Merge pull request #257 from outis1one/claude/asterisk-caller-id-number-y1b3ls
Document Ring vs Page and auto-answer kiosks, in the dashboard itself
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Outis
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# Auto-answer paging kiosks (baresip)
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How to set up a dedicated, always-on auto-answer intercom/paging station for
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`services/asterisk.sh`, and how to mix it into a Ring Group alongside phones
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that ring normally.
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This covers the option that already works today: a lightweight native SIP
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client (`baresip`) on a separate small Linux machine. A browser-based
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("webpage") auto-answer client is possible in principle — any WebRTC SIP JS
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library can answer a call programmatically with none of a mobile phone's
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OS-level call-screen restrictions — but this server has no WebRTC/WebSocket
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transport configured, and building one is real new work (a `transport-wss`
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PJSIP transport, a Caddy WSS passthrough, a custom webpage). Not covered
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here.
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## What this actually is
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Easy Asterisk (the vendor project `services/asterisk.sh` installs) ships a
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`kiosk` device category, described in its own source as *"Fixed auto-answer
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intercoms"* / *"Fixed wall-mount tablets & intercoms."* Installing it sets
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up:
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- `baresip` — a real, fully-scriptable Linux SIP user agent, not a phone app.
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Its answer-mode is plain client config, so nothing about the kiosk's OS
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can silently override it into requiring a manual tap the way a mobile
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phone app often does.
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- An optional push-to-talk daemon (`kiosk-ptt`) — the mic stays muted by
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default (true one-way intercom behavior) and a PTT button unmutes it to
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talk back.
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None of this runs inside the Asterisk container. It runs on a **separate**
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machine — an old PC, a Raspberry Pi, a cheap mini-PC, anything running
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Ubuntu/Debian with a sound card. The vendor script explicitly refuses to
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install baresip when it detects it's running inside Docker
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(`install_baresip_packages()`: *"Docker mode: Baresip not applicable"*) —
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that check is about wherever you happen to run the script, so make sure
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you're running it on the kiosk machine itself, not on the Asterisk server's
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droplet.
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## Set it up
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1. **Create an extension for the kiosk** in the Security Dashboard's
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Extensions tab first (Easy Asterisk device setup section), same as any
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other device. Leave "Mobile" unchecked unless the kiosk is genuinely on a
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cellular connection — a wired/Wi-Fi kiosk on the LAN or a stable remote
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network doesn't need the RTP NAT-keepalive tuning that checkbox adds.
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Note the extension number and the auto-generated password shown once at
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the top of the card.
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2. **Get the vendor script onto the kiosk machine.** It's vendored in this
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repo at `vendor/easy-asterisk/easy-asterisk-v0.10.0.sh` — copy it over
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(`scp`, a USB drive, whatever's convenient) and run it there:
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```bash
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sudo bash easy-asterisk-v0.10.0.sh
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```
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3. **Choose the client-install path** (not server, not full — this machine
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isn't running Asterisk itself). When prompted:
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- **Server (IP or domain)**: your Asterisk box's public IP or domain
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(whatever `asterisk.sh` set up — an FQDN here also turns on TLS
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automatically).
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- **SIP Password**: the password from step 1.
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- **Answer Mode**: choose **1) Auto**.
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- **Extension**: the extension number from step 1.
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It installs `baresip`, PipeWire (for audio), and enables the relevant
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systemd services.
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4. **Log out and back in** (or reboot) if audio doesn't work immediately —
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the installer adds the kiosk user to the audio group, which needs a
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fresh session to take effect. Check with `pactl list sources short` if
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in doubt.
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5. **(Optional) Configure push-to-talk** from the vendor script's own menu:
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Main Menu → Client Management → Configure PTT Button. Without this, the
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kiosk just stays open-mic (fine for a simple listen-and-respond intercom;
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PTT is for anything noisier or more deliberate about when it transmits).
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## Using it in a Ring Group
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No special group type needed for the mix — see the Ring Groups card's own
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"Ring vs Page, and mixing auto-answer devices" help text for why. Short
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version: create (or edit) a plain **Ring**-type group, add the kiosk's
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extension as a member alongside regular phones' extensions. Dialing the
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group's number rings everyone simultaneously; the kiosk answers instantly
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because *it's* configured to, ordinary phones just ring until a person
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picks up or the kiosk's already live. `Page`-type groups still exist for
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signaling auto-answer to devices you don't control the client config of —
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not needed for a kiosk you set up yourself this way.
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## Troubleshooting
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```bash
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journalctl -t kiosk-ptt -f # PTT daemon logs, if configured
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systemctl --user status baresip # run as the kiosk's own Linux user
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pactl list sources short # confirm the mic is visible to PipeWire
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```
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If the kiosk never answers: confirm it actually registered (`baresip`'s own
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CLI shows registration status), and double check the extension/password
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match what the dashboard generated — a stale or mistyped password here
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fails silently from the dashboard's side, since it has no visibility into a
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device that was never issued through it in the first place.
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@@ -3209,6 +3209,12 @@ INDEX_HTML = """<!doctype html>
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<summary>Ring Groups <span class="count" id="room-count"></span></summary>
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<div class="card-body">
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<p class="muted">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.</p>
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<details class="help">
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<summary>Ring vs Page, and mixing auto-answer devices in one group</summary>
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<p class="muted"><b>Ring</b> dials every member at once — first to answer gets the call, everyone else stops ringing. <b>Page</b> 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.</p>
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<p class="muted">You don't need <b>Page</b> 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 <b>Ring</b> group already dials everyone simultaneously, so a device configured to auto-answer (e.g. a dedicated intercom/kiosk running <code>baresip</code> 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.</p>
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<p class="muted">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 <code>baresip</code>-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 <code>docs/kiosk-paging-setup.md</code> in this repo for the full walkthrough.</p>
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</details>
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<div class="row" style="margin-bottom:var(--sp-2)">
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<input type="text" id="ea-room-ext" placeholder="Extension, e.g. 500" style="width:9rem">
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<input type="text" id="ea-room-name" placeholder="Name, e.g. Sales" style="width:10rem">
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