In FreePBX web portal, Add a trunk,
[didforsale]
type=peer
host=[IP ADDRESS OF OUR SERVER]
nat=no
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
insecure=very
context=from-didforsale
Please note: You can find the IP for DIDForSale upon logging into your account. We send the calls to your public IP address and do not require any username, password or registration to our server.
In extension.conf, add these lines. //This will remove + from the callerid, which many people can not handle.
[from-didforsale]
exten => _X.,1,Set(CALLERID(num)=${CALLERID(num):1})
include => from-trunk
Now your system will be able to accept calls from didforsale.com. Now you can do anything you want with the DIDs.
Hope this helps, feel free to comment in this if you want to see any additional information of help.
Thank you,
-Jai
My Settings for Freepbx and it’s working
Login to freePBX administrative interface
Click on Setup
in top right of page
Click on Trunks
in left side navigation
Click Add SIP Trunk
in middle of page
Scroll to Outgoing
Settings and enter didforsale_did
into Trunk Name field
Copy and paste the following into the PEER Details field:
Peer Details:
type=peer
host=209.216.2.211
nat=no
canrinvite=no
disallow=all
allow=ulaw&alaw
dtmfmode=rfc2833
insecure=very
qualify=yes
Click on Submit Changes to add your new SIP trunk to your Asterisk server
Click on the red bar at the top of the screen to apply the changes you just made
in extensions.conf add thease lines
[custom-a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php|1)
exten => _X.,n,Hangup
in FreePBX UI create a Custom Destination;
custom-a2billing,${EXTEN},1
Description: A2Billing
Notes: For a call through service
Click on Inbound Routes
Click on Add Incoming
Route.
You will first want to fill the
DID Number field with your
DID
Make sure to leave the Caller ID Number and
Zaptel channel blank in order to match any
incoming call.
This is useful if you wish to receive all calls
Scroll down to Set Destination
Choose A2Billing Custom Destinations
From Drop Down Menu
Click on Submit Changes to add your new
inbound route to your Asterisk server
Click on the red bar at the top of the screen to apply the changes you
just made
Do NOT edit extensions.conf when using FreePBX
edit extensions_override_freepbx.conf instead
(at least this is what I had to do in Trixbox CE)
Hi, I followed the settings above and my asterisk box says the status is OK when I do a “sip show peers” from the CLI. However when I dial my did I see nothing in the CLI/logs and I just get silence. Any help would be appreciated.. Thanks