Configure Dinstar GSM Gateway with Asterisk based system

Rate this post

Configure Dinstar GSM Gateway with Asterisk based system

In this article, We will explain step by step to install and configure Dinstar GSM Gateway in VICIdial, Goautodial, FreePBX and other Asterisk based PBX servers.

 

The DINSTAR GSM/CDMA gateway enables providers to directly originate/terminate calls from/to local GSM networks. It is a cost-effective SIP/GSM gateway for SOHO, SMEs and system integrators, and also opens up new revenue generating opportunities for service providers. This document is suitable for following products:

  •  DWG2000B GSM/CDMA VoIP Gateway
  •  DWG2000C GSM/CDMA VoIP Gateway
  •  DWG2000E GSM/CDMA VoIP Gateway
  •  DWG2000F GSM/CDMA VoIP Gateway
  •  DWG2000G GSM/CDMA VoIP Gateway

Step1 : Initial Configurations of GSM gateway 

Power on the GSM Gateway and connect the gateway to the Network. By default the Dinstar Gateway is pre-assigned with IP : 192.168.11.1. Make sure this default IP should not have conflict with your existing network. Assign 192.168.11.2/255.255.255.0 to your PC and connect this gateway to your lan via a LAN cable.

  • Now open a browser and type  http://192.168.11.1
  • The default username and password is  admin/admin

Once you are logged in, Click the Quick Config option from the Left pane, in the quick config we will be assigning a new IP and SIP server IP details. In the quick-config network settings page change the default IP to a static from your network. For this guide i am assigning 192.168.1.222

Now browse http://192.168.1.222

Step 2 : Configuring SIP settings in  GSM Gateway

Click – IP Trunk Configuration–> IP Trunk

enter the below settings

INDEX  ——–   31
IP         ——–   192.168.1.19  (this is my asterisk server ip address)
PORT   ——–   5060
Description—-   asterisk
KeepAlive Enable —–  yes (select the check box)

Step 2.1 : SIP Parameter settings

Local SIP Port Settings :
Go to System Configuraiton–> SIP Parameter
Enable below settings
All Ports Register Used Same User ID : Yes
Local SIP port : 5060
Is Register : yes

Step 2.2 : SIP Port Parameters

Go to System Configuration–> Port Parameter
enter the below Parameters
Current Port : 0
SIP User ID : gsm222
Authenticate ID : gsm222
Authenticate Password : dinstar

To VOIP hotline : 1234

Step 3 : Gsm Port Grouping
Go to Port Group Configuration
click new and enter below detailsIndex : 0
Description: asterisk
Select Mode : Cyclic Ascending
and Select all the Check box of port ie:port 0 to Port 7
Note: if you have only 5 sims inserted then select Port 0 to Port 5, then Save .

Step 4 : Routing – Outbound ie: IP-Tel
this step is to create a routing to dial out via Gsm gateway from IPPBX(asterisk)
Goto — Routing Configuration–>IP-Tel Routing
enter Below settings
Index : 31
description: default
Source Prefix : any
Source IP : 31<Asterisk>
Destination Prefix : any
Destination: Port Group: 0<Asterisk>

Step 5 : Routing – Inbound ie; Tel-IP
This step is to create a inbound to receive calls in IPPBXGoto Routing Configuration–> Tel–> IP routing
Enter below details
Index : 31
Description ; default
Source Prefix : any
Source : Port Group — 0<Asterisk>
Destination Prefix : any
Destination : IP (31<Asterisk>)

Step 6 : Some General Gateway settings
Go to  Operation–>Tel-IP Operation Page
Enter Below details
Index : 31
Source Prefix ; any
Source Port : 0<Asterisk>
Destination Prefix: any
Operation: Allow Call & Auto Call
description : Voip Hotline

Step 6.1 : General settings
GoTo System Configuration –> Service Parameter
Select below two options
Allow Call from GSM to IP without Registration : Yes
Allow Call from IP to GSM without Registraiton : Yes

GSM Part is Done , now configure the SIP server  (asterisk server)

Step 7 : Create a SIP Peer in Asterisk
Log in to your Asterisk server in Command line ssh the server
Goto  vi /etc/asterisk/sip.conf
and create a peer as show below at last line

[gsm222]
username=gsm222
secret=dinstar
host=dynamic
type=friend
disallow=all
allow=all
qulify=yes
dtmfmode=auto
context=trunkinbound

  • save the file and exit.
  • go to asterisk cli (asterisk -vvvvvr)
  • type  sip reload
  • now check whether the gsm gateway is registered by typing sip show peers , it will show OK.

 

For any query or issue, feel free to discuss on http://discuss.eduguru.in
%d bloggers like this: