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TELEPHONY

Genesys Cloud

A BYOC Cloud SIP trunk connects Genesys Cloud CX to Synthflow for inbound and outbound calls. Genesys routes calls to a Synthflow agent, and the agent can transfer callers back to live agents or place outbound calls through Genesys.

Genesys Cloud has two integration paths. The BYOC trunk on this page provides basic SIP connectivity, and its throughput may be limited. Full integration, with higher concurrency and priority routing, requires Synthflow to be listed in the Genesys AppFoundry. Contact your Genesys account team and Synthflow sales to discuss availability.

Prerequisites

RequirementDetail
Genesys CloudActive CX org with admin access and BYOC Cloud enabled
SynthflowEnterprise plan with a configured agent (v2.0 or later)
Phone numberAt least one DID provisioned, in E.164 format
NetworkFirewall allows SIP (UDP/TCP 32681 or TLS 32682) and RTP between Genesys and Synthflow, using the region addresses

Genesys Cloud configuration

Configure the trunk in the Genesys Cloud admin console under Admin → Telephony → Trunks → External Trunks.

External trunk

  1. In External Trunks, select Create New.
  2. Enter an External Trunk Name, for example Synthflow AI.
  3. Under Trunk Type, select BYOC Carrier.
  4. Under Subtype, select Generic BYOC Carrier.
  5. Set Trunk State to In-Service.
  6. Select the transport Protocol (UDP or TLS).

Inbound settings

The Inbound section covers calls from Synthflow to Genesys. Inbound SIP routing tells Genesys which organization an incoming call belongs to.

  1. Under Inbound, enter a unique Inbound SIP Termination Identifier, for example synthflow.
  2. Copy the org-specific FQDN that Genesys generates, for example synthflow.byoc.usw2.pure.cloud. You enter it as the SIP Domain in Synthflow.
  3. Select the Number Plan Site that contains the number plans for your DIDs.

The region part of the FQDN (usw2, use1, euw1, and so on) depends on your Genesys Cloud deployment region.

Outbound settings

The Outbound section covers calls from Genesys to Synthflow.

  1. Under Outbound, enter sip.synthflow.ai in Outbound SIP Termination FQDN.
  2. Under SIP Servers or Proxies, enter sip.synthflow.ai as the hostname and 32681 as the port, then select Add.
  3. Turn on Digest Authentication.
  4. Enter sip.synthflow.ai as the Realm.
  5. Enter the User Name and Password. Use the same values you enter as Username and Password on the Synthflow number.

For the US and EU regional addresses, see SIP addresses.

Caller ID

  1. Under Caller ID, enter the Caller Address (your DID in E.164 format) and Caller Name.
  2. Set the Prioritized Caller Selection order for your use case.

SIP access control

  1. Under SIP Access Control, add the Synthflow signaling addresses for your region to the allow list. They are listed under region addresses.
  2. Select Save External Trunk.

Do not turn on Allow All in SIP Access Control. Allow only the specific Synthflow IPs, to prevent unauthorized access.

Site assignment

  1. Go to Telephony → Sites and select your site.
  2. Under Outbound Routes, add the Synthflow AI trunk to the appropriate route.
  3. Check that the site has the right Number Plans for the DIDs you want to route.

For inbound call routing, build a call flow in Genesys Architect that sends incoming calls to a queue, IVR, or transfer destination. Route specific DIDs or IVR branches to the Synthflow trunk for AI handling.

Synthflow configuration

Number import

Import the Genesys DID as a custom number. In Synthflow, select Phone Numbers → New Phone Number → Import a Custom Number, and fill in the Import Phone Number form:

FieldValue
Phone NumberThe DID configured in Genesys, in E.164 format
SIP DomainYour Genesys BYOC inbound FQDN, for example synthflow.byoc.usw2.pure.cloud
UsernameThe digest username set on the Genesys trunk
PasswordThe digest password set on the Genesys trunk
Outbound ProxyLeave blank unless Genesys requires a specific proxy

Synthflow sends outbound calls and transfers back into Genesys through this FQDN. Copy the credentials exactly, because incorrect values cause registration failures.

Agent assignment

  1. Create or select an agent (v2.0 or later).
  2. Assign the number to the agent under Deploy in the agent editor.
  3. Wait 1 to 2 minutes for SIP registration to complete.

Synthflow creates the SIP trunk when you assign the number to the agent.

Call metadata on transfers

Genesys reads call metadata from the SIP User-to-User header and ignores individual X-headers on a cold transfer. If your Architect flow needs values such as a customer intent or a correlation ID, turn on Send as UUI header on the call transfer action. The values then arrive in a single header Genesys can read, and the flow splits the payload on the delimiter you configured.

Call flows

ScenarioFlow
Inbound to the agentCaller → PSTN → Genesys Cloud → Architect flow → BYOC trunk → Synthflow agent
Agent transfers to a live agentSynthflow agent → SIP transfer → Genesys BYOC trunk → queue → live agent
Outbound from the agentSynthflow agent → SIP INVITE → Genesys BYOC trunk → PSTN → callee
Outbound from Genesys through the agentGenesys campaign or flow → BYOC trunk → Synthflow agent handles the call

Verification

TestWhat to check
Inbound callCall the DID from an external phone. Check that it routes through Genesys Architect to the Synthflow agent, with two-way audio.
Transfer to a live agentTrigger a transfer during a Synthflow call. Check that the call lands in the right Genesys queue with caller ID intact.
Outbound from GenesysRoute a Genesys call flow to the Synthflow trunk. Check that the agent answers.

Genesys Cloud has a Call Simulator tool under Telephony that tests dial plans and routing without affecting production. Use it to validate your setup before live testing.

Troubleshooting

IssueFix
Registration failsCheck that the SIP Domain in Synthflow matches the Genesys BYOC FQDN. Check the credentials and firewall rules (UDP/TCP 32681 or TLS 32682).
”No route found” on outboundCheck that the Synthflow trunk is assigned to the right site, and the site has matching number plans.
Genesys rejects callsCheck that the digest authentication credentials match in Genesys and Synthflow. Check the SIP Access Control allow list.
One-way or no audioOpen RTP ports in both directions. Confirm G.711 on both sides.
Caller ID missingSet Caller Address in the Genesys trunk. Check the E.164 format in the SIP From header.
Wrong org, or calls go nowhereThe Inbound SIP Termination Identifier must be unique per org and match the FQDN in the Synthflow SIP Domain.

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