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The VoIP GUI is a set of VoIP software installed in the Dragino VoIP products. It can be used to manage the Device's VoIP settings via web interface.

The products DT01 and OV1 is installed with this set of software.

VoIP Settings

The VoIP part is base on Asterisk. User can configure the VoIP settings in the Web UI or configure Asterisk advance settings via SSH access to the devices.


The Overvew page shows if the device register to SIP/IAX server or not.

Shows Server Register Status

General Settings

This page define the general settings for VoIP and Asterisk.

Genera Settings

VoIP Server Settings

This page defines how device connect to VoIP services. Put your voip server here.

VoIP Servers Settings

Configure each SIP/IAX2 service entry.

VoIP Server Entry Settings

VoIP Clients

We can set the device as a VoIP server. The client page is to define the clients account.

SIP / IAX Client Accounts

Client Entry:

Client Entry Settings

Dial Plan

Dial plan is a set of dial rules. It defines how the system send out dialed numbers to VoIP servers.

Dial Plan Format

Below is the explanation of the dial plan.

Dial Plan Overview
Dial Rule settings.

Definition of dial rule entry:

  • Match Pattern: Define what number will use this dial rule. Same the extension pattern defined in Asterisk, it can be a complete number or a pattern.
    • If extension name starts with '_', it is consider as pattern.
    • When use the pattern, some characters have special meaning:
    • - X - any digit from 0-9
    • - Z - any digit from 1-9
    • - N - any digit from 2-9
    • - [12679] - any digit in the brakets (in the example: 1,2,6,7,9)
    • - . - (dot) wildcard, matches everything remaining
    • _1234. - matches anything starting with 1234 excluding 1234 itself).
    • Note: Do not use '_.', because it will match everything even the predefined extensions!!!


359ZXXXXXX - This will match all dialed numbers that start with 359, and are 10 digits long( including 359)

0XXX. - This will match all dialed numbers that begin with 0 and are minimum 5 digits long (including 0)

  • Sub Number Offset, Sub Number Length

These two setting is used to get a sub-number from the dialed number and use this sub-number as the number dial to the VoIP provider.

The offset tells the position to get the sub-number and the length tells the length of this sub-number. If Length is blank, the sub-number will be the number from the offset to the end of dialed number.


If the dialed number is 9123456, while:

Offset is 0, length is 4. --> the real outgoing number 9123

Offset is 1, length is blank --> the real outgoing number is 123456

  • Add Prefix, Suffix

Add prefix or suffix to the real outgoing number.

  • Use trunk

Select the VoIP service provider to send the outgoing call.

  • VoIP Protocol

The dial out voip protocol , SIP or IAX2, depends on the provider you choose.

Dial Plan Examples

Dial Rule Example
Dial Rule Example

Debug Dial Plan

User can debug how dial plan works and see the actually dial out number via the asterisk CLI.

  • Step 1: Log in the device via SSH
  • Step 2: run asterisk –vvvvvvgrc to access the asterisk CLI.

Then it will shows the debug info when you dial out

Debug Dial Plan in Asterisk CLI

VoIP Set up example

In the below examples, we use DT01 as demonstration in the photo. it also available for the outdoor VoIP ATA OV1.

Configure as SIP ATA

This is the most normal configuration case for DT01/OV1, the structure is as below:

Configure device as SIP ATA

Below is the configure example for using Voipbuster service:

Step 1: Input Service Provider info

Configure device to register to VoIPBuster

Step 2: Configure Dial Plan

Configure dial rule for using the VoIPbuster

After above configure, user will be able to use the normal phone to dial out via VoIP Buster.

Configure as SIP Server

In this application, device connect to SIP provider like above. Besides that, device also acts as a SIP server. So softphones or IP phones can register to the device and make outbound calls via device’s trunks. Structure is as below:

Configure device as SIP Server

User can add clients via the VoIP --> Clients page. VoIP client uses the type soft phone while create.

Create SIP clients

Link two DT01s via IAX2 protocol

Two DT01 can link to each other so extension behind them looks like in the same office and calls between all extensions are free.

Link two ATAs

Set up example:

Step 1:

  • Create a softphone client 8003 in DT01 #1,
  • create a softphone client 6003 in DT01 #2

Step 2: In DT01 #1:

  • In VoIP --> Servers, set up an IAX2 account to register to DT01 #2’s account 6003.
  • In VoIP --> Dial Plan, create a dial rule with below info:
  • Match Pattern: _6.
  • Use Trunk: Select DT01 #2 trunk
  • Protocl: IAX2

Step 3: In DT01 #2::

  • In VoIP --> Servers, set up an IAX2 account to register to DT01 #1’s account 8003.
  • In VoIP --> Dial Plan, create a dial rule with below info:
  • Match Pattern: _8.
  • Use Trunk: Select DT01 #1 trunk
  • Protocl: IAX2

After above configure, all the extensions in DT01 #1 (8xxx) is able to reach the DT01 #2 extensions (6xxx) by simple dial the number 6xxx. Inverse for 6xxx to call to 8xxx.