How to configure openvox gsm gateway in Asterisk

How to configure openvox gsm gateway in Asterisk?

    In this article you are going to learn how to configure the openvox GSM gateway in asterisk based pbx like Vicidial, Freepbx, Goautodial, Elastix, Issabel etc. the Steps are applicable to all openvox GSM gateway model like SWG 3008, SWG 3016 and SWG 3032. If you are planning to use Openvox GSM gateway with vicidial then this article is for you.

Openvox GSM Gateway asterisk vicidial

Openvox GSM gateway overview:

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 of Openvox

● Support SIP/IAX2 Protocol

● Work as both SIP Client and Server mode

Steps to configure openvox gsm gateway

 Follow the below steps to configure the openvox gsm gateway with asterisk or vicidial also check my video tutorial on Openvox gsm gateway configuration

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.

Step 2: Opevox VOIP SIP Endpoint Settings.

Step 3: SIM Group Configuration.

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

Inbound Rules :

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()

For better understanding on dialplan check my article on dialplan basic tutorial

Checking the SIP trunk status:  

in asterisk cli type below command to check sip trunk status.

sip show peers 

sip show registry 

Conclusion:

    Hope this article helpful in configuring your openvox gsm gateway with asterisk or vicidial. for professional support reach me on skype or telegram id: striker24x7