Skip to main content

Documentation Index

Fetch the complete documentation index at: https://asisso-cd509912.mintlify.app/llms.txt

Use this file to discover all available pages before exploring further.

​
Overview

Asterisk ARI (Asterisk REST Interface) allows you to connect Asisso voice agents to your existing Asterisk PBX. ARI provides a WebSocket-based event model for controlling calls via Stasis applications, giving Asisso full control over call flow and audio streaming. This guide focuses on the Asisso-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 (version 16 or later recommended)
  • ARI module enabled in Asterisk
  • chan_websocket (WebSocket channel driver) enabled in your Asterisk build
  • Network connectivity between your Asisso instance and Asterisk
  • Asisso instance running and accessible
If you compiled Asterisk from source, ensure chan_websocket is included during the build. This module is required for external media streaming between Asterisk and Asisso. 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 Asisso. These are minimal examples focused on the Asisso integration — refer to the Asterisk documentation for full configuration details.

​
Enable ARI (ari.conf)

Create an ARI user that Asisso will use to authenticate: 
[general]
enabled = yes

[asisso]
type = user
read_only = no
password = your_secure_password
The username (section name, e.g., asisso) and password here must match the Stasis App Name and App Password you configure in Asisso.

​
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 Asisso can handle them: 
[from-external]
exten => _X.,1,NoOp(Incoming call to ${EXTEN})
 same => n,Stasis(asisso)
 same => n,Hangup()
Replace asisso with the app name you configured in ari.conf and in Asisso.

​
Configure External Media Streaming (websocket_client.conf)

Asisso uses Asterisk’s external media streaming to send and receive audio over WebSocket. Configure a WebSocket client connection that points to your Asisso instance: 
[asisso]
type = websocket_client
uri = ws://your-asisso-host:port/api/v1/telephony/ws/ari
protocols = audio
The section name (e.g., asisso) is the WebSocket Client Name you’ll enter in the Asisso telephony configuration. This name tells Asterisk which WebSocket connection to use for external media streaming during calls.
Refer to the Asterisk WebSocket documentation for additional websocket_client.conf options and TLS configuration.

​
Configuration in Asisso

​
Step 1: Navigate to Telephony Settings

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

​
Step 2: Enter Your ARI Credentials

Configure the following fields:
FieldDescriptionExample
ARI Endpoint URLHTTP base URL of your Asterisk ARI serverhttp://asterisk.example.com:8088
Stasis App NameThe ARI username configured in ari.confasisso
App PasswordThe ARI password configured in ari.confyour_secure_password
WebSocket Client NameThe connection name from websocket_client.confasisso
From ExtensionsOptional SIP extensions or trunk numbers for outbound callsPJSIP/6001 or 6001

​
Step 3: Save and Add Extensions

  1. Click Save Configuration
  2. 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.
  3. 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. Asisso automatically detects these events and activates the workflow assigned to the called extension.

​
How It Works

  1. An external call arrives at Asterisk and the dialplan routes it to Stasis(asisso)
  2. Asterisk fires a StasisStart event over the ARI WebSocket with the channel in Ring state and the dialed extension in the dialplan context
  3. Asisso looks up the called extension in your telephony configuration’s phone numbers, finds the assigned workflow, validates quota, and creates a workflow run
  4. The call is answered, bridged to an external media channel, and your voice agent 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(asisso)
 same => n,Hangup()
…or use a pattern that catches every extension you’ll register in Asisso: 
[from-external]
exten => _X.,1,NoOp(Incoming call to ${EXTEN})
 same => n,Stasis(asisso)
 same => n,Hangup()
Replace asisso with the app name you configured in ari.conf and in Asisso. Step 2: Add the extension as a phone number in Asisso
  1. Go to /telephony-configurations and open your Asterisk ARI configuration
  2. In the Phone numbers section, add a phone number whose address is the SIP extension (e.g. 8000)
  3. Set its Inbound workflow to the agent that should answer
  4. Save
    Adding the extension in Asisso doesn’t change Asterisk’s dialplan — that’s what Step 1 is for. The Asisso 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 voice agent. 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 voice agent respond.

​
Inbound Call Context

When an inbound call activates a workflow, the following context is available to your workflow:
FieldDescription
caller_numberThe caller’s phone number or extension
called_numberThe dialed number or extension
directionAlways inbound
call_idThe Asterisk channel ID
providerAlways ari

​
Troubleshooting

  • Verify the ARI endpoint URL is correct and reachable from your Asisso instance
  • 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
  • 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
  • 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 Asisso URI
  • Ensure the WebSocket Client Name in Asisso matches the section name in websocket_client.conf
  • Verify network connectivity and firewall rules allow WebSocket traffic between Asterisk and Asisso
  • 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
  • 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 Asisso logs for warnings like “no matching phone number registered for config” or “has no inbound_workflow_id assigned”
  • Check the URI in websocket_client.conf points to the correct Asisso host and port
  • Verify the Asisso instance is running and accepting WebSocket connections
  • If using TLS, ensure certificates are correctly configured on both sides

​
Best Practices

  • Keep your Asterisk instance on the same network or a low-latency connection to Asisso for optimal audio quality
  • Use strong passwords for ARI authentication
  • Restrict ARI access to known IP addresses using firewall rules
  • Monitor Asterisk logs alongside Asisso logs when debugging call issues
  • Keep Asterisk updated to the latest stable version for security and compatibility

​
Further Reading