[asterisk-users] SIP CallerID Question

[Dave Fullerton]

Brent Davidson wrote:
> 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?

I don't see anything sticking out as being wrong. For kicks, what is the 
output of "sip show user Office1-user" on office2?