SIP Header Problem

My VOIP provider is trying to ban other SIP client and force to use the software provided by that provider, I just use asterisk simulated as that software, but i just call use as an inbound call, cannot dial out…

I found something strange in the SIP INVITE Header, is it normal?

The VOIP Client Sends
INVITE sip:2888XXXX@219.76.95.XX SIP/2.0
From: "3924XXXX"sip:3924XXXX@219.76.95.40;tag=ce7410-1a448d0a-13c4-45028-7d759-1f57c912-7d759
To: "2888XXXX"sip:2888xxxx@219.76.95.40
Call-ID: ceaae8-1a448d0a-13c4-45028-7d759-1c4c952c-7d759
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.141.68.26:5060;rport;branch=z9hG4bK-7d759-1ea13663-6f567f33
Max-Forwards: 70
Supported: replaces,100rel
Stamp: 1236236004
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL
[size=150]Checksum: 01c780989887e7bbcc8dfaff5851647b[/size]
Contact: sip:3924XXXX@10.141.68.26
ChecksumB: 10.141.68.26
Content-Type: application/sdp
Content-Length: 182

v=0
o=3924XXXX 1236236250 1236236250 IN IP4 10.141.68.26
s=SIP Call
c=IN IP4 10.141.68.26
t=0 0
m=audio 7080 RTP/AVP 0
c=IN IP4 10.141.68.26
a=rtpmap:0 PCMU/8000
a=ptime:20

=========================================
If I use asterisk server send the INVITE header to the gateway, the server return : 403 Forbidden, not either 401 or 407.

What is checksum in the SIP header?
From the user-agent, i saw that the server user-agent is “User-Agent: Cisco-SIPGateway/IOS-12.x”, is such Cisco VOIP gateway can do such “validation” to prevent other VOIP client to login?

thanks