Re: OM blocked on some computers

[Yah's Global Kingdom]

The instructions for setting up the media server, tell you also how to
setup the Turn Server.

On Fri, Feb 23, 2024 at 3:10 PM Ali Alhaidary 
wrote:

>
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/Tutorials+for+installing+OpenMeetings+and+Tools?preview=/27838216/284789626/Installation%20OpenMeetings%207.2.0%20on%20Ubuntu%2022.04%20lts.pdf
> On 2/23/24 11:06, K. Kamhamea wrote:
>
> > Do you have TURN server?
>
> No. Do I need one to vercome the problem? If so, is there an instruction
> how to install it?
>
> Am Fr., 23. Feb. 2024 um 03:09 Uhr schrieb Maxim Solodovnik <
> solomax...@gmail.com>:
>
>> Hello,
>>
>> from mobile (sorry for typos ;)
>>
>>
>> On Fri, Feb 23, 2024, 08:45 Yah's Global Kingdom 
>> wrote:
>>
>>> I have not experienced this problem.
>>>
>>> On Thu, Feb 22, 2024 at 8:04 AM K. Kamhamea 
>>> wrote:
>>>
>>>> I use OM Version 6.2 to communicate with customers. Unfurtunately in
>>>> about 50% of cases OM is blocked, while my website from the same domain but
>>>> different IP is not blocked.
>>>>
>>>
>> Do you have TURN server?
>> Is it open "to the world"?
>>
>>
>> In some cases we could overcome the problem by booting the Win11 laptop
>>>> from an Ubuntu USB. In others I don't know. My questions are:
>>>> 1. Is there a trouble shooting guide for this problem?
>>>> 2. Can this be an IP:Port problem? I still use the 5443 port.
>>>> 3. Is this problem solved with the newer version 7.2?
>>>>
>>>

Re: OM blocked on some computers

[Yah's Global Kingdom]

I have not experienced this problem.

On Thu, Feb 22, 2024 at 8:04 AM K. Kamhamea  wrote:

> I use OM Version 6.2 to communicate with customers. Unfurtunately in about
> 50% of cases OM is blocked, while my website from the same domain but
> different IP is not blocked. In some cases we could overcome the problem by
> booting the Win11 laptop from an Ubuntu USB. In others I don't know. My
> questions are:
> 1. Is there a trouble shooting guide for this problem?
> 2. Can this be an IP:Port problem? I still use the 5443 port.
> 3. Is this problem solved with the newer version 7.2?
>

Re: OM6 testing _ assigning a set of whiteboards to one room

[Yah's Global Kingdom]

I can test in 7.0


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On Sun, Jan 9, 2022 at 6:12 AM jox joe  wrote:

> Environment:
> OM 6 (not the latest version)
> mariadb  Ver 15.1 Distrib 10.6.5-MariaDB, for debian-linux-gnu (x86_64)
> using readline 5.2
>
> Steps of testing:
> STEP 1: creating 20 new whiteboards (this can be ONE lesson in a course)
> STEP 2: saving all whiteboards (SAVE button)
> STEP 3: exit the room
> STEP 4: assign all 20 whiteboards to ONE room (administration/conference
> rooms)
> STEP 5: exit OM
> STEP 6: restart OM-Tomcat
> STEP 7: enter OM and check the given room which should contain the 20 WBs
>
> My result:
> - some whiteboard contents were missing completely(only the empty WB)
> - some whiteboards were not complete (part of the content was missing)
> - I could put back the missing whiteboard manually, but not always
> - sometimes (error message): "something bad happened, check log, contact
> the developers"
>
> Evaluation:
> - the Java module handling this part of the database is suspicious
> (to protect the database consistency "10.6.5-MariaDB" would be able to
> give more support /transaction/)
>
> Notes:
> - I tested the same OM 6 on two different Ubuntu 18.4 servers
> - RAM in the first server: 6 Gbyte
> - RAM in the second server: 8 Gbyte
> - the result was the same on both servers
>
> Possible next step:
> - someone in this community could do the same test
> (optimally with the latest "official" OM version)
>
> QUESTIONS:
> What do you think about this test case?
> What is your opinion about creating/managing (lesson) content in OM?
> (digital OM-Whiteboard means you can create your material in advance
> which can include videos, too)
>
> (Reference number of this message: 2022-01-09-13-08-47)
>
>
>
>

Re: OM 7.0.0-SNAPSHOT

[Yah's Global Kingdom]

Hello Maxim,

Thanks for the response.

I thought text chat previously made an audible sound when someone
transmitted text.  If not there should be an audible sound for chat and the
activities windows.  Preferably, a different sound for each window.  When I
am busy presenting, I am focused on presenting, if someone raises their
hand or types something in text chat, I am not learning of the event until
after the presentation.  So an audible alert for those events would greatly
help me, I don't know about anyone else.  Also now that the SIP client is
working.  The SIP callers need to be listed under the user section via
their callerID.  With ways to manage them like the normal OM users.  Right
now anyone could dial into a conference and there is no way to control them
but perhaps to mute them, but I don't see any way of muting them
individually or disconnecting them.

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On Thu, Jun 16, 2022 at 9:10 PM Maxim Solodovnik 
wrote:

> Hello,
>
> haven't tested this functionality for some time :(
>
> On Thu, 16 Jun 2022 at 00:50, Yah's Global Kingdom 
> wrote:
> >
> > Is there a way to turn on notification sounds for when someone types
> something in text chat
>
> AFAIK there is no Alarm for typing in chat
> Only visual notification in user list
>
> > or when someone raises their hand or requests permission to do anything.?
>
> This should raise browser alert and be displayed in rooms "Activities
> and Actions"
>
> >
> > What does enabling audio in text chat intended to do?
>
> This one should turn off browser level alert
>
> >
> > The download files button doesn't seem to work for moderators only
> admins.
>
> Historically only files of type document/image/recording are downloadable
> You can download in case documents are in personal or public folder
> and not in readOnly mode
>
> file is readOnly in case room belongs to group with restricted flag ON
> And user is NOT admin or moderator :)
>
> >
> >
> >
> >
> >
>
>
>
> --
> Best regards,
> Maxim
>

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OM 7.0.0-SNAPSHOT

[Yah's Global Kingdom]

Is there a way to turn on notification sounds for when someone types
something in text chat or when someone raises their hand or requests
permission to do anything.?

What does enabling audio in text chat intended to do?

The download files button doesn't seem to work for moderators only admins.

Re: SIP Dialer Openmeetings620

[Yah's Global Kingdom]

Hi Maxim,
I have a real provider through Voip.ms,   I just can not get OM sip dialer
to do anything   I would be happy to test and document if I could get it to
work.

On Sun, May 15, 2022 at 9:09 PM Maxim Solodovnik 
wrote:

> Hello Miles,
>
> On Fri, 13 May 2022 at 22:05, Yah's Global Kingdom 
> wrote:
> >
> > Maxim,
> >
> > Is the SIP dialer functional in version 620?
>
> Unfortunately I have no idea how to test it locally :(
> Linphone -> OM works as expected
>
> SIP dialer should help to call to external numbers
> I thought I need to set-up integration with some real provider to check
> this :(
>
> Ant help would be highly appreciated :)
>
> >  Should there configuration information for omsip_user? If so, what is
> the suggested configuration, if not where should this be configured.
> Thanks ahead of time
> >
> > Miles.
>
>
>
> --
> Best regards,
> Maxim
>

SIP Dialer Openmeetings620

[Yah's Global Kingdom]

Maxim,

Is the SIP dialer functional in version 620?  Should there configuration
information for omsip_user? If so, what is the suggested configuration, if
not where should this be configured.  Thanks ahead of time

Miles.

Upgrade fromOpen504 to Open620

[Yah's Global Kingdom]

ndow 
type the following:
; asterisk –rx “database show”
; If you do not receive an output with that resembles openmeetings/rooms/400## 
where “##” will equal
; the extension assigned when you created your room
; If you do not receive the above output check your parameters in
; /opt/om/webapps/openmeetings/WEB-INF/classes/openmeetings.properties
; Go back into the Administrator Panel and remove the PIN number in each room 
save the record with
; no PIN number and then re-enter the pin again resave the record.
; *

[rooms]
exten => 
_400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),SET(PIN=${DB(openmeetings/rooms/${EXTEN})})
exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
exten => _400X!,n,Hangup
exten => _400X!,n(notavail),Answer()
exten => _400X!,n,Playback(invalid)
exten => _400X!,n,Hangup

[rooms-originate]
exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
exten => _400X!,n,Hangup

[rooms-out]
; *
; Extensions for outgoing calls from Openmeetings room.
; *

[rooms-omsip]
exten => 
_400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,omsip_user)
exten => _400X!,n(notavail),Hangup

[home-phones]

exten => 1001,1,Dial(PJSIP/horace-desktop)

exten => 1002,1,Dial(PJSIP/horace-cellphone)

exten => 9000,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()

exten => _XX,1,Set(CALLERID(all)="YAH's Global Kingdom Ministries 
<4803829901>")
same => n,Dial(PJSIP/${EXTEN}@voipms)


;
; VOIP.MS SECTION  
;
; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 0123456789,1,Answer() ;your DID from ITSP
same => n,PLayback(hello)
same => n,WaitExten(30)
same => n,Hangup()

exten=> 1,1,Answer()
same => n,Dial(PJSIP/horace-desktop)

exten => 2,1,Answer()
same => n,Dial(PJSIP/horace-cellphone)

[voipms-outbound]
exten => _1NXXNXX,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _1NXXNXX,n,Hangup()
exten => _NXXNXX,1,Dial(PJSIP/1${EXTEN}@voipms)
exten => _NXXNXX,n,Hangup()
exten => _011.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

SIP.CONF entries
[general]
context=public  ; Default context for incoming calls. Defaults 
to 'default'
allowoverlap=no ; Disable overlap dialing support. (Default is 
yes)
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to 
(0.0.0.0 binds to all)
; Optionally add a port number, 
192.168.1.1:5062 (default is port 5060)
tcpenable=no; Enable server for incoming TCP connections 
(default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 
binds to all interfaces)
; Optionally add a port number, 
192.168.1.1:5062 (default is port 5060)
transport=udp
srvlookup=yes
maxexpiry=43200 ; Maximum allowed time of incoming 
registrations (seconds)
videosupport=yes 
nat = auto_force_rport  ; Set the force_rport option if Asterisk detects NAT 
(default)
nat = auto_comedia  ; Set the comedia option if Asterisk detects NAT
[basic-options](!); a template
dtmfmode=rfc2833
context=from-office
type=friend

[natted-phone](!,basic-options)   ; another template inheriting basic-options
directmedia=no
host=dynamic

[public-phone](!,basic-options)   ; another template inheriting basic-options
directmedia=yes

[my-codecs](!); a template for my preferred codecs
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
; Or, more simply:
;allow=!all,ilbc,g729,gsm,g723,ulaw

[ulaw-phone](!)   ; and another one for ulaw-only
disallow=all
allow=ulaw
; Again, more simply:
;allow=!all,ulaw

[omsip_user]
host=dynamic
secret=
context=rooms-omsip
transport=ws,wss
type=friend
encryption=no
avpf=yes
icesupport=yes
directmedia=no
allow=!all,ulaw,opus,vp8

**
Configurations from PJSIP
**
; Basic UDP t

webrtc error

[Yah's Global Kingdom]

I am seeing this error when entering rooms
SyntaxError: Failed to construct 'RTCPeerConnection': ICE server parse
failed

Where do I fix this?


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Re: Settngs for Sip Manager

[Yah's Global Kingdom]

Thanks for your responses Maxim,
I understand now and I have corrected:
/etc/asterisk/manager.conf
/etc/asterisk/extension.conf

I manually added the extension to the Asterisk internal key-value DB

meetings*CLI> database show
/SIP/Registry/horace  :
98.174.244.227:49952:60:horace:sip:horace@98.174.244.227:49952;transport=UDP;rinstance=0165778b0d1a09b9
/SIP/Registry/horacecell  :
172.58.69.178:45874:60:horacecell:sip:horacecell@172.58.69.178:45874;transport=UDP;rinstance=42c80f10d1bbc35f
/dundi/secret :
HxawUR7CiWPsyMBN73Q2bQ==;HUnEDvN5+h2GVC1pW4GEYw==
/dundi/secretexpiry   : 1630104193
/openmeetings/rooms   : 40011
/pbx/UUID :
7dd6882b-8da9-4099-a6a7-3012970c94ca
6 results found.

I still get invalid extension when I try to dial into the conference.
Also when I create a room and SIP enable it, should openmeetings be
updating the asterisk internal key-value DB?
I think that it should be, so perhaps there is something still wrong
between my /etc/asterisk/manager.conf and the
/opt/om/webapps/openmeetings/WEB-INF/classes/openmeetings.properties
settings.

Thanks for all your help in getting me this far.


On Thu, Aug 26, 2021 at 7:05 AM Maxim Solodovnik 
wrote:

>
>
> On Thu, 26 Aug 2021 at 21:03, Yah's Global Kingdom 
> wrote:
>
>> Maxim, even if my openmeetings database is open504, this needs to be set
>> to openmeetings?
>>
>
> Yes
> This is NOT MySQL OM DB
> This is Asterisk internal key-value DB :)
>
>
>>
>> On Tue, Aug 24, 2021 at 11:09 PM Maxim Solodovnik 
>> wrote:
>>
>>>
>>>
>>> On Wed, 25 Aug 2021 at 09:38, Yah's Global Kingdom 
>>> wrote:
>>>
>>>> Hi Maxim,
>>>>
>>>> Thanks for responding:
>>>> What I am getting is invalid extension,  when I run asterisk -rx
>>>> database show, I don't see anything from openmeetings
>>>>
>>>> meetings*CLI> database show
>>>> /SIP/Registry/horace  : 1.1.1.1:49952
>>>> :60:horace:sip:horace@1.1.1.1.1
>>>> :49952;transport=UDP;rinstance=fd20fb9e9a736274
>>>> /SIP/Registry/horacecell  : 172.58.69.178:45874
>>>> :60:horacecell:sip:horacecell@172.58.69.178:45874
>>>> ;transport=UDP;rinstance=d0bae988fae57944
>>>> /dundi/secret :
>>>> Bk1zG+mtsgGrpKsMJw09EA==;wS1dMgw+F1jfle4crmWRAA==
>>>> /dundi/secretexpiry   : 1629852193
>>>>
>>>> /pbx/UUID :
>>>> 7dd6882b-8da9-4099-a6a7-3012970c94ca
>>>> 5 results found.
>>>>
>>>> I am confused by these two lines in the Extensions.conf:
>>>> exten =>
>>>> _400X!,1,GotoIf($[${DB_EXISTS(open504/rooms/${EXTEN})}]?ok:notavail)
>>>> exten => _400X!,n(ok),SET(PIN=${DB(open504/rooms/${EXTEN})})
>>>>
>>>
>>> Well
>>> I'm not sure why do you have such portion of config :(
>>>
>>> "reference" config looks like
>>> https://openmeetings.apache.org/AsteriskIntegration.html#configure-extensions
>>>
>>> `openmeetings/rooms/` is NOT DB table but some internal Asterisk
>>> key-value DB
>>> so `open504/rooms` looks wrong here
>>> It should be `openmeetings/rooms`
>>>
>>>
>>>
>>>>
>>>> 1.  in the database open504 I don't find a table named rooms, the table
>>>> name in the database is room.
>>>> 2.   I don't know why it is not registering the open504 database with
>>>> Asterisk,  I am thinking the transport Agent is suppose to do that somehow,
>>>> (just me thinking)
>>>>
>>>> meetings*CLI>
>>>> Tables_in_open504  |
>>>> ++
>>>> | address|
>>>> | appointment|
>>>> | chat   |
>>>> | conference_log |
>>>> | configuration  |
>>>> | email_queue|
>>>> | extra_menu |
>>>> | file_log   |
>>>> | group_user |
>>>> | invitation |
>>>> | ldapconfig |
>>>> | meeting_member |
>>>> | menu_group |
>>>> | oauth_mapping  |
>&

Re: Settngs for Sip Manager

[Yah's Global Kingdom]

Maxim, even if my openmeetings database is open504, this needs to be set to
openmeetings?

On Tue, Aug 24, 2021 at 11:09 PM Maxim Solodovnik 
wrote:

>
>
> On Wed, 25 Aug 2021 at 09:38, Yah's Global Kingdom 
> wrote:
>
>> Hi Maxim,
>>
>> Thanks for responding:
>> What I am getting is invalid extension,  when I run asterisk -rx database
>> show, I don't see anything from openmeetings
>>
>> meetings*CLI> database show
>> /SIP/Registry/horace  : 1.1.1.1:49952
>> :60:horace:sip:horace@1.1.1.1.1
>> :49952;transport=UDP;rinstance=fd20fb9e9a736274
>> /SIP/Registry/horacecell  : 172.58.69.178:45874
>> :60:horacecell:sip:horacecell@172.58.69.178:45874
>> ;transport=UDP;rinstance=d0bae988fae57944
>> /dundi/secret :
>> Bk1zG+mtsgGrpKsMJw09EA==;wS1dMgw+F1jfle4crmWRAA==
>> /dundi/secretexpiry   : 1629852193
>>
>> /pbx/UUID :
>> 7dd6882b-8da9-4099-a6a7-3012970c94ca
>> 5 results found.
>>
>> I am confused by these two lines in the Extensions.conf:
>> exten =>
>> _400X!,1,GotoIf($[${DB_EXISTS(open504/rooms/${EXTEN})}]?ok:notavail)
>> exten => _400X!,n(ok),SET(PIN=${DB(open504/rooms/${EXTEN})})
>>
>
> Well
> I'm not sure why do you have such portion of config :(
>
> "reference" config looks like
> https://openmeetings.apache.org/AsteriskIntegration.html#configure-extensions
>
> `openmeetings/rooms/` is NOT DB table but some internal Asterisk
> key-value DB
> so `open504/rooms` looks wrong here
> It should be `openmeetings/rooms`
>
>
>
>>
>> 1.  in the database open504 I don't find a table named rooms, the table
>> name in the database is room.
>> 2.   I don't know why it is not registering the open504 database with
>> Asterisk,  I am thinking the transport Agent is suppose to do that somehow,
>> (just me thinking)
>>
>> meetings*CLI>
>> Tables_in_open504  |
>> ++
>> | address|
>> | appointment|
>> | chat   |
>> | conference_log |
>> | configuration  |
>> | email_queue|
>> | extra_menu |
>> | file_log   |
>> | group_user |
>> | invitation |
>> | ldapconfig |
>> | meeting_member |
>> | menu_group |
>> | oauth_mapping  |
>> | oauth_server   |
>> | om_calendar|
>> | om_file|
>> | om_group   |
>> | om_user|
>> | om_user_right  |
>> | private_message|
>> | private_message_folder |
>> | recording_chunk|
>> | room   |
>> | room_file  |
>> | room_group |
>> | room_hide_element  |
>> | room_moderator |
>> | room_poll  |
>> | room_poll_answer   |
>> | sessiondata|
>> | sipusers   |
>> | soaplogin  |
>> | user_contact   |
>> ++
>> 34 rows in set (0.00 sec)
>>
>>
>>
>> +--+--+--+-+-++
>> | Field| Type | Null | Key | Default | Extra
>>  |
>>
>> +--+--+--+-+-++
>> | id   | bigint(20)   | NO   | PRI | NULL|
>> auto_increment |
>> | deleted  | bit(1)   | NO   | | NULL|
>>  |
>> | inserted | datetime | YES  | | NULL|
>>  |
>> | updated  | datetime | YES  | | NULL|
>>  |
>> | allow_recording  | bit(1)   | NO   | | NULL|
>>  |
>> | allow_user_questions | bit(1)   | NO   | | NULL|
>>  |
>> | appointment  | bit(1)   | NO   | | NULL|
>>  |
>> | audio_only   | bit(1)   | NO   | | NULL|
>>  |
>> | capacity | bigint(20)   | YES  | | NULL|
>>  |
>> | chat_moderated   | bit(1)   | NO   | | NULL|
>>  |
>> | chat_opened  | bit(1)   | NO   | | NULL|
>>  |
>> | closed   | bit(1)   | NO   | | NULL|
>>  |
>> | comment  | text | YES  | | NUL

Re: Settngs for Sip Manager

[Yah's Global Kingdom]

0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = all
write = all

"openmeetings.settings"
sip.hostname=meetings.glorytoyah.org
sip.manager.port=5038
sip.manager.user=open504
sip.manager.password=12345 - Assuming this is the value from manger.conf
[open504]
sip.manager.timeout=1

sip.ws.local.port.min=
sip.ws.local.port.max=7666
## 127.0.0.1 is NOT working here
sip.ws.local.host=public IP
sip.ws.remote.port=8088
sip.ws.remote.user=omsip_user
sip.ws.remote.password=12345



On Mon, Aug 23, 2021 at 7:57 PM Maxim Solodovnik 
wrote:

> Not sure I got the question :(
>
> You can specify these settings by editing
> https://github.com/apache/openmeetings/blob/master/openmeetings-web/src/main/webapp/WEB-INF/classes/openmeetings.properties#L68
> file
> (located at
> ${OM_HOME}/webapps/openmeetings/WEB-INF/classes/openmeetings.properties)
>
> On Asterisk side it is located here:
> https://openmeetings.apache.org/AsteriskIntegration.html#configure-asterisk-manager
>
>
> On Tue, 24 Aug 2021 at 05:27, Yah's Global Kingdom 
> wrote:
>
>> @Maxium
>> Asterisk:
>> Where are these settings located?
>>
>> sip.manager.user=
>> sip.manager.password=
>> sip.manager.timeout=
>>
>
>
> --
> Best regards,
> Maxim
>

Re: Status of SIP Integration in OM 6.00

[Yah's Global Kingdom]

Hi Ali Alhaidary,

Is it possible that you can tell me were you got the settings for
openmeetings.settings in asterisk? Guess what I am asking the below
settings are required to be set in openmeetings.settings file, but I don't
know what they actually equate to in Asterisk.
Asterisk:
Where are these settings located?

sip.manager.user=
sip.manager.password=
sip.manager.timeout=

On Fri, Apr 16, 2021 at 2:15 PM Yah's Global Kingdom 
wrote:

> I would think that the conferencing bridge would be able to handle many to
> one connections as that is the ultimate purpose conferencing...I will have
> to check it out next week, right now I am in a temp location and have to
> move back into my office and set everything back up again. I am hoping that
> using SIP as the protocol that it utilizing the conference bridge which
> would be fantastic.
>
> Miles
>
> On Fri, Apr 16, 2021 at 1:21 PM Ali Alhaidary 
> wrote:
>
>> Yes, running the latest build of 6.1.0, however it was long and complex
>> learning experience for us, and, it depends on the subscription and number
>> of lines offered, and, for some reason that I did not dig too much into,
>> server is overloaded, and unfortunately, in our part of the globe, no
>> operator is willing to offer the service, so we limited lessons to
>> internet, but other than that, it worked very good two way audio.
>>
>> Ali
>> On 4/16/21 10:01 PM, Yah's Global Kingdom wrote:
>>
>> Thanks Ali,
>>
>> I have to look at my notes it has been a minute.  I see emails that you
>> are running 6.+ Are multiple user able to call in to your room conference
>> using SIP?
>>
>> On Thu, Apr 15, 2021 at 2:04 PM Ali Alhaidary <
>> ali.alhaid...@the5stars.org> wrote:
>>
>>> upgrading is a long process, but certainly worth each and every minute
>>> :-)
>>>
>>> Ali
>>> On 4/15/21 11:40 PM, Yah's Global Kingdom wrote:
>>>
>>> Ok thanks I will upgrade and test
>>>
>>> On Tue, Apr 13, 2021 at 5:29 PM Maxim Solodovnik 
>>> wrote:
>>>
>>>> I believe so :)
>>>> But my testing abilities are limited :)
>>>>
>>>> And SIP-video is not implemented
>>>>
>>>> from mobile (sorry for typos ;)
>>>>
>>>>
>>>> On Wed, Apr 14, 2021, 05:11 Yah's Global Kingdom 
>>>> wrote:
>>>>
>>>>> Maxim what exactly does that mean,  Can more than one person, call
>>>>> into the room SIMULTANEOUSILY using SIP?
>>>>>
>>>>> On Sat, Apr 10, 2021 at 7:27 PM Maxim Solodovnik 
>>>>> wrote:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> According to my tests 2-way audio-only SIP should work as expected :)
>>>>>>
>>>>>> from mobile (sorry for typos ;)
>>>>>>
>>>>>>
>>>>>> On Thu, Apr 8, 2021, 04:20 Yah's Global Kingdom 
>>>>>> wrote:
>>>>>>
>>>>>>> Has this been implemented yet?
>>>>>>>
>>>>>>

Settngs for Sip Manager

[Yah's Global Kingdom]

@Maxium
Asterisk:
Where are these settings located?

sip.manager.user=
sip.manager.password=
sip.manager.timeout=

Error 602 when trying to connect conference room

[Yah's Global Kingdom]

When  trying to connect to a conference room I am getting
SIP/2.0 603 Declined

I don't know why asterisk is declining the request to enter the conference
room.  I am including the portion of the dial plan that is being executed
along with a sip debug information.  As stated before I don't see the
Openmeetings Transport agent register or do anything with asterisk, (am I
suppose to?).  The server just declines to let me enter into the conference
room and I unable to determine why.  Can someone help me out with this?

>From extension.conf
exten => _40011,1,GotoIf($[${DB_EXISTS(open504/room/${EXTEN})}]?ok:notavail)
exten => _40011,n(ok),Confbridge(${EXTEN},default_bridge,omsip_user)
exten => _40011,n(notavail),Hangup


>From Sip Debug

<--- SIP read from UDP:x.x.x.x:49952 --->
REGISTER sip:meetings.glorytoyah.org:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
x.x.x.x:49952;branch=z9hG4bK-524287-1---e7e629091d6d0a9e;rport
Max-Forwards: 70
Contact: 
To: 
From: ;tag=f275c241
Call-ID: iM97Q5-Kv-TL_VmglcgQug..
CSeq: 4757 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
User-Agent: Z 5.5.3 v2.10.15.0
Authorization: Digest
username="horace",realm="asterisk",nonce="59b69e27",uri="sip:meetings.glorytoyah.org:5060
;transport=UDP",response="19416047ef96e57180711cf3c65efaeb",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0

<->
--- (14 headers 0 lines) ---
Sending to x.x.x.x:49952 (no NAT)
Sending to x.x.x.x:49952 (no NAT)

<--- Transmitting (no NAT) to x.x.x.x:49952 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
x.x.x.x:49952;branch=z9hG4bK-524287-1---e7e629091d6d0a9e;received=x.x.x.x;rport=49952
From: ;tag=f275c241
To: ;tag=as1ceb5f1f
Call-ID: iM97Q5-Kv-TL_VmglcgQug..
CSeq: 4757 REGISTER
Server: Asterisk PBX 16.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2db3b07b"
Content-Length: 0


<>
Scheduling destruction of SIP dialog 'iM97Q5-Kv-TL_VmglcgQug..' in 32000 ms
(Method: REGISTER)

<--- SIP read from UDP:x.x.x.x:49952 --->
REGISTER sip:meetings.glorytoyah.org:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
x.x.x.x:49952;branch=z9hG4bK-524287-1---eee8b07fdd98a3c5;rport
Max-Forwards: 70
Contact: 
To: 
From: ;tag=f275c241
Call-ID: iM97Q5-Kv-TL_VmglcgQug..
CSeq: 4758 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
User-Agent: Z 5.5.3 v2.10.15.0
Authorization: Digest
username="horace",realm="asterisk",nonce="2db3b07b",uri="sip:meetings.glorytoyah.org:5060
;transport=UDP",response="a447e48656b54e723a36d28222efcf83",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0

<->
--- (14 headers 0 lines) ---
Sending to x.x.x.x:49952 (no NAT)

<--- Transmitting (no NAT) to x.x.x.x:49952 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
x.x.x.x:49952;branch=z9hG4bK-524287-1---eee8b07fdd98a3c5;received=x.x.x.x;rport=49952
From: ;tag=f275c241
To: ;tag=as1ceb5f1f
Call-ID: iM97Q5-Kv-TL_VmglcgQug..
CSeq: 4758 REGISTER
Server: Asterisk PBX 16.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: ;expires=60
Date: Mon, 23 Aug 2021 05:02:51 GMT
Content-Length: 0


<>
Scheduling destruction of SIP dialog 'iM97Q5-Kv-TL_VmglcgQug..' in 32000 ms
(Method: REGISTER)

<--- SIP read from UDP:172.58.69.178:45874 --->
INVITE sip:40...@meetings.glorytoyah.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.0.0.2:34804
;branch=z9hG4bK-524287-1---52f2ea8febffe027;rport
Max-Forwards: 70
Contact: 
To: 
From: ;tag=6d96d10f
Call-ID: wg7pHc0DwIOKu9v8K_sPHQ..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.10.12.3-mod
Allow-Events: presence, kpml, talk
Content-Length: 185

v=0
o=Zoiper 1629694971491 1 IN IP4 172.58.69.178
s=Z
c=IN IP4 172.58.69.178
t=0 0
m=audio 59692 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<->
--- (13 headers 9 lines) ---
Sending to 172.58.69.178:45874 (NAT)
Sending to 172.58.69.178:45874 (NAT)
Using INVITE request as basis request - wg7pHc0DwIOKu9v8K_sPHQ..
Found peer 'horacecell' for 'horacecell' from 172.58.69.178:45874

<--- Reliably Transmitting (NAT) to 172.58.69.178:45874 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.0.0.2:34804
;branch=z9hG4bK-524287-1---52f2ea8febffe027;received=172.58.69.178;rport=45874
From: ;tag=6d96d10f
To: ;tag=as5df773b0
Call-ID: wg7pHc0DwIOKu9v8K_sPHQ..
CSeq: 1 INVITE
Server: Asterisk PBX 16.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b43696f"
Content-Length: 0


<>
Scheduling destruction of SIP dialog 

Re: SIP Integration

[Yah's Global Kingdom]

OK, I am able to register devices and call anything within the internal
context.  But I can not dial a conference room.  Can anyone that is able to
dial a conference from an Asterisk instance please share their Sip.conf and
Extension.conf so I can compare...?

On Wed, Aug 18, 2021 at 12:10 AM Maxim Solodovnik 
wrote:

> `sudo netstat -taupen|grep aster`
>
> lists port 5060 for me 
>
>
> On Wed, 18 Aug 2021 at 06:38, Yah's Global Kingdom 
> wrote:
>
>> The SIP protocol uses port 5060, according to the documentation: SIP
>> Config tcpenble =yes  and tcpbindaddress default port number is 5060.
>>
>> On Mon, Aug 16, 2021 at 9:34 PM Maxim Solodovnik 
>> wrote:
>>
>>>
>>>
>>> On Mon, 16 Aug 2021 at 21:55, Yah's Global Kingdom 
>>> wrote:
>>>
>>>> Please disregard, I have gotten the sip transport to enter the room.
>>>> However, I don't see anything in Asterisk for when the Transport agent
>>>> enters the room or when I try to register a client.
>>>>
>>>
>>> You should "see something in Asterisk" at the moment the SIP user enters
>>> the room (better with Om user in it ...)
>>>
>>>
>>>>   I have nothing listening on ports 5060,5061 or 5062.
>>>>
>>>
>>> Why do you expect something should listen these ports?
>>>
>>>
>>>>
>>>> On Sat, Aug 14, 2021 at 11:29 AM Yah's Global Kingdom <
>>>> yahs...@gmail.com> wrote:
>>>>
>>>>> Update:
>>>>> Asterisk is not listening on ports 5060/5061/5062 although I have
>>>>> updated the sip.conf
>>>>>
>>>>> I am using the guide at
>>>>> https://openmeetings.apache.org/AsteriskIntegration.html to implement
>>>>> Asterisk and VOIP.
>>>>>
>>>>> Before under previous additions, when I entered the room, the SIP
>>>>> transport agent would also enter the room.  Now after upgrading from 5.0 
>>>>> to
>>>>> 6.10 when I enter the room no sip transport agent enters.   What
>>>>> information do I need to provide to anyone so I can troubleshoot this
>>>>> matter.   Is there an upgraded version of this guide
>>>>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration
>>>>>  ?
>>>>> The sipusers table in 6.1 looks nothing like the table in this guide.
>>>>>
>>>>> Sincerely
>>>>> Bro Miles
>>>>> YAH's Global Kingdom Ministries.
>>>>>
>>>>> On Sat, Aug 14, 2021 at 10:55 AM Yah's Global Kingdom <
>>>>> yahs...@gmail.com> wrote:
>>>>>
>>>>>> I am using the guide at
>>>>>> https://openmeetings.apache.org/AsteriskIntegration.html to
>>>>>> implement Asterisk and VOIP.
>>>>>>
>>>>>> Before under previous additions, when I entered the room, the SIP
>>>>>> transport agent would also enter the room.  Now after upgrading from 5.0 
>>>>>> to
>>>>>> 6.10 when I enter the room no sip transport agent enters.   What
>>>>>> information do I need to provide to anyone so I can troubleshoot this
>>>>>> matter.   Is there an upgraded version of this guide
>>>>>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration
>>>>>> ?  The sipusers table in 6.1 looks nothing like the table in this guide.
>>>>>>
>>>>>> Sincerely
>>>>>> Bro Miles
>>>>>> YAH's Global Kingdom Ministries.
>>>>>>
>>>>>
>>>
>>> --
>>> Best regards,
>>> Maxim
>>>
>>
>
> --
> Best regards,
> Maxim
>

SIP Integrations - How to call Confrence room from Asterisk

[Yah's Global Kingdom]

Ok got clients to register in asterisk and I can make a call from my
[Internal] context to another [internal] context number.
However, when I try to dial a conference room I get the Call from ext-
to extension 40011 (room ID) because extension not found in context
internal  How do I set up the dial plan to dial been context(s).  Wheen I
use the "include  option"  i.e.
[room]
include => Internal I still get the same error.

If I move an extension to the rooms [context]  the attendant answers state
the extension for the room is invalid.

I did note that in extensions.conf the connection string for the database
(openmeetings/rooms
should be openmeetings/room as rooms does not exist in the database.

So how do I make this call across contexts i.e. registered ext 1000 call to
room 40011?

Thanks ahead of time.

Re: SIP Integration

[Yah's Global Kingdom]

The SIP protocol uses port 5060, according to the documentation: SIP Config
tcpenble =yes  and tcpbindaddress default port number is 5060.

On Mon, Aug 16, 2021 at 9:34 PM Maxim Solodovnik 
wrote:

>
>
> On Mon, 16 Aug 2021 at 21:55, Yah's Global Kingdom 
> wrote:
>
>> Please disregard, I have gotten the sip transport to enter the room.
>> However, I don't see anything in Asterisk for when the Transport agent
>> enters the room or when I try to register a client.
>>
>
> You should "see something in Asterisk" at the moment the SIP user enters
> the room (better with Om user in it ...)
>
>
>>   I have nothing listening on ports 5060,5061 or 5062.
>>
>
> Why do you expect something should listen these ports?
>
>
>>
>> On Sat, Aug 14, 2021 at 11:29 AM Yah's Global Kingdom 
>> wrote:
>>
>>> Update:
>>> Asterisk is not listening on ports 5060/5061/5062 although I have
>>> updated the sip.conf
>>>
>>> I am using the guide at
>>> https://openmeetings.apache.org/AsteriskIntegration.html to implement
>>> Asterisk and VOIP.
>>>
>>> Before under previous additions, when I entered the room, the SIP
>>> transport agent would also enter the room.  Now after upgrading from 5.0 to
>>> 6.10 when I enter the room no sip transport agent enters.   What
>>> information do I need to provide to anyone so I can troubleshoot this
>>> matter.   Is there an upgraded version of this guide
>>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration
>>>  ?
>>> The sipusers table in 6.1 looks nothing like the table in this guide.
>>>
>>> Sincerely
>>> Bro Miles
>>> YAH's Global Kingdom Ministries.
>>>
>>> On Sat, Aug 14, 2021 at 10:55 AM Yah's Global Kingdom 
>>> wrote:
>>>
>>>> I am using the guide at
>>>> https://openmeetings.apache.org/AsteriskIntegration.html to implement
>>>> Asterisk and VOIP.
>>>>
>>>> Before under previous additions, when I entered the room, the SIP
>>>> transport agent would also enter the room.  Now after upgrading from 5.0 to
>>>> 6.10 when I enter the room no sip transport agent enters.   What
>>>> information do I need to provide to anyone so I can troubleshoot this
>>>> matter.   Is there an upgraded version of this guide
>>>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration
>>>> ?  The sipusers table in 6.1 looks nothing like the table in this guide.
>>>>
>>>> Sincerely
>>>> Bro Miles
>>>> YAH's Global Kingdom Ministries.
>>>>
>>>
>
> --
> Best regards,
> Maxim
>

Re: SIP Integration

[Yah's Global Kingdom]

Please disregard, I have gotten the sip transport to enter the room.
However, I don't see anything in Asterisk for when the Transport agent
enters the room or when I try to register a client.  I have nothing
listening on ports 5060,5061 or 5062.

On Sat, Aug 14, 2021 at 11:29 AM Yah's Global Kingdom 
wrote:

> Update:
> Asterisk is not listening on ports 5060/5061/5062 although I have updated
> the sip.conf
>
> I am using the guide at
> https://openmeetings.apache.org/AsteriskIntegration.html to implement
> Asterisk and VOIP.
>
> Before under previous additions, when I entered the room, the SIP
> transport agent would also enter the room.  Now after upgrading from 5.0 to
> 6.10 when I enter the room no sip transport agent enters.   What
> information do I need to provide to anyone so I can troubleshoot this
> matter.   Is there an upgraded version of this guide
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration
>  ?
> The sipusers table in 6.1 looks nothing like the table in this guide.
>
> Sincerely
> Bro Miles
> YAH's Global Kingdom Ministries.
>
> On Sat, Aug 14, 2021 at 10:55 AM Yah's Global Kingdom 
> wrote:
>
>> I am using the guide at
>> https://openmeetings.apache.org/AsteriskIntegration.html to implement
>> Asterisk and VOIP.
>>
>> Before under previous additions, when I entered the room, the SIP
>> transport agent would also enter the room.  Now after upgrading from 5.0 to
>> 6.10 when I enter the room no sip transport agent enters.   What
>> information do I need to provide to anyone so I can troubleshoot this
>> matter.   Is there an upgraded version of this guide
>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration
>> ?  The sipusers table in 6.1 looks nothing like the table in this guide.
>>
>> Sincerely
>> Bro Miles
>> YAH's Global Kingdom Ministries.
>>
>

Re: SIP Integration

[Yah's Global Kingdom]

Update:
Asterisk is not listening on ports 5060/5061/5062 although I have updated
the sip.conf

I am using the guide at
https://openmeetings.apache.org/AsteriskIntegration.html to implement
Asterisk and VOIP.

Before under previous additions, when I entered the room, the SIP
transport agent would also enter the room.  Now after upgrading from 5.0 to
6.10 when I enter the room no sip transport agent enters.   What
information do I need to provide to anyone so I can troubleshoot this
matter.   Is there an upgraded version of this guide
https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration
?
The sipusers table in 6.1 looks nothing like the table in this guide.

Sincerely
Bro Miles
YAH's Global Kingdom Ministries.

On Sat, Aug 14, 2021 at 10:55 AM Yah's Global Kingdom 
wrote:

> I am using the guide at
> https://openmeetings.apache.org/AsteriskIntegration.html to implement
> Asterisk and VOIP.
>
> Before under previous additions, when I entered the room, the SIP
> transport agent would also enter the room.  Now after upgrading from 5.0 to
> 6.10 when I enter the room no sip transport agent enters.   What
> information do I need to provide to anyone so I can troubleshoot this
> matter.   Is there an upgraded version of this guide
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration
> ?  The sipusers table in 6.1 looks nothing like the table in this guide.
>
> Sincerely
> Bro Miles
> YAH's Global Kingdom Ministries.
>

SIP Integration

[Yah's Global Kingdom]

I am using the guide at
https://openmeetings.apache.org/AsteriskIntegration.html to implement
Asterisk and VOIP.

Before under previous additions, when I entered the room, the SIP
transport agent would also enter the room.  Now after upgrading from 5.0 to
6.10 when I enter the room no sip transport agent enters.   What
information do I need to provide to anyone so I can troubleshoot this
matter.   Is there an upgraded version of this guide
https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration
?  The sipusers table in 6.1 looks nothing like the table in this guide.

Sincerely
Bro Miles
YAH's Global Kingdom Ministries.

Re: Era in MYSQL RealTime

[Yah's Global Kingdom]

I am using version 16.13.0.  I don't know what would be making the query,
or if there are other fields that it requires.  The documentation
say's field order is important.

On Thu, Aug 12, 2021 at 7:22 PM Maxim Solodovnik 
wrote:

> OM creates "sipusers" DB table
> but column "insecure" is NOT created
>
> I saw no issues with this
>
> what version of Asterisk are you using?
>
> I guess you can manually add the desired column to this DB table :)
>
> On Thu, 12 Aug 2021 at 09:29, Yah's Global Kingdom 
> wrote:
>
>> I am getting the following era in Asterisk trying to implement VOIP SIP
>> integration
>> resconfig_mysq.c: 513 realtime_multi_mysql:  MySQL RealTime: Failed to
>> query database: unknown column 'insecure' in 'where clause'
>>
>> I "assume" this is coming from sipusers table.  However I know what the
>> order of the fields are and were this field should be added.
>>
>> Miles
>>
>
>
> --
> Best regards,
> Maxim
>

Era in MYSQL RealTime

[Yah's Global Kingdom]

I am getting the following era in Asterisk trying to implement VOIP SIP
integration
resconfig_mysq.c: 513 realtime_multi_mysql:  MySQL RealTime: Failed to
query database: unknown column 'insecure' in 'where clause'

I "assume" this is coming from sipusers table.  However I know what the
order of the fields are and were this field should be added.

Miles

Re: OpenMeetings Back/Restore

[Yah's Global Kingdom]

jira created


On Tue, Aug 10, 2021 at 10:52 PM Maxim Solodovnik 
wrote:

> Well,
>
> I tried to upload huge video file as backup
> And I got an error, but this took significant amount of time
> I'll try to improve this
> Can you create the JIRA?
>
> On Wed, 11 Aug 2021 at 12:36, Maxim Solodovnik 
> wrote:
>
>> There might be an issue with file upload UI
>> I'll take a deeper look and will report back :)
>>
>> meanwhile you can use "command line admin"
>> basic instructions are at Admin->Backup page
>> More info here: https://openmeetings.apache.org/CommandLineAdmin.html
>>
>> On Wed, 11 Aug 2021 at 11:42, Yah's Global Kingdom 
>> wrote:
>>
>>> actual files size is 3.98G Chrome has a limit of 4GB..
>>> Perhaps you can share what the import command does and I can do it
>>> manually.
>>>
>>>
>>> On Tue, Aug 10, 2021 at 9:37 PM Yah's Global Kingdom 
>>> wrote:
>>>
>>>> I change the limit to 10GB the file size is 4GB.   File is already on
>>>> the server.
>>>>
>>>> On Tue, Aug 10, 2021 at 6:19 PM Maxim Solodovnik 
>>>> wrote:
>>>>
>>>>> What is your current limit?
>>>>> What is the file size?
>>>>>
>>>>> This limit is applied by browser, so the behavior might be not very
>>>>> user friendly :(
>>>>>
>>>>> On Wed, 11 Aug 2021 at 06:35, Yah's Global Kingdom 
>>>>> wrote:
>>>>>
>>>>>> O.K. Thanks. I thought applied literally to just file uploads, not to
>>>>>> the size of the backup file which is on the server and doesn't need to be
>>>>>> uploaded.   However, when I click the import link and select the back up
>>>>>> file which is in .zip format.  Nothing happens, no status bar on the 
>>>>>> screen
>>>>>> , nothing.  This is version 6.10.
>>>>>>
>>>>>> On Mon, Aug 9, 2021 at 8:21 PM Maxim Solodovnik 
>>>>>> wrote:
>>>>>>
>>>>>>> Admin->Config
>>>>>>>
>>>>>>> max.upload.size
>>>>>>>
>>>>>>> More info:
>>>>>>> https://openmeetings.apache.org/GeneralConfiguration.html
>>>>>>>
>>>>>>> On Tue, 10 Aug 2021 at 07:04, Yah's Global Kingdom <
>>>>>>> yahs...@gmail.com> wrote:
>>>>>>>
>>>>>>>> Is there anyway to increase the size of the OpenMeetings
>>>>>>>> Back/Restore.  When I backuped my system it is 4GB..I go to restore it 
>>>>>>>> and
>>>>>>>> it says the Max 100MB?  Got to be a way to change that.
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Best regards,
>>>>>>> Maxim
>>>>>>>
>>>>>>
>>>>>
>>>>> --
>>>>> Best regards,
>>>>> Maxim
>>>>>
>>>>
>>
>> --
>> Best regards,
>> Maxim
>>
>
>
> --
> Best regards,
> Maxim
>

Re: OpenMeetings Back/Restore

[Yah's Global Kingdom]

actual files size is 3.98G Chrome has a limit of 4GB..
Perhaps you can share what the import command does and I can do it manually.


On Tue, Aug 10, 2021 at 9:37 PM Yah's Global Kingdom 
wrote:

> I change the limit to 10GB the file size is 4GB.   File is already on the
> server.
>
> On Tue, Aug 10, 2021 at 6:19 PM Maxim Solodovnik 
> wrote:
>
>> What is your current limit?
>> What is the file size?
>>
>> This limit is applied by browser, so the behavior might be not very user
>> friendly :(
>>
>> On Wed, 11 Aug 2021 at 06:35, Yah's Global Kingdom 
>> wrote:
>>
>>> O.K. Thanks. I thought applied literally to just file uploads, not to
>>> the size of the backup file which is on the server and doesn't need to be
>>> uploaded.   However, when I click the import link and select the back up
>>> file which is in .zip format.  Nothing happens, no status bar on the screen
>>> , nothing.  This is version 6.10.
>>>
>>> On Mon, Aug 9, 2021 at 8:21 PM Maxim Solodovnik 
>>> wrote:
>>>
>>>> Admin->Config
>>>>
>>>> max.upload.size
>>>>
>>>> More info:
>>>> https://openmeetings.apache.org/GeneralConfiguration.html
>>>>
>>>> On Tue, 10 Aug 2021 at 07:04, Yah's Global Kingdom 
>>>> wrote:
>>>>
>>>>> Is there anyway to increase the size of the OpenMeetings
>>>>> Back/Restore.  When I backuped my system it is 4GB..I go to restore it and
>>>>> it says the Max 100MB?  Got to be a way to change that.
>>>>>
>>>>
>>>>
>>>> --
>>>> Best regards,
>>>> Maxim
>>>>
>>>
>>
>> --
>> Best regards,
>> Maxim
>>
>

Re: OpenMeetings Back/Restore

[Yah's Global Kingdom]

I change the limit to 10GB the file size is 4GB.   File is already on the
server.

On Tue, Aug 10, 2021 at 6:19 PM Maxim Solodovnik 
wrote:

> What is your current limit?
> What is the file size?
>
> This limit is applied by browser, so the behavior might be not very user
> friendly :(
>
> On Wed, 11 Aug 2021 at 06:35, Yah's Global Kingdom 
> wrote:
>
>> O.K. Thanks. I thought applied literally to just file uploads, not to the
>> size of the backup file which is on the server and doesn't need to be
>> uploaded.   However, when I click the import link and select the back up
>> file which is in .zip format.  Nothing happens, no status bar on the screen
>> , nothing.  This is version 6.10.
>>
>> On Mon, Aug 9, 2021 at 8:21 PM Maxim Solodovnik 
>> wrote:
>>
>>> Admin->Config
>>>
>>> max.upload.size
>>>
>>> More info:
>>> https://openmeetings.apache.org/GeneralConfiguration.html
>>>
>>> On Tue, 10 Aug 2021 at 07:04, Yah's Global Kingdom 
>>> wrote:
>>>
>>>> Is there anyway to increase the size of the OpenMeetings Back/Restore.
>>>> When I backuped my system it is 4GB..I go to restore it and it says the Max
>>>> 100MB?  Got to be a way to change that.
>>>>
>>>
>>>
>>> --
>>> Best regards,
>>> Maxim
>>>
>>
>
> --
> Best regards,
> Maxim
>

Re: OpenMeetings Back/Restore

[Yah's Global Kingdom]

O.K. Thanks. I thought applied literally to just file uploads, not to the
size of the backup file which is on the server and doesn't need to be
uploaded.   However, when I click the import link and select the back up
file which is in .zip format.  Nothing happens, no status bar on the screen
, nothing.  This is version 6.10.

On Mon, Aug 9, 2021 at 8:21 PM Maxim Solodovnik 
wrote:

> Admin->Config
>
> max.upload.size
>
> More info:
> https://openmeetings.apache.org/GeneralConfiguration.html
>
> On Tue, 10 Aug 2021 at 07:04, Yah's Global Kingdom 
> wrote:
>
>> Is there anyway to increase the size of the OpenMeetings Back/Restore.
>> When I backuped my system it is 4GB..I go to restore it and it says the Max
>> 100MB?  Got to be a way to change that.
>>
>
>
> --
> Best regards,
> Maxim
>

OpenMeetings Back/Restore

[Yah's Global Kingdom]

Is there anyway to increase the size of the OpenMeetings Back/Restore.
When I backuped my system it is 4GB..I go to restore it and it says the Max
100MB?  Got to be a way to change that.

Re: Status of SIP Integration in OM 6.00

[Yah's Global Kingdom]

I would think that the conferencing bridge would be able to handle many to
one connections as that is the ultimate purpose conferencing...I will have
to check it out next week, right now I am in a temp location and have to
move back into my office and set everything back up again. I am hoping that
using SIP as the protocol that it utilizing the conference bridge which
would be fantastic.

Miles

On Fri, Apr 16, 2021 at 1:21 PM Ali Alhaidary 
wrote:

> Yes, running the latest build of 6.1.0, however it was long and complex
> learning experience for us, and, it depends on the subscription and number
> of lines offered, and, for some reason that I did not dig too much into,
> server is overloaded, and unfortunately, in our part of the globe, no
> operator is willing to offer the service, so we limited lessons to
> internet, but other than that, it worked very good two way audio.
>
> Ali
> On 4/16/21 10:01 PM, Yah's Global Kingdom wrote:
>
> Thanks Ali,
>
> I have to look at my notes it has been a minute.  I see emails that you
> are running 6.+ Are multiple user able to call in to your room conference
> using SIP?
>
> On Thu, Apr 15, 2021 at 2:04 PM Ali Alhaidary 
> wrote:
>
>> upgrading is a long process, but certainly worth each and every minute :-)
>>
>> Ali
>> On 4/15/21 11:40 PM, Yah's Global Kingdom wrote:
>>
>> Ok thanks I will upgrade and test
>>
>> On Tue, Apr 13, 2021 at 5:29 PM Maxim Solodovnik 
>> wrote:
>>
>>> I believe so :)
>>> But my testing abilities are limited :)
>>>
>>> And SIP-video is not implemented
>>>
>>> from mobile (sorry for typos ;)
>>>
>>>
>>> On Wed, Apr 14, 2021, 05:11 Yah's Global Kingdom 
>>> wrote:
>>>
>>>> Maxim what exactly does that mean,  Can more than one person, call into
>>>> the room SIMULTANEOUSILY using SIP?
>>>>
>>>> On Sat, Apr 10, 2021 at 7:27 PM Maxim Solodovnik 
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> According to my tests 2-way audio-only SIP should work as expected :)
>>>>>
>>>>> from mobile (sorry for typos ;)
>>>>>
>>>>>
>>>>> On Thu, Apr 8, 2021, 04:20 Yah's Global Kingdom 
>>>>> wrote:
>>>>>
>>>>>> Has this been implemented yet?
>>>>>>
>>>>>

Re: Status of SIP Integration in OM 6.00

[Yah's Global Kingdom]

Thanks Ali,

I have to look at my notes it has been a minute.  I see emails that you are
running 6.+ Are multiple user able to call in to your room conference using
SIP?

On Thu, Apr 15, 2021 at 2:04 PM Ali Alhaidary 
wrote:

> upgrading is a long process, but certainly worth each and every minute :-)
>
> Ali
> On 4/15/21 11:40 PM, Yah's Global Kingdom wrote:
>
> Ok thanks I will upgrade and test
>
> On Tue, Apr 13, 2021 at 5:29 PM Maxim Solodovnik 
> wrote:
>
>> I believe so :)
>> But my testing abilities are limited :)
>>
>> And SIP-video is not implemented
>>
>> from mobile (sorry for typos ;)
>>
>>
>> On Wed, Apr 14, 2021, 05:11 Yah's Global Kingdom 
>> wrote:
>>
>>> Maxim what exactly does that mean,  Can more than one person, call into
>>> the room SIMULTANEOUSILY using SIP?
>>>
>>> On Sat, Apr 10, 2021 at 7:27 PM Maxim Solodovnik 
>>> wrote:
>>>
>>>> Hello,
>>>>
>>>> According to my tests 2-way audio-only SIP should work as expected :)
>>>>
>>>> from mobile (sorry for typos ;)
>>>>
>>>>
>>>> On Thu, Apr 8, 2021, 04:20 Yah's Global Kingdom 
>>>> wrote:
>>>>
>>>>> Has this been implemented yet?
>>>>>
>>>>

Re: Status of SIP Integration in OM 6.00

[Yah's Global Kingdom]

Ok thanks I will upgrade and test

On Tue, Apr 13, 2021 at 5:29 PM Maxim Solodovnik 
wrote:

> I believe so :)
> But my testing abilities are limited :)
>
> And SIP-video is not implemented
>
> from mobile (sorry for typos ;)
>
>
> On Wed, Apr 14, 2021, 05:11 Yah's Global Kingdom 
> wrote:
>
>> Maxim what exactly does that mean,  Can more than one person, call into
>> the room SIMULTANEOUSILY using SIP?
>>
>> On Sat, Apr 10, 2021 at 7:27 PM Maxim Solodovnik 
>> wrote:
>>
>>> Hello,
>>>
>>> According to my tests 2-way audio-only SIP should work as expected :)
>>>
>>> from mobile (sorry for typos ;)
>>>
>>>
>>> On Thu, Apr 8, 2021, 04:20 Yah's Global Kingdom 
>>> wrote:
>>>
>>>> Has this been implemented yet?
>>>>
>>>

Re: Status of SIP Integration in OM 6.00

[Yah's Global Kingdom]

Maxim what exactly does that mean,  Can more than one person, call into the
room SIMULTANEOUSILY using SIP?

On Sat, Apr 10, 2021 at 7:27 PM Maxim Solodovnik 
wrote:

> Hello,
>
> According to my tests 2-way audio-only SIP should work as expected :)
>
> from mobile (sorry for typos ;)
>
>
> On Thu, Apr 8, 2021, 04:20 Yah's Global Kingdom  wrote:
>
>> Has this been implemented yet?
>>
>

Status of SIP Integration in OM 6.00

[Yah's Global Kingdom]

Has this been implemented yet?

Upgrade and testing OM 5.1

[Yah's Global Kingdom]

Hello openmeetings users,
I have tried to upload a video that is approx 300MBs.  This is the error I
am getting:
process: convert to MP4 :: 05c584c2-3b5a-4869-87cd-d1d196a228d8 command:
/usr/bin/ffmpeg -y -i /opt/open504/temp/video8949136119815950200.mp4 -c:v
h264 -c:a aac -pix_fmt yuv420p -vf pad=ceil(iw/2)*2:ceil(ih/2)*2
/opt/open504/webapps/openmeetings/data/upload/files/05c584c2-3b5a-4869-87cd-d1d196a228d8/05c584c2-3b5a-4869-87cd-d1d196a228d8.mp4
exception: java.lang.IllegalThreadStateException: process hasn't exited
error: Exception after 00:20:00.003 of work; process hasn't exited
exitValue: -1 optional: false out:

Additionally, @maxim  I implemented the asterisk server in OM 5.1.  I just
can't remember how to dial the conference room or how I would have to
configure a sip client to do so.  Any ideas?

Re: Server Lost Connection Error

[Yah's Global Kingdom]

Here is the Jira  https://issues.apache.org/jira/browse/OPENMEETINGS-2402

On Mon, Jun 29, 2020 at 6:24 PM Maxim Solodovnik 
wrote:

> This one seems to be reproducible :(((
> Investigating it
> Would appreciate if you can file JIRA
>
> On Mon, 29 Jun 2020 at 09:23, Yah's Global Kingdom 
> wrote:
>
>> Not sure what steps you want.
>> 1 Go into a room,
>> 2. Select files
>> 3. Select filed to upload
>> 4. file.mp4 size 349mb
>> 5 click start upload
>> I have been successful upload files of 150 MB.
>>
>> On Sun, Jun 28, 2020 at 8:27 AM Maxim Solodovnik 
>> wrote:
>>
>>> What are your steps?
>>>
>>> Default Tomcat maxPostSize is 2MB
>>> I definitely can upload more than 2MB
>>>
>>> You can add maxPostSize attribute to
>>> https://github.com/apache/openmeetings/blob/master/openmeetings-server/src/main/assembly/conf/server.xml#L55
>>>  (as
>>> described here http://tomcat.apache.org/tomcat-9.0-doc/config/http.html)
>>> But I doubt it will help
>>>
>>> Would appreciate to get steps ...
>>>
>>> On Sun, 28 Jun 2020 at 21:43, Yah's Global Kingdom 
>>> wrote:
>>>
>>>> This was working before but it had to be adjusted in the
>>>> jee-container.xml, I believe that file was tied to red5-server but I am not
>>>> 100% sure of that.  But it does not exist under this version that I can
>>>> find.
>>>>
>>>> On Sun, Jun 28, 2020 at 7:41 AM Yah's Global Kingdom 
>>>> wrote:
>>>>
>>>>> actually 200 20GB
>>>>>
>>>>> On Sun, Jun 28, 2020 at 7:39 AM Yah's Global Kingdom <
>>>>> yahs...@gmail.com> wrote:
>>>>>
>>>>>> 2GB
>>>>>>
>>>>>> On Sun, Jun 28, 2020 at 1:52 AM Maxim Solodovnik <
>>>>>> solomax...@gmail.com> wrote:
>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> what do you mean by "I have already increased the max upload size"
>>>>>>> Admin->Config->"max.upload.size" ?
>>>>>>>
>>>>>>> What is your value?
>>>>>>>
>>>>>>> On Sat, 27 Jun 2020 at 23:11, Yah's Global Kingdom <
>>>>>>> yahs...@gmail.com> wrote:
>>>>>>>
>>>>>>>> Name
>>>>>>>> Glory To YAH
>>>>>>>> Version
>>>>>>>> 5.0.0-M4
>>>>>>>> Revision
>>>>>>>> 9753e61
>>>>>>>> Build date
>>>>>>>> 2020-04-19T03:54:23Z
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> I am trying to upload files larger than 100MB, I have already
>>>>>>>> increased the max upload size, however, I am still getting server lost
>>>>>>>> connection errors.  Is there another time-out that needs to be changed?
>>>>>>>> the jee-container.xml does not exist in this version that I can find.
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Best regards,
>>>>>>> Maxim
>>>>>>>
>>>>>>
>>>
>>> --
>>> Best regards,
>>> Maxim
>>>
>>
>
> --
> Best regards,
> Maxim
>

Re: Server Lost Connection Error

[Yah's Global Kingdom]

Not sure what steps you want.
1 Go into a room,
2. Select files
3. Select filed to upload
4. file.mp4 size 349mb
5 click start upload
I have been successful upload files of 150 MB.

On Sun, Jun 28, 2020 at 8:27 AM Maxim Solodovnik 
wrote:

> What are your steps?
>
> Default Tomcat maxPostSize is 2MB
> I definitely can upload more than 2MB
>
> You can add maxPostSize attribute to
> https://github.com/apache/openmeetings/blob/master/openmeetings-server/src/main/assembly/conf/server.xml#L55
>  (as
> described here http://tomcat.apache.org/tomcat-9.0-doc/config/http.html)
> But I doubt it will help
>
> Would appreciate to get steps ...
>
> On Sun, 28 Jun 2020 at 21:43, Yah's Global Kingdom 
> wrote:
>
>> This was working before but it had to be adjusted in the
>> jee-container.xml, I believe that file was tied to red5-server but I am not
>> 100% sure of that.  But it does not exist under this version that I can
>> find.
>>
>> On Sun, Jun 28, 2020 at 7:41 AM Yah's Global Kingdom 
>> wrote:
>>
>>> actually 200 20GB
>>>
>>> On Sun, Jun 28, 2020 at 7:39 AM Yah's Global Kingdom 
>>> wrote:
>>>
>>>> 2GB
>>>>
>>>> On Sun, Jun 28, 2020 at 1:52 AM Maxim Solodovnik 
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> what do you mean by "I have already increased the max upload size"
>>>>> Admin->Config->"max.upload.size" ?
>>>>>
>>>>> What is your value?
>>>>>
>>>>> On Sat, 27 Jun 2020 at 23:11, Yah's Global Kingdom 
>>>>> wrote:
>>>>>
>>>>>> Name
>>>>>> Glory To YAH
>>>>>> Version
>>>>>> 5.0.0-M4
>>>>>> Revision
>>>>>> 9753e61
>>>>>> Build date
>>>>>> 2020-04-19T03:54:23Z
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> I am trying to upload files larger than 100MB, I have already
>>>>>> increased the max upload size, however, I am still getting server lost
>>>>>> connection errors.  Is there another time-out that needs to be changed?
>>>>>> the jee-container.xml does not exist in this version that I can find.
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Best regards,
>>>>> Maxim
>>>>>
>>>>
>
> --
> Best regards,
> Maxim
>

Re: Server Lost Connection Error

[Yah's Global Kingdom]

This was working before but it had to be adjusted in the jee-container.xml,
I believe that file was tied to red5-server but I am not 100% sure of
that.  But it does not exist under this version that I can find.

On Sun, Jun 28, 2020 at 7:41 AM Yah's Global Kingdom 
wrote:

> actually 200 20GB
>
> On Sun, Jun 28, 2020 at 7:39 AM Yah's Global Kingdom 
> wrote:
>
>> 2GB
>>
>> On Sun, Jun 28, 2020 at 1:52 AM Maxim Solodovnik 
>> wrote:
>>
>>> Hello,
>>>
>>> what do you mean by "I have already increased the max upload size"
>>> Admin->Config->"max.upload.size" ?
>>>
>>> What is your value?
>>>
>>> On Sat, 27 Jun 2020 at 23:11, Yah's Global Kingdom 
>>> wrote:
>>>
>>>> Name
>>>> Glory To YAH
>>>> Version
>>>> 5.0.0-M4
>>>> Revision
>>>> 9753e61
>>>> Build date
>>>> 2020-04-19T03:54:23Z
>>>>
>>>>
>>>>
>>>>
>>>> I am trying to upload files larger than 100MB, I have already increased
>>>> the max upload size, however, I am still getting server lost connection
>>>> errors.  Is there another time-out that needs to be changed?  the
>>>> jee-container.xml does not exist in this version that I can find.
>>>>
>>>
>>>
>>> --
>>> Best regards,
>>> Maxim
>>>
>>

Re: Server Lost Connection Error

[Yah's Global Kingdom]

actually 200 20GB

On Sun, Jun 28, 2020 at 7:39 AM Yah's Global Kingdom 
wrote:

> 2GB
>
> On Sun, Jun 28, 2020 at 1:52 AM Maxim Solodovnik 
> wrote:
>
>> Hello,
>>
>> what do you mean by "I have already increased the max upload size"
>> Admin->Config->"max.upload.size" ?
>>
>> What is your value?
>>
>> On Sat, 27 Jun 2020 at 23:11, Yah's Global Kingdom 
>> wrote:
>>
>>> Name
>>> Glory To YAH
>>> Version
>>> 5.0.0-M4
>>> Revision
>>> 9753e61
>>> Build date
>>> 2020-04-19T03:54:23Z
>>>
>>>
>>>
>>>
>>> I am trying to upload files larger than 100MB, I have already increased
>>> the max upload size, however, I am still getting server lost connection
>>> errors.  Is there another time-out that needs to be changed?  the
>>> jee-container.xml does not exist in this version that I can find.
>>>
>>
>>
>> --
>> Best regards,
>> Maxim
>>
>

Re: Server Lost Connection Error

[Yah's Global Kingdom]

2GB

On Sun, Jun 28, 2020 at 1:52 AM Maxim Solodovnik 
wrote:

> Hello,
>
> what do you mean by "I have already increased the max upload size"
> Admin->Config->"max.upload.size" ?
>
> What is your value?
>
> On Sat, 27 Jun 2020 at 23:11, Yah's Global Kingdom 
> wrote:
>
>> Name
>> Glory To YAH
>> Version
>> 5.0.0-M4
>> Revision
>> 9753e61
>> Build date
>> 2020-04-19T03:54:23Z
>>
>>
>>
>>
>> I am trying to upload files larger than 100MB, I have already increased
>> the max upload size, however, I am still getting server lost connection
>> errors.  Is there another time-out that needs to be changed?  the
>> jee-container.xml does not exist in this version that I can find.
>>
>
>
> --
> Best regards,
> Maxim
>

Server Lost Connection Error

[Yah's Global Kingdom]

Name
Glory To YAH
Version
5.0.0-M4
Revision
9753e61
Build date
2020-04-19T03:54:23Z




I am trying to upload files larger than 100MB, I have already increased the
max upload size, however, I am still getting server lost connection
errors.  Is there another time-out that needs to be changed?  the
jee-container.xml does not exist in this version that I can find.

Re: Mobile App

[Yah's Global Kingdom]

Thanks, I wish I could heip but I know nothing about mobile programming


On Wed, Jun 24, 2020 at 9:43 PM Maxim Solodovnik 
wrote:

> We had in plans to create special mobile CSS
> Unfortunately there is not much progress on this :(
>
> On Thu, 25 Jun 2020 at 01:42, Yah's Global Kingdom 
> wrote:
>
>> Yes it works in the mobile browser but as far as my experience has been,
>> you have to resize to get to different things. Esp from a mobile phone.
>>
>> On Tue, Jun 23, 2020 at 7:08 PM Maxim Solodovnik 
>> wrote:
>>
>>> There is no mobile app
>>> Latest version seems to work in mobile browser
>>>
>>> (from mobile, sorry for typos)
>>>
>>> On Wed, Jun 24, 2020, 07:18 Yah's Global Kingdom 
>>> wrote:
>>>
>>>> Is there an Openmeetings Mobile App?  If so how do I get a copy of it?
>>>>
>>>
>
> --
> Best regards,
> Maxim
>

Re: Mobile App

[Yah's Global Kingdom]

Yes it works in the mobile browser but as far as my experience has been,
you have to resize to get to different things. Esp from a mobile phone.

On Tue, Jun 23, 2020 at 7:08 PM Maxim Solodovnik 
wrote:

> There is no mobile app
> Latest version seems to work in mobile browser
>
> (from mobile, sorry for typos)
>
> On Wed, Jun 24, 2020, 07:18 Yah's Global Kingdom 
> wrote:
>
>> Is there an Openmeetings Mobile App?  If so how do I get a copy of it?
>>
>

Mobile App

[Yah's Global Kingdom]

Is there an Openmeetings Mobile App?  If so how do I get a copy of it?

Re: OM5 -> SIP

[Yah's Global Kingdom]

I want to second or third or +1000 on the SIP Implementation for OM5.

On Sat, Jun 20, 2020 at 8:05 AM Maxim Solodovnik 
wrote:

> No news so far
>
> On Sat, 20 Jun 2020 at 17:02, R. Scholz 
> wrote:
>
>> Hello,
>>
>> are there news with the SIP-implemetation in OM5?
>> I got many times this question.
>>
>> Best regards,
>>
>> René
>>
>>
>
> --
> Best regards,
> Maxim
>

Re: SIP CLIENT FOR 5.0

[Yah's Global Kingdom]

thanks Maxim it really is needed


On Sun, Apr 12, 2020 at 11:27 PM Maxim Solodovnik 
wrote:

> I would like to release M4, then check what I can do with SIP integration
> https://issues.apache.org/jira/browse/OPENMEETINGS-2239
>
> On Sun, 12 Apr 2020 at 04:33, Yah's Global Kingdom 
> wrote:
>
>> Maxim, with the corona virus pandemic, it is imperative that we are able
>> to have a sip client for 5.0, ca we move this up in the priorities?
>>
>
>
> --
> Best regards,
> Maxim
>

SIP CLIENT FOR 5.0

[Yah's Global Kingdom]

Maxim, with the corona virus pandemic, it is imperative that we are able to
have a sip client for 5.0, ca we move this up in the priorities?

Re: [DISCUSSION] is it time to drop old crypt mechanisms

[Yah's Global Kingdom]

No Objections here.

On Tue, Dec 17, 2019 at 2:08 AM Maxim Solodovnik 
wrote:

> Hello All,
>
> Some time ago I have changed password hashing code to be more secure
> Same time backward compatibility with old hashing types was added
> I would like to remove support of old hashes in OM 5
>
> Any objections?
>
> --
> WBR
> Maxim aka solomax
>

Clustering

[Yah's Global Kingdom]

What clustering software is recommended to be used with OM?  I am looking
at Apache Httpd or pound.  Pound seems to be the simplest to use, but I
want to know what is already working with OM.

Re: VOIP for 5.0.0.M

[Yah's Global Kingdom]

Thanks Maxim,

I got it to work and you are correct I needed the SpecialSSLHostConfig in
the server.xml

On Thu, Oct 17, 2019 at 2:29 AM Maxim Solodovnik 
wrote:

> Actually there is no need to specify *AprProtocol to use let's encrypt
> certificates without conversions
> Here is simple step-by-step guide:
> https://community.letsencrypt.org/t/using-letsencrypt-certificates-on-tomcat-8-x-on-windows/28548/7
> all you need is "Special SSLHostConfig"
> Documentation is here:
> https://tomcat.apache.org/tomcat-9.0-doc/config/http.html#SSL_Support_-_SSLHostConfig
>
> On Thu, 17 Oct 2019 at 13:16, René Scholz <
> rene.sch...@abakus-edv-systems.de> wrote:
>
>> Hello,
>>
>> hm, that looks complicated. In my configuration it was not necessary to
>> define a protocol like you have done.
>> The error-message shows that the choosen protocol requires a library. Its
>> possible that this is the error, but I dont know
>> if your certificate match to this protocol.
>>
>> I am afraid without deeper knowledge of your certificates and (maybe very
>> complicated and high-secured)
>> network-configuration I have no further idea what goes wrong.
>>
>> I have only rudimentary knowledge about certificates - in my
>> configuration "behind a NAT" the https-certificate
>> was the lesser evil.
>>
>> Best regrads,
>>
>> René
>>
>>
>>
>>
>>
>> Am 16.10.2019 um 15:25 schrieb Yah's Global Kingdom:
>>
>> Rene, I apologize and thanks for your help!  I did use the lines you sent
>> me and changed the necessary information.  .
>> The private key is using http11NioProtocol, the format you provided goes
>> into the Http11AprProtocol section.
>>
>>  I got this error:
>>
>> 16-Oct-2019 05:58:47.266 SEVERE [main]
>> org.apache.catalina.util.LifecycleBase.handleSubClassException Failed to
>> initialize component
>> [Connector[org.apache.coyote.http11.Http11AprProtocol-5443]]
>> org.apache.catalina.LifecycleException: The configured protocol
>> [org.apache.coyote.http11.Http11AprProtocol] requires the APR/native
>> library which is not available
>>
>> When I use the Http11NioProtocol I get this error.   My keystore only has
>> one key in it the private key.
>>
>> 16-Oct-2019 06:05:35.065 INFO [main]
>> org.apache.coyote.AbstractProtocol.init Initializing ProtocolHandler
>> ["http-nio-5080"]
>> 16-Oct-2019 06:05:35.107 INFO [main]
>> org.apache.coyote.AbstractProtocol.init Initializing ProtocolHandler
>> ["https-jsse-nio-5443"]
>> 16-Oct-2019 06:05:35.352 SEVERE [main]
>> org.apache.catalina.util.LifecycleBase.handleSubClassException Failed to
>> initialize component [Connector[HTTP/1.1-5443]]
>> org.apache.catalina.LifecycleException: Protocol handler initialization
>> failed
>> at
>> org.apache.catalina.connector.Connector.initInternal(Connector.java:983)
>> at org.apache.catalina.util.LifecycleBase.init(LifecycleBase.java:136)
>> at
>> org.apache.catalina.core.StandardService.initInternal(StandardService.java:533)
>> at org.apache.catalina.util.LifecycleBase.init(LifecycleBase.java:136)
>> at
>> org.apache.catalina.core.StandardServer.initInternal(StandardServer.java:1059)
>> at org.apache.catalina.util.LifecycleBase.init(LifecycleBase.java:136)
>> at org.apache.catalina.startup.Catalina.load(Catalina.java:584)
>> at org.apache.catalina.startup.Catalina.start(Catalina.java:621)
>> at java.base/jdk.internal.reflect.NativeMethodAccessorImpl.invoke0(Native
>> Method)
>> at
>> java.base/jdk.internal.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:62)
>> at
>> java.base/jdk.internal.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:43)
>> at java.base/java.lang.reflect.Method.invoke(Method.java:566)
>> at org.apache.catalina.startup.Bootstrap.start(Bootstrap.java:344)
>> at org.apache.catalina.startup.Bootstrap.main(Bootstrap.java:475)
>> Caused by: java.lang.IllegalArgumentException: Cannot store
>> non-PrivateKeys
>> at
>> org.apache.tomcat.util.net.AbstractJsseEndpoint.createSSLContext(AbstractJsseEndpoint.java:99)
>> at
>> org.apache.tomcat.util.net.AbstractJsseEndpoint.initialiseSsl(AbstractJsseEndpoint.java:71)
>> at org.apache.tomcat.util.net.NioEndpoint.bind(NioEndpoint.java:218)
>> at
>> org.apache.tomcat.util.net.AbstractEndpoint.bindWithCleanup(AbstractEndpoint.java:1124)
>> at
>> org.apache.tomcat.util.net.AbstractEndpoint.init(AbstractEndpoint.java:1137)
>> a

Re: VOIP for 5.0.0.M

[Yah's Global Kingdom]

Thanks Ren'e,  I got it to work I appreciate your taking the time to help
me.  That brings up another question.  Do your camera work from behind a
NAT without a stun or turn server?

On Thu, Oct 17, 2019 at 2:16 AM René Scholz <
rene.sch...@abakus-edv-systems.de> wrote:

> Hello,
>
> hm, that looks complicated. In my configuration it was not necessary to
> define a protocol like you have done.
> The error-message shows that the choosen protocol requires a library. Its
> possible that this is the error, but I dont know
> if your certificate match to this protocol.
>
> I am afraid without deeper knowledge of your certificates and (maybe very
> complicated and high-secured)
> network-configuration I have no further idea what goes wrong.
>
> I have only rudimentary knowledge about certificates - in my configuration
> "behind a NAT" the https-certificate
> was the lesser evil.
>
> Best regrads,
>
> René
>
>
>
>
>
> Am 16.10.2019 um 15:25 schrieb Yah's Global Kingdom:
>
> Rene, I apologize and thanks for your help!  I did use the lines you sent
> me and changed the necessary information.  .
> The private key is using http11NioProtocol, the format you provided goes
> into the Http11AprProtocol section.
>
>  I got this error:
>
> 16-Oct-2019 05:58:47.266 SEVERE [main]
> org.apache.catalina.util.LifecycleBase.handleSubClassException Failed to
> initialize component
> [Connector[org.apache.coyote.http11.Http11AprProtocol-5443]]
> org.apache.catalina.LifecycleException: The configured protocol
> [org.apache.coyote.http11.Http11AprProtocol] requires the APR/native
> library which is not available
>
> When I use the Http11NioProtocol I get this error.   My keystore only has
> one key in it the private key.
>
> 16-Oct-2019 06:05:35.065 INFO [main]
> org.apache.coyote.AbstractProtocol.init Initializing ProtocolHandler
> ["http-nio-5080"]
> 16-Oct-2019 06:05:35.107 INFO [main]
> org.apache.coyote.AbstractProtocol.init Initializing ProtocolHandler
> ["https-jsse-nio-5443"]
> 16-Oct-2019 06:05:35.352 SEVERE [main]
> org.apache.catalina.util.LifecycleBase.handleSubClassException Failed to
> initialize component [Connector[HTTP/1.1-5443]]
> org.apache.catalina.LifecycleException: Protocol handler initialization
> failed
> at org.apache.catalina.connector.Connector.initInternal(Connector.java:983)
> at org.apache.catalina.util.LifecycleBase.init(LifecycleBase.java:136)
> at
> org.apache.catalina.core.StandardService.initInternal(StandardService.java:533)
> at org.apache.catalina.util.LifecycleBase.init(LifecycleBase.java:136)
> at
> org.apache.catalina.core.StandardServer.initInternal(StandardServer.java:1059)
> at org.apache.catalina.util.LifecycleBase.init(LifecycleBase.java:136)
> at org.apache.catalina.startup.Catalina.load(Catalina.java:584)
> at org.apache.catalina.startup.Catalina.start(Catalina.java:621)
> at java.base/jdk.internal.reflect.NativeMethodAccessorImpl.invoke0(Native
> Method)
> at
> java.base/jdk.internal.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:62)
> at
> java.base/jdk.internal.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:43)
> at java.base/java.lang.reflect.Method.invoke(Method.java:566)
> at org.apache.catalina.startup.Bootstrap.start(Bootstrap.java:344)
> at org.apache.catalina.startup.Bootstrap.main(Bootstrap.java:475)
> Caused by: java.lang.IllegalArgumentException: Cannot store non-PrivateKeys
> at
> org.apache.tomcat.util.net.AbstractJsseEndpoint.createSSLContext(AbstractJsseEndpoint.java:99)
> at
> org.apache.tomcat.util.net.AbstractJsseEndpoint.initialiseSsl(AbstractJsseEndpoint.java:71)
> at org.apache.tomcat.util.net.NioEndpoint.bind(NioEndpoint.java:218)
> at
> org.apache.tomcat.util.net.AbstractEndpoint.bindWithCleanup(AbstractEndpoint.java:1124)
> at
> org.apache.tomcat.util.net.AbstractEndpoint.init(AbstractEndpoint.java:1137)
> at org.apache.coyote.AbstractProtocol.init(AbstractProtocol.java:574)
> at
> org.apache.coyote.http11.AbstractHttp11Protocol.init(AbstractHttp11Protocol.java:74)
> at org.apache.catalina.connector.Connector.initInternal(Connector.java:980)
> ... 13 more
> Caused by: java.security.KeyStoreException: Cannot store non-PrivateKeys
> at
> java.base/sun.security.provider.JavaKeyStore.engineSetKeyEntry(JavaKeyStore.java:262)
> at
> java.base/sun.security.util.KeyStoreDelegator.engineSetKeyEntry(KeyStoreDelegator.java:111)
> at java.base/java.security.KeyStore.setKeyEntry(KeyStore.java:1174)
> at
> org.apache.tomcat.util.net.SSLUtilBase.getKeyManagers(SSLUtilBase.java:324)
> at
> org.apache.tomcat.util.net.SSLUtilBase.createSSLContext(SSLUtilBase.java:247)
> at
> org.apache.tom

Re: VOIP for 5.0.0.M

[Yah's Global Kingdom]

Rene, I apologize and thanks for your help!  I did use the lines you sent
me and changed the necessary information.  .
The private key is using http11NioProtocol, the format you provided goes
into the Http11AprProtocol section.

 I got this error:

16-Oct-2019 05:58:47.266 SEVERE [main]
org.apache.catalina.util.LifecycleBase.handleSubClassException Failed to
initialize component
[Connector[org.apache.coyote.http11.Http11AprProtocol-5443]]
org.apache.catalina.LifecycleException: The configured protocol
[org.apache.coyote.http11.Http11AprProtocol] requires the APR/native
library which is not available

When I use the Http11NioProtocol I get this error.   My keystore only has
one key in it the private key.

16-Oct-2019 06:05:35.065 INFO [main]
org.apache.coyote.AbstractProtocol.init Initializing ProtocolHandler
["http-nio-5080"]
16-Oct-2019 06:05:35.107 INFO [main]
org.apache.coyote.AbstractProtocol.init Initializing ProtocolHandler
["https-jsse-nio-5443"]
16-Oct-2019 06:05:35.352 SEVERE [main]
org.apache.catalina.util.LifecycleBase.handleSubClassException Failed to
initialize component [Connector[HTTP/1.1-5443]]
org.apache.catalina.LifecycleException: Protocol handler initialization
failed
at org.apache.catalina.connector.Connector.initInternal(Connector.java:983)
at org.apache.catalina.util.LifecycleBase.init(LifecycleBase.java:136)
at
org.apache.catalina.core.StandardService.initInternal(StandardService.java:533)
at org.apache.catalina.util.LifecycleBase.init(LifecycleBase.java:136)
at
org.apache.catalina.core.StandardServer.initInternal(StandardServer.java:1059)
at org.apache.catalina.util.LifecycleBase.init(LifecycleBase.java:136)
at org.apache.catalina.startup.Catalina.load(Catalina.java:584)
at org.apache.catalina.startup.Catalina.start(Catalina.java:621)
at java.base/jdk.internal.reflect.NativeMethodAccessorImpl.invoke0(Native
Method)
at
java.base/jdk.internal.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:62)
at
java.base/jdk.internal.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:43)
at java.base/java.lang.reflect.Method.invoke(Method.java:566)
at org.apache.catalina.startup.Bootstrap.start(Bootstrap.java:344)
at org.apache.catalina.startup.Bootstrap.main(Bootstrap.java:475)
Caused by: java.lang.IllegalArgumentException: Cannot store non-PrivateKeys
at
org.apache.tomcat.util.net.AbstractJsseEndpoint.createSSLContext(AbstractJsseEndpoint.java:99)
at
org.apache.tomcat.util.net.AbstractJsseEndpoint.initialiseSsl(AbstractJsseEndpoint.java:71)
at org.apache.tomcat.util.net.NioEndpoint.bind(NioEndpoint.java:218)
at
org.apache.tomcat.util.net.AbstractEndpoint.bindWithCleanup(AbstractEndpoint.java:1124)
at
org.apache.tomcat.util.net.AbstractEndpoint.init(AbstractEndpoint.java:1137)
at org.apache.coyote.AbstractProtocol.init(AbstractProtocol.java:574)
at
org.apache.coyote.http11.AbstractHttp11Protocol.init(AbstractHttp11Protocol.java:74)
at org.apache.catalina.connector.Connector.initInternal(Connector.java:980)
... 13 more
Caused by: java.security.KeyStoreException: Cannot store non-PrivateKeys
at
java.base/sun.security.provider.JavaKeyStore.engineSetKeyEntry(JavaKeyStore.java:262)
at
java.base/sun.security.util.KeyStoreDelegator.engineSetKeyEntry(KeyStoreDelegator.java:111)
at java.base/java.security.KeyStore.setKeyEntry(KeyStore.java:1174)
at
org.apache.tomcat.util.net.SSLUtilBase.getKeyManagers(SSLUtilBase.java:324)
at
org.apache.tomcat.util.net.SSLUtilBase.createSSLContext(SSLUtilBase.java:247)
at
org.apache.tomcat.util.net.AbstractJsseEndpoint.createSSLContext(AbstractJsseEndpoint.java:97)
... 20 more
here is the relevant part of my server.xml that includes the original
configuration plus the two configurations I have tried to use to get this
to work commented out.   is my servername.domainname.org perhaps you
can look and see what I have done wrong.




  
  
  
  
  
  
  
  

  
  







   


-->




On Wed, Oct 16, 2019 at 1:50 AM René Scholz <
rene.sch...@abakus-edv-systems.de> wrote:

> Hello,
>
> why don't you try out the config-part I sent you?
> Make a backup of your sever.xml, edit the part for your connector-port,
> restart your OM, pray a little bit and open your browser with https and
> your port.
>
> Whats the result?
>
> When you mean that something goes wrong replace it with your backuped
> server.xml.
>
> Best regards,
>
> René
>
> Am 15.10.2019 um 22:30 schrieb Yah's Global Kingdom:
>
> Your saying I don't have to use a keystore with these certs?
>
> On Mon, Oct 14, 2019 at 4:06 AM Maxim Solodovnik 
> wrote:
>
>> With this config import is redundant
>> you can use your keys as-is :)
>>
>> On Sun, 13 Oct 2019 at 21:11, Yah's Global Kingdom 
>> wrote:
>>
>>> Thanks for the information, if I might ask which of these keys did you
>>> impo

Re: VOIP for 5.0.0.M

[Yah's Global Kingdom]

Your saying I don't have to use a keystore with these certs?

On Mon, Oct 14, 2019 at 4:06 AM Maxim Solodovnik 
wrote:

> With this config import is redundant
> you can use your keys as-is :)
>
> On Sun, 13 Oct 2019 at 21:11, Yah's Global Kingdom 
> wrote:
>
>> Thanks for the information, if I might ask which of these keys did you
>> import into your keystore for openmeetings?
>>
>> On Sat, Oct 12, 2019 at 1:36 PM R. Scholz <
>> rene.sch...@abakus-edv-systems.de> wrote:
>>
>>> Hello,
>>>
>>> this is the part in my server.xml in the conf-dir of my openmeeting I
>>> use without problems:
>>>
>>> >> SSLEnabled="true">
>>>   
>>> >>  certificateKeyFile="/etc/letsencrypt/live/
>>> subdomain.domain.de/privkey.pem"
>>>  certificateChainFile="/etc/letsencrypt/live/
>>> subdomain.domain.de/fullchain.pem" />
>>>   
>>> 
>>>
>>> With best regards,
>>>
>>> René
>>>
>>>
>>>
>>> Am 12.10.2019 um 17:35 schrieb Yah's Global Kingdom:
>>>
>>> Ok understood for the VOIP implementation.  Hopefully, there will be
>>> time for in the near future as it was feature that was really appreciated
>>> and used.
>>> On a different note.  I am using LetsEncrypt for ssl certificates.  The
>>> wiki at https://openmeetings.apache.org/HTTPS.html does not seem to
>>> apply as you can not submit a .csr file to lets encrypt and it only works
>>> on port 443. I have changed /conf/server.conf to 443 but the server still
>>> refuses to connect.  Are there any instructions for how to make OM 5.0.0.M2
>>> OR M3 work with LetEncrypt and Certbot?  Thanks for all your help Maxim.
>>>
>>> On Thu, Oct 10, 2019 at 12:45 PM Maxim Solodovnik 
>>> wrote:
>>>
>>>> Yes, sure
>>>> unfortunately my time is very limited
>>>> not sure i can provide any estimates
>>>>
>>>> On Thu, 10 Oct 2019 at 09:16, Yah's Global Kingdom 
>>>> wrote:
>>>>
>>>>> Is there a plan to implement VOIP for this version of Openmeetings?
>>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>
>>>
>
> --
> WBR
> Maxim aka solomax
>

Re: VOIP for 5.0.0.M

[Yah's Global Kingdom]

Thanks for the information, if I might ask which of these keys did you
import into your keystore for openmeetings?

On Sat, Oct 12, 2019 at 1:36 PM R. Scholz 
wrote:

> Hello,
>
> this is the part in my server.xml in the conf-dir of my openmeeting I use
> without problems:
>
>  SSLEnabled="true">
>   
>   certificateKeyFile="/etc/letsencrypt/live/
> subdomain.domain.de/privkey.pem"
>  certificateChainFile="/etc/letsencrypt/live/
> subdomain.domain.de/fullchain.pem" />
>   
> 
>
> With best regards,
>
> René
>
>
>
> Am 12.10.2019 um 17:35 schrieb Yah's Global Kingdom:
>
> Ok understood for the VOIP implementation.  Hopefully, there will be time
> for in the near future as it was feature that was really appreciated and
> used.
> On a different note.  I am using LetsEncrypt for ssl certificates.  The
> wiki at https://openmeetings.apache.org/HTTPS.html does not seem to apply
> as you can not submit a .csr file to lets encrypt and it only works on port
> 443. I have changed /conf/server.conf to 443 but the server still refuses
> to connect.  Are there any instructions for how to make OM 5.0.0.M2 OR M3
> work with LetEncrypt and Certbot?  Thanks for all your help Maxim.
>
> On Thu, Oct 10, 2019 at 12:45 PM Maxim Solodovnik 
> wrote:
>
>> Yes, sure
>> unfortunately my time is very limited
>> not sure i can provide any estimates
>>
>> On Thu, 10 Oct 2019 at 09:16, Yah's Global Kingdom 
>> wrote:
>>
>>> Is there a plan to implement VOIP for this version of Openmeetings?
>>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>
>

Re: VOIP for 5.0.0.M

[Yah's Global Kingdom]

Ok understood for the VOIP implementation.  Hopefully, there will be time
for in the near future as it was feature that was really appreciated and
used.
On a different note.  I am using LetsEncrypt for ssl certificates.  The
wiki at https://openmeetings.apache.org/HTTPS.html does not seem to apply
as you can not submit a .csr file to lets encrypt and it only works on port
443. I have changed /conf/server.conf to 443 but the server still refuses
to connect.  Are there any instructions for how to make OM 5.0.0.M2 OR M3
work with LetEncrypt and Certbot?  Thanks for all your help Maxim.

On Thu, Oct 10, 2019 at 12:45 PM Maxim Solodovnik 
wrote:

> Yes, sure
> unfortunately my time is very limited
> not sure i can provide any estimates
>
> On Thu, 10 Oct 2019 at 09:16, Yah's Global Kingdom 
> wrote:
>
>> Is there a plan to implement VOIP for this version of Openmeetings?
>>
>
>
> --
> WBR
> Maxim aka solomax
>

VOIP for 5.0.0.M

[Yah's Global Kingdom]

Is there a plan to implement VOIP for this version of Openmeetings?