Jitsi-Meet Jigasi + Freeswitch

hello,
i installed jitsi-meet and jigasi, jigasi is also registered to my freeswitch server. i can make calls from a jitsi-meet room to my freeswitch, but i can´t get calls from my freeswitch to the jitsi-meet room. i know i need to add a custom header with the room id. but can/should it be an additional field or should it be in the TO field of the SIP header?
maybe somebody with a freeswitch could post here an example of the freeswitch dialplan to make a call from a freeswitch to a jitsi-meet room.
thank you so much!
thomas

net.java.sip.communicator.impl.protocol.sip.acc1.JITSI_MEET_ROOM_HEADER_NAME=X-Room-Name
[...]
net.java.sip.communicator.impl.protocol.sip.SKIP_REINVITE_ON_FOCUS_CHANGE_PROP=true

in jigasi/sip-communicator.properties and then:

  <extension name="conference_jitsi">
    <condition field="destination_number" expression="^[number reserved for jitsi]$">
      <action application="set" data="inherit_codec=true"/>
      <action application="set" data="proxy_media=true"/>
      <action inline="true" application="set" data="sip_h_X-Room-Name=[room-name]@muc.meet.jitsi"/>
      <action application="bridge" data="user/jitsi@${domain_name}"/>
    </condition>
  </extension>

Replace [number reserved for jitsi] and [room-name] with actual number and name. Later you will probably use some IVR for meeting number to room name mapping.

BTW: inherit_codec and proxy_media came from tests with codec related issues. They are not necessary and proxy_media must not be enabled if you want to use FS IVR.

hello,
thank you very much for posting your dialplan to make an inbound call to jitsi-meet. i did try it, but my calls are still getting in the siptest room and not in the destination conference room. in the sip-log i can see the required header field, but jigasi doesnt interpret the header in the correct way. here is the pcap log from the inbound call:

Via: SIP/2.0/UDP 185.164.4.41;rport=5060;branch=z9hG4bK20r4X0g7D936S;received=185.164.4.41
Route: sip:2036@193.170.143.130:47744;transport=udp;registering_acc=sip5_peterseil_com
Max-Forwards: 69
From: “Thomas DECT” sip:1038@185.164.4.41;tag=Fr1QZrBm1740c
To: sip:2036@10.1.1.130:5060;transport=udp;registering_acc=sip5_peterseil_com
Call-ID: 6326c5d8-f5de-1238-42b6-339c5c4a0bb5
CSeq: 18711521 INVITE
Contact: sip:mod_sofia@185.164.4.41:5060
User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE
Supported: timer,path,replaces
Allow-Events: talk,hold,conference,presence,as-feature-event,dialog,line-seize,call-info,sla,include-session-description,presence.winfo,message-summary,refer
Content-Type: application/sdp
Content-Disposition: session
X-Room-Name: thomas3@muc.meet.jitsi
X-FS-Support: update_display,send_info
Remote-Party-ID: “Thomas DECT” sip:1038@185.164.4.41;party=calling;screen=yes;privacy=off
Content-Length: 400

v=0
o=FreeSWITCH 1945466611 1945466612 IN IP4 185.164.4.41
s=FreeSWITCH
c=IN IP4 185.164.4.41
t=0 0
m=audio 16634 RTP/AVP 8 0 2 102 100 99 97 101 18
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=rtcp:7083
a=ptime:20
B��^`�^K^@^X^B^@^@^X^B^@^@^@^@^@^@^@^@^@^@^@^@^@^@^H^@E^@^B
�^T^@^@5^Q�^A^@^@^@^@��^D)^S�^S�^A�^@^@SIP/2.0 180 Ringing
CSeq: 18711521 INVITE
Call-ID: 6326c5d8-f5de-1238-42b6-339c5c4a0bb5
From: “Thomas DECT” sip:1038@185.164.4.41;tag=Fr1QZrBm1740c
To: sip:2036@10.1.1.130:5060;transport=udp;registering_acc=sip5_peterseil_com;tag=89ba5150
Via: SIP/2.0/UDP 185.164.4.41;rport=5060;branch=z9hG4bK20r4X0g7D936S;received=185.164.4.41
Contact: “2036” sip:2036@10.1.1.130:5060;transport=udp;registering_acc=sip5_peterseil_com
User-Agent: Jigasi1.1.38-g8f3c241Linux
Content-Length: 0

C��^(^K^D^@^X^B^@^@^X^B^@^@^@^@^@^@^@^@^@^@^@^@^@^@^H^@E^@^B
�^T^@^@5^Q�^A^@^@^@^@��^D)^S�^S�^A�^@^@SIP/2.0 180 Ringing
CSeq: 18711521 INVITE
Call-ID: 6326c5d8-f5de-1238-42b6-339c5c4a0bb5
From: “Thomas DECT” sip:1038@185.164.4.41;tag=Fr1QZrBm1740c
To: sip:2036@10.1.1.130:5060;transport=udp;registering_acc=sip5_peterseil_com;tag=89ba5150
Via: SIP/2.0/UDP 185.164.4.41;rport=5060;branch=z9hG4bK20r4X0g7D936S;received=185.164.4.41
Contact: “2036” sip:2036@10.1.1.130:5060;transport=udp;registering_acc=sip5_peterseil_com
User-Agent: Jigasi1.1.38-g8f3c241Linux
Content-Length: 0

thank you very much for your help!
thomas

Can you post the jigasi log during a dial in try?

hello,
here is the jigasi.log:
2020-04-17 21:16:32.687 INFORMATION: [66] org.jitsi.jigasi.SipGateway.incomingCallReceived().196 [ctx=158715099267961409378] Incoming call received…
2020-04-17 21:16:33.737 INFORMATION: [68] org.jitsi.jigasi.SipGatewaySession.run().1464 [ctx=158715099267961409378]Using default JVB room name property siptest
2020-04-17 21:16:33.742 INFORMATION: [68] org.jitsi.jigasi.JvbConference.start().422 [ctx=158715099267961409378] Starting JVB conference room: siptest
2020-04-17 21:16:33.939 INFORMATION: [68] org.jitsi.jigasi.JvbConference.setXmppProvider().553 [ctx=158715099267961409378] Using ProtocolProviderServiceJabberImpl(Jabber:26bd25fb@meet.cooltools.education/26bd25fb)
2020-04-17 21:16:34.140 INFORMATION: [71] impl.protocol.jabber.OperationSetBasicTelephonyJabberImpl.registrationStateChanged().127 Jingle : ON
2020-04-17 21:16:34.140 INFORMATION: [71] org.jitsi.jigasi.JvbConference.registrationStateChanged().606 [ctx=158715099267961409378] Registering XMPP.
2020-04-17 21:16:34.161 INFORMATION: [71] impl.protocol.jabber.ProtocolProviderServiceJabberImpl.authenticated().2535 Authenticated: false
2020-04-17 21:16:34.174 INFORMATION: [71] org.jitsi.jigasi.JvbConference.joinConferenceRoom().685 [ctx=158715099267961409378] Joining JVB conference room: siptest
2020-04-17 21:16:34.334 INFORMATION: [76] impl.protocol.jabber.ChatRoomJabberImpl.joined().1256 siptest@conference.meet.cooltools.education/focus has joined the siptest@conference.meet.cooltools.education chat room.
2020-04-17 21:16:36.748 INFORMATION: [92] org.jitsi.jigasi.SipGatewaySession.handleCallState().1318 [ctx=158715099267961409378] SIP call ended: CallPeerChangeEvent: type=CallPeerStatusChange oldV=net.java.sip.communicator.serv$
2020-04-17 21:16:36.749 INFORMATION: [92] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1384 [ctx=158715099267961409378] SIP peer state: Disconnected

it seems to me that jigasi doesnt collect any information from the sip-header, it just sends the call to the default room (siptest).
thank you so much for your help!
thomas

Can you please double check the sip.properties JITSI_MEET_ROOM_HEADER_NAME=X-Room-Name setting (or post the relevant part of your sip.properties)

Is this pcap from the jigasi incoming interface? If so, you have to find why jigasi does not recognise the room name header. If not, make a pcap there and check. Maybe some provider remove custom headers.

Hello
I use freeswitch to connect with jigasi, use softphone to dial into the video conference room through freeswitch, and the call quality is normal, but I forward it to freeswitch through the trunk line, and after using the phone number to dial into the conference room, there will be a buzzing sound
The information after dialing in is as follows:
m=audio 65448 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Hi, following the instructions above, editing sip-communicator.properties and adding an extension to bridge the jigasi works well. However, it will only work on the default siptest room (org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME=siptest). Is there a way that we can define and get the header so that any room on jigasi could allow them to join?

Thanks,