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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:

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

Помогите разобраться Asterisk+Polycom IP6000

sip.conf

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 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 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' '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 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 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: 7070