[Freeswitch-users] default muted media on conference ?!

Mirko Brankovic mirkobrankovic at gmail.com
Sat Jul 2 12:31:32 MSD 2016

What I think is the problem is the vpn. If you are using sip it is 99% the

If you can try without it just as a test. Also try 1on1 call and different
uac if possible.
Just to eliminate possibilities.

On Jul 2, 2016 9:30 AM, "Anonim Stefan" <fanx07 at gmail.com> wrote:

> 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
> [1] https://postimg.org/image/4cwxn91bb/
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