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

[Daniel Huhardeaux <devel tootai net>]

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

--
Daniel