@ -506,7 +564,7 @@ export async function setLegMetadata(
 * dtmf_digit, recording_done, tool_recording_done, tool_transcription_done,
 * leg_added, leg_removed, sip_unhandled
 */
-export function onProxyEvent(event: string, handler: (data: any) => void): void {
+export function onProxyEvent<K extends keyof TProxyEventMap>(event: K, handler: (data: TProxyEventMap[K]) => void): void {
  if (!bridge) throw new Error('proxy engine not initialized');
  bridge.on(`management:${event}`, handler);
 }
@@ -522,7 +580,7 @@ export async function sendProxyCommand<K extends keyof TProxyCommands>(
  params: TProxyCommands[K]['params'],
 ): Promise<TProxyCommands[K]['result']> {
  if (!bridge || !initialized) throw new Error('proxy engine not initialized');
-  return bridge.sendCommand(method as string, params as any) as any;
+  return bridge.sendCommand(method, params) as Promise<TProxyCommands[K]['result']>;
 }

 /** Shut down the proxy engine. */
--- a/ts/runtime/proxy-events.ts
+++ b/ts/runtime/proxy-events.ts
@@ -0,0 +1,187 @@
+import { onProxyEvent } from '../proxybridge.ts';
+import type { VoiceboxManager } from '../voicebox.ts';
+import type { StatusStore } from './status-store.ts';
+import type { IProviderMediaInfo, WebRtcLinkManager } from './webrtc-linking.ts';
+
+export interface IRegisterProxyEventHandlersOptions {
+  log: (msg: string) => void;
+  statusStore: StatusStore;
+  voiceboxManager: VoiceboxManager;
+  webRtcLinks: WebRtcLinkManager;
+  getBrowserDeviceIds: () => string[];
+  sendToBrowserDevice: (deviceId: string, data: unknown) => boolean;
+  broadcast: (type: string, data: unknown) => void;
+  onLinkWebRtcSession: (callId: string, sessionId: string, media: IProviderMediaInfo) => void;
+  onCloseWebRtcSession: (sessionId: string) => void;
+}
+
+export function registerProxyEventHandlers(options: IRegisterProxyEventHandlersOptions): void {
+  const {
+    log,
+    statusStore,
+    voiceboxManager,
+    webRtcLinks,
+    getBrowserDeviceIds,
+    sendToBrowserDevice,
+    broadcast,
+    onLinkWebRtcSession,
+    onCloseWebRtcSession,
+  } = options;
+
+  onProxyEvent('provider_registered', (data) => {
+    const previous = statusStore.noteProviderRegistered(data);
+    if (previous) {
+      if (data.registered && !previous.wasRegistered) {
+        log(`[provider:${data.provider_id}] registered (publicIp=${data.public_ip})`);
+      } else if (!data.registered && previous.wasRegistered) {
+        log(`[provider:${data.provider_id}] registration lost`);
+      }
+    }
+    broadcast('registration', { providerId: data.provider_id, registered: data.registered });
+  });
+
+  onProxyEvent('device_registered', (data) => {
+    if (statusStore.noteDeviceRegistered(data)) {
+      log(`[registrar] ${data.display_name} registered from ${data.address}:${data.port}`);
+    }
+  });
+
+  onProxyEvent('incoming_call', (data) => {
+    log(`[call] incoming: ${data.from_uri} -> ${data.to_number} via ${data.provider_id} (${data.call_id})`);
+    statusStore.noteIncomingCall(data);
+
+    if (data.ring_browsers === false) {
+      return;
+    }
+
+    for (const deviceId of getBrowserDeviceIds()) {
+      sendToBrowserDevice(deviceId, {
+        type: 'webrtc-incoming',
+        callId: data.call_id,
+        from: data.from_uri,
+        deviceId,
+      });
+    }
+  });
+
+  onProxyEvent('outbound_device_call', (data) => {
+    log(`[call] outbound: device ${data.from_device} -> ${data.to_number} (${data.call_id})`);
+    statusStore.noteOutboundDeviceCall(data);
+  });
+
+  onProxyEvent('outbound_call_started', (data) => {
+    log(`[call] outbound started: ${data.call_id} -> ${data.number} via ${data.provider_id}`);
+    statusStore.noteOutboundCallStarted(data);
+
+    for (const deviceId of getBrowserDeviceIds()) {
+      sendToBrowserDevice(deviceId, {
+        type: 'webrtc-incoming',
+        callId: data.call_id,
+        from: data.number,
+        deviceId,
+      });
+    }
+  });
+
+  onProxyEvent('call_ringing', (data) => {
+    statusStore.noteCallRinging(data);
+  });
+
+  onProxyEvent('call_answered', (data) => {
+    if (statusStore.noteCallAnswered(data)) {
+      log(`[call] ${data.call_id} connected`);
+    }
+
+    if (!data.provider_media_addr || !data.provider_media_port) {
+      return;
+    }
+
+    const target = webRtcLinks.noteCallAnswered(data.call_id, {
+      addr: data.provider_media_addr,
+      port: data.provider_media_port,
+      sipPt: data.sip_pt ?? 9,
+    });
+
+    if (!target) {
+      log(`[webrtc] media info cached for call=${data.call_id}, waiting for session accept`);
+      return;
+    }
+
+    onLinkWebRtcSession(data.call_id, target.sessionId, target.media);
+  });
+
+  onProxyEvent('call_ended', (data) => {
+    if (statusStore.noteCallEnded(data)) {
+      log(`[call] ${data.call_id} ended: ${data.reason} (${data.duration}s)`);
+    }
+
+    broadcast('webrtc-call-ended', { callId: data.call_id });
+
+    const sessionId = webRtcLinks.cleanupCall(data.call_id);
+    if (sessionId) {
+      onCloseWebRtcSession(sessionId);
+    }
+  });
+
+  onProxyEvent('sip_unhandled', (data) => {
+    log(`[sip] unhandled ${data.method_or_status} Call-ID=${data.call_id?.slice(0, 20)} from=${data.from_addr}:${data.from_port}`);
+  });
+
+  onProxyEvent('leg_added', (data) => {
+    log(`[leg] added: call=${data.call_id} leg=${data.leg_id} kind=${data.kind} state=${data.state}`);
+    statusStore.noteLegAdded(data);
+  });
+
+  onProxyEvent('leg_removed', (data) => {
+    log(`[leg] removed: call=${data.call_id} leg=${data.leg_id}`);
+    statusStore.noteLegRemoved(data);
+  });
+
+  onProxyEvent('leg_state_changed', (data) => {
+    log(`[leg] state: call=${data.call_id} leg=${data.leg_id} -> ${data.state}`);
+    statusStore.noteLegStateChanged(data);
+  });
+
+  onProxyEvent('webrtc_ice_candidate', (data) => {
+    broadcast('webrtc-ice', {
+      sessionId: data.session_id,
+      candidate: {
+        candidate: data.candidate,
+        sdpMid: data.sdp_mid,
+        sdpMLineIndex: data.sdp_mline_index,
+      },
+    });
+  });
+
+  onProxyEvent('webrtc_state', (data) => {
+    log(`[webrtc] session=${data.session_id?.slice(0, 8)} state=${data.state}`);
+  });
+
+  onProxyEvent('webrtc_track', (data) => {
+    log(`[webrtc] session=${data.session_id?.slice(0, 8)} track=${data.kind} codec=${data.codec}`);
+  });
+
+  onProxyEvent('webrtc_audio_rx', (data) => {
+    if (data.packet_count === 1 || data.packet_count === 50) {
+      log(`[webrtc] session=${data.session_id?.slice(0, 8)} browser audio rx #${data.packet_count}`);
+    }
+  });
+
+  onProxyEvent('voicemail_started', (data) => {
+    log(`[voicemail] started for call ${data.call_id} caller=${data.caller_number}`);
+  });
+
+  onProxyEvent('recording_done', (data) => {
+    log(`[voicemail] recording done: ${data.file_path} (${data.duration_ms}ms) caller=${data.caller_number}`);
+    voiceboxManager.addMessage('default', {
+      callerNumber: data.caller_number || 'Unknown',
+      callerName: null,
+      fileName: data.file_path,
+      durationMs: data.duration_ms,
+    });
+  });
+
+  onProxyEvent('voicemail_error', (data) => {
+    log(`[voicemail] error: ${data.error} call=${data.call_id}`);
+  });
+}
--- a/ts/runtime/status-store.ts
+++ b/ts/runtime/status-store.ts
@@ -0,0 +1,313 @@
+import type { IAppConfig } from '../config.ts';
+import type {
+  ICallAnsweredEvent,
+  ICallEndedEvent,
+  ICallRingingEvent,
+  IDeviceRegisteredEvent,
+  IIncomingCallEvent,
+  ILegAddedEvent,
+  ILegRemovedEvent,
+  ILegStateChangedEvent,
+  IOutboundCallEvent,
+  IOutboundCallStartedEvent,
+  IProviderRegisteredEvent,
+} from '../shared/proxy-events.ts';
+import type {
+  IActiveCall,
+  ICallHistoryEntry,
+  IDeviceStatus,
+  IProviderStatus,
+  IStatusSnapshot,
+  TLegType,
+} from '../shared/status.ts';
+
+const MAX_HISTORY = 100;
+const CODEC_NAMES: Record<number, string> = {
+  0: 'PCMU',
+  8: 'PCMA',
+  9: 'G.722',
+  111: 'Opus',
+};
+
+export class StatusStore {
+  private appConfig: IAppConfig;
+  private providerStatuses = new Map<string, IProviderStatus>();
+  private deviceStatuses = new Map<string, IDeviceStatus>();
+  private activeCalls = new Map<string, IActiveCall>();
+  private callHistory: ICallHistoryEntry[] = [];
+
+  constructor(appConfig: IAppConfig) {
+    this.appConfig = appConfig;
+    this.rebuildConfigState();
+  }
+
+  updateConfig(appConfig: IAppConfig): void {
+    this.appConfig = appConfig;
+    this.rebuildConfigState();
+  }
+
+  buildStatusSnapshot(
+    instanceId: string,
+    startTime: number,
+    browserDeviceIds: string[],
+    voicemailCounts: Record<string, number>,
+  ): IStatusSnapshot {
+    const devices = [...this.deviceStatuses.values()];
+    for (const deviceId of browserDeviceIds) {
+      devices.push({
+        id: deviceId,
+        displayName: 'Browser',
+        address: null,
+        port: 0,
+        aor: null,
+        connected: true,
+        isBrowser: true,
+      });
+    }
+
+    return {
+      instanceId,
+      uptime: Math.floor((Date.now() - startTime) / 1000),
+      lanIp: this.appConfig.proxy.lanIp,
+      providers: [...this.providerStatuses.values()],
+      devices,
+      calls: [...this.activeCalls.values()].map((call) => ({
+        ...call,
+        duration: Math.floor((Date.now() - call.startedAt) / 1000),
+        legs: [...call.legs.values()].map((leg) => ({
+          ...leg,
+          pktSent: 0,
+          pktReceived: 0,
+          transcoding: false,
+        })),
+      })),
+      callHistory: this.callHistory,
+      contacts: this.appConfig.contacts || [],
+      voicemailCounts,
+    };
+  }
+
+  noteDashboardCallStarted(callId: string, number: string, providerId?: string): void {
+    this.activeCalls.set(callId, {
+      id: callId,
+      direction: 'outbound',
+      callerNumber: null,
+      calleeNumber: number,
+      providerUsed: providerId || null,
+      state: 'setting-up',
+      startedAt: Date.now(),
+      legs: new Map(),
+    });
+  }
+
+  noteProviderRegistered(data: IProviderRegisteredEvent): { wasRegistered: boolean } | null {
+    const provider = this.providerStatuses.get(data.provider_id);
+    if (!provider) {
+      return null;
+    }
+
+    const wasRegistered = provider.registered;
+    provider.registered = data.registered;
+    provider.publicIp = data.public_ip;
+    return { wasRegistered };
+  }
+
+  noteDeviceRegistered(data: IDeviceRegisteredEvent): boolean {
+    const device = this.deviceStatuses.get(data.device_id);
+    if (!device) {
+      return false;
+    }
+
+    device.address = data.address;
+    device.port = data.port;
+    device.aor = data.aor;
+    device.connected = true;
+    return true;
+  }
+
+  noteIncomingCall(data: IIncomingCallEvent): void {
+    this.activeCalls.set(data.call_id, {
+      id: data.call_id,
+      direction: 'inbound',
+      callerNumber: data.from_uri,
+      calleeNumber: data.to_number,
+      providerUsed: data.provider_id,
+      state: 'ringing',
+      startedAt: Date.now(),
+      legs: new Map(),
+    });
+  }
+
+  noteOutboundDeviceCall(data: IOutboundCallEvent): void {
+    this.activeCalls.set(data.call_id, {
+      id: data.call_id,
+      direction: 'outbound',
+      callerNumber: data.from_device,
+      calleeNumber: data.to_number,
+      providerUsed: null,
+      state: 'setting-up',
+      startedAt: Date.now(),
+      legs: new Map(),
+    });
+  }
+
+  noteOutboundCallStarted(data: IOutboundCallStartedEvent): void {
+    this.activeCalls.set(data.call_id, {
+      id: data.call_id,
+      direction: 'outbound',
+      callerNumber: null,
+      calleeNumber: data.number,
+      providerUsed: data.provider_id,
+      state: 'setting-up',
+      startedAt: Date.now(),
+      legs: new Map(),
+    });
+  }
+
+  noteCallRinging(data: ICallRingingEvent): void {
+    const call = this.activeCalls.get(data.call_id);
+    if (call) {
+      call.state = 'ringing';
+    }
+  }
+
+  noteCallAnswered(data: ICallAnsweredEvent): boolean {
+    const call = this.activeCalls.get(data.call_id);
+    if (!call) {
+      return false;
+    }
+
+    call.state = 'connected';
+
+    if (data.provider_media_addr && data.provider_media_port) {
+      for (const leg of call.legs.values()) {
+        if (leg.type !== 'sip-provider') {
+          continue;
+        }
+
+        leg.remoteMedia = `${data.provider_media_addr}:${data.provider_media_port}`;
+        if (data.sip_pt !== undefined) {
+          leg.codec = CODEC_NAMES[data.sip_pt] || `PT${data.sip_pt}`;
+        }
+        break;
+      }
+    }
+
+    return true;
+  }
+
+  noteCallEnded(data: ICallEndedEvent): boolean {
+    const call = this.activeCalls.get(data.call_id);
+    if (!call) {
+      return false;
+    }
+
+    this.callHistory.unshift({
+      id: call.id,
+      direction: call.direction,
+      callerNumber: call.callerNumber,
+      calleeNumber: call.calleeNumber,
+      providerUsed: call.providerUsed,
+      startedAt: call.startedAt,
+      duration: data.duration,
+      legs: [...call.legs.values()].map((leg) => ({
+        id: leg.id,
+        type: leg.type,
+        metadata: leg.metadata || {},
+      })),
+    });
+
+    if (this.callHistory.length > MAX_HISTORY) {
+      this.callHistory.pop();
+    }
+
+    this.activeCalls.delete(data.call_id);
+    return true;
+  }
+
+  noteLegAdded(data: ILegAddedEvent): void {
+    const call = this.activeCalls.get(data.call_id);
+    if (!call) {
+      return;
+    }
+
+    call.legs.set(data.leg_id, {
+      id: data.leg_id,
+      type: data.kind,
+      state: data.state,
+      codec: data.codec ?? null,
+      rtpPort: data.rtpPort ?? null,
+      remoteMedia: data.remoteMedia ?? null,
+      metadata: data.metadata || {},
+    });
+  }
+
+  noteLegRemoved(data: ILegRemovedEvent): void {
+    this.activeCalls.get(data.call_id)?.legs.delete(data.leg_id);
+  }
+
+  noteLegStateChanged(data: ILegStateChangedEvent): void {
+    const call = this.activeCalls.get(data.call_id);
+    if (!call) {
+      return;
+    }
+
+    const existingLeg = call.legs.get(data.leg_id);
+    if (existingLeg) {
+      existingLeg.state = data.state;
+      if (data.metadata) {
+        existingLeg.metadata = data.metadata;
+      }
+      return;
+    }
+
+    call.legs.set(data.leg_id, {
+      id: data.leg_id,
+      type: this.inferLegType(data.leg_id),
+      state: data.state,
+      codec: null,
+      rtpPort: null,
+      remoteMedia: null,
+      metadata: data.metadata || {},
+    });
+  }
+
+  private rebuildConfigState(): void {
+    const nextProviderStatuses = new Map<string, IProviderStatus>();
+    for (const provider of this.appConfig.providers) {
+      const previous = this.providerStatuses.get(provider.id);
+      nextProviderStatuses.set(provider.id, {
+        id: provider.id,
+        displayName: provider.displayName,
+        registered: previous?.registered ?? false,
+        publicIp: previous?.publicIp ?? null,
+      });
+    }
+    this.providerStatuses = nextProviderStatuses;
+
+    const nextDeviceStatuses = new Map<string, IDeviceStatus>();
+    for (const device of this.appConfig.devices) {
+      const previous = this.deviceStatuses.get(device.id);
+      nextDeviceStatuses.set(device.id, {
+        id: device.id,
+        displayName: device.displayName,
+        address: previous?.address ?? null,
+        port: previous?.port ?? 0,
+        aor: previous?.aor ?? null,
+        connected: previous?.connected ?? false,
+        isBrowser: false,
+      });
+    }
+    this.deviceStatuses = nextDeviceStatuses;
+  }
+
+  private inferLegType(legId: string): TLegType {
+    if (legId.includes('-prov')) {
+      return 'sip-provider';
+    }
+    if (legId.includes('-dev')) {
+      return 'sip-device';
+    }
+    return 'webrtc';
+  }
+}
--- a/ts/runtime/webrtc-linking.ts
+++ b/ts/runtime/webrtc-linking.ts
@@ -0,0 +1,66 @@
+export interface IProviderMediaInfo {
+  addr: string;
+  port: number;
+  sipPt: number;
+}
+
+export interface IWebRtcLinkTarget {
+  sessionId: string;
+  media: IProviderMediaInfo;
+}
+
+export class WebRtcLinkManager {
+  private sessionToCall = new Map<string, string>();
+  private callToSession = new Map<string, string>();
+  private pendingCallMedia = new Map<string, IProviderMediaInfo>();
+
+  acceptCall(callId: string, sessionId: string): IProviderMediaInfo | null {
+    const previousCallId = this.sessionToCall.get(sessionId);
+    if (previousCallId && previousCallId !== callId) {
+      this.callToSession.delete(previousCallId);
+    }
+
+    const previousSessionId = this.callToSession.get(callId);
+    if (previousSessionId && previousSessionId !== sessionId) {
+      this.sessionToCall.delete(previousSessionId);
+    }
+
+    this.sessionToCall.set(sessionId, callId);
+    this.callToSession.set(callId, sessionId);
+
+    const pendingMedia = this.pendingCallMedia.get(callId) ?? null;
+    if (pendingMedia) {
+      this.pendingCallMedia.delete(callId);
+    }
+    return pendingMedia;
+  }
+
+  noteCallAnswered(callId: string, media: IProviderMediaInfo): IWebRtcLinkTarget | null {
+    const sessionId = this.callToSession.get(callId);
+    if (!sessionId) {
+      this.pendingCallMedia.set(callId, media);
+      return null;
+    }
+
+    return { sessionId, media };
+  }
+
+  removeSession(sessionId: string): string | null {
+    const callId = this.sessionToCall.get(sessionId) ?? null;
+    this.sessionToCall.delete(sessionId);
+    if (callId) {
+      this.callToSession.delete(callId);
+    }
+    return callId;
+  }
+
+  cleanupCall(callId: string): string | null {
+    const sessionId = this.callToSession.get(callId) ?? null;
+    this.callToSession.delete(callId);
+    this.pendingCallMedia.delete(callId);
+    if (sessionId) {
+      this.sessionToCall.delete(sessionId);
+    }
+    return sessionId;
+  }
+}
--- a/ts/shared/proxy-events.ts
+++ b/ts/shared/proxy-events.ts
@@ -0,0 +1,145 @@
+import type { TLegType } from './status.ts';
+
+export interface IIncomingCallEvent {
+  call_id: string;
+  from_uri: string;
+  to_number: string;
+  provider_id: string;
+  ring_browsers?: boolean;
+}
+
+export interface IOutboundCallEvent {
+  call_id: string;
+  from_device: string | null;
+  to_number: string;
+}
+
+export interface IOutboundCallStartedEvent {
+  call_id: string;
+  number: string;
+  provider_id: string;
+}
+
+export interface ICallRingingEvent {
+  call_id: string;
+}
+
+export interface ICallAnsweredEvent {
+  call_id: string;
+  provider_media_addr?: string;
+  provider_media_port?: number;
+  sip_pt?: number;
+}
+
+export interface ICallEndedEvent {
+  call_id: string;
+  reason: string;
+  duration: number;
+  from_side?: string;
+}
+
+export interface IProviderRegisteredEvent {
+  provider_id: string;
+  registered: boolean;
+  public_ip: string | null;
+}
+
+export interface IDeviceRegisteredEvent {
+  device_id: string;
+  display_name: string;
+  address: string;
+  port: number;
+  aor: string;
+  expires: number;
+}
+
+export interface ISipUnhandledEvent {
+  method_or_status: string;
+  call_id?: string;
+  from_addr: string;
+  from_port: number;
+}
+
+export interface ILegAddedEvent {
+  call_id: string;
+  leg_id: string;
+  kind: TLegType;
+  state: string;
+  codec?: string | null;
+  rtpPort?: number | null;
+  remoteMedia?: string | null;
+  metadata?: Record<string, unknown>;
+}
+
+export interface ILegRemovedEvent {
+  call_id: string;
+  leg_id: string;
+}
+
+export interface ILegStateChangedEvent {
+  call_id: string;
+  leg_id: string;
+  state: string;
+  metadata?: Record<string, unknown>;
+}
+
+export interface IWebRtcIceCandidateEvent {
+  session_id: string;
+  candidate: string;
+  sdp_mid?: string;
+  sdp_mline_index?: number;
+}
+
+export interface IWebRtcStateEvent {
+  session_id?: string;
+  state: string;
+}
+
+export interface IWebRtcTrackEvent {
+  session_id?: string;
+  kind: string;
+  codec: string;
+}
+
+export interface IWebRtcAudioRxEvent {
+  session_id?: string;
+  packet_count: number;
+}
+
+export interface IVoicemailStartedEvent {
+  call_id: string;
+  caller_number?: string;
+}
+
+export interface IRecordingDoneEvent {
+  file_path: string;
+  duration_ms: number;
+  caller_number?: string;
+}
+
+export interface IVoicemailErrorEvent {
+  call_id: string;
+  error: string;
+}
+
+export type TProxyEventMap = {
+  provider_registered: IProviderRegisteredEvent;
+  device_registered: IDeviceRegisteredEvent;
+  incoming_call: IIncomingCallEvent;
+  outbound_device_call: IOutboundCallEvent;
+  outbound_call_started: IOutboundCallStartedEvent;
+  call_ringing: ICallRingingEvent;
+  call_answered: ICallAnsweredEvent;
+  call_ended: ICallEndedEvent;
+  sip_unhandled: ISipUnhandledEvent;
+  leg_added: ILegAddedEvent;
+  leg_removed: ILegRemovedEvent;
+  leg_state_changed: ILegStateChangedEvent;
+  webrtc_ice_candidate: IWebRtcIceCandidateEvent;
+  webrtc_state: IWebRtcStateEvent;
+  webrtc_track: IWebRtcTrackEvent;
+  webrtc_audio_rx: IWebRtcAudioRxEvent;
+  voicemail_started: IVoicemailStartedEvent;
+  recording_done: IRecordingDoneEvent;
+  voicemail_error: IVoicemailErrorEvent;
+};
--- a/ts/shared/status.ts
+++ b/ts/shared/status.ts
@@ -0,0 +1,89 @@
+import type { IContact } from '../config.ts';
+
+export type TLegType = 'sip-device' | 'sip-provider' | 'webrtc' | 'tool';
+export type TCallDirection = 'inbound' | 'outbound';
+
+export interface IProviderStatus {
+  id: string;
+  displayName: string;
+  registered: boolean;
+  publicIp: string | null;
+}
+
+export interface IDeviceStatus {
+  id: string;
+  displayName: string;
+  address: string | null;
+  port: number;
+  aor: string | null;
+  connected: boolean;
+  isBrowser: boolean;
+}
+
+export interface IActiveLeg {
+  id: string;
+  type: TLegType;
+  state: string;
+  codec: string | null;
+  rtpPort: number | null;
+  remoteMedia: string | null;
+  metadata: Record<string, unknown>;
+}
+
+export interface IActiveCall {
+  id: string;
+  direction: TCallDirection;
+  callerNumber: string | null;
+  calleeNumber: string | null;
+  providerUsed: string | null;
+  state: string;
+  startedAt: number;
+  legs: Map<string, IActiveLeg>;
+}
+
+export interface IHistoryLeg {
+  id: string;
+  type: string;
+  metadata: Record<string, unknown>;
+}
+
+export interface ICallHistoryEntry {
+  id: string;
+  direction: TCallDirection;
+  callerNumber: string | null;
+  calleeNumber: string | null;
+  providerUsed: string | null;
+  startedAt: number;
+  duration: number;
+  legs: IHistoryLeg[];
+}
+
+export interface ILegStatus extends IActiveLeg {
+  pktSent: number;
+  pktReceived: number;
+  transcoding: boolean;
+}
+
+export interface ICallStatus {
+  id: string;
+  direction: TCallDirection;
+  callerNumber: string | null;
+  calleeNumber: string | null;
+  providerUsed: string | null;
+  state: string;
+  startedAt: number;
+  duration: number;
+  legs: ILegStatus[];
+}
+
+export interface IStatusSnapshot {
+  instanceId: string;
+  uptime: number;
+  lanIp: string;
+  providers: IProviderStatus[];
+  devices: IDeviceStatus[];
+  calls: ICallStatus[];
+  callHistory: ICallHistoryEntry[];
+  contacts: IContact[];
+  voicemailCounts: Record<string, number>;
+}
--- a/ts/sipproxy.ts
+++ b/ts/sipproxy.ts
@@ -1,36 +1,20 @@
 /**
- * SIP proxy — entry point.
+ * SIP proxy bootstrap.
 *
- * Spawns the Rust proxy-engine which handles ALL SIP protocol mechanics.
- * TypeScript is the control plane:
- * - Loads config and pushes it to Rust
- * - Receives high-level events (incoming calls, registration, etc.)
- * - Drives the web dashboard
- * - Manages IVR, voicemail, announcements
- * - Handles WebRTC browser signaling (forwarded to Rust in Phase 2)
- *
- * No raw SIP ever touches TypeScript.
+ * Spawns the Rust proxy-engine, wires runtime state/event handling,
+ * and starts the web dashboard plus browser signaling layer.
 */

 import fs from 'node:fs';
 import path from 'node:path';

-import { loadConfig } from './config.ts';
-import type { IAppConfig } from './config.ts';
+import { loadConfig, type IAppConfig } from './config.ts';
 import { broadcastWs, initWebUi } from './frontend.ts';
-import {
-  initWebRtcSignaling,
-  sendToBrowserDevice,
-  getAllBrowserDeviceIds,
-  getBrowserDeviceWs,
-} from './webrtcbridge.ts';
-import { initAnnouncement } from './announcement.ts';
-import { PromptCache } from './call/prompt-cache.ts';
+import { initWebRtcSignaling, getAllBrowserDeviceIds, sendToBrowserDevice } from './webrtcbridge.ts';
 import { VoiceboxManager } from './voicebox.ts';
 import {
  initProxyEngine,
  configureProxyEngine,
-  onProxyEvent,
  hangupCall,
  makeCall,
  shutdownProxyEngine,
@@ -38,648 +22,200 @@ import {
  webrtcIce,
  webrtcLink,
  webrtcClose,
-  addLeg,
-  removeLeg,
 } from './proxybridge.ts';
-import type {
-  IIncomingCallEvent,
-  IOutboundCallEvent,
-  ICallEndedEvent,
-  IProviderRegisteredEvent,
-  IDeviceRegisteredEvent,
-} from './proxybridge.ts';
-
-// ---------------------------------------------------------------------------
-// Config
-// ---------------------------------------------------------------------------
+import { registerProxyEventHandlers } from './runtime/proxy-events.ts';
+import { StatusStore } from './runtime/status-store.ts';
+import { WebRtcLinkManager, type IProviderMediaInfo } from './runtime/webrtc-linking.ts';

 let appConfig: IAppConfig = loadConfig();

 const LOG_PATH = path.join(process.cwd(), 'sip_trace.log');
-
-// ---------------------------------------------------------------------------
-// Logging
-// ---------------------------------------------------------------------------
-
 const startTime = Date.now();
 const instanceId = `${Date.now()}-${Math.random().toString(36).slice(2, 8)}`;

+const statusStore = new StatusStore(appConfig);
+const webRtcLinks = new WebRtcLinkManager();
+const voiceboxManager = new VoiceboxManager(log);
+
+voiceboxManager.init(appConfig.voiceboxes ?? []);
+initWebRtcSignaling({ log });
+
 function now(): string {
  return new Date().toISOString().replace('T', ' ').slice(0, 19);
 }

-function log(msg: string): void {
-  const line = `${now()} ${msg}\n`;
+function log(message: string): void {
+  const line = `${now()} ${message}\n`;
  fs.appendFileSync(LOG_PATH, line);
  process.stdout.write(line);
-  broadcastWs('log', { message: msg });
+  broadcastWs('log', { message });
 }

-// ---------------------------------------------------------------------------
-// Shadow state — maintained from Rust events for the dashboard
-// ---------------------------------------------------------------------------
-
-interface IProviderStatus {
-  id: string;
-  displayName: string;
-  registered: boolean;
-  publicIp: string | null;
+function errorMessage(error: unknown): string {
+  return error instanceof Error ? error.message : String(error);
 }

-interface IDeviceStatus {
-  id: string;
-  displayName: string;
-  address: string | null;
-  port: number;
-  connected: boolean;
-  isBrowser: boolean;
-}
-
-interface IActiveLeg {
-  id: string;
-  type: 'sip-device' | 'sip-provider' | 'webrtc' | 'tool';
-  state: string;
-  codec: string | null;
-  rtpPort: number | null;
-  remoteMedia: string | null;
-  metadata: Record<string, unknown>;
-}
-
-interface IActiveCall {
-  id: string;
-  direction: string;
-  callerNumber: string | null;
-  calleeNumber: string | null;
-  providerUsed: string | null;
-  state: string;
-  startedAt: number;
-  legs: Map<string, IActiveLeg>;
-}
-
-interface IHistoryLeg {
-  id: string;
-  type: string;
-  metadata: Record<string, unknown>;
-}
-
-interface ICallHistoryEntry {
-  id: string;
-  direction: string;
-  callerNumber: string | null;
-  calleeNumber: string | null;
-  startedAt: number;
-  duration: number;
-  legs: IHistoryLeg[];
-}
-
-const providerStatuses = new Map<string, IProviderStatus>();
-const deviceStatuses = new Map<string, IDeviceStatus>();
-const activeCalls = new Map<string, IActiveCall>();
-const callHistory: ICallHistoryEntry[] = [];
-const MAX_HISTORY = 100;
-
-// WebRTC session ↔ call linking state.
-// Both pieces (session accept + call media info) can arrive in any order.
-const webrtcSessionToCall = new Map<string, string>(); // sessionId → callId
-const webrtcCallToSession = new Map<string, string>(); // callId → sessionId
-const pendingCallMedia = new Map<string, { addr: string; port: number; sipPt: number }>(); // callId → provider media info
-
-// Initialize provider statuses from config (all start as unregistered).
-for (const p of appConfig.providers) {
-  providerStatuses.set(p.id, {
-    id: p.id,
-    displayName: p.displayName,
-    registered: false,
-    publicIp: null,
-  });
-}
-
-// Initialize device statuses from config.
-for (const d of appConfig.devices) {
-  deviceStatuses.set(d.id, {
-    id: d.id,
-    displayName: d.displayName,
-    address: null,
-    port: 0,
-    connected: false,
-    isBrowser: false,
-  });
-}
-
-// ---------------------------------------------------------------------------
-// Initialize subsystems
-// ---------------------------------------------------------------------------
-
-const promptCache = new PromptCache(log);
-const voiceboxManager = new VoiceboxManager(log);
-voiceboxManager.init(appConfig.voiceboxes ?? []);
-
-// WebRTC signaling (browser device registration).
-initWebRtcSignaling({ log });
-
-// ---------------------------------------------------------------------------
-// Status snapshot (fed to web dashboard)
-// ---------------------------------------------------------------------------
-
-function getStatus() {
-  // Merge SIP devices (from Rust) + browser devices (from TS WebSocket).
-  const devices = [...deviceStatuses.values()];
-  for (const bid of getAllBrowserDeviceIds()) {
-    devices.push({
-      id: bid,
-      displayName: 'Browser',
-      address: null,
-      port: 0,
-      connected: true,
-      isBrowser: true,
-    });
-  }
-
+function buildProxyConfig(config: IAppConfig): Record<string, unknown> {
  return {
-    instanceId,
-    uptime: Math.floor((Date.now() - startTime) / 1000),
-    lanIp: appConfig.proxy.lanIp,
-    providers: [...providerStatuses.values()],
-    devices,
-    calls: [...activeCalls.values()].map((c) => ({
-      ...c,
-      duration: Math.floor((Date.now() - c.startedAt) / 1000),
-      legs: [...c.legs.values()].map((l) => ({
-        id: l.id,
-        type: l.type,
-        state: l.state,
-        codec: l.codec,
-        rtpPort: l.rtpPort,
-        remoteMedia: l.remoteMedia,
-        metadata: l.metadata || {},
-        pktSent: 0,
-        pktReceived: 0,
-        transcoding: false,
-      })),
-    })),
-    callHistory,
-    contacts: appConfig.contacts || [],
-    voicemailCounts: voiceboxManager.getAllUnheardCounts(),
+    proxy: config.proxy,
+    providers: config.providers,
+    devices: config.devices,
+    routing: config.routing,
+    voiceboxes: config.voiceboxes ?? [],
+    ivr: config.ivr,
  };
 }

-// ---------------------------------------------------------------------------
-// Start Rust proxy engine
-// ---------------------------------------------------------------------------
+function getStatus() {
+  return statusStore.buildStatusSnapshot(
+    instanceId,
+    startTime,
+    getAllBrowserDeviceIds(),
+    voiceboxManager.getAllUnheardCounts(),
+  );
+}
+
+function requestWebRtcLink(callId: string, sessionId: string, media: IProviderMediaInfo): void {
+  log(`[webrtc] linking session=${sessionId.slice(0, 8)} to call=${callId} media=${media.addr}:${media.port} pt=${media.sipPt}`);
+  void webrtcLink(sessionId, callId, media.addr, media.port, media.sipPt).then((ok) => {
+    log(`[webrtc] link result: ${ok}`);
+  });
+}
+
+async function configureRuntime(config: IAppConfig): Promise<boolean> {
+  return configureProxyEngine(buildProxyConfig(config));
+}
+
+async function reloadConfig(): Promise<void> {
+  try {
+    const previousConfig = appConfig;
+    const nextConfig = loadConfig();
+
+    appConfig = nextConfig;
+    statusStore.updateConfig(nextConfig);
+    voiceboxManager.init(nextConfig.voiceboxes ?? []);
+
+    if (nextConfig.proxy.lanPort !== previousConfig.proxy.lanPort) {
+      log('[config] proxy.lanPort changed; restart required for SIP socket rebinding');
+    }
+    if (nextConfig.proxy.webUiPort !== previousConfig.proxy.webUiPort) {
+      log('[config] proxy.webUiPort changed; restart required for web UI rebinding');
+    }
+
+    const configured = await configureRuntime(nextConfig);
+    if (configured) {
+      log('[config] reloaded - proxy engine reconfigured');
+    } else {
+      log('[config] reload failed - proxy engine rejected config');
+    }
+  } catch (error: unknown) {
+    log(`[config] reload failed: ${errorMessage(error)}`);
+  }
+}

 async function startProxyEngine(): Promise<void> {
-  const ok = await initProxyEngine(log);
-  if (!ok) {
+  const started = await initProxyEngine(log);
+  if (!started) {
    log('[FATAL] failed to start proxy engine');
    process.exit(1);
  }

-  // Subscribe to events from Rust BEFORE sending configure.
-  onProxyEvent('provider_registered', (data: IProviderRegisteredEvent) => {
-    const ps = providerStatuses.get(data.provider_id);
-    if (ps) {
-      const wasRegistered = ps.registered;
-      ps.registered = data.registered;
-      ps.publicIp = data.public_ip;
-      if (data.registered && !wasRegistered) {
-        log(`[provider:${data.provider_id}] registered (publicIp=${data.public_ip})`);
-      } else if (!data.registered && wasRegistered) {
-        log(`[provider:${data.provider_id}] registration lost`);
-      }
-      broadcastWs('registration', { providerId: data.provider_id, registered: data.registered });
-    }
-  });
-
-  onProxyEvent('device_registered', (data: IDeviceRegisteredEvent) => {
-    const ds = deviceStatuses.get(data.device_id);
-    if (ds) {
-      ds.address = data.address;
-      ds.port = data.port;
-      ds.connected = true;
-      log(`[registrar] ${data.display_name} registered from ${data.address}:${data.port}`);
-    }
-  });
-
-  onProxyEvent('incoming_call', (data: IIncomingCallEvent) => {
-    log(`[call] incoming: ${data.from_uri} → ${data.to_number} via ${data.provider_id} (${data.call_id})`);
-    activeCalls.set(data.call_id, {
-      id: data.call_id,
-      direction: 'inbound',
-      callerNumber: data.from_uri,
-      calleeNumber: data.to_number,
-      providerUsed: data.provider_id,
-      state: 'ringing',
-      startedAt: Date.now(),
-      legs: new Map(),
-    });
-
-    // Notify browsers of the incoming call, but only if the matched inbound
-    // route asked for it. `ring_browsers !== false` preserves today's
-    // ring-by-default behavior for any Rust release that predates this
-    // field or for the fallback "no route matched" case (where Rust still
-    // sends `true`). Note: this is an informational toast — browsers do
-    // NOT race the SIP device to answer. First-to-answer-wins requires
-    // a multi-leg fork which is not yet implemented.
-    if (data.ring_browsers !== false) {
-      const browserIds = getAllBrowserDeviceIds();
-      for (const bid of browserIds) {
-        sendToBrowserDevice(bid, {
-          type: 'webrtc-incoming',
-          callId: data.call_id,
-          from: data.from_uri,
-          deviceId: bid,
-        });
-      }
-    }
-  });
-
-  onProxyEvent('outbound_device_call', (data: IOutboundCallEvent) => {
-    log(`[call] outbound: device ${data.from_device} → ${data.to_number} (${data.call_id})`);
-    activeCalls.set(data.call_id, {
-      id: data.call_id,
-      direction: 'outbound',
-      callerNumber: data.from_device,
-      calleeNumber: data.to_number,
-      providerUsed: null,
-      state: 'setting-up',
-      startedAt: Date.now(),
-      legs: new Map(),
-    });
-  });
-
-  onProxyEvent('outbound_call_started', (data: any) => {
-    log(`[call] outbound started: ${data.call_id} → ${data.number} via ${data.provider_id}`);
-    activeCalls.set(data.call_id, {
-      id: data.call_id,
-      direction: 'outbound',
-      callerNumber: null,
-      calleeNumber: data.number,
-      providerUsed: data.provider_id,
-      state: 'setting-up',
-      startedAt: Date.now(),
-      legs: new Map(),
-    });
-
-    // Notify all browser devices — they can connect via WebRTC to listen/talk.
-    const browserIds = getAllBrowserDeviceIds();
-    for (const bid of browserIds) {
-      sendToBrowserDevice(bid, {
-        type: 'webrtc-incoming',
-        callId: data.call_id,
-        from: data.number,
-        deviceId: bid,
-      });
-    }
-  });
-
-  onProxyEvent('call_ringing', (data: { call_id: string }) => {
-    const call = activeCalls.get(data.call_id);
-    if (call) call.state = 'ringing';
-  });
-
-  onProxyEvent('call_answered', (data: { call_id: string; provider_media_addr?: string; provider_media_port?: number; sip_pt?: number }) => {
-    const call = activeCalls.get(data.call_id);
-    if (call) {
-      call.state = 'connected';
-      log(`[call] ${data.call_id} connected`);
-
-      // Enrich provider leg with media info from the answered event.
-      if (data.provider_media_addr && data.provider_media_port) {
-        for (const leg of call.legs.values()) {
-          if (leg.type === 'sip-provider') {
-            leg.remoteMedia = `${data.provider_media_addr}:${data.provider_media_port}`;
-            if (data.sip_pt !== undefined) {
-              const codecNames: Record<number, string> = { 0: 'PCMU', 8: 'PCMA', 9: 'G.722', 111: 'Opus' };
-              leg.codec = codecNames[data.sip_pt] || `PT${data.sip_pt}`;
-            }
-            break;
-          }
-        }
-      }
-    }
-
-    // Try to link WebRTC session to this call for audio bridging.
-    if (data.provider_media_addr && data.provider_media_port) {
-      const sessionId = webrtcCallToSession.get(data.call_id);
-      if (sessionId) {
-        // Both session and media info available — link now.
-        const sipPt = data.sip_pt ?? 9;
-        log(`[webrtc] linking session=${sessionId.slice(0, 8)} to call=${data.call_id} media=${data.provider_media_addr}:${data.provider_media_port} pt=${sipPt}`);
-        webrtcLink(sessionId, data.call_id, data.provider_media_addr, data.provider_media_port, sipPt).then((ok) => {
-          log(`[webrtc] link result: ${ok}`);
-        });
-      } else {
-        // Session not yet accepted — store media info for when it arrives.
-        pendingCallMedia.set(data.call_id, {
-          addr: data.provider_media_addr,
-          port: data.provider_media_port,
-          sipPt: data.sip_pt ?? 9,
-        });
-        log(`[webrtc] media info cached for call=${data.call_id}, waiting for session accept`);
-      }
-    }
-  });
-
-  onProxyEvent('call_ended', (data: ICallEndedEvent) => {
-    const call = activeCalls.get(data.call_id);
-    if (call) {
-      log(`[call] ${data.call_id} ended: ${data.reason} (${data.duration}s)`);
-      // Snapshot legs with metadata for history.
-      const historyLegs: IHistoryLeg[] = [];
-      for (const [, leg] of call.legs) {
-        historyLegs.push({
-          id: leg.id,
-          type: leg.type,
-          metadata: leg.metadata || {},
-        });
-      }
-      // Move to history.
-      callHistory.unshift({
-        id: call.id,
-        direction: call.direction,
-        callerNumber: call.callerNumber,
-        calleeNumber: call.calleeNumber,
-        startedAt: call.startedAt,
-        duration: data.duration,
-        legs: historyLegs,
-      });
-      if (callHistory.length > MAX_HISTORY) callHistory.pop();
-      activeCalls.delete(data.call_id);
-
-      // Notify browser(s) that the call ended.
-      broadcastWs('webrtc-call-ended', { callId: data.call_id });
-
-      // Clean up WebRTC session mappings.
-      const sessionId = webrtcCallToSession.get(data.call_id);
-      if (sessionId) {
-        webrtcCallToSession.delete(data.call_id);
-        webrtcSessionToCall.delete(sessionId);
-        webrtcClose(sessionId).catch(() => {});
-      }
-      pendingCallMedia.delete(data.call_id);
-    }
-  });
-
-  onProxyEvent('sip_unhandled', (data: any) => {
-    log(`[sip] unhandled ${data.method_or_status} Call-ID=${data.call_id?.slice(0, 20)} from=${data.from_addr}:${data.from_port}`);
-  });
-
-  // Leg events (multiparty) — update shadow state so the dashboard shows legs.
-  onProxyEvent('leg_added', (data: any) => {
-    log(`[leg] added: call=${data.call_id} leg=${data.leg_id} kind=${data.kind} state=${data.state}`);
-    const call = activeCalls.get(data.call_id);
-    if (call) {
-      call.legs.set(data.leg_id, {
-        id: data.leg_id,
-        type: data.kind,
-        state: data.state,
-        codec: data.codec ?? null,
-        rtpPort: data.rtpPort ?? null,
-        remoteMedia: data.remoteMedia ?? null,
-        metadata: data.metadata || {},
-      });
-    }
-  });
-
-  onProxyEvent('leg_removed', (data: any) => {
-    log(`[leg] removed: call=${data.call_id} leg=${data.leg_id}`);
-    activeCalls.get(data.call_id)?.legs.delete(data.leg_id);
-  });
-
-  onProxyEvent('leg_state_changed', (data: any) => {
-    log(`[leg] state: call=${data.call_id} leg=${data.leg_id} → ${data.state}`);
-    const call = activeCalls.get(data.call_id);
-    if (!call) return;
-    const leg = call.legs.get(data.leg_id);
-    if (leg) {
-      leg.state = data.state;
-      if (data.metadata) leg.metadata = data.metadata;
-    } else {
-      // Initial legs (provider/device) don't emit leg_added — create on first state change.
-      const legId: string = data.leg_id;
-      const type = legId.includes('-prov') ? 'sip-provider' : legId.includes('-dev') ? 'sip-device' : 'webrtc';
-      call.legs.set(data.leg_id, {
-        id: data.leg_id,
-        type,
-        state: data.state,
-        codec: null,
-        rtpPort: null,
-        remoteMedia: null,
-        metadata: data.metadata || {},
-      });
-    }
-  });
-
-  // WebRTC events from Rust — forward ICE candidates to browser via WebSocket.
-  onProxyEvent('webrtc_ice_candidate', (data: any) => {
-    // Find the browser's WebSocket by session ID and send the ICE candidate.
-    broadcastWs('webrtc-ice', {
-      sessionId: data.session_id,
-      candidate: { candidate: data.candidate, sdpMid: data.sdp_mid, sdpMLineIndex: data.sdp_mline_index },
-    });
-  });
-
-  onProxyEvent('webrtc_state', (data: any) => {
-    log(`[webrtc] session=${data.session_id?.slice(0, 8)} state=${data.state}`);
-  });
-
-  onProxyEvent('webrtc_track', (data: any) => {
-    log(`[webrtc] session=${data.session_id?.slice(0, 8)} track=${data.kind} codec=${data.codec}`);
-  });
-
-  onProxyEvent('webrtc_audio_rx', (data: any) => {
-    if (data.packet_count === 1 || data.packet_count === 50) {
-      log(`[webrtc] session=${data.session_id?.slice(0, 8)} browser audio rx #${data.packet_count}`);
-    }
-  });
-
-  // Voicemail events.
-  onProxyEvent('voicemail_started', (data: any) => {
-    log(`[voicemail] started for call ${data.call_id} caller=${data.caller_number}`);
-  });
-
-  onProxyEvent('recording_done', (data: any) => {
-    log(`[voicemail] recording done: ${data.file_path} (${data.duration_ms}ms) caller=${data.caller_number}`);
-    // Save voicemail metadata via VoiceboxManager.
-    voiceboxManager.addMessage?.('default', {
-      callerNumber: data.caller_number || 'Unknown',
-      callerName: null,
-      fileName: data.file_path,
-      durationMs: data.duration_ms,
-    });
-  });
-
-  onProxyEvent('voicemail_error', (data: any) => {
-    log(`[voicemail] error: ${data.error} call=${data.call_id}`);
-  });
-
-  // Send full config to Rust — this binds the SIP socket and starts registrations.
-  const configured = await configureProxyEngine({
-    proxy: appConfig.proxy,
-    providers: appConfig.providers,
-    devices: appConfig.devices,
-    routing: appConfig.routing,
+  registerProxyEventHandlers({
+    log,
+    statusStore,
+    voiceboxManager,
+    webRtcLinks,
+    getBrowserDeviceIds: getAllBrowserDeviceIds,
+    sendToBrowserDevice,
+    broadcast: broadcastWs,
+    onLinkWebRtcSession: requestWebRtcLink,
+    onCloseWebRtcSession: (sessionId) => {
+      void webrtcClose(sessionId);
+    },
  });

+  const configured = await configureRuntime(appConfig);
  if (!configured) {
    log('[FATAL] failed to configure proxy engine');
    process.exit(1);
  }

-  const providerList = appConfig.providers.map((p) => p.displayName).join(', ');
-  const deviceList = appConfig.devices.map((d) => d.displayName).join(', ');
+  const providerList = appConfig.providers.map((provider) => provider.displayName).join(', ');
+  const deviceList = appConfig.devices.map((device) => device.displayName).join(', ');
  log(`proxy engine started | LAN ${appConfig.proxy.lanIp}:${appConfig.proxy.lanPort} | providers: ${providerList} | devices: ${deviceList}`);
-
-  // Generate TTS audio (WAV files on disk, played by Rust audio_player).
-  try {
-    await initAnnouncement(log);
-
-    // Pre-generate prompts.
-    await promptCache.generateBeep('voicemail-beep', 1000, 500, 8000);
-    for (const vb of appConfig.voiceboxes ?? []) {
-      if (!vb.enabled) continue;
-      const promptId = `voicemail-greeting-${vb.id}`;
-      if (vb.greetingWavPath) {
-        await promptCache.loadWavPrompt(promptId, vb.greetingWavPath);
-      } else {
-        const text = vb.greetingText || 'The person you are trying to reach is not available. Please leave a message after the tone.';
-        await promptCache.generatePrompt(promptId, text, vb.greetingVoice || 'af_bella');
-      }
-    }
-    if (appConfig.ivr?.enabled) {
-      for (const menu of appConfig.ivr.menus) {
-        await promptCache.generatePrompt(`ivr-menu-${menu.id}`, menu.promptText, menu.promptVoice || 'af_bella');
-      }
-    }
-    log(`[startup] prompts cached: ${promptCache.listIds().join(', ') || 'none'}`);
-  } catch (e) {
-    log(`[tts] init failed: ${e}`);
-  }
 }

-// ---------------------------------------------------------------------------
-// Web UI
-// ---------------------------------------------------------------------------
-
-initWebUi(
+initWebUi({
+  port: appConfig.proxy.webUiPort,
  getStatus,
  log,
-  (number, deviceId, providerId) => {
-    // Outbound calls from dashboard — send make_call command to Rust.
+  onStartCall: (number, deviceId, providerId) => {
    log(`[dashboard] start call: ${number} device=${deviceId || 'any'} provider=${providerId || 'auto'}`);
-    // Fire-and-forget — the async result comes via events.
-    makeCall(number, deviceId, providerId).then((callId) => {
+    void makeCall(number, deviceId, providerId).then((callId) => {
      if (callId) {
        log(`[dashboard] call started: ${callId}`);
-        activeCalls.set(callId, {
-          id: callId,
-          direction: 'outbound',
-          callerNumber: null,
-          calleeNumber: number,
-          providerUsed: providerId || null,
-          state: 'setting-up',
-          startedAt: Date.now(),
-          legs: new Map(),
-        });
+        statusStore.noteDashboardCallStarted(callId, number, providerId);
      } else {
        log(`[dashboard] call failed for ${number}`);
      }
    });
-    // Return a temporary ID so the frontend doesn't show "failed" immediately.
+
    return { id: `pending-${Date.now()}` };
  },
-  (callId) => {
-    hangupCall(callId);
+  onHangupCall: (callId) => {
+    void hangupCall(callId);
    return true;
  },
-  () => {
-    // Config saved — reconfigure Rust engine.
-    try {
-      const fresh = loadConfig();
-      Object.assign(appConfig, fresh);
-
-      // Update shadow state.
-      for (const p of fresh.providers) {
-        if (!providerStatuses.has(p.id)) {
-          providerStatuses.set(p.id, {
-            id: p.id, displayName: p.displayName, registered: false, publicIp: null,
-          });
-        }
-      }
-      for (const d of fresh.devices) {
-        if (!deviceStatuses.has(d.id)) {
-          deviceStatuses.set(d.id, {
-            id: d.id, displayName: d.displayName, address: null, port: 0, connected: false, isBrowser: false,
-          });
-        }
-      }
-
-      // Re-send config to Rust.
-      configureProxyEngine({
-        proxy: fresh.proxy,
-        providers: fresh.providers,
-        devices: fresh.devices,
-        routing: fresh.routing,
-      }).then((ok) => {
-        if (ok) log('[config] reloaded — proxy engine reconfigured');
-        else log('[config] reload failed — proxy engine rejected config');
-      });
-    } catch (e: any) {
-      log(`[config] reload failed: ${e.message}`);
-    }
-  },
-  undefined, // callManager — legacy, replaced by Rust proxy-engine
-  voiceboxManager, // voiceboxManager
-  // WebRTC signaling → forwarded to Rust proxy-engine.
-  async (sessionId, sdp, ws) => {
+  onConfigSaved: reloadConfig,
+  voiceboxManager,
+  onWebRtcOffer: async (sessionId, sdp, ws) => {
    log(`[webrtc] offer from browser session=${sessionId.slice(0, 8)} sdp_type=${typeof sdp} sdp_len=${sdp?.length || 0}`);
    if (!sdp || typeof sdp !== 'string' || sdp.length < 10) {
      log(`[webrtc] WARNING: invalid SDP (type=${typeof sdp}), skipping offer`);
      return;
    }
+
    log(`[webrtc] sending offer to Rust (${sdp.length}b)...`);
    const result = await webrtcOffer(sessionId, sdp);
    log(`[webrtc] Rust result: ${JSON.stringify(result)?.slice(0, 200)}`);
    if (result?.sdp) {
      ws.send(JSON.stringify({ type: 'webrtc-answer', sessionId, sdp: result.sdp }));
      log(`[webrtc] answer sent to browser session=${sessionId.slice(0, 8)}`);
-    } else {
-      log(`[webrtc] ERROR: no answer SDP from Rust`);
+      return;
    }
+
+    log('[webrtc] ERROR: no answer SDP from Rust');
  },
-  async (sessionId, candidate) => {
+  onWebRtcIce: async (sessionId, candidate) => {
    await webrtcIce(sessionId, candidate);
  },
-  async (sessionId) => {
+  onWebRtcClose: async (sessionId) => {
+    webRtcLinks.removeSession(sessionId);
    await webrtcClose(sessionId);
  },
-  // onWebRtcAccept — browser has accepted a call, linking session to call.
-  (callId: string, sessionId: string) => {
+  onWebRtcAccept: (callId, sessionId) => {
    log(`[webrtc] accept: callId=${callId} sessionId=${sessionId.slice(0, 8)}`);

-    // Store bidirectional mapping.
-    webrtcSessionToCall.set(sessionId, callId);
-    webrtcCallToSession.set(callId, sessionId);
-
-    // Check if we already have media info for this call (provider answered first).
-    const media = pendingCallMedia.get(callId);
-    if (media) {
-      pendingCallMedia.delete(callId);
-      log(`[webrtc] linking session=${sessionId.slice(0, 8)} to call=${callId} media=${media.addr}:${media.port} pt=${media.sipPt}`);
-      webrtcLink(sessionId, callId, media.addr, media.port, media.sipPt).then((ok) => {
-        log(`[webrtc] link result: ${ok}`);
-      });
-    } else {
-      log(`[webrtc] session ${sessionId.slice(0, 8)} accepted, waiting for call_answered media info`);
+    const pendingMedia = webRtcLinks.acceptCall(callId, sessionId);
+    if (pendingMedia) {
+      requestWebRtcLink(callId, sessionId, pendingMedia);
+      return;
    }
+
+    log(`[webrtc] session ${sessionId.slice(0, 8)} accepted, waiting for call_answered media info`);
  },
-);
+});

-// ---------------------------------------------------------------------------
-// Start
-// ---------------------------------------------------------------------------
+void startProxyEngine();

-startProxyEngine();
+process.on('SIGINT', () => {
+  log('SIGINT, exiting');
+  shutdownProxyEngine();
+  process.exit(0);
+});

-process.on('SIGINT', () => { log('SIGINT, exiting'); shutdownProxyEngine(); process.exit(0); });
-process.on('SIGTERM', () => { log('SIGTERM, exiting'); shutdownProxyEngine(); process.exit(0); });
+process.on('SIGTERM', () => {
+  log('SIGTERM, exiting');
+  shutdownProxyEngine();
+  process.exit(0);
+});
--- a/ts/voicebox.ts
+++ b/ts/voicebox.ts
@@ -29,12 +29,14 @@ export interface IVoiceboxConfig {
  greetingVoice?: string;
  /** Path to uploaded WAV greeting (overrides TTS). */
  greetingWavPath?: string;
-  /** Seconds to wait before routing to voicemail (default 25). */
-  noAnswerTimeoutSec: number;
-  /** Maximum recording duration in seconds (default 120). */
-  maxRecordingSec: number;
-  /** Maximum stored messages per box (default 50). */
-  maxMessages: number;
+  /** Seconds to wait before routing to voicemail. Defaults to 25 when
+   *  absent — both the config loader and `VoiceboxManager.init` apply
+   *  the default via `??=`. */
+  noAnswerTimeoutSec?: number;
+  /** Maximum recording duration in seconds. Defaults to 120. */
+  maxRecordingSec?: number;
+  /** Maximum stored messages per box. Defaults to 50. */
+  maxMessages?: number;
 }

 export interface IVoicemailMessage {
@@ -148,6 +150,35 @@ export class VoiceboxManager {
  // Message CRUD
  // -------------------------------------------------------------------------

+  /**
+   * Convenience wrapper around `saveMessage` — used by the `recording_done`
+   * event handler, which has a raw recording path + caller info and needs
+   * to persist metadata. Generates `id`, sets `timestamp = now`, defaults
+   * `heard = false`, and normalizes `fileName` to a basename (the WAV is
+   * expected to already live in the box's directory).
+   */
+  addMessage(
+    boxId: string,
+    info: {
+      callerNumber: string;
+      callerName?: string | null;
+      fileName: string;
+      durationMs: number;
+    },
+  ): void {
+    const msg: IVoicemailMessage = {
+      id: crypto.randomUUID(),
+      boxId,
+      callerNumber: info.callerNumber,
+      callerName: info.callerName ?? undefined,
+      timestamp: Date.now(),
+      durationMs: info.durationMs,
+      fileName: path.basename(info.fileName),
+      heard: false,
+    };
+    this.saveMessage(msg);
+  }
+
  /**
   * Save a new voicemail message.
   * The WAV file should already exist at the expected path.
--- a/ts/webrtcbridge.ts
+++ b/ts/webrtcbridge.ts
@@ -5,8 +5,8 @@
 * - Browser device registration/unregistration via WebSocket
 * - WS → deviceId mapping
 *
- * All WebRTC media logic (PeerConnection, RTP, transcoding) lives in
- * ts/call/webrtc-leg.ts and is managed by the CallManager.
+ * All WebRTC media logic (PeerConnection, RTP, transcoding, mixer wiring)
+ * lives in the Rust proxy-engine. This module only tracks browser sessions.
 */

 import { WebSocket } from 'ws';
@@ -39,7 +39,7 @@ export function initWebRtcSignaling(cfg: IWebRtcSignalingConfig): void {

 /**
 * Handle a WebRTC signaling message from a browser client.
- * Only handles registration; offer/ice/hangup are routed through CallManager.
+ * Only handles registration; offer/ice/hangup are routed through frontend.ts.
 */
 export function handleWebRtcSignaling(
  ws: WebSocket,
@@ -51,7 +51,7 @@ export function handleWebRtcSignaling(
    handleRegister(ws, message.sessionId!, message.userAgent, message._remoteIp);
  }
  // Other webrtc-* types (offer, ice, hangup, accept) are handled
-  // by the CallManager via frontend.ts WebSocket handler.
+  // by the frontend.ts WebSocket handler and forwarded to Rust.
 }

 /**
@@ -64,13 +64,6 @@ export function sendToBrowserDevice(deviceId: string, data: unknown): boolean {
  return true;
 }

-/**
- * Get the WebSocket for a browser device (used by CallManager to create WebRtcLegs).
- */
-export function getBrowserDeviceWs(deviceId: string): WebSocket | null {
-  return deviceIdToWs.get(deviceId) ?? null;
-}
-
 /**
 * Get all registered browser device IDs.
 */
--- a/ts_web/00_commitinfo_data.ts
+++ b/ts_web/00_commitinfo_data.ts
@@ -3,6 +3,6 @@
 */
 export const commitinfo = {
  name: 'siprouter',
-  version: '1.20.2',
+  version: '1.25.0',
  description: 'undefined'
 }
--- a/ts_web/elements/sipproxy-devices.ts
+++ b/ts_web/elements/sipproxy-devices.ts
@@ -41,11 +41,10 @@ export class SipproxyDevices extends DeesElement {
            },
          },
          {
-            key: 'contact',
+            key: 'address',
            header: 'Contact',
            renderer: (_val: any, row: any) => {
-              const c = row.contact;
-              const text = c ? (c.port ? `${c.address}:${c.port}` : c.address) : '--';
+              const text = row.address ? (row.port ? `${row.address}:${row.port}` : row.address) : '--';
              return html`<span style="font-family:'JetBrains Mono',monospace;font-size:.75rem">${text}</span>`;
            },
          },
--- a/ts_web/elements/sipproxy-view-overview.ts
+++ b/ts_web/elements/sipproxy-view-overview.ts
@@ -186,11 +186,10 @@ export class SipproxyViewOverview extends DeesElement {
            },
          },
          {
-            key: 'contact',
+            key: 'address',
            header: 'Contact',
            renderer: (_val: any, row: any) => {
-              const c = row.contact;
-              const text = c ? (c.port ? `${c.address}:${c.port}` : c.address) : '--';
+              const text = row.address ? (row.port ? `${row.address}:${row.port}` : row.address) : '--';
              return html`<span style="font-family:'JetBrains Mono',monospace;font-size:.75rem">${text}</span>`;
            },
          },
--- a/ts_web/elements/sipproxy-view-providers.ts
+++ b/ts_web/elements/sipproxy-view-providers.ts
@@ -164,173 +164,346 @@ export class SipproxyViewProviders extends DeesElement {
        iconName: 'lucide:plus',
        type: ['header'] as any,
        actionFunc: async () => {
-          await this.openAddModal();
-        },
-      },
-      {
-        name: 'Add Sipgate',
-        iconName: 'lucide:phone',
-        type: ['header'] as any,
-        actionFunc: async () => {
-          await this.openAddModal(PROVIDER_TEMPLATES.sipgate, 'Sipgate');
-        },
-      },
-      {
-        name: 'Add O2/Alice',
-        iconName: 'lucide:phone',
-        type: ['header'] as any,
-        actionFunc: async () => {
-          await this.openAddModal(PROVIDER_TEMPLATES.o2, 'O2/Alice');
+          await this.openAddStepper();
        },
      },
    ];
  }

-  // ---- add provider modal --------------------------------------------------
+  // ---- add provider stepper ------------------------------------------------

-  private async openAddModal(
-    template?: typeof PROVIDER_TEMPLATES.sipgate,
-    templateName?: string,
-  ) {
-    const { DeesModal } = await import('@design.estate/dees-catalog');
+  private async openAddStepper() {
+    const { DeesStepper } = await import('@design.estate/dees-catalog');
+    type TDeesStepper = InstanceType<typeof DeesStepper>;
+    // IStep / menuOptions types: we keep content typing loose (`any[]`) to
+    // avoid having to import tsclass IMenuItem just for one parameter annotation.

-    const formData = {
-      displayName: templateName || '',
-      domain: template?.domain || '',
-      outboundProxyAddress: template?.outboundProxy?.address || '',
-      outboundProxyPort: String(template?.outboundProxy?.port ?? 5060),
+    type TProviderType = 'Custom' | 'Sipgate' | 'O2/Alice';
+    interface IAccumulator {
+      providerType: TProviderType;
+      displayName: string;
+      domain: string;
+      outboundProxyAddress: string;
+      outboundProxyPort: string;
+      username: string;
+      password: string;
+      // Advanced — exposed in step 4
+      registerIntervalSec: string;
+      codecs: string;
+      earlyMediaSilence: boolean;
+    }
+
+    const accumulator: IAccumulator = {
+      providerType: 'Custom',
+      displayName: '',
+      domain: '',
+      outboundProxyAddress: '',
+      outboundProxyPort: '5060',
      username: '',
      password: '',
-      registerIntervalSec: String(template?.registerIntervalSec ?? 300),
-      codecs: template?.codecs ? template.codecs.join(', ') : '9, 0, 8, 101',
-      earlyMediaSilence: template?.quirks?.earlyMediaSilence ?? false,
+      registerIntervalSec: '300',
+      codecs: '9, 0, 8, 101',
+      earlyMediaSilence: false,
    };

-    const heading = template
-      ? `Add ${templateName} Provider`
-      : 'Add Provider';
+    // Snapshot the currently-selected step's form (if any) into accumulator.
+    const snapshotActiveForm = async (stepper: TDeesStepper) => {
+      const form = stepper.activeForm;
+      if (!form) return;
+      const data: Record<string, any> = await form.collectFormData();
+      Object.assign(accumulator, data);
+    };

-    await DeesModal.createAndShow({
-      heading,
-      width: 'small',
-      showCloseButton: true,
+    // Overwrite template-owned fields. Keep user-owned fields (username,
+    // password) untouched. displayName is replaced only when empty or still
+    // holds a branded auto-fill.
+    const applyTemplate = (type: TProviderType) => {
+      const tpl =
+        type === 'Sipgate' ? PROVIDER_TEMPLATES.sipgate
+        : type === 'O2/Alice' ? PROVIDER_TEMPLATES.o2
+        : null;
+      if (!tpl) return;
+      accumulator.domain = tpl.domain;
+      accumulator.outboundProxyAddress = tpl.outboundProxy.address;
+      accumulator.outboundProxyPort = String(tpl.outboundProxy.port);
+      accumulator.registerIntervalSec = String(tpl.registerIntervalSec);
+      accumulator.codecs = tpl.codecs.join(', ');
+      accumulator.earlyMediaSilence = tpl.quirks.earlyMediaSilence;
+      if (
+        !accumulator.displayName ||
+        accumulator.displayName === 'Sipgate' ||
+        accumulator.displayName === 'O2/Alice'
+      ) {
+        accumulator.displayName = type;
+      }
+    };
+
+    // --- Step builders (called after step 1 so accumulator is populated) ---
+
+    const buildConnectionStep = (): any => ({
+      title: 'Connection',
      content: html`
-        <div style="display:flex;flex-direction:column;gap:12px;padding:4px 0;">
+        <dees-form>
          <dees-input-text
            .key=${'displayName'}
            .label=${'Display Name'}
-            .value=${formData.displayName}
-            @input=${(e: Event) => { formData.displayName = (e.target as any).value; }}
+            .value=${accumulator.displayName}
+            .required=${true}
          ></dees-input-text>
          <dees-input-text
            .key=${'domain'}
            .label=${'Domain'}
-            .value=${formData.domain}
-            @input=${(e: Event) => { formData.domain = (e.target as any).value; }}
+            .value=${accumulator.domain}
+            .required=${true}
          ></dees-input-text>
          <dees-input-text
            .key=${'outboundProxyAddress'}
            .label=${'Outbound Proxy Address'}
-            .value=${formData.outboundProxyAddress}
-            @input=${(e: Event) => { formData.outboundProxyAddress = (e.target as any).value; }}
+            .value=${accumulator.outboundProxyAddress}
          ></dees-input-text>
          <dees-input-text
            .key=${'outboundProxyPort'}
            .label=${'Outbound Proxy Port'}
-            .value=${formData.outboundProxyPort}
-            @input=${(e: Event) => { formData.outboundProxyPort = (e.target as any).value; }}
+            .value=${accumulator.outboundProxyPort}
          ></dees-input-text>
+        </dees-form>
+      `,
+      menuOptions: [
+        {
+          name: 'Continue',
+          iconName: 'lucide:arrow-right',
+          action: async (stepper: TDeesStepper) => {
+            await snapshotActiveForm(stepper);
+            stepper.goNext();
+          },
+        },
+      ],
+    });
+
+    const buildCredentialsStep = (): any => ({
+      title: 'Credentials',
+      content: html`
+        <dees-form>
          <dees-input-text
            .key=${'username'}
            .label=${'Username / Auth ID'}
-            .value=${formData.username}
-            @input=${(e: Event) => { formData.username = (e.target as any).value; }}
+            .value=${accumulator.username}
+            .required=${true}
          ></dees-input-text>
          <dees-input-text
            .key=${'password'}
            .label=${'Password'}
            .isPasswordBool=${true}
-            .value=${formData.password}
-            @input=${(e: Event) => { formData.password = (e.target as any).value; }}
+            .value=${accumulator.password}
+            .required=${true}
          ></dees-input-text>
+        </dees-form>
+      `,
+      menuOptions: [
+        {
+          name: 'Continue',
+          iconName: 'lucide:arrow-right',
+          action: async (stepper: TDeesStepper) => {
+            await snapshotActiveForm(stepper);
+            stepper.goNext();
+          },
+        },
+      ],
+    });
+
+    const buildAdvancedStep = (): any => ({
+      title: 'Advanced',
+      content: html`
+        <dees-form>
          <dees-input-text
            .key=${'registerIntervalSec'}
            .label=${'Register Interval (sec)'}
-            .value=${formData.registerIntervalSec}
-            @input=${(e: Event) => { formData.registerIntervalSec = (e.target as any).value; }}
+            .value=${accumulator.registerIntervalSec}
          ></dees-input-text>
          <dees-input-text
            .key=${'codecs'}
            .label=${'Codecs (comma-separated payload types)'}
-            .value=${formData.codecs}
-            @input=${(e: Event) => { formData.codecs = (e.target as any).value; }}
+            .value=${accumulator.codecs}
          ></dees-input-text>
          <dees-input-checkbox
            .key=${'earlyMediaSilence'}
            .label=${'Early Media Silence (quirk)'}
-            .value=${formData.earlyMediaSilence}
-            @newValue=${(e: CustomEvent) => { formData.earlyMediaSilence = e.detail; }}
+            .value=${accumulator.earlyMediaSilence}
          ></dees-input-checkbox>
-        </div>
+        </dees-form>
      `,
      menuOptions: [
        {
-          name: 'Cancel',
-          iconName: 'lucide:x',
-          action: async (modalRef: any) => {
-            modalRef.destroy();
+          name: 'Continue',
+          iconName: 'lucide:arrow-right',
+          action: async (stepper: TDeesStepper) => {
+            await snapshotActiveForm(stepper);
+            // Rebuild the review step so its rendering reflects the latest
+            // accumulator values (the review step captures values at build time).
+            stepper.steps = [...stepper.steps.slice(0, 4), buildReviewStep()];
+            await (stepper as any).updateComplete;
+            stepper.goNext();
          },
        },
-        {
-          name: 'Create',
-          iconName: 'lucide:check',
-          action: async (modalRef: any) => {
-            if (!formData.displayName.trim() || !formData.domain.trim()) {
-              deesCatalog.DeesToast.error('Display name and domain are required');
-              return;
-            }
-            try {
-              const providerId = slugify(formData.displayName);
-              const codecs = formData.codecs
+      ],
+    });
+
+    const buildReviewStep = (): any => {
+      const resolvedId = slugify(accumulator.displayName);
+      return {
+        title: 'Review & Create',
+        content: html`
+          <dees-panel>
+            <div
+              style="display:grid;grid-template-columns:auto 1fr;gap:6px 16px;font-size:.85rem;padding:8px 4px;"
+            >
+              <div style="color:#94a3b8;">Type</div>
+              <div>${accumulator.providerType}</div>
+              <div style="color:#94a3b8;">Display Name</div>
+              <div>${accumulator.displayName}</div>
+              <div style="color:#94a3b8;">ID</div>
+              <div style="font-family:'JetBrains Mono',monospace;">${resolvedId}</div>
+              <div style="color:#94a3b8;">Domain</div>
+              <div>${accumulator.domain}</div>
+              <div style="color:#94a3b8;">Outbound Proxy</div>
+              <div>
+                ${accumulator.outboundProxyAddress || accumulator.domain}:${accumulator.outboundProxyPort}
+              </div>
+              <div style="color:#94a3b8;">Username</div>
+              <div>${accumulator.username}</div>
+              <div style="color:#94a3b8;">Password</div>
+              <div>${'*'.repeat(Math.min(accumulator.password.length, 12))}</div>
+              <div style="color:#94a3b8;">Register Interval</div>
+              <div>${accumulator.registerIntervalSec}s</div>
+              <div style="color:#94a3b8;">Codecs</div>
+              <div>${accumulator.codecs}</div>
+              <div style="color:#94a3b8;">Early-Media Silence</div>
+              <div>${accumulator.earlyMediaSilence ? 'yes' : 'no'}</div>
+            </div>
+          </dees-panel>
+        `,
+        menuOptions: [
+          {
+            name: 'Create Provider',
+            iconName: 'lucide:check',
+            action: async (stepper: TDeesStepper) => {
+              // Collision-resolve id against current state.
+              const existing = (this.appData.providers || []).map((p) => p.id);
+              let uniqueId = resolvedId;
+              let suffix = 2;
+              while (existing.includes(uniqueId)) {
+                uniqueId = `${resolvedId}-${suffix++}`;
+              }
+
+              const parsedCodecs = accumulator.codecs
                .split(',')
                .map((s: string) => parseInt(s.trim(), 10))
                .filter((n: number) => !isNaN(n));

              const newProvider: any = {
-                id: providerId,
-                displayName: formData.displayName.trim(),
-                domain: formData.domain.trim(),
+                id: uniqueId,
+                displayName: accumulator.displayName.trim(),
+                domain: accumulator.domain.trim(),
                outboundProxy: {
-                  address: formData.outboundProxyAddress.trim() || formData.domain.trim(),
-                  port: parseInt(formData.outboundProxyPort, 10) || 5060,
+                  address:
+                    accumulator.outboundProxyAddress.trim() || accumulator.domain.trim(),
+                  port: parseInt(accumulator.outboundProxyPort, 10) || 5060,
                },
-                username: formData.username.trim(),
-                password: formData.password,
-                registerIntervalSec: parseInt(formData.registerIntervalSec, 10) || 300,
-                codecs,
+                username: accumulator.username.trim(),
+                password: accumulator.password,
+                registerIntervalSec: parseInt(accumulator.registerIntervalSec, 10) || 300,
+                codecs: parsedCodecs.length ? parsedCodecs : [9, 0, 8, 101],
                quirks: {
-                  earlyMediaSilence: formData.earlyMediaSilence,
+                  earlyMediaSilence: accumulator.earlyMediaSilence,
                },
              };

-              const result = await appState.apiSaveConfig({
-                addProvider: newProvider,
-              });
-
-              if (result.ok) {
-                modalRef.destroy();
-                deesCatalog.DeesToast.success(`Provider "${formData.displayName}" created`);
-              } else {
-                deesCatalog.DeesToast.error('Failed to save provider');
+              try {
+                const result = await appState.apiSaveConfig({
+                  addProvider: newProvider,
+                });
+                if (result.ok) {
+                  await stepper.destroy();
+                  deesCatalog.DeesToast.success(
+                    `Provider "${newProvider.displayName}" created`,
+                  );
+                } else {
+                  deesCatalog.DeesToast.error('Failed to save provider');
+                }
+              } catch (err: any) {
+                console.error('Failed to create provider:', err);
+                deesCatalog.DeesToast.error('Failed to create provider');
+              }
+            },
+          },
+        ],
+      };
+    };
+
+    // --- Step 1: Provider Type ------------------------------------------------
+    //
+    // Note: `DeesStepper.createAndShow()` dismisses on backdrop click; a user
+    // mid-form could lose work. Acceptable for v1 — revisit if users complain.
+
+    const typeOptions: { option: string; key: TProviderType }[] = [
+      { option: 'Custom', key: 'Custom' },
+      { option: 'Sipgate', key: 'Sipgate' },
+      { option: 'O2 / Alice', key: 'O2/Alice' },
+    ];
+    const currentTypeOption =
+      typeOptions.find((o) => o.key === accumulator.providerType) || null;
+
+    const typeStep: any = {
+      title: 'Choose provider type',
+      content: html`
+        <dees-form>
+          <dees-input-dropdown
+            .key=${'providerType'}
+            .label=${'Provider Type'}
+            .options=${typeOptions}
+            .selectedOption=${currentTypeOption}
+            .enableSearch=${false}
+            .required=${true}
+          ></dees-input-dropdown>
+        </dees-form>
+      `,
+      menuOptions: [
+        {
+          name: 'Continue',
+          iconName: 'lucide:arrow-right',
+          action: async (stepper: TDeesStepper) => {
+            // `dees-input-dropdown.value` is an object `{option, key, payload?}`,
+            // not a plain string — extract the `key` directly instead of using
+            // the generic `snapshotActiveForm` helper (which would clobber
+            // `accumulator.providerType`'s string type via Object.assign).
+            const form = stepper.activeForm;
+            if (form) {
+              const data: Record<string, any> = await form.collectFormData();
+              const selected = data.providerType;
+              if (selected && typeof selected === 'object' && selected.key) {
+                accumulator.providerType = selected.key as TProviderType;
              }
-            } catch (err: any) {
-              console.error('Failed to create provider:', err);
-              deesCatalog.DeesToast.error('Failed to create provider');
            }
+            if (!accumulator.providerType) {
+              accumulator.providerType = 'Custom';
+            }
+            applyTemplate(accumulator.providerType);
+            // (Re)build steps 2-5 with current accumulator values.
+            stepper.steps = [
+              typeStep,
+              buildConnectionStep(),
+              buildCredentialsStep(),
+              buildAdvancedStep(),
+              buildReviewStep(),
+            ];
+            await (stepper as any).updateComplete;
+            stepper.goNext();
          },
        },
      ],
-    });
+    };
+
+    await DeesStepper.createAndShow({ steps: [typeStep] });
  }

  // ---- edit provider modal -------------------------------------------------
--- a/ts_web/elements/sipproxy-view-routes.ts
+++ b/ts_web/elements/sipproxy-view-routes.ts
--- a/ts_web/state/appstate.ts
+++ b/ts_web/state/appstate.ts
@@ -2,72 +2,12 @@
 * Application state — receives live updates from the proxy via WebSocket.
 */

-export interface IProviderStatus {
-  id: string;
-  displayName: string;
-  registered: boolean;
-  publicIp: string | null;
-}
+import type { IContact } from '../../ts/config.ts';
+import type { ICallHistoryEntry, ICallStatus, IDeviceStatus, IProviderStatus } from '../../ts/shared/status.ts';

-export interface IDeviceStatus {
-  id: string;
-  displayName: string;
-  contact: { address: string; port: number } | null;
-  aor: string;
-  connected: boolean;
-  isBrowser: boolean;
-}
-
-export interface ILegStatus {
-  id: string;
-  type: 'sip-device' | 'sip-provider' | 'webrtc' | 'tool';
-  state: string;
-  remoteMedia: { address: string; port: number } | null;
-  rtpPort: number | null;
-  pktSent: number;
-  pktReceived: number;
-  codec: string | null;
-  transcoding: boolean;
-  metadata?: Record<string, unknown>;
-}
-
-export interface ICallStatus {
-  id: string;
-  state: string;
-  direction: 'inbound' | 'outbound' | 'internal';
-  callerNumber: string | null;
-  calleeNumber: string | null;
-  providerUsed: string | null;
-  createdAt: number;
-  duration: number;
-  legs: ILegStatus[];
-}
-
-export interface IHistoryLeg {
-  id: string;
-  type: string;
-  metadata: Record<string, unknown>;
-}
-
-export interface ICallHistoryEntry {
-  id: string;
-  direction: 'inbound' | 'outbound' | 'internal';
-  callerNumber: string | null;
-  calleeNumber: string | null;
-  providerUsed: string | null;
-  startedAt: number;
-  duration: number;
-  legs?: IHistoryLeg[];
-}
-
-export interface IContact {
-  id: string;
-  name: string;
-  number: string;
-  company?: string;
-  notes?: string;
-  starred?: boolean;
-}
+export type { IContact };
+export type { ICallHistoryEntry, ICallStatus, IDeviceStatus, IProviderStatus };
+export type { ILegStatus } from '../../ts/shared/status.ts';

 export interface IAppState {
  connected: boolean;
Author	SHA1	Message	Date
Juergen Kunz	89ae12318e	v1.25.0 Some checks failed Docker (tags) / release (push) Failing after 3s Details	2026-04-14 18:52:13 +00:00
Juergen Kunz	feb3514de4	feat(proxy-engine): add live TTS streaming interactions and incoming number range support	2026-04-14 18:52:13 +00:00
Juergen Kunz	adfc4726fd	v1.24.0 Some checks failed Docker (tags) / release (push) Failing after 2s Details	2026-04-14 16:35:54 +00:00
Juergen Kunz	06c86d7e81	feat(routing): require explicit inbound DID routes and normalize SIP identities for provider-based number matching	2026-04-14 16:35:54 +00:00
Juergen Kunz	cff70ab179	v1.23.0 Some checks failed Docker (tags) / release (push) Failing after 3s Details	2026-04-14 10:45:59 +00:00
Juergen Kunz	51f7560730	feat(runtime): refactor runtime state and proxy event handling for typed WebRTC linking and shared status models	2026-04-14 10:45:59 +00:00
Juergen Kunz	5a280c5c41	v1.22.0 Some checks failed Docker (tags) / release (push) Failing after 2s Details	2026-04-12 20:45:08 +00:00
Juergen Kunz	59d8c2557c	feat(proxy-engine): add on-demand TTS caching for voicemail and IVR prompts	2026-04-12 20:45:08 +00:00
Juergen Kunz	cfadd7a2b6	v1.21.0 Some checks failed Docker (tags) / release (push) Failing after 3s Details	2026-04-11 20:04:56 +00:00
Juergen Kunz	80f710f6d8	feat(providers): replace provider creation modal with a guided multi-step setup flow	2026-04-11 20:04:56 +00:00
Juergen Kunz	9ea57cd659	v1.20.5 Some checks failed Docker (tags) / release (push) Failing after 3s Details	2026-04-11 19:20:14 +00:00
Juergen Kunz	c40c726dc3	fix(readme): improve architecture and call flow documentation with Mermaid diagrams	2026-04-11 19:20:14 +00:00
Juergen Kunz	37ba7501fa	v1.20.4 Some checks failed Docker (tags) / release (push) Failing after 3s Details	2026-04-11 19:17:39 +00:00
Juergen Kunz	24924a1aea	fix(deps): bump @design.estate/dees-catalog to ^3.71.1	2026-04-11 19:17:38 +00:00
Juergen Kunz	7ed76a9488	v1.20.3 Some checks failed Docker (tags) / release (push) Failing after 3s Details	2026-04-11 19:02:52 +00:00
Juergen Kunz	a9fdfe5733	fix(ts-config,proxybridge,voicebox): align voicebox config types and add missing proxy bridge command definitions	2026-04-11 19:02:52 +00:00