Thanks Maxim,
Let me make sure I understand about the sip transport. It should not be
popping in and out of the room? On my box I keep getting Sip Transport has
exited the room.
When properly configured, should I be able to call land and cell phones without
a need for another server or;
1. Do I need to subscribe to a VOIP service provider
2. Configure Asterisk as a sip trunk to use google voice or some other
solution?
Also if you could have someone correct this line in the instructions of
extensions.conf it will help to eliminate at least one error
[rooms-red5sip]
exten =>
_400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil)
<<<<<<<<<<<< should be “notavail”
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
Thanks ahead of time
From: Maxim Solodovnik [mailto:[email protected]]
Sent: Monday, July 21, 2014 4:00 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
jsvc can be used to start java application as service
I don't really like it (it was unstable when I used it)
I prefer 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 <[email protected]> wrote:
Additonally the VOIP and SIP integration 3.0 instructions do not mention
installing jsvc. Is it still a requirement to install jsvc under 3.0 as it was
under 2.0?:
apt-get install jsvc
From: Maxim Solodovnik [mailto:[email protected]]
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 <[email protected]> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:[email protected]]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <[email protected]> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:[email protected]]
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 of "Red5sip log : Error
o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <[email protected]> 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.NullPointerException: Null
From: Maxim Solodovnik [mailto:[email protected]]
Sent: Friday, July 18, 2014 8:43 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 Miles <[email protected]> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:[email protected]]
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 <[email protected]> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:[email protected]]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are
testing?
On 18 July 2014 21:48, Horace Miles <[email protected]> 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/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions
menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No
Service provider / method not found; to handle calls like onBWCheck, add a
service provider. (it is my understanding this error is to be expect as it is
not being used?)
So where would I start to try and figure out why there is no sip dialer
available?
From: Maxim Solodovnik [mailto:[email protected]]
Sent: Friday, July 18, 2014 7:15 AM
To: 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 <[email protected]> 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 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.e. how to
make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
