Jitsi-Meet Jigasi FreePBX Asterisk Integration

I’ve read a lot of topics on this forum as well as some things off of github and I seem to be stuck. I have a working Jitsi-Meet Installation, working FreePBX Installation, I’ve installed Jigasi on the Jitsi-Meet deployment, but I did not use the correct settings.

My Jitsi-Meet deployment is a public deployment so I do not use authentication on it. If anyone can, can someone share a working example of Jigasi’s config file as well as a working Jigasi sip-communicator.properties?

@damencho Any Advice?

I’ve gone through the logs and something isn’t configured properly I know that much and it stems from the fact that Jigasi isn’t able to register with the SIP FreePBX Server so I need help on what a working config looks like so I can edit my configuration files accordingly.

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Can you upload logs, to see the problem. A working properties file you can find in the source code: https://github.com/jitsi/jigasi/blob/master/jigasi-home/sip-communicator.properties

@damencho I think I figured out what’s wrong and it’s half the FreePBX side and the other half Jitsi-Meet. I wasn’t getting any registrations on chan_sip in FreePBX. I’ve been able to isolate that problem and fix it. Now I need to know how to manually enter the settings properly in Jitsi-Meet Jigasi config and sip-communicator.properties.

I’ve heard that when you type the password into the jigasi automated setup that it encrypts the password, how can I manually encrypt the password to enter it into sip-communicator.properties and config files manually?

Sorry for the Double Post.
@damencho Alright there was a configuration issue with FreePBX. I’m able to get extensions registered, the problem was the freepbx firewall. I needed to whitelist an ip.

I checked the jigasi.log and figured out what I needed to do for the Jitsi-Meet deployment with Jigasi.
Apparently I had to re-encrypt/hash the password via Base64 add that into both jigasi/config and jigasi/sip-communicator.properties.

Which fixed one problem, I had to set authentication to always trusted which i’m only using Let’s Encrypt certificates on Jitsi-Meet deployment and FreePBX deployment.

I've used evilcreamsicle's jitsi_curling.sh
    #!/bin/bash
     
    jit_rm=$(curl --silent https://jitsi-api.jitsi.net/conferenceMapper?id=$1 | cut -d \, -f 3 | cut -d \: -f 2 | cut -d \" -f 2 | cut -d \@ -f 1)
    echo "SET VARIABLE JITSI \"${jit_rm}\" "

And extensions_override_freepbx.conf example modifying with my extension.
    [ext-local]
    exten => 1002,1,Set(__RINGTIMER=${IF($["${DB(AMPUSER/1002/ringtimer)}" > "0"]?${DB(AMPUSER/1002/ringtimer)}:${RINGTIMER_DEFAULT})})
    exten => 1002,2,Read(Pin,"custom/my_system_recording")
    exten => 1002,3,Verbose(result is: ${Pin})
    exten => 1002,4,AGI(jitsi_curling.sh,${Pin})
    exten => 1002,5,Verbose(result is: ${JITSI})
    exten => 1002,6,SIPAddHeader(Jitsi-Conference-Room: ${JITSI})
    exten => 1002,7,Macro(exten-vm,novm,1002,0,0,0)
    exten => 1002,8(dest),Set(__PICKUPMARK=)
    exten => 1002,9,Goto(${IVR_CONTEXT},return,1)
    exten => 1002,hint,SIP/1002&Custom:DND1002,CustomPresence:1002

But the problem is I can only call in to the siptest room the call gets answered by the freepbx server and joins the jitsi-meet siptest room as the city and state of the phone number instead of just listing the phone number.

However I cannot get Jigasi to answer the freepbx call on any other conference room other than siptest.

@damencho What did I forget? Here’s what I’ve got so far.

Jigasi’s jigasi.log (43.3 KB)
Jigasi’s sip-communicator.properties.txt (8.7 KB)
Jigasi’s config.txt (355 Bytes)

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You can open the pcap file from jigasi log folder with wireshark and check the incoming sip traffic do you see your custom header Jitsi-Conference-Room?

@damencho I honestly don’t know what i’m looking for i’m looking at the pcap in wireshark and nothing looks out of the ordinary.

I’ve got some misconfiguration but I can’t figure out why it’ll only call into siptest conference room but no other conference room.

I used evilcreamsicle’s confirguations for jitsi_curling.sh and extensions_override_freepbx.conf and I can achieve call in on siptest but no other conference room.

Do you see the Jitsi-Conference-Room header in the incoming sip INVITE?

@damencho, @nightstryke
I got presicely to same place as nightstryke, the examples provided does not realy explain how to set up to Freepbx (Asterisk) IVR.
Nor how to get Jitsi to display the dialin numbers.
In fact if someone would share a ‘Jigasi<->Freepbx setup for dummies’ it would help me a lot…

@janlov You’re not the only one who wants guide as I’ve wrapped my head around this and can’t get it to work properly.

Have you searched the forum? Not once an asterisk examples had been posted. Also explaining about the numbers, it is a simple json file with number that need to be set up in config.js

@nightstryke

I faced the same issue as you. In my case, damencho’s suggestion of checking whether the incoming sip traffic contained the correct headers showed that it did not!

In case you have created a pjsip and not legacy sip extension in FreePBX, the command SIPAddHeader(Jitsi-Conference-Room: ${JITSI}) does not work as it is deprecated for pjsip. You can use the following instead:

Set(HASH(__SIPHEADERS,Jitsi-Conference-Room)=${JITSI})
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@Planeshifter
I’m using Chan_Sip not PJSIP but that fixed my issue!

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I’m in the same boat at the moment. FreePBX answers, ask for pin, hit pound, than nothing happens.
It’s like it just hangs after running jitsi_curling.sh Any advice? Thank you.

[2020-04-02 11:46:21] VERBOSE[24686][C-0000000c] pbx.c: Executing [2000@from-did-direct:2] Read(“SIP/VoipDutilCom-0000000c”, “Pin,“custom/enter_pin””) in new stack
[2020-04-02 11:46:21] VERBOSE[24686][C-0000000c] file.c: <SIP/VoipDutilCom-0000000c> Playing ‘custom/enter_pin.slin’ (language ‘en’)
[2020-04-02 11:46:33] VERBOSE[24686][C-0000000c] app_read.c: User entered ‘2686911238’
[2020-04-02 11:46:33] VERBOSE[24686][C-0000000c] pbx.c: Executing [2000@from-did-direct:3] Verbose(“SIP/VoipDutilCom-0000000c”, “result is: 2686911238”) in new stack
[2020-04-02 11:46:33] VERBOSE[24686][C-0000000c] app_verbose.c: result is: 2686911238
[2020-04-02 11:46:33] VERBOSE[24686][C-0000000c] pbx.c: Executing [2000@from-did-direct:4] AGI(“SIP/VoipDutilCom-0000000c”, “jitsi_curling.sh,2686911238”) in new stack
[2020-04-02 11:46:33] VERBOSE[24686][C-0000000c] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/jitsi_curling.sh

/var/lib/asterisk/agi-bin/jitsi_curling.sh

#!/bin/bash

jit_rm=$(curl --silent https://jitsi-api.jitsi.net/conferenceMapper?id=1 | cut -d \, -f 3 | cut -d \: -f 2 | cut -d \" -f 2 | cut -d \@ -f 1) echo "SET VARIABLE JITSI \"{jit_rm}" "

/etc/asterisk/extensions_override_freepbx.conf

[ext-local]
exten => 2000,1,Set(__RINGTIMER={IF(["{DB(AMPUSER/2000/ringtimer)}" > "0"]?{DB(AMPUSER/2000/ringtimer)}:{RINGTIMER_DEFAULT})}) exten => 2000,2,Read(Pin,"custom/enter_pin") exten => 2000,3,Verbose(result is: {Pin})
exten => 2000,4,AGI(jitsi_curling.sh,{Pin}) exten => 2000,5,Verbose(result is: {JITSI})
exten => 2000,6,Set(HASH(__SIPHEADERS,Jitsi-Conference-Room)={JITSI}) exten => 2000,7,Macro(exten-vm,novm,2000,0,0,0) exten => 2000,8(dest),Set(__PICKUPMARK=) exten => 2000,9,Goto({IVR_CONTEXT},return,1)
exten => 2000,hint,SIP/2000&Custom:DND2000,CustomPresence:2000

@nightstryke - can you post details on how you configured your FreePBX extension? I am currently stuck at a similar point.

Incoming calls are working to the point that the room is joined (even with correct caller ID), but then the call gets dropped. I suspect a codec issue, but am not sure.

What version of jigasi/asterisk are you using?

This is beginning to drive me nuts… :frowning:

Regards,
Thomas

I’ll see if I can write something up on this as there’s not good guidelines or documentation and without @Planeshifter’s Help I would not have gotten my Jitsi-Meet/Jigasi Deployment to work with my FreePBX Deployment.

The Short Uncompiled Version:
I followed @EvilCreamsicle’s Guide Here:
http://evilcreamsicle.com/index.php/2018/03/28/jitsi-meet-w-call-in-and-active-directory/

One if you’re using the latest version of FreePBX make sure you’re using Legacy Chan_Sip, set the extension’s IP to your Jitsi-Meet/Jigasi Server’s IP.
Then you should make sure that the extension uses port 5160 in the latest FreePBX version legacy chan_sip uses port 5160.
Whitelist the Jitsi-Meet/Jigasi Server’s IP in FreePBX System Admin Intrusion Detection.
Also make sure to whitelist the IP in the Firewall of FreePBX under Networks, you should have responsive firewall on too and enabled for legacy sip.

As far as @EvilCreamsicle’s Guide when you get to:
Edit the file /etc/asterisk/extensions_override_freepbx.conf

Change it from this:
[ext-local]
exten => 9999,1,Set(__RINGTIMER=${IF($["${DB(AMPUSER/9999/ringtimer)}" > "0"]?${DB(AMPUSER/9999/ringtimer)}:${RINGTIMER_DEFAULT})})
exten => 9999,2,Read(Pin,"custom/my_system_recording")
exten => 9999,3,Verbose(result is: ${Pin})
exten => 9999,4,AGI(jitsi_curling.sh,${Pin})
exten => 9999,5,Verbose(result is: ${JITSI})
exten => 9999,6,SIPAddHeader(Jitsi-Conference-Room: ${JITSI})
exten => 9999,7,Macro(exten-vm,novm,9999,0,0,0)
exten => 9999,8(dest),Set(__PICKUPMARK=)
exten => 9999,9,Goto(${IVR_CONTEXT},return,1)
exten => 9999,hint,SIP/9999&Custom:DND9999,CustomPresence:9999

Then Change to this:
[ext-local]
exten => 9999,1,Set(__RINGTIMER=${IF($["${DB(AMPUSER/9999/ringtimer)}" > "0"]?${DB(AMPUSER/9999/ringtimer)}:${RINGTIMER_DEFAULT})})
exten => 9999,2,Read(Pin,"custom/my_system_recording")
exten => 9999,3,Verbose(result is: ${Pin})
exten => 9999,4,AGI(jitsi_curling.sh,${Pin})
exten => 9999,5,Verbose(result is: ${JITSI})
exten => 9999,6,Set(HASH(__SIPHEADERS,Jitsi-Conference-Room)=${JITSI})
exten => 9999,7,Macro(exten-vm,novm,9999,0,0,0)
exten => 9999,8(dest),Set(__PICKUPMARK=)
exten => 9999,9,Goto(${IVR_CONTEXT},return,1)
exten => 9999,hint,SIP/9999&Custom:DND9999,CustomPresence:9999

Now this is super important.
Edit your Jitsi-Meet Jigasi sip-communicator.properties at /etc/jitsi/jigasi/sip-communicator.properties
Add these lines to the file near the SIP settings.
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PROXY_PORT=5160 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PREFERRED_TRANSPORT=UDP

The reason being chan_sip by default uses UDP by default for extension whether they’re remote or local.
Also as I said earlier to make sure your extension is set to port 5160 because of FreePBX’s default settings you also have to tell Jigasi that it’s using port 5160 and not the old standard 5060 which is now used by Chan_PJSIP.

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Adding an update to my post.
Feel kind of dumb not realizing this before I posted but I have a dual nic FreePBX. One network is dedicated to our phone systems and has no public route. When the bash script was trying to query jitsi’s API (https://jitsi-api.jitsi.net/conferenceMapper?id=), it was routing out the wrong interface. So it couldn’t actually run. That problem has now been fixed. I am now working to the point of it trying to connect to jigasi, which is doing all kinds of crazy now. I have had the conference room show my phone number connect than disconnect. I have had it just ring and ring until it errors out. I’ll post an update when I know more. Thank you everyone for all the info you share.

@Michael_Brown I just formatted and edited my earlier post, read it and check your FreePBX settings.

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@nightstryke: did you do anything special with the codecs in FreePBX or jigasi?
I suspect I’m running into an issue here, that’s why I want to check if we are running the same versions of asterisk/jigasi here.

Joining the room worked for me before, but the call gets dropped immediately after that.
For joining to work, I had to change the asterisk code from

exten => 9999,6,Set(HASH(__SIPHEADERS,Jitsi-Conference-Room)=${JITSI})

to

exten => 9999,6,Set(HASH(__SIPHEADERS,Jitsi-Conference-Room)=${JITSI}@muc.meet.jitsi)

I am using a docker setup, whatever jigasi version from log:

Jigasi 2020-04-02 14:24:12.467 INFO: [10] org.jitsi.version.AbstractVersionActivator.log() VersionService registered: Jigasi 1.1.38-g8f3c241

Asterisk is a PBX in a flash setup on a raspberry Pi

Asterisk 13.21.1 built by root @ raspbx on a armv7l running Linux on 2018-09-09 20:20:03 UTC

I have checked the SIP Settings with jitsi desktop, from there I could call in both directions. So I have no idea what the actual issue is. Actually, it could be on the asterisk side, though I don’t believe it. Right after connecting, I see an error in asterisk:

**app_dial.c** : **1000** **do_forward** : Not accepting call completion offers from call-forward recipient Local/1022@from-internal-00000089;1

I could’nt really find out what that means, as I actually didn’t forward a call. Google gets few results and all seem having to do with a call forward.

More interesting is jigasis logfile, right after connecting and joining the conference there is a severe Error and then the call gets disconnected (look at timestamp 19:33:03.051):

Jigasi 2020-04-02 19:32:57.798 INFO: [1225] org.jitsi.jigasi.SipGateway.incomingCallReceived().188 Incoming call received...
Jigasi 2020-04-02 19:32:57.894 INFO: [1227] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /10.36.35.2:5160
Jigasi 2020-04-02 19:32:58.800 INFO: [1226] org.jitsi.jigasi.SipGatewaySession.run().1281 [ctx=15858487777981075833164] Wait thread cancelled
Jigasi 2020-04-02 19:32:58.803 INFO: [1225] org.jitsi.jigasi.JvbConference.setXmppProvider().539 [ctx=15858487777981075833164] Using ProtocolProviderServiceJabberImpl(Jabber:47f6172b@meet.jitsi/47f6172b)
Jigasi 2020-04-02 19:32:58.896 INFO: [1228] impl.protocol.jabber.OperationSetBasicTelephonyJabberImpl.registrationStateChanged().125 Jingle : ON 
Jigasi 2020-04-02 19:32:58.897 INFO: [1228] org.jitsi.jigasi.JvbConference.registrationStateChanged().577 [ctx=15858487777981075833164] Registering XMPP.
Jigasi 2020-04-02 19:32:58.923 INFO: [1228] impl.protocol.jabber.ProtocolProviderServiceJabberImpl.authenticated().2535 Authenticated: false
Jigasi 2020-04-02 19:32:58.926 INFO: [1228] org.jitsi.jigasi.JvbConference.joinConferenceRoom().648 [ctx=15858487777981075833164] Joining JVB conference room: test@muc.meet.jitsi
Jigasi 2020-04-02 19:32:59.023 INFO: [1234] impl.protocol.jabber.ChatRoomJabberImpl.joined().1256 test@muc.meet.jitsi/focus has joined the test@muc.meet.jitsi chat room.
Jigasi 2020-04-02 19:32:59.033 INFO: [1234] impl.protocol.jabber.ChatRoomJabberImpl.joined().1256 test@muc.meet.jitsi/d9f3cf50 has joined the test@muc.meet.jitsi chat room.
Jigasi 2020-04-02 19:33:00.003 INFO: [1249] impl.protocol.jabber.IceUdpTransportManager.createIceAgent().346 End gathering harvester within 636 ms
Jigasi 2020-04-02 19:33:01.977 INFO: [1249] impl.protocol.jabber.CallPeerMediaHandlerJabberImpl.harvestCandidates().1198 End candidate harvest within 1244 ms
Jigasi 2020-04-02 19:33:02.000 INFO: [1249] org.jitsi.jigasi.JvbConference.incomingCallReceived().1024 [ctx=15858487777981075833164] Got invite from focus
Jigasi 2020-04-02 19:33:02.027 WARNING: [1249] impl.protocol.sip.CallPeerMediaHandlerSipImpl.createMediaDescriptions().225 No active device for video was found!
Jigasi 2020-04-02 19:33:02.079 WARNING: [1260] service.protocol.media.DynamicPayloadTypeRegistry.addMapping().270 Remote party is trying to remap payload type 97 and reassign it from rtpmap:-1 speex/16000 to rtpmap:-1 iLBC/8000. We'll go along but there might be issues because of this. We'll also expect to receive rtpmap:-1 iLBC/8000 with PT=98
Jigasi 2020-04-02 19:33:02.080 INFO: [1262] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /10.36.35.2:5160
Jigasi 2020-04-02 19:33:02.086 INFO: [1262] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1204 [ctx=15858487777981075833164] SIP peer state: Connecting
Jigasi 2020-04-02 19:33:02.087 INFO: [1263] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /10.36.35.2:5160
Jigasi 2020-04-02 19:33:02.113 INFO: [1263] org.jitsi.jigasi.SipGatewaySession.handleCallState().1138 [ctx=15858487777981075833164] SIP call ended: CallPeerChangeEvent: type=CallPeerStatusChange oldV=net.java.sip.communicator.service.protocol.CallPeerState:Connecting newV=net.java.sip.communicator.service.protocol.CallPeerState:Disconnected for peer=Thomas <1022@10.36.35.2>;status=Disconnected
Jigasi 2020-04-02 19:33:02.113 INFO: [1265] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /10.36.35.2:5160
Jigasi 2020-04-02 19:33:02.127 INFO: [1263] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1204 [ctx=15858487777981075833164] SIP peer state: Disconnected
Jigasi 2020-04-02 19:33:02.437 INFO: [1261] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1003 Dynamic PT map: 126=rtpmap:-1 telephone-event/8000; 111=rtpmap:-1 opus/48000/2 fmtp:useinbandfec=1;minptime=10; 103=rtpmap:-1 unknown/90000; 
Jigasi 2020-04-02 19:33:02.437 INFO: [1261] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1020 PT overrides [103->104 ]
Jigasi 2020-04-02 19:33:02.616 INFO: [1261] service.protocol.media.CallPeerMediaHandler.start().1961 Starting
Jigasi 2020-04-02 19:33:03.009 INFO: [1260] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1003 Dynamic PT map: 101=rtpmap:-1 telephone-event/8000; 96=rtpmap:-1 speex/32000; 97=rtpmap:-1 speex/16000; 99=rtpmap:-1 speex/8000; 98=rtpmap:-1 iLBC/8000; 
Jigasi 2020-04-02 19:33:03.009 INFO: [1260] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1020 PT overrides [96->119 97->117 98->97 99->110 ]
Jigasi 2020-04-02 19:33:03.051 SEVERE: [1260] impl.protocol.sip.CallPeerSipImpl.answer().1334 Failed to create an SDP description for an OK response to an INVITE request!
java.lang.NullPointerException
	at net.java.sip.communicator.service.protocol.media.MediaHandler.configureStream(MediaHandler.java:703)
	at net.java.sip.communicator.service.protocol.media.MediaHandler.initStream(MediaHandler.java:953)
	at net.java.sip.communicator.service.protocol.media.CallPeerMediaHandler.initStream(CallPeerMediaHandler.java:1189)
	at net.java.sip.communicator.impl.protocol.sip.CallPeerMediaHandlerSipImpl.createMediaDescriptionsForAnswer(CallPeerMediaHandlerSipImpl.java:832)
	at net.java.sip.communicator.impl.protocol.sip.CallPeerMediaHandlerSipImpl.processUpdateOffer(CallPeerMediaHandlerSipImpl.java:519)
	at net.java.sip.communicator.impl.protocol.sip.CallPeerMediaHandlerSipImpl.processOffer(CallPeerMediaHandlerSipImpl.java:447)
	at net.java.sip.communicator.impl.protocol.sip.CallPeerSipImpl.answer(CallPeerSipImpl.java:1320)
	at net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.answerCallPeer(OperationSetBasicTelephonySipImpl.java:1955)
	at org.jitsi.jigasi.CallManager$AnswerCallThread.run(CallManager.java:298)
	at java.util.concurrent.Executors$RunnableAdapter.call(Executors.java:511)
	at java.util.concurrent.FutureTask.run(FutureTask.java:266)
	at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1149)
	at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:624)
	at java.lang.Thread.run(Thread.java:748)
Jigasi 2020-04-02 19:33:03.085 INFO: [1282] impl.protocol.sip.SipLogger.logInfo().196 Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /10.36.35.2:5160
Jigasi 2020-04-02 19:33:03.691 INFO: [1261] org.jitsi.jigasi.JvbConference.callStateChanged().1122 [ctx=15858487777981075833164] JVB conference call IN_PROGRESS.
Jigasi 2020-04-02 19:33:03.917 SEVERE: [1320] net.sf.fmj.media.Log.error()   Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
Jigasi 2020-04-02 19:33:03.919 SEVERE: [1320] net.sf.fmj.media.Log.error() Failed to prefetch: net.sf.fmj.media.ProcessEngine@29881665
Jigasi 2020-04-02 19:33:03.926 SEVERE: [1319] net.sf.fmj.media.Log.error() Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@29881665

Jigasi 2020-04-02 19:33:07.115 INFO: [1266] org.jitsi.jigasi.SipGatewaySession.sipCallEnded().629 [ctx=15858487777981075833164] Sip call ended: Call: id=1585848777795266673403 peers=0
Jigasi 2020-04-02 19:33:07.118 INFO: [1266] org.jitsi.jigasi.JvbConference.stop().500 [ctx=15858487777981075833164] Removing account Jabber:47f6172b@meet.jitsi/47f6172b
Jigasi 2020-04-02 19:33:07.136 INFO: [1266] impl.protocol.jabber.OperationSetBasicTelephonyJabberImpl.registrationStateChanged().132 Jingle : OFF 
Jigasi 2020-04-02 19:33:07.141 INFO: [1266] org.jitsi.jigasi.AbstractGateway.notifyCallEnded().128 [ctx=15858487777981075833164] Removed session for call. Sessions:0
Jigasi 2020-04-02 19:33:07.161 SEVERE: [1269] org.jitsi.impl.neomedia.RTPConnectorOutputStream.log() Failed to send a packet to target /172.22.0.6:10000:java.io.IOException: No active socket.
Jigasi 2020-04-02 19:33:07.173 SEVERE: [1269] org.jitsi.impl.neomedia.RTPConnectorOutputStream.log() Failed to send a packet to target /172.22.0.6:10000:java.io.IOException: No active socket.
Jigasi 2020-04-02 19:33:07.174 SEVERE: [1328] org.jitsi.jigasi.JvbConference.callStateChanged().1114 [ctx=15858487777981075833164] Call change event for different call ? Call: id=15858487793131054358458 peers=0 : null
Jigasi 2020-04-02 19:33:07.196 SEVERE: [1268] org.jitsi.impl.neomedia.RTPConnectorOutputStream.log() Failed to send a packet to target /172.22.0.6:10000:java.io.IOException: No active socket.
Jigasi 2020-04-02 19:33:07.202 SEVERE: [1328] org.jitsi.impl.neomedia.RTPConnectorOutputStream.log() Failed to send a packet to target /172.22.0.6:10000:java.io.IOException: No active socket.

For jitsi desktop, I needed to disable opus to work with my asterisk; I copied those settings over, but, well… see above.

For me, this still smells like a codec issue, but I can’t nail it down. I played a lot with codec settings, but no success so far.

Regards
Thomas

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@tbrettinger I’m not using docker so I can’t help you. I’m using straight Ubuntu 16.04 and the debian/ubuntu builds provided by jitsi.org

I hate to say it but your problem might stem from using docker, it may not have the necessary dependencies installed that a standalone Ubuntu 16.04 build of Jitsi-Meet may provide.

I’m also using a virtual machine deployment of freepbx (Version FreePBX 15.0.16.44 with Asterisk 16.9.0)

Hey Nightstryke, what do you mean by “set the extension’s IP to your jitsi-meet/jigasi server’s IP?”
Is that a setting in freepbx. I’m not sure.

I almost have this working. You and I are doing about the same setup. Virutal, Debian 10 Jitsi Server, FreePBX 15 Server. I can call and put in pin, hit pound, the conference room shows the dial in connecting but than shows it disconnected.The phone keeps ringing for a bit than dies with a busy tone.

I can share the sip.pcap is that helps?

Thank you much!