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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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.

I added the #!define WITH_NAT option, and now the call can only be made one way. RTPProxy was started like so: $ rtpproxy -l 192.168.1.101 -s udp:localhost:7722 -u kamailio root@kamailioA:~# netstat -pln | egrep "kamailio|rtpproxy" tcp 0 0 192.168.1.101:5060 0.0.0.0:* LISTEN 10112/kamailio tcp 0 0 127.0.0.1:5060 0.0.0.0:* LISTEN 10112/kamailio udp 0 0 192.168.1.101:5060 0.0.0.0:* 10081/kamailio udp 0 0 127.0.0.1:5060 0.0.0.0:* 10081/kamailio udp 0 0 127.0.0.1:7722 0.0.0.0:* 10042/rtpproxy raw 0 0 0.0.0.0:255 0.0.0.0:* 7 10081/kamailio unix 2 [ ACC ] STREAM LISTENING 33357 10102/kamailio /var/run/kamailio//kamailio_ctl My full config is at https://gist.github.com/ticklingcontest/e315972c80c82f6dfa23920c7725d60b BTW, my entire setup, kamailio, asterisk and the phones etc. are in one private network. I think setting realtime endpoint with "direct_media=no" is pointless as all of these interactions are fronted by Kamailio. What's going on here? Any help is appreciated. On Wed, Jul 27, 2016 at 10:15 AM, Daniel Tryba &lt;d.tryba(a)pocos.nl&gt; wrote: _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users