If I do remember correctly
SIP transport should enter the room and be in room as long as there other
users in it.
Possibility to call to phone numbers depends on your Asterisk config.
I'll try to fix documentation ASAP
On 21 July 2014 18:55, Horace Miles <horace.mi...@myit-solutions.com> wrote:
> 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:solomax...@gmail.com]
> *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 <horace.mi...@myit-solutions.com>
> 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: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
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
