Description: How to configure Yeastar Gsm Gateway with asterisk/vicidial/goautodial
Yeastar TG GSM/CDMA/UMTS Gateways connect GSM/CDMA/UMTS network to VoIP network directly and support two-way communication: GSM/CDMA/UMTS to VoIP and VoIP to GSM/CDMA/UMTS. It is the best solution ever to connect IP-based telephone systems, and softswithces to GSM/CDMA/UMTS network; and also the best fallback solution when landline goes down
The Yeast Gsm gateway comes in 5 models ie ( single prot, 2 port ,4 port ,8port ,16port)
Step 1 : Configuring the IP to the yeastar 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
Navigate to System == network preferences == Lan settings.
change the IP address as per your local Network , subnetmask & gateway ip and connect the Gsm Gateway to your Lan Network .
(as per this guide i consider 192.168.1.111 as my Gsm gateway ip & 192.168.1.100 my vicidial/goautodial/asterisk server IP)
Note: the Ip address should be static not from your DHCP pool.
Step 2 : Configuring Mobile group
- insert all your SIM cards in the respective slots
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: SIPGROUP1
- 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,
- save.
Step 3 : Creating a VOIP trunk in gsm gateway
Click VOIP trunk from the VOIP setting tab.
Click add Voip trunk button and fill the below details
- Trunk Type : service provider
- Type : SIP
- Provider name: asterisk
- Hostname/IP : enter the ip address of your asterisk or vicidial or goautodial server (192.168.1.100)
- save
Step 4 : Setting Outbound and Inbound route in Gsm Gateway
navigate to Route Settings where you can see two options
1. Mobile to IP
2. IP to Mobile
Outbound Setup:
Click IP to Mobile
Click Add IP to Mobile Route button and fill below details
- Simple mode Yes
- Call Source : select asterisk from dropdown, the one we created in previous step
- Call Destination : select the group (SIPGROUP1) from drop down, this created in step1
- save.
Inbound Setup :
Click Mobile to IP
click ADD Mobile to IP route and fill below details
- Simple Mode : Yes
- Route Name : give a name say inbound
- call source : SIPGROUP1 (the one created in step 1)
- Hot line : 2222 (this is a extension we will be creating in asterisk)
- save
Step 5 : applying changes
once any setting done in Gsm Gateway you have to press Apply Changes.
Step 6 : Configuring SIP trunk in Asterisk/vicidial/goautodial
you can create sip trunk either in GUI or in command line, i prefer in command line
- SSH your asterisk server
- go to vi /etc/asterisk/sip.conf
- add the below settings in last line of sip.conf
[2222]
host=192.168.1.111
fromdomain=192.168.1.111
type=friend
disallow=all
allow=all
context=trunkinbond
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 dialplan
you can write dialplan in command line or in gui, i prefer command line
- ssh to server
- goto 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/2222/${EXTEN:1},,Tto)
exten => _9X.,3,Hangup
For plain asterisk
exten => _9X.,1,Dial(SIP/2222/${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
For more detail about dialplan check this link
Inbound setup :
If you are using vicidial or goautodial then go to GUI
- ADD new DID
- DID number : 2222
- DID route : set it based on your requirement either ingroup or exten.
For those use plain asterisk
go to extensions.conf
create a context ie trunkinbound
[trunkinbound]
exten => _X.,1,Dial(SIP/100)
the above dialplan will dial extension 100 for all the incomming calls.
change it as per your requirement.
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