I am no longer getting the sip transport popping in and out of the room.
1. Should I see any instance of the sip transport in the room
2. When using the sip dialer upon clicking call nothing happens.
a. I check the openmeetings log and I can see the following:
1. getSipNumber: room_ID: 13, sipNumber: 40013
2. Asterisk –rx “originate
Local/6234123251@rooms-out extension 40013@rooms-originate”
Should I now be looking at the Asterisk logs to find out why the call is not
being placed? If so 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 <horace.mi...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
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 <horace.mi...@myit-solutions.com> 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 of "Red5sip log : Error
o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <horace.mi...@myit-solutions.com> 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:solomax...@gmail.com]
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 <horace.mi...@myit-solutions.com> wrote:
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 <horace.mi...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax...@gmail.com]
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 <horace.mi...@myit-solutions.com> 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:solomax...@gmail.com]
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 <horace.mi...@myit-solutions.com> 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
