[c-nsp] ISDN to VoIP dial-peer Question

[Dan Armstrong]

I'm trying to setup a seemingly simple application with an AS-5400XM as 
a PSTN gateway for a hosted VoIP service.  Sip proxy & users on one 
side, PRI on the other side.  I setup 2 dialpeers, one for each.  I just 
want every call coming off the ISDN PRI to be sent to the SIP proxy, and 
vice versa. 

I (foolishly) used .T in both dial peer configurations, in hopes of 
accomplishing this without any major configuration:

dial-peer voice 1 voip
 destination-pattern .T
 session protocol sipv2
 session target sip-server
 codec g711ulaw
!
dial-peer voice 70 pots
 destination-pattern .T
 direct-inward-dial
 port 7/0:1:D


The problem is that the pots dial peer also matches itself much (most) 
of the time, and when a call comes in, it gets sent back out to the 
telco, who sends it back to me, and only then do we send it to the SIP 
server.  This is causing almost every call from PSTN to use up 3 
channels on the PRI!  The recommended solution is to list all the DIDs 
on the SIP side in my dialpeer.... however there are thousands of DIDs, 
few of them are sequential.  We're LNPing customer numbers onto the PRI 
all the time - to manually keep a list of the DIDs inside each AS-5400's 
dial-peer config is completely impractical.

Surely I'm not the first person to encounter this?  Is there a simple 
solution here?  Can the 5400 consult an outside directory?  Can it be 
told not to send a call back out a dial peer that it received it on?  Is 
there some fancy prefixing method I haven't thought of?