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

[Damien Sandras <dsandras seconix com>]

Le dimanche 20 novembre 2005 à 22:51 +0100, Daniel Huhardeaux a écrit :
> 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
> 

Then this doesn't make sense =)

Can you debug further? (Please upgrade first).
-- 
 _      Damien Sandras
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