I can change the sip port to any number, and when I unload and reload chan_sip.so, I always get == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 64.1.16.172:5060 == Using TOS bits 4 == Parsing '/etc/asterisk/sip_notify.conf': Found == Registered application 'SIPDtmfMode'
Is there any other way to make the port change work? Also I never got an answer about how to prevent unregistered sip phones from sending inbound SIP calls. I can send calls regardless if my softphone is registered or not, when autocreatepeer=no. This is flaw that makes Asterisk very insecure. F.A. _______________________________________________ --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
