---
title: "Connect FreePBX, FusionPBX, or Asterisk to Cloud SIP"
url: "https://enfonica.com/help/sip-trunking/connect-freepbx"
summary: "Add your SIP domain as a trunk or gateway that doesn't register, accept calls from both Enfonica signaling addresses, and use +E.164 numbers for caller IDs and inbound routes. Then point your phone numbers at the PBX through a load balancer in the Enfonica Console."
type: how-to
audience: everyone
products: ["Cloud SIP", "Phone Numbers"]
works_with: ["FreePBX", "FusionPBX", "Asterisk with chan_pjsip"]
who_can_do_this: "Whoever manages your PBX and your Enfonica project"
---

# Connect FreePBX, FusionPBX, or Asterisk to Cloud SIP

**Short answer:** Add your SIP domain as a trunk or gateway that doesn't register, accept calls from both Enfonica signaling addresses, and use +E.164 numbers for caller IDs and inbound routes. Then point your phone numbers at the PBX through a load balancer in the Enfonica Console.

**Applies to:** Cloud SIP, Phone Numbers  
**Works with:** FreePBX, FusionPBX, Asterisk with chan_pjsip  
**Who can do this:** Whoever manages your PBX and your Enfonica project

## Before you start

- An Enfonica project with the phone numbers you want the PBX to use.
- Admin access to your PBX, and to its configuration files if you run Asterisk directly.
- Your PBX's public IP address, or a dynamic DNS name if the address changes.
- Access to the firewall in front of the PBX, or the person who manages it.

## How these PBXs connect to Cloud SIP

FreePBX is built on Asterisk, and FusionPBX is built on FreeSWITCH. All three connect to Cloud SIP the same way. Your PBX (private branch exchange, your phone system) sends outgoing calls to your SIP domain, which is your SIP trunk on Enfonica. Cloud SIP sends incoming calls to the address you give it in the Console.

The PBX doesn't register to Cloud SIP. Cloud SIP recognizes your outgoing calls by the SIP domain in each call, plus your PBX's public IP address, a username and password, or both. A trunk or gateway set to register gets `405 Method not allowed`.

This guide gives the value each setting needs. FreePBX and FusionPBX lay out their settings differently between releases, so match each value to the field that sets it on your system.

## Create your SIP domain

1. Sign in to the [Enfonica Console](https://console.enfonica.com/), and go to **Cloud SIP**.
2. Select **New SIP domain**, and create a **Standard** SIP domain, such as `acme-pbx`. It's ready instantly, with a name like `acme-pbx.sip.enfonica.com`.
3. Under **Termination settings**, set **Call termination mode** to **PSTN Trunk**, so that outgoing calls go to the phone network.
4. Choose how Cloud SIP knows that a call is yours:

   - If your PBX has a static public IP address, under **IP access control list**, select **Add IP range**, and enter the address with `/32` on the end, for example `203.0.113.10/32`.
   - If its address changes, leave the IP list empty. Under **Users**, select **Add user**, and create a username and password instead.
   - You can also use both. A call must then come from a listed address and carry the password.
5. Select **Save Changes**. Changes take up to a minute to apply.

## Asterisk and FreePBX (chan\_pjsip)

Use a chan\_pjsip trunk with no registration. On Asterisk, this `pjsip.conf` example has everything the trunk needs. Replace `acme-pbx` with your SIP domain. If you authenticate by IP address only, leave out `outbound_auth` and the `[enfonica-auth]` section.

```
[enfonica]
type=endpoint
transport=transport-udp
context=from-enfonica
disallow=all
allow=alaw,ulaw
outbound_auth=enfonica-auth
aors=enfonica
from_domain=acme-pbx.sip.enfonica.com

[enfonica-auth]
type=auth
auth_type=userpass
username=your-username
password=your-password

[enfonica]
type=aor
contact=sip:acme-pbx.sip.enfonica.com
qualify_frequency=60

[enfonica-identify]
type=identify
endpoint=enfonica
match=34.87.239.197,34.87.254.143
```

Make sure that the `transport-udp` transport and the `from-enfonica` context exist elsewhere in your configuration. Incoming calls from Enfonica arrive in that context.

In FreePBX, you set the same options on the chan\_pjsip trunk's screens. Field names vary between releases, so match each value in this table to the field that sets it.

| What the setting does | Value | pjsip option |
| --- | --- | --- |
| Where outgoing calls go | Your SIP domain, for example `acme-pbx.sip.enfonica.com`, port 5060 | `contact` |
| The domain in the From header | Your SIP domain | `from_domain` |
| Registration | None | No `registration` section |
| Authentication | Outbound only, with your SIP user's username and password, or none for IP-only | `outbound_auth` |
| Which addresses incoming calls come from | `34.87.239.197` and `34.87.254.143` | `match` |
| Codecs | G.711 A-law and u-law (G.729 also works) | `allow` |

Next, set up the calls themselves:

- **Outgoing calls**. Send them to the trunk, for example with `Dial(PJSIP/${EXTEN}@enfonica)` in Asterisk, or with an outbound route that uses the trunk in FreePBX.
- **Caller ID**. Set every outgoing call's caller ID to a number you hold, in +E.164 (full international format with a plus), for example `+61255501234`. In Asterisk, add `Set(CALLERID(num)=+61255501234)` before the `Dial`.
- **Incoming calls**. Cloud SIP sends numbers in +E.164, so create an inbound route for each number in that format, such as `+61255501234`.

## FusionPBX

FusionPBX connects through a gateway for outgoing calls and an access control list for incoming calls.

1. Add a gateway, with your SIP domain as the address it sends calls to, for example `acme-pbx.sip.enfonica.com`. If you created a SIP user, enter its username and password.
2. Set the gateway's **Register** to **False**.
3. Go to **Advanced > Access Controls**, and open **Providers**.
4. Add a node with type **allow**, CIDR `34.87.239.197/32`, and the domain left empty.
5. Add a second node with type **allow**, CIDR `34.87.254.143/32`, and the domain left empty.
6. Select **Reload ACL**.
7. Create your inbound routes with each number in +E.164, for example `+61255501234`.
8. Set outbound caller IDs in the same format.

FusionPBX is multi-tenant, so one server can keep many customers apart, such as households on a residential voice service. FreePBX is single-tenant. On a shared FreePBX, you'd need to stop one customer's extensions from calling another's.

## Allow Enfonica through your firewall

Cloud SIP sends call signaling and call audio from different addresses. In the firewall in front of your PBX, allow all four addresses in both directions. Include any cloud provider firewall.

| Traffic | IP addresses | Ports |
| --- | --- | --- |
| Signaling | `34.87.239.197`<br>`34.87.254.143` | 5060 (UDP), 5061 (TLS) |
| Media (call audio) | `35.189.19.106`<br>`35.189.6.179` | All UDP ports |

Cloud SIP supports UDP and TLS, not TCP. Some SIP messages are large enough to be split into fragments, so allow UDP fragments as well. For more information, see [Firewall configuration for Cloud SIP](https://enfonica.com/docs/cloud-sip/firewall-configuration/).

## Send incoming calls to your PBX

Because the PBX doesn't register, you need to tell Cloud SIP where it is. Create a load balancer once, and then point each number at it.

1. In the Console, go to **Cloud SIP**, and open your SIP domain.
2. Under **Load balancers**, select **Add load balancer**, and enter an **Alias ID**, such as `pbx`.
3. Select **Add endpoint**.
4. In **Endpoint URI**, after the `sip:` that's already shown, enter your PBX's public IP address or hostname, for example `203.0.113.10` or `pbx.example.com`. If the address changes, use a dynamic DNS name.
5. Select **Add**, select **Save**, and then select **Save Changes**.
6. Go to **Phone Numbers > My numbers**, and open a number.
7. Under **Call settings**, in **Incoming call handlers**, select **Add**, and then select **SIP**.
8. Select **Load balancer**, and then select your load balancer.
9. Select **Save Changes**.

Repeat steps 6 to 9 for each number.

> **Warning:** If you delete a load balancer, numbers that pointed at it stop receiving calls. Move them to the new load balancer first.

## Check that it worked

1. On Asterisk, run `pjsip show aors`. The Enfonica contact shows as available, because Cloud SIP answers your PBX's keepalive checks.
2. Call a mobile from an extension. The mobile rings and shows your caller ID.
3. Call one of your Enfonica numbers from a mobile. It rings on your PBX, and you can hear each other both ways.
4. In the Console, go to **Cloud Voice > Calls**. Both test calls are listed there with their SIP traces.

If a test call fails, its SIP response code points to the cause. To find the code, see [Find a call and read its SIP trace](https://enfonica.com/help/call-troubleshooting/find-call-sip-trace). To learn what each code means, see [Outbound calls aren't going through](https://enfonica.com/help/call-troubleshooting/outbound-calls-fail). When numbers fail to dial, the format is often the cause. For every format, see [Which number format should my PBX use?](https://enfonica.com/help/sip-trunking/number-formats)

## Questions

### My PBX says the trunk can't register. What's wrong?

Nothing is wrong with the trunk. Cloud SIP doesn't accept registration, so turn registration off:

- On Asterisk, remove the `registration` section.
- In FreePBX, choose no registration on the trunk.
- On a FusionPBX gateway, set **Register** to **False**.

Calls still authenticate with your IP address or your username and password.

### Incoming calls never reach the PBX. Where do I look?

Open the call in **Cloud Voice > Calls**. If its SIP trace shows Cloud SIP sending the call with no reply from your PBX, a firewall in front of the PBX is almost always the cause. Make sure that both signaling addresses are allowed, including in your cloud provider's firewall. For other causes, see [Calls aren't reaching your PBX](https://enfonica.com/help/call-troubleshooting/inbound-calls-not-arriving).

### My PBX is on site behind NAT, with an address that changes. Will it work?

Yes, with two changes:

- Use a username and password for outgoing calls.
- Give the load balancer a dynamic DNS name that follows your address, and have your router forward SIP to the PBX.

It's simpler to host the PBX in the cloud with a static IP address, and connect your on-site handsets to it.

### Can I connect handsets or softphones directly to Cloud SIP?

No. Cloud SIP is a trunk for phone systems and doesn't accept registrations. Register handsets, softphones, and ATAs (analog telephone adapters) to your PBX as extensions. The PBX connects them to the phone network through Cloud SIP.

## What to send us

Include these and we can find the call straight away.

- The number that made the call and the number it called, in full international format (for example +61255501234).
- The date and time of the call, with your timezone.
- Whether the call was inbound or outbound.
- The call ID, if the Console shows the call.
- What happened: an error message, silence, a busy tone or a dropped call.

[Contact support](https://enfonica.com/contact-support?article=%2Fhelp%2Fsip-trunking%2Fconnect-freepbx&topic=call-fault)

## Related

- [Connect your PBX to Cloud SIP](https://enfonica.com/help/sip-trunking/connect-pbx-to-cloud-sip.md)
- [Which number format should my PBX use?](https://enfonica.com/help/sip-trunking/number-formats.md)
- [Outbound calls aren't going through](https://enfonica.com/help/call-troubleshooting/outbound-calls-fail.md)
- [Calls aren't reaching your PBX](https://enfonica.com/help/call-troubleshooting/inbound-calls-not-arriving.md)
- [SIP 405 Method Not Allowed](https://enfonica.com/help/sip-errors/405-method-not-allowed.md)
- [Firewall configuration for Cloud SIP (docs)](https://enfonica.com/docs/cloud-sip/firewall-configuration/)

---

Web version: https://enfonica.com/help/sip-trunking/connect-freepbx