Hi. Il dom, 2004-05-23 alle 01:52, Tony Hoyle ha scritto: > Surely it depends on who's calling me - if they're using a SIP phone it'll > come in over the SIP port, and if they're using an IAX phone it'll come in > over the IAX port - ie there's this context in the default iax.conf: > > [guest] > type=user > context=default > callerid="Guest IAX User"
for letting unauthorized user to call you over IAX(2). Like a pstn call... everyone can call you if the have your number (or IP in Voip calls) If you don't want that, just delete that entry :) > btw. how many rtp streams do I need? I only have 1 phone at the moment (max. > will be about 4 I think). mmh... I dunno the values of that association, but bear in mind that: * are only UDP ports * are opened only during a RTP session, in a dynamic way so leaving open ports 10000 to 20000 UDP as in default rtp.conf isn't a problem, since there's not any port open... (unless you run any udp service on that interval :) ) and a portscan will detect these port as closed. only during a call, * and the phone will handshake an RTP port and use that. otherwise will be closed. Matteo. -- Brancaleoni Matteo <[EMAIL PROTECTED]> Espia - Emmegi Srl _______________________________________________ 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