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

# BYO SIP trunk

> Connect your own SIP provider and use your existing numbers

Already have numbers and a SIP provider (Twilio, Telnyx, Plivo, Easybell, or any standards-compliant trunk)? Connect them directly without porting anything.

For provider-specific FQDN direction, inbound authentication, and signaling allowlists, open the [SIP provider guides](/telephony/providers/overview).

## Setup checklist

1. **Copy the platform SIP URI** from **Settings → Numbers → Add a number → Add SIP integration** (or Carrier import). Pick a **SIP region** (default: global) so inbound signaling terminates where you need it (e.g. EU).
2. At your provider, set that URI as the **origination / forwarding destination** for your numbers.
3. Create the trunk in the platform (form below) — inbound auth + outbound termination.
4. Assign numbers to assistants and test inbound, then one outbound test call.

## Trunk type

| Type                   | When to use                                                     | Numbers                                                                                                                                                                                                                         |
| ---------------------- | --------------------------------------------------------------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| **Phone number (DID)** | Most carriers (recommended)                                     | One E.164 DID on the trunk; listed under **Numbers**                                                                                                                                                                            |
| **SIP Extension**      | Carrier requires national or extension-style outbound caller ID | One explicit **Primary DID** is required for inbound matching — full E.164 (`+…`) **or** national digits without country code (e.g. `0741926265`). This is not a wildcard; national-only primaries are not listed under Numbers |

## Inbound (receiving calls)

* **Your phone number (DID)** — public E.164 customers dial; must match what the carrier forwards to the platform SIP URI.
* **Inbound authentication**
  * **SIP username/password** — only when the carrier explicitly sends digest credentials to the forwarding destination.
  * **Provider source IPs** — use when the carrier forwards to the platform FQDN without downstream digest and publishes stable SIP-signaling IPs/CIDRs. Do not use media ranges or overly wide networks.
* At the provider: forward / originate to the platform SIP URI you copied.

Incoming calls are matched to the assistant you assign, exactly like marketplace numbers.

## Outbound (calling out)

* **Termination address** — provider SIP host only (e.g. `sip.telnyx.com`). No `sip:` prefix, no port.
* **Transport** — `AUTO` (recommended), `UDP`, `TCP`, or `TLS`. Secure trunking always uses TLS.
* **Outbound region** — where the platform originates the call. Prefer **Automatic**, or the country closest to your customers / carrier POP.
* **Outbound calling number format** — how the FROM number is sent to the carrier. Must match the carrier’s setting (e.g. Telnyx Origination Number Format):
  * International with `+` (recommended for most)
  * International without `+`
  * National (no country code)
* **Credentials**
  * **Shared** (recommended) — one username/password for inbound and outbound.
  * **Separate** — distinct inbound vs outbound secrets when the carrier requires it.
* **Outbound authentication** — username/password (recommended). The platform has **no static outbound IPs**, so carrier IP allowlists usually fail. Use “no credentials” only if the carrier explicitly allows unauthenticated outbound.

Outbound calls and [campaigns](/campaigns/overview) can then use your trunk and your caller IDs.

## Advanced

Expand **Advanced** when creating or editing a BYO trunk:

| Setting                               | Meaning                                                           | Recommendation                                                                                               |
| ------------------------------------- | ----------------------------------------------------------------- | ------------------------------------------------------------------------------------------------------------ |
| **Secure trunking (TLS + SRTP)**      | Encrypted signaling + media                                       | Enable when the carrier supports Secure Trunking / TLS+SRTP                                                  |
| **Media encryption**                  | Allow (prefer SRTP) or Require SRTP                               | Allow unless compliance requires Require                                                                     |
| **Custom outbound headers**           | X-\* on outbound INVITEs (max 10)                                 | Empty unless carrier docs require a header                                                                   |
| **Inbound headers (200 OK)**          | Custom X-\* in the SIP 200 OK                                     | Usually empty                                                                                                |
| **Header → attribute maps**           | Make selected SIP header values available to assistants and tools | Only if you need carrier metadata                                                                            |
| **Include SIP headers as attributes** | None / all X-\* / all headers                                     | **None** (map only what you need)                                                                            |
| **Media codecs**                      | Extra SDP codecs (PCMU, PCMA, G.722, AMR-WB)                      | Leave platform defaults; add AMR-WB only if required. Avoid “Only listed codecs” unless you know peer codecs |
| **Media / ringing timeouts**          | Optional overrides (seconds)                                      | Leave empty (platform defaults) unless troubleshooting                                                       |

**HD Voice (G.722):** enable on the **Telnyx** connection/codec settings when needed (supported with Telnyx, not Twilio).

## Editing an existing trunk

Existing trunks are edited from **Numbers → Configure → Carrier Settings** (not from a separate trunk list on Add SIP integration).

## EU routing

For EU customers, select the **EU SIP region** when creating the connection. Consult your agreement and data-processing documentation for the applicable regional commitments.

## Limits and behavior

* E.164 numbers on BYO trunks count against the same plan number allowance as purchased numbers.
* Calls over your own trunk avoid the platform's per-destination carrier surcharge — your provider bills termination directly. Plan minutes are still consumed; see [billing](/billing/minutes).
* Assistant features such as flows, warm transfer, and recording work identically on BYO trunks.

## Troubleshooting

**Outbound calls fail or don't connect** — check the termination host, transport, and credentials against your provider's documentation first: a wrong transport (UDP vs. TLS), or an extra `sip:` prefix or port on the termination address, is the most common cause. Then compare the outbound calling number format and the codec settings with what the provider expects. Change one setting at a time and place a single test call after each, so **History** shows you which change fixed it.

**Inbound calls don't reach the assistant** — the provider must send calls to the platform's SIP **FQDN**, never to a raw IP address; sending to an IP is the most common inbound failure. With Provider source IPs, confirm every current signaling IP or CIDR is entered — a stale or incomplete list drops calls silently. Then confirm the DID matches exactly what the provider sends: a SIP Extension trunk also needs its Primary DID set, since it isn't a wildcard.

**Call transfers over SIP REFER fail** — confirm the destination provider actually supports SIP REFER; not every carrier does, and that's the most common cause. If one URI format fails, try the alternatives in order: with the port (`sip:+1234567890@sip-server:5060`), without the port, then a bare `sip:+1234567890`. Rule out the destination itself as well — check that the target number is reachable and not blocked before assuming the trunk is at fault. When a transfer was attempted and failed, the call's event log holds `call_transfer_failed` or `warm_transfer_failed`; see [Inbound & outbound calls](/telephony/inbound-outbound#failure-guidance) for the shared failure shape.

Test inbound and outbound independently — one working does not confirm the other. Still stuck? Contact support with the call ID, the exact trunk configuration, and, for transfer issues, the SIP URI format you tried.

## API and MCP

Create and manage trunks via the Public API (`POST /api/v1/sip-trunks` and related endpoints in the API reference), or with the MCP tools `create_sip_trunk`, `list_sip_trunks`, `get_sip_trunk`, and `delete_sip_trunk`. Both surfaces support the same customer-facing settings as the UI, and passwords are never returned.

<Tip>
  Test inbound first: call one of your numbers and check it appears in **History** with the right assistant. Then verify outbound with a single test call before wiring the trunk into campaigns.
</Tip>