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

[Tickling Contest]

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!
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