Inbound calls not working

[osmanhalifa]

Good Day I have vicibox setup on my server. My outbound calls are working fine but however my inbound calls don't work. When I dial the number from my mobile phone I get " I'm sorry we cannot connect your call 233", even in CLI nothing shows also read the manager manual still cant find a way around it. Tried SIP debug and nothing on there I cant notice anything that relates to my DID, I created the DID extension and routed it to my phone extension to test but nothing. My setup is as follows:

My internal LAN consists of a router that users a HSDPA+ connection to the internet (ADSL SPEEDS ARE BAD WERE I AM), I have 5 pcs connected to a switch and my router connected to my switch. I contacted my carrier and they say everything is fine on their side. I'm using Switch2voip as my carrier for both outbound and inbound calls they have given me my DID as follows: 13232031640.

Now I suspect I need to setup port forwarding on my router and get a static IP from a service provider, OR something is wrong on my sip.conf or extensions.conf file on asterisk.

My carrier settings are as follows:

Account entry:
[Switch2Voip]
username=971***5********
type=friend
secret=5*******
context=trunkinbound
progressinband=never
port=5060
nat=auto
insecure=invite
ignoresdpversion=yes
host=sip.switch2voip.us
dtmfmode=rfc2833
canreinvite=no
allow=g729&g711&g723
fromuser=97********

dial plan entry:
exten => _91.,1,Set(callerid(num)=+1XXXXXXXXXX)
exten => _91.,2,Set(callerid(ani)=Phone number)
exten => _91.,3,AGI(agi://127.0.0.1:4577/call_log)
exten => _91.,4,Dial(sip/${EXTEN:1}@Switch2Voip,55,o)
exten => _91,5,Hangup

DID setup
DID Extension: 13232031640
DID Description:
Active: y
Admin User Group: all
DID Route: PHONE
Record Call:
Extension: 9998811112
Extension Context: default
Voicemail Box:
Phone Extension: 0111
Server IP: 10.0.0.25



SIP DEBUG:
OPTIONS sip:0111@10.0.0.3:6018;cpd=on SIP/2.0
Via: SIP/2.0/UDP 10.0.0.25:5060;branch=z9hG4bK7fd2b64a;rport
From: "asterisk" <sip:asterisk@10.0.0.25>;tag=as0ffe06f7
To: <sip:0111@10.0.0.3:6018;cpd=on>
Contact: <sip:asterisk@10.0.0.25>
Call-ID: 644679fd082e2b7d4e6f1fb87ac15d8c@10.0.0.25
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 10 Sep 2013 08:23:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Sep 10 04:23:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 10 04:23:02]
<--- SIP read from 10.0.0.3:6018 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.25:5060;branch=z9hG4bK7fd2b64a;rport=5060
Contact: <sip:10.0.0.3:6018>
To: <sip:0111@10.0.0.3:6018;cpd=on>;tag=3f59652c
From: "asterisk"<sip:asterisk@10.0.0.25>;tag=as0ffe06f7
Call-ID: 644679fd082e2b7d4e6f1fb87ac15d8c@10.0.0.25
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: eventlist
User-Agent: eyeBeam release 3010n stamp 19039
Content-Length: 0






ROUTER: DLINK DSL 275OU

OpenSUSE 12.1

VICIBOX 4.0.3

Asterisk Version: 1.4.44-vici

SERVER BOX
DUALCORE 3.0
4GIGS OF RAM
500GIG SATA HD

[DruRoland]

AFAIK, AT&T/T-Mo use private 10.X.X.X IP addresses for their devices. This will prevent you from being able to receive calls.