Connect FreePBX, FusionPBX, or Asterisk to Cloud SIP

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, 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.

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.

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.

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.

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. To learn what each code means, see Outbound calls aren't going through. When numbers fail to dial, the format is often the cause. For every format, see Which number format should my PBX use?

Questions