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FreePBX and Asterisk

Send calls from your FreePBX or Asterisk phone system to your receptionist over a SIP trunk.

Your PBX keeps its carrier and numbers, and passes the calls you choose to Reception.ai over a SIP trunk. You add a second trunk on the PBX that points at Reception.ai, then route a DID, a time condition, or an IVR option to it.

Before you start

  • FreePBX 15 or later, or Asterisk with PJSIP, reachable from the internet.
  • A static public IP address for the PBX. Reception.ai only accepts calls from addresses you allow.
  • Someone comfortable with your PBX’s trunks, routes and firewall.

Setup

1

Add the SIP trunk in Reception.ai

Go to Integrations, select Add integration, and choose SIP Trunk. Under Inbound allowed addresses, add your PBX’s public IP address. If you also set an inbound username and password, use the same pair on the PBX trunk below.

2

Add the number

Go to Receptionists, select the Phone tile, enter the number callers dial, with its country code, and select Add SIP Trunk number. Reception.ai shows the SIP server host and the request URI to use.

3

Turn on TCP in FreePBX

Reception.ai doesn’t accept UDP. In FreePBX, go to Settings → Asterisk SIP Settings → SIP Settings [chan_pjsip], set the tcp transport to Yes, and select Submit and Apply Config. Restart Asterisk so the new transport loads. TLS also works if you’ve set up a certificate in Certificate Management.

4

Create the trunk to Reception.ai

Go to Connectivity → Trunks → Add Trunk → Add SIP (chan_pjsip) Trunk. On the General tab, enter a Trunk Name, such as reception. Set CID Options to Allow Any CID, so the receptionist sees each caller’s own number. On the pjsip Settings tab:

FieldValue
RegistrationNone
AuthenticationNone, or Outbound with Username and Secret if you set inbound credentials in Reception.ai
SIP ServerThe Reception.ai SIP server host
SIP Server Port5060 for TCP, or 5061 for TLS
TransportThe TCP (or TLS) transport, for example 0.0.0.0-tcp
5

Add an outbound route that sends +E.164

Go to Connectivity → Outbound Routes and add a route named Reception that uses only the reception trunk. Add a dial pattern with an unused prefix, such as 88, a match pattern of X., and a prepend of +. A call to 8814155550123 then leaves the trunk as +14155550123, which is the format Reception.ai requires.

6

Send the DID to Reception.ai

Go to Applications → Misc Destinations and add one that dials 88 followed by the number in E.164 format without +, for example 8814155550123. Then go to Connectivity → Inbound Routes, open the route for your DID, and set its destination to that misc destination. You can also use it as the destination of a time condition, ring group failover or IVR option.

7

Fix NAT and audio

Go to Settings → Asterisk SIP Settings → General SIP Settings. Set External Address to the PBX’s public IP and list your private subnets under Local Networks. Open the RTP Port Ranges shown on that page (UDP, 10000–20000 by default) in your firewall.

8

Test it

Call your DID from another phone and check that your receptionist answers.

Plain Asterisk

On Asterisk without FreePBX, the same setup in pjsip.conf looks like this. Replace SIP_SERVER with the host Reception.ai shows.

[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5060
external_signaling_address=PBX_PUBLIC_IP
external_media_address=PBX_PUBLIC_IP
local_net=192.168.0.0/16
[reception]
type=aor
contact=sip:SIP_SERVER:5060;transport=tcp
[reception]
type=endpoint
transport=transport-tcp
aors=reception
context=from-reception
disallow=all
allow=ulaw,alaw

In the dialplan, send the call with the number in E.164 format:

exten => YOUR_DID,1,Dial(PJSIP/+14155550123@reception)

If you set inbound credentials in Reception.ai, add a type=auth section with them and reference it with outbound_auth= on the endpoint.

Limitations

  • Transfers. Your receptionist transfers calls by placing a new call to the Outbound SIP address. Point it at your SIP carrier, using a carrier account that accepts a username and password. Routing transfers back into the PBX isn’t covered here: Reception.ai calls from a changing pool of addresses, so the PBX can’t match those calls to a trunk by IP.
  • Your PBX carries every call twice, once from the carrier and once to Reception.ai, so it needs a free channel for each leg.

Troubleshooting

ProblemWhat to check
Calls fail before ringingCheck that the trunk uses the TCP or TLS transport, not UDP, and that the port matches it.
Calls are rejectedAdd the PBX’s public IP to Inbound allowed addresses. Check in the Asterisk console that the INVITE’s request URI starts with + and the country code.
The receptionist sees the PBX’s number as the callerSet CID Options on the Reception.ai trunk to Allow Any CID.
One-way or no audioSet External Address and Local Networks, and open the RTP port range in the firewall.
The receptionist doesn’t answerMake sure the number is added on the receptionist’s Phone tile and matches the number the PBX sends.