exten => 1702,1,Dial(Sip/2RULE170) ; // If 1701 is

dialed the Dial() application is triggered to connect the

user 1RULE170
sip.conf file
[general]

exten => 111,1,Playback(demo-moreinfo) ; // If 111 is

dialed the Playback application is triggered and demomoreinfo recording is palyed

port = 5060 ; //Port that asterisk server operates or port

to bind to.

exten => 111,2,Playback(demo-thanks) ; // is played

after demo-moreinfo played and finished.

context = default ; //Default context for incoming calls

which is corresponding to the default in extensions.conf
file[4]

exten => 111,3,Hangup ; // It will hang the call for a

disallow=all ; // Disallows the use of all codecs

allow=gsm ; // Enables the codec preferred for
Asterisk.
allow=ulaw // Enables a codec based on -law
algorithm, which provides more dynamic range than Alaw.
allow=alaw // Enables codec based on A-law
algorithm, which requires less CPU processing power
than -law.

Sip.conf
[rule25]
type=friend
host= rule25.domain// Instead of using dynamic, the
host call placed to the IP of rule25.domainname
username=RULE170
fromuser=RULE170 // Is used to override the name in
section title.
secret=RULE170

Creating a user profile in sip.conf

nat=yes
canreinvite=no ; // it is blocking end points connecting
directly. By using this, Asterisk remains in the middle of
the connecting path, which is necessary to detect DTMF.
type=friend ; Asterisk will allow calls to and from this
phone[4]
username=1RULE170; user name
secret=RULE170; password of the user
host=dynamic ; inform Asterisk that the phone will tell
us where it is in the network.
nat=yes ; takes senders ip address and port ignoring
address information in SIP and SDP header from this
peer
Extensions.conf
[default]
exten => 1701,1,Dial(Sip/1RULE170) ; // If 1701 is
dialed the Dial() application is triggered to connect the
Sip channel user 1RULE170.