> ## Documentation Index > Fetch the complete documentation index at: https://plivo.com/docs/llms.txt > Use this file to discover all available pages before exploring further. # Troubleshoot Transfer to Human Agent > Test an AI-to-human transfer, read its hangup causes and Dial status, and fix failed, silent, or stuck transfers. > [!NOTE] > > * **What:** Test a transfer from your AI agent to a human agent, and fix one that fails or has no audio. > * **Start with:** a transfer to a phone number. If it connects, the fault is on the SIP path. The setup for each option is on its own page: [DID forward](/docs/voice/use-cases/transfer-to-human-agent/did-forward/), [SIP auth forward](/docs/voice/use-cases/transfer-to-human-agent/sip-auth-forward/), and [Multiparty Call](/docs/voice/use-cases/transfer-to-human-agent/multiparty-call/). *** ## Test Your Transfer **1. Transfer to a phone number first** Point a test transfer at a phone number that reaches the same team. If this connects, your XML, your application, and your Plivo account are all working — which narrows any later failure to the SIP path. **2. Transfer to the SIP endpoint** Run the same transfer against your SIP endpoint. A failure here, after the phone number transfer succeeded, points at the firewall or the SIP credentials rather than at your agent logic. **3. Check audio in both directions** Speak from each end. Audio in one direction only means the RTP media ranges are whitelisted in one direction only. **4. Confirm your application reads DialStatus** Your Dial action URL should receive `DialStatus` and `DialHangupCause`, so your application can fall back or inform the caller when a handoff fails. *** ## Hangup Causes and Dial Status When a transfer fails, the Dial action URL and hangup callback include specific values: | HangupCauseName | Code | DialStatus | Meaning | | - | - | - | - | | `sip_auth_failed` | `4240` | `failed` | SIP credentials were rejected by the remote endpoint | | `sip_auth_timeout` | `4250` | `timeout` | Remote endpoint did not respond to the digest challenge | | `no-answer` | — | `no-answer` | Agent did not pick up within the dial timeout | | `busy` | — | `busy` | Agent endpoint is busy | Your agent application receives `DialStatus` and `DialHangupCause` on the Dial action URL, so it can detect a failed handoff and respond (e.g., retry with a different endpoint, fall back to a phone number, or inform the caller). Every hangup code is listed in [Hangup Causes](/docs/voice/troubleshooting/hangup-causes/). *** ## Troubleshooting | Issue | Solution | | - | - | | Transfer never arrives, and nothing appears in your contact center's logs | Only the RTP media ranges are whitelisted, so the INVITE is dropped before it arrives. Whitelist the [External SIP endpoints](/docs/voice/concepts/firewall-network-configuration/#external-sip-endpoints) on port 5060, and confirm your endpoint is listening on that port | | The Dial action URL reports an empty `DialBLegUUID` | Nobody answered, so no B-leg was created. Check the SIP URI in `` against the [Dial XML reference](/docs/voice/xml/routing/), then check that your endpoint is reachable | | Transfer fails with `sip_auth_failed` (`4240`) | Verify the SIP credentials match exactly. Check your contact center's auth realm. Some platforms authenticate by IP address instead of digest — if yours does, whitelist Plivo's addresses rather than sending credentials | | Transfer fails with `sip_auth_timeout` (`4250`) | Your endpoint is unreachable, or its SIP service isn't running. Check both before re-testing credentials | | Agent answers, but there's no audio | Configure your SIP endpoint to advertise its **public** IP address. A private address in its `Contact` header or SDP leaves Plivo's audio with nowhere to send to | | Call is rejected before the agent's phone rings | No shared audio codec. Offer PCMU or PCMA on your SIP endpoint | | Audio works in one direction only | The RTP media ranges are whitelisted in one direction only. Add a rule for each direction | | Call connects, but the line stays open after one side hangs up | Plivo's in-dialog `BYE` may be sent from a different source port than the one that opened the dialog. A firewall rule pinned to source port 5060 drops it. Whitelist Plivo's addresses without pinning a source port | | Transfer XML doesn't run while your AI agent's `` with `keepCallAlive="true"` is open | Close your AI agent's WebSocket when it decides to transfer. If transfers still drop the call or never run, use a [Multiparty Call](/docs/voice/use-cases/transfer-to-human-agent/multiparty-call/) | | In a Multiparty Call warm transfer, the caller hears your AI agent brief the human agent | Hold the caller with `hold` set to `true`, not `mute`. A muted participant still hears everyone in the Multiparty Call | | In a Multiparty Call, the held caller hears silence instead of hold music | Make your `customerHoldMusicUrl` return XML with a `` element, not an audio file. See [Custom Hold Music](/docs/voice/xml/multiparty-call/#custom-hold-music) | | Caller hears silence during ring | Use the `dialMusic` attribute on `` to play hold music while the agent is being reached | *** ## Related * [Transfer to Human Agent](/docs/voice/use-cases/transfer-to-human-agent/) — When to transfer, and how to choose an approach * [Option 1: DID Forward](/docs/voice/use-cases/transfer-to-human-agent/did-forward/) — Transfer to a phone number * [Option 2: SIP Auth Forward](/docs/voice/use-cases/transfer-to-human-agent/sip-auth-forward/) — Transfer to a SIP endpoint with credentials * [Option 3: Multiparty Call](/docs/voice/use-cases/transfer-to-human-agent/multiparty-call/) — Cold or warm transfer in a Multiparty Call * [What Your Contact Center Must Do](/docs/voice/use-cases/transfer-to-human-agent/sip-auth-forward/#what-your-contact-center-must-do) — Firewall and SIP settings for your contact center * [Hangup Causes](/docs/voice/troubleshooting/hangup-causes/) — Every hangup code and its meaning