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

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 :

Ok got it. Thanks


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?


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