I apologize if this is not the proper forum for this question. I've set up openmeetings 3 in a ubuntu guest on top of a linux host. I've also got an asterisk server on the vm along with red5sip.
*Everything* works except for one thing: when dialed in to a room through asterisk, no audio reaches the telephone handset through red5sip. Audio *from* the phone makes it into the conference on both sides (asterisk conf bridge and openmeetings flash audio). And: if more than one phone dials in via sip to the asterisk bridge, those phones have bi-directional audio between each other. The only problem is that no audio seems to go towards the asterisk bridge from the red5sip daemon. I've looked through logs and haven't found anything obvious - same with sniffing. I could really use a pointer on this one.
