With editor you prefer open /etc/asterisk/sip.conf for editing. Add next in [general] section:

[general]
context=default Default context for incoming calls
recordhistory=no Do you want to record your calls
realm=example.com Realm for digest authentication
bindport=5060 UDP port for SIP client
bindaddr=0.0.0.0 IP address to bind to (0.0.0.0 bind to all)
srvlookup=yes Enable DNS SRV lookups on outbound calls
tos=reliability lowdelay,throughput,reliability,mincost,none
disallow=all First disallow all codec's
allow=alaw Then allow what you need
allow=ulaw
allow=gsm
localnet=192.168.0.0/255.255.0.0 All RFC 1918 addresses are local networks
qualify=yes
nat=no
canreinvite=no

On next link you can find all sip options: SIP options Now is time to add some SIP client. On the end of file add this:

[600] This is extension number
type=friend
host=dynamic
callerid=kotahila This will be on your friend VoIP phone when you call him :)
disallow=all
allow=ulaw
dtmfmode=rfc2833
mailbox=601
context=numberplan Which dial plan user is allow to use

So, when you want to call kotahila you will dial 600. Extension number depends of number of SIP client which you have connected on your server. I use SIP client to connect end user with server, and if I want to connect 2 Asterisk servers I will use IAX2. After this you need to restart asterisk service.