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EnableX provides Internal SIP Gateway and facilitates the legacy world of SIP endpoints and SIP Gateways to communicate with EnableX multi-party call. EnableX has also partnered with the Telephony Service Provider to provide a way to connect the EnableX conference using normal PSTN phone or mobile phone without any internet connectivity!  

This Solution Document explains how to enable SIP/PSTN Interconnect Service and make inbound / outbound calls to/from EnableX platform.


In order to enable SIP/PSTN Interconnect Server, the below mentioned provisioning tasks need to be taken up: 

For PSTN Dial-in   

To adhere to the Legal Guidelines, you may use a phone number procured by EnableX.  You may opt to choose either of the following type of Phone Numbers:

  • A Shared Phone Number: You may opt to choose a shared phone number (i.e. shared among many other applications) for your application too. From your Portal, you need to choose a shared phone number for your application.  Once chosen, its ready for use.
  • Dedicated Phone Number:  You may opt for a dedicated number for your application only. From your Portal, you may request to buy a new Phone Number submitting documents required by Telco Authorities. After procurement, the phone number will be assigned to your Account for use.

For PSTN Dial-out   

Alike PSTN Dial-In Provisioning, you may opt for shared or dedicated phone number for Dial-Out Service. The same PSTN Number may be used for both Dial-In and Dial-Out.

Note that the same phone number will be your valid caller-id for all outbound calls generated from EnableX Session.

For SIP Trunking 

For SIP Provisioning, you need to submit your SIP Server Information with us and EnableX will direct all SIP Calls to the provisioned server.

To submit SIP Server Information:

  • Login to Portal
  • Go to My Apps
  • Go to App’s Settings
  • Open the SIP Settings Tab

Joining Session through SIP/PSTN Calls 

Enable a Room to accept SIP 

A room must be defined with SIP Enabled option to make or receiving calls in a Session. To enable SIP, use the following in the JSON Payload in Room Creation Server API Call:

	“sip”: {
		“enabled”:  true

Refer Create Room API Call

The Response JSON to create SIP enabled Room, will have additional SIP Access Information as:

	“sip”: {
		“enabled”:  true,
		“pin”: “99999”,
		“meeting_id”: “99999”

The Meeting ID and PIN are important data to be used to get connected to the right session using SIP Endpoint or PSTN Dial-in.

Join Session through Dial-In

From SIP End Points: On getting connected through SIP Endpoints:

  1. You will use a SIP Server Address provisioned through Portal (explained above).
  2. You need to use PIN and Meeting ID when prompted.

From PSTN/GSM Phone: To join a session through inbound call from PSTN/GSM Phones:

  1. Dial the Shared Number you chosen or Dedicated Number you purchased and configured with your Application (to know phone number configured, refer API Documentation)
  2. On getting connected, IVR prompts you for Meeting ID and PIN.  

Invite to join Session through Dial-Out

Being in Session, you can initiate outbound call either to PSTN/GMS Phone Number or to a SIP end Point using respective Client API toolkit. On receiving the call, the called party is put into the RTC Session.

Please refer API Documentation for EnxRoom.makeOutbound() method to know more details about the API call and its responses.