Key events & Combinations of keys

Hi,
I'm listening to key events and it's fine one one key is pressed;
if (event.getKeyCode() == KeyEvent.VK_UP)
}But I need to know when a combination of keys is pressed (like ctrl-a). How can I know if two specifics keys are pressed together?
Thank you...

Read JavaDoc. java.awt.event.InputEvent class.
if (event.getKeyCode() == KeyEvent.VK_UP && event.isControlDown())
}

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    *&      Module  STATUS_0100  OUTPUT
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      if tree1 is initial.
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      endif.
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      append l_event to lt_events.
      l_event-eventid = cl_gui_column_tree=>eventid_node_context_menu_req.
      append l_event to lt_events.
      l_event-eventid = cl_gui_column_tree=>eventid_item_context_menu_req.
      append l_event to lt_events.
      l_event-eventid = cl_gui_column_tree=>eventid_item_double_click.
      append l_event to lt_events.
      l_event-eventid = cl_gui_column_tree=>eventid_item_keypress.
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                textline1    = sy-subrc
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                start_row    = 6.               "#EC NOTEXT
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    * create container for alv-tree
      data: l_tree_container_name(30) type c,
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      l_tree_container_name = 'TREE1'.
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                cntl_system_error           = 2
                create_error                = 3
                lifetime_error              = 4
                lifetime_dynpro_dynpro_link = 5.
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            no_toolbar          = ''
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            cntl_system_error            = 2
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            lifetime_error               = 4
            illegal_node_selection_mode  = 5
            failed                       = 6
            illegal_column_name          = 7.
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    Code Formatted by: Alvaro Tejada Galindo on Feb 14, 2008 9:54 AM
    Edited by: Lukasz Sekowski on Feb 14, 2008 4:11 PM

    I've finally fixed the problem:
    there was:
    item_selection      = space
    and should be:
    item_selection      = 'X'

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    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Proxy-Authenticate: Digest realm="sip.skype.com", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", algorithm=MD5
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
    Content-Length: 0
    <------------->
    --- (8 headers 0 lines) ---
    set_destination: Parsing <sip:[email protected]> for address/port to send to
    set_destination: set destination to 63.209.144.201:5060
    Transmitting (NAT) to 63.209.144.201:5061:
    ACK sip:[email protected] SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 10.5.2
    Content-Length: 0
    Audio is at 9302
    Adding codec 100003 (ulaw) to SDP
    Adding codec 100004 (alaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 63.209.144.201:5061:
    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX 10.5.2
    Proxy-Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:[email protected]", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", response="6efb0e37178bae868f0a1e0ddf110e3c"
    Date: Thu, 23 Aug 2012 23:22:34 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 370
    v=0
    o=root 1671301052 1671301053 IN IP4 192.168.1.15
    s=Asterisk PBX 10.5.2
    c=IN IP4 192.168.1.15
    t=0 0
    m=audio 9302 RTP/SAVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 100 Trying
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
    Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    Content-Length: 0
    <------------->
    --- (8 headers 0 lines) ---
    Really destroying SIP dialog '[email protected]' Method: REGISTER
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 180 Ringing
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 103 INVITE
    User-Agent: SipGW 8
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
    Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    Content-Length: 0
    <------------->
    --- (9 headers 0 lines) ---
    list_route: hop: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    -- SIP/skype-000000b1 is ringing
    <--- Transmitting (NAT) to 192.168.1.16:5060 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>;tag=as3f27fa61
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Server: Asterisk PBX 10.5.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Contact: <sip:[email protected]:5060>
    Content-Length: 0
    <------------>
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 408 Request Timeout
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
    Content-Length: 0
    <------------->
    --- (7 headers 0 lines) ---
    [2012-08-23 19:22:45] WARNING[17932]: chan_sip.c:20947 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '[email protected]'. Giving up.
    set_destination: Parsing <sip:[email protected]> for address/port to send to
    set_destination: set destination to 63.209.144.201:5060
    Transmitting (NAT) to 63.209.144.201:5061:
    ACK sip:[email protected]:5061;maddr=63.209.144.201;transport=tls SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 103 ACK
    User-Agent: Asterisk PBX 10.5.2
    Content-Length: 0
    Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
    == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [19739928881@home:2] Hangup("SIP/scott_office-000000b0", "") in new stack
    == Spawn extension (home, 19739928881, 2) exited non-zero on 'SIP/scott_office-000000b0'
    Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
    <--- Reliably Transmitting (NAT) to 192.168.1.16:5060 --->
    SIP/2.0 503 Service Unavailable
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>;tag=as3f27fa61
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Server: Asterisk PBX 10.5.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0
    <------------>
    <--- SIP read from UDP:192.168.1.16:5060 --->
    ACK sip:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>;tag=as3f27fa61
    Call-ID: [email protected]
    CSeq: 101 ACK
    Max-Forwards: 70
    Contact: "Scott's Office" <sip:[email protected]:5060>
    User-Agent: Cisco/SPA504G-7.5.2b
    Content-Length: 0
    <------------->
    --- (10 headers 0 lines) ---
    Really destroying SIP dialog '[email protected]' Method: INVITE
    Really destroying SIP dialog '[email protected]' Method: ACK

    I wound up calling skype support. This is the final sip.conf looks like. Hope it helps. Good luck.
    Scott
    [general]
    context=default_context
    allowguest=no
    alwaysauthreject=yes
    allowoverlap=no
    udpbindaddr=0.0.0.0
    tlsenable=yes
    tlsbinddir=0.0.0.0
    tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
    tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
    tlscipher=ALL
    tlsclientmethod=tlsv1
    tcpenable=yes
    tcpbindaddr=0.0.0.0
    transport=udp,tcp,tls
    srvlookup=yes
    dynamic_exclude_static = yes
    buggymwi=yes
    contactpermit=192.168.1.0/24
    register => tls://[email protected]
    [skype]
    type=friend
    context=from-skype
    dtmfmode=rfc2833
    host=sip.skype.com
    username=user
    fromuser=user
    secret=pass
    disallow=all
    allow=ulaw
    allow=alaw
    nat=yes
    fromdomain=sip.skype.com
    insecure=port,invite
    transport=tls
    srtpcapable=yes
    encryption=yes

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