This guide explains production-tested fixes for ASR and TTS issues on the Kore.ai Voice Gateway. Each section identifies the symptom, root cause, and the exact configuration or code change required.
All parameters are applied using:
userSessionUtils.setCallControlParam('paramName', value);
Microsoft Azure ASR
Utterance Splitting — Speech Split into Multiple Parts
Symptom
The ASR splits one continuous user utterance into multiple turns. For example, "Hello, I want to check my balance" arrives as two separate inputs: "Hello" followed by "I want to check my balance". This causes intent mismatches, loop behaviour, or the bot responding 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:
// Standard conversational flows userSessionUtils.setCallControlParam( 'AzureSegmentationSilenceTimeoutMs', 1200 ); // Longer conversational utterances (agentic app flows) userSessionUtils.setCallControlParam( 'AzureSegmentationSilenceTimeoutMs', 2000 ); // Do NOT exceed 3000 — significantly increases latency.
Parameter Reference
| Parameter | Value / Range | Purpose |
|---|---|---|
AzureSegmentationSilenceTimeoutMs |
800–2000 ms | End-of-speech silence window. Lower = faster response. Higher = fewer splits. |
sttMinConfidence |
0.4–0.6 | Lower to accept more inputs; raise to filter background noise. |
botNoInputTimeoutMS |
5000–8000 ms | Increase for non-English speakers or callers who pause before speaking. |
Digit and Number Misrecognition
Symptom
Numbers are misinterpreted by the ASR engine. Common patterns:
- Spoken Hindi numbers like
ek(one) combined with22,000become1,202,000 - Card numbers with repeated digits are truncated or garbled
- Confidence scores are above threshold (for example, 75.7%), but the transcription is still incorrect
Fix
// 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 Guidance
| 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 recognised across any bots. An Azure region outage typically causes either a concurrency limit breach or expired API credentials.
Immediate Recovery Steps
- Check Azure Service Health — Verify the Azure region status at the Azure Service Health dashboard. Confirm whether the outage is regional or global.
- Verify API Credentials — Confirm the Azure Speech API key and endpoint have not expired. Rotate credentials if necessary and update them in the Kore.ai platform.
- Switch to Fallback ASR — Activate the fallback ASR configuration (different region or secondary vendor) to restore service while the primary region recovers.
- Monitor Concurrency Limits — Review your Azure subscription concurrency quota. If the limit has been breached, request a quota increase or shed load to the fallback.
Prevention — Configure Before Go-Live
// Primary and Fallback MUST use different labels. // Using the same label for both provides no fallback protection. Primary ASR: Label = your-primary-label // primary region Fallback ASR: Label = your-fallback-label // different region, same vendor
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" when using a mismatched voice.
Fix
Use SSML tags in your Message nodes:
<!-- 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 wrong | <sub alias="correct pronunciation"> |
Replaces text with specified pronunciation |
| Need a pause | <break time="500ms"/> |
Inserts a 0.5-second pause |
| Speak slower | <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 happens because LLM outputs contain formatting characters (\n, markdown bold, bullet points) that TTS cannot interpret.
Fix
Add a Script node before the Message node to sanitise the LLM output:
// 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;
// Reference context.ttsOutput in your Message node
Hindi TTS Failure — Voice-Language Mismatch
Symptom
On language switch to Hindi mid-call, TTS throws a synthAudio requires language error. Logs show language: undefined and voice: undefined. Digits like 6987 may be spoken incorrectly when an English voice receives Devanagari numerals.
Root Cause
The voice name and language code do not match. For example, using language=hi-IN with voice=en-IN-AartiIndicNeural — which is an English-Indian voice, not a Hindi voice.
Fix
Add a Script node before the first Hindi message node:
userSessionUtils.setCallControlParam('ttsLanguage', 'hi-IN');
userSessionUtils.setCallControlParam('voiceName', 'hi-IN-SwaraNeural');
userSessionUtils.setCallControlParam('ttsProvider', 'microsoft');
// CRITICAL: voice name and language MUST match.
// 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 — do not use for Hindi text |
Configured Voice Not Applying at Runtime
Symptom
The voice configured in Voice Preferences (for example, hi-IN-SwaraNeural) is not applied during live calls. The bot uses the default English voice instead, with no error shown in the UI.
Resolution
| Issue | Resolution |
|---|---|
| Root Cause | TTS initialises at the first message node. If automation or a script runs first, TTS uses defaults before the voice configuration loads. |
| Fix 1 (Recommended) | Add a short message node as the very first node in your Experience Flow — before any script or automation node. |
| Fix 2 | After saving the 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. |
Deepgram ASR / TTS
Utterance Splitting — utteranceEndMs Tuning
Symptom
The user says a long sentence, but Deepgram commits mid-sentence due to natural pauses. Setting deepgramUtteranceEndMs to 3000 or 4000 still 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
// 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');
Parameter Reference
| Parameter | Value / 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 collects 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
var recognizerConfig = {
vendor: 'deepgram',
model: 'nova-3-medical', // Use nova-3 for non-medical flows
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 |
Optimised for telephony audio | Call centre deployments | Stable |
enhanced |
High accuracy, slightly higher latency | Accuracy-critical flows | Stable |
Google ASR and 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 — DTMF is the reliable alternative for 16-digit card numbers.
Fix
// 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.
Google ASR Model Reference
| Model | Description | Use Case | Status |
|---|---|---|---|
chirp_2 |
Best for telephony + accents | General IVR | Recommended |
chirp_telephony |
Optimised for 8 kHz / 16 kHz audio | Telephony deployments | Recommended |
telephony |
Legacy — stable, widely tested | Default | Good default |
telephony_short |
Fast, for short utterances (< 5 s) | Menu / Y-N commands | Recommended |
long |
Batch file processing | Not for real-time IVR | Do not use for 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 UI.
Root Cause
The ttsLabel configured in the Voice Preferences UI does not 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:
// 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');
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 synthesised successfully, but no audio is delivered to the caller.
Fix Options
| Option | Action |
|---|---|
| Option 1 — Disable Background Streaming | Go to Experience Flow > Edit > Background Streaming > Disabled. Confirm audio plays correctly. |
| Option 2 — Use a 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. |
Sarvam TTS
TTS Fails on Currency, Commas, and Special Characters
Symptom
Sarvam TTS fails on messages containing the ₹ symbol, commas in Indian number format (for example, 3,40,650.23), or mixed punctuation. The Sarvam API does not handle Unicode currency symbols or Indian number formatting natively.
Fix
Pre-process text in a Script node before passing it to Sarvam TTS:
function sanitiseForSarvam(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 and semicolons
.replace(/\.(?!\d)/g, ' ') // periods not followed by a digit
.replace(/[()\[\]{}]/g, '') // brackets
.replace(/\s{2,}/g, ' ') // collapse multiple spaces
.trim();
}
context.sarvamInput = sanitiseForSarvam(context.llmResponse);
// Provider configuration
userSessionUtils.setCallControlParam('ttsProvider', 'custom:sarvamTTS');
userSessionUtils.setCallControlParam('voiceName', 'simran');
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 no audio plays to the caller.
Fix
Disable Background Streaming when using Sarvam TTS. Go to Experience Flow > Edit > Background Streaming > Disabled.
Call Flow — Barge-in, Transfer, and Transcription
Aggressive Barge-in — Bot Interrupts User Mid-Speech
Symptom
The bot plays filler music in under 3 seconds, cutting off the user’s utterance. The ActionHookDelayProcessor fires too early, especially when there is a language mismatch between the TTS text and the voice configuration.
Fix
// 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 to give the caller time to state their full intent. |
| Barge in on legal or 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
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 first:
var transferCmd = {
type: 'command',
command: 'redirect',
queueCommand: true, // Critical — waits for TTS to complete before transfer
data: [{
verb: 'transfer',
destination: 'sip:agent@your-sip-domain.com'
}]
};
// Execute using your flow's command execution method.>
Transcription Not Displaying — Twilio or ASR Timeout
Symptom
The Transcripts tab is empty after completed calls. This can have two distinct root causes: Twilio webhook misconfiguration or an ASR timeout that is too short.
Diagnosis and Fix
| Root Cause | How to Diagnose | Fix |
|---|---|---|
| Twilio webhook misconfiguration | The Twilio Media Stream webhook is missing the Kore transcript endpoint. Transcription is enabled, but results are not sent to the platform. | Verify your webhook URL in the Twilio Console includes the correct Kore transcript endpoint for your environment. Contact Kore support 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 is not set too high. |
| ChannelOverrideTemplate override | Template-level timeout overrides Experience Flow settings silently. | Set botNoInputTimeoutMS explicitly in the ChannelOverrideTemplate itself, not just in the Experience 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 or number collection. |
continuousASRTimeoutInMS |
Azure, Google | 1500–3000 ms | Wait after the 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 the no-input event. Increase for non-English. |
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 or 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 | Maximum 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 is playing. |