Asterisk ARI

Overview

Asterisk ARI (Asterisk REST Interface) allows you to connect SentiVue Talk workflows to your existing Asterisk PBX. ARI provides a WebSocket-based event model for controlling calls via Stasis applications, giving SentiVue Talk full control over call flow and audio streaming. This guide focuses on the SentiVue Talk-specific configuration. For general Asterisk installation and administration, refer to the official Asterisk documentation.

Prerequisites

Before setting up the ARI integration, ensure you have:

A running Asterisk instance with chan_websocket and res_websocket_client modules available. Known-working setups: (a) Asterisk 22+, (b) Asterisk 20 LTS with these modules included
ARI module enabled in Asterisk
chan_websocket (WebSocket channel driver) and res_websocket_client (loads websocket_client.conf) enabled in your Asterisk build. Verify with asterisk -rx "module show like chan_websocket" and asterisk -rx "module show like res_websocket_client" — both should report Running.
Your Asterisk server’s ARI endpoint reachable from the internet, since SentiVue Talk connects out to it
A SentiVue Talk account

If you compiled Asterisk from source, ensure both chan_websocket and res_websocket_client are included during the build. These modules are required for external media streaming between Asterisk and SentiVue Talk. Refer to the Asterisk build system documentation for details on enabling modules.

Asterisk Configuration

The following Asterisk configuration files need to be set up to work with SentiVue Talk. These are minimal examples focused on the SentiVue Talk integration — refer to the Asterisk documentation for full configuration details.

Enable ARI (`ari.conf`)

Create an ARI user that SentiVue Talk will use to authenticate:

[general]
enabled = yes

[sentivue]
type = user
read_only = no
password = your_secure_password

The username (section name, e.g., sentivue) and password here must match the Stasis App Name and App Password you configure in SentiVue Talk.

Enable the HTTP Server (`http.conf`)

ARI requires the Asterisk HTTP server to be enabled:

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

Configure the Stasis Dialplan (`extensions.conf`)

Route incoming calls to your Stasis application so SentiVue Talk can handle them:

[from-external]
exten => _X.,1,NoOp(Incoming call to ${EXTEN})
 same => n,Stasis(sentivue)
 same => n,Hangup()

Replace sentivue with the app name you configured in ari.conf and in SentiVue Talk.

Configure External Media Streaming (`websocket_client.conf`)

SentiVue Talk uses Asterisk’s external media streaming to send and receive audio over WebSocket. Configure a WebSocket client connection that points at SentiVue Talk, so your Asterisk streams call audio out to it:

[sentivue]
type = websocket_client
uri = wss://app.sentivue.com/talk/api/v1/telephony/ws/ari
protocols = media
tls_enabled = yes
ca_list_file = /etc/ssl/certs/ca-certificates.crt

tls_enabled = yes is required even though the URI scheme is wss:// — without it Asterisk will not negotiate TLS and the connection will fail. The ARI credentials (Stasis App Name and App Password) must match what you configure in the SentiVue Talk dashboard under Telephony Settings.

The section name (e.g., sentivue) is the WebSocket Client Name you’ll enter in the SentiVue Talk telephony configuration. This name tells Asterisk which WebSocket connection to use for external media streaming during calls.

SentiVue Talk’s external media channel uses G.711 μ-law (ulaw). Make sure any PJSIP endpoint or SIP trunk that places or receives calls through SentiVue Talk allows ulaw (e.g. allow=ulaw in the endpoint config).

Refer to the Asterisk WebSocket documentation for additional websocket_client.conf options and TLS configuration.

Apply the configuration changes

After editing any of the files above, reload the affected Asterisk modules from the Asterisk CLI (asterisk -rvvv):

ari reload                                 # picks up ari.conf changes
dialplan reload                            # picks up extensions.conf changes
module reload res_websocket_client.so      # picks up websocket_client.conf changes

Changes to http.conf require a full Asterisk reload (core reload) or a service restart.

Configuration in SentiVue Talk

Step 1: Navigate to Telephony Settings

Navigate to /telephony-configurations and click Add configuration
Select Asterisk ARI as your provider

Step 2: Enter Your ARI Credentials

Configure the following fields:

Field	Description	Example
ARI Endpoint URL	HTTP base URL of your Asterisk ARI server	`http://asterisk.example.com:8088`
Stasis App Name	The ARI username configured in `ari.conf`	`sentivue`
App Password	The ARI password configured in `ari.conf`	`your_secure_password`
WebSocket Client Name	The connection name from `websocket_client.conf`	`sentivue`
From Extensions	Optional SIP extensions or trunk numbers for outbound calls	`PJSIP/6001` or `6001`

Step 3: Save and Add Extensions

Click Save Configuration
Open the configuration you just created and add each SIP extension that should be reachable as a phone number (e.g. 8000). For inbound, you’ll assign a workflow to each extension separately — see Inbound Calling below.
Create a test workflow and initiate a test call to verify the connection.

Inbound Calling

Unlike other telephony providers that use HTTP webhooks for inbound calls, ARI delivers inbound calls as StasisStart events on the ARI WebSocket. SentiVue Talk automatically detects these events and activates the workflow assigned to the called extension.

How It Works

An external call arrives at Asterisk and the dialplan routes it to Stasis(sentivue)
Asterisk fires a StasisStart event over the ARI WebSocket with the channel in Ring state and the dialed extension in the dialplan context
SentiVue Talk looks up the called extension in your telephony configuration’s phone numbers, finds the assigned workflow, validates quota, and creates a workflow run
The call is answered, bridged to an external media channel, and your workflow begins

Workflow assignment is per extension, so different extensions on the same Asterisk can route to different agents.

Setting Up Inbound Calls

Step 1: Configure the Asterisk dialplan Ensure your dialplan routes the extensions you care about into the Stasis application. Either route a specific extension:

[from-external]
exten => 8000,1,NoOp(Incoming call to 8000)
 same => n,Stasis(sentivue)
 same => n,Hangup()

…or use a pattern that catches every extension you’ll register in SentiVue Talk:

[from-external]
exten => _X.,1,NoOp(Incoming call to ${EXTEN})
 same => n,Stasis(sentivue)
 same => n,Hangup()

Replace sentivue with the app name you configured in ari.conf and in SentiVue Talk. Step 2: Add the extension as a phone number in SentiVue Talk

Go to /telephony-configurations and open your Asterisk ARI configuration
In the Phone numbers section, add a phone number whose address is the SIP extension (e.g. 8000)
Set its Inbound workflow to the agent that should answer
Save
Adding the extension in SentiVue Talk doesn’t change Asterisk’s dialplan — that’s what Step 1 is for. The SentiVue Talk entry tells the StasisStart handler which workflow to run when a call to that extension reaches the Stasis app.

Repeat Step 2 for each extension that should reach a workflow. Step 3: Test an inbound call Place a call to one of the extensions you configured. You should see the assigned workflow activate and the workflow respond.

Inbound Call Context

When an inbound call activates a workflow, the following context is available to your workflow:

Field	Description
`caller_number`	The caller’s phone number or extension
`called_number`	The dialed number or extension
`direction`	Always `inbound`
`call_id`	The Asterisk channel ID
`provider`	Always `ari`

Call handling setup

Asterisk numbers use the Standard setup: calls reach your agent through the Asterisk REST Interface. This supports inbound calls, outbound calls, and cold transfer to a SIP endpoint on your server (PJSIP/sales-queue) or an external number. Warm transfer and live translation need the Advanced setup, which is currently available on Twilio only. Support for Asterisk is in progress.

Troubleshooting

Cannot connect to ARI endpoint

Verify the ARI endpoint URL is correct and reachable from the public internet, since SentiVue Talk connects out to it
Check that the Asterisk HTTP server is running (http.conf has enabled = yes)
Ensure firewall rules allow traffic on the ARI port (default: 8088)
Confirm the ARI module is loaded: run module show like res_ari in the Asterisk CLI

Authentication failed

Verify the Stasis App Name matches the ARI user section name in ari.conf
Check the App Password matches the password in ari.conf
Ensure there are no extra spaces in the credentials

No audio during calls

Verify chan_websocket is loaded: run module show like chan_websocket in the Asterisk CLI
Check that websocket_client.conf is correctly configured with the right SentiVue Talk URI
Ensure the WebSocket Client Name in SentiVue Talk matches the section name in websocket_client.conf
Verify network connectivity and firewall rules allow WebSocket traffic between Asterisk and SentiVue Talk

Calls not reaching SentiVue Talk

Ensure the dialplan routes calls to Stasis(your_app_name)
Verify the app name in the dialplan matches the ARI user in ari.conf
Check Asterisk CLI for errors: asterisk -rvvv
Confirm the ARI WebSocket connection is active

Inbound calls are immediately hung up

Verify the called extension is added as a phone number under your ARI configuration in /telephony-configurations and has an Inbound workflow assigned
Confirm the workflow exists and belongs to the same organization as the ARI config
Check that your organization has available quota
Review SentiVue Talk logs for warnings like “no matching phone number registered for config” or “has no inbound_workflow_id assigned”

WebSocket client connection issues

Check the URI in websocket_client.conf matches the SentiVue Talk URL shown above
Verify your Asterisk server can reach that URL outbound (DNS, egress firewall, proxy)
Ensure tls_enabled = yes is set and ca_list_file points at a valid CA bundle

Best Practices

Restrict ARI access to known IP addresses using firewall rules, and use a strong App Password
The ARI WebSocket drops if a keepalive goes unanswered for 10 seconds, so a link that stalls that long ends calls in progress
Read Asterisk’s logs alongside SentiVue Talk’s when debugging a call — each side only sees its own half

Getting started

Core Concepts

Models and providers

Building workflows

Telephony

Billing and access

Overview

Prerequisites

Asterisk Configuration

Enable ARI (`ari.conf`)

Enable the HTTP Server (`http.conf`)

Configure the Stasis Dialplan (`extensions.conf`)

Configure External Media Streaming (`websocket_client.conf`)

Apply the configuration changes

Configuration in SentiVue Talk

Step 1: Navigate to Telephony Settings

Step 2: Enter Your ARI Credentials

Step 3: Save and Add Extensions

Inbound Calling

How It Works

Setting Up Inbound Calls

Inbound Call Context

Call handling setup

Troubleshooting

Best Practices

Further Reading

​Overview

​Prerequisites

​Asterisk Configuration

​Enable ARI (ari.conf)

​Enable the HTTP Server (http.conf)

​Configure the Stasis Dialplan (extensions.conf)

​Configure External Media Streaming (websocket_client.conf)

​Apply the configuration changes

​Configuration in SentiVue Talk

​Step 1: Navigate to Telephony Settings

​Step 2: Enter Your ARI Credentials

​Step 3: Save and Add Extensions

​Inbound Calling

​How It Works

​Setting Up Inbound Calls

​Inbound Call Context

​Call handling setup

​Troubleshooting

​Best Practices

​Further Reading

Overview

Prerequisites

Asterisk Configuration

Enable ARI (`ari.conf`)

Enable the HTTP Server (`http.conf`)

Configure the Stasis Dialplan (`extensions.conf`)

Configure External Media Streaming (`websocket_client.conf`)

Apply the configuration changes

Configuration in SentiVue Talk

Step 1: Navigate to Telephony Settings

Step 2: Enter Your ARI Credentials

Step 3: Save and Add Extensions

Inbound Calling

How It Works

Setting Up Inbound Calls

Inbound Call Context

Call handling setup

Troubleshooting

Best Practices

Further Reading