[Users] Question on SIP authentication with users from OpenSER

Hi, This is more an Asterisk question, I know, but I am sure a good many of you have similar setups.... We are using Asterisk 1.2.3 with RealTime for PSTN and Voicemail where users register with an OpenSER cluster (2 nodes currently). When they request PSTN they are forwarded to * where they have entries in SIP realtime database. This ensures that they get their correct CallerID and context, etc. This is working fine at present, where I have the SIP users set up with the following relevant SIP entries: name username callerid "User" <XXXXXXX> canreinvite no context context dtmfmode RFC2833 host 123.123.123.123 insecure port type friend username username Note that I have set the host to the IP of the OpenSER server, and there is no secret. I have the OpenSER servers set up as peers also. My questions are: 1. Is this the best way to to set this up? 2. I have many users, and I need to be certain that a) the username exists and b) that the request came from one of our OpenSER servers. Will the above ensure that both the username AND the host are correct? I have seen instances where if I have a static SIP entry with the same host= line, a non-existent user will be accepted as this static user. 3. How can have more than one possible host= setting for a user (i.e. they could come in from either of our OpenSER servers. Thanks! -- -Barry Flanagan