If you are using SIP then you can do this by making sure that canreinvite is set to yes in sip.conf for those 2 sip clients and: 1. No transcoding is taking place (you are using the same codec on both ends). 2. You don't have anything in the dial command that forces asterisk to keep the stream (like t or T, etc.). 3. And that the 2 clients can reach each other nicely (if they are both behind different NATs then there might be a problem, even if when asterisk has no problem communictating with them).
On 11/5/05, Arik Funke <[EMAIL PROTECTED]> wrote: > Hello, > > can somebody tell me if following is possible and if yes, how? > > Assume we have a Asterisk server to which two VoIP clients are connected > over the internet (i.e. not internally). Now I would like to avoid > having the connection run over the server but would like the server to > tell the clients that they should contacts each other directly. Thus I > would obviously avoid traffic, load on the server and reduce delays. > > Now how do I go about this? Thanks in advance for the help. > > Cheers, > Arik > > > PS: The clients have dynamic IP adresses and are possibly behind nat > server... > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
