On Tue, 2008-12-16 at 22:20 -0200, Alberto Furtado wrote: > In my configuration the Asterisk Box is a user, it sends calls to the > Sipxpbx. > Sipxpbx is doing the Trunk Termination, not the Origination. > > I have had excellent results with Sipxpbx in our simulations and > I am beginning a test run in a practical environment. My tests show > that Sipxpbx can handle the large load that will be generated > by our Termination Services, where Asterisk Servers fail to handle > the throughput. But Sipxpbx has to be able to accept an Asterisk > box as a user, the same way it accepts the FXS and E1 SIP > gateways that we tested.
You don't need to register to originate calls. If your dial plans don't require an permissions (which I would do only on a closed network), then sipXecs won't challenge for authentication. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
