Tried to Cancel, Convenient​ly Dropped Call, Another RIDICULOUS Bill

I called on June 19th to cancel my service. My line (which had no phone attached to it) kept calling 911 and directory assistance... My bill is normally $55, and ended up $190. The next bill was $119. So after being on the phone and sent to several different representatives who kept "transferring me to THAT department," I finally got to someone who could honor my request.
It wasn't until threatening me with imagery of what post-apocalyptic life would be for me without a landline and me repeatedly saying, "just cancel my service," that she agreed to put in my cancellation request.
Conveniently, my call was dropped immediately as those words left her lips.
Now, correct me if I'm wrong, but when she first got on the line and requested that I confirm my mobile number in case our call was disconnected, that meant a disconnection would result in a call back from that agent.
Nearly a month later, I've had no response from that representative. In fact, the only contact I've had is the generic, "Thanks for choosing Verizon! Here are some helpful links for future use of your service!"
The service I called to have CANCELED.
So, out of spite, I've been waiting for a call regarding my account. Since I never received one, I assumed my final bill was being mailed to me. I can handle not being spoken to when I'm canceling my service.... but I figured the bill was coming.
NOPE. Just received the email notification that my bill is available to view! And guess what? It's not my final bill, but again, my $55 bill is now $128. For an account that should have been shut down.
So, not only do I feel that Verizon is conveniently ignoring my request as to gain more money from me for some sort of problem in the wiring... BUT, as has been the case throughout this entire ordeal, I have the police beating on my door from the hours of 1:00 AM to 5:00 AM.
I fully expect a response regarding this post by someone with some kind of authority regarding my issues. I am being billed for items that are absolutely NOT my doing (as I have NOTHING plugged into my walls TO make it fire off and call 911 and directory assistance) and I've already been put in a position to pay these astronomical bills because not doing so will only reflect on my credit score.
To say I'm disappointed in the service I've received from this company is an understatement. I am highly offended by how I was treated and how I've been completely written off just so it can make another dime.

Hello leilanilujan,
We'll be closing your private support case for the time being since we have not heard from you for an extended period of time. If you still require assistance, please make a public posting in this thread and we'll be more than happy to begin working with you again.
Thank you,
Art

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  • CUCM 8.6 Dropped call transfers involving SIP phones

    Hi All,
    I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone.  This scenario works:
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    I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492).  The trace snippet shown above is from this log.
    I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it.  I can try to provide more info about our CUCM configuration if needed.
    Thanks in advance!

    Leslie, so here is what I found from the traces....
    To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
    SCCP
    When the transfer key is pressed
    1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
    NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
    2.CUCM tells the held party to listen MOH from MOH server
    3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
    4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
    5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
    6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
    7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
    For SIP signalling. when the first transfer key is pressed
    1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
    2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
    NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
    3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
    4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
    5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
    6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
    7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
    Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
    lets look at succesful SCCP-----SCCP-----SIP without MTP
    Point 4 above
    ++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
    (0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
    Point 5 above
    ++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
    (0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845.  myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
    Point 6 Above
    +++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
    (0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847.  myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
    (0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847.  myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
    Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
    +++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
    [885626,NET]
    INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 0.0.0.0
    m=audio 24560 RTP/AVP 9 101
    a=rtpmap:9 G722/8000
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    a=inactive-----------------------------------------------------Inactive
    Still part of Point 6 for SIP signalling
    ++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
    //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
    [885628,NET]
    SIP/2.0 200 OK
    v=0
    o=- 18077 11099 IN IP4 10.10.10.104
    s=yasdjip
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    a=recvonly-------------------------------------a=recvonly
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    +++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
    [885630,NET]
    INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
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    /SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
    [885634,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
    Contact:
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    Call-ID: [email protected]
    v=0
    o=- 18077 11099 IN IP4 10.10.10.104
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    m=audio 16574 RTP/AVP 9 101
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    (0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
    (0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
    remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
    +++++++++++=Next Phone sends its ACK+++++++++++++++
    (0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
    +++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
    [885635,NET]
    ACK sip:[email protected]:62220;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
    From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
    To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
    Date: Tue, 19 Feb 2013 21:44:45 GMT
    Call-ID: [email protected]
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    CSeq: 103 ACK
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    o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
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    m=audio 20352 RTP/AVP 9 101
    a=rtpmap:9 G722/8000
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    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Now at this point all is well...and the call is connected....
    Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
    From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
    [881160,NET]
    ACK sip:[email protected]:53361;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
    From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Date: Tue, 19 Feb 2013 17:38:50 GMT
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    Max-Forwards: 70
    CSeq: 102 ACK
    o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
    t=0 0
    m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=sendonly---------------------------------------------------------sendonly
    +++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
    [881161,NET]
    INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
    From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Date: Tue, 19 Feb 2013 17:39:04 GMT
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 164
    v=0
    o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
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    c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
    t=0 0
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    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=inactive---------------------------------------------------------------------media inactive
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    and here is what we got..
    ++Trying which is expected++++
    //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
    [881162,NET]
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
    From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    CSeq: 103 INVITE
    To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Content-Length: 0
    ++++++++Then we get a BYE+++++++++++++++
    /SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
    [881163,NET]
    BYE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
    Contact:
    Max-Forwards: 70
    From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
    Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
    Supported: replaces, path
    User-Agent: Acrobits Softphone Business/2.4.8
    To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
    Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
    CSeq: 3 BYE
    Content-Length: 0
    So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
    The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
    Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
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    SIP------Media------MTP------------Media-------SCCP Phone
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    SIP-------------media----MTP--------media---------MTP
    The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
    ++++++++Ivite to 492 ++++++++++++++
    INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
    From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
    To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
    Date: Tue, 19 Feb 2013 21:24:59 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 102 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 214
    v=0
    o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
    t=0 0
    m=audio 25038 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    +++++++Invite to 491 +++++++++++++++++
    //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
    [885429,NET]
    INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
    From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
    To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
    Date: Tue, 19 Feb 2013 21:24:59 GMT
    Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 237
    v=0
    o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.195----------------------------------------MTP
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 25030 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Wao! That was a long one isnt it...It was fun too.
    So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
    Please rate all useful posts
    "opportunity is a haughty goddess who waste no time with those who are unprepared"

  • IPhone 5 drops calls as fast as my confidence in Verizon

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    >>Personal information removed to comply with the Verizon Wireless Terms of Service<<
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