I'll try to find server with configured Asterisk and try to double-check
On 22 July 2014 20:37, Horace Miles <horace.mi...@myit-solutions.com> wrote:
> Thanks I made the change prior to sending the email. There appears to be
> something else missing:
>
> There appears to be a entry missing in the /etc/asterisk/func_odbc.conf:
> file for ${EXTEN}
>
> I am probably wrong. But I can’t figure out how this is making the call
> to the database.
>
> I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file
> and I am not sure how to construct one there that would work.
>
> Would I simply add
>
> [EXTEN]
> dsn=asterisk
> readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO
> GET THE ROOMID INTO THIS VARIABLE
>
>
>
> Thanks ahead of time
>
>
>
> Miles
>
> *From:* Maxim Solodovnik [mailto:solomax...@gmail.com]
> *Sent:* Tuesday, July 22, 2014 6:23 AM
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> yes, this line need to be corrected
>
> openmeetings/rooms -> openmeetings/room
>
>
>
> guess this is the problem
>
>
>
> On 22 July 2014 19:43, Horace Miles <horace.mi...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> I have been trying to figure this out, I am knew to it all but on a steep
> learning curve.
>
>
>
> I do have a question about the asterisk extensions.conf
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>
>
>
> Does the above line check the openmeetings database rooms table for the
> confno and returns ok if it finds it and notavail if it doesn’t?
>
>
>
> I am getting the following warning:
>
> Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (
> 98.0.0.0:5070) to extension ‘40016’ rejected because extension not found
> in context “rooms-red5sip” I don’t recall seeing this error before. But
> if the “exten” line is checking the database openmeetings and looking for
> rooms table it does not exist. There is a table name room but no rooms.
>
>
>
> Am I reading this correctly?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax...@gmail.com]
> *Sent:* Tuesday, July 22, 2014 4:27 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> AFAIK dial in/out conference room was working as expected
>
> Not sure if we still have infrastructure test current version
>
>
>
> Will try to ask someone
>
>
>
> On 21 July 2014 20:17, Horace Miles <horace.mi...@myit-solutions.com>
> wrote:
>
> Ok on the sip transport, I will try to figure out why it keep popping in
> and out.
>
> However, I am not understanding concerning the Asterisk config. The
> asterisk config I am using is from the install and modification as stated
> from the VOIP/SIP 3.0 integration package. The instructions don’t say a
> sip trunk or outside provider is required. However, I am unable to
> succesfull make calls from a conference room to a phone and visa versa.
> Which I thought I was suppose to be able to do after the integration. Per
> the below instructions I guess I am asking what am I missing from below?
>
> *Feature Matrix:*
>
> *Feature*
>
> *Description*
>
> 1) Dial-In
>
> A phone number is provided which you can give to anybody to "Dial-In" via
> usual landlane/phone into the conference room of OpenMeetings - Every room
> has its own phone number. Currently room gets number
> like 400<Id of room>. Maybe should move phone prefix to settings,
> currently it hardcoded.
>
> 2) Dial-Out
>
> The users in the conference room can call anybody outside of the
> conference room by entering the phone number in the conference room - In
> room actions menu exist "SIP dialer". When user clicked dialer window
> appears.
> Currently calls can't be dropped from Openmeetings, tbd
>
> 3) Multiple Dial-In
>
> You can give away multiple numbers and do the same as described in case
> (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk).
> It is possible to create multiple extensions (phone numbers) in Asterisk
> configuration that will be redirects to single conference room.
>
> 4) Multiple Dial-Out
>
> You can dial multiple numbers from within the conference room - From
> within conference can be dialed multiple numbers.
>
> *Main difference to native Red5-Phone project*
>
>
>
>
>
>
>
> *rom:* Maxim Solodovnik [mailto:solomax...@gmail.com]
> *Sent:* Monday, July 21, 2014 5:36 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> 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
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
