This is how it works by default or you couldn't get a call from a remote SIP users. The call will drop into the 's' extension (assuming 1.x here - haven't looked at the changes in 1.2 yet) in whatever context you have configured in sip.conf (default?). The authentication details are important when you want to give more priviledge to a user (outbound PSTN for example) which you really should have in 'default' context as external callers could cost you a lot of money!
Matt. On 24/11/05, Mauro Zanin <[EMAIL PROTECTED]> wrote: > Hi there, > is there any way to call Asterisk from a SIP phone, where you don't know > name and password of the caller? > I want to allow customers of a company to place a call over the Internet > without being registered on the Asterisk. This could be a very large number > of SIP clients, only a few will phone at the same time, but I don't want to > create an entry in SIP.conf for each probable caller! _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users