ht488 und FXO caller ID

Step 1:
Add a SIP trunk with these details:

Maximum Channels: 1

Trunk Name: HT-488
<or whatever device name you want, remember what you enter, it will be needed in Steps 2, 5 & 6>

PEER Details:
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=dynamic
secret=<the password you want the HT-488 to use to authenticate to your Asterisk Server>
type=friend

Leave "USER Context", "USER Details", and "Register String" all blank

Step 2:
Add a CUSTOM Trunk with these details:

Maximum Channels: 1
Custom Dial String: SIP/HT-488,,D(w$OUTNUM$)
<replace 'HT-488' with device name you used in Step 1 if you changed device name in Step 1>

Step 3: <Fixes a bug in FreePBX, may not be necessary, test first without this step>
Modify your extensions.conf as follows:
In the macro-dialout-trunk section, look for the following lines:
exten => s,21,Dial(${pre_num:4}${the_num}${post_num})
exten => s,22,Goto(s-${DIALSTATUS},1)

Replace it with the following lines:
exten => s,21,Set(full_dialstring=${pre_num:4}${the_num}${post_num})
exten => s,22,Set(DialArg1=${CUT(full_dialstring,\,,1)})
exten => s,23,Set(DialArg2=${CUT(full_dialstring,\,,2)})
exten => s,24,Set(DialArg3=${CUT(full_dialstring,\,,3)})
exten => s,25,Dial(${DialArg1},${DialArg2},${DialArg3})
exten => s,26,Goto(s-${DIALSTATUS},1)

Step 4:
In your HT-488's web based configuration, go to the 'BASIC SETTINGS' Section
Set number of rings to answer PSTN line from '4' to '1'
Set "Forward to VoIP" to : s

Step 5:
In your HT-488's web based configuration, go to the 'FXO Port' Section
Enter the login details that you specified in 'PEER Details' in Step 1
Be sure to set your User ID and Auth ID to be 'HT-488'
<or device name you used in Step 1 if you changed device name in Step 1>

Step 6:
Use: "SIP/HT-488,,D(w$OUTNUM$)" as your trunk when setting up your outgoing routes.
<replace 'HT-488' with device name you used in Step 1 if you changed device name in Step 1>