asterisk users - Mar 2005 - Why even have set CallerID option?

Why even have the ability to set callerid name/number if end offices don't
honor it?

For example, I have a SIP UA registered and in the sip.conf I have:

    callerid="Mark Mane <2815692712>"

When that phone makes an outbound local call, asterisk will terminate it on
PRI connected to asterisk box to Time Warner.

When the called party looks at their caller id display screen it shows the
number that is in sip.conf, but does not show the name I have set in the
sip.conf; instead it shows our company name (since we "own" the
number).

If it is the responsibility of the last end office to do a data-dip and
select out the name, then that means I cannot control the callerid name,
correct?

So I guess that callerid name is only useful for VoIP<->VoIP calls that go
thru asterisk?

-Matthew

Matthew Boehm wrote:
>Why even have the ability to set callerid name/number if end offices
don't
>honor it?
>
>For example, I have a SIP UA registered and in the sip.conf I have:
>
>    callerid="Mark Mane <2815692712>"
>
>When that phone makes an outbound local call, asterisk will terminate it on
>PRI connected to asterisk box to Time Warner.
>
>When the called party looks at their caller id display screen it shows the
>number that is in sip.conf, but does not show the name I have set in the
>sip.conf; instead it shows our company name (since we "own" the
number).
>
>If it is the responsibility of the last end office to do a data-dip and
>select out the name, then that means I cannot control the callerid name,
>correct?
>  
>
Close enough, yeah.
>So I guess that callerid name is only useful for VoIP<->VoIP calls
that go
>thru asterisk?
>
>-Matthew
>  
>
Yup.  Which is actually very helpful for me.  My offices are going to 
have about 50 phones, and the callerid on the phones will be extremely 
helpful for sip-sip calls. 

Sean

mboehm@cytelcom.com (Matthew Boehm) writes:
> Why even have the ability to set callerid name/number if end offices
don't
> honor it?
VOIP is bigger than just PSTN-gatewayed calls via some specific
company.  The end goal is to connect the VOIP islands directly.  That
is already happening at some large companies where they call their
supplier directly on a purely voip link.  For a concrete example look
at the sip-edu program. It is a growing group of universities that
exchange SIP calls directly.  (Some even have their asterisk and SER
config notes on line.)  In all cases the caller's calling-number and
calling-name stuff will get passed to the callee.

     http://voip.internet2.edu/SIP.edu/

-wolfgang