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Configuring SIP Server
In this example, Asterisk acts as a registrar and a proxy. It
accepts registration from VoIP network, binding the peer's SIP
URL with its location (such as IP address). For call request
from various peers, Asterisk first checks if it is authorized. If not,
it simply denies it. And for incoming calls requesting other
extensions belonging to it, Asterisk finds the corresponding
location, and relays the SIP message to it.
First make sure Asterisk is configured with user information.
Note that some other SIP servers do not need you to configure
this in advance, so in that case simply skip this step.
Asterisk SIP configuration file is located in /etc/asterisk/sip.conf.
A sample configuration is shown below, with brief
semicolon-separated explanation on the right:
bindport=5060
; service port
203
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