Hello Openmeetings Users,
I still consider myself to be a noob at this. Trying to rap my head around all the different technology. I am running Openmeetings 3.0.2. release on Ubuntu 12.04 LTS Core 2 Quad CPU Q6700 @ 2.66GHz x4 32 bit OS I have successfully integrated SIP using the SIP 3.0 integrate instructions. I have successfully integrated Asterisk 11.20.0 I can make call out from OM and I can receive calls in to a room. I am trying to figure out how can I separate the incoming calls to show up on their separate channel within a room showing their callerID. I am finding very little documentation on red5sip transport. Since this is using confbridge from asterisk is there some setting in the conferencebridge that could bring the calls in on a separate transport. The idea is to be able to identify people that have called in and to be able to individually control their mic, i.e. muting and unmuting them on an individual basis. This would make an already really great webapp really really great. I am still trying to learn asterisk but from what I have read the conference bridge should just make the users join the conference/room. I am going to load a separate asterisk server with confbridge and to see if the users in a conference call can be controlled on a individual basis. But I am really thinking that has to do with the SIP transport. Not being able to control individual callers in a conference room is a really big problem. Right now it is all or nothing. Can anyone share any light on this matter. Cross domain peering servers anyone know of any good one that may be free or really really cheap that they recommend using? --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus
