[SR-Users] Kamailio LB: how to get Asterisk to do the RTP

With the help of members from this mailing list (many thanks!), I finally got Asterisk fronted by Kamailio for LB and REGISTERs and I am able to make a call using the setup that looks like this: [Kamailio 4.4.2]<->[Asterisk 13.7.2] Kamailio manages REGISTERs, but also forwarding them to Asterisk. I am able to make a call, but I get only one way audio or no audio depending on which client made the call (SipDroid->Zoiper I hear one way audio on Zoiper, but no audio if the call is made the other way). I noticed that Kamailio forced direct media between the endpoints in this situation, but my application really needs Asterisk to handle it. How do I do this? Should I start by forwarding INVITEs to Asterisk? How do I do that? Any help is appreciated. Thanks!