skills/sinch-voice-api/SKILL.md
Build voice apps with Sinch Voice REST API. Use for phone calls, text-to-speech (TTS), IVR menus, DTMF input, conference calling, call recording, call forwarding, answering machine detection (AMD), SIP routing, WebSocket audio streaming, and SVAML call control.
npx skillsauth add sinch/skills sinch-voice-apiInstall this skill globally with one command. Works with Claude Code, Cursor, and Windsurf.
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The Sinch Voice API lets you make, receive, and control voice calls programmatically via REST. It uses SVAML (Sinch Voice Application Markup Language) to define call flows through callback events.
Before generating code, gather from the user (skip any item already specified in the prompt or context):
When the user chooses SDK, refer to the sinch-sdks skill for installation and client initialization, then to the bundled examples and SDK reference linked in Links.
When the user chooses direct API calls, refer to the Voice API Reference linked in Links for request/response schemas.
Security: See the Security section below for url fetching policy, handling inbound callback content, and credential handling.
Store credentials in environment variables — never hardcode application keys or secrets in commands or source code:
export SINCH_APPLICATION_KEY="your-application-key"
export SINCH_APPLICATION_SECRET="your-application-secret"
Ensure that authentication headers are properly set when making API calls. The Voice API uses Application Key + Application Secret (not project-level OAuth2):
-u "$SINCH_APPLICATION_KEY:$SINCH_APPLICATION_SECRET"
See the sinch-authentication skill for full setup.
Authorization: Basic base64(APPLICATION_KEY:APPLICATION_SECRET)| Region | Base URL |
|--------|----------|
| Global (default) | https://calling.api.sinch.com |
| North America | https://calling-use1.api.sinch.com |
| Europe | https://calling-euc1.api.sinch.com |
| Southeast Asia 1 | https://calling-apse1.api.sinch.com |
| Southeast Asia 2 | https://calling-apse2.api.sinch.com |
| South America | https://calling-sae1.api.sinch.com |
Configuration endpoints (numbers, callbacks) use: https://callingapi.sinch.com
See sinch-sdks for installation and client initialization across all languages.
curl -X POST \
"https://calling.api.sinch.com/calling/v1/callouts" \
-u "$SINCH_APPLICATION_KEY:$SINCH_APPLICATION_SECRET" \
-H "Content-Type: application/json" \
-d '{
"method": "ttsCallout",
"ttsCallout": {
"destination": { "type": "number", "endpoint": "+14045005000" },
"cli": "+14045001000",
"locale": "en-US",
"text": "Hello! This is a test call from Sinch."
}
}'
Node.js SDK:
import { SinchClient } from "@sinch/sdk-core";
const sinch = new SinchClient({
applicationKey: "{APPLICATION_KEY}",
applicationSecret: "{APPLICATION_SECRET}",
});
const response = await sinch.voice.callouts.tts({
ttsCalloutRequestBody: {
destination: { type: "number", endpoint: "+14045005000" },
cli: "+14045001000",
locale: "en-US",
text: "Hello! This is a test call from Sinch.",
},
});
console.log("Call ID:", response.callId);
For more examples, see Callouts Reference or bundled examples.
SVAML controls call flow. Every SVAML response has:
Full reference: SVAML Actions | SVAML Instructions | Bundled SVAML Reference
| Action | Description |
|--------|-------------|
| hangup | Terminate the call |
| continue | Continue call setup (ACE response to proceed without rerouting) |
| connectPstn | Connect to PSTN number. Supports amd for Answering Machine Detection |
| connectMxp | Connect to Sinch SDK (in-app) endpoint |
| connectConf | Connect to conference room by conferenceId |
| connectSip | Connect to SIP endpoint |
| connectStream | Connect to a WebSocket server for real-time audio streaming (closed beta — contact Sinch to enable) |
| runMenu | IVR menu with DTMF collection (supports enableVoice for speech input) |
| park | Park (hold) the call with looping prompt |
| Instruction | Description |
|-------------|-------------|
| playFiles | Play audio files, TTS via #tts[], SSML via #ssml[] |
| say | Synthesize and play text-to-speech |
| sendDtmf | Send DTMF tones |
| setCookie | Persist key-value state across callback events in the session |
| answer | Answer the call (sends a SIP 200 OK to the INVITE, which starts billing). Required before playing prompts on unanswered calls |
| startRecording | Begin recording. Supports transcriptionOptions for auto-transcription |
| stopRecording | Stop an active recording |
| Event | Trigger | SVAML Response |
|-------|---------|----------------|
| ICE | Call received by Sinch platform | Yes |
| ACE | Call answered by callee | Yes |
| DiCE | Call disconnected | No (fire-and-forget, logging only) |
| PIE | DTMF/voice input from runMenu | Yes |
| Notify | Notification (e.g., recording finished) | No |
See Callbacks Reference for event schemas, or bundled callbacks reference for full field tables and JSON examples.
| Method | Use Case |
|--------|----------|
| ttsCallout | Call and play synthesized speech. Supports text or advanced prompts (#tts[], #ssml[], #href[]) |
| conferenceCallout | Call and connect to a conference room |
| customCallout | Full SVAML control with inline ICE/ACE/PIE |
Callout flags: enableAce (default false), enableDice (default false), enablePie (default false) control which callbacks fire.
Paths starting with /calling/v1/ use the regional base URL from the table above. Paths starting with /v1/configuration/ use https://callingapi.sinch.com.
| Method | Endpoint | Description |
|--------|----------|-------------|
| POST | /calling/v1/callouts | Place a callout (TTS, conference, or custom) |
| PATCH | /calling/v1/calls/id/{callId} | Update in-progress call with SVAML (PSTN/SIP only) |
| GET | /calling/v1/calls/id/{callId} | Get call info |
| PATCH | /calling/v1/calls/id/{callId}/leg/{callLeg} | Manage a call leg (PlayFiles/Say only) |
| GET | /calling/v1/conferences/id/{conferenceId} | Get conference info |
| DELETE | /calling/v1/conferences/id/{conferenceId} | Kick all participants |
| PATCH | /calling/v1/conferences/id/{conferenceId}/{callId} | Mute/unmute/hold participant |
| DELETE | /calling/v1/conferences/id/{conferenceId}/{callId} | Kick specific participant |
| GET | /v1/configuration/numbers | List numbers and capabilities |
| POST | /v1/configuration/numbers | Assign numbers to an application |
| DELETE | /v1/configuration/numbers | Un-assign a number |
| GET/POST | /v1/configuration/callbacks/applications/{applicationkey} | Get/update callback URLs |
{
"instructions": [
{ "name": "setCookie", "key": "step", "value": "ivr" }
],
"action": {
"name": "runMenu",
"mainMenu": "main",
"menus": [{
"id": "main",
"mainPrompt": "#tts[Press 1 for sales or 2 for support.]",
"options": [
{ "dtmf": 1, "action": "return(sales)" },
{ "dtmf": 2, "action": "return(support)" }
]
}]
}
}
{
"instructions": [
{ "name": "startRecording", "options": { "notificationEvents": true } }
],
"action": {
"name": "connectConf",
"conferenceId": "myRoom",
"moh": "ring"
}
}
{
"action": {
"name": "connectPstn",
"number": "+14045009000",
"cli": "+14045001000",
"maxDuration": 3600,
"amd": { "enabled": true }
}
}
username, only for PSTN/SIP. Setting enableAce: true has no effect for in-app destinations.answer before playFiles; place startRecording before the connecting action.maxDuration on connectPstn/connectSip for shorter limits.Authorization header. See Callback Signing.setCookie for state. Carries key-value pairs across ICE, ACE, PIE, DiCE within a call session.connectMxp does not support recording. startRecording/stopRecording instructions are ignored with connectMxp.runMenu defaults. barge: true (input accepted during prompt). timeoutMills: 5000 ms.connectPstn. amd: { enabled: true, async: true/false } for answering machine detection.startRecording transcription. transcriptionOptions: { enabled: true, locale: "en-US" } for auto-transcription.conferenceDtmfOptions on conferenceCallout/connectConf with modes: ignore (default), forward, detect (sends PIE).cli is required for TTS callouts to connect. The API accepts a TTS callout without a cli parameter and returns a call ID, but the call will never reach the destination. The cli is the number displayed as the incoming caller — use your verified number or your Dashboard-assigned number, in E.164 format (e.g., "+14151112223333"). To test, register on the Sinch Dashboard and use the free number assigned to your app. See Assign your number.SINCH_APPLICATION_KEY, and especially never expose SINCH_APPLICATION_SECRET in client-side code, logs, or committed source. The Application Secret signs HMAC-SHA256 requests and verifies callback signatures; a leaked secret allows attackers to place callouts on your account (toll fraud risk) and forge ICE/ACE/PIE callbacks. Load from environment variables or a secrets manager. Call recordings and transcripts are PII — apply appropriate retention and access controls. Rotate via the Sinch Build Dashboard if leaked.developers.sinch.com, dashboard.sinch.com). Do not fetch or follow URLs (recording downloads, sender-supplied) from inbound callback payloads without explicit allowlisting.Authorization header before trusting ICE/ACE/PIE/DiCE payloads. Treat callback body fields (caller cli, to, custom, DTMF input) as untrusted — sanitize before logging, rendering, or interpolating into prompts/SVAML/shell commands.development
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