Extension.conf problem

Asterisk Asterisk Support
1

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

2

If you go step by step in the book you should have any any problems. You an also:

  1. Join the asterisk users list (lists.digium.com)
  2. Hop on IRC (irc.freenode.net #asterisk)
  3. Search on around on voip-info.org
  4. GOOGLE !!!

Getting back to your question, your dialplan seems OK. What are you trying to do ?