Genesys SIP Trunk Integration

Overview

This guide describes the steps required to integrate Genesys Cloud with the Sestek Virtual Agent (VA) via a SIP trunk. The VA will operate over this SIP trunk, and agent escalations will be handled using SIP REFER messages.

Step 1 — Number Plan Configuration

Number plans are used to transfer calls from Genesys to Sestek's IVR system.

1. Navigate to Admin > Sites (new UI: Telephony > Sites) and either create a new site or select an existing one.
2. Under Number Plans, create a number plan for Sestek.
3. Select a Classification based on your preferences.

⚠️ If you are using a number list for the number plan, ensure that all numbers intended for the Knovvu Virtual Agent integration are included.

📢 Please inform Sestek of the assigned number so they can complete the required configuration on the IVR side.

Step 2 — External Trunk Configuration

1. Navigate to Admin > Telephony > Trunks (new UI: Digital and Telephony > Telephony > Trunks) and create a new External Trunk.
2. Set the trunk type to Generic BYOC Carrier.

3. Set the protocol to UDP or TLS — TLS is recommended.
4. Under the Inbound section, enter a name of your choice in the Termination Identifier field.

📢 Please notify Sestek of the FQDN to be used during setup.

5. Select the Number Plan configured in Step 1 for Virtual Translator operation.
6. Under the Outbound section, fill in SIP Servers or Proxies with the IP and port information provided by Sestek.

If an SBC is in place, enter the Output SIP Termination FQDN in the following format:

customer.machinehostname

7. Under the SIP Access Control section, enter the same IP and port information provided by Sestek.

⚠️ for enabling Custom SIP Header transmission enable "Conversation Headers" property under protocol section!

Step 3 — Outbound Route Configuration

1. Navigate to Admin > Telephony > Sites > Outbound Routes (new UI: Telephony > Sites), click on your site, and create a new Outbound Route.
2. Under External Trunks, select the trunk created in Step 2.
3. Choose the same Classification used in the Number Plan from Step 1.

Step 4 — Flow Configuration

⚠️ Steps marked as "customer prerequisite" must be completed by the customer before proceeding.

1. Navigate to Admin > Architect and create the main call flow according to your requirements.

   • Within the main call flow, configure the redirect menu using the Transfer to Number option.
   • Enter the number assigned in the number plan — this connects the call to the Sestek IVR.
   • (This step must be completed by the customer — prerequisite.)

2. Once all changes are made, publish the flow.

Step 5 — Call Routing Configuration

1. Navigate to Admin > Routing > Call Routing.
2. Add a new Call Route and select the inbound call flow created in Step 4
3. Provide the inbound call number to Sestek.

Architecture Note

The Knovvu Virtual Agent will serve calls via the configured SIP trunk. For agent escalations, a 2-step transfer (attended/controlled transfer) will be used. Custom SIP headers must be included in the initial INVITE message sent to Genesys, as Genesys Architect's Get Raw SIP Headers action only reads headers from the initial INVITE on the inbound call leg. Headers carried in REFER messages are not accessible via this mechanism.

Example Message:

INVITE sip:<DIALED_NUMBER>@<GENESYS_EDGE_IP>:<PORT>;transport=tcp SIP/2.0
To: "<TRUNK_NAME>" <sip:<DIALED_NUMBER>@<GENESYS_EDGE_IP>:<PORT>;user=phone>
From: "<CALLER_NAME>" <sip:<CALLER_EXT>@<ORIGINATING_IP>;user=phone>;tag=<RANDOM_TAG>
Call-ID: <UNIQUE_CALL_ID>
Via: SIP/2.0/TCP <SBC_IP>:<PORT>;branch=<BRANCH_ID>
CSeq: 1 INVITE
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, REFER
Supported: norefersub, timer
Accept: application/sdp
Contact: "<CALLER_NAME>" <sip:<CALLER_EXT>@<SBC_IP>:<PORT>;transport=tcp>

X-Interaction-ID: <UNIQUE_INTERACTION_IDENTIFIER>
X-Caller-ID: <CALLER_NUMBER_OR_ID>
X-Language: <LANGUAGE_CODE>
X-Queue: <TARGET_QUEUE_NAME>
X-Category: <CALL_CATEGORY>
X-Sentiment: <SENTIMENT_VALUE>

Content-Type: application/sdp
User-Agent: <USER_AGENT_STRING>
Content-Length: <CONTENT_LENGTH>
v=0
o=- <SESSION_ID> <SESSION_VERSION> IN IP4 <MEDIA_IP>
s=-
c=IN IP4 <MEDIA_IP>
t=0 0
m=audio <RTP_PORT> RTP/AVP 96 0 8 101
a=rtpmap:96 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Reference Links