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 <d.tr...@pocos.nl> wrote: > On Wed, Jul 27, 2016 at 01:54:07AM -0400, SamyGo wrote: > > 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'd suggest to use rtpengine for all calls, it fixes most problems and > uses nearly no resources (with the kernel plugin) > > > _______________________________________________ > 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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