asterisk users - Jun 2005 - Passing called number in SIP

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

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).
>Chris
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-- 
Andres
Network Admin
http://www.telesip.net