Genesys Cloud CX
Last updated
Last updated
In this user guide, you will learn how to connect Genesys Cloud CX with Syntphony CAI through the Voice Gateway.
By connecting them, you can make phone calls directly from Genesys to your virtual assistant. Genesys Cloud CX and the Voice Gateway follow a set of SIP (Session Initiation Protocol) standards, allowing them to work together smoothly and efficiently. This integration enables seamless communication between the two services, making it easier for you to handle calls and interactions with customers.
This guide is only for Genesys Cloud CX. If you use Genesys Engage, contact Syntphony CAI technical support team for further assistance.
To initiate call transfers between destinations, you must first perform some basic configurations. These initial steps involve setting up phone numbers and SIP trunks, which are essential for establishing the connection using SIP protocols.
When initiating a call transfer from Genesys to the Voice Bot, it is necessary to include the Genesys SIP Trunk within the Voice Gateway. This step ensures the seamless routing of the call to the designated Voice Bot, specifying the desired destination for the transfer.
To do this, it will be necessary to create a request for sip trunk configuration and virtual assistant assignment in the Voice Gateway at the following support link:
In the form, you will have to attach the configuration file indicating all the data in the "Template DNIS" tab:
In this case, indicating that the origin is Genesys Pure Cloud, it will not be necessary to fill in the information in the "Customer SBC Data" tab.
Once all the information has been entered and successfully applied in the Voice Gateway, the data to configure the SIP trunk in Genesys will be provided.
1. Add the Voice Gateway SIP Trunk
Through the Admin UI > Telephony > Trunks > External Trunks, the Voice Gateway SIP Trunk can be added to Genesys. While configuring the SIP Trunk, there are a few important fields.
The External Trunk Named defines the SIP Trunk name within Genesys.
The Inbound SIP Termination Identifier is the termination URI, unique within the Genesys Cloud organization's region. The termination URI will be used by Voice Gateway to direct SIP traffic to Genesys Cloud.
In the SIP Servers or Proxies section, at least one Voice Gateway port must be specified to define where the outgoing requests should be sent, regardless of the request's destination address.
Ensure that all 3 Voice Gateway IP Addresses are added to SIP Access Control to whitelist them and permit SIP access.
2. DID Numbers
To add and configure the DID numbers, go to Telephony > DID Numbers. Assigning the DID number to a person, call flow, or phone is possible. If queuing is used, the DID number should be assigned to a call flow. The assignee will be the flow configured within Genesys.
3. Call Routes
As mentioned above, a flow must be configured to define what should happen if a number receives a call. To do so, the flow can be created in Architect > Flows: Inbound Call. It is possible to define whether the call should be routed to a specific number or added to a queue.
4. Number Plans
Configuring a number plan through Telephony > Sites > Number Plans is necessary. This number will be needed in the next steps when setting up the Genesys SIP Trunk within the Voice Gateway.
In certain scenarios, transferring the call back to Genesys may be necessary to, for example, establish a connection with a human agent. This process can be effortlessly handled through the Voice Flow within Syntphony CAI, allowing for efficient management and smooth connectivity.
To configure the handover back to Genesys you will have to add a cell response specifying in the technical text field the information for the sip transfer of the call (https://docs.eva.bot/user-guide/using-eva/develop-your-bot/voice-agents#transfer-to-human).
To declare you want a call to be transferred using the transfer property you can use the following example (remember to replace the information inside the quotation marks):
The property “dest” defines the destination to be transferred and “uui” the custom message that would be transferred along with the call via the user-to-user SIP header.
To identify in Genesys that the incoming transfer is a callback, it is recommended to filter from the architect flow by UUI (if you want to route by DID number, you should use another specific number).
For details on call control capabilities available from Syntphony CAI, visit Voice Agent
