Page 133 - Asterisk™: The Future of Telephony

simple extensions.conf file. We are going to build up a dialplan based on this simple
configuration. The dialplan for each server will be very similar to the other one, but for
clarity we will show both. The new lines we’re adding to the file will be italicized.
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
Osaka extensions.conf:
[globals]

[general]
autofallthrough=yes
[default]

[incoming_calls]
[phones]
include => internal
include => remote
[internal]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/${EXTEN},30)
exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _1XXX,n,Hangup()
[remote]
exten => _2XXX,1,NoOp()

Connecting Two Asterisk Boxes Together via SIP | 105