> For a complete page index, fetch https://docs.synthflow.ai/llms.txt. For full documentation content, fetch https://docs.synthflow.ai/llms-full.txt. # Talkdesk > Connect Talkdesk to Synthflow via SIP trunking for AI handling and transfers back to live agents. Talkdesk's External SIP Trunking connects Talkdesk to Synthflow over a SIP trunk. Talkdesk routes calls to a Synthflow agent, and the agent can transfer callers back to live Talkdesk agents. ## Prerequisites | Requirement | Detail | | ------------ | ----------------------------------------------------------------------------------------------------------------------------------------------------------------- | | Talkdesk | Active account with admin access, and Talkdesk CX Cloud with SIP Trunking enabled | | Synthflow | Enterprise plan with a configured agent (v2.0 or later) | | Phone number | At least one DID provisioned in Talkdesk, in E.164 format | | Network | Firewall allows SIP (UDP/TCP `32681` or TLS `32682`) and RTP media between Talkdesk and Synthflow, using the [region addresses](/ip-allow-lists#region-addresses) | ## Talkdesk configuration Configure the trunk and routing in **Talkdesk Admin**. ### SIP trunk 1. In Talkdesk Admin, go to **Integrations** → **SIP Trunking**. 2. Select **Add SIP Trunk** and configure the trunk with the values below. 3. Under **Allowed IPs**, add the Synthflow signaling addresses for your region, listed under [region addresses](/ip-allow-lists#region-addresses). 4. Select **Save**. | Field | Value | | ------------------------ | ------------------------------------------------------------------------------------------------------- | | **Name** | `Synthflow AI` | | **SIP Server / Proxy** | `sip.synthflow.ai`, or the [SIP address for your region](/connect-over-sip#sip-addresses) | | **Port** | `32681` (UDP/TCP) or `32682` (TLS) | | **Transport** | UDP/TCP or TLS | | **Authentication Type** | IP-based (allow the Synthflow IPs) or Digest | | **SIP Username** | Only for Digest. A username you set here. Enter the same value as **Username** on the Synthflow number. | | **SIP Password** | Only for Digest. A password you set here. Enter the same value as **Password** on the Synthflow number. | | **Max Concurrent Calls** | Based on your Synthflow plan and expected call volume | ### Inbound routing Route inbound calls to Synthflow through Talkdesk's call routing engine. 1. In Talkdesk Admin, go to **Voice** → **Ring Groups** or **IVR**, depending on your setup. 2. Create a routing action, or edit an existing IVR menu. 3. Add an **External Transfer** or **SIP Transfer** step: * Set the destination to the DID of your Synthflow number. * Select the **Synthflow AI** SIP trunk as the outbound route. 4. Set a timeout and a fallback to a Talkdesk agent ring group. We recommend a 30-second timeout. 5. Save and activate the routing configuration. For skills-based routing, use Talkdesk's routing rules engine to send specific call types (for example by language, department, or IVR selection) to different Synthflow agent numbers. ### Transfers to live agents 1. Note your Talkdesk SIP domain, or the DID of the target agent ring group. 2. Set the Synthflow [call transfer](/call-transfers) action to route to the Talkdesk ring group DID through the SIP trunk. 3. Talkdesk receives the transfer and routes it to the next available agent in the ring group. ## Synthflow configuration ### Number import Import the Talkdesk 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 provisioned in Talkdesk, in E.164 format | | **SIP Domain** | Your Talkdesk SBC gateway hostname or IP address | | **Username** | The Talkdesk trunk's SIP username, if it uses Digest | | **Password** | The Talkdesk trunk's SIP password, if it uses Digest | | **Outbound Proxy** | Your Talkdesk SBC proxy address, if required | The **SIP Domain** routes Synthflow calls back into Talkdesk for PSTN termination and agent transfers. 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 flows | Scenario | Flow | | ---------------------------------------- | ----------------------------------------------------------------------------- | | Inbound to the agent | Caller → PSTN → Talkdesk → IVR or ring group → SIP transfer → Synthflow agent | | Agent transfers to a live agent | Synthflow agent → SIP transfer → Talkdesk SBC → ring group → live agent | | Outbound from Talkdesk through the agent | Talkdesk IVR or campaign → SIP trunk → Synthflow agent handles the call | ## Verification | Test | What to check | | ------------------------ | -------------------------------------------------------------------------------------------------------------------------------- | | Inbound call | Call the Talkdesk DID from an external phone. Check that it routes to the Synthflow agent, with two-way audio. | | Transfer to a live agent | Trigger an escalation during a Synthflow call. Check that the call lands in the right Talkdesk ring group with caller ID intact. | | Outbound from Talkdesk | Route a Talkdesk IVR call to the Synthflow trunk. Check that the agent answers. | ## Troubleshooting | Issue | Fix | | ---------------------------- | ------------------------------------------------------------------------------------------------------------------------------------------------------- | | Registration fails | Check the **SIP Domain**, credentials, and firewall (UDP/TCP `32681` or TLS `32682`). Check that the Synthflow IPs are in the Talkdesk **Allowed IPs**. | | Calls do not reach Synthflow | Check that the IVR or ring group routing action points to the Synthflow DID through the right SIP trunk. | | One-way or no audio | Open RTP ports in both directions. Confirm G.711 is negotiated on both sides. | | Caller ID missing | Set the `From` header to the caller's E.164 number. Check that Talkdesk passes the ANI in the SIP INVITE. | | Transfer fails | Check that the Talkdesk ring group DID is correct and the SIP trunk is active. Check the Talkdesk transfer logs. | ## Resources * [Talkdesk documentation](https://docs.talkdesk.com) * [Talkdesk support](https://support.talkdesk.com) * [Telephony integrations overview](/telephony-integrations) * [X-EI SIP header](/pass-call-context#x-ei-header)