Hi Tickling, Could you please share with us your working config?
With kind regards, Jurijs On Wed, Jul 27, 2016 at 8:54 AM, SamyGo <govoi...@gmail.com> wrote: > 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 >> >> > > _______________________________________________ > 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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