Open Source - How to Configure a phone for Asterisk through it's GUI

Manually configuring a Digium phone without DPMA requires that your Asterisk system have a proper sip device configuration.

Example of sip.conf.  
 

pjsip.conf

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0
 
[105]
type = endpoint
context = internal
disallow = all
allow = ulaw
aors = 105
auth = auth105
 
[105]
type = aor
max_contacts = 1
 
[105]
type=auth
auth_type=userpass
password=1234
username=105
 
[106]
type = endpoint
context = internal
disallow = all
allow = ulaw
aors = 106
auth = auth106
 
[106]
type = aor
contact = sip:106@192.0.2.1:5060
 
[auth106]
type=auth
auth_type=userpass
password=1234
username=106

sip.conf

[105]

type=friend

host=dynamic

context=phones

secret=!*123abc*!

disallow=all

allow=ulaw,alaw,g729

allowsubscribe=yes

callcounter = yes

canreinvite=no

nat=no

In order to configure your phone To connect to Asterisk you will need to do the following:

1. Open the phone web interface by opening your web browser and tying the IP of the phone. By default the user and password is  admin / 789

2. Select the Line Tab and then select Line 1

3. Complete the following information:

   1. User ID extension:  105  

   2. Authorization Name: <Leave blank>

   3. Line Label on Phone: 105

   4. Caller ID Name: 105

   5. Digitmap: <Leave as Defaults>

   6. Password: !*123abc*! (This needs to match the extensions password set in pjsip.conf or sip.conf)

   7. Voicemail Extension or SIP URI: 800 

   8. IP Address or Hostname: <IP of your Asterisk system>

   9. Port: 5060 

4. Click Submit Phone Settings 

Once the configuration has been saved your pone screen should look like the example below.
 

Note: If your see the message Access to this Web User Interface has been disabled when opening the phone GUI on your web browse this means that your phone has already been configured by DPMA or XML file. You will need to reset the phone to factory default in order to have access to its GUI.