diff --git a/docs/kiosk-paging-setup.md b/docs/kiosk-paging-setup.md new file mode 100644 index 0000000..79a8e0d --- /dev/null +++ b/docs/kiosk-paging-setup.md @@ -0,0 +1,104 @@ +# 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: + + ```bash + 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 + +```bash +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. diff --git a/services/security-dashboard.sh b/services/security-dashboard.sh index 3482b15..2bd99ca 100644 --- a/services/security-dashboard.sh +++ b/services/security-dashboard.sh @@ -3209,6 +3209,12 @@ INDEX_HTML = """ Ring Groups

Pick extensions, give them a name, and they ring (or page/auto-answer) together as one dialable extension — and can optionally be assigned a personal number below, so an inbound call to that number rings every current member.

+
+ Ring vs Page, and mixing auto-answer devices in one group +

Ring dials every member at once — first to answer gets the call, everyone else stops ringing. Page tells Asterisk to signal auto-answer to every member via SIP headers, for devices that honor it, turning the same simultaneous dial into a one-way intercom-style broadcast instead.

+

You don't need Page just to mix an auto-answering device with normally-ringing phones in the same group, though — auto-answer is really a property of the device's own SIP client configuration, not something Asterisk enforces per member. A plain Ring group already dials everyone simultaneously, so a device configured to auto-answer (e.g. a dedicated intercom/kiosk running baresip in Answer Mode: Auto) picks up instantly, while ordinary phones in the same group just keep ringing until a person answers — no extra setting needed here for that mix.

+

For a dedicated always-on auto-answer device (a wall-mounted intercom, a paging station), Easy Asterisk — the vendor project this installer builds on — has a built-in baresip-based kiosk client for exactly that. It installs on a separate small Linux machine (an old PC, a Raspberry Pi), not this Asterisk server itself. See docs/kiosk-paging-setup.md in this repo for the full walkthrough.

+