Asterisk 13: Set Up Local SIP Channels

First, we will create the user/phone account in sip.conf:

secret=password1111 ;; *change this!*
qualify=yes ;; keep NAT open
mailbox=20 ;; voicemail boxl

This creates a SIP phone called “phone1”, with the username “phone1”, using the password “password1111”. This also links it to the from-internal context in extensions.conf where I have the rest of my extensions. If you want it to go to a different part of the configuration, change that line here.

Now, edit voicemail.conf and add the new mailbox (The default context is quite a ways down):

; Note: The rest of the system must reference mailboxes defined here as mailbox@default.

20 => 20,User1,

In the Asterisk console, reload:


In the SIP client, the SIP address is now phone1@asterisk.ip.address. The password is what was set in sip.conf. You should now be able to dial other extensions! In some SIP clients, you may need to turn off/disable sRTP Encryption.

Now, the next thing we could do is make the SIP phone an extension in Asterisk. Edit extensions.conf and add the following to [from-internal] context (or whatever your normal extensions context is):

exten => 300,1,Dial(SIP/phone1,20)

In the Asterisk console, reload the dialplan:

dialplan reload

Your SIP phone is now extension 300!

There are a TON of different options for SIP phones, so this is just the beginning.