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
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