Jitsi Asterisk

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Hi, I'm having an issue with all my calls going out my SIP provider. I'm using Jitsi as a softphone registering to a local Asterisk PBX. I register as extension 4053 to asterisk server at (alias IP - real IP addr. Is and dial a phone number that. Check out Jitsi as a Service. Connect the users of your website or app. Get branding & tight access controls. Have notifications, transcriptions & recordings delivered straight to your backend Learn more Start & join meetings for free No account needed. Connect your calendar to view all your meetings in Jitsi. Setup and configuration of Jigasi - Jitsi’s SIP Gateway element for connecting to SIP telephony.

  1. Jitsi Jigasi Asterisk
  2. Jitsi Asterisk
  3. Jitsi Asterisk Logo
  4. Jitsi Asterisk Sip
  5. Jitsi Meet Asterisk Integration
  6. Jitsi Asterisk Setup
  7. Jitsi Asterisk Free

For this tutorial , I’m starting from the basis that you have already setup Jitsi-meet , but have not yet installed Jigasi. Also, that you have a separate instance running FreePBX (setup with a DID from a Voip provider) and can receive an inbound call, place an outbound call and that audio works both ways.

If you have audio issues or dropped calls (just using softphone… no jigasi yet) , you need to clear that up first.

Current Specs for reference -

Jitsi Server
• OS: Ubuntu Server 20.04
• Hardware: Quad-Core 2.4Ghz , 16GB DDR3, 120GB SSD

FreePBX Server
• OS: FreePBX Official SNG image: FreePBX 14 • Linux 7.6 • Asterisk 16
• Hardware: Quad-Core 2.4Ghz , 4GB DDR3, 32GB SSD

1. FreePBX - Create an extension to use with Jigasi

• FreePBX > Applications > Extensions
• +Add Extension
• Add new CHAN_SIP Extension
• User Extension: Pick an unused number (I’m using 888)
• Display Name: Call it whatever you want (I’m using Jitsi)
• Secret: Copy this, you’ll need it for Jigasi later

Pro Tip: Pause here. Register this extension with a soft phone. Direct your Inbound Route to send calls to this new extension and make sure inbound/outbound calls work. Once confirmed, remove the extension from your softphone and proceed.

2. Jitsi Server – Install Jigasi

• From terminal:

• When prompted, enter your SIP username. This will be the [email protected]
Example: [email protected]

Jitsi asterisk download

• Password will be the ‘secret’ copied from when you created the extension in step 1.

3. Add some SIP related configuration settings to Jigasi

• From terminal:

• Add the following lines under the SERVER_ADDRESS line:
Be sure to change domain and ports to match your configuration

• Add or un-comment the following line:

4. Add credentials for Jigasi

If you’re using authentication, Jigasi will need it’s own credentials.

• I’m using prosody. prosodyctl register [username] [Jitsi Domain] [Password]

• Add the new credentials to /etc/jitsi/jigasi/sip-communicator.properties
• From terminal:

• Add the following lines, replacing with the credentials you just created:
Note: password is clear text

• Restart Jigasi with the following terminal command

5. Pause here and do some testing

At this point, Jigasi’s SIP extension should be registered, let’s validate that by heading over to FreePBX…

• FreePBX > Admin > Asterisk CLI
• Run CLI command: SIP show Peers
• The extension should show “OK” if registered properly

Dialing the extension from another SIP endpoint (desk phone or softphone) should route you to the default Jitsi Room “siptest”

• Pull up a jitsi meeting via web browser , use the name: siptest
• from another registered endpoint, dial your jitsi extension (my example is 888)
• If this doesn’t work , check logs
Jigasi log - (sudo nano /var/log/jitsi/jigasi.log)
FreePBX – Reports > Asterisk logfiles

Only Proceed to step 6 if you can successfully dial into the siptest room

6. Provision Jitsi Dial in feature
Inorder for the Dial-in box to populate, Jitsi requires hosting a JSON file containing the phone numbers. I will be using the FreePBX server for this.

• Create a file titled jitsi_numbers.json, mirror contents of the file below

• Create another file titled jitsiNumberList.php, mirror contents of the file below

• Place both files on the FreePBX server under /var/www/html
I use Filezilla for linux to transfer files via ftp

• Enable CORS in Apache. On the FreePBX server, Run terminal command:

• Locate and add an Access-Control-Allow-Origin header
replace “https://meet.domain.com” with your jitsi meet domain.
Should look like this:

• Restart httpd

7. Jitsi – Edit config.js

• Provide Jitsi with the locations of our phone list and API to use
• From Terminal:


• Add the following lines.
First line, swap out “ voip.domain.com” with your FreePBX FQDN or IP
Second line is using Jitsi’s API

• At this point, joining a web-meeting should now show the share box populated with your dial-in details

Jitsi Jigasi Asterisk

8. FreePBX – Custom IVR
Now for the mystical PBX magic. We need to be able to collect the meeting pin from callers, send that pin off to an API in order to retrieve the conference room name. Then, add the conference room name in the form of a SIP header so that Jigasi knows where we want to go… and ideally, provision password entry and some other IVR-ish bells and whistles.

To begin prepping the IVR-ish application, we need to create a custom destination.
• FreePBX > Admin > Custom Destination
• Target: Jitsi-Conference-Entry,s,1

Next, we’ll use the custom destination with an inbound route
• Note: This assumes that you already have a trunk setup for inbound calling
• FreePBX > Connectivity > Inbound Routes
• DID: add the DID callers will dial to reach your conference
you should have already tested inbound calling with an extension, in which case, we’re just setting the custom destination instead of an extension destination

• Set Destination: This will be the custom destination we created above

Jitsi Asterisk

Lastly, we’ll add some Dial Plan magic

• FreePBX > Config Edit > extensions_custom.conf

• paste in the below dial plan code
• change the 888 to whatever you’re using as your Jitsi SIP extention
• exten => s,2,Set(Jitsi=888)

  1. Complete!

Jitsi Asterisk Sip

You should now have a fully functional IVR , along with dial-in options displayed in Jitsi.


• IVR – all system recordings used are included with a standard FreePBX 14 install with asterisk 16. If you’re using something else, you may need to swap out some recording names for compatibility.

• IVR - the line below uses an MOH (music on hold group) called ‘silence’ to override the first ring attempt and just play nothing. You won’t have this by default, but it’s ok… the system will use your default MOH in it’s place. If you want to use this, create an MOH group call Silence and place a blank 5 or 10 second recording in it.

Asterisk and Jitsi are running happily on the same machine.
Asterisk is accepting regular SIP calls on port 5060 and talks to jigasi on 5160.

Jitsi Meet Asterisk Integration

jigasi is also running, and successfully registers via PJSIP as an endpoint on Asterisk.

But when I try to call into a conference room from Asterisk to jigasi, whether I specify and conference room in the SIP header field X-Room_Name or not (and therefore I should go to the default room), jigasi does not appear to even respond to the Invite from Asterisk.

I have looked through other threads, but have so far been unable to resolve this issue.

Jitsi Asterisk Setup

Please can you point me in the right direction?

Jitsi Asterisk Free

All the best,