[Asterisk-Users] Call routing

[Daniel Salama]

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
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