Description: How to configure the openvox gsm gateway in asterisk,vicidial,freepbx
Models : Openvox gsm gateway model SWG 3008/SWG 3016 SWG 3032
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Openvox GSM Gateway asterisk vicidial
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Abount OpenVox
OpenVox Communication is a global leading provider of the VoIP Gateways, IPPBX and open source Asterisk Telephony Cards with excellent quality and experience. In 2013, OpenVox USA Branch was set up in California, the top economic and financial strength state in the United States, and marked an important step in the global expansion. OpenVox has earned a good reputation over 100 countries worldwide with excellent product quality, professional technical support and efficient logistic services
OpenVox SWG-3016/32 series wireless gateways which can compatible with a series of modules(2G/ 3G/4G) , enabling interconnection between GSM / WCDMA /LTE network and VoIP network safely and effi-ciently. They can bring you excellent HD voice service with multiple codecs, including G.711U, G.711A,GSM, G.722, G.726, G.729, and also flexible SMS service with multiple SMS API. The SWG-3016/32 se-ries gateways is perfect compatible with Asterisk, 3CX, FreePBX, FreeSWITCH SIP server and VOS VoIP system platform.
Major Features
● Support SIP/IAX2 Protocol
● Work as both SIP Client and Server mode
Asterisk
Asterisk is a free and No:1 open source framework for building communications applications and is sponsored by Sangoma.
Vicidial
VICIDIAL is a software suite that is designed to interact with the Asterisk Open-Source PBX Phone system to act as a complete inbound/outbound contact center suite with inbound email support as well.
Steps Involved
1. Gateway Network Configuration
2. VOIP SIP endpoints configuration
3. SIM Groups configuration
4. Call routing Configuration - inbound/outbound
5. asterisk SIP settings and Dialplan.
Step 1: Openvox GSM Gateway Network Settings
The Default Factory set IP address of openvox gsm gateway is 172.16.98.1
The default Login Credentials are admin/admin
For Best Practice change the Default ip of the GSM Gateway to your LAN IP address range,
To change the IP address connect the gateway to your LAN or connect one to one to your PC/LAPTOP with Same Subnet IP of gateway ie:172.16.98.2/24 as shown below.
Access the GSM Gateway portal by browsing http://172.16.98.1 with any of your favorite browsers, t
Login with the default credentials are admin/admin.
Now under Network tab/menu , change the IP Address by setting the TYPE to Static as shown below and set your LAN IP address to this Gateway, i have used192.168.1.252 as static ip to my device.
save the configuration, followed to that the device is restart,
Now access your GSM Gateway with the new IP Address 192.168.1.252.
Step 2: VOIP SIP Endpoint Settings.
Next you need to Create a SIP Peer in GSM Gateway, which will be registered as SIP trunk in your asterisk /SIP server to Make and receive calls
Login to your GSM Gateway WebPortal.
Navigate to VOIP > VOIP Endpoints from the TOP Menu
Under VOIP Endpoints Select Add New SIP Endpoint
Now in the New SIP Endpoint section, fill the details as shown below
Name : Provide a Name Better use number eg:6003
username : enter a username for SIP Peer eg: 6003
Password : password to register 6003 eg:1234
Registration: Server (So gateway register as trunk in Asterisk/sip )
Hostname : Dynamic
transport : UDP
NAT : No
Save and Apply config.
Step 3: SIM Group Configuration.
Next we need to Group all the SIM Ports to a single group, so that we can dial in all SIM Ports in RoundRobin fashion.
Navigate to Routing > Groups
Click Create New Group and Add the below details.
GroupName : SIMGROUP
Type : Module
Policy : RoundRobin
Members: Select ALL or Select the SIM's to you wan in group
Save and Apply.
Step 4: Call Routing Rules in OPENVOX
Next we need to Create two call routing rules
1. Outbound Rules
2. Inbound rules.
Outbound Rules used to dialout via SIM Ports for the calls initiated via Asteris or any SIP servers.
Inbound Rules used to Route the Incoming calls landing on GSM Gateway to a destination like Asterisk server Extensions , IVR etc.
Outbound Rules:
Navigate to Routing > Call routing Rules > New Call Routing Rule
Enter the Below details:
Routing Name : OUTBOUND
Call Comes From: 6003 (SIP Endpoint created in STEP 2)
Send Call Through : SIMGROUP (Group created in STEP 3)
SAVE and APPLY
Navigate to Routing > Call routing Rules > New Call Routing Rule
Enter the Below details:
Routing Name : INBOUND
Call Comes From: SIMGROUP (Group created in STEP 3)
Send Call Through : 6003 (SIP Endpoint created in STEP 2)
SAVE and APPLY
Step 5: Asterisk SIP Trunk and Dialplan Settings
Now we are all set in Openvox GSM gateway configuration, now we need to create SIP trunk /carrier and dialplan in asterisk to dialout and dial in.
In Your asterisk or vicidial use the below SIP turnk configuration
either enter in sip.conf or admin> carriers in gui
register => 6003:1234@192.168.1.252:5060/6003
[6003]
secret=1234
fromuser=6003
username=6003
host=192.168.1.252 ; this is my gsm ip
type=friend
context=trunkinbound ; set this to your incoming context
disallow=all
allow=aulaw,alaw,gsm
insecure=port
dtmfmode=rfc2833
DialPlan
For vicidial outbound dialplan
exten => _9X.,1,AGI(agi://127.0.0.1/call_log)
exten => _9X.,n,Dial(SIP/6003/${ETEN:1},,Tto)
exten => _9X.,n,Hangup()
For Asterisk
exten => _9X.,1,Dial(SIP/6003/${ETEN:1},,Tto)
exten => _9X.,n,Hangup()
For inbound
In Vicidial Create a DID as 6003
or use below dialplan in asterisk incoming context.
[inbound]
exten => 6003,1,Answer()
exten => 6003,n,Dial(SIP/1001)
exten => 6003,n,Hangup()
Checking the SIP trunk status:
in asterisk cli type below command to check sip trunk status.
sip show peers
sip show registry
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