[Asterisk-Users] realtime sip confusion

[snacktime]

On 6/27/05, Steve Blair <blairs at isc.upenn.edu> wrote:
> 
> 
> 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?

It makes sense because it mirrors how sip.conf works, as opposed to
doing something different.   To me it looks like a limitation of SIP,
whereas IAX was designed to work in a PBX environment.

I'm not clear on a lot of this, but with the way SIP works I can't see
an easy way to get the callerid and the called number without using
some custom and/or little used headers.  I would be very interested in
hearing about how this is customarily done.  Obviously the providers
are getting that information from their upstream proxies, otherwise
they wouldn't be able to route the calls.  Why that information isn't
passed downstream I don't know.  Maybe it requires customization of *
beyond what a provider wants to support?

Chris