> For a complete page index, fetch https://docs.synthflow.ai/llms.txt. For full documentation content, fetch https://docs.synthflow.ai/llms-full.txt. # Genesys Cloud > Connect Genesys Cloud to Synthflow via BYOC Cloud SIP trunking for inbound, outbound, and transfer flows. 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 | Requirement | Detail | | ------------- | ---------------------------------------------------------------------------------------------------------------------------------------------------------- | | Genesys Cloud | Active CX org with admin access and BYOC Cloud enabled | | Synthflow | Enterprise plan with a configured agent (v2.0 or later) | | Phone number | At least one DID provisioned, in E.164 format | | Network | Firewall allows SIP (UDP/TCP `32681` or TLS `32682`) and RTP between Genesys and Synthflow, using the [region addresses](/ip-allow-lists#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](/connect-over-sip#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](/ip-allow-lists#region-addresses). 2. Select **Save External Trunk**. > **Warning** > > 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](/phone-numbers#adding-custom-numbers). In Synthflow, select **Phone Numbers** → **New Phone Number** → **Import a Custom Number**, and fill in the **Import Phone Number** form: | Field | Value | | ------------------ | ---------------------------------------------------------------------------- | | **Phone Number** | The DID configured in Genesys, in E.164 format | | **SIP Domain** | Your Genesys BYOC inbound FQDN, for example `synthflow.byoc.usw2.pure.cloud` | | **Username** | The digest username set on the Genesys trunk | | **Password** | The digest password set on the Genesys trunk | | **Outbound Proxy** | Leave 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](/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](/call-transfers#transfer-modes). If your Architect flow needs values such as a customer intent or a correlation ID, turn on [Send as UUI header](/sip-transfers#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 | Scenario | Flow | | --------------------------------------- | ----------------------------------------------------------------------------- | | Inbound to the agent | Caller → PSTN → Genesys Cloud → Architect flow → BYOC trunk → Synthflow agent | | Agent transfers to a live agent | Synthflow agent → SIP transfer → Genesys BYOC trunk → queue → live agent | | Outbound from the agent | Synthflow agent → SIP INVITE → Genesys BYOC trunk → PSTN → callee | | Outbound from Genesys through the agent | Genesys campaign or flow → BYOC trunk → Synthflow agent handles the call | ## Verification | Test | What to check | | ------------------------ | ------------------------------------------------------------------------------------------------------------------------------- | | Inbound call | Call 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 agent | Trigger a transfer during a Synthflow call. Check that the call lands in the right Genesys queue with caller ID intact. | | Outbound from Genesys | Route 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 | Issue | Fix | | ------------------------------ | ---------------------------------------------------------------------------------------------------------------------------------------------------- | | Registration fails | Check 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 outbound | Check that the Synthflow trunk is assigned to the right site, and the site has matching number plans. | | Genesys rejects calls | Check that the digest authentication credentials match in Genesys and Synthflow. Check the **SIP Access Control** allow list. | | One-way or no audio | Open RTP ports in both directions. Confirm G.711 on both sides. | | Caller ID missing | Set **Caller Address** in the Genesys trunk. Check the E.164 format in the SIP `From` header. | | Wrong org, or calls go nowhere | The **Inbound SIP Termination Identifier** must be unique per org and match the FQDN in the Synthflow **SIP Domain**. | ## Resources * [Create a BYOC Cloud trunk](https://help.genesys.cloud/articles/create-a-byoc-cloud-trunk/) (Genesys) * [Configure SIP routing for a BYOC Cloud trunk](https://help.mypurecloud.com/articles/configure-sip-routing-for-a-byoccloud-trunk/) (Genesys) * [External trunk settings](https://help.genesys.cloud/articles/external-trunk-settings/) (Genesys) * [Telephony integrations overview](/telephony-integrations) * [X-EI SIP header](/pass-call-context#x-ei-header) > Connect Genesys Cloud to Synthflow via BYOC Cloud SIP trunking for inbound, outbound, and transfer flows.