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.contest@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!

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