[Asterisk-Users] Passing called number in SIP

[snacktime]

On 6/27/05, Andres <andres at telesip.net> wrote:
> 
> 
> snacktime wrote:
> 
> >I thought maybe one of the providers here could answer this question.
> >When using IAX to make calls, it passes the called number.  Being
> >designed to work with a PBX this makes sense.   SIP works differently
> >though and I'm curious why providers don't have a way to pass the
> >called number on their DID's, or choose not to do so.  Providers that
> >themselves use upstream SIP proxies are obviously getting both the
> >callerid and the called number.  However all incoming calls I get to
> >my DID's have the caller id as the SIP user being called.
> >
> >I'm guessing that it takes using some additional SIP headers to get
> >both called number and callerid, and that most providers probably
> >don't want to have to support that for their clients.
> >
> >That's my very rough guess.  Can anyone shed some light on the real reason?
> >
> >
> Chris,
> 
> The SIP Message has both a "From" Header (caller id), and a "To" Header
> (called number).  If you are not getting those properly then either you
> have broken Asterisk implementation (like the ones between 1.0.4 and
> 1.0.7), or your SIP provider is messing things up.  To figure out who is
> at fault then use Etheral to capture the SIP INVITE directly before it
> hits your Asterisk and see if is is missing the "To" Header.  (I really
> doubt it is).

Strange now it's working with the called number being passed in the
"To" field.  I must have just not seen it before.

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