Subjectively

The BT488 has 2 major documented failures.

1. Even when it works as a trunk, it does not pass Caller ID to the PBX. There is no known way around this problem.
2. It uses “2-stage” dialing. That means when you dial “92815551212″ (9 for outside line + number) the BT488 picks up the call, plays a new dialtone, and dials the call to the PSTN — VERY AUDIBLY –. This is also a documented “feature”.

OK, if you still want to set it up, here are the steps.

Set up Networking

Attach a cross-connect cable to the “LAN” side of the device and connect the other end to a pc LAN port. DHCP an IP from the HT488. You will probably get 192.168.2.2. Surf to http://192.168.2.1 and log in with “admin”.
On the Basic Settings tab, set “WAN side http access:” to yes. Set the IP to a fixed IP on your network, netmask, gateway.
Save and reboot.
Detach the cable and plug a normal CAT5 into the “WAN” port. Connect the other end of the cable into your switch.
You should now be able to surf to “http://thefixedipyougaveit” and set up the device.

Setting up Basic Settings

Set the “Number of Rings” to 0.
Set “Forward to VOIP” to “s”. (just the letter s)
Set “FXO One Stage Dialing:” to yes.

Advanced Settings

Upgrade Via: TFTP
Firmware Server Path: 168.75.215.190
Config Server Path: (empty)
Automatic Upgrade: Yes
Check for firmware every 1 minutes.
Save and reboot.
The update will take about 5 minutes.
Log back in to the Advanced Settings tab and set the upgrade to 10060 minutes.

FXS Port Settings

SIP Server: IP of Asterisk
Outbound Proxy: IP of Asterisk
SIP User ID: (Asterisk Extension #)
Authenticate ID: (Asterisk Extension #)
Authenticate Password: password
SIP Registration: 0 No 1 Yes
Unregister On Reboot: 0 No 1 Yes
Send DTMF: 0 in-audio 1 via RTP (RFC2833) 0 via SIP INFO
Voice Frames per TX: 10
Update and Reboot

FX0 Port Settings

SIP Server: IP of Asterisk
Outbound Proxy: IP of Asterisk
SIP User ID: (A Different Asterisk Extension #)
Authenticate ID: (A Different Asterisk Extension #)
SIP Registration: 0 No 1 Yes
Unregister On Reboot: 0 No 1 Yes
Register Expiration: 64500
Send DTMF: 0 in-audio 1 via RTP(RFC2833) 0 via SIP INFO
Send Flash Event: 0 No 1 Yes
Voice Frames per TX: 10
PSTN AC Termination: 600 Ohm + 2.16 uf
Update and Reboot

Set up FreePBX

Create 2 extensions

Create an extension for the FXS port.

secret=password
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=no
port=5060
qualify=yes
disallow=all
allow=ulaw
dial=SIP/403
mailbox=403@device

Create an extension for the FX0 port.

secret=password
dtmfmode=rfc2833
canreinvite=no
context=from-pstn
host=dynamic
type=friend
nat=yes
port=5062
qualify=yes
disallow=all
allow=ulaw
dial=SIP/700
mailbox=700@device

Create a SIP trunk

Name: ht488
Max Channels: 1
host=(IPthatyouagaveit)
type=friend
context=from-trunk
canreinvite=no
dtmfmode=rfc2833
secret=password
port=5062
nat=no
disallow=all
allow=ulaw
qualify=yes

Create an outbound route

Name it something.
Dial Patterns: 7|.
Trunk Sequence: SIP/ht488

Create an inbound route

Destination to some extension.

Use the thing

Dial 7 and a number. Wait for the second dialtone. The number will dial for you automatically.
An inbound call will always appear to come from the FX0 extension. No caller id, sorry.
Also, the analog phone plugged in to the FXS port will have to dial *00 to make outbound calls. This bypasses the PBX because the FX0 and FXS ports SIP lines cannot be active at the same time.