Setting up the SIP Trunk
Last updated
Last updated
Both Genesys and Parloa have support for Digest authentication; however, to keep a lower level of complexity, IP authentication is used in this guide. When creating the release in Parloa, follow the configuration below:
Platform: Select Phone 2
Platform Settings: Choose Peering / Inbound registration for SIP Trunk connection.
Select Whitelisted IPs for Authentication, and enter Genesys region IPs. For eu-central-1, these are 18.197.177.95, 52.28.196.118, 18.197.58.44, and 18.184.11.49 at the time of this writing. You may refer here for the latest IP addresses and for other regions.
Use SIP URI as username@[subdomain].voip.parloa.com for bot availability.
The settings dialog should look similar to the image below:
Genesys allows its users the option to set up SIP trunks for interconnection between Genesys and their carrier and/or PBX. In our case, even though Parloa won’t be offering proper carrier services, it will be connected as a standard (Bring Your Own Carrier) trunk to make use of the greater range of options that Genesys offers for this kind of connection. https://help.mypurecloud.com/articles/about-byoc-cloud/ is the official documentation that provides much more details about this kind of setup.
Below is a step-by-step guide to complete the setup in a blank Genesys account. If one or more steps are completed, you can freely jump to the next one.
In Admin → Locations, create a location with the site's physical address.
Set the address as available for sites.
Create a site in Admin → Telephony → Sites, selecting the media model as Cloud.
Go to Admin → Telephony → Trunks → External Trunks.
Select Generic BYOC Carrier as the Type.
Set the trunk to "In-Service" and provide it with a descriptive name.
Choose the desired transport protocol. (TLS is highly recommended!)
In the Inbound / Termination section, enter a descriptive value for the Termination Identifier. Leave the Termination Header blank as initial calls from Parloa to Genesys are not expected in this setup.
In the Outbound section, you have two options for configuring the DID with Parloa:
Option 1 – Configure an Outbound SIP DNIS. This will overwrite all SIP invites to Parloa with a specific SIP URI. This setup requires one SIP trunk per Parloa bot.
Option 2 – Do not configure an Outbound SIP DNIS. This allows one trunk towards Parloa to operate effectively, where the DNIS will be dynamically replaced with the virtual DID. This configuration is suitable for using the same trunk with multiple Parloa bots.
Set the Parloa proxy address in the SIP Servers or Proxies field. Use the address provided in the Parloa configuration from the previous steps: [subdomain].voip.parloa.com. The port can be left blank as Genesys will automatically use the correct port based on the Transport type.
Do not enable Digest as it is not used in this setup.
In SIP Access Control, define the IPs of hosts that should be allowed to contact Genesys. For example, add the IP of Parloa's proxy, such as 20.107.52.237.
Enable "Take Back and Transfer" to instruct Genesys to accept and process SIP REFER messages.
Generally, leave the settings in the External Trunk Configuration unchanged, except for the Protocol subsection.
Navigate back to Admin → Telephony → Sites.
Verify that in the site created in the Create Location and Site, the default numbering plans are available.
Depending on the complexity of your existing configurations, you might need to consider creating a separate numbering plan for the virtual numbers that will be used. If necessary, follow these steps:
Create a specific numbering plan for the virtual numbers.
Set up a specific outbound route for this classification to route calls to the trunk.
However, if the default plans and classifications suffice for your setup, you can streamline the process:
Simply assign the newly created external trunk to the outbound route.
This configuration will enable incoming calls to be routed correctly both internally in Genesys and to the external trunk.
In the Architect module, create a flow dedicated to handling Inbound calls. Keep this flow as simple as possible, with the sole purpose of sending calls to the trunk.
Within this flow, access the Main Menu and perform the following actions:
Remove the Disconnect component.
Add a Transfer to Number component.
In the DTMF (Dual-Tone Multi-Frequency) menu, select # (though this choice isn't critical).
Check the box labeled "DTMF goes to this menu choice from any menu."
Set a virtual E164 number in the Number field. Note that this number will be present in the SIP To header but will might overridden by trunk configuration, depending on your Outbound SIP DNIS settings
To enhance the user experience and reduce connection time, set Pre-Transfer Audio to a brief 100ms of blank audio.
Customize the Failed-Transfer message to be more descriptive, as it serves as an indicator of a failed transfer. You can use basic Text-to-Speech (TTS) like "Connection Failed." Other options in this section can remain unchanged.
In the Main Menu section, you can further optimize the call flow for quicker connection:
Set both the "Initial Greeting" and "Menu Prompt" to 100ms of blank audio, as mentioned above.
Choose Transfer to Number as the default choice.
After making these adjustments, remember to Save and Publish the changes. These modifications will take effect the next time a call is routed through this specific route.
In Routing → Call Routing, create a new Regular Route that directs all incoming calls to the Call Flow you've just configured above.
To allow external calls to flow into the portal, you must have at least one valid DID (Direct Inward Dialing) available. This DID can either be purchased through Genesys or routed into the portal through another BYOC (Bring Your Own Carrier) carrier.
Navigate to Admin → Telephony → DID Numbers.
In this menu, you have the option to assign DIDs to various entities, including a person, a phone, or a call flow.
To set up DIDs for your specific scenario:
Assign the DID to a Call Flow.
From the drop-down menu, select the Route to which this Call Flow application was attached.
In scenarios where a customer needs to transition from interacting with a bot to speaking with a human agent:
Ensure you have completed Step 5 in Section 3.1 of your setup.
Inside the flow for the respective bot in Parloa’s portal, insert a Call Control block.
Configure this Call Control block to initiate a REFER to the configured SIP URI. While the domain part of the URI is not critical because Genesys only reads the number part, it's advisable to set it to something like:
Replace xxxxxxxxxxx with the virtual DID configured in Genesys, which should be routed to an inbound flow.
Termination Identifier should match the name set for “Inbound SIP Termination Identifier” in the trunk setup.
Note that the crucial part here is the virtual DID. It can be any E164 number that is not reachable from the PSTN but only from within the system.
You can create these virtual DIDs in Admin → Telephony → DID Numbers. Genesys treats them as real numbers, so you can assign them to persons or call flows in the same manner as described in Section 3.3.
For easier management and distinction from real numbers, it's recommended to tag these DIDs or ranges with descriptive tags.
Scroll down and set both the "Number of times to repeat menu" and "Menu selection timeout" to 0.
