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

[Daniel Huhardeaux <devel tootai net>]

Damien Sandras a écrit :

> 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).
>  
Upgrade done: if RTP ports stays to 6970:6990, same behaviour. Two 
calls, that's all.

If I modify and put 16384:16388 I now get a Error Popup window: Erreur 
Générique - Function pthread_mutex_destroy failed. Push validated, GM 
hangs up properly with an "abnormal end of call"