RZFeeser_old_telephone_boothThis is a basic extensions.conf (the Asterisk dialplan) configuration for Asterisk v1.8. As you can see, only two lines are actually necessary in extensions.conf for the SIP user agents defined in sip.conf to be able to call one another.

If you’re confused, than examine this dialplan in conjunction with my previous blog post on configuring a basic sip.conf in Asterisk 1.8. It may also enhance your understanding of Asterisk to examine the script I posted that complies any version of Asterisk in Ubutnu.

The default location for extension.conf on an Ubuntu 10.04 and 12.04 system is:


The configuration file is commented for maximum clarity. The active code appears in bold. If you have any questions, feel free to post a comment, or contact me directly using this site.





; basic asterisk v1.8 box
; by r.zachary feeser
; /etc/asterisk/extensions.conf

exten => _40X,1,Dial(SIP/${EXTEN})   ; if a match is made on any number 400 – 409, then attempt to establish a new outgoing connection on a channel, and then link it to the existing input channel (i.e. send a SIP INVITE to the matched extension)
exten => _40X,n,Hangup()    ; hangs up the calling channel unconditionally when the call ends