Sorry my brother for the late reply
Happy New Years to you and Gloria
-Original Message-
From: Alvaro [mailto:zurca...@gmail.com]
Sent: Friday, January 01, 2016 12:39 AM
To: user@openmeetings.apache.org
Subject: Happy Year 2016!
Happy Year 2016 to all!
Alvaro
---
This email has bee
n Sun, Nov 1, 2015 at 9:29 PM, Horace Miles
wrote:
I can get into the conference room just fine. However dialing out does not
work.
Also should callers that have dialed in on their phone be able to talk in the
conference or listen only?
Does anyone have any sample code to dial out of
Hello Openmeetings Users,
I still consider myself to be a noob at this. Trying to rap my head around
all the different technology.
I am running Openmeetings 3.0.2. release on Ubuntu 12.04 LTS
Core 2 Quad CPU Q6700 @ 2.66GHz x4 32 bit OS
I have successfully integrated SIP using the SIP
I can get into the conference room just fine. However dialing out does not
work.
Also should callers that have dialed in on their phone be able to talk in
the conference or listen only?
Does anyone have any sample code to dial out of conference rooms in
Openmeeting?
---
This email has bee
Maxim,
Have there been any updates to the SIP integration since the 3.0 instructions
were put out. i.e. any way to kick users that are connected via sip etc?
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
Sent: Saturday, September 05, 2015 9:40 PM
To: Awotipe Oluwaseun
Cc: Openmeetings
Asterisk does not use the openmeetings database, it uses it’s own database
that is installed when you install asterisk
From: Pierre Smits [mailto:pierre.sm...@gmail.com]
Sent: Thursday, September 03, 2015 1:10 PM
To: user@openmeetings.apache.org
Subject: Fwd: Sip integration
FYI
To keep
ri, Feb 13, 2015 at 4:28 PM, Horace Miles
wrote:
What I specifically have done.
Started with a openmeeting 2.0 iso install.
Removed the 2.0 database and red52 openmeetings folders
Created empty openmeetings database as open53
Download openmeetings3.03 zip file
Unzipped into /usr/lib/re
-start
I'm not sure how can I help without being able to reproduce :(
Maybe any specific steps?
On Fri, Feb 13, 2015 at 10:11 AM, Horace Miles
wrote:
Ok thank Maxim,
I have no clue where to go from here.. I don’t know how it ties the codec.rate
and the microphonerate variables. Someh
, February 12, 2015 8:35 PM
To: Openmeetings user-list
Subject: Re: FW: red5sip error
works as expected here without any modifications
On Fri, Feb 13, 2015 at 8:55 AM, Horace Miles
wrote:
Maxim,
Thanks the SVN checkout fixed the build error.. but the redsip will not start
because there seems to
Maxim,
Thanks the SVN checkout fixed the build error.. but the redsip will not start
because there seems to be a problem between the codec.rate and the
microphonebestrate variable/?
root@omeet:/usr/lib/red5sip/red5sip_3.0# sudo ./red5sip.sh
Starting Red5SIP
12 Feb 19:14:02 - [main]:[ERROR] o.r.
When I start red5sip I I get an error at
Red5.codec.rate=22
Microphoneratebest = 22 in the public/config.xml
It doesn't appear to think these two are the same variables.
Hi, I am trying to build the red5sip client using the integration steps
for openmeetings 303. I have successfully upgraded from 2.0 to 3.03.
However when trying to enable sip and do the sip integration I get the
following errors when trying to ant the red5sip. Can anyone help me
correct this err
What is "php5-ffmpeg"? And how is it connected with OM?
WBR, Maxim
(from mobile, sorry for the typos)
On Feb 3, 2015 7:17 AM, "Horace Miles" wrote:
Has anyone gotten openmeetings to successful load with php5-ffmpeg? I keep
getting missing headers.
Has anyone gotten openmeetings to successful load with php5-ffmpeg? I
keep getting missing headers.
http://openmeetings.apache.org/WebappNamePath.html
first block
On Fri, Nov 28, 2014 at 8:49 PM, Horace Miles
wrote:
Thank you very much.
From: Peter Dähn [mailto:da...@vcrp.de]
Sent: Friday, November 28, 2014 5:56 AM
To: user@openmeetings.apache.org
Subject: Re: Changing default logo
Hi
According to the instructions below, multiple instances of openmeetings
can be run on the same server. Is it possible to make the different
instances use a different database? If so, how?
Running multiple Openmeetings instances on the same server
If you want to run multiple Openmeetings inst
p;api=v2
Greetings Peter
Am 28.11.2014 um 13:41 schrieb Horace Miles:
Can anyone direct me to the instructions for how to change the logo in
Openmeetings? I thought I saw these instructions long ago, but cant seem to
find them now..
Thanks Miles
Can anyone direct me to the instructions for how to change the logo in
Openmeetings? I thought I saw these instructions long ago, but cant
seem to find them now..
Thanks Miles
Can red5 server run more than one webapp on port 1935?
Thank you
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
Sent: Wednesday, September 17, 2014 6:47 AM
To: Openmeetings user-list
Subject: Re: Red5 Version
3.0.0-3.0.2 uses 1.0.0 rev.4393
3.0.3 uses 1.0.3 RELEASE
On 17 September 2014 20:24, Horace Miles
wrote:
What version of
What version of red5 is openmeetings 3.0 based upon>?
Is there a way to change the room layout in openmeetings. i.e. If
someone does not want the whiteboard, is there a way to make the chat
window grow dynamically to fill up that space?
September 2014 20:06, Horace Miles wrote:
Hi Everyone,
I am experiencing the following problem. I am using Openmeetings 3.00. I have
integrated VOIP. Still trying to learn all that Openmeetings has to offer.
When I make a call from an ext in Asterisk, the number called is automatically
Hi Everyone,
I am experiencing the following problem. I am using Openmeetings 3.00.
I have integrated VOIP. Still trying to learn all that Openmeetings
has to offer.
When I make a call from an ext in Asterisk, the number called is
automatically placed on the blacklist. Does anyone know
...@gmail.com]
Sent: Wednesday, August 27, 2014 4:50 AM
To: Openmeetings user-list
Subject: Re: OM 3.0 VOIP Integration.
I believe Sip Transport should display caller count as follows:
Sip Transport (XX)
where XX is count of SIP users in the room
On 27 August 2014 01:40, Horace Miles wrote:
For
For incoming calls into the conference room, should the openmeeting show
the caller in the conference. I get sound to the phone, but the caller
does not show up in the room. Is this by design?
Is the nomenclature for calling into a conference
exten@ip address or
ex...@fqdn.tld ?
trace?
what version are you using?
On 21 August 2014 20:22, Horace Miles wrote:
When uploading a mpeg4 I receive the following error:
key: processFLV 0
process: convertToFLV
command: null
exception: java.lang.NullPointerException
error: null
exitValue: java.lang.NullPointerException
08-22 07:08:48.585 o.a.o.r.r.ScopeApplicationAdapter:2623
[NioProcessor-7] - getSipConferenceMembersNumber: (0)
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
Sent: Thursday, August 21, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: Openmeetings 3.0 Error when uploading
using?
On 21 August 2014 20:22, Horace Miles wrote:
When uploading a mpeg4 I receive the following error:
key: processFLV 0
process: convertToFLV
command: null
exception: java.lang.NullPointerException
error: null
exitValue: java.lang.NullPointerException
out: null
Where can I go
mailing list we can ask?
On 21 August 2014 19:34, Horace Miles wrote:
Maxim,
Thanks but I didn’t know any other way to initialize the astDB database table.
If you know of a better way, we can do it that way. The only other alternative
I could see was to start from scratch, which I really
When uploading a mpeg4 I receive the following error:
key: processFLV 0
process: convertToFLV
command: null
exception: java.lang.NullPointerException
error: null
exitValue: java.lang.NullPointerException
out: null
Where can I go or do to fix this error?
20:39, Horace Miles wrote:
Maxim,
Thanks for all your help I have gotten everything to 3.0 VOIP and SIP
integration to work.
I believe if the documentation is changed as follows:
Add the following comment section above the room extensions
: Re: User Conferences
what do you mean? sort of public rooms but created by users?
On 20 August 2014 02:48, Horace Miles wrote:
Is there a way for users to create conference that are permenant?
--
WBR
Maxim aka solomax
Is there a way for users to create conference that are permenant?
Sorry Maxim,
I thought I had sent this to te users/dev lists
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
Sent: Monday, August 18, 2014 8:41 PM
To: Тимур Тлеукенов
Cc: Horace Miles; Openmeetings user-list
Subject: Fwd: Pointer on WB
@Timur, could you please answer this
Disregard this request
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
Sent: Monday, August 18, 2014 8:41 PM
To: Тимур Тлеукенов
Cc: Horace Miles; Openmeetings user-list
Subject: Fwd: Pointer on WB
@Timur, could you please answer this question
@Horace, please send your questions to
Maxim,
Thanks for all your help I have gotten everything to 3.0 VOIP and SIP
integration to work.
I believe if the documentation is changed as follows:
Add the following comment section above the room extensions entry:
; *
; The below
documentation
should include some information on this to pre-clude future novice users as
myself from making this same mistake.
Miles
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
Sent: Monday, August 11, 2014 8:20 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Thanks Maxim,
Why would my openmeetings database not have a rooms table but have one name
room?
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
Sent: Monday, August 11, 2014 7:50 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Subject: Re: Pointer on WB
sudo
Thanks Timur,
That is what the Asterisk documentations says that dbexist does. However, as I
am reading the code it is being used to query the openmeetings database and not
the AstDB. The code below is from the Openmeetings SOAP and VOIP Integration
wiki.
exten =>
_400X!,1,GotoIf($[${DB_EXI
ttp://openmeetings.apache.org/red5sip-integration_3.0.html
On 10 August 2014 22:09, Horace Miles wrote:
I took the firewall down. Still had the same problem.
I reverse the logic in this line of the extensions.conf
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?notav
eout
?
can you check it with all firewalls disabled?
On 8 August 2014 23:51, Horace Miles wrote:
Maxim,
Whenever you have time I understand. Here are all of my configurations by file
name. I hope it will help.
HERE IS THE CONFIGURATION FOR EACH FILE FOR THE ASTERISK INTEGRATION.
bridge from asterisk to red5 and performs audio/video
transcoding rtp <->rtmp
according to your issue it seems like creadentials specified in settings file
are invalid for your Asterisk, can it be a problem?
Will try to reproduce your problem as soon as i will get some time
On 7 Au
Re: Pointer on WB
Hello Horace,
sorry for keeping silence, a little bit bit busy right now
SIP transport set up the bridge from asterisk to red5 and performs audio/video
transcoding rtp <->rtmp
according to your issue it seems like creadentials specified in settings file
are in
sue it seems like creadentials specified in settings file
are invalid for your Asterisk, can it be a problem?
Will try to reproduce your problem as soon as i will get some time
On 7 August 2014 02:53, Horace Miles wrote:
Maxim,
Perhaps if I knew exactly what sip transport does, I might be a
on WB
Simple test if everything works is:
1) go to Admin->Conference rooms
2) select room
3) Check enable SIP
4) SIP number should appear in room panel (maybe after save)
is it works for you?
On 2 August 2014 00:36, Horace Miles wrote:
Ok found red5sip.enable value = yes
Aster
Simple test if everything works is:
1) go to Admin->Conference rooms
2) select room
3) Check enable SIP
4) SIP number should appear in room panel (maybe after save)
is it works for you?
On 2 August 2014 00:36, Horace Miles wrote:
Ok found red5sip.enable value = yes
Asterisk
27; in 32000 ms
(Method: INVITE)
Retransmitting #6 (no NAT) to 127.0.0.1:5070:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
127.0.1.1:5070;branch=z9hG4bK4721088;received=127.0.0.1;rport=5070
From: "red5sip_user" ;tag=z9hG4bK79605539
To: ;tag=as4a2fdbcb
Call-ID: 400642563986@127.0.1.1
CSeq: 2
I am still trying to integrate RED5SIP and VOIP into Openmeetings 3.0 The
connection is be Declined as not authorized but I can not figure out why. Here
are the relative log and debug files. Hopefully someone can help me figure
this out.
Thanks Miles
Asterisk messages log
Aug 5 06:0
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am
assuming Admin config is Openmee
be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT.My understanding was
that Openmeetings to be access from the internet needed to be on a public
address. That address would be the one in
...@gmail.com]
Sent: Sunday, July 27, 2014 12:07 AM
To: Horace Miles; Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally red5sip connects to red5 server directly, not to the swf client,
so contents of config.xml is ignored while connecting by red5sip
On 27 July 2014
[mailto:solomax...@gmail.com]
Sent: Friday, July 25, 2014 7:22 AM
To: Horace Miles
Subject: Re: VOIP and Sip Integration
hope you will be able to fix it, please let ne know if additional help is
required
On 25 July 2014 20:53, Horace Miles wrote:
Hey thanks for the files.
I compared
efer to write init.d script
I see no errors in your log
If everything is OK SIP transport should be in the room
All 3 logs should be checked to have no errors
I usually run asterisk in debug mode while setting everything up
On 20 July 2014 23:55, Horace Miles wrote:
Additonally
: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax...@gmail.com
where do I check?
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles wrote:
Ok I
18 July 2014 23:22, Horace Miles wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead
Error
o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles wrote:
Openmeetings log says confBridgeList authentication is failing. I will check
to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerExcept
AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room."
usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Mi
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles wrote:
Probably not, since I
:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax...@gmail.com
t; checked for the room you are
testing?
On 18 July 2014 21:48, Horace Miles wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have
completed all the steps according to the integration document.
I found the following document on the wiki:
https://cwiki.apache.org/
: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with
Asterisk. That being said, I can find no documentation that
It is nice that Openmeetings provided a way to integrate VOIP and Sip
with Asterisk. That being said, I can find no documentation that tells
the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i
Upgraded from 2.0 to Openmeetings 3.0.2 Rev 1598809 Build date 31 May
2014
I have the following configuration public IP
I can connect from the public IP on the local machine. However, when I
try to connect on the public IP from a remote host no connection is
made.
I have configure the
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