Why am I dropping calls on my I phone 5s?

I have dropped more in a week with the 5s the in 2. Years with the 4s.

if it works with headset connected then it's stuck it's a hardware issue
https://discussions.apple.com/thread/3896498?start=0&tstart=0

Similar Messages

  • Why am I dropping calls and only get 3G?

    I switched to Verizon because my previous provider stopped covering my area. Now I'm having problems with Verizon dropping calls and not having internet at home. Any idea why the service has gotten worse?
    galaxy s3
    66072

        @local101,
    Thanks for the additional details! I show you're in an extended area and may see the service fluctuate between 3G and 4G. Verizon Wireless radio frequency (RF) engineers and system performance engineers regularly evaluate network coverage and performance to identify locations that may require additional coverage. Based on their analyses and recommendations cell site construction or other improvements are then planned and budgeted accordingly.
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    JohnB_VZW
    Follow us on Twitter @VZWSupport

  • 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:
    Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
    These scenarios do not work:
    SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
    SIP phone calls Cisco phone, which transfers the original call to another SIP phone
    I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem.  I am just unable to understand what the trace means:
    10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
    10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
    10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
    10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
    I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
    G711U
    G711A
    G722
    ILBC
    GSM
    and our SIP software is  also set to accept the first codec offered by the remote side.  It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
    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
    a=ptime:20
    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
    c=IN IP4 10.10.10.104
    t=0 0
    a=ptime:20
    a=recvonly-------------------------------------a=recvonly
    Finally Point 7 above..
    +++++++++++++=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
    +++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
    /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
    s=yasdjip
    c=IN IP4 10.10.10.104
    t=0 0
    m=audio 16574 RTP/AVP 9 101
    a=rtpmap:101 TELEPHONE-EVENT/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    +Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
    (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]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 237
    v=0
    o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
    s=SIP Call
    c=IN IP4 10.10.10.137
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 20352 RTP/AVP 9 101
    a=rtpmap:9 G722/8000
    a=ptime:20
    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
    s=SIP Call
    c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
    t=0 0
    m=audio 4000 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=inactive---------------------------------------------------------------------media inactive
    At this point, we should get a response back from the sip phone...
    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:
    for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
    SIP------Media------MTP------------Media-------SCCP Phone
    When the new destination is dialled and transfer is commited,
    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"

  • Constantly dropping calls no matter the phone

    All Verizon phones constantly drop calls in my house.  Signal constantly bounces from 2-4 bars when the phones aren't moving.  The only place I can make calls without dropping signal is if I step out in my back yard.  How do I put in a ticket or have someone check this out.  The problem has been here since I moved in 2 years ago, however it's progressively gotten worse.  Phones are HTC Winphone 8x and Samsung Brightside, but family's Droid X2, Iphone 4, Iphone 5, LG showtime, and a couple other basic phones all have same issue.

        Hi mtpica112471!
    I apologize for the inconvenience because I realize the importance of operable service! Let's look into this. Current zip code? There can challenges present with indoor coverage.
    Thanks,
    AyaniB_VZW
    Follow us on Twitter @VZWSupport

  • Why is Adobe dropping support for pre 3GS phones?!

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    Apple have dropped support for pre-3GS because to pull of the features and performance in iOS 4.3 they had to develop just for the ARM7 chip, and not for the ARM6. They probably have a good sense of the number of people that have the older devices, and think it's ok to not update the software on those devices.
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    I am trying to find out how to make AIR 2.6 and AIR 2.0 live together in Flash CS5. If I work it out I'll let you know.

  • Why cant i hear calls on my i phone 5

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    if it works with headset connected then it's stuck it's a hardware issue
    https://discussions.apple.com/thread/3896498?start=0&tstart=0

  • Drop calls on landline phone

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    Your calls should not be dropping like that. You can reset the ONT box by using the following steps...
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    Anthony_VZ
    **If someones post has helped you, please acknowledge their assistance by clicking the red thumbs up button to give them Kudos. If you are the original poster and any response gave you your answer, please mark the post that had the answer as the solution**
    Notice: Content posted by Verizon employees is meant to be informational and does not supersede or change the Verizon Forums User Guidelines or Terms or Service, or your Customer Agreement Terms and Conditions or plan

  • Why i cant make calls even if i had balance and strong signal on my phone

    hi apple
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    pleasee please help me with this matter.
    thanks

    try removing battery for ten mins then try a soft reset the code is *#7370# security code 12345 back up any data you wish to keep such as contacts etc
    before that may i ask have you removed the sim card or did the phone get a knock have you contacted your operator?
    if you have done so try the code above .
    Message Edited by jimmyireland on 22-Jan-2010 07:18 PM
    If  i have helped at all a click on the white star below would be nice thanks.
    Now using the Lumia 1520

  • Why do I keep dropping calls

    I have a Iphone 4 why after talking for about five min. I loose my call

    mummyof2 wrote:
    Why do I keep losing service/dropping calls with every outgoing call it's not my location it happens where ever I am. It will not let me call out even to customer service or 611. It also will go straight to voicemail when someone calls me, but low and behold I cant even call my voicemail. This is ridiculous and has been happening for almost 2 weeks. What if I have an emergency and need to call out or someone is trying to reach me, that's the whole point of my cell phone. Please help I'm very unhappy! I have a galaxy note 2Hi , Thank you for your post. We're happy to help. Were there any changes to your device when this started happening? Do you have your device in Airplane Mode? Please check to see if you're running the most current software version through Settings>About device>Software update>Check for updates. Are you hearing a recording when you try to make a call? Check in Settings>Call settings>Additional settings>Call barring>verfiy that All outgoing calls is unchecked. Also, make sure that your SIM card is inserted correctly and not damaged. We have some additional steps to try. First let's try a Soft Reset by removing your battery for 2-3 minutes. This will re-optimize your device. You may also try running your device in Safe Mode for 2-3 hours: http://bit.ly/1vkcs06 . This will only run stock applications to see if any third party apps are causing this. Let us know if you still need assistance! Thanks, Ema with Samsung Support

  • Why does my iPhone 4S drop calls with good service

    Why when I have 3 or more bars does my iPhone 3S drop calls?

    Go to Settings > General > Accessability > Incoming Calls then set to default.

  • Why does my iphone keep dropping calls

    Why does my iphone4 keep dropping calls??? It has been doing it for over a week. Does anyone have an answer.... AT&T says they are not having any problems and I have done everything but restore my phone.

    Unfortunately, this has plagued the AT&T iPhone since the original model came out. It's most likely a dead zone causing the issue (I have the sameissue from time to time). You can also get a Micro Cell from AT&T, it's basically just an access point that allows you to pick up a cellular signal at home or work (where ever you set it up).
    To eliminate any hardware issues, make sure you're on the latest build of the OS (4.3.3 for the AT&T iPhone 4) - I noticed you're still on 4.3.2. If you don't have a case, I suggest the Apple bumper or any case that covers the metal bezel on your phone, that will help as well.
    Also, you can "reboot" your phone. To do this, hold down the metal sleep button on the upper-right corner of your phone until the "slide to power off" slider is displayed, then power off the phone. Wait a few minutes and turn it back on.
    Hope this helps!

  • Dropped calls and no service in my house...why am I even paying a phone bill?

    I've been a Verizon customer for probably around 6 or 7 years now. I've had plenty of different devices and was always happy with my phone. A few years ago, service started to get horrible in my house. I would barely get service and my phone would drop calls. I have spent hours over the past 2 years on the phone with reps. Some were very helpful, and some were very frustrating. I have put in service tickets and even changed out my phone to try to get better service. Let me also note that AT&T customers get perfect service inside of my house. I don't even want to hear the excuse that Verizon doesn't guarantee service inside of a building due to building materials. The last time I checked, I didn't rebuild my house in the past few years with new materials. I use my cell phone for business when I'm at home, so it is extremely important that it works inside my house. I put my most recent service ticket in almost 2 weeks ago and there is STILL no resolution. I also won't accept if the only resolution is me buying a network extender.. I'd rather pay an early termination fee and leave before I'd ever spend money on another Verizon device. I have had 2 reps in the past week tell me that they would call me back and I never hear anything. I also have been told that they would resolve it by a certain date, only to find out that now they are saying it's going to stay open until they can figure it out. I rely on my phone for work now more than ever since I recently starting working from home, so how can I accept the fact that I am waiting around and paying for a phone that doesn't work when I need it the most? For all of my trouble, they offered me a $20 credit on my account. I shouldn't even use the word "offered" because I was the one who mentioned that I should be reimbursed for my time. I'm so sick of wasting my time on the phone and I am truly disgusted with Verizon. As a long time customer who has spent thousands of dollars over the year paying for Verizon service, I would expect to be treated much better than this.

    I am not in the same boat, Verizon works absolutely everywhere I go and I am super happy with the service. If the service started not working however, and I got no resolve, I would just cancel it and switch providers. No point in paying for service you can't use, especially where you live.

  • Why am I getting dropped calls every since I upgraded to IOS 7?

    I upgraded to IOS7 and I have been having dropped calls all the time.  I talked to Verizon and we went through a number of things to fix it but it did not work.  Has anyone else had this problem? If so what have you done. 

        I know how important service is for you and your wife olehippy52 and I'm sorry you are now having issues at home. I also want to apologize for the experience you had at the store location. The information given to you should have been provided with some investigation and a better explanation. We can look into the area issues especially if you had service before and now have problems. What is your zipcode? What two phones do you and your wife have? You can usually dial *228, send and option #2 to update 3g phones. You can reach out to us here via private message, by phone to talk with tech support, chat, or social media. Once we speak with you, we can get started with an investigation to get this fixed and keep you with our family. We look forward to hearing from you soon.
    KinquanaH_VZW
    follow us on Twitter @vzwsupport

  • Dropping calls like crazy..why!!??

    I normally have great service at home..but the past few days, I'm dropping calls like crazy..have to go outside and stand in the driveway. NOT convenient! Also I keep losing my wifi when i go to the other end of the house, it starts using my 3G, never had that issue before either. Is there something I can do? Beyond frustrated....

    To power cycle device press the power button until you get the slider on screen to power device down....  Once its powered down press the power button again to power device back on..   Also you can press power button and home button to do a hard reboot.
    When it is reported to retest your device all you have to do is simply use the device like usual and see if the issue continue.

  • 4.3.2 iPhone upgrade and dropped calls.

    Since upgrading to the 4.3.2 OS three days ago, I've been experiencing numerous dropped calls on the order of one every minute.  Calls linger in limbo before connecting and then drop.  Antena signal reads 3 bars.  I'm missing calls too.  I live in the same area and have NEVER had a problem with dropped calls until this update and it is really causing problems.  Why?  Is the phone failing?  Is this an update problem?  Is this a network problem?  Is anyone else experiencing this?

    I am experiencing the same issue here in Germany, Network is T-Mobile.
    Since upgrade to 4.3.2 I have dropped calls regularly with my iPhone4!!
    This is getting more and more annoying.
    What happens is: I am starting a call or am receiving a call. The call stays up for 1-5 minutes. During that time (randomly differs from 30 seconds to 5 minutes) all of a sudden the person on the other side does not hear me anymore! I am still able to hear him/her! After around 5-10 seconds after that happening, the connection is lost/cancelled. This is happening independently of the amount of bars, meaning it does not matter whether I am having 5, 3 or just one bar.
    Please Apple solve this!! I cannot use my phone anymore the way it is supposed to work! Never had this issue before the upgrade in the same places - in fact, before the upgrade, I did not experience ANY dropped call whatsoever!!
    BTW: I tried reboot, reset of all settings, backup, etc. - no positive effect.

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