Process for porting phone # to FIOS?

Can someone point me to instructions on how to port my # from Comcast/Xfinity to FIOS? 
We had FIOS for years until five weeks ago, switched to Xfinity and ported the home # I've had for 10 years, then switched back to FIOS yesterday.  During the ordering process with FIOS we were clear that we wanted to port in our #.  During the last contact with FIOS before installation they told my husband to wait a week after switching to call to port the #, and the installer yesterday issued us a new # (which he said would be temporary).  FIOS service was closed by the time the installation was done yesterday, and is closed today.  What is the process for porting the #?  Is there a place online we can do this when FIOS is closed?  How long before Xfinity releases the #?  If I lose this # that is connected to basically every other account I have, I will be furious.

In March I switched my phone from Vonage to VZ digital voice.  I did this so I was able to get a better bundled price.
 I informed the VZ CS that my intent is to move my phone from Vonage to VZ.  They told me that it will take 5 days for this to occur and a truck roll would need to be scheduled.  On the day of the appointment my number continued to work with vonage.  The VZ tech arrived, wired up the home phone line to the ONT and the phone line was dead.  She went to the truck and whatever magic occurred she returned 5 minutes later and told me to see if I had dial tone.   Upon picking up the phone there was dial tone.  I called my cell and it displayed the caller ID of my phone.  I then called my home phone and it worked too.  
Since you already have dial tone I'm sure no truck roll will be necessary.  I do understand there is a 5 day porting process but once you call TS and tell them your intent it should be pretty straight forward.  When I plaecd my original order it was documented on that order.  Not sure if this helps but hope it does.
Jim

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    |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.xx.4.xx on port 56714 index 31809 with 973 bytes:
    [6387070,NET]
    NOTIFY sip:[email protected] SIP/2.0
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    To: <sip:[email protected]>
    From: <sip:[email protected]>;tag=00ffb00bc50a00340000499f-00006ab4
    Call-ID: [email protected]
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    Contact: <sip:[email protected]:56714;transport=TCP>
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    Content-Length: 350
    Content-Type: application/dialog-info+xml
    Content-Disposition: session;handling=required
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    Here is a more detailed explanation of how SIP calls notify cucm when they go off hook to make a call. The digit dialled here is 4080
    +++++ Analysis of SIP Phone making a call +++++++++
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    00869539.002 |14:58:13.837 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 976 bytes:
    [46240,NET]
    NOTIFY sip:[email protected] SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
    To: <sip:[email protected]>
    From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:13 GMT
    CSeq: 11 NOTIFY
    Event: dialog
    Subscription-State: active
    Max-Forwards: 70
    Contact: <sip:[email protected]:52910;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 350
    Content-Type: application/dialog-info+xml
    Content-Disposition: session;handling=required
    <?xml version="1.0" encoding="UTF-8" ?>
    <dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="10" state="partial" entity="sip:[email protected]">
    <dialog id="6" call-id="[email protected]" local-tag="544e42f26d0b001d00007cc9-000044a3"><state>trying</state></dialog>
    </dialog-info>
    ++++ CUCM SIP stack processes the new connection for the phone+++++++
    00869540.001 |14:58:13.837 |AppInfo  |//SIP/Stack/Info/0x0/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 (SIP_NETWORK_MSG), for event 1 (SIPSPI_EV_NEW_MESSAGE)
    00869540.002 |14:58:13.837 |AppInfo  |//SIP/Stack/Transport/0x0/sipTransportProcessNWNewConnMsg: context=(nil)
    00869540.003 |14:58:13.837 |AppInfo  |//SIP/Stack/Transport/0x0/sipConnectionManagerProcessNewConnMsg: gConnTab=0xe81c0d70, addr=10.50.16.1, port=52910, connid=2748, transport=TCP
    ++++ Next CUCM allocates a call id for this call +++++
    00869546.002 |14:58:13.838 |AppInfo  |LineControl(66) - Get call instance=1 for CI=24419584
    +++Next CUCM sends a 200 OK to the NOTIFY request for the new dialog ++++
    00869555.007 |14:58:13.839 |AppInfo  |//SIP/Stack/Transport/0x0xe7df4d48/sipTransportPostSendMessage: Posting send for msg=0xefbe9910, addr=10.50.16.1, port=52910, connId=2748 for
    00869555.008 |14:58:13.839 |AppInfo  |//SIP/Stack/Info/0x0/act_dialog_pending_resp_event: Changing from State: SUBSCRIBE_STATE_DIALOG_PENDING to state SUBSCRIBE_STATE_ACTIVE
    00869556.000 |14:58:13.839 |SdlSig   |SIPSPISignal                           |wait                           |SIPTcp(1,100,71,1)               |SIPHandler(1,100,79,1)           |1,100,14,31314.75^10.50.16.1^SEP00909E9D106C |*TraceFlagOverrode
    00869556.001 |14:58:13.839 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46241,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
    From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
    To: <sip:[email protected]>;tag=1822746380
    Date: Mon, 16 Feb 2015 12:58:13 GMT
    Call-ID: [email protected]
    CSeq: 11 NOTIFY
    Server: Cisco-CUCM10.5
    Content-Length: 0
    ++++ The IP Phone sends its connection ID to CUCM, its ip address and its port number+++++++++
    00869541.001 |14:58:13.838 |AppInfo  |SIPStationInit: connID=2748, SEP00909E9D106C, 10.50.16.1:52910, Routed signal by connection index to (1,100,73,66)
    ++++ Next CUCM informs us that the NOTIFY message is for an offhook event ++++++
    00869542.003 |14:58:13.838 |AppInfo  |SIPStationD(66) - processCommonDialogNotifyInd: Notified Dialogs - Did 6 State trying
    00869542.004 |14:58:13.838 |AppInfo  |SIPStationD(66) - processCommonDialogNotifyInd:   Did 6 Sending Notified SIPOffHook to new Cdfc
    00869542.010 |14:58:13.838 |AppInfo  |SIPStationD(66) - processSIPOffHook Primary Call Not-Found
    00869543.000 |14:58:13.838 |SdlSig   |SIPOffHookInd 
    +++ The next thing is the USER dials a digit on the phone ++++++
    This is where it gets a little complicated. So lets examine this. The first digit that is dialled generates an INVITE to CUCM like this:
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    00869559.002 |14:58:14.064 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 1445 bytes:
    [46242,NET]
    INVITE sip:[email protected];user=phone SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
    From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
    To: <sip:[email protected];user=phone>
    Call-ID: [email protected]
    Max-Forwards: 70
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 101 INVITE
    User-Agent: Cisco-SIPIPCommunicator/9.1.1
    Contact: <sip:[email protected]:52910;transport=tcp>
    Expires: 180
    Accept: application/sdp
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
    Remote-Party-ID: "Emre ESEN" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes
    Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1
    Allow-Events: kpml,dialog
    Content-Length: 373
    Content-Type: application/sdp
    Content-Disposition: session;handling=optional
    v=0
    o=Cisco-SIPUA 21020 0 IN IP4 10.50.16.1
    s=SIP Call
    t=0 0
    m=audio 20250 RTP/AVP 0 8 18 9 116 124 101
    c=IN IP4 10.50.16.1
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:9 G722/8000
    a=rtpmap:116 iLBC/8000
    a=fmtp:116 mode=20
    a=rtpmap:124 ISAC/16000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    +++++ NEXT CUCM sends a trying for the INVITE it received +++++++++++
    00869562.001 |14:58:14.065 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46243,NET]
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
    From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
    To: <sip:[email protected];user=phone>
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: presence
    Content-Length: 0
    ++++NOW CUCM evaluates the DTMF supported by the phone to determine how to inform the phones to send the remaining dtmf digits++++
    From the INVITE cucm concludes that KPML and rtp-nte is supported
    00869566.009 |14:58:14.066 |AppInfo  |setEndpointsDtmfCaps: KPML Supported.
    00869566.010 |14:58:14.066 |AppInfo  |setEndpointsDtmfCaps: Detected inband DTMF support
    Next CUCM generates kpml event pkg which is going to be used to receive the remaining digits from the phone
    00869590.001 |14:58:14.067 |AppInfo  |SIPEventPkg::SIPEventPkg 0xe4a1d1e0 scbId[16725], event name[kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3], id[]
    +++ Next CUCM sends a SUBSCRIBE to the IP phone for kpml event +++++
    00869594.001 |14:58:14.068 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46244,NET]
    SUBSCRIBE sip:[email protected]:52910 SIP/2.0
    Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
    From: <sip:[email protected]>;tag=480227084
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 101 SUBSCRIBE
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    User-Agent: Cisco-CUCM10.5
    Event: kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3
    Expires: 7200
    Contact: <sip:[email protected]:5060;transport=tcp>
    Accept: application/kpml-response+xml
    Max-Forwards: 70
    Content-Type: application/kpml-request+xml
    Content-Length: 424
    <?xml version="1.0" encoding="UTF-8" ?>
    <kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0">
      <pattern criticaldigittimer="1000" extradigittimer="500" interdigittimer="15000" persist="persist">
        <regex tag="Backspace OK">[x#*+]|bs</regex>
      </pattern>
      </kpml-request>
     +++ Next we get a 200 OK to the SUBSCRIBE from the ip phone ++++
     00869595.002 |14:58:14.118 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 459 bytes:
    [46245,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
    From: <sip:[email protected]>;tag=480227084
    To: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 101 SUBSCRIBE
    Server: Cisco-SIPIPCommunicator/9.1.1
    Contact: <sip:[email protected]:52910;transport=TCP>
    Expires: 7200
    Content-Length: 0
    +++ NEXT the IP phones sends the remaining digit dialled on the phone to CUCM +++
    00869603.002 |14:58:14.183 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 573 bytes:
    [46247,NET]
    NOTIFY sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
    To: <sip:[email protected]>;tag=480227084
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 1000 NOTIFY
    Event: kpml
    Subscription-State: active; expires=7200
    Max-Forwards: 70
    Contact: <sip:[email protected]:52910;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 0
    00869608.001 |14:58:14.183 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46248,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    To: <sip:[email protected]>;tag=480227084
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    Call-ID: [email protected]
    CSeq: 1000 NOTIFY
    Server: Cisco-CUCM10.5
    Content-Length: 0
    +++Next the IP phone sends the next digit. Here its important to note that the NOTIFY doesnt contain the next digit,
    the NOTIFY is still the same as the first digit but the next digit is carried in the xml document attached to the NOTIFY.
    At this point I will insert a paragraph from the RFC 4730 for SIP KPML
    +++++++++++++
    The event package uses SUBSCRIBE
       messages and allows for XML documents that define and describe filter
       specifications for capturing key presses (DTMF Tones) entered at a
       presentation-free User Interface SIP User Agent (UA).  The event
       package uses NOTIFY messages and allows for XML documents to report
       the captured key presses (DTMF tones), consistent with the filter
       specifications, to an Application Server +++++++++++++++++++++++++++
    00869609.002 |14:58:14.209 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
    [46249,NET]
    NOTIFY sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
    To: <sip:[email protected]>;tag=480227084
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
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    CSeq: 1001 NOTIFY
    Event: kpml
    Subscription-State: active; expires=7200
    Max-Forwards: 70
    Contact: <sip:[email protected]:52910;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 209
    Content-Type: application/kpml-response+xml
    Content-Disposition: session;handling=required
    <?xml version="1.0" encoding="UTF-8"?>
    <kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
    00869622.001 |14:58:14.210 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46250,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    To: <sip:[email protected]>;tag=480227084
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    Call-ID: [email protected]
    CSeq: 1001 NOTIFY
    Server: Cisco-CUCM10.5
    Content-Length: 0
    +++ Again we get the next digit ++++
    00869624.002 |14:58:14.262 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
    [46251,NET]
    NOTIFY sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
    To: <sip:[email protected]>;tag=480227084
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 1002 NOTIFY
    Event: kpml
    Subscription-State: active; expires=7200
    Max-Forwards: 70
    Contact: <sip:[email protected]:52910;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 209
    Content-Type: application/kpml-response+xml
    Content-Disposition: session;handling=required
    <?xml version="1.0" encoding="UTF-8"?>
    <kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="8" tag="Backspace OK"/>
    00869637.001 |14:58:14.263 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
    [46252,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    To: <sip:[email protected]>;tag=480227084
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    Call-ID: [email protected]
    CSeq: 1002 NOTIFY
    Server: Cisco-CUCM10.5
    Content-Length: 0
    +++ Finally we get the last digit ++++
    00869638.002 |14:58:14.390 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
    [46253,NET]
    NOTIFY sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00006c1c
    To: <sip:[email protected]>;tag=480227084
    From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
    Call-ID: [email protected]
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    CSeq: 1003 NOTIFY
    Event: kpml
    Subscription-State: active; expires=7200
    Max-Forwards: 70
    Contact: <sip:[email protected]:52910;transport=TCP>
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
    Content-Length: 209
    Content-Type: application/kpml-response+xml
    Content-Disposition: session;handling=required
    <?xml version="1.0" encoding="UTF-8"?>
    <kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
    Once digit collection is completed CUCM proceeds to finalise its digit analysis process.
    Note that digit analysis is carried out for each digit that is recieved. I have only included the final DA here
    00869648.003 |14:58:14.391 |AppInfo  |Digit Analysis: star_DaReq: Matching SIP URL, Numeric User, user=4080
    00869648.004 |14:58:14.391 |AppInfo  |Digit Analysis: getDaRes data: daRes.ssType=[0] Intercept DAMR.sstype=[0], TPcount=[0], DAMR.NotifyCount=[0], DaRes.NotifyCount=[0]
    00869648.005 |14:58:14.391 |AppInfo  |Digit Analysis: getDaRes - Remote Destination [4080] isURI[0]
    00869648.012 |14:58:14.391 |AppInfo  |Digit analysis: match(pi="2", fqcn="9106", cn="9106",plv="5", pss="", TodFilteredPss="", dd="4080",dac="0")
    00869648.013 |14:58:14.391 |AppInfo  |Digit analysis: analysis results
    00869648.014 |14:58:14.391 |AppInfo  ||PretransformCallingPartyNumber=9106
    |CallingPartyNumber=9106
    |DialingPartition=
    |DialingPattern=4XXX
    |FullyQualifiedCalledPartyNumber=4080
    |DialingPatternRegularExpression=(4[0-9][0-9][0-9])
    |DialingWhere=
    +++++Once this is done CUCM then proceeds to send the call out to to the intended destination as configured in the RL ++++
    00869701.001 |14:58:14.435 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.250.0.13 on port 5060 index 2754
    [46256,NET]
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce931ee3d74
    From: "Emre ESEN" <sip:[email protected]>;tag=16726~813ee89e-33db-4d58-9f6a-61542cc840ee-24419585
    To: <sip:[email protected]>
    Date: Mon, 16 Feb 2015 12:58:14 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces

  • SQLite3-WinRT / SQLite3Component for Windows Phone 8.1 (HTML5 & JavaScript)

    Dear all
    I am a newbie in the developpement for Windows 8.1 and Windows Phone 8.1.
    After a few weeks reading books and watching videos to learn developping Windows Store Apps 8.1 with HTML5 and JavaScript, I have started my very first serious application as a Universal Windows Store App using visual studio 2013 community edition. I would
    like to use SQLite for persisiting the data of my application.
    I installed the necessary extensions to VS2013 both for windows 8.1 and windows phone 8.1 (SDK + SQLite extension)
    I cloned the SQLite3-WinRT projet from GitHub, and, as VS2013 asked it, run the targetting process to upgrade to Windows 8.1.
    I successfully installed and tested SQLite3-WinRT using Javascript for the WIndows 8.1 part of my project.
    But when i tried to add the SQLite3Component project as a reference to my Windows Phone 8.1 project, VS2013 complained that SQLite3Component project is an invalid reference for a windows phone project... I can understand this as the SQLite3Component project
    seems to be a windows 8 only project.
    I have been looking for a solution for two days now without success and decided therefore to post my question here: is there a SQLite3Component project targetted for WINdows Phone 8(.1)? If not, what are my alternatives for persisting potentially large amount
    of data with advanced search queries locally on a windows phone?
    Thank you for any help the comunity could give me.
    Regards
    François

    While not specifically talking about JavaScript/HTML5 this article series outlines how to use SQLite in Universal App projects:
    http://blog.tpcware.com/2014/04/universal-app-with-sqlite-part-1/
    I have already used it from C# and given that we're talking about WinRT components it should be accessible from WinJS as well. Hope it helps.

  • Voip application for windows phone 8.1 fails store submission

    A voip application with the new 8.1 architecture based on a separate process for the voip server is not accepted by the store.
    Voip architecture is based on 8.1 sample: https://code.msdn.microsoft.com/windowsapps/ChatterBox-VoIP-WP81-64921ad4
    Errors during submission:
    3011: The package is missing RpalManifest.xml. Update it and then try again.
    1028: The native API api-ms-win-core-processthreads-l1-1-2.dll:ExitProcess() isn’t allowed in assembly doubango.BackEndServerHost.exe. Update it and then try again.

    Update.  I am currently working with the sample team to update the sample.   We found the problem to be in the way the BackEndServerHost VIsual studio 
    project was created.   While we work on re-creating that from scratch, testing, and re-publishing I wanted to post what we have done so far in case anyone 
    wants to move forward.
    First there are a couple bugs in the sample, maybe you worked around those already in your development but in case not here they are.
    In the backendaudio.cpp BackEndCapture::InitCapture I changed the code:
    if (SUCCEEDED(hr))
    WAVEFORMATEX format = {};
    if (SUCCEEDED(hr))
    FillPcmFormat(format, m_pwfx->nChannels, m_pwfx->nSamplesPerSec, m_pwfx->wBitsPerSample);
    m_sourceFrameSizeInBytes = (format.wBitsPerSample / 8) * format.nChannels;
    hr = m_pDefaultCaptureDevice->Initialize(AUDCLNT_SHAREMODE_SHARED, 0x88140000, 1000 * 10000, 0, &format, NULL);
    The m_pwfx was already retrieved via GetMixFormat earlier, just use that.  Don’t create a custom one.
    In the BackEndAudio::Stop routine change the order of cleanup:
    if (m_pDefaultRenderDevice)
    m_pDefaultRenderDevice->Release();
    m_pDefaultRenderDevice = NULL;
    if (m_pDefaultCaptureDevice)
    m_pDefaultCaptureDevice->Release();
    m_pDefaultCaptureDevice = NULL;
    if (m_pCaptureClient)
    m_pCaptureClient->Release();
    m_pCaptureClient = NULL;
    The next step is to basically delete the BackEndServerHost project and create a new one with the following steps.
    1)  Add a new C++ Dirext X App (Windows Phone) to the solution and name it VoipBackendServerHost  NOTE: using "." in the name causes problems, so don't use the same name as the original sample.
    2)  Remove the Common and Content Folders
    3)  Remove App1Main.cpp and App1Main.h
    4)  Contents of app.cpp should be:
    #include "pch.h"
    #include "App.h"
    #include <windows.applicationmodel.core.h>
    #include <wrl.h>
    #include <string>
    #include <stdexcept>
    using namespace ABI::Windows::ApplicationModel::Core;
    using namespace ABI::Windows::Foundation;
    using namespace Microsoft::WRL;
    using namespace Wrappers;
    HRESULT __cdecl MyGetActivationFactory(_In_ HSTRING activatableClassId, _COM_Outptr_ IInspectable **factory);
    class GetCustomClass : public RuntimeClass<RuntimeClassFlags<RuntimeClassType::WinRtClassicComMix>,
    IGetActivationFactory,
    CloakedIid<IAgileObject >>
    public:
    IFACEMETHODIMP GetActivationFactory(_In_ HSTRING activatableClassId, _COM_Outptr_ IInspectable **factory)
    return MyGetActivationFactory(activatableClassId, factory);
    private:
    HMODULE m_hMod;
    [Platform::MTAThread]
    int main(Platform::Array<Platform::String^>^)
    HRESULT hr = Initialize(RO_INIT_MULTITHREADED);
    if (FAILED(hr))
    throw std::runtime_error(std::string("Failed to Initialize(RO_INIT_MULTITHREADED), HRESULT: ").append(std::to_string(hr)));
    // Scoping for smart pointers
    ComPtr<ICoreApplication> spApplicationFactory;
    hr = GetActivationFactory(HStringReference(RuntimeClass_Windows_ApplicationModel_Core_CoreApplication).Get(), &spApplicationFactory);
    if (FAILED(hr))
    throw std::runtime_error(std::string("Failed to GetActivationFactory(RuntimeClass_Windows_ApplicationModel_Core_CoreApplication), HRESULT: ").append(std::to_string(hr)));
    ComPtr<IGetActivationFactory> spGetActivationFactory = Make<GetCustomClass>();
    spApplicationFactory->RunWithActivationFactories(spGetActivationFactory.Get());
    Uninitialize();
     5) Right click and bring up Project properties.  go To Linker->Input.
     6) Set the Configuration and Platform dropdowns to change all configurations all platforms at the same time
     7) Choose Edit for Additional Dependencies and set to this:
    $(SolutionDir)$(PlatformTarget)\$(Configuration)\BackEnd\PhoneVoIPApp.BackEnd.lib
    WindowsPhoneCore.lib
    RuntimeObject.lib
     8) Under Cofniguration Properties->General Set the Target Name to PhoneVoIP.BackEnd.HostServer and Output directory to $(SolutionDir)UI
     9) APply the settings. Then click OK
    10) Build the Proxy Stub
    11) Build the Agents
    12) Build the new Project
    13) Expand the UI (Windows Phone Silverlight 8.1) Project and find the PhoneVoIP.BackEnd.HOstServer.exe and remove it.
    14) Right click UI and Choose Add Existing and add the PhoneVoIP.BackEnd.HostServer
    This should hopefully get you moving forward.  Please test and make sure your application is working. The end result should pass WACK now.  Please let me know if there are any issues.  It is a lot of steps and I hope I didn't miss any.
    Bret Bentzinger (MSFT) @awehellyeah

  • Submit: Great Skype for Windows Phone ideas

    We have just launched a new Windows Phone idea board to let you all share and exchange your ideas on how to improve and develop Skype for Windows Phone.
    Before you start sharing your ideas, here are some guidelines to help you get started:
    Idea Submission Guidelines
    Search before posting. Do a quick search and make sure you are not submitting a duplicate. If the idea already exists show your support by giving kudos to the original idea rather than watering down the potential impact across multiple submissions.
    Draft your idea. Be as clear as possible, tell us what your idea is, how it would benefit users and why you think we should consider it. That will not only help us to better understand what you want, but we may even be able to solve your problem more quickly another way. Please avoid text speak and keep the idea short and to the point. Ideas are available within a limited number of topic sections of the community as they are not discussion threads.
    Currently you can choose one of the following topic labels for your ideas:
    Video Messaging
    We‘ve just launched Video Messaging on Windows Phone so you can now send video messages to family and friends, even when they’re not online. Let us know what you think of Video Messaging on Windows Phone and any ideas you have for how we can make it better. 
    Phone Integration
    Tell us how you would like Skype to be integrated with Windows Phone. Would things be easier if Skype was better integrated with the People Hub? Integrated into the messaging app? Or in the dial pad? 
    Contacts
    How can we make it easier for you to find friends and family on Skype and add them as a contact? What are your thoughts on how contacts are managed on Skype for Windows Phone? Do you have any recommended improvements?
    Calling
    Tell us how you would like Skype to be integrated with Windows Phone. Would things be easier if Skype was better integrated with the People Hub? Integrated into the messaging app? Or in the dial pad? 
    Instant Messaging
    Video calling is one of Skype’s most popular features. What do/don’t you like about Video calling on Skype for Windows Phone and do you have any ideas for how we can make video calling even better? 
    Some things to include:
    Start with a clear title
    add appropriate label(s)
    list features and benefits
    Review the ideas of others. Look at other ideas to see what your fellow community members are requesting and feel free to exchange on them in the comments section. While we take into account the numbers of kudos an idea receives, we also consider the amount and quality of member feedback when making our decisions.
    Add value. Vote for those items that you like and agree with by giving the idea a kudos. It's really a significant way to recognize someone's effort. We are using votes as a barometer to find out more about what our contributor community thinks is important, so we can factor this into our prioritization process.
    Submit a new idea. Adding an idea is just like posting a message on a board. To create an idea, click on the “New Idea” button, located on the left side of the page content tab, and in the left column under the actions heading click “create an idea”. Be sure to enter a clear title and description of your idea so others can decide how they would like to vote. Remember, the more complete your thought, the more likely it is to get noticed! You can use simple HTML, format the text, or add links or images. You can also preview your idea and check your spelling before you post the idea.
    Do not be discouraged. Sometimes what seems simple may not be so simple to build. Some ideas take longer than others, and some ideas will not be possible within the scope of what we want to accomplish in the next year or so.
    Be respectful. Comply with the Community Guidelines and Skype etiquette.
    Disclaimer: Skype may or may not use your comments, ideas, or suggestions. Skype reserves the right to use for any purpose including commercial purposes, comments, ideas, suggestions, code, products or services (“Content”) that are posted on this community without any obligation to compensate the individual or entity that provided the Content. By posting any Content in the community, you hereby relinquish any right you may have in such Content and to any future compensation for publication, use, distribution, license or sale of the same. By posting Content, you represent and warrant that you either own or control all of the rights to that Content, and such Content does not violate any third party rights or these Skype Terms of Service. You agree that all opinions expressed by users of this community are expressed in their individual capacities, and not as representatives of Skype.
    Frequently Asked Questions
    What is the Idea Board all about?
    Idea Board is a new part of the Skype Community, where members can share their ideas about existing products and services. Anyone in the community can see and vote on the ideas you post.
    We encourage you to give kudos and comment on submitted ideas as we are more likely to take the top kudoed and most discussed ones into consideration.
    How do I navigate the Ideas Board? 
    Finding your way around the ideas board is a pretty straightforward task, here are some of the things you will see to help you get around.
    •  New Ideas – The most recent ideas.
    •  Hot Ideas – The most popular ideas right now.
    •  Top Ideas – The most kudoed ideas.
    The sidebar widgets
    In the sidebar you'll find more ways to sort the ideas board, see new and popular ideas and the top rated contributors to the idea board. It's also really easy to view only the ideas with certain status', for instance if you only want to see ideas that are “Accepted”.
    How to check if an idea exists already?
    Use the search function filtering by the ideas board – or click here (direct link to search function)
    What do the votes/kudos mean?
    Just as you can give kudos in the community boards and blogs, you can vote for ideas you like on the Idea Board. Ideas with high vote totals are more likely to be recognized and accepted by the teams at Skype who develop our products and services. You can also leave comments on ideas, but remember to be kind!
    What do the different idea statuses mean?
    We strive to review submitted ideas on a biweekly/monthly cycle. Each idea will go through a life cycle that will be assigned statuses at each milestone. The statuses are explained here:
    New Idea - Sparkly new idea, submitted by you, and ready to be vetted by the community.
    Comments Requested - Items set as Comments Requested will be open to the community for feedback and votes.
    Under consideration - Idea has been forwarded to appropriate business team who is determining feasibility.
    More information needed - More information is needed before our Product Management team can take this forward, please review the comments left by the team.
    Duplicate - Ideas in this status have already been submitted in some form or another. We will link these items over to the other idea so you can review the status or track the progress there.
    Already exists - This is available in the product. 
    Future possibility - Idea is a good one but not something we can include in a near-term product or software release.
    Accepted - Idea has been accepted and has been taken forward for implementation, check back soon for updates.
    Not right now - We like the idea but just not for right now, we will revisit it in the future.
    Implemented - You asked for it, we did it!
    My idea has been accepted. How long will it take to be implemented?
    Sadly, we will not be able to provide dates for the implementation of any accepted ideas within the community due to the number of factors involved. Please be patient and let us make sure it is completely ready before we make it available.
    How will I know if an idea has been implemented?
    You can track the status of your ideas as your ideas are sent to the people here at Skype who can make them happen and you will see the status of your idea is updated. We will be telling you if the idea is something we are working on now or would like to do in the future, or if it’s something that is not feasible for us at this time. To ensure you don’t miss any updates select “Subscribe” from the “Idea Options” menu when viewing your idea.
    What ideas have been implemented so far?
    You can check "Accepted and Implemented ideas" topic here.
    Can I edit or delete my ideas comments?
    No, you can't. Be sure to check your spelling and preview your comment before you post it; you can't edit a comment once it's posted.
    Who and how often moderates/ manages the ideas board?
    The community team is monitoring the board on the daily basis and product managers will come in on a monthly basis to update statuses of the ideas and post their comments.
    If Skype decides not to pursue my idea, will you tell me why?
    We do not always provide the reasons for our decisions regarding ideas submitted for review but will strive to provide an explanation where possible.
    I have an idea to improve the Windows Phone software itself?
    Microsoft has set up a Windows Phone Feature Suggestion Box for exactly that purpose. Share your idea here: http://windowsphone.uservoice.com
    If I am not comfortable submitting my idea over the Internet, is there another way I can do so?
    No.

    I would really appreciate if you guys work on the feature “ custom backgrounds for chat notifications”.If users will be able to choose any image of their liking in the custom background feature if would add value. Currently some of the chat messengers provide this facility to the users which makes interesting to the users while improving user friendliness and satisfaction. Options that would include areTake a photoChoose from albumsSearch from webDeleteThat appears to be a reasonably robust selection. In search, a user can use a built in engine to pull up images based on search and then select it as it their choice. Once you chose an image, it will be asked to confirm and ‘set’ as the universal chat window for all contacts. 

  • I have a hard time understanding why I can't contact support about YOUR problem. I plugged my iPhone into my pc. iTunes came up. It said there was an update available for the phone. I accepted that. Clicked on the "download only" because I didn't have the

    I have a hard time understanding why I can't contact support about THEIR problem. I plugged my iPhone into my pc. iTunes came up. It said there was an update available for the phone. I accepted that. Clicked on the "download only" because I didn't have the time to install it at that point. It downloaded but didn't stop. It continued to the 'backing up your iPhone' segment then froze. I waited at least an hour with no change. Finally I unplugged my phone..iTunes wasn't responding.I rebooted my pc then tried the same thing again. Same outcome even though I let it "Back Up" for 7-8 hrs while I slept. Same Story...............         iTunes 10.4.0.80 is the software I'm using. When my phone is 'plugged' in what I see on it's face is "Sync in Progress".....for hours!

    Error 1611
    This error typically occur when security software interferes with the restore and update process. FollowTroubleshooting security software issues to resolve this issue. In rare cases, this error may be a hardware issue. If the errors persist on another computer, the device may need service.
    Synced media like apps and music are not included in the iPod backup that iTunes makes. You can redownload iTunes purchases by:
    Downloading past purchases from the App Store, iBookstore, and iTunes Store

  • Paid $60 - for a phone number rather than a subscription? Please chime in and help a newbie

    Hi, first a quick rant: I'm an EXTREMELY frustrated new user. I have spent the last hour trying to figure out what I just purchased and what I can do to get an explaination, but whoever created this website did so without a new user in mind. Quickly: I need help trying to determine what I just purchased and what to do from here. As Quickly as I can explain what's going on: My friend in Croatia told me he spent $60 for unlimited calling. He can apparently call anywhere to phones - as we made several calls to landlines in both the states and england from his computer. We've since left his place. As I'm traveling to many countries, I figured this was a great idea, I'll pay up front instead of purchasing credits. I whipped out my credit card and paid $60 after chosing a number. After about an hour, the transaction cleared on my account with a renewal in 12 months. I went to make a call and it said I didn't have enough credit. After trying to figure out what happened all I can determine is that I might have paid for the use of a phone number for 12 months and not a subscription. Now of course after trying for several hours trying to find out how to contact Microsoft or Skype - or Whatever help center - for the past hour and finding they deflect all help to their help pages that comes nowhere NEAR helping me or explaining what has happened - and if I made a mistake how to correct it. I apparently need to spend more money for a premium account to get actual advice or any kind of resolution. Any company that takes money and expects a new user just to GET it, without providing some human interaction is a horrid company. So I paid for something I didn't want and can't use. I THINK. But I can't request a refund or help because I haven't paid for a premium account. Can anyone shed some light on this? Thanks so much; I realize you're probably just a user of the service like myself.

    Skype Rip Off people on Renewal $30 Number $30 Calling Plan Why Skype and Microsoft Sucks: My Subscription Order No.[removed for privacy] paid and delivery but not working I would like to know why the subscription I bought is not available I can make phone calls please fix the problem Skype is not a solution but a big problem since microsoft took over very disappointing
    Yes they told me I will have to pay $30 for my number and $30 for my 12 months subscription now they said they did not they are liarsMy subscription is not working neither after several months tried to fix the problem no one from skype neither from Microsoft tried to contact me is very painful to paid skype for a phone services and they just do nothing but laughing in your face Skype and Microsoft let me tell you something in your face you both sucks big time you are nightmare what a pityI would love to sue skype since they are not providing with the service I contract with then my business is suffering for negligence in delivery my subscription that I bought but is not working long time ago I will follow a class action against skype please keep updatedPORT A SKYPE NUMBER http://community.skype.com/t5/Skype-Number/Port-Skype-Online-Number-to-another-provider/td-p/1529827https://consumercomplaints.fcc.gov/hc/en-us/articles/[removed for privacy]-Keeping-Your-Telephone-Number-When-Changing-Service-ProvidersWhen all is said and done Skype is a phone service provider and as such they must allow your number to move. It is required under Section 212 of the Small Business Regulatory Enforcement Fairness Act of 1996. The SBREFA '96 required portability in the largest 100 Metropolitan statistical areas. Then in 2003, the requirement was expanded to ALL carriers regardless of geographical location.
    In order for a number to be ported, roughly speaking you must live in the area code's geography. So, you can't port a New York number to Iowa.
    If you have a problem porting your phone number from one service provider to another, first try to resolve it with the service provider. If you cannot resolve the problem directly, you can file a complaint with the FCC. There is no charge for filing a complaint. You can file your complaint using an FCC online complaint form. You can also file your complaint with the FCC’s Consumer Center by calling 1-888-CALL-FCC (1-888-225-5322) or 1-888-TELL-FCC (1-888-835-5322) for TTY; faxing 1-866-418-0232; or writing to the Federal Communications Commission at:
    Federal Communications Commission
    Consumer and Governmental Affairs Bureau
    Consumer Inquiries and Complaints Division
    445 12th Street, SW
    Washington, DC 20554

  • Word processing for MacBook Pro

    I want to buy word processing for my MacBook Pro. (Please keep in mind I don't know much about computers and buying things that have to be installed scares me that I'm going to ruin the computer!) The computer is one year old and did not come with word processing.
    Here are my concerns:
    1. I think I need to go to the Mac Store and buy Pages, Numbers, and Keynote all separately, right. I can't just buy iWork and get it all together cheaper?
    2. It is called Pages '09. So this is 2013. I imagine any day now they are going to come out with Pages Some-other-year and I'll have an outdated word processor in a month. Is there an update planned? How does one handle this? Would I have to update for another fee? Would my Pages '09 no longer work?
    3. So, after I fork over my money, what is going to happen? How complicated is this? Will I need a password? Can I just press a button and it will install automatically? Or are there multiple steps after purchasing? (Years ago I bought something for my desktop, they told us we didn't have enough memory, we tried to install more memory and ruined the computer. This is why I am so hesitant. Please be patient with my fears.)
    Thank you,
    Karen

    Apple has made buying, installing & updating their software about as easy as possible.  Certainly someone in an Apple Store can help you and that may be the most comfortable route for you - but you would need to bring your Mac into the store.  On the other hand, if you purchased AppleCare, you could call them and they will help you on the phone and you won't have to take your Mac into a store.
    Regardless of whether you do it on your own or go to an Apple Store, you will purchase Pages, Numbers & Keynote through the Mac App Store (online).  You will need an Apple ID - you may have created one when you started up your Mac the first time after you bought it.  If you created one then, that would be the best Apple ID to use instead of creating another one.
    If you don't have an Apple ID, you can create it online when you open the Mac App Store (it's an application on your Mac).   Once you sign in to the store, a few clicks and $60 later you will be able to download & install all three applications.  The Mac App Store app will do this for you.  Later on, when there are updates, the Mac App Store app will indicate there are updates.  A few more clicks and your updates will automatically download & install.

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