Помогите разобраться Asterisk+Polycom IP6000

Помогите пожалуйста разобраться. Пытаюсь подсоединить Polycom к MeetMe Trixbox перепробовал все как написано в форумах не как не получается, может что то не разглядел.

sip.conf

[7004]
deny=0.0.0.0/0.0.0.0
disallow=all
type=friend
qualify=yes
port=5060
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=7004@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/7004
context=from-sip
canreinvite=no
callerid=device <7004>
allow=g729,alaw,ulaw
call-limit=50

Когда делаю sip set debug ip 10.14.0.133 То пишит:

<--- SIP read from UDP://10.14.0.132:5060 --->
INVITE sip:7070@10.14.0.131 SIP/2.0 
Via: SIP/2.0/UDP 10.14.0.132:5060;branch=z9hG4bKb050e80ba47
From: <sip:7003@10.14.0.131>;tag=b050e80ba4
To: <sip:7070@10.14.0.131>
Call-ID: b0626c50-7f5e-e86b-800b-0002a40252aa@10.14.0.132
CSeq: 7 INVITE
Supported: replaces, timer, 100rel, early-session
Min-SE: 1800
Date: Wed, 03 Oct 2012 16:07:12 GMT
Session-Expires: 1800
User-Agent: AddPac SIP Gateway
Contact: <sip:7003@10.14.0.132>
Accept: application/sdp
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 357
Max-Forwards: 70

v=0
o=7003 1349280432 1349280432 IN IP4 10.14.0.132
s=AddPac Gateway SDP
c=IN IP4 10.14.0.132
b=AS:256
t=1349280432 0
m=audio 23014 RTP/AVP 18 8 0 101
a=ptime:20
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 23016 RTP/AVP 99
a=framerate:27.0
a=rtpmap:99 H264/90000

<------------->
--- (17 headers 16 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
Sending to 10.14.0.132 : 5060 (no NAT)
Using INVITE request as basis request - b0626c50-7f5e-e86b-800b-0002a40252aa@10.14.0.132
Found user '7003' for '7003'

<--- Reliably Transmitting (no NAT) to 10.14.0.132:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.14.0.132:5060;branch=z9hG4bKb050e80ba47;received=10.14.0.132
From: <sip:7003@10.14.0.131>;tag=b050e80ba4
To: <sip:7070@10.14.0.131>;tag=as2122f902
Call-ID: b0626c50-7f5e-e86b-800b-0002a40252aa@10.14.0.132
CSeq: 7 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e5b59b5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'b0626c50-7f5e-e86b-800b-0002a40252aa@10.14.0.132' in 32000 ms (Method: INVITE)
trixbox1*CLI> 
<--- SIP read from UDP://10.14.0.132:5060 --->
ACK sip:7070@10.14.0.131 SIP/2.0
Via: SIP/2.0/UDP 10.14.0.132:5060;branch=z9hG4bKb050e80ba47
From: <sip:7003@10.14.0.131>;tag=b050e80ba4
To: <sip:7070@10.14.0.131>;tag=as2122f902
Call-ID: b0626c50-7f5e-e86b-800b-0002a40252aa@10.14.0.132
CSeq: 7 ACK
Content-Length: 0
Max-Forwards: 70


<------------->
--- (8 headers 0 lines) ---
trixbox1*CLI> 
<--- SIP read from UDP://10.14.0.132:5060 --->
INVITE sip:7070@10.14.0.131 SIP/2.0
Via: SIP/2.0/UDP 10.14.0.132:5060;branch=z9hG4bKb050e80ba48
From: <sip:7003@10.14.0.131>;tag=b050e80ba4
To: <sip:7070@10.14.0.131>
Call-ID: b0626c50-7f5e-e86b-800b-0002a40252aa@10.14.0.132
CSeq: 8 INVITE
Supported: replaces, timer, 100rel, early-session
Min-SE: 1800
Date: Wed, 03 Oct 2012 16:07:12 GMT
Session-Expires: 1800
User-Agent: AddPac SIP Gateway
Authorization: Digest username="7003", realm="asterisk", nonce="1e5b59b5", uri="sip:7070@10.14.0.131", response="6e55a32b3215e5a4aa38b03169604699", algorithm=MD5
Contact: <sip:7003@10.14.0.132>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 357
Max-Forwards: 70