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FusionPBX Connection

Configure the connection to your FusionPBX server.

PBX Server iThe full URL to your FusionPBX web interface. Used for recording downloads via HTTPS and session verification. Must include the protocol (https://).

ESL (Event Socket Layer)

Direct TCP connection to FreeSWITCH for call control and real-time events.

Connection iESL replaces the old HTTP click-to-call method. Your Node.js backend connects to FreeSWITCH on port 8021 (TCP) and sends originate commands, subscribes to call events, and tracks call state in real-time. The PBX must have port 8021 open and firewalled to your dialer box IP (92.242.160.80).

SIP / WebRTC Softphone

Browser-based softphone using SIP.js over WebSocket.

WebSocket Configuration iSIP.js connects from the browser directly to FreeSWITCH via SIP over WebSocket Secure (WSS). This requires mod_sofia to have a ws-binding configured on port 5066 with TLS. The browser registers as a SIP extension and receives calls bridged from ESL.

Dialer Defaults

Default values for the auto-dialer queue.

Call Settings

Recording Download

How call recordings are fetched from the PBX for transcription.

Download Method iChoose how recordings are transferred from the PBX to the dialer server. HTTPS uses FusionPBX web login to download via the browser interface. SFTP connects directly to the PBX filesystem via SSH โ€” faster and more reliable, uses the exact file path from ESL RECORD_STOP events.

HTTPS Settings iAuthenticates against FusionPBX web UI and downloads recordings via /app/call_recordings/download.php. The CDR UUID from ESL events maps to the download URL. Sessions are cached for 30 minutes.

SFTP Settings iConnects via SCP/SSH directly to the PBX server. When a call ends, the ESL RECORD_STOP event provides the exact file path on the PBX (e.g. /var/lib/freeswitch/recordings/domain/uuid.wav). The dialer SCPs that exact file โ€” no searching needed. Prefer key-based auth over passwords.

Transcription

Speech-to-text and compliance analysis settings.

Vosk Engine iVosk runs locally on the dialer server โ€” no cloud APIs, no audio data leaves the box. Real-time mode streams PCM audio from the browser via WebSocket. Post-call mode transcribes downloaded WAV files. Both run compliance keyword matching.

Announcement

Auto-play a message when a call connects. The mic is muted during playback and unmuted when the announcement finishes.

Auto-Play Settings iWhen enabled, the announcement automatically plays into the call as soon as the connection is established. The operator's microphone is muted during playback and automatically unmuted when it finishes. You can also manually trigger the announcement using the Announce button in the softphone panel.

Upload Announcement iUpload a WAV or MP3 file to use as the announcement. The file will replace the current announcement. For best results, use a clear recording at 16kHz or higher sample rate.

๐Ÿ“

Drag & drop an audio file here, or click to browse

WAV or MP3, max 10MB

Preview

Listen to the current announcement audio:

User Management

Manage dialer application accounts.

My Account

Manage your password and two-factor authentication.

Change Password

Two-Factor Authentication iAdds TOTP-based 2FA using Google Authenticator, Authy, or any TOTP app. After enabling, you'll need to enter a 6-digit code from your app each time you log in.