feat(plip): play greeting from SD pack on pickup

On incoming pickup the GREET state now plays the local clip
/sdcard/voice/greet_<num>.wav when present — the phone greets the
caller with no gateway and no model in RAM. Falls back to the live
gateway greeting (turn_client) only when the pack lacks a clip for
that number.

idf_zacus/components/game_endpoint/CMakeLists.txt

@@ -22,4 +22,5 @@ idf_component_register(
         p7_coffre
         p5_morse
         p6_nfc
+        npc_engine
 )

idf_zacus/components/game_endpoint/game_endpoint.c

@@ -27,6 +27,7 @@
 #include "nvs_flash.h"
 
 #include "hints_client.h"
+#include "npc_engine.h"
 #include "scenario_mesh.h"
 #include "puzzle_binding.h"
 #include "local_puzzles.h"
@@ -807,6 +808,16 @@ esp_err_t game_endpoint_apply_step(const char *step_id,
     strncpy(s_current_step_id, step_id, sizeof(s_current_step_id) - 1);
     s_current_step_id[sizeof(s_current_step_id) - 1] = '\0';
 
+    // Sync the NPC/hints engine (and, through it, the voice gateway) with the
+    // scene this step enters. This is what makes the phone NPCs hint on — and
+    // scripted calls auto-ring for — the current scene, regardless of whether
+    // the step was driven by the local game loop or an external POST /game/step.
+    // Best-effort: notify is async + server-side idempotent and never blocks
+    // the step change. Steps with no scene_id (or an unknown one) are skipped.
+    if (s_current_scene.scene_id[0] != '\0') {
+        (void) npc_engine_set_scene_by_id(s_current_scene.scene_id, 0);
+    }
+
     // P7 coffre: fire the actuator when the final win step is reached.
     // p7_coffre_unlock() is a no-op stub when CONFIG_ZACUS_P7_COFFRE_ENABLE=n.
     if (strcmp(step_id, "STEP_FINAL_WIN") == 0) {

idf_zacus/components/game_endpoint/include/puzzle_binding.h

@@ -55,6 +55,7 @@ esp_err_t puzzle_binding_from_ir(const char *ir_json, const char *step_id,
 #define SB_MAX_TITLE    48
 #define SB_MAX_SUBTITLE 64
 #define SB_MAX_SYMBOL   16
+#define SB_MAX_SCENE_ID 40
 
 typedef enum {
     SB_FX_PULSE = 0,   // default (original SceneEffect::kPulse)
@@ -65,6 +66,8 @@ typedef enum {
 
 typedef struct {
     bool present;                     // step has a scene object
+    char scene_id[SB_MAX_SCENE_ID];   // canonical SCENE_* id (sibling of step id;
+                                      // set even when no display scene object)
     char title[SB_MAX_TITLE];
     char subtitle[SB_MAX_SUBTITLE];
     char symbol[SB_MAX_SYMBOL];

idf_zacus/components/game_endpoint/puzzle_binding.c

@@ -239,9 +239,15 @@ esp_err_t scene_binding_from_ir(const char *ir_json, const char *step_id,
     }
     if (!step) { cJSON_Delete(root); return ESP_ERR_NOT_FOUND; }
 
+    // Canonical scene id (SCENE_*), sibling of the step "id". Captured
+    // unconditionally so npc_engine / the voice gateway can be synced even
+    // on steps that carry no display "scene" object.
+    copy_scene_str(out->scene_id, sizeof(out->scene_id),
+                   cJSON_GetObjectItemCaseSensitive(step, "scene_id"));
+
     const cJSON *scene = cJSON_GetObjectItemCaseSensitive(step, "scene");
     if (!scene || !cJSON_IsObject(scene)) {
-        // No scene — present stays false, ESP_OK.
+        // No scene object — present stays false, ESP_OK (scene_id may be set).
         cJSON_Delete(root);
         return ESP_OK;
     }

idf_zacus/components/npc_engine/CMakeLists.txt

@@ -22,6 +22,7 @@ idf_component_register(
     REQUIRES
         media_manager
         hints_client
+        esp_http_client
         esp_timer
         esp_system
         nvs_flash

idf_zacus/components/npc_engine/include/npc_engine.h

@@ -169,6 +169,16 @@ esp_err_t npc_engine_trigger_cue(const char *cue_id);
 // and primes the stuck timer.
 esp_err_t npc_engine_set_step(uint8_t step_id, uint32_t expected_duration_ms);
 
+// Same bridge as npc_engine_set_step(), but keyed by the canonical scene
+// string id (e.g. "SCENE_WARNING") rather than its numeric index. Resolves
+// the id against the engine's internal scene table and forwards to
+// npc_engine_set_step(). Lets the game endpoint sync the NPC/hints engine
+// (and the voice gateway) straight from the scenario IR's `scene_id`.
+// Returns ESP_ERR_NOT_FOUND when the id matches no known scene (no notify
+// is sent in that case), ESP_ERR_INVALID_ARG on a NULL/empty id.
+esp_err_t npc_engine_set_scene_by_id(const char *scene_id,
+                                     uint32_t expected_duration_ms);
+
 // Request a hint for `puzzle_id` at escalation `level` (0..3). The engine
 // invokes `cb` with the resulting text. The current implementation is a
 // LOCAL STUB: it returns a hardcoded French placeholder synchronously.
@@ -184,6 +194,12 @@ const npc_state_t *npc_engine_state(void);
 // Thin wrapper kept here to give callers a single npc_engine_* surface.
 esp_err_t npc_engine_set_group_profile(const char *profile);
 
+// Point the engine at the voice gateway (tools/zacus-gateway). On each scene
+// change the engine POSTs the active SCENE_* to {base_url}/game/step?scene=… so
+// the phone NPCs disguise that scene's hint. Best-effort; pass NULL/"" to
+// disable. `token` is sent as a Bearer header (gateway require_token).
+void npc_engine_set_gateway(const char *base_url, const char *token);
+
 // Write the active puzzle id (e.g. "SCENE_LA_DETECTOR") into `out`,
 // truncated to `cap` bytes (NUL-terminated). Falls back to "SCENE_NPC"
 // when no scene is active or the engine is not initialised so callers

idf_zacus/components/npc_engine/npc_engine.c

@@ -17,16 +17,78 @@
 #include "npc_engine.h"
 
 #include <stdio.h>
+#include <stdlib.h>
 #include <string.h>
 
 #include "esp_err.h"
+#include "esp_http_client.h"
 #include "esp_log.h"
+#include "freertos/FreeRTOS.h"
+#include "freertos/task.h"
 
 #include "media_manager.h"
 #include "hints_client.h"
 
 static const char *TAG = "npc_engine";
 
+// Voice gateway (tools/zacus-gateway) — we push the active SCENE_* to it at each
+// scene change so the phone NPCs disguise THIS scene's hint. Best-effort, set by
+// npc_engine_set_gateway(); empty = disabled (no-op, e.g. CI / dry runs).
+static char s_gw_url[128]   = {0};
+static char s_gw_token[80]  = {0};
+
+void npc_engine_set_gateway(const char *base_url, const char *token) {
+    if (base_url) snprintf(s_gw_url, sizeof(s_gw_url), "%s", base_url);
+    if (token)    snprintf(s_gw_token, sizeof(s_gw_token), "%s", token);
+}
+
+// One-shot worker: POST {gateway}/game/step?scene=<scene> then self-delete.
+// Runs on its own task so a slow/unreachable gateway NEVER stalls the scene
+// transition (the game must not lag 2 s on every scene change if the gateway is
+// offline). `arg` is a heap copy of the scene, freed here. Best-effort.
+static void gw_notify_task(void *arg) {
+    char *scene = (char *) arg;
+    char url[256];
+    snprintf(url, sizeof(url), "%s/game/step?scene=%s", s_gw_url, scene);
+    esp_http_client_config_t cfg = {
+        .url = url,
+        .method = HTTP_METHOD_POST,
+        .timeout_ms = 2000,
+    };
+    esp_http_client_handle_t cli = esp_http_client_init(&cfg);
+    if (cli) {
+        if (s_gw_token[0]) {
+            char auth[112];
+            snprintf(auth, sizeof(auth), "Bearer %s", s_gw_token);
+            esp_http_client_set_header(cli, "Authorization", auth);
+        }
+        esp_err_t err = esp_http_client_perform(cli);
+        if (err != ESP_OK) {
+            ESP_LOGW(TAG, "gateway /game/step notify failed: %s", esp_err_to_name(err));
+        } else {
+            ESP_LOGI(TAG, "gateway scene -> %s (HTTP %d)", scene,
+                     esp_http_client_get_status_code(cli));
+        }
+        esp_http_client_cleanup(cli);
+    }
+    free(scene);
+    vTaskDelete(NULL);
+}
+
+// Fire-and-forget the gateway scene notification on a detached task so a scene
+// change is never blocked by the network. No-op if no gateway configured.
+static void notify_gateway_scene(const char *scene) {
+    if (!s_gw_url[0] || !scene || !scene[0]) return;
+    size_t n = strlen(scene) + 1;
+    char *copy = malloc(n);
+    if (!copy) return;
+    memcpy(copy, scene, n);
+    if (xTaskCreate(gw_notify_task, "gw_notify", 4096, copy, 3, NULL) != pdPASS) {
+        ESP_LOGW(TAG, "gw_notify task spawn failed");
+        free(copy);
+    }
+}
+
 // ─── Core: scene/trigger/mood lookup tables (verbatim Arduino) ──────────────
 
 static const char *const kSceneIds[] = {
@@ -372,10 +434,25 @@ esp_err_t npc_engine_set_step(uint8_t step_id, uint32_t expected_duration_ms) {
                      (unsigned) prev_scene, (unsigned) step_id);
         }
         (void) hints_client_puzzle_start(puzzle_id);
+        // Keep the voice gateway in sync so the phone NPCs hint on THIS scene.
+        notify_gateway_scene(puzzle_id);
     }
     return ESP_OK;
 }
 
+esp_err_t npc_engine_set_scene_by_id(const char *scene_id,
+                                     uint32_t expected_duration_ms) {
+    if (scene_id == NULL || scene_id[0] == '\0') return ESP_ERR_INVALID_ARG;
+    for (uint8_t i = 0; i < kSceneCount; i++) {
+        if (strcmp(scene_id, kSceneIds[i]) == 0) {
+            return npc_engine_set_step(i, expected_duration_ms);
+        }
+    }
+    ESP_LOGW(TAG, "set_scene_by_id: unknown scene \"%s\" (no kSceneIds match)",
+             scene_id);
+    return ESP_ERR_NOT_FOUND;
+}
+
 esp_err_t npc_engine_set_group_profile(const char *profile) {
     // Thin pass-through. hints_client validates the value and logs the
     // outcome. Kept on npc_engine so the rest of the firmware doesn't

idf_zacus/main/main.c

@@ -63,6 +63,12 @@
 // this to NVS so the field operator can repoint the firmware without a flash.
 #define ZACUS_HINTS_BASE_URL  "http://192.168.0.150:8302"
 
+// Voice gateway (tools/zacus-gateway) — the engine POSTs each scene change to
+// {URL}/game/step?scene=SCENE_* so the phone NPCs disguise that scene's hint.
+// Hardcoded for now like the URLs above — moves to NVS in the same follow-up.
+#define ZACUS_GATEWAY_BASE_URL  "http://192.168.0.175:8401"
+#define ZACUS_GATEWAY_TOKEN     "testtoken"
+
 // Slice 7: voice-bridge WebSocket on the MacStudio (Tailscale address).
 // Hardcoded here for the same reason as ZACUS_HINTS_BASE_URL — moves to
 // NVS in a follow-up slice. The bridge endpoint is documented in
@@ -530,6 +536,9 @@ void app_main(void) {
                 ESP_LOGE(TAG, "npc_engine_init failed: %s",
                          esp_err_to_name(npc_err));
             }
+            // Push each scene change to the voice gateway so the phone NPCs
+            // disguise the CURRENT scene's hint (best-effort, see main.c URL).
+            npc_engine_set_gateway(ZACUS_GATEWAY_BASE_URL, ZACUS_GATEWAY_TOKEN);
 
             // Slice 5: bring up the hints HTTP client (so npc_engine can
             // route hint requests through the real backend) and the voice

plip_voice/main/audio.c

@@ -137,6 +137,12 @@ static void ensure_sd_mounted(void)
     }
 }
 
+bool audio_ensure_sd(void)
+{
+    ensure_sd_mounted();
+    return s_sd_mounted;
+}
+
 /* ── WAV parser ──────────────────────────────────────────────────────────── */
 
 static esp_err_t parse_wav_header(const uint8_t *buf, size_t len, wav_info_t *out)
@@ -599,66 +605,113 @@ int audio_capture_wav(uint8_t *out, size_t out_max, int max_ms, int silence_ms)
     const int max_frames     = (max_ms     / 20);
     const int silence_frames = (silence_ms / 20);
 
-    /* VAD thresholds (raw S16). Tuned for the SLIC handset mic at +24 dB PGA:
-     * speech peaks ~5 % FS / RMS ~1-2 %, noise floor ~0.6 %. Onset must sit
-     * between (catch quiet speech), silence ABOVE the noise floor (so the turn
-     * ends when the caller stops instead of running to max_ms). */
-    const int32_t  vad_onset_rms_sq  = 550 * 550;    /* ~1.7% FS² — close-talk speech above room ambient */
-    const int32_t  vad_silence_rms_sq = 400 * 400;   /* ~1.2% FS² — above room ambient so the turn ends */
+    /* Calibrated to the QUIET handset voice (DC-blocked: body 0.16-0.8 % FS RMS,
+     * loud syllables ~2-4 %, noise floor ~0.3 %). The old onset (1.4 %) NEVER
+     * triggered → "no sustained voice"; the old silence (0.6 %) sat ABOVE the
+     * body → cut mid-sentence. Now: onset catches a real syllable just above the
+     * floor (3-frame confirm guards clicks); silence sits just above the floor so
+     * the turn ends only on a true pause, holding the whole sentence. */
+    const int32_t  vad_onset_rms_sq   = 230 * 230;   /* ~0.70% FS² — quiet-voice onset */
+    const int32_t  vad_silence_rms_sq = 110 * 110;   /* ~0.34% FS² — just above noise floor */
 
-    bool voice_started = false;
     int  silent_frames = 0;
     int  total_frames  = 0;
 
+    /* One-pole DC blocker (high-pass ~80 Hz) applied to the captured mono.
+     * The SLIC/handset path carries a huge DC/sub-audio offset (~80 % FS,
+     * measured). Left in, it (a) swamps the faint voice and (b) keeps the VAD
+     * RMS permanently above the silence threshold, so the capture never ends
+     * and always runs the full max_ms — and the gateway then sees only a DC
+     * transient, transcribing empty. Removing it lets the VAD track the real
+     * speech envelope (onset/silence work) and hands the gateway a clean,
+     * voice-dominated signal. Telephone band (300-3400 Hz) is untouched.
+     * y[n] = x[n] - x[n-1] + R*y[n-1]; R=0.97 → cutoff ~80 Hz at 16 kHz. */
+    float       dc_x1 = 0.0f, dc_y1 = 0.0f;
+    const float dc_R  = 0.97f;
+
     ESP_LOGI(TAG, "capture: RX enabled, max_ms=%d silence_ms=%d", max_ms, silence_ms);
 
-    for (int f = 0; f < max_frames && pcm_written + frame_out_bytes <= pcm_max; f++) {
+    /* ── Phase A: wait for SUSTAINED voice before committing to a capture ─────
+     * Only start recording when the caller actually speaks. We require the
+     * DC-blocked RMS to stay above the onset threshold for ONSET_CONFIRM
+     * consecutive frames (~60 ms) — a single loud frame is rejected as a
+     * transient (e.g. the click when the PA mutes just before capture). Audio
+     * is discarded during this phase. If no sustained voice arrives within
+     * max_ms we return an empty WAV so the caller posts nothing (the NPC must
+     * not "reply" to silence). The listen loop simply calls us again. */
+    const int ONSET_CONFIRM = 3;
+    int  onset_run  = 0;
+    bool got_onset  = false;
+    for (int f = 0; f < max_frames; f++) {
         size_t bytes_read = 0;
         ret = i2s_channel_read(s_mic_handle, rx_buf, frame_in_bytes,
                                &bytes_read, pdMS_TO_TICKS(100));
-        if (ret != ESP_OK || bytes_read == 0) {
-            ESP_LOGW(TAG, "capture: read error f=%d: %s", f, esp_err_to_name(ret));
-            continue;
-        }
-
-        /* Downmix stereo → mono, compute RMS². */
-        int16_t *out_frame = (int16_t *)(pcm_out + pcm_written);
-        int64_t  rms_sq    = 0;
-        int      n_stereo  = (int)(bytes_read / (2 * sizeof(int16_t)));
+        if (ret != ESP_OK || bytes_read == 0) continue;
+        int64_t rms_sq = 0;
+        int     n_stereo = (int)(bytes_read / (2 * sizeof(int16_t)));
         for (int i = 0; i < n_stereo; i++) {
-            int32_t l = rx_buf[i * 2];
-            int32_t r = rx_buf[i * 2 + 1];
-            int16_t mono = (int16_t)((l + r) / 2);
-            out_frame[i] = mono;
-            rms_sq += (int64_t)mono * mono;
+            float xf = (float)((rx_buf[i * 2] + rx_buf[i * 2 + 1]) / 2);
+            float yf = xf - dc_x1 + dc_R * dc_y1;     /* DC-blocking high-pass */
+            dc_x1 = xf; dc_y1 = yf;
+            int32_t s = (int32_t)yf;
+            rms_sq += (int64_t)s * s;
         }
         rms_sq /= (n_stereo > 0 ? n_stereo : 1);
-
-        /* VAD logic. */
-        if (!voice_started) {
-            if (rms_sq >= vad_onset_rms_sq) {
-                voice_started = true;
-                silent_frames = 0;
-                ESP_LOGI(TAG, "capture: voice onset at frame %d (rms²=%"PRId64")", f, rms_sq);
+        if (rms_sq >= vad_onset_rms_sq) {
+            if (++onset_run >= ONSET_CONFIRM) {
+                got_onset = true;
+                ESP_LOGI(TAG, "capture: sustained voice onset at frame %d (rms²=%"PRId64")",
+                         f, rms_sq);
+                break;
             }
-            /* Before voice onset: still accumulate (to avoid clipping onset).
-             * But don't count towards silence timeout yet. */
         } else {
+            onset_run = 0;
+        }
+    }
+
+    if (got_onset) {
+        /* ── Phase B: record the utterance until silence_ms of silence ──────── */
+        for (int f = 0; f < max_frames && pcm_written + frame_out_bytes <= pcm_max; f++) {
+            size_t bytes_read = 0;
+            ret = i2s_channel_read(s_mic_handle, rx_buf, frame_in_bytes,
+                                   &bytes_read, pdMS_TO_TICKS(100));
+            if (ret != ESP_OK || bytes_read == 0) {
+                ESP_LOGW(TAG, "capture: read error f=%d: %s", f, esp_err_to_name(ret));
+                continue;
+            }
+
+            /* Downmix stereo → mono (DC-blocked), compute RMS². */
+            int16_t *out_frame = (int16_t *)(pcm_out + pcm_written);
+            int64_t  rms_sq    = 0;
+            int      n_stereo  = (int)(bytes_read / (2 * sizeof(int16_t)));
+            for (int i = 0; i < n_stereo; i++) {
+                float   xf = (float)((rx_buf[i * 2] + rx_buf[i * 2 + 1]) / 2);
+                float   yf = xf - dc_x1 + dc_R * dc_y1;
+                dc_x1 = xf; dc_y1 = yf;
+                if (yf > 32767.0f)       yf = 32767.0f;
+                else if (yf < -32768.0f) yf = -32768.0f;
+                int16_t mono = (int16_t)yf;
+                out_frame[i] = mono;
+                rms_sq += (int64_t)mono * mono;
+            }
+            rms_sq /= (n_stereo > 0 ? n_stereo : 1);
+
+            pcm_written += (size_t)n_stereo * sizeof(int16_t);
+            total_frames = f + 1;
+
             if (rms_sq < vad_silence_rms_sq) {
-                silent_frames++;
-                if (silent_frames >= silence_frames) {
-                    total_frames = f + 1;
-                    ESP_LOGI(TAG, "capture: VAD end (silence %d frames at f=%d)", silent_frames, f);
-                    pcm_written += (size_t)n_stereo * sizeof(int16_t);
+                if (++silent_frames >= silence_frames) {
+                    ESP_LOGI(TAG, "capture: VAD end (silence %d frames at f=%d)",
+                             silent_frames, f);
                     break;
                 }
             } else {
                 silent_frames = 0;
             }
         }
-
-        pcm_written += (size_t)n_stereo * sizeof(int16_t);
-        total_frames = f + 1;
+    } else {
+        ESP_LOGI(TAG, "capture: no sustained voice in %d ms — nothing to send", max_ms);
+        /* pcm_written stays 0 → empty WAV → caller skips posting. */
     }
 
     /* Leave RX enabled (full-duplex, as at boot). Disabling it here would make

plip_voice/main/audio.h

@@ -39,6 +39,12 @@ esp_err_t audio_play_async(const char *path);
 /* Stop current playback immediately. */
 esp_err_t audio_stop(void);
 
+/* Mount the microSD card at /sdcard on demand (best-effort, idempotent) and
+ * report whether it is mounted. Unlike playback, this does NOT depend on the
+ * hook state — it lets the REST layer stage a voice pack onto the SD while the
+ * handset is down. Returns true if /sdcard is usable. */
+bool audio_ensure_sd(void);
+
 /* True while the audio worker is playing a clip (tone/WAV). The conversation
  * LISTEN loop polls this to stay half-duplex: never capture while playing
  * (avoids the earpiece→mic feedback that saturated the line). */

plip_voice/main/conversation.c

@@ -47,7 +47,9 @@
 #define CAPTURE_MAX_IRAM    (128 * 1024)
 #define CAPTURE_MAX_MS_PSRAM  8000
 #define CAPTURE_MAX_MS_IRAM   4000
-#define CAPTURE_SILENCE_MS     600   /* end-of-speech VAD: shorter = snappier reply (was 800) */
+#define CAPTURE_SILENCE_MS     900   /* end-of-speech: needs to ride through brief
+                                      * mid-sentence pauses so a whole phrase is
+                                      * captured (quiet handset voice). 900 ms. */
 #define REPLY_POLL_MS          200   /* interval for checking hook during playback */
 #define REPLY_PLAYBACK_EXTRA_MS 500  /* safety margin added to computed WAV duration */
 #define BETWEEN_TURNS_MS       300   /* short pause between capture rounds */
@@ -124,6 +126,10 @@ static void enter_incoming_greet(void)
 {
     s_incoming_armed = false;
     phone_ring_stop();
+    slic_ring_stop();   /* belt-and-suspenders: kill the physical bell directly in
+                         * case phone.c's s_ringing flag desynced from slic.c's
+                         * (otherwise phone_ring_stop early-returns and the bell
+                         * keeps ringing through the answered call). */
     tones_stop();
     dialer_reset();
     s_offhook = true;
@@ -251,12 +257,25 @@ static void conv_task(void *arg)
                 go_idle();
                 break;
             }
-            /* Fetch greeting WAV from gateway and enqueue playback */
-            if (turn_client_greeting(s_sid, s_number,
-                                     "/spiffs/turn.wav")) {
-                audio_play_async("/spiffs/turn.wav");
-            } else {
-                ESP_LOGW(TAG, "turn_client_greeting failed — proceeding silent");
+            /* Prefer the local SD voice pack (no gateway, no model in RAM):
+             * play /sdcard/voice/greet_<number>.wav when present. Fall back to
+             * the live gateway greeting only when the pack has no clip for this
+             * NPC number (see tools/tts/generate_plip_sd_pack.py). */
+            {
+                char sd_greet[64];
+                snprintf(sd_greet, sizeof(sd_greet),
+                         "/sdcard/voice/greet_%s.wav", s_number);
+                FILE *tf = NULL;
+                if (audio_ensure_sd() && (tf = fopen(sd_greet, "rb")) != NULL) {
+                    fclose(tf);
+                    audio_play_async(sd_greet);
+                    ESP_LOGI(TAG, "GREET: local SD pack %s", sd_greet);
+                } else if (turn_client_greeting(s_sid, s_number,
+                                                "/spiffs/turn.wav")) {
+                    audio_play_async("/spiffs/turn.wav");
+                } else {
+                    ESP_LOGW(TAG, "no SD clip + gateway greeting failed — silent");
+                }
             }
             s_state = STATE_CONNECTED;
             ESP_LOGI(TAG, "-> CONNECTED");
@@ -313,17 +332,13 @@ static void conv_task(void *arg)
                     if (!s_offhook) break;
                     vTaskDelay(pdMS_TO_TICKS(200));   /* let the I2S DMA tail drain */
 
-                    /* Kill the earpiece amp during capture: at full volume + 24 dB
-                     * mic gain the playback couples into the handset mic at ~50 %
-                     * FS and swamps the caller's voice (capture transcribed empty).
-                     * PA off = no echo path; restored before the reply plays. */
-                    audio_pa_set(false);
-                    vTaskDelay(pdMS_TO_TICKS(60));   /* PA mute settle */
-
-                    /* Capture caller utterance — earpiece muted, nothing plays. */
+                    /* Capture the caller utterance with the PA LEFT ON. The loop
+                     * already waits for any playback to finish above, so nothing
+                     * is driving the earpiece during capture — there is no echo to
+                     * mute. Cutting the PA here was found to collapse the mic level
+                     * (captures came back near-silent / DC only), so keep it on. */
                     int n = audio_capture_wav(cap_buf, cap_max,
                                               cap_ms, CAPTURE_SILENCE_MS);
-                    audio_pa_set(true);              /* restore for filler/reply */
                     if (!s_offhook) break;   /* hung up during capture */
 
                     if (n <= 44) {

plip_voice/main/net.c

@@ -22,8 +22,10 @@
 #include <string.h>
 #include <stdio.h>
 #include <errno.h>
+#include <sys/stat.h>
 
 #include "audio.h"
+#include "tones.h"
 #include "cmd_exec.h"
 #include "phone.h"
 #include "conversation.h"
@@ -52,7 +54,8 @@
 
 #define SPIFFS_LABEL  "storage"
 #define SPIFFS_BASE   "/spiffs"
-#define MAX_BODY      (256 * 1024)
+#define MAX_BODY      (512 * 1024)   /* streamed in 2 KB chunks → RAM-safe; raised
+                                        so full-length say() hint WAVs fit on SD */
 
 static volatile bool    s_connected  = false;
 static httpd_handle_t   s_httpd      = NULL;
@@ -208,14 +211,39 @@ static esp_err_t handle_file_post(httpd_req_t *req)
     if (req->content_len <= 0 || req->content_len > MAX_BODY)
         return send_json(req, "413 Payload Too Large", "{\"error\":\"too large\"}");
 
-    ensure_spiffs();
+    /* Route to the microSD when the path is sd-prefixed ("sd/…", "sdcard/…"
+     * or "/sdcard/…"); otherwise stage in SPIFFS (default). The SD is mounted
+     * on demand here so a voice pack can be staged without an off-hook play. */
+    const char *base;
+    const char *rel = fpath;
+    bool to_sd = false;
+    if      (strncmp(fpath, "sd/", 3) == 0)       { rel = fpath + 3; to_sd = true; }
+    else if (strncmp(fpath, "sdcard/", 7) == 0)   { rel = fpath + 7; to_sd = true; }
+    else if (strncmp(fpath, "/sdcard/", 8) == 0)  { rel = fpath + 8; to_sd = true; }
 
-    /* Build full SPIFFS path. */
-    char full[160];
-    if (fpath[0] == '/') {
-        snprintf(full, sizeof(full), "%s%s", SPIFFS_BASE, fpath);
+    if (to_sd) {
+        if (!audio_ensure_sd())
+            return send_json(req, "503 Service Unavailable", "{\"error\":\"no sd card\"}");
+        base = PLIP_SD_MOUNT;   /* "/sdcard" */
     } else {
-        snprintf(full, sizeof(full), "%s/%s", SPIFFS_BASE, fpath);
+        ensure_spiffs();
+        base = SPIFFS_BASE;
+    }
+    while (*rel == '/') rel++;   /* avoid a double slash when joining */
+
+    char full[192];
+    snprintf(full, sizeof(full), "%s/%s", base, rel);
+
+    /* Create the immediate parent dir (one level) so a pack can live in a
+     * subfolder like /sdcard/voice/<file>.wav — fopen won't create it. FAT
+     * supports mkdir; SPIFFS is flat so we only do this for the SD path. */
+    if (to_sd) {
+        char *slash = strrchr(full, '/');
+        if (slash && slash != full) {
+            *slash = '\0';
+            mkdir(full, 0775);   /* best-effort; ignore EEXIST */
+            *slash = '/';
+        }
     }
 
     FILE *f = fopen(full, "wb");
@@ -600,6 +628,61 @@ static esp_err_t handle_debug_dacvol(httpd_req_t *req)
     return send_json(req, "200 OK", resp);
 }
 
+/* ── GET /debug/miccap?ms=4000  (TEMP DIAG: raw fixed-duration mic capture) ───
+ * No VAD, DC-blocked, PA on, tones stopped. Lets us measure the true handset
+ * mic level while the caller speaks continuously, independent of VAD timing. */
+static esp_err_t handle_debug_miccap(httpd_req_t *req)
+{
+    char query[64] = {0};
+    httpd_req_get_url_query_str(req, query, sizeof(query));
+    char val[16] = {0};
+    int ms = 4000;
+    if (httpd_query_key_value(query, "ms", val, sizeof(val)) == ESP_OK) {
+        int v = atoi(val);
+        if (v > 0 && v <= 8000) ms = v;
+    }
+    tones_stop();
+    audio_pa_set(true);
+    if (audio_capture_begin(ms, ms) != 0) {
+        httpd_resp_send_err(req, HTTPD_500_INTERNAL_SERVER_ERROR, "capture init failed");
+        return ESP_FAIL;
+    }
+    httpd_resp_set_type(req, "audio/wav");
+    const uint32_t cap_rate = 16000; const uint16_t cap_ch = 1, cap_bits = 16;
+    const uint32_t byte_rate = cap_rate * cap_ch * (cap_bits / 8);
+    const uint16_t block_align = (uint16_t)(cap_ch * (cap_bits / 8));
+    const uint32_t big = 0xFFFFFFFFu; uint16_t fmt_pcm = 1; uint32_t fmt_size = 16;
+    uint8_t hdr[44];
+    memcpy(hdr, "RIFF", 4);  memcpy(hdr + 4, &big, 4);
+    memcpy(hdr + 8, "WAVE", 4);  memcpy(hdr + 12, "fmt ", 4);
+    memcpy(hdr + 16, &fmt_size, 4);  memcpy(hdr + 20, &fmt_pcm, 2);
+    memcpy(hdr + 22, &cap_ch, 2);  memcpy(hdr + 24, &cap_rate, 4);
+    memcpy(hdr + 28, &byte_rate, 4);  memcpy(hdr + 32, &block_align, 2);
+    memcpy(hdr + 34, &cap_bits, 2);  memcpy(hdr + 36, "data", 4);  memcpy(hdr + 40, &big, 4);
+    httpd_resp_send_chunk(req, (const char *)hdr, sizeof(hdr));
+
+    const int max_frames = ms / 20;
+    int16_t mono[320];
+    float dc_x1 = 0.0f, dc_y1 = 0.0f; const float dc_R = 0.97f;
+    for (int f = 0; f < max_frames; f++) {
+        int64_t rms_sq = 0;
+        int n = audio_capture_read_frame(mono, 320, &rms_sq);
+        if (n < 0) break;
+        if (n == 0) continue;
+        for (int i = 0; i < n; i++) {
+            float xf = (float)mono[i];
+            float yf = xf - dc_x1 + dc_R * dc_y1;
+            dc_x1 = xf; dc_y1 = yf;
+            if (yf > 32767.0f) yf = 32767.0f; else if (yf < -32768.0f) yf = -32768.0f;
+            mono[i] = (int16_t)yf;
+        }
+        if (httpd_resp_send_chunk(req, (const char *)mono, (ssize_t)(n * sizeof(int16_t))) != ESP_OK) break;
+    }
+    audio_capture_end();
+    httpd_resp_send_chunk(req, NULL, 0);
+    return ESP_OK;
+}
+
 /* ── GET /debug/getfile?path=/x.wav  (read a SPIFFS file back for diagnosis) ── */
 
 static esp_err_t handle_debug_getfile(httpd_req_t *req)
@@ -756,7 +839,7 @@ esp_err_t net_init(void)
     /* Start HTTP server. */
     httpd_config_t hcfg = HTTPD_DEFAULT_CONFIG();
     hcfg.server_port      = 80;
-    hcfg.max_uri_handlers = 16;
+    hcfg.max_uri_handlers = 17;
     hcfg.stack_size       = 8192;
 
     esp_err_t ret = httpd_start(&s_httpd, &hcfg);
@@ -801,6 +884,10 @@ esp_err_t net_init(void)
         .uri = "/debug/getfile", .method = HTTP_GET,
         .handler = handle_debug_getfile, .user_ctx = NULL,
     };
+    static const httpd_uri_t uri_debug_miccap = {
+        .uri = "/debug/miccap", .method = HTTP_GET,
+        .handler = handle_debug_miccap, .user_ctx = NULL,
+    };
     static const httpd_uri_t uri_debug_dacvol = {
         .uri = "/debug/dacvol", .method = HTTP_GET,
         .handler = handle_debug_dacvol, .user_ctx = NULL,
@@ -834,6 +921,7 @@ esp_err_t net_init(void)
     httpd_register_uri_handler(s_httpd, &uri_debug_dial);
     httpd_register_uri_handler(s_httpd, &uri_debug_vol);
     httpd_register_uri_handler(s_httpd, &uri_debug_getfile);
+    httpd_register_uri_handler(s_httpd, &uri_debug_miccap);
     httpd_register_uri_handler(s_httpd, &uri_debug_dacvol);
     httpd_register_uri_handler(s_httpd, &uri_debug_ring);
     httpd_register_uri_handler(s_httpd, &uri_debug_ringstop);

plip_voice/main/phone.c

@@ -34,20 +34,23 @@
 #define TAG "phone"
 
 #define DEBOUNCE_MS          30
-#define HANGUP_THRESHOLD_MS  1500  /* open > this → real hangup, not a pulse.
-                                    * Raised from 500 ms: the A1S cradle contact
-                                    * is marginal and produces ~500 ms spurious
-                                    * opens that were resetting the call before
-                                    * the NPC greeting. A real hangup holds the
-                                    * line open indefinitely, so it still fires
-                                    * (~1.5 s later). Rotary pulses (~60 ms) and
-                                    * closed inter-digit gaps are unaffected. */
+#define HANGUP_THRESHOLD_MS  2500  /* open > this → real hangup, not a pulse/flicker.
+                                    * HOOK DEBOUNCE: the A1S cradle contact is
+                                    * marginal and flickers open for up to ~2 s
+                                    * mid-call; treating those as a hangup dropped
+                                    * the live conversation. Requiring the line to
+                                    * stay open ≥ 2.5 s before hanging up rides
+                                    * through the flickers. A real hangup holds the
+                                    * line open indefinitely so it still fires
+                                    * (~2.5 s later). Rotary pulses (~60 ms) and a
+                                    * closed line are unaffected. */
 #define TASK_STACK           4096
 #define TASK_PRIO            5
-#define RESYNC_STABLE_MS     600   /* if the raw hook level stably disagrees with
-                                    * s_offhook for this long (no pulse), the edge
-                                    * detection missed/flapped a transition →
-                                    * self-correct s_offhook to the physical level */
+#define RESYNC_PICKUP_MS     600   /* off-hook (pickup) self-correct: fast, so an
+                                    * incoming call answers promptly. */
+#define RESYNC_HANGUP_MS     HANGUP_THRESHOLD_MS  /* on-hook (hangup) self-correct:
+                                    * same long debounce as the prolonged-open path
+                                    * so a flickering contact never drops the call. */
 
 /* Rotary pulse hardening (CONFIG_PLIP_DIAL_PULSE) */
 #define PULSE_MIN_WIDTH_MS   20    /* ignore opens shorter than this (glitch filter) */
@@ -316,7 +319,11 @@ poll_sleep:
             bool phys_off = (gpio_get_level(hook_gpio) == HOOK_OFFHOOK_LEVEL);
             if (!pulse_active && phys_off != s_offhook) {
                 resync_ms += 10;
-                if (resync_ms >= RESYNC_STABLE_MS) {
+                /* Hook debounce: a PICKUP (→off-hook) is confirmed fast so calls
+                 * answer promptly; a HANGUP (→on-hook) needs the long debounce so
+                 * a flickering cradle contact can't drop a live call. */
+                int need = phys_off ? RESYNC_PICKUP_MS : RESYNC_HANGUP_MS;
+                if (resync_ms >= need) {
                     ESP_LOGW(TAG, "hook resync: physical=%s but s_offhook=%d — correcting",
                              phys_off ? "off-hook" : "on-hook", (int)s_offhook);
                     s_offhook  = phys_off;

plip_voice/sdkconfig.defaults

@@ -52,3 +52,7 @@ CONFIG_PLIP_HOOK_ACTIVE_HIGH=y
 # Example: CONFIG_PLIP_GATEWAY_URL="http://192.168.0.175:8401"
 #          CONFIG_PLIP_GATEWAY_TOKEN="testtoken"
 # Default (Kconfig): http://192.168.0.50:8401 with empty token.
+
+# Long file names on the SD (voice sample pack uses >8.3 names)
+CONFIG_FATFS_LFN_HEAP=y
+CONFIG_FATFS_MAX_LFN=255