Simple Asterisk Phone System…

asterisk-by-digiumBelow I detail how to set up a basic asterisk dial plan with 2 extensions, a SIP Trunk provider, and incoming calls ringing both extensions.

in /etc/asterisk/sip.conf add:

[100] ; extension 100
username=100
secret=100
host=dynamic
nat=yes
type=friend
qualify=yes
context=phones

[101] ; extension 101
username=101
secret=101
host=dynamic
nat=yes
type=friend
qualify=yes
context=phones

[SIP-PROVIDER] ; Trunk to/from SIP Provider
type=friend
host=IP-OF-SIP-PROVIDER
port=5060
username=USERNAME-FOR-TRUNK
secret=PASSWORD-FOR-TRUNK
callerid=CLI
nat=yes
qualify=yes
dtmfmode=rfc2833
disallow=all
allow=g722
allow=alaw
allow=ulaw
allow=gsm
allow=ilbc
deny=0.0.0.0/0.0.0.0
permit=IP-OF-SIP-PROVIDER
context=from-pstn
language=en_GB

In /etc/asterisk/extensions.conf, add:

[to-pstn] ; Any number dialled with 9XXX will go out this trunk as xxxxx
exten => 9|X.,1,Dial(SIP/${EXTEN}@SIP-PROVIDER,1000)
exten => 9|X.,2,hangup

[from-pstn] ; Any Incoming call from the trunk should call both ext 100 & 101
exten = _X.,1,Dial(SIP/100&SIP/100,120)
exten = _X.,2,hangup

[phones] ; container for phones
exten = 100,1,Dial(SIP/100,120)
exten = 100,2,hangup
exten = 101,1,Dial(SIP/101,120)
exten = 101,2,hangup

include => to-pstn

reload asterisk

asterisk -rx "reload"

program some sip phones with the details above IE username=100, Password =100, IP=IP-OF-ASTERISK, and make calls.

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