On Jan 16, 2009, at 10:38 AM, Gabriel Ortiz Lour wrote: > Hi all, > > Suposing that 2 SIP phone register at a remote (internet) > asterisk, what is the best way, if any, to make the RTP traffic go > phone to phone, whithout using the internet conection (asterisk)?
Allow reinvite? Assuming both are not behind NAT. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
