sorry, not "red5.enable" "sip.enable" should be "false", this is how you get rid of the SIP Applet.
Then you can concentrate on getting red5SIP integration running. Sebastian 2012/8/4 [email protected] <[email protected]> > 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-integration.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]
