13
- January
2017
Posted By : admin
Steps to Configure GSM Gateway for use with vicidial or Asterisk
GSM gateway devices present a great and easy way to start a call center without using a PRI line. They help call centers to start with GSM/mobile SIMs and can be used from as small a setup for 2 ports (SIMs) upto 16 ports. They thus help small setups to be mobile, change the number (SIM) on outbound calls frequently.
The Yeastar GSM gateway is sold by the company in 5 different models ie ( single prot, 2 port ,4 port ,8port ,16port) They all differ in the number of SIM cards they support. The instructions below hold out for all the models except the number of rows one will see in the number of SIM modules etc.
Step 1 :  Configuring the local LAN IP address on the GSM gateway
By default the yeastar gsm gateway come with factory default ip  : 192.168.5.150
Configure your Laptop/pc with a ip(192.168.5.151) same subnet.
Open a browser and browse 192.168.5.150
username : admin
password : password
Goto
System => Network Preferences => LAN Settings.
Make the changes to the IP address as per your local Network , Subnet Mask and Gateway IP Address and connect the GSM Gateway to your LAN.
Our Configuration for this tutorial:
GSM Gateway: 192.168.1.249
Vicidial/Asterisk server IP: 192.168.1.250

Note: The IP address for both should be static and not from your DHCP pool.

Step 2 : Configuring Mobile group
Power off the GSM gateway and insert all the SIM cards in the respective slots.
Power on the Gateway.
Open the IP address of the GSM gateway (192.168.1.249) in the browser window and login.
Click on the Tab Gateway (on top right corner)
Select Mobile Group
Click Add New Mobile Group and fill the below details
  • Group Name : Give name say: GSMGroup
  • Startegy         : select either sequence or balance
  • Group members :  select the slots in which you have inserted sim, if all slots have sim then select all. You can make multiple groups as per your need, like seprate groups for inbound and or outbound calls etc
  • click on save to save your settings.
Step 3 : Creating a VOIP trunk  in GSM gateway
Click VOIP Trunk from the VOIP Setting tab on the left hand side menu.
Click Add Voip Trunk button and fill the details as given below
  • Trunk Type : service provider
  • Type            : SIP
  • Provider name: asterisk
  • Hostname/IP :  enter the ip address of your asterisk or vicidial or goautodial server (192.168.1.250)
  • click on save to save your settings.
Step 4 : Setting Outbound and Inbound route in GSM Gateway
Navigate to Route Settings on the left hand menu where you can see:
1. Mobile to IP
2. IP to Mobile
For Outbound Call Setup:
Click IP to Mobile
Click Add IP to Mobile Route button and fill below details
  • Simple mode: Yes
  • Call Source  : From the dropdown menu select asterisk from dropdown, (we created this in previous step)
  • Call Destination : From the dropdown menu select the group (GSMGROUP), We created in step1
  • click on Save to save the settings.
For Inbound Call Setup:
Click Mobile to IP
click ADD Mobile to IP route and fill below details
  • Simple Mode : Yes
  • Route Name   : give any name say inbound
  • call source   :   GSMGROUP1  (the one created in step 1)
  • Hot line       : 8899   (this is a extension we will be creating in our Asterisk)
  • save
Step 5 : Applying changes
Everytime you make any change in any of the setting, click on Apply Changes to apply the changes to a the GSM Gateway.
Step 6 :  Configuring the SIP trunk in Asterisk/vicidial/goautodial
You can create sip trunk either in GUI or from the command line, we prefer the command line
  • SSH your asterisk server
  • Open the sip.conf file  vi /etc/asterisk/sip.conf
  • add the below settings at the end of the existing entries in the file sip.conf
[8899]
host=192.168.1.249
fromdomain=192.168.1.249
type=friend
disallow=all
allow=all
context=default
qualify=yes
*Note: context can be any as per your inbound setup.
  • save the file and reload asterisk to register the trunk
  • you can check the status of trunk by type sip show peers in asterisk cli.
Step 7 : Setting outbound & inbound
Outbound :
For outbound you need to make changes to the dialplan. Changes can be made from GUI or command line.
  • ssh to server
  • open vi /etc/asterisk/extensions.conf
  • use the below dialplan under default context
for vicidial based system
exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,2,Dial(SIP/8899/${EXTEN:1},,Tto)
exten => _9X.,3,Hangup
For plain asterisk
exten => _9X.,1,Dial(SIP/8899/${EXTEN:1})
exten => _9X.,2,Hangup
  • save the file ,exit,  reload the asterisk
  • Make a test call by dialling any number with prefix 9
  • Use  9  as dialprefix in the Campaign settings in vicidial/goautodial
Inbound setup :
If you are using vicidial or goautodial then go to GUI
  • ADD new DID
  • DID number :  8899
  • DID route  : set it based on your requirement either ingroup or exten.
For those use plain asterisk
go to extensions.conf
goto context default (its existing in all vicidal systems) or put the lines below int he same context as that entered in the previous step in sip.conf.
[default]
exten => _X.,1,Dial(SIP/8001)
The above dialplan entry will dial extension 8001 for all the incoming calls. you can change it as per your requirements for IVR menu or routing to agents.
reload the dialplan from the Asterisk CLI and you are all set to go!!
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Comments

  • hey im not able to dial particulate numbers example 9880XXXXXX…. starting with 98 series im not able to dial… in dialer its taking after 3 second its automatically disconnected … can u help us.. im using dial plan
    exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
    exten => _9X.,2,Dial(SIP/8899/${EXTEN:1},,Tto)
    exten => _9X.,3,Hangup

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