Asterisk - rapida.ai documentation

Asterisk is the world’s most popular open-source telephony platform. Rapida integrates with Asterisk using two transport methods — AudioSocket (native TCP protocol) and WebSocket (chan_websocket) — enabling real-time bidirectional audio streaming for AI-powered voice conversations.

Both transports follow the same two-phase flow: an HTTP webhook registers the call, then a media connection (TCP or WebSocket) carries the audio. Choose the transport that fits your Asterisk version and network topology.

Transport Comparison

Feature	AudioSocket (Native)	WebSocket
Asterisk Module	`res_audiosocket`	`chan_websocket`
Protocol	Raw TCP	WebSocket (wss://)
Port	4573	443 (standard HTTPS)
Audio Codec	SLIN 16-bit 8kHz	μ-law 8kHz
Firewall	Requires TCP 4573 open	Uses standard HTTPS port
TLS	Not built-in	Native TLS via wss://
Asterisk Version	16+	20+
Best For	LAN / private network	Cloud / NAT traversal

Architecture Overview

Both transports share the same backend pipeline. The only difference is the Streamer layer that bridges the media connection to the AI pipeline.

Two-Phase Call Setup

Every Asterisk call (inbound or outbound, AudioSocket or WebSocket) follows this pattern:

Phase 1 — Webhook: An HTTP request registers the call and gets a contextId
Phase 2 — Media: A transport connection (TCP or WebSocket) streams audio using that contextId

This design decouples call setup from media transport, allowing the same webhook to serve both AudioSocket and WebSocket.

Prerequisites

Asterisk Server

Asterisk 16+ (AudioSocket) or 20+ (WebSocket) with the required module loaded

Rapida Account

Active Rapida account with an API key (rpd-xxx)

Voice Assistant

A configured Rapida voice assistant with phone deployment enabled

Network Access

TCP 4573 (AudioSocket) or HTTPS 443 (WebSocket) to Rapida

For outbound calls, you also need Asterisk ARI enabled with credentials stored in a Rapida vault credential.

Quick Start — Inbound Calls

Choose your transport and follow the matching setup:

AudioSocket (Native TCP)
WebSocket

1. Verify Module

asterisk -rx "module show like audiosocket"

If not loaded:

# /etc/asterisk/modules.conf
[modules]
load = res_audiosocket.so

2. Configure Dialplan

# /etc/asterisk/extensions.conf

[globals]
RAPIDA_API_KEY = rpd-xxx-your-api-key
RAPIDA_HOST    = websocket-01.in.rapida.ai
RAPIDA_AS_HOST = socket-01.in.rapida.ai
RAPIDA_PORT    = 4573
ASSISTANT_ID   = 123456789

[rapida-inbound]
exten => _X.,1,Answer()
 same => n,Set(CHANNEL(audioreadformat)=slin)
 same => n,Set(CHANNEL(audiowriteformat)=slin)
 ; Phase 1: Call the webhook to get contextId
 same => n,Set(CTX=${CURL(https://${RAPIDA_HOST}/v1/talk/asterisk/call/${ASSISTANT_ID}?from=${CALLERID(num)}&x-api-key=${RAPIDA_API_KEY})})
 same => n,GotoIf($["${CTX}" = ""]?error)
 ; Phase 2: Connect AudioSocket with contextId as UUID
 same => n,AudioSocket(${CTX},${RAPIDA_AS_HOST}:${RAPIDA_PORT})
 same => n,Hangup()
 same => n(error),Playback(an-error-has-occurred)
 same => n,Hangup()

3. Reload

asterisk -rx "dialplan reload"

1. Verify Module

asterisk -rx "module show like chan_websocket"

If not loaded:

# /etc/asterisk/modules.conf
[modules]
load = chan_websocket.so

2. Configure Dialplan

# /etc/asterisk/extensions.conf

[globals]
RAPIDA_API_KEY = rpd-xxx-your-api-key
RAPIDA_HOST    = websocket-01.in.rapida.ai
ASSISTANT_ID   = 123456789

[rapida-inbound-ws]
exten => _X.,1,Answer()
 ; Phase 1: Call the webhook to get contextId
 same => n,Set(CTX=${CURL(https://${RAPIDA_HOST}/v1/talk/asterisk/call/${ASSISTANT_ID}?from=${CALLERID(num)}&x-api-key=${RAPIDA_API_KEY})})
 same => n,GotoIf($["${CTX}" = ""]?error)
 ; Phase 2: Connect WebSocket with contextId in URL path
 same => n,WebSocket(wss://${RAPIDA_HOST}/v1/talk/asterisk/ctx/${CTX}?x-api-key=${RAPIDA_API_KEY})
 same => n,Hangup()
 same => n(error),Playback(an-error-has-occurred)
 same => n,Hangup()

3. Reload

asterisk -rx "dialplan reload"

Inbound Call Flow — Detailed

Sequence Diagram

What Happens at Each Step

Step	Component	Action
1	Asterisk dialplan	Answers call, sets SLIN audio format
2	`CURL()` function	HTTP GET to Rapida webhook — creates conversation, saves `CallContext` to Redis
3	Webhook response	Returns `contextId` as plain text (UUID)
4a	`AudioSocket()` app	Opens TCP to Rapida:4573, sends `contextId` as UUID frame
4b	`WebSocket()` app	Opens WSS to `wss://host/v1/talk/asterisk/ctx/{contextId}`
5	Rapida transport	Atomically reads and deletes `CallContext` from Redis (prevents replay)
6	AI pipeline	STT → LLM → TTS loop until hangup

Outbound Call Flow — Detailed

Outbound calls are initiated via the Rapida SDK (gRPC CreatePhoneCall RPC) or REST API. Rapida uses Asterisk’s ARI (Asterisk REST Interface) to originate the call.

Prerequisites for Outbound

Enable ARI on Asterisk

# /etc/asterisk/ari.conf
[general]
enabled = yes
pretty = yes

[rapida]
type = user
read_only = no
password = your-ari-password

# /etc/asterisk/http.conf
[general]
enabled = yes
bindaddr = 0.0.0.0
bindport = 8088

Reload:

asterisk -rx "module reload res_ari.so"
asterisk -rx "module reload res_http_websocket.so"

Create Vault Credential in Rapida

In the Rapida dashboard, go to Credentials → Create Credential and select Asterisk. Provide:

Field	Required	Example	Description
`ari_url`	Yes	`http://asterisk.local:8088`	Base URL of Asterisk ARI
`ari_user`	Yes	`rapida`	ARI username
`ari_password`	Yes	`your-ari-password`	ARI password
`endpoint_technology`	No	`PJSIP`	SIP technology (default: PJSIP)
`trunk`	No	`my-sip-trunk`	SIP trunk name for routing

Configure Outbound Dialplan

ARI can originate calls in two modes. Choose one:

Stasis App Mode (Recommended)
Dialplan Mode

ARI originates the channel directly into a Stasis application — no dialplan needed for the outbound leg. The channel is handled programmatically.

# /etc/asterisk/stasis.conf (or use existing ARI app)
; No dialplan needed — ARI handles the channel via the "rapida" app

Rapida sends: POST /ari/channels?endpoint=PJSIP/+15551234567&app=rapida

ARI originates the channel into a dialplan context, which routes to AudioSocket or WebSocket:

# /etc/asterisk/extensions.conf

[rapida-outbound]
exten => s,1,Set(CHANNEL(audioreadformat)=slin)
 same => n,Set(CHANNEL(audiowriteformat)=slin)
 ; RAPIDA_CONTEXT_ID is passed as a channel variable by ARI
 same => n,AudioSocket(${RAPIDA_CONTEXT_ID},${RAPIDA_AS_HOST}:${RAPIDA_PORT})
 same => n,Hangup()

[rapida-outbound-ws]
exten => s,1,NoOp(Outbound WebSocket call)
 ; RAPIDA_CONTEXT_ID is passed as a channel variable by ARI
 same => n,WebSocket(wss://${RAPIDA_HOST}/v1/talk/asterisk/ctx/${RAPIDA_CONTEXT_ID}?x-api-key=${RAPIDA_API_KEY})
 same => n,Hangup()

Set deployment options in Rapida to route via dialplan:

context: rapida-outbound (or rapida-outbound-ws)
extension: s

Assign Credential to Phone Deployment

In the assistant’s Phone Deployment, select the Asterisk vault credential and configure:

Option	Value	Description
Telephony Provider	`asterisk`	Selects Asterisk integration
Credential	(your vault credential)	ARI connection details
Phone Number	`+15559876543`	Caller ID for outbound
Context	(optional)	Dialplan context for outbound
Extension	(optional)	Dialplan extension (default: `s`)
App	(optional)	Stasis app name (default: `rapida`)

Outbound Sequence Diagram

SDK Code Examples (Outbound)

Python
Node.js
Go
cURL

from rapida import Rapida

client = Rapida(api_key="rpd-xxx-your-key")

call = client.calls.create(
    assistant_id=123456789,
    to_number="+15551234567",
    from_number="+15559876543",  # optional, uses deployment default
    metadata={"campaign": "follow-up"},
)

print(f"Call queued: conversation_id={call.conversation.id}")

import { Rapida } from 'rapida';

const client = new Rapida({ apiKey: 'rpd-xxx-your-key' });

const call = await client.calls.create({
  assistantId: 123456789,
  toNumber: '+15551234567',
  fromNumber: '+15559876543',
  metadata: { campaign: 'follow-up' },
});

console.log(`Call queued: ${call.conversation.id}`);

import "github.com/rapidaai/rapida"

client := rapida.NewClient("rpd-xxx-your-key")

call, err := client.Calls.Create(ctx, &rapida.CreateCallRequest{
    AssistantID: 123456789,
    ToNumber:    "+15551234567",
    FromNumber:  "+15559876543",
    Metadata:    map[string]any{"campaign": "follow-up"},
})

curl -X POST https://api.rapida.ai/v1/talk/call \
  -H "Authorization: Bearer rpd-xxx-your-key" \
  -H "Content-Type: application/json" \
  -d '{
    "assistant": { "assistant_id": 123456789 },
    "to_number": "+15551234567",
    "from_number": "+15559876543",
    "metadata": { "campaign": "follow-up" }
  }'

AudioSocket Protocol Reference

Frame Format

Each frame has a 3-byte header followed by a variable-length payload:

+----------+------------------+----------------------+
| Type     | Length            | Payload              |
| 1 byte   | 2 bytes (BE)     | 0-65535 bytes        |
+----------+------------------+----------------------+

Frame Types

Type	Hex	Direction	Description
Hangup	`0x00`	Both	Call termination
UUID	`0x01`	Asterisk → Rapida	Context ID (sent as first frame)
Silence	`0x02`	Asterisk → Rapida	Silence indicator
Audio	`0x10`	Both	Audio data (SLIN 16-bit)
Error	`0xFF`	Both	Error condition

Audio Format

Property	AudioSocket (SLIN)	WebSocket (μ-law)	Rapida Internal
Encoding	Signed Linear 16-bit	μ-law 8-bit	Linear PCM 16-bit
Sample Rate	8000 Hz	8000 Hz	16000 Hz
Channels	Mono	Mono	Mono

Rapida handles all resampling and codec conversion transparently.

Status Callbacks

Rapida sends status updates to Asterisk via ARI events. You can also configure a webhook on the assistant to receive call lifecycle events.

Event Callback URL

POST https://{RAPIDA_HOST}/v1/talk/asterisk/ctx/{contextId}/event?x-api-key={apiKey}

This endpoint is automatically configured during outbound call setup. For inbound calls, Asterisk can optionally POST events to this URL.

Event Types

Event	Trigger
`webhook`	Initial call registration
`channel_created`	ARI channel created (outbound)
`call.connected`	Media connection established
`call.ended`	Call terminated normally
`call.failed`	Error during call

Credential Configuration

Vault Credential Fields

When creating an Asterisk credential in the Rapida vault, provide:

{
  "ari_url": "http://your-asterisk-server:8088",
  "ari_user": "rapida",
  "ari_password": "your-ari-password",
  "endpoint_technology": "PJSIP",
  "trunk": "my-sip-trunk"
}

Field	Required	Default	Description
`ari_url`	Yes (outbound only)	—	Base URL of Asterisk ARI HTTP server
`ari_user`	Yes (outbound only)	—	ARI username
`ari_password`	Yes (outbound only)	—	ARI password
`endpoint_technology`	No	`PJSIP`	Channel technology: `PJSIP`, `SIP`, `IAX2`, `DAHDI`
`trunk`	No	—	SIP trunk for routing: builds `PJSIP/trunk/number`

Vault credentials are encrypted at rest. The ARI password is never exposed in API responses or logs.

Phone Deployment Options

These options are configured on the assistant’s phone deployment:

Option	Default	Description
`phone`	—	Default caller ID for outbound calls
`context`	(none)	Asterisk dialplan context for outbound
`extension`	`s`	Dialplan extension
`app`	`rapida`	Stasis application name
`endpoint_technology`	`PJSIP`	Overrides vault credential
`trunk`	—	Overrides vault credential

Full Configuration Examples

Example 1: Inbound Only (AudioSocket)

Simplest setup — no ARI needed, no vault credential required for inbound-only.

Asterisk Config
Network Requirements

# /etc/asterisk/extensions.conf
[globals]
RAPIDA_API_KEY = rpd-xxx-your-api-key
RAPIDA_HOST    = websocket-01.in.rapida.ai
RAPIDA_PORT    = 4573
ASSISTANT_ID   = 123456789

[from-pstn]
; Route all inbound calls to Rapida AI
exten => _X.,1,Answer()
 same => n,Set(CHANNEL(audioreadformat)=slin)
 same => n,Set(CHANNEL(audiowriteformat)=slin)
 same => n,Set(CTX=${CURL(https://${RAPIDA_HOST}/v1/talk/asterisk/call/${ASSISTANT_ID}?from=${CALLERID(num)}&x-api-key=${RAPIDA_API_KEY})})
 same => n,GotoIf($["${CTX}" = ""]?error)
 same => n,AudioSocket(${CTX},${RAPIDA_AS_HOST}:${RAPIDA_PORT})
 same => n,Hangup()
 same => n(error),Playback(an-error-has-occurred)
 same => n,Hangup()

Asterisk → Rapida: TCP 4573 (AudioSocket)
Asterisk → Rapida: HTTPS 443 (webhook API)

Example 2: Inbound Only (WebSocket)

No firewall port needed beyond HTTPS — ideal for cloud-hosted Asterisk.

# /etc/asterisk/extensions.conf
[globals]
RAPIDA_API_KEY = rpd-xxx-your-api-key
RAPIDA_HOST    = websocket-01.in.rapida.ai
ASSISTANT_ID   = 123456789

[from-pstn-ws]
exten => _X.,1,Answer()
 same => n,Set(CTX=${CURL(https://${RAPIDA_HOST}/v1/talk/asterisk/call/${ASSISTANT_ID}?from=${CALLERID(num)}&x-api-key=${RAPIDA_API_KEY})})
 same => n,GotoIf($["${CTX}" = ""]?error)
 same => n,WebSocket(wss://${RAPIDA_HOST}/v1/talk/asterisk/ctx/${CTX}?x-api-key=${RAPIDA_API_KEY})
 same => n,Hangup()
 same => n(error),Playback(an-error-has-occurred)
 same => n,Hangup()

Example 3: Inbound + Outbound (Full Setup)

Complete setup with ARI for outbound calls and AudioSocket for media.

[globals]
RAPIDA_API_KEY = rpd-xxx-your-api-key
RAPIDA_HOST    = websocket-01.in.rapida.ai
RAPIDA_PORT    = 4573
ASSISTANT_ID   = 123456789

; ---- Inbound: PSTN → Rapida AI ----
[rapida-inbound]
exten => _X.,1,Answer()
 same => n,Set(CHANNEL(audioreadformat)=slin)
 same => n,Set(CHANNEL(audiowriteformat)=slin)
 same => n,Set(CTX=${CURL(https://${RAPIDA_HOST}/v1/talk/asterisk/call/${ASSISTANT_ID}?from=${CALLERID(num)}&x-api-key=${RAPIDA_API_KEY})})
 same => n,GotoIf($["${CTX}" = ""]?error)
 same => n,AudioSocket(${CTX},${RAPIDA_AS_HOST}:${RAPIDA_PORT})
 same => n,Hangup()
 same => n(error),Playback(an-error-has-occurred)
 same => n,Hangup()

; ---- Outbound: Rapida ARI → Customer ----
; ARI originates into this context; RAPIDA_CONTEXT_ID is a channel variable
[rapida-outbound]
exten => s,1,Set(CHANNEL(audioreadformat)=slin)
 same => n,Set(CHANNEL(audiowriteformat)=slin)
 same => n,AudioSocket(${RAPIDA_CONTEXT_ID},${RAPIDA_AS_HOST}:${RAPIDA_PORT})
 same => n,Hangup()

[general]
enabled = yes
pretty = yes

[rapida]
type = user
read_only = no
password = your-ari-password

[general]
enabled = yes
bindaddr = 0.0.0.0
bindport = 8088

{
  "ari_url": "http://asterisk.local:8088",
  "ari_user": "rapida",
  "ari_password": "your-ari-password",
  "endpoint_technology": "PJSIP",
  "trunk": "my-sip-trunk"
}

{
  "phone": "+15559876543",
  "context": "rapida-outbound",
  "extension": "s"
}

Asterisk → Rapida: TCP 4573 (AudioSocket)
Asterisk → Rapida: HTTPS 443 (webhook API)
Rapida → Asterisk: TCP 8088 (ARI, outbound only)

Example 4: WebSocket-Only (Inbound + Outbound)

Zero custom ports — everything runs over HTTPS.

extensions.conf
Rapida Deployment Options

[globals]
RAPIDA_API_KEY = rpd-xxx-your-api-key
RAPIDA_HOST    = websocket-01.in.rapida.ai
ASSISTANT_ID   = 123456789

; ---- Inbound ----
[rapida-inbound-ws]
exten => _X.,1,Answer()
 same => n,Set(CTX=${CURL(https://${RAPIDA_HOST}/v1/talk/asterisk/call/${ASSISTANT_ID}?from=${CALLERID(num)}&x-api-key=${RAPIDA_API_KEY})})
 same => n,GotoIf($["${CTX}" = ""]?error)
 same => n,WebSocket(wss://${RAPIDA_HOST}/v1/talk/asterisk/ctx/${CTX}?x-api-key=${RAPIDA_API_KEY})
 same => n,Hangup()
 same => n(error),Playback(an-error-has-occurred)
 same => n,Hangup()

; ---- Outbound (ARI routes here with RAPIDA_CONTEXT_ID) ----
[rapida-outbound-ws]
exten => s,1,WebSocket(wss://${RAPIDA_HOST}/v1/talk/asterisk/ctx/${RAPIDA_CONTEXT_ID}?x-api-key=${RAPIDA_API_KEY})
 same => n,Hangup()

{
  "phone": "+15559876543",
  "context": "rapida-outbound-ws",
  "extension": "s"
}

Call Lifecycle

State Machine

Captured Metrics

Each call automatically records:

Metric	Description
`call.duration`	Total call time (seconds)
`call.talk_duration`	Active speech time
`call.status`	`completed`, `failed`, `no_answer`
`call.disconnected_by`	`user`, `assistant`, `system`

Troubleshooting

Connection Refused (AudioSocket)

Symptoms: AudioSocket() returns immediately, no audioChecklist:

Verify Rapida host is reachable: nc -zv socket-01.in.rapida.ai 4573
Check firewall allows TCP 4573 outbound
Ensure res_audiosocket.so is loaded
Verify the contextId (token) is non-empty in the dialplan

WebSocket Connection Failed

Symptoms: WebSocket() fails, no audioChecklist:

Verify WSS URL is correct: wss://websocket-01.in.rapida.ai/v1/talk/asterisk/ctx/{contextId}?x-api-key={your-api-key}
Ensure chan_websocket.so is loaded
Check TLS certificate trust on the Asterisk server
Verify Asterisk version is 20+ for chan_websocket

Call Context Not Found

Symptoms: Transport connects but immediately drops with “call context not found”Checklist:

Context ID expires after 5 minutes — ensure fast connection after webhook
Each contextId is single-use (atomically deleted on first connection)
Don’t reuse contextIds across calls
Verify the webhook returned a valid contextId (not an error message)

No Audio / One-Way Audio (AudioSocket)

Symptoms: AI speaks but doesn’t hear, or vice versaChecklist:

Set CHANNEL(audioreadformat)=slin and CHANNEL(audiowriteformat)=slin
Verify res_audiosocket module is loaded
Check Asterisk timing: asterisk -rx "timing test"

Choppy Audio / Delays

Symptoms: Audio artifacts, long pausesChecklist:

Check network latency to Rapida: ping websocket-01.in.rapida.ai (target < 100ms)
Verify no packet loss on the path
For WebSocket: ensure no proxy is buffering WebSocket frames
Reduce network hops if possible

ARI Outbound Call Fails

Symptoms: SDK call returns error, no ring on customer phoneChecklist:

Verify ARI is enabled: asterisk -rx "http show status"
Check ARI credentials in vault: ari_url, ari_user, ari_password
Verify endpoint format: PJSIP/trunk/number or PJSIP/number
Check ARI URL is reachable from Rapida: must not be behind NAT without port forwarding
Review Asterisk CLI: asterisk -rvvv for ARI request logs

RAPIDA_CONTEXT_ID Empty in Outbound Dialplan

Symptoms: Outbound call connects but AudioSocket/WebSocket has no contextChecklist:

Verify ARI channel variables are configured: {"variables": {"RAPIDA_CONTEXT_ID": "..."}}
Check the dialplan references ${RAPIDA_CONTEXT_ID} (exact case)
Verify deployment options include context and extension

Debug Commands

# Check modules
asterisk -rx "module show like audiosocket"
asterisk -rx "module show like chan_websocket"

# Enable verbose debugging
asterisk -rx "core set debug 5"

# Monitor active channels
asterisk -rx "core show channels verbose"

# Check ARI status
asterisk -rx "http show status"

# View dialplan
asterisk -rx "dialplan show rapida-inbound"

# Test network connectivity
nc -zv socket-01.in.rapida.ai 4573   # AudioSocket
curl -I https://websocket-01.in.rapida.ai  # HTTPS/WebSocket

High Availability

Multi-Node Deployment

Dialplan with failover (AudioSocket):

[rapida-inbound-ha]
exten => _X.,1,Answer()
 same => n,Set(CHANNEL(audioreadformat)=slin)
 same => n,Set(CHANNEL(audiowriteformat)=slin)
 same => n,Set(CTX=${CURL(https://${RAPIDA_HOST}/v1/talk/asterisk/call/${ASSISTANT_ID}?from=${CALLERID(num)}&x-api-key=${RAPIDA_API_KEY})})
 same => n,GotoIf($["${CTX}" = ""]?error)
 same => n,AudioSocket(${CTX},rapida-primary:4573)
 same => n,GotoIf($["${AUDIOSOCKET_STATUS}" = "FAILED"]?failover)
 same => n,Hangup()
 same => n(failover),AudioSocket(${CTX},rapida-secondary:4573)
 same => n,Hangup()
 same => n(error),Playback(an-error-has-occurred)
 same => n,Hangup()

API Reference

Inbound Webhook

GET /v1/talk/asterisk/call/{assistantId}

Parameter	Location	Required	Description
`assistantId`	Path	Yes	Rapida assistant ID
`from`	Query	Yes	Caller phone number
`x-api-key`	Query or Header	Yes	API key for authentication
`channel_id`	Query	No	Asterisk channel ID

Response: Plain text contextId (UUID)

WebSocket Media Endpoint

GET /v1/talk/asterisk/ctx/{contextId}
Upgrade: websocket

Parameter	Location	Required	Description
`contextId`	Path	Yes	Context ID from webhook response
`x-api-key`	Query	Yes	API key for authentication

Status Callback

POST /v1/talk/asterisk/ctx/{contextId}/event

Receives ARI-style JSON events with a type field.

Voice Deployment Options

Overview of all voice deployment channels

Create an Assistant

Build your voice AI assistant

Webhooks

Configure event notifications

API Reference

Call API documentation

​Transport Comparison

​Architecture Overview

​Two-Phase Call Setup

​Prerequisites

Asterisk Server

Rapida Account

Voice Assistant

Network Access

​Quick Start — Inbound Calls

​1. Verify Module

​2. Configure Dialplan

​3. Reload

​1. Verify Module

​2. Configure Dialplan

​3. Reload

​Inbound Call Flow — Detailed

​Sequence Diagram

​What Happens at Each Step

​Outbound Call Flow — Detailed

​Prerequisites for Outbound

​Outbound Sequence Diagram

​SDK Code Examples (Outbound)

​AudioSocket Protocol Reference

​Frame Format

​Frame Types

​Audio Format

​Status Callbacks

​Event Callback URL

​Event Types

​Credential Configuration

​Vault Credential Fields

​Phone Deployment Options

​Full Configuration Examples

​Example 1: Inbound Only (AudioSocket)

​Example 2: Inbound Only (WebSocket)

​Example 3: Inbound + Outbound (Full Setup)

​Example 4: WebSocket-Only (Inbound + Outbound)

​Call Lifecycle

​State Machine

​Captured Metrics

​Troubleshooting

​Debug Commands

​High Availability

​Multi-Node Deployment

​API Reference

​Inbound Webhook

​WebSocket Media Endpoint

​Status Callback

​Related Resources

Voice Deployment Options

Create an Assistant

Webhooks

API Reference

Transport Comparison

Architecture Overview

Two-Phase Call Setup

Prerequisites

Quick Start — Inbound Calls

1. Verify Module

2. Configure Dialplan

3. Reload

1. Verify Module

2. Configure Dialplan

3. Reload

Inbound Call Flow — Detailed

Sequence Diagram

What Happens at Each Step

Outbound Call Flow — Detailed

Prerequisites for Outbound

Outbound Sequence Diagram

SDK Code Examples (Outbound)

AudioSocket Protocol Reference

Frame Format

Frame Types

Audio Format

Status Callbacks

Event Callback URL

Event Types

Credential Configuration

Vault Credential Fields

Phone Deployment Options

Full Configuration Examples

Example 1: Inbound Only (AudioSocket)

Example 2: Inbound Only (WebSocket)

Example 3: Inbound + Outbound (Full Setup)

Example 4: WebSocket-Only (Inbound + Outbound)

Call Lifecycle

State Machine

Captured Metrics

Troubleshooting

Debug Commands

High Availability

Multi-Node Deployment

API Reference

Inbound Webhook

WebSocket Media Endpoint

Status Callback

Related Resources