asterisk users - Jan 2008 - CallerID Number incorrect in SIP packet

I am having an issue with the CallerID Number not being passed to my
phone in the SIP packet.  The CallerID Name is passed just fine and
displayed on the phone with no issue.  I have done a NoOp() in my
extension.conf and successfully seen both the CallerID name and number
correctly. So that leads me to believe that Asterisk is handeling it
correctly.  However, when I do a packet capture of the SIP packet sent
from the Asterisk server to the phone, I do not see the CallerID Number
but instead see the registered user name of the phone:
 
 
 
The lutgrins-G-2433 is the user name that my phone is registered as.  I
would expect to see <sip:7893425141 at 10.31.4.18 instead of what I am
seeing.  Both the phone and the server are running on the same network
segment (no NAT involved).
 
Any help would be appreciated.
 
I am running Asterisk version 1.4.11
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in sip.conf under the definition for the sip user add
callerid=whatever

  ----- Original Message ----- 
  From: Lutgring, Sam 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, January 08, 2008 4:37 PM
  Subject: [asterisk-users] CallerID Number incorrect in SIP packet


  I am having an issue with the CallerID Number not being passed to my phone in
the SIP packet.  The CallerID Name is passed just fine and displayed on the
phone with no issue.  I have done a NoOp() in my extension.conf and successfully
seen both the CallerID name and number correctly. So that leads me to believe
that Asterisk is handeling it correctly.  However, when I do a packet capture of
the SIP packet sent from the Asterisk server to the phone, I do not see the
CallerID Number but instead see the registered user name of the phone:



  The lutgrins-G-2433 is the user name that my phone is registered as.  I would
expect to see <sip:7893425141 at 10.31.4.18 instead of what I am seeing. 
Both the phone and the server are running on the same network segment (no NAT
involved).

  Any help would be appreciated.

  I am running Asterisk version 1.4.11


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I have a server behind a firewall.  It is publicly addressed.  Should 
NOT be trying to NAT (how would I know).

The connection is a SIP trunk to Broadvoice.  I am calling the 
Broadvoice # from my cell and the call is being routed to my server.

With one firewall the INVITE contains information for the RTP session to 
be with broadvoice servers with different addresses.  It works.

With the other firewall the INVITE does NOT contain any other IP 
addresses, and the call goes through but no voice (duh).

I have captures of both.  I would include them in this message, but I am 
a little concerned that if some of you get the Registration, you will 
crack my secret...

PLEASE help me out on this.  I am absolutely pulling out my hair on it 
(and I don't have much).  I have stared at the Wireshark displays and 
just don't see it.  I have turned on logging on the failing firewall, 
and am not seeing any messages being dropped or rejected.