What I think is the problem is the vpn. If you are using sip it is 99% the cause.
If you can try without it just as a test. Also try 1on1 call and different uac if possible.
Just to eliminate possibilities.
Regards,
Mirko
Hello,I am trying to establish a conference group call using the setup pictured at [1]. Thus, when a user calls to a group extension, all the users in that group are called.I have set the sip-ip and rtp-ip bind IPs. Signaling works fine, the phones in the group ring. When I answer I even hear the wave playback conference starter/joiner. I can see bindings on the RTP public IP:PORT (using netstat). I can ngrep packets coming in and out for that ports.The problem I have is that I can hear no media.I've been debugging and eliminated the possible networking firewall issues using netcat over UDP and some high ports, between public and private RTP ip, in both directions (firewall allows everything).Thus, I would say that the problem is on the application side. This kind of looks like a scenario where all conference users are muted, even if I don't have any rules like that in the dialplan group extension. The freeswitch ACLs default="allow" and I don't see any "blocked by acl" logs. The logs look fine: "... muxing call legs".Any ideas what might be the problem?Thanks in advance,Stefan
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