can't connect to my sip provider

Hi,
I was running asterisk to dial out using callwithus.com no problem. Suddenly, today I found out Asterisk is having problem connecting to callwithus and I could not find any problem on my side. I can use SIP client to connect to callwithus but not my Asterisk. What problem could it be? It was working fine before and no configuration change was made. Please help me.
Here is the sip.conf and the log:

---
Really destroying SIP dialog '74c6543e2891fb1138ca9d3c3721b360@192.168.1.101' Method: REGISTER
Really destroying SIP dialog '842eaa23-623ea9d4-60569@216.127.66.119' Method: OPTIONS
Retransmitting #1 (NAT) to 64.85.162.136:5060:
REGISTER sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK41e1a45f;rport
From: ;tag=as29a4b9a6
To:
Call-ID: 74c6543e2891fb1138ca9d3c3721b360 [!at] 192.168.1.101 (replace the [!at] with a @)
CSeq: 984 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact:
Event: registration
Content-Length: 0


---
Retransmitting #2 (NAT) to 64.85.162.136:5060:
REGISTER sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK41e1a45f;rport
From: ;tag=as29a4b9a6
To:
Call-ID: 74c6543e2891fb1138ca9d3c3721b360 [!at] 192.168.1.101 (replace the [!at] with a @)
CSeq: 984 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact:
Event: registration
Content-Length: 0


---
Retransmitting #3 (NAT) to 64.85.162.136:5060:
REGISTER sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK41e1a45f;rport
From: ;tag=as29a4b9a6
To:
Call-ID: 74c6543e2891fb1138ca9d3c3721b360 [!at] 192.168.1.101 (replace the [!at] with a @)
CSeq: 984 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact:
Event: registration
Content-Length: 0


---
Reliably Transmitting (NAT) to 64.85.162.136:5060:
OPTIONS sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK7cc60ff5;rport
From: "asterisk" ;tag=as151102d8
To:
Contact:
Call-ID: 27f46518154096a95533169853d806ac [!at] 192.168.1.101 (replace the [!at] with a @)
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 01 Apr 2008 22:39:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


[general]

port = 5060
bindaddr = 0.0.0.0
context = others
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=yes
canreinvite=no
extenip=58.251.75.228
localhost=192.168.1.1/255.255.255.0
register=>USER:PASS@sip.callwithus.com/XXXXXXX
[callwithus]
type=friend
host=sip.callwithus.com
username=XXXXXXXX
secret=XXXXX
qualify=no
insecure=invite
context=fax-out