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

["Damien Sandras" <dsandras seconix com>]

> 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?