Re: [bigbluebutton-setup] Problem with WebCRT Error 1004

[Fred Dixon]

Hi Adam,

Can you try 

  http://demo.bigbluebutton.org/

and confirm you can join the audio there.  On your server, can you post the output of the command

  sudo bbb-conf --check

Regards,... Fred

On Wed, Jan 27, 2016 at 3:34 AM, adam feit <adamber...@gmail.com> wrote:

Hi, I need help.
After installing BBB me only works Flash audio.

Error that reports the BBB is: Detected seuraavilla the WebRTC issue: Error 1004: Failure on call (reason = Failure SIP Code). Do you want to try Flash instead?
In the console: Tue Jan 26 2016 3:54:35 p.m. GMT + 0100 | sip.inviteclientcontext | Error: Attempted to send a BYE in the session terminated.

Please help me
Adam

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[adam feit]

Hi Fred, if I connect to the demo, so I sound functioning normally.

[adam feit]

My system runs on HTTP. Not a problem, I do not have HTTPS?

[Fred Dixon]

Hi Adam,

Your going to want to setup HTTPS, see

  http://docs.bigbluebutton.org/1.0/10install.html#configuring-https-on-bigbluebutton

Also, we recommend you install BigBlueButton 1.0-beta as it should be easier to setup with HTTPS.  See

  http://docs.bigbluebutton.org/1.0/10install.html

Regards,... Fred

  

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[adam feit]

But i need this system on HTTP without SSL certificated. Its possible?

On Thursday, January 28, 2016 at 2:14:07 PM UTC+1, Fred Dixon wrote:

Hi Adam,

Your going to want to setup HTTPS, see

  http://docs.bigbluebutton.org/1.0/10install.html#configuring-https-on-bigbluebutton

Also, we recommend you install BigBlueButton 1.0-beta as it should be easier to setup with HTTPS.  See

  http://docs.bigbluebutton.org/1.0/10install.html

Regards,... Fred

  

On Thu, Jan 28, 2016 at 7:13 AM, adam feit <adamber...@gmail.com> wrote:

My system runs on HTTP. Not a problem, I do not have HTTPS?

On Thursday, January 28, 2016 at 10:00:12 AM UTC+1, adam feit wrote:

Hi Fred, if I connect to the demo, so I sound functioning normally.

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[Fred Dixon]

Hi Adam,

> But i need this system on HTTP without SSL certificated. Its possible?

Your going to want to upgrade your system to use HTTPS; otherwise, users of Chrome 47 (and above) will no longer be able to use WebRTC for audio.

What version of BigBlueButton server are you running?

Regards,... Fred

On Thu, Jan 28, 2016 at 8:32 AM, adam feit <adamber...@gmail.com> wrote:

But i need this system on HTTP without SSL certificated. Its possible?

On Thursday, January 28, 2016 at 2:14:07 PM UTC+1, Fred Dixon wrote:

Hi Adam,

Your going to want to setup HTTPS, see

  http://docs.bigbluebutton.org/1.0/10install.html#configuring-https-on-bigbluebutton

Also, we recommend you install BigBlueButton 1.0-beta as it should be easier to setup with HTTPS.  See

  http://docs.bigbluebutton.org/1.0/10install.html

Regards,... Fred

  

On Thu, Jan 28, 2016 at 7:13 AM, adam feit <adamber...@gmail.com> wrote:

My system runs on HTTP. Not a problem, I do not have HTTPS?

On Thursday, January 28, 2016 at 10:00:12 AM UTC+1, adam feit wrote:

Hi Fred, if I connect to the demo, so I sound functioning normally.

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[adam feit]

I have version 1.0 BETA and on second server 0.9.

Yesterday me worked normally webRTC, but today is problem...

I try connect to webRTC and system say: Detected the following WebRTC issue: Error 1004: Failure on call (reason=SIP Failure Code). Do you want to try Flash instead?

In the console is error: Thu Jan 28 2016 14:56:07 GMT+0100 | sip.inviteclientcontext | Error: Attempted to send BYE in a terminated session.

On Thursday, January 28, 2016 at 2:51:30 PM UTC+1, Fred Dixon wrote:

Hi Adam,

> But i need this system on HTTP without SSL certificated. Its possible?

Your going to want to upgrade your system to use HTTPS; otherwise, users of Chrome 47 (and above) will no longer be able to use WebRTC for audio.

What version of BigBlueButton server are you running?

Regards,... Fred

On Thu, Jan 28, 2016 at 8:32 AM, adam feit <adamber...@gmail.com> wrote:

But i need this system on HTTP without SSL certificated. Its possible?

On Thursday, January 28, 2016 at 2:14:07 PM UTC+1, Fred Dixon wrote:

Hi Adam,

Your going to want to setup HTTPS, see

  http://docs.bigbluebutton.org/1.0/10install.html#configuring-https-on-bigbluebutton

Also, we recommend you install BigBlueButton 1.0-beta as it should be easier to setup with HTTPS.  See

  http://docs.bigbluebutton.org/1.0/10install.html

Regards,... Fred

  

On Thu, Jan 28, 2016 at 7:13 AM, adam feit <adamber...@gmail.com> wrote:

My system runs on HTTP. Not a problem, I do not have HTTPS?

On Thursday, January 28, 2016 at 10:00:12 AM UTC+1, adam feit wrote:

Hi Fred, if I connect to the demo, so I sound functioning normally.

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[Fred Dixon]

Hi Adam,

> I have version 1.0 BETA and on second server 0.9.

Cool.  A few questions:

1. Did both initially work without the WebRTC error?

2. Are both of these servers giving you the same error?  (If so, did they both start giving you the same error around the same time).

3. Have you made any custom changes to the server?  (If so, can you back out the changes and see if WebRTC audio works).

Regards,... Fred

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[adam feit]

Both servers give me the same error. I did not make any changes. Even the 1.0 version I installed only today.

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[Chad Pilkey]

The console error that you pasted is just a message the follows the actual failure cause. The real cause should be higher up in the log.

[adam feit]

Hi all, I keep failing to deploy WebRTC ever ... I do not know what to do

On Wednesday, January 27, 2016 at 2:21:20 PM UTC+1, Fred Dixon wrote:

Hi Adam,

Can you try 

  http://demo.bigbluebutton.org/

and confirm you can join the audio there.  On your server, can you post the output of the command

  sudo bbb-conf --check

Regards,... Fred

On Wed, Jan 27, 2016 at 3:34 AM, adam feit <adamber...@gmail.com> wrote:

Hi, I need help.
After installing BBB me only works Flash audio.

Error that reports the BBB is: Detected seuraavilla the WebRTC issue: Error 1004: Failure on call (reason = Failure SIP Code). Do you want to try Flash instead?
In the console: Tue Jan 26 2016 3:54:35 p.m. GMT + 0100 | sip.inviteclientcontext | Error: Attempted to send a BYE in the session terminated.

Please help me
Adam

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[Fred Dixon]

Hi Adam,

> In the console: Tue Jan 26 2016 3:54:35 p.m. GMT + 0100 | sip.inviteclientcontext | Error: Attempted to send a BYE in the session terminated.

...

> The console error that you pasted is just a message the follows the actual failure cause. The real cause should be higher up in the log.

Did you check higher up on the console logs?  Try posting the full console log if it's unclear to the error..

Regards,... Fred

On Fri, Feb 5, 2016 at 6:26 AM, adam feit <adamber...@gmail.com> wrote:

Hi all, I keep failing to deploy WebRTC ever ... I do not know what to do

On Wednesday, January 27, 2016 at 2:21:20 PM UTC+1, Fred Dixon wrote:

Hi Adam,

Can you try 

  http://demo.bigbluebutton.org/

and confirm you can join the audio there.  On your server, can you post the output of the command

  sudo bbb-conf --check

Regards,... Fred

On Wed, Jan 27, 2016 at 3:34 AM, adam feit <adamber...@gmail.com> wrote:

Hi, I need help.
After installing BBB me only works Flash audio.

Error that reports the BBB is: Detected seuraavilla the WebRTC issue: Error 1004: Failure on call (reason = Failure SIP Code). Do you want to try Flash instead?
In the console: Tue Jan 26 2016 3:54:35 p.m. GMT + 0100 | sip.inviteclientcontext | Error: Attempted to send a BYE in the session terminated.

Please help me
Adam

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[adam feit]

Hi, this is my whole extract from the console:


An unbalanced tree was written using document.write() causing data from the network to be reparsed. For more information https://developer.mozilla.org/en/Optimizing_Your_Pages_for_Speculative_Parsing BigBlueButton.html:27:0
BigBlueButton SWF is ready. bbb_api_bridge.js:596:7
Successfully connected on test connection. {"server":"77.93.202.240","userId":"8bdphrd5eisg_1","port":":1935","meetingName":"Webinar","app":"video/portTest","protocol":"RTMP","meetingId":"81a5899ad9d8a70a9b52a50c05fbdb4ff300c4b1-1454679968043","username":"admin"} bbblogger.js:23:7
Successfully connected to BBB App. {"user":{"userId":"8bdphrd5eisg_2","externalMeetingId":"77379d1db58c9ac8c4892fdd4d8a15dfab5e4d2f","meetingName":"Webinar","meetingId":"81a5899ad9d8a70a9b52a50c05fbdb4ff300c4b1-1454679968043","userName":"admin"}} bbblogger.js:23:7
Received [UserJoinedEvent] bbb_api_bridge.js:567:7
No listeners for [UserJoinedEvent] bbb_api_bridge.js:543:13
Received [SwitchedPresenterEvent] bbb_api_bridge.js:567:7
No listeners for [SwitchedPresenterEvent] bbb_api_bridge.js:543:13
Received [NewRoleEvent] bbb_api_bridge.js:567:7
No listeners for [NewRoleEvent] bbb_api_bridge.js:543:13
Received [UserJoinedEvent] bbb_api_bridge.js:567:7
No listeners for [UserJoinedEvent] bbb_api_bridge.js:543:13
Received [SwitchedLayoutEvent] bbb_api_bridge.js:567:7
No listeners for [SwitchedLayoutEvent] bbb_api_bridge.js:543:13
Joining the audio test first bbb_webrtc_bridge_sip.js:81:2
User info callback [myUserID=8bdphrd5eisg,myUsername=admin,myAvatarURL=http://77.93.202.240/client/avatar.png,myRole=MODERATOR,amIPresenter=true,dialNumber=613-555-1234,voiceBridge=79974]. bbb_webrtc_bridge_sip.js:43:1
8bdphrd5eisg-bbbID-admin bbb_webrtc_bridge_sip.js:55:4
user 8bdphrd5eisg-bbbID-admin calling to 919679974 bbb_webrtc_bridge_sip.js:219:2
Fetching STUN/TURN server info for user agent bbb_webrtc_bridge_sip.js:129:2
Creating new user agent bbb_webrtc_bridge_sip.js:154:2
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | configuration parameters after validation: sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · viaHost: "2qkrv0lbe4sv.invalid" sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · uri: sip:8bdphrd5eisg...@77.93.202.240 sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · wsServers: [{"ws_uri":"ws://77.93.202.240/ws","sip_uri":"<sip:77.93.202.240;transport=ws;lr>","weight":0,"status":0,"scheme":"WS"}] sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · password: NOT SHOWN sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · registerExpires: 600 sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · register: false sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · registrarServer: sip:77.93.202.240 sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · wsServerMaxReconnection: 3 sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · wsServerReconnectionTimeout: 4 sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · connectionRecoveryMinInterval: 2 sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · connectionRecoveryMaxInterval: 30 sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · usePreloadedRoute: false sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · userAgentString: "BigBlueButton" sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · noAnswerTimeout: 60000 sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · stunServers: ["stun:stun.freeswitch.org"] sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · turnServers: [] sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · traceSip: true sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · hackViaTcp: false sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · hackIpInContact: false sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · hackWssInTransport: false sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · autostart: false sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · rel100: "none" sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · mediaHandlerFactory: function defaultFactory(session, options) {
  return new MediaHandler(session, options);
} sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · authenticationFactory: undefined sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · displayName: "8bdphrd5eisg-bbbID-admin" sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · instanceId: "34c7f7d7-872e-444c-afaa-ad99bb79d00c" sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · sipjsId: "3pemc" sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · hostportParams: "77.93.202.240" sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · authorizationUser: "8bdphrd5eisg-bbbID-admin" sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · media: undefined sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event connecting sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event connected sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event disconnected sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event newTransaction sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event transactionDestroyed sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event registered sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event unregistered sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event registrationFailed sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event invite sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event newSession sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event message sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event ack sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event cancel sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event bye sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event options sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event info sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | adding event notify sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.clientcontext | adding event progress sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.clientcontext | adding event accepted sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.clientcontext | adding event rejected sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.clientcontext | adding event failed sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.clientcontext | adding event cancel sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.registercontext | adding event registered sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.registercontext | adding event unregistered sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.registercontext | new listener added to event failed sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.registercontext | new listener added to event registered sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.registercontext | new listener added to event unregistered sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | new listener added to event connected sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | new listener added to event disconnected sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | user requested startup... sip.js:2636:3
"Fri Feb 05 2016 14:46:21 GMT+0100 | sip.transport | connecting to WebSocket ws://77.93.202.240/ws" sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | emitting event connecting sip.js:2636:3
"Fri Feb 05 2016 14:46:21 GMT+0100 | sip.transport | WebSocket ws://77.93.202.240/ws connected" sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | connection state set to 0 sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | emitting event connected sip.js:2636:3
Use of document.createAttribute() is deprecated. Use element.setAttribute() instead. imtranslatorOverlay.js:2352:10
Use of setAttributeNode() is deprecated. Use setAttribute() instead. imtranslatorOverlay.js:2354:1
getUserMicMedia: success bbb_webrtc_bridge_sip.js:235:5
Setting options..  bbb_webrtc_bridge_sip.js:271:2
Calling to 919679974.... bbb_webrtc_bridge_sip.js:298:2
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event progress sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event accepted sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event rejected sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event failed sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event cancel sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event connecting sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event terminated sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event dtmf sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event invite sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | skipping event cancel - Event exists sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event refer sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event bye sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event hold sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event unhold sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event muted sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.clientcontext | adding event unmuted sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event userMediaRequest sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event userMedia sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event userMediaFailed sip.js:2636:3
RTCIceServer.url is deprecated! Use urls instead. <unknown>
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event userMediaRequest sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event userMedia sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event userMediaFailed sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event iceGathering sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event iceComplete sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event iceConnected sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event iceCompleted sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event iceFailed sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event iceDisconnected sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event iceClosed sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event getDescription sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event setDescription sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event dataChannel sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | adding event addStream sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | new listener added to event userMediaRequest sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | new listener added to event userMedia sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | new listener added to event userMediaFailed sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | acquiring local media sip.js:2636:3
call connecting bbb_webrtc_bridge_sip.js:303:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.inviteclientcontext | new listener added to event connecting sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.inviteclientcontext | new listener added to event progress sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.inviteclientcontext | new listener added to event failed sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.inviteclientcontext | new listener added to event bye sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.inviteclientcontext | new listener added to event accepted sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | new listener added to event iceFailed sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | new listener added to event iceConnected sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | new listener added to event iceCompleted sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | acquired local media streams sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.inviteclientcontext | emitting event connecting sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | ICE candidate received: candidate:0 1 UDP 2122252543 2a00:1028:8d1f:afba:5124:4d55:b0a6:f919 60166 typ host sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | ICE candidate received: candidate:2 1 UDP 2122187007 2a00:1028:8d1f:afba:6127:7a8c:395b:8f78 60167 typ host sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | ICE candidate received: candidate:4 1 UDP 2122055935 10.0.0.33 60168 typ host sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | ICE candidate received: candidate:6 1 UDP 2122121471 2001:0:9d38:6abd:24b5:3947:aab8:1411 60169 typ host sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | ICE candidate received: candidate:0 2 UDP 2122252542 2a00:1028:8d1f:afba:5124:4d55:b0a6:f919 54654 typ host sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | ICE candidate received: candidate:2 2 UDP 2122187006 2a00:1028:8d1f:afba:6127:7a8c:395b:8f78 54655 typ host sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | ICE candidate received: candidate:4 2 UDP 2122055934 10.0.0.33 54656 typ host sip.js:2636:3
Fri Feb 05 2016 14:46:23 GMT+0100 | sip.invitecontext.mediahandler | ICE candidate received: candidate:6 2 UDP 2122121470 2001:0:9d38:6abd:24b5:3947:aab8:1411 54657 typ host sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.invitecontext.mediahandler | ICE candidate received: candidate:5 1 UDP 1685856255 85.71.235.238 60168 typ srflx raddr 10.0.0.33 rport 60168 sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.invitecontext.mediahandler | ICE candidate received: candidate:5 2 UDP 1685856254 85.71.235.238 54656 typ srflx raddr 10.0.0.33 rport 54656 sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.invitecontext.mediahandler | ICE Gathering Completed sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.invitecontext.mediahandler | emitting event iceComplete sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.invitecontext.mediahandler | emitting event getDescription sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.ua | emitting event newTransaction sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transaction.ict | adding event stateChanged sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transaction.ict | emitting event stateChanged sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transport | sending WebSocket message:

INVITE sip:9196...@77.93.202.240 SIP/2.0

Via: SIP/2.0/WS 2qkrv0lbe4sv.invalid;branch=z9hG4bK5161837

Max-Forwards: 70

To: <sip:9196...@77.93.202.240>

From: "8bdphrd5eisg-bbbID-admin" <sip:8bdphrd5eisg...@77.93.202.240>;tag=a1ju6ihm99

Call-ID: 3pemc65322ma1n39sgba

CSeq: 2436 INVITE

Contact: <sip:19c3...@2qkrv0lbe4sv.invalid;transport=ws;ob>

Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY

Content-Type: application/sdp

Supported: outbound

User-Agent: BigBlueButton

Content-Length: 1625



v=0

o=mozilla...THIS_IS_SDPARTA-43.0.4 8574080507790804439 0 IN IP4 0.0.0.0

s=-

t=0 0

a=sendrecv

a=fingerprint:sha-256 50:2C:32:01:2E:B6:23:A8:6A:D8:84:7D:14:8E:38:B4:3C:94:89:3E:03:FB:90:84:D5:A5:48:18:E9:B5:96:B8

a=ice-options:trickle

a=msid-semantic:WMS *

m=audio 60168 UDP/TLS/RTP/SAVPF 109 9 0 8

c=IN IP4 85.71.235.238

a=candidate:0 1 UDP 2122252543 2a00:1028:8d1f:afba:5124:4d55:b0a6:f919 60166 typ host

a=candidate:2 1 UDP 2122187007 2a00:1028:8d1f:afba:6127:7a8c:395b:8f78 60167 typ host

a=candidate:4 1 UDP 2122055935 10.0.0.33 60168 typ host

a=candidate:6 1 UDP 2122121471 2001:0:9d38:6abd:24b5:3947:aab8:1411 60169 typ host

a=candidate:0 2 UDP 2122252542 2a00:1028:8d1f:afba:5124:4d55:b0a6:f919 54654 typ host

a=candidate:2 2 UDP 2122187006 2a00:1028:8d1f:afba:6127:7a8c:395b:8f78 54655 typ host

a=candidate:4 2 UDP 2122055934 10.0.0.33 54656 typ host

a=candidate:6 2 UDP 2122121470 2001:0:9d38:6abd:24b5:3947:aab8:1411 54657 typ host

a=candidate:5 1 UDP 1685856255 85.71.235.238 60168 typ srflx raddr 10.0.0.33 rport 60168

a=candidate:5 2 UDP 1685856254 85.71.235.238 54656 typ srflx raddr 10.0.0.33 rport 54656

a=sendrecv

a=end-of-candidates

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level

a=ice-pwd:5ba7bb632fd29650ee066bae34410cc7

a=ice-ufrag:81f42080

a=mid:sdparta_0

a=msid:{ed1d6b33-cc96-4bf1-bb9d-7052c4ce50c1} {72edb597-6100-47d1-86f1-eac18fd0b231}

a=rtcp:54656 IN IP4 85.71.235.238

a=rtcp-mux

a=rtpmap:109 opus/48000/2

a=rtpmap:9 G722/8000/1

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=setup:actpass

a=ssrc:1993978282 cname:{499c9067-489e-4476-8976-bdfa44303870}


 sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transport | received WebSocket text message:

SIP/2.0 100 Trying

Via: SIP/2.0/WS 2qkrv0lbe4sv.invalid;branch=z9hG4bK5161837;received=77.93.202.240;rport=58539

From: "8bdphrd5eisg-bbbID-admin" <sip:8bdphrd5eisg...@77.93.202.240>;tag=a1ju6ihm99

To: <sip:9196...@77.93.202.240>

Call-ID: 3pemc65322ma1n39sgba

CSeq: 2436 INVITE

User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150327T050105Z~f2a5271009~64bit

Content-Length: 0




 sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transaction.ict | emitting event stateChanged sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.inviteclientcontext | emitting event progress sip.js:2636:3
call progress: SIP/2.0 100 Trying

Via: SIP/2.0/WS 2qkrv0lbe4sv.invalid;branch=z9hG4bK5161837;received=77.93.202.240;rport=58539

From: "8bdphrd5eisg-bbbID-admin" <sip:8bdphrd5eisg...@77.93.202.240>;tag=a1ju6ihm99

To: <sip:9196...@77.93.202.240>

Call-ID: 3pemc65322ma1n39sgba

CSeq: 2436 INVITE

User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150327T050105Z~f2a5271009~64bit

Content-Length: 0



 bbb_webrtc_bridge_sip.js:314:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transport | received WebSocket text message:

SIP/2.0 502 Bad Gateway

Via: SIP/2.0/WS 2qkrv0lbe4sv.invalid;branch=z9hG4bK5161837;received=77.93.202.240;rport=58539

Max-Forwards: 69

From: "8bdphrd5eisg-bbbID-admin" <sip:8bdphrd5eisg...@77.93.202.240>;tag=a1ju6ihm99

To: <sip:9196...@77.93.202.240>;tag=4jQ5j12X5HZ9g

Call-ID: 3pemc65322ma1n39sgba

CSeq: 2436 INVITE

User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150327T050105Z~f2a5271009~64bit

Accept: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, refer

Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"

Content-Length: 0

Remote-Party-ID: "919679974" <sip:9196...@77.93.202.240>;party=calling;privacy=off;screen=no




 sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transaction.ict | emitting event stateChanged sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transport | sending WebSocket message:

ACK sip:9196...@77.93.202.240 SIP/2.0

Via: SIP/2.0/WS 2qkrv0lbe4sv.invalid;branch=z9hG4bK5161837

To: <sip:9196...@77.93.202.240>;tag=4jQ5j12X5HZ9g

From: "8bdphrd5eisg-bbbID-admin" <sip:8bdphrd5eisg...@77.93.202.240>;tag=a1ju6ihm99

Call-ID: 3pemc65322ma1n39sgba

CSeq: 2436 ACK




 sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.inviteclientcontext | closing INVITE session 3pemc65322ma1n39sgbaadhef1e9jj sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.invitecontext.mediahandler | closing PeerConnection sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.invitecontext.mediahandler | ICE connection state changed to "closed" sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.invitecontext.mediahandler | emitting event iceClosed sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.inviteclientcontext | emitting event failed sip.js:2636:3
call failed with cause: SIP Failure Code bbb_webrtc_bridge_sip.js:318:3
Stopping webrtc audio test bbb_webrtc_bridge_sip.js:87:2
Hanging up current session bbb_webrtc_bridge_sip.js:409:2
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.inviteclientcontext | Error: Attempted to send BYE in a terminated session. sip.js:2636:3
[12]</module.exports/LoggerFactory.prototype.print() sip.js:2636
[12]</module.exports/</LoggerFactory.prototype[targetName]() sip.js:2653
[12]</module.exports/</Logger.prototype[targetName]() sip.js:2647
[22]</module.exports/Session.prototype.bye() sip.js:4583
webrtc_hangup() bbb_webrtc_bridge_sip.js:413
stopWebRTCAudioTest() bbb_webrtc_bridge_sip.js:89
<anonymous> BigBlueButton.html
BBB.webRTCCallFailed() bbb_api_bridge.js:447
webRTCCallback() bbb_webrtc_bridge_sip.js:11
make_call/<() bbb_webrtc_bridge_sip.js:322
[7]</module.exports/EventEmitter.prototype.emit/<() sip.js:1072
forEach() self-hosted:211
[7]</module.exports/EventEmitter.prototype.emit() sip.js:1070
[22]</module.exports/Session.prototype.failed() sip.js:5292
[22]</module.exports/InviteClientContext.prototype.receiveInviteResponse() sip.js:6418
[17]</module.exports/RequestSender.prototype.receiveResponse() sip.js:3495
[26]</module.exports/InviteClientTransaction.prototype.receiveResponse() sip.js:7314
[27]</module.exports/Transport.prototype.onMessage() sip.js:7967
[27]</module.exports/Transport.prototype.connect/this.ws.onmessage() sip.js:7840

Fri Feb 05 2016 14:46:24 GMT+0100 | sip.ua | user requested closure... sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.ua | closing registerContext sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.registercontext | already unregistered sip.js:2636:3
"Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transport | closing WebSocket ws://77.93.202.240/ws" sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.inviteclientcontext | emitting event rejected sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transaction.ict | Timer D expired for INVITE client transaction z9hG4bK5161837 sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transaction.ict | emitting event stateChanged sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.ua | emitting event transactionDestroyed sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transport | WebSocket disconnected (code: 1000) sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.ua | connection state set to 1 sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.ua | emitting event disconnected sip.js:2636:3

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[Chad Pilkey]

Adam,

The error message coming from the server is "502 Bad Gateway". The official FreeSWITCH documentation explains it better, but basically it means that the destination on the server couldn't be reached. There have been two people who have posted in the past about Bad Gateway errors (one and two). The first one was caused by a misplaced "s" in one of the FreeSWITCH config files that the person had edited and the I suspected a similar issue with the second post as well, but they never updated so I don't know if they got it working or not.

Have you manually edited any of the FreeSWITCH configuration files? Are there any errors present in the output of "bbb-conf --check"?

On Friday, February 5, 2016 at 8:47:44 AM UTC-5, adam feit wrote:

Hi, this is my whole extract from the console:


An unbalanced tree was written using document.write() causing data from the network to be reparsed. For more information https://developer.mozilla.org/en/Optimizing_Your_Pages_for_Speculative_Parsing BigBlueButton.html:27:0
BigBlueButton SWF is ready. bbb_api_bridge.js:596:7
Successfully connected on test connection. {"server":"77.93.202.240","userId":"8bdphrd5eisg_1","port":":1935","meetingName":"Webinar","app":"video/portTest","protocol":"RTMP","meetingId":"81a5899ad9d8a70a9b52a50c05fbdb4ff300c4b1-1454679968043","username":"admin"} bbblogger.js:23:7
Successfully connected to BBB App. {"user":{"userId":"8bdphrd5eisg_2","externalMeetingId":"77379d1db58c9ac8c4892fdd4d8a15dfab5e4d2f","meetingName":"Webinar","meetingId":"81a5899ad9d8a70a9b52a50c05fbdb4ff300c4b1-1454679968043","userName":"admin"}} bbblogger.js:23:7
Received [UserJoinedEvent] bbb_api_bridge.js:567:7
No listeners for [UserJoinedEvent] bbb_api_bridge.js:543:13
Received [SwitchedPresenterEvent] bbb_api_bridge.js:567:7
No listeners for [SwitchedPresenterEvent] bbb_api_bridge.js:543:13
Received [NewRoleEvent] bbb_api_bridge.js:567:7
No listeners for [NewRoleEvent] bbb_api_bridge.js:543:13
Received [UserJoinedEvent] bbb_api_bridge.js:567:7
No listeners for [UserJoinedEvent] bbb_api_bridge.js:543:13
Received [SwitchedLayoutEvent] bbb_api_bridge.js:567:7
No listeners for [SwitchedLayoutEvent] bbb_api_bridge.js:543:13
Joining the audio test first bbb_webrtc_bridge_sip.js:81:2
User info callback [myUserID=8bdphrd5eisg,myUsername=admin,myAvatarURL=http://77.93.202.240/client/avatar.png,myRole=MODERATOR,amIPresenter=true,dialNumber=613-555-1234,voiceBridge=79974]. bbb_webrtc_bridge_sip.js:43:1
8bdphrd5eisg-bbbID-admin bbb_webrtc_bridge_sip.js:55:4
user 8bdphrd5eisg-bbbID-admin calling to 919679974 bbb_webrtc_bridge_sip.js:219:2
Fetching STUN/TURN server info for user agent bbb_webrtc_bridge_sip.js:129:2
Creating new user agent bbb_webrtc_bridge_sip.js:154:2
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | configuration parameters after validation: sip.js:2636:3
Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · viaHost: "2qkrv0lbe4sv.invalid" sip.js:2636:3

Fri Feb 05 2016 14:46:21 GMT+0100 | sip.ua | · uri: sip:8bdphrd5ei...@77.93.202.240 sip.js:2636:3

INVITE sip:91...@77.93.202.240 SIP/2.0

Via: SIP/2.0/WS 2qkrv0lbe4sv.invalid;branch=z9hG4bK5161837

Max-Forwards: 70

To: <sip:91...@77.93.202.240>

From: "8bdphrd5eisg-bbbID-admin" <sip:8bdphrd5ei...@77.93.202.240>;tag=a1ju6ihm99

Call-ID: 3pemc65322ma1n39sgba

CSeq: 2436 INVITE

Contact: <sip:19c3sbk3@2qkrv0lbe4sv.invalid;transport=ws;ob>

From: "8bdphrd5eisg-bbbID-admin" <sip:8bdphrd5ei...@77.93.202.240>;tag=a1ju6ihm99

To: <sip:91...@77.93.202.240>

Call-ID: 3pemc65322ma1n39sgba

CSeq: 2436 INVITE

User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150327T050105Z~f2a5271009~64bit

Content-Length: 0




 sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transaction.ict | emitting event stateChanged sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.inviteclientcontext | emitting event progress sip.js:2636:3
call progress: SIP/2.0 100 Trying

Via: SIP/2.0/WS 2qkrv0lbe4sv.invalid;branch=z9hG4bK5161837;received=77.93.202.240;rport=58539

From: "8bdphrd5eisg-bbbID-admin" <sip:8bdphrd5ei...@77.93.202.240>;tag=a1ju6ihm99

To: <sip:91...@77.93.202.240>

Call-ID: 3pemc65322ma1n39sgba

CSeq: 2436 INVITE

User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150327T050105Z~f2a5271009~64bit

Content-Length: 0



 bbb_webrtc_bridge_sip.js:314:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transport | received WebSocket text message:

SIP/2.0 502 Bad Gateway

Via: SIP/2.0/WS 2qkrv0lbe4sv.invalid;branch=z9hG4bK5161837;received=77.93.202.240;rport=58539

Max-Forwards: 69

From: "8bdphrd5eisg-bbbID-admin" <sip:8bdphrd5ei...@77.93.202.240>;tag=a1ju6ihm99

To: <sip:91...@77.93.202.240>;tag=4jQ5j12X5HZ9g

Call-ID: 3pemc65322ma1n39sgba

CSeq: 2436 INVITE

User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150327T050105Z~f2a5271009~64bit

Accept: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, refer

Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"

Content-Length: 0

Remote-Party-ID: "919679974" <sip:91...@77.93.202.240>;party=calling;privacy=off;screen=no

 sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transaction.ict | emitting event stateChanged sip.js:2636:3
Fri Feb 05 2016 14:46:24 GMT+0100 | sip.transport | sending WebSocket message:

ACK sip:91...@77.93.202.240 SIP/2.0

Via: SIP/2.0/WS 2qkrv0lbe4sv.invalid;branch=z9hG4bK5161837

To: <sip:91...@77.93.202.240>;tag=4jQ5j12X5HZ9g

[adam feit]

Hi, thanks for responde...
This is my check

BigBlueButton Server 0.9.0 (742)
                    Kernel version: 2.6.32-042stab108.8
                      Distribution: Ubuntu 14.04.2 LTS (64-bit)
                            Memory: 5120 MB

/var/www/bigbluebutton/client/conf/config.xml (bbb-client)
                Port test (tunnel): 77.93.202.240
                              Red5: 77.93.202.240
              useWebrtcIfAvailable: false

/opt/freeswitch/conf/sip_profiles/external.xml (FreeSWITCH)
                    websocket port: 5066
                    WebRTC enabled: false

/etc/nginx/sites-available/bigbluebutton (nginx)
                       server name: 77.93.202.240
                              port: 80
                    bbb-client dir: /var/www/bigbluebutton

/var/lib/tomcat7/webapps/bigbluebutton/WEB-INF/classes/bigbluebutton.properties (bbb-web)
                      bbb-web host: 77.93.202.240

/usr/share/red5/webapps/bigbluebutton/WEB-INF/red5-web.xml (red5)
                  voice conference: FreeSWITCH
                     capture video: true
                   capture desktop: true

/usr/local/bigbluebutton/core/scripts/bigbluebutton.yml (record and playback)
                     playback host: 77.93.202.240


** Potential problems described below **
# The following properties in /var/lib/tomcat7/webapps/bigbluebutton/WEB-INF/classes/bigbluebutton.properties have no value:
#     defaultWelcomeMessage
defaultWelcomeMessageFooter

...

[Fred Dixon]

Hi Adam,

In addition to the output of `sudo bbb-conf --check`, Chad had asked

> Have you manually edited any of the FreeSWITCH configuration files? 

A previous thread from a user had the same issue and it turned out to be an error in the FreeSWITCH configuration files after editing.

Regards,... Fred

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[adam feit]

Hi Fred,
Can you send me a link to a thread that came out this problem?

Thanks Adam

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[Fred Dixon]

Hi Adam,

I did a search for 1004 in bigbluebutton-setup and found this thread

 https://groups.google.com/d/msg/bigbluebutton-setup/F2MlW6Voj-0/R1qFl0cnCwAJ

and this thread

  https://groups.google.com/d/msg/bigbluebutton-setup/DyJoNJ0RrfY/Rt4pNTTzAwAJ

Hope that helps.

Regards,... Fred

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http://bigbluebutton.org/

@bigbluebutton

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