> ## Documentation Index
> Fetch the complete documentation index at: https://koreai.mintlify.app/llms.txt
> Use this file to discover all available pages before exploring further.

# ASR and TTS Troubleshooting

This document covers the most common speech recognition (ASR) and text-to-speech (TTS) issues encountered in Voice Gateway deployments. Each section is structured as: Symptom > Root Cause > Fix, with parameter reference tables and known limitations noted where applicable.

All runtime parameters are set using:

```js theme={null}
userSessionUtils.setCallControlParam('paramName', value);
```

***

## Microsoft Azure ASR

### Utterance Splitting — Speech Split into Multiple Parts

**Symptom**

A continuous user utterance arrives as multiple separate inputs. For example, `Hello, I want to check my balance` is received as two turns: `Hello` then `I want to check my balance`. This causes intent mismatches, loop behavior, or responses to partial input.

**Root Cause**

Azure Speech uses `AzureSegmentationSilenceTimeoutMs` to detect end-of-speech. Natural pauses of 300 ms or more between words can trigger a premature commit. The default value is too aggressive for conversational IVR use cases.

**Fix**

Add a Script node at the start of your Experience Flow:

```js theme={null}
// Standard conversational IVR
userSessionUtils.setCallControlParam(
  'AzureSegmentationSilenceTimeoutMs', 1200
);

// Longer conversational utterances (Agentic app flows)
userSessionUtils.setCallControlParam(
  'AzureSegmentationSilenceTimeoutMs', 2000
);

// Do NOT exceed 3000 — increases latency significantly.
```

<Tip>Start with 1200 ms and increase in 200 ms increments if splitting persists. Test with real callers in your target language before going live.</Tip>

**Parameter Reference**

| Parameter                           | Range        | Purpose                                                                          |
| :---------------------------------- | :----------- | :------------------------------------------------------------------------------- |
| `AzureSegmentationSilenceTimeoutMs` | 800-2000 ms  | End-of-speech silence detection. Lower = faster response. Higher = fewer splits. |
| `sttMinConfidence`                  | 0.4-0.6      | Reduce to accept more valid inputs; raise to filter background noise.            |
| `botNoInputTimeoutMS`               | 5000-8000 ms | Increase for non-English speakers or callers who pause before speaking.          |

<Info>For more details, refer to [Speech Customization](/ai-for-service/channels/voice-gateway/speech-customization)</Info>

***

### Digit and Number Misinterpretation

**Symptom**

Numbers are misinterpreted by the ASR engine. Common patterns include:

* Spoken Hindi numbers such as `ek` combined with 22,000 becoming 1,202,000.
* Card numbers with repeated digits being truncated or garbled.
* Confidence scores above threshold (for example, 75.7%) but transcription is incorrect.

**Fix**

```js theme={null}
// Enable continuous ASR at the digit-collection entity node
userSessionUtils.setCallControlParam('continuousASR', true);
userSessionUtils.setCallControlParam('continuousASRTimeoutInMS', 2000);

// DTMF fallback — strongly recommended for 16-digit card numbers
userSessionUtils.setCallControlParam('dtmfCollectMaxDigits', 16);
userSessionUtils.setCallControlParam('dtmfCollectInterDigitTimeoutMS', 4000);
```

**Input Type Recommendations**

| Input Type               | Recommended Method       | Configuration Notes                                               | Error Rate    |
| :----------------------- | :----------------------- | :---------------------------------------------------------------- | :------------ |
| 4-8 digit PIN            | Speech ASR               | Enable `continuousASR`                                            | Low           |
| 10-digit phone number    | Speech or DTMF           | Speech for English; DTMF for non-English                          | Medium        |
| 16-digit card number     | DTMF preferred           | ASR struggles with repeated digits-use DTMF as primary            | High with ASR |
| Currency amounts (Hindi) | Speech + lower threshold | `sttMinConfidence=0.4` + `AzureSegmentationSilenceTimeoutMs=1500` | Medium        |

***

### Widespread ASR Failure-Outage or Concurrency Breach

**Symptom**

All IVR calls fail simultaneously. The bot responds with the default error message and no speech is recognized across any bots. An Azure region outage typically causes a concurrency limit breach or expired API credentials.

**Immediate Recovery Steps**

1. Verify the affected Azure region at Azure Status. Confirm whether the issue is a full outage or a concurrency limit breach.
2. Switch active calls to the fallback ASR label configured in your Voice Gateway settings.
3. If the issue is caused by expired API credentials, update the Azure Speech key in the platform channel configuration and save the changes.
4. Place a test call and confirm that speech recognition resumes before you re-enable production traffic.

**Prevention — Configure Before Go-Live**

```
Primary ASR:  Label = your-primary-label    (primary region)
Fallback ASR: Label = your-fallback-label   (different region, same vendor)
```

<Warning>Primary and Fallback must use different labels. Using the same label for both provides no fallback protection.</Warning>

<Note>Always configure fallback ASR/TTS before going to production. Use the same vendor but a different region for optimal compatibility. For vendor-level fallback, configure a secondary vendor (for example, Deepgram as a fallback for Azure).</Note>

***

## Microsoft Azure TTS

### Mispronounced Words-SSML Phoneme and Say-As Tags

**Symptom**

Azure Neural Voice mispronounces product names, currency amounts, acronyms, or non-English terms. Digits may be read as words — for example, `6987` spoken as `Che Nau Sath Aath Feet` when using a mismatched voice.

**Fix**

Use SSML tags in your Message nodes:

```xml theme={null}
<!-- Currency amounts -->
<speak>
  <say-as interpret-as="currency" language="hi-IN">
    ₹3,40,650.23
  </say-as>
</speak>

<!-- Digit-by-digit (prevents misread artifacts) -->
<speak>
  Your OTP is <say-as interpret-as="digits">6987</say-as>
</speak>

<!-- Acronym spelled out -->
<speak>
  <say-as interpret-as="spell-out">EMI</say-as>
</speak>

<!-- Pause before key information -->
<speak>
  Your balance is <break time="300ms"/>
  <say-as interpret-as="currency">₹5000</say-as>
</speak>
```

**SSML Tag Reference**

| Problem                   | SSML Fix                              | Effect                                        |
| :------------------------ | :------------------------------------ | :-------------------------------------------- |
| Currency read incorrectly | `<say-as interpret-as="currency">`    | Reads with correct language context           |
| Digits read as words      | `<say-as interpret-as="digits">`      | Forces digit-by-digit: `six nine eight seven` |
| Acronym mispronounced     | `<say-as interpret-as="spell-out">`   | Spells out letter by letter: `E-M-I`          |
| Brand name incorrect      | `<sub alias="correct pronunciation">` | Replaces text with specified pronunciation    |
| Need a pause              | `<break time="500ms"/>`               | Inserts a 0.5-second pause                    |
| Speech too fast           | `<prosody rate="slow">`               | Reduces speech rate for complex information   |

***

### TTS Reading Newlines from LLM or Agentic Responses

**Symptom**

When using LLM or Agentic app responses in voice flows, the TTS engine reads literal newline characters aloud — saying `'backslash n backslash n'` instead of pausing. This occurs because LLM outputs contain formatting characters (`\n`, markdown bold, bullet points) that TTS can't interpret.

**Fix**

Add a Script node before the Message node to sanitize the LLM output:

```js theme={null}
// Replace 'YourLLMNode' with your actual node name
var raw = context.steps.YourLLMNode.response || '';

var clean = raw
  .replace(/\\n/g, ' ')        // literal \n strings from LLM
  .replace(/\n/g, ' ')          // actual newline characters
  .replace(/\r/g, ' ')          // carriage returns
  .replace(/\*\*/g, '')         // markdown bold
  .replace(/\*/g, '')           // markdown bullets/italic
  .replace(/#+\s/g, '')         // markdown headers
  .replace(/^\s*[-•]\s*/gm, '') // bullet characters
  .replace(/\s{2,}/g, ' ')      // collapse multiple spaces
  .trim();

context.ttsOutput = clean;
// Use context.ttsOutput in your Message node
```

<Warning>There is no call control parameter for this fix. Sanitization must be applied in a Script node before the Message node. Adjust the variable path (`context.steps.YourLLMNode.response`) to match your specific flow design.</Warning>

***

### Hindi TTS Failure-Voice-Language Mismatch

**Symptom**

On mid-call language switch to Hindi, TTS throws a `'synthAudio requires language'` error. Logs show `'language: undefined'` and `'voice: undefined'`. Digits may be spoken incorrectly when an English voice receives Devanagari numerals.

**Root Cause**

The voice name and language code don't match. For example, using `language=hi-IN` with `voice=en-IN-AartiIndicNeural` (an English-Indian voice, not a Hindi voice).

**Fix**

Add a Script node before the first Hindi message node:

```js theme={null}
userSessionUtils.setCallControlParam('ttsLanguage', 'hi-IN');
userSessionUtils.setCallControlParam('voiceName', 'hi-IN-SwaraNeural');
userSessionUtils.setCallControlParam('ttsProvider', 'microsoft');

// WRONG:   language=hi-IN + voice=en-IN-AartiIndicNeural
// CORRECT: language=hi-IN + voice=hi-IN-SwaraNeural
```

**Supported Hindi Voices**

| Voice Name               | Gender | Language | Notes                                      |
| ------------------------ | ------ | -------- | ------------------------------------------ |
| `hi-IN-SwaraNeural`      | Female | `hi-IN`  | Recommended natural Hindi.                 |
| `hi-IN-MadhurNeural`     | Male   | `hi-IN`  | Alternate male voice.                      |
| `en-IN-AartiIndicNeural` | Female | `en-IN`  | English-Indian (don't use for Hindi text). |

<Note>The system doesn't validate voice-language matching during configuration. Manually verify that the `voiceName` language prefix matches the `ttsLanguage` code.</Note>

***

### Configured Voice Not Applying

**Symptom**

The voice configured in Voice Preferences (for example, `hi-IN-SwaraNeural`) isn't applied during live calls. The bot uses the default English voice instead, and no error displays.

**Root Cause**

TTS initializes at the first message node. If automation or a script runs first, TTS uses defaults before the voice configuration loads.

**Resolution Options**

| Option              | Action                                                                                                                                                                                          |
| :------------------ | :---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| Fix 1 (Recommended) | Add a short Message node as the first node in your Flow-before any script or automation node.                                                                                                   |
| Fix 2               | After saving voice config in the UI, edit the flow description field and re-save to force re-registration.                                                                                      |
| Fix 3 (Script)      | Set voice via script: `userSessionUtils.setCallControlParam('voiceName', 'hi-IN-SwaraNeural');` before the message node.                                                                        |
| Fix 4 (Google)      | If you see `'No credentials for Google with labels: undefined'` — set `ttsLabel` using a script, not the UI. See [No Credentials for Google](#no-credentials-for-google-label-undefined-error). |

***

## Deepgram ASR / TTS

### Utterance Splitting — utteranceEndMs Tuning

**Symptom**

A user says a long sentence, but Deepgram commits mid-sentence due to natural pauses. Setting `deepgramUtteranceEndMs` to 3000 or 4000 ms results in splits.

**Root Cause**

`deepgramUtteranceEndMs` alone is insufficient. It must be paired with `deepgramEndpointing` to handle micro-pauses within sentences. The endpointing parameter controls Voice Activity Detection (VAD) sensitivity for short gaps; `utteranceEndMs` controls the final silence window.

**Fix**

```js theme={null}
// Recommended for Agentic/Conversational flows
userSessionUtils.setCallControlParam('deepgramUtteranceEndMs', 1500);
userSessionUtils.setCallControlParam('deepgramEndpointing', 500);
userSessionUtils.setCallControlParam('sttMinConfidence', 0.4);
userSessionUtils.setCallControlParam('deepgramNumerals', true);
userSessionUtils.setCallControlParam('deepgramPunctuate', true);

// For Spanish / Latin American Spanish
userSessionUtils.setCallControlParam('sttLanguage', 'es-419');
```

<Note>
  **Why smaller `utteranceEndMs` works better:** `endpointing=500` handles micro-pauses within speech. `utteranceEndMs=1500` handles the longer gap at the end. Setting `utteranceEndMs=3000-4000` delays the final commit but doesn't fix mid-sentence splits.
</Note>

**Parameter Reference**

| Parameter                | Range        | Purpose                                                                                   |
| ------------------------ | ------------ | ----------------------------------------------------------------------------------------- |
| `deepgramUtteranceEndMs` | 800-2000 ms  | End-of-utterance silence window. Start at 1200; increase by 200 ms if splitting persists. |
| `deepgramEndpointing`    | 300-600 ms   | VAD sensitivity. 300 ms = responsive; 600 ms = more patient with pauses.                  |
| `deepgramNumerals`       | `true/false` | Converts spoken numbers to digits (for example, `five hundred` > `500`).                  |
| `deepgramNer`            | `true/false` | Named Entity Recognition — improves proper noun and entity recognition.                   |
| `deepgramPunctuate`      | `true/false` | Adds punctuation to transcripts-useful for NLU processing.                                |
| `deepgramKeyterms`       | String array | Boosts recognition of specific listed words.                                              |

***

### Digit and RX Number Recognition

**Symptom**

The bot is collecting multi-digit numbers (for example, 5-7 digit prescription or account numbers). Callers pause between digits. The bot splits the utterance or misses digits.

**Production-Tested Configuration**

```js theme={null}
var recognizerConfig = {
  vendor: 'deepgram',
  model: 'nova-3-medical', // Use nova-3 for non-medical
  language: 'en-US',
  interim: true,
  deepgramOptions: {
    endpointing: 300,
    utteranceEndMs: 1000,
    keyterms: [
      'zero', 'one', 'two', 'three', 'four',
      'five', 'six', 'seven', 'eight', 'nine'
    ],
    numerals: true
  }
};

userSessionUtils.setCallControlParam('dtmfBargein', false);
userSessionUtils.setCallControlParam('listenDuringPrompt', true);
```

**Model Selection**

| Model              | Best For                               | Use Case                        | Status      |
| ------------------ | -------------------------------------- | ------------------------------- | ----------- |
| `nova-3`           | Best general accuracy                  | Default IVR flows               | Recommended |
| `nova-3-medical`   | Medical/clinical terms, RX numbers     | Healthcare, pharmacy, insurance | Recommended |
| `nova-2-phonecall` | Optimized for telephony audio          | Call centre deployments         | Stable      |
| `enhanced`         | High accuracy, slightly higher latency | Accuracy-critical flows         | Stable      |

***

## Google ASR & TTS

### 16-Digit Card Number — Repeated Digit Failure

**Symptom**

Google ASR fails on sequences with 8 or more consecutive identical digits (for example, `509099999999990002`). The bot may barge in early. This is a known Google ASR limitation.

**Fix**

```js theme={null}
// Recommended: DTMF input for card numbers
userSessionUtils.setCallControlParam('dtmfCollectMaxDigits', 16);
userSessionUtils.setCallControlParam('dtmfCollectInterDigitTimeoutMS', 4000);
userSessionUtils.setCallControlParam('dtmfCollectTermDigit', '#');

// If speech must be used — enable continuous ASR
userSessionUtils.setCallControlParam('continuousASR', true);
userSessionUtils.setCallControlParam('continuousASRTimeoutInMS', 2500);

// Note: AzureSegmentationSilenceTimeoutMs does NOT affect Google ASR.
```

**Model Selection**

| Model             | Description                                      | Use Case               | Status                      |
| ----------------- | ------------------------------------------------ | ---------------------- | --------------------------- |
| `chirp_2`         | Best for telephony + accents                     | Recommended for IVR    | Recommended                 |
| `chirp_telephony` | Optimized for 8 kHz/16 kHz audio                 | Telephony deployments  | Recommended                 |
| `telephony`       | Legacy — stable, widely tested                   | Default                | Good default                |
| `telephony_short` | Fast, for short utterances (less than 5 seconds) | Menu / yes-no commands | Recommended                 |
| `long`            | Batch file processing                            | —                      | Don't use for real-time IVR |

***

### No Credentials for Google — Label: undefined Error

**Symptom**

TTS fails with: `'No text-to-speech service credentials for Google with labels: undefined'`. This occurs even after setting `ttsLabel` in the [Voice Preferences](/ai-for-service/channels/voice-gateway/configure-voice-gateway#voice-preferences).

**Root Cause**

The `ttsLabel` configured in the Voice Preferences UI doesn't propagate to the TTS engine at runtime in some configurations. The label must be set using a Script node.

**Fix**

Add a Script node at the start of your Experience Flow:

```js theme={null}
// TTS configuration
userSessionUtils.setCallControlParam('ttsProvider', 'google');
userSessionUtils.setCallControlParam('ttsLabel', 'your_google_label');
userSessionUtils.setCallControlParam('ttsLanguage', 'en-US');
userSessionUtils.setCallControlParam('voiceName', 'en-US-Chirp3-HD-Aoede');

// ASR configuration (same approach)
userSessionUtils.setCallControlParam('sttProvider', 'google');
userSessionUtils.setCallControlParam('sttLabel', 'your_google_label');
```

<Warning>The Voice Preferences UI setting for `ttsLabel` doesn't propagate at runtime for Google TTS. Always set the label using a script.</Warning>

***

### Background Streaming-Prompts Not Played

**Symptom**

When using Google TTS with Chirp voices and Background Streaming enabled at the Experience Flow level, follow-up prompts (for example, `Is there anything else?`) are logged as synthesized successfully, but no audio is delivered to the caller.

**Fix Options**

| Option                                  | Action                                                                                                            |
| --------------------------------------- | ----------------------------------------------------------------------------------------------------------------- |
| Option 1 — Disable Background Streaming | Go to Flows > Edit > Background Streaming > Disabled. Confirm audio plays correctly.                              |
| Option 2 — Use non-Chirp voice          | Switch to a Google Neural voice (for example, `en-US-Wavenet-F`). Confirmed compatible with Background Streaming. |
| Option 3 — Use Deepgram TTS             | Deepgram TTS has native streaming support and is more stable with Background Streaming enabled.                   |

<Note>Google Chirp voices aren't compatible with Background Streaming.</Note>

***

## Sarvam TTS

### TTS Fails on Currency, Commas, Special Characters

**Symptom**

Sarvam TTS fails on messages containing the `₹` symbol, commas in Indian number format (`3,40,650.23`), or mixed punctuation. The Sarvam API doesn't handle Unicode currency symbols or Indian number formatting natively.

**Fix**

Pre-process text in a Script node before passing to Sarvam TTS:

```js theme={null}
function sanitizeForSarvam(text) {
  return text
    .replace(/₹([\d,]+\.?\d*)/g, function(m, n) {
      var num = parseFloat(n.replace(/,/g, ''));
      if (num >= 100000) return (num / 100000).toFixed(0) + ' lakh rupaye';
      if (num >= 1000) return (num / 1000).toFixed(0) + ' hazaar rupaye';
      return num + ' rupaye';
    })
    .replace(/[,;]/g, ' ')      // remove commas/semicolons
    .replace(/\.(?!\d)/g, ' ')  // periods NOT followed by a digit
    .replace(/[()\[\]{}]/g, '') // brackets
    .replace(/\s{2,}/g, ' ')    // collapse multiple spaces
    .trim();
}

context.sarvamInput = sanitizeForSarvam(context.llmResponse);

// Provider configuration
userSessionUtils.setCallControlParam('ttsProvider', 'custom:sarvamTTS');
userSessionUtils.setCallControlParam('voiceName', 'simran');
```

<Warning>Sarvam doesn't support SSML tags. Use plain text only. All text formatting must be handled in the sanitization script before passing to Sarvam TTS.</Warning>

***

### Background Streaming — Audio Not Played

**Symptom**

When Background Streaming is enabled at the Experience Flow level with Sarvam TTS, audio synthesis is logged as successful in Interactions, but zero audio plays to the caller.

**Fix**

Disable Background Streaming when using Sarvam TTS.

**Steps:** Go to **Experience Flow > Edit > Background Streaming > Disabled**.

<Note>Sarvam TTS isn't compatible with Background Streaming. A platform fix is being tracked. Until resolved, Background Streaming must be disabled for any flow using Sarvam TTS. No configuration workaround is available.</Note>

***

## Call Flow — Barge-in, Transfer & Transcription

### Aggressive Barge-in — Bot Interrupts User Mid-Speech

**Symptom**

The bot plays filler music in under 3 seconds, interrupting the user's utterance. The `ActionHookDelayProcessor` fires too early, especially when there is a language mismatch between TTS text and the voice configuration.

**Fix**

```js theme={null}
// Disable barge-in for specific nodes (legal, OTP, transfer messages)
userSessionUtils.setCallControlParam('node.bargein', false);

// Delay filler music — prevent premature firing
userSessionUtils.setCallControlParam('actionHookDelayInMs', 8000);

// Session-level barge-in sensitivity
userSessionUtils.setCallControlParam('session.bargein', true);
userSessionUtils.setCallControlParam('session.bargeInSensitivity', 'low');
```

**Scenario Guide**

| Scenario                           | Recommended Fix                                                                                                             |
| :--------------------------------- | :-------------------------------------------------------------------------------------------------------------------------- |
| Bot interrupts greetings           | Set `actionHookDelayInMs=8000` or higher. Give the caller time to state full intent.                                        |
| Barge-in on legal/OTP prompts      | Set `node.bargein=false` on those specific message nodes.                                                                   |
| Background noise triggers barge-in | Set `bargeInSensitivity=low`. Ensure noise suppression is active in your telephony setup.                                   |
| Transfer message cut short         | Use `queueCommand: true` — see [Transfer Fires Before TTS Message Completes](#transfer-fires-before-tts-message-completes). |

***

### Transfer Fires Before TTS Message Completes

**Symptom**

Agent transfer fires before the farewell or transfer message finishes playing. The caller hears a partial message before being transferred.

**Root Cause**

A direct Agent Transfer node placed after a Message node creates a race condition — the transfer may execute before TTS playback completes.

**Fix**

Use `queueCommand: true` in a Script node to ensure TTS completes before transfer executes:

```js theme={null}
var transferCmd = {
  type: 'command',
  command: 'redirect',
  queueCommand: true, // Critical parameter — waits for TTS to complete
  data: [{
    verb: 'transfer',
    destination: 'sip:agent@your-sip-domain.com'
  }]
};

// Execute using your flow's command execution method.
```

<Note>The `queueCommand: true` parameter instructs the Voice Gateway to wait until current TTS playback is fully complete before executing the transfer. Use this for any action that follows a voice prompt.</Note>

<Info>For additional information, refer to [Utility Functions in Voice Gateway](/ai-for-service/channels/voice-gateway/utility-functions-in-voice-gateway)</Info>

***

### Transcription Not Displaying (Twilio / ASR Timeout)

**Symptom**

The Transcripts tab is empty after completed calls.

**Diagnosis & Fix**

| Root Cause                       | How to Diagnose                                                                                                                       | Fix                                                                                                                                                                                  |
| :------------------------------- | :------------------------------------------------------------------------------------------------------------------------------------ | :----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| Twilio webhook misconfiguration  | The Twilio Media Stream webhook is missing the transcript endpoint. Transcription is enabled but results aren't sent to the platform. | Verify your webhook URL in the Twilio Console includes the correct transcript endpoint for your environment. Contact [Support](https://support.kore.ai/) to confirm the correct URL. |
| ASR timeout too short            | ASR timeout is set too low (for example, 2000 ms). Calls end with `NO_INPUT` before transcription completes.                          | Increase `botNoInputTimeoutMS` to `8000` ms or higher. Verify `sttMinConfidence` isn't set too high.                                                                                 |
| ChannelOverrideTemplate override | Template-level timeout silently overrides the Experience Flow settings.                                                               | Set `botNoInputTimeoutMS` explicitly in the `ChannelOverrideTemplate` itself, not just in the Flow.                                                                                  |

***

## Quick Reference-All Parameters

All parameters are set using `userSessionUtils.setCallControlParam('paramName', value)`.

| Parameter                           | Vendor        | Value Range       | Purpose                                                             |
| :---------------------------------- | :------------ | :---------------- | :------------------------------------------------------------------ |
| `AzureSegmentationSilenceTimeoutMs` | Azure ASR     | 800-2000 ms       | End-of-speech silence. Increase to reduce utterance splits.         |
| `continuousASR`                     | Azure, Google | `true/false`      | Enable for digit/number collection.                                 |
| `continuousASRTimeoutInMS`          | Azure, Google | 1500-3000 ms      | Wait after last input in continuous mode.                           |
| `sttMinConfidence`                  | All vendors   | 0.4-0.7           | Minimum confidence to accept an utterance. Lower = more permissive. |
| `sttLanguage`                       | All vendors   | BCP-47 code       | ASR language (for example, `en-US`, `hi-IN`, `es-419`, `ja-JP`).    |
| `sttProvider`                       | All vendors   | Provider string   | ASR vendor selection (`microsoft`, `google`, `deepgram`).           |
| `botNoInputTimeoutMS`               | All vendors   | 5000-15000 ms     | Wait before no-input event. Increase for non-English callers.       |
| `deepgramUtteranceEndMs`            | Deepgram      | 800-2000 ms       | End-of-utterance silence. Pair with `deepgramEndpointing`.          |
| `deepgramEndpointing`               | Deepgram      | 300-600 ms        | VAD sensitivity. 300 ms = responsive; 600 ms = patient.             |
| `deepgramNumerals`                  | Deepgram      | `true/false`      | Spoken numbers → digits (*five hundred* → `500`).                   |
| `deepgramPunctuate`                 | Deepgram      | `true/false`      | Adds punctuation to transcripts.                                    |
| `deepgramKeyterms`                  | Deepgram      | String array      | Boost recognition of specific words.                                |
| `ttsLanguage`                       | All TTS       | BCP-47 code       | TTS language. Must match `voiceName` language.                      |
| `ttsProvider`                       | All TTS       | Provider string   | TTS vendor (`microsoft`, `google`, `custom:sarvamTTS`).             |
| `ttsLabel`                          | Google TTS    | Label string      | Must be set using a script, not UI only.                            |
| `voiceName`                         | All TTS       | Voice identifier  | Must match `ttsLanguage`.                                           |
| `node.bargein`                      | All vendors   | `true/false`      | Enable/disable barge-in per node.                                   |
| `session.bargeInSensitivity`        | All vendors   | `low/medium/high` | Session-level barge-in sensitivity.                                 |
| `actionHookDelayInMs`               | All vendors   | 5000-10000 ms     | Delay before filler music fires.                                    |
| `dtmfCollectMaxDigits`              | All vendors   | 1-20              | Max DTMF digits to collect (card numbers: `16`).                    |
| `dtmfCollectInterDigitTimeoutMS`    | All vendors   | 3000-5000 ms      | Wait between DTMF digits.                                           |
| `dtmfCollectTermDigit`              | All vendors   | `#` or `*`        | Termination digit for DTMF collection.                              |
| `listenDuringPrompt`                | All vendors   | `true/false`      | Allow ASR to listen while TTS plays.                                |

***