Constant Buzzing and Can't make outgoing calls

For the past month or so, our phone line has had a constant buzzing. Now, we cannot even get a dialtone to make outgoing calls. We do receive calls, but it doesn't do us much good since the buzzing prevents us from talking to anyone anyway.

I am having the same exact complaint, "Constant Buzzing on line, and inbound or outbound conversation cannot be performed".  I am in southern New Jersey just outside of Philly, and this problem was first recognized approximately September 19th.  I went out of town and felt that the problem would be remedied by my return.  On September 24th the same problem was still present on the line.  I used to be a Computer Network Engineer years ago and have a telephone handheld "butt set" (handheld phone that the phone techs use to connect directly to the wires.  I disconnected all of my premise equipment and connected to only the line coming in from the pole, and the problem still persists which rules out any "interior" equipment malfunction.
Original service appointment was scheduled for Thursday September 29 between 8AM and Noon, however when Thursday arrived, I noticed on the internet that my expected service visit time had been moved back (without any heads-up) to Friday September 30 between 8AM and Noon.  Friday 12 Noon comes and goes with no results yet, and upon looking at the internet again I noticed that the expected service visit time had been moved back once again to Friday September 30 at 8 PM.
Very poor repair service for a "phone not usable" condition.

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    <------------->
    --- (8 headers 0 lines) ---
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    Transmitting (NAT) to 63.209.144.201:5061:
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    Adding codec 100003 (ulaw) to SDP
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    Content-Length: 370
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    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
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    To: <sip:[email protected]>;tag=as3f27fa61
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    CSeq: 101 INVITE
    Server: Asterisk PBX 10.5.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
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    Content-Length: 0
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    <--- SIP read from TLS:63.209.144.201:5061 --->
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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: 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.
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    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 --->
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    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 --->
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    I wound up calling skype support. This is the final sip.conf looks like. Hope it helps. Good luck.
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    allowoverlap=no
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    tlsbinddir=0.0.0.0
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    srvlookup=yes
    dynamic_exclude_static = yes
    buggymwi=yes
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    [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
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