Hello Sebastian, I think I'm very close to get this work, I have a problem with the SIP Transport, In the log events I see this:
Caused by: javax.persistence.NoResultException: Query "SELECT m.members FROM MeetMe AS m WHERE m.confno=:confno" selected no result, but expected unique result. I don´t know where can be the error, the log in the asterisk CLI is: [Aug 9 05:18:48] NOTICE[11229]: chan_sip.c:22825 handle_request_invite: Call from 'test2' (127.0.0.1:5074) to extension ' ' rejected because extension not found in context 'rooms'. I think the script is not getting the ID of the conference room or something like that. Thanks a lot. Leonardo Peña A. CCNA Security - CCIP Cyma Ingeniería Ltda Tels: 5402830 - 3473268 Ext. 103 Cels: 317 516 67 63 – 311 829 20 81 BB Pin: 21383E42 Dir: Cra 14A No 71A - 59 Of. 602 Salvemos el planeta. NO Imprima este mensaje si no es necesario. -----Mensaje original----- De: Leonardo Peña Aristizabal [mailto:[email protected]] Enviado el: sábado, 04 de agosto de 2012 01:53 p.m. Para: '[email protected]' Asunto: RE: VoIP Integration With OpenMeetings Ok, I put just false and works Sebastian I think this is strange when I try to connect to a conference room, there is a user already in there that is call "SIP Transport" but when I get in to that user just disappears, I think that why it's not working? Thanks. Leonardo Peña A. CCNA Security - CCIP Cyma Ingeniería Ltda Tels: 5402830 - 3473268 Ext. 103 Cels: 317 516 67 63 – 311 829 20 81 BB Pin: 21383E42 Dir: Cra 14A No 71A - 59 Of. 602 P Salvemos el planeta. NO Imprima este mensaje si no es necesario. -----Mensaje original----- De: [email protected] [mailto:[email protected]] Enviado el: sábado, 04 de agosto de 2012 01:42 p.m. Para: [email protected] Asunto: Re: VoIP Integration With OpenMeetings sip.enable should be 0 or No or false. Whatever value but NOT yes red5sip.enable should be "yes" Sebastian 2012/8/4 Leonardo Peña Aristizabal <[email protected]> > Sorry, for this question: > > 1: enable > 0:disable > > or, > > Yes:enable > No:disable > > And you mean the sip.enable parameters? Or red5sip.enable? > > Thanks > > Leonardo Peña A. > CCNA Security - CCIP > Cyma Ingeniería Ltda > Tels: 5402830 - 3473268 Ext. 103 > Cels: 317 516 67 63 – 311 829 20 81 > BB Pin: 21383E42 > Dir: Cra 14A No 71A - 59 Of. 602 > > P Salvemos el planeta. NO Imprima este mensaje si no es necesario. > > > -----Mensaje original----- > De: [email protected] [mailto:[email protected]] Enviado el: > sábado, 04 de agosto de 2012 01:20 p.m. > Para: [email protected] > Asunto: Re: VoIP Integration With OpenMeetings > > Hi Leonardo, > > no I did not say anothing about "work without red5SIP". > I just said "SIP Applet is completely wrong" > > There are 2 integration of SIP /VoIP. > You should use red5SIP. > So first thing todo would be to disable the red5.enable configuration > value in Administration > Configuration. > > I don't know what number you've configured for dialing into conference > rooms. I guess this can be configured in Asterisk. > > Sebastian > > 2012/8/4 Leonardo Peña Aristizabal <[email protected]> > > > Uhmm Ok, so you mean that if i just configure the asterisk, create > > the extensions and the MeetMe can work without Red5Sip, in that way > > how can I connect a sip phone with the conference rooms?**** > > > > ** ** > > > > Until the correct registration of a sip phone in the asterisk I’m > > going well I just getting stuck in the integration to openmeetings I > > mean I supposed that I have to dial 400(X) and just get connected to > > the room?*** > > * > > > > ** ** > > > > Thanks.**** > > > > ** ** > > > > Leonardo Peña A.**** > > > > CCNA Security - CCIP**** > > > > Cyma Ingeniería Ltda**** > > > > Tels: 5402830 - 3473268 Ext. 103 **** > > > > Cels: 317 516 67 63 – 311 829 20 81**** > > > > BB Pin: 21383E42 > > Dir: Cra 14A No 71A - 59 Of. 602**** > > > > [image: Descripción: logo cymaingPequeño]**** > > > > *P** **Salvemos el planeta. NO Imprima este mensaje si no es > > necesario.*** > > ** > > > > ** ** > > > > *De:* [email protected] [mailto:[email protected]] *Enviado > > el:* sábado, 04 de agosto de 2012 12:56 p.m. > > *Para:* Leonardo Peña Aristizabal > > *Asunto:* Re: VoIP Integration With OpenMeetings**** > > > > ** ** > > > > Hi Leonardo, > > > > you've mixed up the integration docs. > > The SIP Applet should not be enabled at all for the red5SIP integration. > > > > Please send further queries to the official mailing lists or to the > > commercial support. > > > > Sebastian**** > > > > 2012/8/4 Leonardo Peña Aristizabal > > <[email protected]>**** > > > > Hello Sebastian**** > > > > **** > > > > I’m working in the VoIP Integration with Openmeetings but I’m having > > some troubles , what I got is this:**** > > > > **** > > > > Debian 6 x64**** > > > > Openmeetings: 2.0.0.r1361497-14-07-2012_1108**** > > > > Asterisk in the same server: Asterisk 1.6.2.9-2+squeeze6**** > > > > **** > > > > I follow the exact instructions son this link: > > https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integrat > > io > > n.html > > **** > > > > **** > > > > And everything ok, but when I enter to a conference room I get the > > message SIP Applet is not ready!, and in the users appear SIP > > Transport but when I get in with my camera and the other settings > > that user disappears and nothing happened, I try to connect to the > > conference room with a Sip Phone just dialing 400(2,3,4,5) but the > > same just appear connect in the sip phone but in the conference room > > nothing happened.**** > > > > **** > > > > So I don´t if something is wrong if you need more information just > > tell > me. > > **** > > > > **** > > > > Thanks a lot.**** > > > > **** > > > > Leonardo Peña A.**** > > > > CCNA Security - CCIP**** > > > > Cyma Ingeniería Ltda**** > > > > Tels: 5402830 - 3473268 Ext. 103 **** > > > > Cels: 317 516 67 63 – 311 829 20 81**** > > > > BB Pin: 21383E42 > > Dir: Cra 14A No 71A - 59 Of. 602**** > > > > **** > > > > *P** **Salvemos el planeta. NO Imprima este mensaje si no es > > necesario.*** > > ** > > > > **** > > > > > > > > > > -- > > Sebastian Wagner > > https://twitter.com/#!/dead_lock > > http://www.webbase-design.de > > http://www.wagner-sebastian.com > > [email protected]**** > > > > > > -- > Sebastian Wagner > https://twitter.com/#!/dead_lock > http://www.webbase-design.de > http://www.wagner-sebastian.com > [email protected] > > -- Sebastian Wagner https://twitter.com/#!/dead_lock http://www.webbase-design.de http://www.wagner-sebastian.com [email protected]
