Struggling with callcentric inbound

freepbx 2.5.2 and Asterisk 1.6.0.15 here, new setup trying to add Callcentric DID and coming up short.
I'm not sure exactly where it's failing, if it's my end or callcentric's.
Used this page:
http://sysadminman.net/blog/2008/callcentric-trunk-setup-with-asteriskfr...
To set up my trunk.
Outgoing Settings:
Trunk Name: callcentric
PEER Details:
username=1777xxxxxxx
type=peer
secret=PASSWORD
qualify=yes
nat=no
insecure=very
host=callcentric.com
fromuser=1777xxxxxxx
fromdomain=callcentric.com
dtmfmode=rfc2833
disallow=all
context=custom-get-did-from-sip
canreinvite=yes
allow=ulaw
Register String:
1777xxxxxxx:PASSWORD@callcentric.com/1777xxxxxxx
Also have some dial rules in there. Everything else is blank or default.
This works for outbound with an outbound route I have.
In /etc/asterisk/extensions_custom.conf I have:
[custom-get-did-from-sip]
exten => _.,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
Then in Inbound Routes
Description: Callcentric DID
DID Number: 1513xxxxxxx (my DID)
Signal Ringing: On (I've tried this both ways)
Destination: Doesn't matter, right now it's set to voicemail for extension 502.
What happens is, watching asterisk logs I make an incoming call to my DID. I see it trigger, I see the correct DID in the logs, I see it routing how I have it set, voicemail, ringing an extension whatever.
When I set it to ring an extension (a copy of X-Lite Beta) it rings. I answer. The dialing phone keeps on ringing.
When I set it to voicemail the logs show:
--
--
--
--
--
but the dialing phone keeps on ringing. Immediately after that bit of log I get:
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'SIP/callcentric-b67045b0' in macro 'vm'
== Spawn extension (ext-local, vmb502, 1) exited non-zero on 'SIP/callcentric-b67045b0'
I'm stumped. No matter what I do the dialing phone never hears an answer, and just keeps ringing.
Where the heck do I even start?



Callcentric setup - from Callcentric Tech Support
Callcentric has the best tech support of anybody I've worked with in this industry. You can't call them but if you submit a support request they will go above and beyond to answer questions about Asterisk and FreePBX not really related to their service.
Here is the setup they helped me put together. If this doesn't help login to your callcentric acct and submit a support request.
Trunk Name: Callcentric
Dial Rules: 1513+NXXXXXX
Peer Details:
username=17779998888
type=peer
secret=password
insecure=port,invite
host=callcentric.com
fromuser=17779998888
fromdomain=callcentric.com
disallow=all
context=from-pstn
allow=g729&ulaw&alaw
User Details is left Blank
Register String: 17779998888:password@callcentric.com
INBOUND ROUTE SETTINGS
route name: from-cc
DID Number: 17779998888
Pick a destination
-----------------------
If all else fails add an inbound route called "from-pstn" with no DID number specified and pick your own destination
I don't suppose your
I don't suppose your Asterisk/Trixbox server is behind a NAT'd connection is it? From what you say it looks as though the calls are coming in correctly.