asterisk users - Dec 2010 - Fw: Sip Hangup after critical packet SIP DEBUG attached

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HI
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I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn)
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Out going calls from asterisk to the ata works fine
Incoming calls from the ata to asterisk cuts off with the error msg
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Maximum retries exceeded on transmission 70854efe-4157e3a8 at 10.168.7.103 for 
seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Dec? 6 13:52:43] WARNING[3921]: chan_sip.c:3858 retrans_pkt:
?Hanging up call 70854efe-4157e3a8 at 10.168.7.103 - no reply to our critical 
packet (see doc/sip-retransmit.txt).

I been googling this error and it was mentioned to use
t1min= 500 however its only delaying the problem.
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any ideas on what is the cause of this problem.
Only 2-3 atas are having this problem the rest are fine.
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Here is the sip debug
the sip invites are not being received
and in one of the message a busy response was sent back.
?Retransmitting #4 (no NAT) to 10.168.7.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103
From: Ridge <sip:287 at 10.10.0.1>;tag=f314fd35733eba9bo0
To: <sip:204 at 10.10.0.1>;tag=as4593172b
Call-ID: f54a1cbd-891ce0b3 at 10.168.7.103
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:204 at 41.146.208.131>
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 777980638 777980638 IN IP4 41.146.208.131
s=Asterisk PBX 1.6.2.13
c=IN IP4 41.146.208.131
t=0 0
m=audio 19726 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
?
Retransmitting #5 (no NAT) to 10.168.7.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103
From: Ridge <sip:287 at 10.10.0.1>;tag=f314fd35733eba9bo0
To: <sip:204 at 10.10.0.1>;tag=as4593172b
Call-ID: f54a1cbd-891ce0b3 at 10.168.7.103
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:204 at 41.146.208.131>
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 777980638 777980638 IN IP4 41.146.208.131
s=Asterisk PBX 1.6.2.13
c=IN IP4 41.146.208.131
t=0 0
m=audio 19726 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Reliably Transmitting (no NAT) to 10.168.7.103:5060:
OPTIONS sip:287 at 10.168.7.103:5060 SIP/2.0
Via: SIP/2.0/UDP 41.146.208.131:5060;branch=z9hG4bK226c4b89;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 41.146.208.131>;tag=as21bdce7e
To: <sip:287 at 10.168.7.103:5060>
Contact: <sip:asterisk at 41.146.208.131>
Call-ID: 062f8f6c4e7f5929487f3db12a93f7c2 at 41.146.208.131
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 06 Dec 2010 12:47:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
?
---
<--- SIP read from UDP:10.168.7.103:5060 --->
SIP/2.0 486 Busy Here
To: <sip:287 at 10.168.7.103:5060>;tag=18c8b9ab85ca5068i0
From: "asterisk" <sip:asterisk at 41.146.208.131>;tag=as21bdce7e
Call-ID: 062f8f6c4e7f5929487f3db12a93f7c2 at 41.146.208.131
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 41.146.208.131:5060;branch=z9hG4bK226c4b89
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
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<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '062f8f6c4e7f5929487f3db12a93f7c2 at
41.146.208.131'
Method: OPTIONS
Retransmitting #6 (no NAT) to 10.168.7.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103
From: Ridge <sip:287 at 10.10.0.1>;tag=f314fd35733eba9bo0
To: <sip:204 at 10.10.0.1>;tag=as4593172b
Call-ID: f54a1cbd-891ce0b3 at 10.168.7.103
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:204 at 41.146.208.131>
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 777980638 777980638 IN IP4 41.146.208.131
s=Asterisk PBX 1.6.2.13
c=IN IP4 41.146.208.131
t=0 0
m=audio 19726 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
?
?
zakir
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