[Asterisk-Users] realtime sip confusion

[Steve Blair]

snacktime wrote:

>In case someone else made the same mistake I did, and because I can't
>find this information posted anywhere, here is what I found out  about
>realtime sip.
>
>You can use it to register UA's that are registering to asterisk, and
>you can use it for peer context's for outgoing calls, but you cannot
>use it for incoming calls from gateways you have registered with.  I
>would have thought that when a call came in it would query either for
>the hostname of the gateway you registered with, or maybe the
>extension you registered as, but instead it looks up the username of
>the caller, which for incoming calls will usually be the caller id.
>
>It makes sense when you stop and think about it, but it's not exactly
>intuitive at first.
>
>  
>
Thanks for the note but why do you say it makes sense? If the username 
of the caller is
used to identify a peer that seems really bad. If used this way then I'd 
have to define every
number that is likely to call into my Asterisk box. Could you explain?

Thanks

>Chris
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at 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
>  
>

-- 
  
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104  


voice: 215-573-8396 

       215-746-8001

fax: 215-898-9348    

sip:blairs at upenn.edu