Extension.conf problem

Hi
I used the Safari book Asterisk the future of telephony to trunk to asterisk servers together. It worked ok but I have no idea how the extensions.confs work. The explanation in the book wasn't that good. Can someone take me through it in laymans terms as I am still fairly new to asterisk

Toronto extensions.conf:
[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
include => internal
include => remote
[internal]
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(SIP/${EXTEN},30)
exten => _2XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _2XXX,n,Hangup()
[remote]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/osaka/${EXTEN})
exten => _1XXX,n,Hangup()
[osaka_incoming]
include => internal


Thanks in advance for your help

i think the idea is that your registered subscribers are given the default context of 'phones', which includes the local and remote contexts. A single digit prefix is then used to route the call out on the apropriate channel, where internal calls are delivered locally via SIP, and remote calls are sent out on the osaka entity defined in your sip.conf (also using SIP)

Thanks for the esponse what I am having a problem is with the following lines
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(SIP/${EXTEN},30)

I don't understand what they mean ?
H

OK

exten => _2XXX,1,NoOp()
Means when you dial a 4 digit number starting with a 2 then first do a NoOp ie no operation but NoOp(Hello World) would show hello world on the asterisk cli

exten => _2XXX,n,Dial(SIP/${EXTEN},30)
Now dial the sip peer that is the number you dialed so for example if you dialed 2122 then it SIP/2122 will be called for 30 seconds then ! go to the timeout extension or if thats missing it will hangup.

Hope that helps

Thanks that did help enough for me to expalin it in my project.