Re: [GnomeMeeting-devel-list] SIP, RTP ports and firewall

[Daniel Huhardeaux <devel tootai net>]

Damien Sandras a écrit :

> > Daniel Huhardeaux a écrit :
> >
> >    
> >
> > > Daniel Huhardeaux a écrit :
> > >
> > >      
> > >
> > > > Damien Sandras a écrit :
> > > >
> > > >        
> > > >
> > > > > [...]
> > > > >
> > > > >
> > > > > If the snapshot you are using is correct, the key name is :
> > > > > /apps/gnomemeeting-snapshot/protocols/ports/rtp_port_range
> > > > >
> > > > >
> > > > >          
> > > > Ok got it. Thanks
> > > >
> > > >        
> > > Outch:
> > >
> > > Please modify RTP port range to 16384:16388. Doing this, when I place
> > > a call it never stop to ring I have to kill GM :-(
> > >
> > > Some else see this?
> > >
> > >      
> > I think there is something wrong with this: I put 6970:6990 for RTP
> > port. I can now pass 2 calls and have no audio on the third. Does this
> > have a sense? FYI my asterisk uses 6991:7170
> >
> >    
>
>
> on the LAN?
>  
Yes

--
Daniel