diff --git a/tools/zacus-gateway/main.py b/tools/zacus-gateway/main.py
index 7b6f547..5e07ad4 100644
--- a/tools/zacus-gateway/main.py
+++ b/tools/zacus-gateway/main.py
@@ -1908,12 +1908,17 @@ def _build_wav(pcm: bytes, sr: int, ch: int, sw: int) -> bytes:
     return out.getvalue()
 
 
-def _normalize_wav_for_stt(wav_bytes: bytes, target_peak: float = 0.5,
-                           max_gain: float = 25.0) -> bytes:
-    """Boost a quiet capture up to `target_peak` before STT. The PLIP handset
-    mic comes in very low (~1-4 % FS through the SLIC), too quiet for Kyutai to
-    transcribe — it hears only the noise floor and hallucinates. Peak-normalise
-    in software (gain capped so pure silence isn't blown up into noise)."""
+def _normalize_wav_for_stt(wav_bytes: bytes, target_peak: float = 0.7,
+                           max_gain: float = 40.0, hp_cutoff_hz: float = 110.0) -> bytes:
+    """Condition a PLIP handset capture for Kyutai STT: high-pass then normalise.
+
+    Two problems with the SLIC handset path: (1) it injects low-frequency rumble
+    / DC drift that swamps the speech and makes Kyutai output nothing — a
+    high-pass (~110 Hz, box moving-average subtraction) removes it; (2) the mic
+    comes in very quiet (~2-5 % FS even at +24 dB PGA), so peak-normalise after
+    filtering (gain capped so silence isn't blown up). WITHOUT the high-pass the
+    exact same capture transcribes empty; WITH it, clean French — verified on the
+    bench."""
     if _np is None:
         return wav_bytes
     parsed = _wav_pcm(wav_bytes)
@@ -1923,12 +1928,17 @@ def _normalize_wav_for_stt(wav_bytes: bytes, target_peak: float = 0.5,
     if sw != 2 or not pcm:
         return wav_bytes
     arr = _np.frombuffer(pcm, dtype=_np.int16).astype(_np.float32)
+    # High-pass = signal minus its low-frequency moving average. Window length
+    # sets the cutoff (~sr/k Hz); k≈145 @16 kHz → ~110 Hz.
+    k = max(2, int(sr / max(hp_cutoff_hz, 1.0)))
+    if arr.size > k:
+        lp = _np.convolve(arr, _np.ones(k, dtype=_np.float32) / k, mode="same")
+        arr = arr - lp
     peak = float(_np.max(_np.abs(arr))) or 1.0
     gain = min((target_peak * 32767.0) / peak, max_gain)
-    if gain <= 1.05:  # already loud enough
-        return _build_wav(pcm, sr, ch, sw)
     arr = _np.clip(arr * gain, -32768, 32767).astype(_np.int16)
-    logging.info("STT normalise: peak %.1f%% FS → gain x%.1f", 100 * peak / 32768, gain)
+    logging.info("STT conditioning: high-pass %.0f Hz, peak %.1f%% FS → gain x%.1f",
+                 hp_cutoff_hz, 100 * peak / 32768, gain)
     return _build_wav(arr.tobytes(), sr, ch, sw)