ubuntu-post-install/docs/kiosk-paging-setup.md

Auto-answer paging kiosks (baresip)

How to set up a dedicated, always-on auto-answer intercom/paging station for services/asterisk.sh, and how to mix it into a Ring Group alongside phones that ring normally.

This covers the option that already works today: a lightweight native SIP client (baresip) on a separate small Linux machine. A browser-based ("webpage") auto-answer client is possible in principle — any WebRTC SIP JS library can answer a call programmatically with none of a mobile phone's OS-level call-screen restrictions — but this server has no WebRTC/WebSocket transport configured, and building one is real new work (a transport-wss PJSIP transport, a Caddy WSS passthrough, a custom webpage). Not covered here.

What this actually is

Easy Asterisk (the vendor project services/asterisk.sh installs) ships a kiosk device category, described in its own source as "Fixed auto-answer intercoms" / "Fixed wall-mount tablets & intercoms." Installing it sets up:

• baresip — a real, fully-scriptable Linux SIP user agent, not a phone app. Its answer-mode is plain client config, so nothing about the kiosk's OS can silently override it into requiring a manual tap the way a mobile phone app often does.
• An optional push-to-talk daemon (kiosk-ptt) — the mic stays muted by default (true one-way intercom behavior) and a PTT button unmutes it to talk back.

None of this runs inside the Asterisk container. It runs on a separate machine — an old PC, a Raspberry Pi, a cheap mini-PC, anything running Ubuntu/Debian with a sound card. The vendor script explicitly refuses to install baresip when it detects it's running inside Docker (install_baresip_packages(): "Docker mode: Baresip not applicable") — that check is about wherever you happen to run the script, so make sure you're running it on the kiosk machine itself, not on the Asterisk server's droplet.

Set it up

1. Create an extension for the kiosk in the Security Dashboard's Extensions tab first (Easy Asterisk device setup section), same as any other device. Leave "Mobile" unchecked unless the kiosk is genuinely on a cellular connection — a wired/Wi-Fi kiosk on the LAN or a stable remote network doesn't need the RTP NAT-keepalive tuning that checkbox adds. Note the extension number and the auto-generated password shown once at the top of the card.

2. Get the vendor script onto the kiosk machine. It's vendored in this repo at vendor/easy-asterisk/easy-asterisk-v0.10.0.sh — copy it over (scp, a USB drive, whatever's convenient) and run it there:

   sudo bash easy-asterisk-v0.10.0.sh
   

3. Choose the client-install path (not server, not full — this machine isn't running Asterisk itself). When prompted:

   • Server (IP or domain): your Asterisk box's public IP or domain (whatever asterisk.sh set up — an FQDN here also turns on TLS automatically).
   • SIP Password: the password from step 1.
   • Answer Mode: choose 1) Auto.
   • Extension: the extension number from step 1.

   It installs baresip, PipeWire (for audio), and enables the relevant systemd services.

4. Log out and back in (or reboot) if audio doesn't work immediately — the installer adds the kiosk user to the audio group, which needs a fresh session to take effect. Check with pactl list sources short if in doubt.

5. (Optional) Configure push-to-talk from the vendor script's own menu: Main Menu → Client Management → Configure PTT Button. Without this, the kiosk just stays open-mic (fine for a simple listen-and-respond intercom; PTT is for anything noisier or more deliberate about when it transmits).

Using it in a Ring Group

No special group type needed for the mix — see the Ring Groups card's own "Ring vs Page, and mixing auto-answer devices" help text for why. Short version: create (or edit) a plain Ring-type group, add the kiosk's extension as a member alongside regular phones' extensions. Dialing the group's number rings everyone simultaneously; the kiosk answers instantly because it's configured to, ordinary phones just ring until a person picks up or the kiosk's already live. Page-type groups still exist for signaling auto-answer to devices you don't control the client config of — not needed for a kiosk you set up yourself this way.

Troubleshooting

journalctl -t kiosk-ptt -f              # PTT daemon logs, if configured
systemctl --user status baresip         # run as the kiosk's own Linux user
pactl list sources short                # confirm the mic is visible to PipeWire

If the kiosk never answers: confirm it actually registered (baresip's own CLI shows registration status), and double check the extension/password match what the dashboard generated — a stale or mistyped password here fails silently from the dashboard's side, since it has no visibility into a device that was never issued through it in the first place.