Jitsi SIP Gateway: Add Dial-In With Jigasi

You have a self-hosted Jitsi Meet server and it works fine in the browser. Then someone asks the obvious question: can people dial in from a phone? Or can the room call out to a landline for the one board member who refuses to use a laptop. That is what a SIP gateway gets you, and on Jitsi the piece that does it is Jigasi.

On a plain Ubuntu 24.04 Jitsi box the install itself is short. The part that trips people is the config: the brewery room, the XMPP control user, and getting a SIP account that actually reaches the phone network. This guide follows the current Jigasi package and the standard jitsi-meet layout, not the old build-from-source steps you still find in 2019 forum posts. Where a value depends on your own domain or provider, I have called it out so you are not copy-pasting blind.

What is Jigasi, the Jitsi SIP gateway?

Jigasi is short for Jitsi Gateway to SIP. It is a server-side application that lets regular SIP clients and phone callers join a Jitsi Meet conference. Think of it as a translator that sits between two worlds: WebRTC on the Jitsi side, SIP on the telephony side. When a phone caller connects, Jigasi joins the meeting room as a normal participant and relays audio both ways. Everyone in the browser hears the caller, and the caller hears the room.

That single component is the whole reason a Jitsi SIP gateway exists. Without it, Jitsi is browser-only. With it, you get real telephony: dial-in numbers, dial-out invites, and (as a bonus) meeting transcription, because Jigasi can also route audio to a speech-to-text engine.

Dial-in vs dial-out

People say "SIP gateway" and mean two different things. Worth splitting them, because the setup effort is not the same.

• Dial-in. A phone user calls a number and lands in a specific Jitsi room, usually after punching in a PIN. This needs a real phone number (a DID) from a provider, and a bit of routing logic to map the number or PIN to a room. More moving parts.
• Dial-out. From inside a Jitsi meeting you hit the phone icon and type a SIP address or number, and Jigasi calls out to it. This is the simpler one to stand up first, and it is what the classic "invite a SIP participant" button does.

My advice: get dial-out working end to end before you touch dial-in. It proves your Jigasi install, your brewery config, and your SIP account are all correct, without also debugging PSTN number routing at the same time. Dial-in is dial-out plus an inbound number mapping, so do the hard-to-debug part last.

Install Jigasi on Ubuntu 24.04

If Jitsi Meet came from the official Jitsi apt repository, Jigasi is in the same repo. So there is nothing new to add to your sources on Ubuntu 24.04. Update and install:

sudo apt update
sudo apt install jigasi

During install, debconf will stop and ask you two things: the SIP account username (in the form user@sip-provider.com) and its password. Enter the credentials from your SIP provider. This is the account Jigasi uses to place and receive calls. The installer writes them into the config for you, so you rarely need to hand-edit the SIP account afterward.

Configure the brewery room in Prosody

Jicofo discovers available Jigasi instances through a shared MUC (multi-user chat) room called the brewery. On a standard jitsi-meet install the internal MUC component already exists, so usually you only need to confirm it. Open your Prosody virtual host config:

sudo nano /etc/prosody/conf.d/meet.example.com.cfg.lua

Check that you have an internal MUC component like this. If it is missing, add it, and make sure both focus and jigasi are listed as admins:

Component "internal.auth.meet.example.com" "muc"
    storage = "memory"
    modules_enabled = { "ping"; }
    admins = {
        "focus@auth.meet.example.com",
        "jigasi@auth.meet.example.com"
    }
    muc_room_locking = false
    muc_room_default_public_jids = true

Restart Prosody after any change:

sudo prosodyctl restart

Register the Jigasi XMPP user

Jigasi logs into Prosody as its own user to join the brewery and the meeting rooms. The apt install usually registers this jigasi user for you, so this step is mostly a check and a fix if it did not. To create it manually (pick a strong password, this is not the SIP password):

sudo prosodyctl register jigasi auth.meet.example.com YOUR_STRONG_PASSWORD

Now tell Jigasi to use it. Open its main config file:

sudo nano /etc/jitsi/jigasi/sip-communicator.properties

Set the XMPP control account so it matches the user you just registered. The password here is Base64-encoded, not plain text. Generate it with echo -n 'YOUR_STRONG_PASSWORD' | base64 and paste the result:

org.jitsi.jigasi.xmpp.acc.USER_ID=jigasi@auth.meet.example.com
org.jitsi.jigasi.xmpp.acc.PASS=BASE64_ENCODED_PASSWORD
org.jitsi.jigasi.xmpp.acc.ANONYMOUS_AUTH=false
org.jitsi.jigasi.xmpp.acc.SERVER_ADDRESS=127.0.0.1
org.jitsi.jigasi.xmpp.acc.VColibri.BOSH_URL_PATTERN=https://meet.example.com/http-bind?room={roomName}

The same file holds the SIP account block that the installer wrote from your debconf answers. If you skipped it or need to change providers, the SIP password there is Base64 too. Do not paste a raw password into either field, Jigasi will fail to authenticate and the logs will not make it obvious why.

Point Jicofo at the Jigasi brewery

Jicofo needs to know the brewery room name so it can find Jigasi and show the dial button. Good news: the jigasi package usually sets this for you during install. So check before you edit anything.

On current Jitsi, Jicofo is configured through the HOCON file at /etc/jitsi/jicofo/jicofo.conf, not the old sip-communicator.properties. Open it:

sudo nano /etc/jitsi/jicofo/jicofo.conf

Confirm the jigasi brewery is set, using the same internal MUC domain from your Prosody config. If the block is missing, add it, nesting it inside the existing top-level jicofo block rather than starting a second one:

jicofo {
  jigasi {
    brewery-jid = "JigasiBrewery@internal.auth.meet.example.com"
  }
}

Restart Jicofo so it picks up the change:

sudo systemctl restart jicofo

Start Jigasi and reload Jitsi

The package installs Jigasi as a systemd service, so there is no ./jigasi.sh script to babysit like the old build-from-source days. Restart Jigasi and Jicofo:

sudo systemctl restart jigasi
sudo systemctl restart jicofo
sudo systemctl status jigasi

Once Jigasi is up it registers with Prosody and joins the brewery room. Reload your Jitsi Meet URL in the browser and open a meeting. You should now see a telephone icon in the toolbar (usually under the invite or more-actions menu). Click it, enter a SIP address or number, and Jigasi places the call. That is dial-out working.

Choosing a SIP provider

Jigasi is only half the picture. It speaks SIP, but something has to carry that SIP call onto the actual phone network (or to another SIP endpoint). Here is how the common options compare for a Jitsi setup.

My honest take: if you just want dial-in numbers working this week, use Twilio or Telnyx and move on. Self-hosting FreeSWITCH is the right call when you already operate a PBX or you have a hard requirement to keep voice on your own infrastructure. Otherwise you are signing up to run a whole extra service to save a few cents a minute.

Transcription, the other Jigasi mode

Worth knowing even if you came here for phones. Jigasi is not only a SIP gateway. It can also join a meeting purely to transcribe it, feeding the audio to a speech-to-text backend and dropping captions into the chat. The audio plumbing is the same, which is why one component does both jobs.

You do not need transcription to run dial-in or dial-out, and enabling it is a separate config path from the SIP account. If live captions matter to you, plan it as its own task rather than something that comes for free with the phone gateway. For teams handling sensitive calls, also look at our writeup on Jitsi Meet end-to-end encryption, because a SIP or transcription leg changes what "encrypted meeting" actually means once a call leaves the browser.

Troubleshooting the SIP gateway

Where this usually goes wrong. The install almost never breaks. The config does. Start with the logs, they are the source of truth:

sudo journalctl -u jigasi -f

• No telephone icon in the toolbar. Jicofo cannot see Jigasi. Check that jicofo.jigasi.brewery-jid in jicofo.conf matches the room Jigasi joins, confirm the jigasi XMPP user is registered, then restart jicofo and jigasi in that order. The button appears only when a Jigasi instance is present in the brewery room.
• Jigasi fails to authenticate to XMPP. Almost always a Base64 mistake. The PASS value must be the Base64 of the exact password you gave prosodyctl register. Re-encode with echo -n (the -n matters, a trailing newline breaks it) and try again.
• The call connects then drops instantly. Usually the SIP side. Confirm your SIP account is registered with the provider and that outbound calling is enabled on the trunk. Test the same credentials in a plain softphone to isolate whether it is Jigasi or the provider.
• Audio one way or no audio. Firewall or NAT on the media ports. If your Jitsi already runs a TURN server, make sure Jigasi's media can traverse it too, and that the SIP provider's RTP range is not blocked.

One more common gotcha: adding a second top-level jicofo block in jicofo.conf instead of nesting jigasi inside the existing one. HOCON will not merge them the way you expect, and your brewery setting quietly does nothing. Keep one jicofo block and put jigasi under it.

Where this fits in a bigger Jitsi setup

Adding SIP is one piece. If you are building out a full self-hosted stack, recording is usually the next request after telephony, and that is a separate component (Jibri) with its own setup, covered in our guide on setting up Jibri for Jitsi Meet recording on Ubuntu. And if you are still deciding whether to self-host at all, our overview of what Jitsi is and what it does lays out the tradeoffs before you commit a server to it.

None of this is hard once you have done it. But standing up Jitsi, wiring Jigasi, terminating SSL, and keeping Prosody, Jicofo, and the bridge in sync is a real afternoon of work, and every one of those pieces has a way to fail quietly. If you would rather skip the wiring, the Meetrix Jitsi Meet AMI ships with the core stack pre-configured so you can add the SIP gateway on a known-good base instead of debugging a fresh install at the same time.