Setting up PLAR with Cisco Unity Connection Call Handler

Question

  This is a lab setup and Im doing it to learn.  No customer involvement.
Setup
Analog phones - FXS Port - 2951 Router - FXS port - FXO Port - 2951 Router - CUCM 9 - UC 9
                        |------------PSTN Emulator-------------|    |-----MGCP GW----------|
I have a CTI Route Point configured as DN 7000 and it has the default VM profile.  The CTI RP is set to FWD All to VM. 
The FXO port is set to PLAR to 7000.
When I dial from the PSTN analog phones through the FXO port, I hear the first ring, the FXO port answers, then I hear what sounds like the recorded message beginning to play.  Immediately after, I hear the recorded message "You cannot be transferred to this number.  Check the number and try again."
I dont do anything and within a second or two, I hear the recorded message for the system call handler start.
I did some more testing.  I added an E1 trunk between the PSTN and the MGCP gateway.  In the CUCM, I created another CTI Route Point with a DN that I could dial from my PSTN cloud.  I also set that CTI Route point up to Call FWD all to VM.
When I dial using the E1 trunk, the call hits the system call handler as expected and I hear my recorded greeting (as expected).
However, calling through the FXS-FXO tie line consistently gets me the error message recording followed by my recorded greeting.
Im currently using loopstart on the tie line, though I have also tried ground start with no difference.
Any ideas?
Jeff              

Answer 1

If I understand correctly, you want outside calls to go directly to the call handler but internal calls to ring whatever phone this extension is on.
If I am understanding correctly, then this will probably resolve it.  For the purposes of the explanation I will assume that extension 1000 is the number in question:
Create a new partition, we'll just call it ToVM or something like that
Create a new CTI route point with extension 1000 and put it in the ToVM partition, forward all calls for this CTI route point to voicemail
If you don't have one already, create a calling search space for the voice gateway.  Call it Gateway-CSS.  This should have the same partitions that the gateway can normally call, but it should also have the ToVM partition and that partition should be HIGHER in the list than the partition that has the normal extension 1000 on it.
Apply Gateway-CSS to the gateway
Configure the normal extension 1000 (not in the ToVM partition) the way that you want it to work.
Now when external callers dial 1000 they will go to the call handler because that partition is higher in the CSS and the CTI Route point should be hit first.  Internal callers will ring the phone (or whatever it is) because they only have access to the regular internal partition (and not ToVM) that 1000 is in.
This is all assuming that I understand you correctly!