> ## Documentation Index
> Fetch the complete documentation index at: https://docs.julian.11x.ai/llms.txt
> Use this file to discover all available pages before exploring further.

# SIP Calling

> Connect 11x to your phone system over SIP: receive calls from it, and transfer live calls back to it.

11x connects to cloud phone systems such as Zoom Phone in both directions. Your phone system can send calls to a
11x agent, and the agent can live-transfer callers back to a queue or extension that only has a SIP address.

```text theme={}
Caller -> your phone system (IVR) -> 11x agent -> your queue
                 inbound: to 11x         outbound: transfer back
```

| | What it does |
| - | - |
| **Inbound** | Your phone system sends a call to the 11x agent over SIP |
| **Outbound** | The 11x agent transfers the caller to a SIP address of yours |

## Inbound: send calls to 11x

Your phone system sends the call straight to a SIP address that routes to one 11x agent. The call is handled like any
other inbound call to that agent: same greeting, call record, and post-call steps.

**What the 11x team provides**

| Item | Example | Notes |
| - | - | - |
| **SIP address** | `sip:+16505550100@acme.sip.twilio.com` | Each SIP domain routes to one agent |
| **Authentication** | Username and password, or an IP allowlist | Calls without valid credentials or from unlisted IPs are rejected |

**What we need from you**

| Item | Example | Why it's needed |
| - | - | - |
| **Caller's number** | `From: sip:+14155550123@pbx.example.com` | 11x creates the contact from the number before the `@` |
| **Source IP addresses** | Your phone system's SIP signaling IPs | Only if you authenticate by IP allowlist instead of credentials |

> [!NOTE]
>
> The caller's number must be the user part of the SIP `From` address, in E.164 format (`+14155550123`) or as US digits
> (`14155550123`). A call whose `From` user isn't a phone number, such as `sip:frontdesk@pbx.example.com`, is rejected.

## Outbound: transfer calls to your team

11x can live-transfer a caller to a SIP address, such as a call queue in Zoom Phone. The caller stays on the line and
is connected to the next available rep. Phone and SIP transfers behave the same for the caller.

**What we need from you**

| Item | Example | Why it's needed |
| - | - | - |
| **SIP address** | `sip:support-queue@pbx.example.com` | Where 11x sends the call |
| **SIP credentials** | Username and password | Only if your provider authenticates incoming SIP calls |
| **Source IP allowlist** | Provided by the 11x team | Only if your provider accepts SIP calls from known IP addresses |
| **SIP headers** | `X-Account-Id: 12345` | Optional. Passes context from the call to your phone system |
| **Caller ID preference** | Caller's number or the agent's number | What your rep sees when the transfer rings |

> [!NOTE]
>
> Some phone systems expose a queue's SIP address only inside their own account. Confirm with your provider that the
> address accepts calls from an external SIP source before going live.

**Configure the transfer.** Open the agent in 11x and go to:

```text theme={}
Instructions > Tools > Transfer Call > Add New Transfer Tool
```

| Field | Value |
| - | - |
| **Name** | A unique name the agent uses for this tool, such as `transfer_to_support` |
| **Destination Type** | SIP |
| **SIP URI** | Your destination's SIP address |
| **SIP Username / Password** | Your SIP credentials, if your provider requires them |
| **SIP Headers** | Static headers, or headers the agent fills in during the call |
| **Use Agent Caller ID** | Off shows the caller's number to your rep; on shows the agent's number |
| **When to use** | When the agent should pick this transfer |
| **Briefing** | None (Blind Transfer) for a queue, where no one waits for a briefing |
| **Fallback to AI on transfer failure** | On, so the caller returns to the agent if the transfer does not connect |

Save the transfer tool, then reference it by name in the agent's prompt where it describes when to transfer:

```text theme={}
When transferring the caller to the Support team, use the transfer_to_support tool.
```

## Test before going live

1. The 11x team first tests each direction with a SIP softphone standing in for your phone system.
2. **Inbound:** call through your IVR and confirm the 11x agent answers and the call record shows the caller's number.
3. **Outbound:** ask the agent for a person and confirm a rep in your queue answers, with the expected caller ID.
4. Test a failed transfer (for example, with no reps available) and confirm the caller is handled as expected.

> [!NOTE]
>
> Use a real phone or SIP client for test calls. The **Test** button in 11x starts a browser call, which cannot place a
> SIP transfer.


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