Asterisk Incoming Caller ID

On reflection, that probably should have been:
exten = _!,n,Set(CALLERID(all)="${CID}" <${CALLERID(num)}>)

I actually have tried that:
exten = _!,n,Set(CALLERID(all)="${CID}" <${CALLERID(num)}>)
I also tried the same line with the number in email format.
As for printing the number to screen.
I am saving all the caller id values to a database like so:
exten = _!,n,Set(CL=${ODBC_SPInsertCallerId(${CALLERID(name)},${CALLERID(num)},${CALLERID(ani)},${CALLERID(dnid)},${CALLERID(rdnis)},${CALLERID(pres)},${CALLERID(ton)})})
name, num, all, and ani are all initially set to the incoming caller ID number.
When I print ${CALLERID(num)} to screen, it has the correct caller id.
The odbc look up is an internal caller id database. That too is retrieving the name as expected and it does come up on the phone. The phones are linksys spa942. If I do not set the caller ID, I get the caller ID number with the called extension underneath. If I set the name, I get the name with the called extension underneath. I have tried using the "o" option when transfering to the correct extension:
exten => s,n,Dial(SIP/${ARG1},20,o). This made no difference.

When I make calls to an SPA941 from Asterisk, the LCD display always shows this:
From: <name>
<number>

Where <name> is the caller ID name and <number> is the caller ID number. The only difference I can think of is that I have never used the SetCallerPres command in the dial plan or the usecallingpres parameter in the peer definition. So perhaps you need to remove those lines from your peer definition and dial plan macro.

Other possibilities include: An obscure setting on the SPA942, but I tried several that I thought might make a difference and none did; Another line in the Asterisk dial plan that is not contained in the macro, but I assume you would know if that was the case; A setting in sip.conf or users.conf that is over-riding the caller ID, but I think all the settings available are there to set the caller ID for calls made from the peer, not for calls made to it; an additional SIP header (specifically the P-Asserted-Identity header) has got in there somehow and is being used in preference to the From header.

Can you capture the entire SIP INVITE packet sent to the Linksys phone at the start of the call? I normally use the Asterisk CLI command "sip set debug" (or "sip set debug on" if using v1.6) to capture SIP packets.