No "sip domain name" field in trunk configuration

Hello,
A customer' s UC320 throw a "sip 484" error,
Sep 22 16:10:44 UC320W user.debug voice: SIP/2.0 484 Address Incomplete
I think the problem is the lack of the  "sip domain name" field in the  trunk configuration as i can see one here;
https://supportforums.cisco.com/servlet/JiveServlet/showImage/102-15041-40-10
https://supportforums.cisco.com/docs/DOC-15041#UC_320_SIP_Trunk_Configuration
See attached the screenshot of the state of the trun, it misses his domain after the @
The UC320 is upgraded to 2.2.2 .

The case number is; 623254153
This is what the provider can see, (they say they cant handle "notify" requests);
U 2012/09/21 14:18:28.183877 IP.IP.IP.IP:5060 -> ip.ip.ip.ip:5060
NOTIFY sip:sip.myprovider.net SIP/2.0.
Via: SIP/2.0/UDP IP.IP.IP.IP:5060;branch=z9hG4bK-d56a8b2a.
From: "NIT" [email protected]>;tag=d78ae5c25878b8eco5.
To: .
Call-ID: [email protected]
CSeq: 211 NOTIFY.
Max-Forwards: 70.
Contact: "NIT" [email protected]:5060;ref=MYSIPID>.
Event: keep-alive.
User-Agent: Cisco/UC320W-2.2.2(1).
Allow-Events: talk, hold, conference, x-spa-cti.
Content-Length: 0.
U 2012/09/21 14:18:28.184243 ip.ip.ip.ip:5060 -> IP.IP.IP.IP:5060
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP IP.IP.IP.IP:5060;branch=z9hG4bK-d56a8b2a.
From: "NIT" [email protected]>;tag=d78ae5c25878b8eco5.
To: ;tag=696c4aa18cbb1138a6b8a116e3582f18.6a97.
Call-ID: [email protected]
CSeq: 211 NOTIFY.
Server: Alphalink SIP Proxy 2.0.
Content-Length: 0.

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    voice service voip
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      ipv4 192.0.2.0 255.255.255.0
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    incoming called-number 9T
    session protocol sipv2
    codec g711ulaw
    dtmf-relay rtp-nte
    Outbound Dial-Peer for calls from CUBE to CUCM (SP will be sending 10 digits inbound)
    dial-peer voice 200 voip
    description *** Outbound LAN side dial-peer ***
    destination-pattern [2-9].........
    session protocol sipv2
    session target ipv4:
    codec g711ulaw
    dtmf-relay rtp-nte
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    Inbound Dial-Peer for calls from SP to CUBE
    dial-peer voice 100 voip
    description *** Inbound WAN side dial-peer ***------------------(catch-all for all inbound PSTN calls)
    incoming called-number [2-9].........
    session protocol sipv2
    codec g711ulaw
    dtmf-relay rtp-nte
    Outbound Dial-Peer for calls from CUBE to SP
    dial-peer voice 200 voip
    description *** Outbound WAN side dial-peer ***
    translation-profile outgoing Digitstrip
    destination-pattern 9[2-9].........
    session protocol sipv2
    voice-class sip bind control source gig0/1
    voice-class sip bind media source gig0/1
    session target ipv4::XXXX (where XXXX is the port number your provider is using if different from 5060)
    codec g711ulaw
    dtmf-relay rtp-nte
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    Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs (if supported by provider)
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    Fax calls – T.38, modem pass-through--whichever one you decide to use
    Please rate all useful posts
    "opportunity is a haughty goddess who waste no time with those who are unprepared"

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