[Asterisk-Users] RTP Packets going over caller and calle !!

2004-08-02 Thread Carlos Arnt
Hi, I Have a problem here, if anyone know a method to avoid please tell me . Using * with the option canreinvite=yes i can in theory tell to my * box, send RTP Packet directly from one Sip device to another one, then "In Theory", i will not use my own internet connection. So this mean that

Re: [Asterisk-Users] RTP Packets going over caller and calle !!

2004-08-02 Thread Greg Hill
On Mon, 2 Aug 2004, Carlos Arnt wrote: I Have a problem here, if anyone know a method to avoid please tell me. Using * with the option canreinvite=yes i can in theory tell to my * box, send RTP Packet directly from one Sip device to another one, then In Theory, i will not use my own internet