Inbound call routing not working

I am new to Asterisk and come from th Broadsoft world… I am looking at creating an IVR application for a non-profit but am having this problem:

Running A@H 2.8, the most recent version
FreePBX Administration 2.0.1
Shorewall Firewall

Getting an 404 error on inbound calls from 2 different Voip SP’s

Ethereal shows Error 404 Service Univalable from the asterisk response

Here are the asterisk logs from the call:

Jun 1 03:38:35 DEBUG[4572] chan_sip.c: Auto destroying call '[email protected]
Jun 1 03:38:39 DEBUG[4572] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Request 102: Match Found
Jun 1 03:38:39 DEBUG[4572] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Request 102: Match Found
Jun 1 03:38:40 DEBUG[4572] chan_sip.c: Setting NAT on RTP to 0
Jun 1 03:38:40 DEBUG[4572] chan_sip.c: Checking SIP call limits for device 19783384654
Jun 1 03:38:40 NOTICE[4572] pbx.c: Cannot find extension context 'global’
Jun 1 03:38:40 DEBUG[4572] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Response 102: Match Found
Jun 1 03:38:42 DEBUG[5619] manager.c: Manager received command 'Command’
Jun 1 03:38:42 DEBUG[5619] manager.c: Manager received command 'Command’
Jun 1 03:38:42 DEBUG[5619] manager.c: Manager received command ‘Command’

I see this as the problem: Jun 1 03:38:40 NOTICE[4572] pbx.c: Cannot find extension context ‘global’

I have it going to Core ext 202… but i have tred putting it into a que and AA

Here are my FreePX entries… let me know if you need any .conf files… though i have not customized anything in them:

Incoming settings
User Context : Viatalk

authuser=19785551212
canreinvite=yes
dtmf=inband
dtmfmode=inband
fromdomain=neptune.vtnoc.net
fromuser=19785551212
host=neptune.vtnoc.net
insecure=very
nat=no
qualify=yes
secret=kjhdsfkjh
type=peer
username=19785551212

Inbound route
DID 19785551212
core ext 202

Thanks for any help!

Yup. Add:

context=from-pstn

To your inbound settings for the Trunk.