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Contact Connect Jitsi Meet Instance to Asterisk using Jigasi (Apr 2024/ translated from German) Instructions for connecting Jitsi to Asterisk have been planned for a long time. We now have a really cool setup: Asterisk (FrePBX) connected to SIP provider and thus incoming and outgoing calls via the telephone network Linphone on the desktop and via WireGuard or OpenVPN VPN on the mobile device, always with an encrypted connection Direct, encrypted, server-to-server connections to friends via IAX2 trunks, enabling peer-to-peer telephony without third-party providers Integration of Jitsi into the Matrix Synapse Server for Element Messenger as a client The entire installation is of course on-premise in-house What is still missing is the integration of users of the public telephone network or the servers connected with a direct connection into the Jitsi conferences. We previously upgraded our current Seup from Debian Bullseye (11) to Bookworm (12). Important notes can be found in the blog. Now we expand our Jitsi setup to include Jigasi: https://github.com/jitsi/jigasi As preparation, we set up a SIP account in Asterisk/FreePBX: 99@sip.domain.tld with password superpassword It should be noted that if the Jitsi server is in the WAN and the Asterisk in the LAN, then firewall rules and NAT settings are required for communication. The DNS should also be adjusted. Debian Jitsi Repo has a package for this and we can use: apt install jigasi to install. SIP account and password must be entered as part of the installation. Then check the Asterisk LOG to see whether the SIP account registers properly. I had after setup and: systemctl restart jigasi.service systemctl restart prosody.service systemctl restart jicofo.service systemctl restart jitsi-videobridge2.service systemctl restart nginx a password error: chan_sip.c: Registration from ‘“99” sip:99@sip.domain.tld ’ failed for ‘JitsiIP:5060’ - Wrong password The password for the Jigasi account was then reset in the Asterisk and with echo -n PASSWORD | base64 - -n is really important : ) now saved as Base64 string in /etc/jitsi/jigasi/config and sip-communicator.properties. The default room should also be set there: org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME=room99@conference.jitsi.DOMAIN.com This room will be put through if no header is sent in the SIP invite. And if self-signed certificates are used: net.java.sip.communicator.service.gui.ALWAYS_TRUST_MODE_ENABLED=true must be set. Now we need a connection from Jigasi to the Prosody XMPP server, which is set up on the Jitsi server. To do this, we check or, if the entry does not exist, we add it at the end of /etc/prosody/conf.avail/jitsi.DOMAIN.com.cfg.lua: -Internal muc component for enabling Jigasi Brewery Rooms Component "internal.auth.jitsi.DOMAIN.com" "muc" storage = "memory" modules_enabled = { "ping"; } admins = { "focus@auth.jitsi.DOMAIN.com", "jigasi@auth.jitsi.DOMAIN.com" } muc_room_locking = false muc_room_default_public_jids = true Here, especially if authentication is necessary for the meeting leader, an error message like this can occur: No supported and enabled SASL Mechanism provided by server. Server announced mechanisms: [PLAIN, SCRAM-SHA-1, SCRAM-SHA-256, SCRAM-SHA-1-PLUS, SCRAM-SHA-256-PLUS]. Registered SASL mechanisms with Smack: [SASL Mech: GSSAPI, Prio: 100, SASL Mech: SCRAM-SHA-1-PLUS, Prio: 100, SASL Mech: SCRAM-SHA-1, Prio: 110, SASL Mech: DIGEST-MD5 , Prio: 200, SASL Mech: CRAM-MD5, Prio: 300, SASL Mech: PLAIN, Prio: 400, SASL Mech: X-OAUTH2, Prio: 410, SASL Mech: EXTERNAL, Prio: 500, SASL Mech: ANONYMOUS, Priority: 500]. Enabled SASL mechanisms for this connection: [ANONYMOUS]. Blacklisted SASL mechanisms: [SCRAM-SHA-1-PLUS] Then it helps to create a user prosodyctl adduser sipclient@jitsi.DOMAIN.com and in /etc/jitsi/jigasi/sip-communicator.properties enable # If you want jigasi to perform authenticated login instead of anonymous login # to the XMPP server, you can set the following properties. org.jitsi.jigasi.xmpp.acc.USER_ID=sipclient@jitsi.DOMAIN.com org.jitsi.jigasi.xmpp.acc.PASS=THE_PASSWORD org.jitsi.jigasi.xmpp.acc.ANONYMOUS_AUTH=false to solve this problem. It is important that the room must be opened before the first caller wants to join. Otherwise the line is busy. When the last web user leaves the room, all SIP Clients ended. Using it Once the room, in our case named room99, has been opened, users can call +xxYourNumber - 99 and will be put through to room99. Jitsi tip Depending on the Asterisk dial-out restrictions, users can also be added via telephone via …->Settings->Add Activate lobby function also asks dial-in users about access Asterisk tip: If outgoing calls from the Jitsi SIP account should be blocked, simply add a context in the extensions_custom.conf: [from-no-external-call] exten => 61,1,goto(from-internal,${EXTEN},1) exten => 62,1,goto(from-internal,${EXTEN},1) exten => _.,1,hangup() Here 61 and 62 must be selected internally, otherwise the call will hang up. Simply set the extension from-internal to from-no-external-call. Links Setup Jitsi Meet and Integration with Matrix Synapse Server http://wmv7y4tehgsvghaabiqvrm76uag7c6vdxufzoorjso3escefkiwo4tid.onion/Self%20Hosted%20Jitsi%20Meet%20Instance%20%28Mar%202021/German%29/ http://wmv7y4tehgsvghaabiqvrm76uag7c6vdxufzoorjso3escefkiwo4tid.onion/de/how-to-setup-your-matrix-synapse-messaging-homeserver-english/ Further instructions in the blog, especially for the LineageOS setup on the mobile device Questions? If you like this blog post, please feel free to join my public space. Enter with /join #kmj:matrix.ctseuro.com in the chat box of the Element Matrix Messenger ( https://element.io ) or with this link: https://matrix.to/#/#kmj:matrix.ctseuro.com Only ask questions in this room! Commercial customers CTS IT Solutions ( https://ctssupport.at ), which I founded, has been offering commercial customers perfect and well-tested IT solutions since 1985! The specialists will be happy to advise you on all questions relating to IT! Her Karl M. Joch http://wmv7y4tehgsvghaabiqvrm76uag7c6vdxufzoorjso3escefkiwo4tid.onion/contact/