I have two asterisk boxes connected using IAX. There are two T1s on each box. I have all my dialing rules in one of the asterisk boxes and all of my agents register on the same box where I have all the dialing rules. See diagram below: Asterisk_1 <--2xT1--> PSTN || || Asterisk_2 <--2xT1--> PSTN || || SIP_Agents I'm wondering how can I configure extensions.conf in Asterisk_1 so that EVERY incoming call (regardless of DID or CallerID or whatever) received from PSTN in Asterisk_1 is routed via IAX to Asterisk_2? Basically, anything that arrives from Zap/g1 or Zap/g2 in Asterisk_1 should be automatically routed to Asterisk_2 preserving all call features, such as DID, CallerID, etc. Any ideas? Thanks, Daniel