[asterisk-users] SIP CallerID Question

[Brent Davidson]

Dave Fullerton wrote:
> Check the entries for office1 and office2 servers in sip.conf. If they 
> have a callerid= entry comment it out and do a SIP reload. When it is 
> set asterisk overrides the caller ID sent to it.
>
> -Dave
There aren't any callerid= entries in any of my sip peer entries, and 
I'm not overriding the callerID anywhere in my dial plan.

Would the way I route the extensions make any difference?  Each office 
has it's own server and prefix by which it is accessed from another 
office.  So for office1 to dial extension 12 at office2 he would dial 1012.

In my Dialplan I have (AEL syntax):

  _10XX => {
    Dial(SIP/${EXTEN:2}@Office2,,Tt);
    Hangup;
  }

And in my SIP.conf on Office 1

[Office2]
username=Office1-user
fromuser=Office1-user
host=XXX.XXX.XXX.XXX (edited out)
type=peer
context=internal
secret= password
dtmfmode=rfc2833
disallow=all
allow=speex
call-limit=20
qualify=yes
canreinvite=no

In My Sip.Conf on Office2:

[Office1-user]
username=Office1
host=XXX.XXX.XXX.XXX (edited out)
type=user
context=internal
secret=password
dtmfmode=rfc2833
disallow=all
allow=speex
call-limit=20
canreinvite=no

Separating into peer and user entries was the only way I was able to get 
calls to go through and be authenticated properly.  Would this setup 
have any bearing on the caller ID?