[Asterisk-Users] Passing called number in SIP

[snacktime]

On 6/27/05, Andres <andres at telesip.net> wrote:
> 
> >
> >However,  it's not really passing the called number per say.  What
> >it's doing is putting the extension I have in my register statement
> >into the "To" field.  I'm assuming the "To" field is actually being
> >populated with whatever * set the "Contact" field to when it
> >registered.    This seems to mean that I need a unique username for
> >every SIP DID I have if I want to be able to route them to different
> >context's.
> >
> >Is there a standard way of handling this issue when you have multiple
> >SIP DID's ?
> >
> >Chris
> >
> >
> >
> 
> Say you have a block of 100 DIDs with Level 3 for example.  You can just
> configure something like this in your incoming context:
> 
> exten => _30355597[0-9][0-9],1,Dial(SIP/${EXTEN},30)
> 

I don't get it:)  If I have 100 DID's but only one register statement,
isn't the called number for all 100 going to be the one extension name
I registered as?  Or am I missing something?  At least with the
provider I am testing with the called number is always the extension
in my register statement, regardless of what the DID really is.

For example I have 4 DID's with this one provider.  With the following
register statement they will all come in with the sip user/called
number as 1112223333:

register => user:pass at provider.com/1112223333

With this register statement they all come in to sip user/called number "s":

register => user:pass at provider.com


What happens if I put a register statement for every DID?  Wouldn't that work?

Chris