Hi Again, You need to enable NAT handling in your Kamailio (#!define WITH_NAT), then depending upon how your clients will interact with asterisk you may or may not need a media proxy, like RTPproxy. If asterisks can send/receive media directly from the internet then its ok for now, else you definitely need to have rtpproxy/rtpengine in there.
Regards, Sammy On Tue, Jul 26, 2016 at 10:29 PM, Tickling Contest < tickling.cont...@gmail.com> wrote: > 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! > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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