Configuring incoming DID to IVR menu

Asterisk Asterisk Support
1

I’m fairly new at configuring Asterisk. Running Asterisk 1.4 with FreePBX (installed AsteriskNOW).

I am trying to setup Asterisk to be used as an IVR application only (no internal extensions, no sip phones, etc…). Just an IVR menu that will utilize AGI() to access scripts on the server that communicate with a local MySQL database. My problem is that I cannot correctly configure an inbound route to Asterisk. I have a DID pointing to the box at port 5060. The box is not behind a NAT, and has port 5060 open.

Here is what my sip.config file looks like:

[general]
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
context=from-sip-external
callerid=Unknown
tos=0x68

And my extensions.conf looks like this:

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[from-sip-external]
include => ext-did
include => from-pstn
exten => _X.,1,Answer()
exten => _X.,2,Wait(1)
exten => _X.,3,Goto(default,s,1)
exten => 1888XXXXXXX,1,Answer() ;I’ve X’d out the real number
exten => 1888XXXXXXX,2,Wait(1)
exten => 1888XXXXXXX,3,Goto(default,s,1)

[default] ;the simple IVR menu just for testing
exten => s,1,Wait(1)
exten => s,2,BackGround(dir-intro-oper)
exten => s,3,Wait(1)
exten => s,4,HangUp()

All I want to do is set this up so when someone calls the DID (1888XXXXXXX), or many other DIDs in the future, Asterisk answers it and forwards it to the [default] context which basically is the IVR menu. I am told from my phone provider that the DID is pointing to the box correctly, and no other configuration needs to be done outside. No authentication is necessary either.

What am I missing? Any help is greatly appreciated, thank you!

2

Presumably, this just isnt working for you. You didn’t actually say whats wrong.
It would be a great help if you could post the sip debug messages so we can see what’s causing the calls to fail, or even, if they are reaching your server in the first place.

Where are the calls orginating from? A voip provider, or a isdn/pstn gateway within your network?
If its a voip provider running asterisk, with your set up below, this will only work if your box has a public static ip address.
Check that your “DID” is set to call your server follows “SIP/${EXTEN}@your-ip-address:5060”

Give me some more info on your set up please and pelase post the SIP debug for a call. :smiley: