Connect your PBX to Cloud SIP

Is there a guide for your phone system?

These systems have their own guide, with every field named the way that system names it:

• Connect 3CX to Cloud SIP
• Connect Microsoft Teams Direct Routing to Enfonica
• Connect Yeastar to Cloud SIP
• Connect FreePBX, FusionPBX, or Asterisk to Cloud SIP
• Connect an AI voice agent to Cloud SIP

For any other SIP PBX (phone system) or SBC (session border controller), use this guide.

How Cloud SIP connects to your PBX

Cloud SIP is a SIP trunk without registration. Your PBX sends outgoing calls to your SIP domain. Cloud SIP recognizes them by the domain in each call, your PBX's public IP address, and, if you add one, a username and password.

Because your PBX doesn't register, Cloud SIP can't learn where it is. You tell it once, with a load balancer. Incoming calls to your numbers then go to that address.

Create your SIP domain

A SIP domain is your trunk. It's ready instantly when you create it, with its own name in the format <name>.sip.enfonica.com.

1. Sign in to the Enfonica Console, and open your project.
2. Go to Cloud SIP, and select New SIP domain.
3. For Type, select Standard.
4. Enter a SIP domain ID, such as acme, and complete the steps to create the SIP domain.
5. Under Termination settings, set Call termination mode to PSTN Trunk, so that outgoing calls go to the phone network.

Secure your SIP domain

Choose how Cloud SIP knows that a call really comes from your PBX. You can use either method, or both:

• IP access control list. Under IP access control list, select Add IP range, and enter your PBX's public IP address in CIDR notation, for example 203.0.113.10/32. Only devices in the networks you list can make calls.
• Users. Under Users, select Add user, and create a username and password. Every call must then authenticate with them.

Changes take up to a minute to apply after you select Save Changes.

Using both is the most secure setup. A call must then come from a listed network and also carry the password. If the system that sends calls is a cloud platform whose IP addresses change, use a user only. A call from an unlisted address is refused before its password is checked.

Allow Enfonica through your firewall

Your firewall must accept call signaling and call audio from Enfonica's Sydney edge. They come from different addresses, so allow all four addresses in both directions.

If you use UDP, also allow fragmented UDP packets, if your firewall can. Long SIP messages are split into fragments, and only the first fragment carries the port number. A port-based rule can drop the rest.

Many PBXs also block addresses automatically when they see a lot of SIP traffic or failed authentication. Add both signaling addresses to your PBX's allow list or whitelist, so that they're never blocked.

Point your PBX's trunk at Cloud SIP

In your PBX, add a SIP trunk with these settings. Some systems call it a SIP provider, peer, or gateway. The caller ID step matters most, because an invalid caller ID is the most common reason that a new trunk's calls are refused.

1. Set the host, server, or registrar to your SIP domain, for example acme.sip.enfonica.com. Use the name rather than an IP address. Leave any outbound proxy setting empty.
2. Use UDP on port 5060, or TLS on port 5061. Plain TCP isn't supported. UDP is the simplest place to start.
3. Turn registration off. If your PBX asks, choose a trunk type that makes calls without registering.
4. If you created a user, enter its username and password as the trunk's authentication. Enter the username on its own, such as pbx1, not pbx1@acme.sip.enfonica.com.
5. Allow G.711 A-law (PCMA), G.711 u-law (PCMU), or G.729, the codecs that Cloud SIP supports. Leave wideband codecs, such as Opus or G.722, off the trunk. They only make the PBX convert the audio of every call.
6. Set each extension's outbound caller ID, or the trunk's default caller ID, to a number in your Enfonica project in +E.164 (full international format with a plus), for example +61255501234. A name, a username, or a number missing its area code is refused with 484 Address Incomplete.
7. Send the numbers your users dial in +E.164, for example +61491570156. National format, such as 0491570156, also works, because Cloud SIP converts it using Australia as your home country. Never send the 0011 international prefix.

For every format that Cloud SIP accepts, with examples, including 13, 1300, and 1800 numbers, see Which number format should my PBX use?

Send incoming calls to your PBX

Create a load balancer that knows your PBX's address, 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 hostname or IPv4 address, for example pbx.example.com. IPv6 addresses aren't supported. Calls use UDP on port 5060 unless you add a port. For TLS on port 5061, add ;transport=tls.
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, listed as pbx@acme.sip.enfonica.com.
9. Select Save Changes.
10. Repeat steps 6 to 9 for each number.
11. In your PBX, add each number as a DID (direct inward dialing number) or inbound route in +E.164, for example +61255501234, exactly as Cloud SIP sends it.

If your PBX can't accept a plus sign, set the number that Cloud SIP sends instead. For the number's handler, select SIP URI rather than Load balancer, and add a to header, for example sip:pbx@acme.sip.enfonica.com?to=0255501234.

To share calls across several PBXs or sites, with failover, see Send incoming calls to more than one PBX or site.

Check that it worked

1. Call a mobile from an extension on your PBX. The mobile rings and shows the caller ID your PBX sent.
2. Call one of your Enfonica numbers from a mobile. It rings on your PBX, and you can hear each other both ways.
3. 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 learn what each code means, see Outbound calls aren't going through. For incoming calls, see Calls aren't reaching your PBX.

Questions