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

# PBX & contact center platforms

> Connect your own 3CX, Asterisk, Starface, Genesys Cloud, Five9, or Aircall deployment as a SIP trunk

If your organization runs its own PBX or a contact-center platform, you can connect it the same way you'd connect any SIP trunk — no number porting required. Start with [SIP provider guides](/telephony/providers/overview) for the shared mechanism (the platform SIP FQDN, the exact DID, and inbound authentication); this page covers what's specific to six PBX and contact-center platforms, including a few things that look broken but are expected.

| Platform           | Access requirement                                | Watch for                                                   |
| ------------------ | ------------------------------------------------- | ----------------------------------------------------------- |
| 3CX                | StartUP PRO, or a self-managed/Dedicated instance | Inbound needs a two-part routing rule, not just the trunk   |
| Asterisk / FreePBX | Self-hosted                                       | NAT settings; calls dropping at exactly 30 seconds          |
| Starface           | Any                                               | Transfers must be SIP transfers, never PSTN forwarding      |
| Genesys Cloud      | BYOC Carrier                                      | Non-standard ports 32681/32682; never allow all SIP sources |
| Five9              | Enterprise, via a support ticket                  | 12-hour hard call cap; SIP REFER only                       |
| Aircall            | Enterprise plan                                   | Inbound only — no outbound caller ID over SIP               |

## 3CX

3CX's **Generic SIP Trunk (IP Based)** connection type is only available on **StartUP PRO** and self-managed/Dedicated instances — the entry StartUP tier doesn't expose it.

1. Create a Generic SIP Trunk in 3CX with IP-based authentication, not SIP registration.
2. Point it at the platform SIP FQDN, set the trunk transport to TCP (UDP fragments large INVITEs), and restrict codecs to PCMA/PCMU with RFC2833 DTMF — disable Opus and G.722.
3. Set the trunk's **From: Display-Name** and **Remote-Party-ID** to `OriginatorCallerID` so the assistant sees the real caller, not 3CX's own system number.
4. Configure inbound in two parts: an outbound rule with a distinct prefix (for example `999`) routed to the trunk, and an inbound rule that dials `999<number>` back through that same outbound rule.
5. Assign the DID on the platform side and test.

<Frame caption="Set the trunk's transport protocol to TCP">
  <img src="https://mintcdn.com/ouraicall/YHkeF3SwYhU0rNTj/images/famulor_3cx_tcp_protocol_en.png?fit=max&auto=format&n=YHkeF3SwYhU0rNTj&q=85&s=ddd1b913ef77717ec02a5a4987a4c146" alt="3CX trunk Options tab with Transport Protocol set to TCP and SRTP Mode disabled" width="1024" height="1024" data-path="images/famulor_3cx_tcp_protocol_en.png" />
</Frame>

<Frame caption="Pass through the real caller ID with OriginatorCallerID">
  <img src="https://mintcdn.com/ouraicall/YHkeF3SwYhU0rNTj/images/famulor_3cx_caller_id_settings_en.png?fit=max&auto=format&n=YHkeF3SwYhU0rNTj&q=85&s=8433ecd3c782421ec1d9d3e28ce37334" alt="3CX trunk Caller ID Options with From: Display Name and Remote Party ID - Calling Party: Display Name both set to OriginatorCallerID" width="1024" height="1024" data-path="images/famulor_3cx_caller_id_settings_en.png" />
</Frame>

<Warning>
  The outbound rule alone delivers nothing on its own. The pairing in step 4 — an outbound rule plus a matching inbound rule dialing through it — is what actually routes an inbound call to the assistant. Skipping it is the most common cause of "trunk configured, calls never arrive."
</Warning>

## Asterisk / FreePBX

Configure a static, non-registering trunk in both directions — Asterisk never sends the platform a SIP REGISTER, and the platform never expects one.

### FreePBX (GUI)

Add a generic SIP trunk pointed at the platform SIP FQDN and disable registration. Restrict codecs to PCMA/PCMU. If Asterisk sits behind NAT, set the external media and signaling addresses and the local network range so RTP negotiates correctly, and forward the RTP port range at the firewall.

### Raw config (pjsip.conf)

```text theme={null}
[platform]
type=endpoint
context=from-platform
disallow=all
allow=ulaw
allow=alaw
aors=platform
auth=platform-auth
outbound_auth=platform-auth
from_domain=<platform-sip-fqdn>
from_user=<did-in-e164-without-plus>
direct_media=no
rewrite_contact=yes

[platform]
type=aor
contact=sip:<platform-sip-fqdn>

[platform-auth]
type=auth
auth_type=userpass
username=<username>
password=<password>

[platform]
type=identify
endpoint=platform
match=<platform-sip-fqdn>
```

The sample uses whatever transport your Asterisk install already defines. If you name your transports explicitly, add a matching `transport=` line to the endpoint — the platform SIP URI advertises TCP by default.

If calls connect but drop at exactly 30 seconds, the endpoint is missing a re-INVITE across NAT. `direct_media=no` and `rewrite_contact=yes` (already in the sample above) fix it.

## Starface

Connect Starface with a host/IP-based SIP line, not SIP REGISTER — the line will show **not registered** in Starface's status view afterward. That's expected, not a fault.

1. Create the SIP line as host/IP-based, pointed at the platform SIP FQDN, and assign the number.
2. Enable **CLIP No Screening** so the real caller ID passes through instead of the trunk's own number.
3. For time-based routing to the assistant, use a line-prefix dial string such as `**<line-digit>*<number>`.
4. If audio is one-way or missing, set **Behind NAT? yes** on both the Starface server and the line.

<Warning>
  Never use ordinary PSTN call forwarding from the assistant to transfer a call on Starface — it creates a call loop between Starface and the platform. Configure every assistant transfer on a Starface trunk as a SIP transfer instead.
</Warning>

## Genesys Cloud

Connect the platform as an external trunk under **BYOC Carrier**.

1. Create the trunk with inbound (the FQDN Genesys Cloud generates) and outbound (pointed at the platform SIP FQDN) configured as separate legs.
2. Open the non-standard ports at your firewall — **TCP/UDP 32681** and **TLS 32682** — instead of the usual 5060/5061. A firewall that only opens the standard ports fails silently.
3. Order codecs PCMU then PCMA, and remove G.722.
4. Restrict SIP Access Control to only the platform's current signaling addresses.

<Warning>
  Never set SIP Access Control to **Allow All**. An open trunk is typically found and used for toll fraud within hours.
</Warning>

## Five9

Provisioning the platform as an external SIP destination on a Five9 SBC is an enterprise-only integration.

1. Open a support ticket with Five9 — there's no self-service path.
2. Set codecs to G.711 only (no Opus/G.722).
3. Configure human transfers with SIP REFER — Five9 doesn't honor SIP 302 redirects.
4. Target a fully-qualified URI for every REFER (`sip:queue@sbc-{region}.five9.com`); a bare extension is silently dropped.

<Warning>
  Five9 hard-drops any call at exactly 12 hours, regardless of activity. Account for this ceiling in long-running flows and unattended campaign calls — there is no way to extend it from the platform side.
</Warning>

## Aircall

Aircall has no wholesale SIP trunk. **SIP Forwarding** (Enterprise plan only) covers inbound only — it delivers calls to the platform, but Aircall has no SIP path for sending a call back out with an Aircall number as caller ID. To use one number in both directions, port it to a SIP-capable carrier for outbound (see [SIP provider guides](/telephony/providers/overview)) while keeping Aircall for inbound. Aircall also doesn't support SIP REFER — a transfer should place a fresh outbound call rather than attempt a cold SIP transfer.

## Validate

Call the DID from an external phone and confirm a new inbound record in **History**, the assigned assistant, and two-way audio. For platforms with a distinct outbound leg (3CX, Asterisk, Genesys Cloud), place one outbound test call too — an inbound success doesn't confirm outbound is configured correctly, and vice versa.

See also [SIP provider guides](/telephony/providers/overview), [BYO SIP trunk](/telephony/sip-trunks), and [Other and custom SIP providers](/telephony/providers/other-custom).

Sources: [3CX SIP trunks](https://www.3cx.com/docs/manual/sip-trunks/), [Asterisk documentation](https://docs.asterisk.org/), [Starface knowledge base](https://knowledge.starface.de/), [Genesys BYOC Cloud](https://help.mypurecloud.com/articles/about-byoc-cloud/), [Five9 documentation](https://documentation.five9.com/), [Aircall support](https://support.aircall.io/). Accessed 22 August 2026.