Unable to call skype to skype

First of all, Ive been trying for two days to get help with no success.  I have been unable to call skype to skype for several weeks why?  I have no problem calling from my tablet to my family's  land line why is this happening

Up and running now on mobile. THANK YOU for resolving the issue.

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    please contact Skype customer service
    TIME ZONE - US EASTERN. LOCATION - PHILADELPHIA, PA, USA.
    I recommend that you always run the latest Skype version: Windows & Mac
    If my advice helped to fix your issue please mark it as a solution to help others.
    Please note that I generally don't respond to unsolicited Private Messages. Thank you.

  • Unlimited World - unable to call landlines (AUSTRA...

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    Go to Solution.

    Are you calling domestic landlines (not business landlines)?
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    TIME ZONE - US EASTERN. LOCATION - PHILADELPHIA, PA, USA.
    I recommend that you always run the latest Skype version: Windows & Mac
    If my advice helped to fix your issue please mark it as a solution to help others.
    Please note that I generally don't respond to unsolicited Private Messages. Thank you.

  • Unlimited World subscription but unable to call co...

    Hi,
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    Hi,
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    Assinaturas / Skype Premium / Contacto com atendimento ao cliente da Skype /
    Se a minha resposta foi útil, dá Kudos ↘. Obrigada! Eu posso não responder às mensagens privadas não solicitadas.
    Si mi respuesta fue útil, dé Kudos ↘. Gracias! Puedo no responder los mensajes privados no solicitados.

  • Snom phones in secondary subnet unable to call out - SIP CANCEL in SIP log

    I've been trying to diagnose this very strange problem we are having. All our servers and some SNOM phones are in the subnet 192.168.100.0, the main building. They all work fine. Phones located in two other buildings connected with high-speed fiber use subnets
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    We are running Lync Server 2013 Standard Edition, with the latest updates applied. Mediation role is co-located. Edge server is setup and I think I have configured everything correctly. I have two network adapters, one external facing and one internal facing.
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    entry is routable lync2013edge.network.domain.ca). Phones used are the SNOM 720, I have the latest updates applied (8.8.3.27 UC)
    On the actual SNOM phone, I will dial 7804636201. It will call and start ringing the other party. Almost exactly 10 seconds later I will hear a busy signal and then the phone displays "Media Connectivity Failure". I ran a log on SIP from the FE
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    TL_VERBOSE(TF_PARSE) [0]411C.2DE8::02/24/2015-17:23:42.240.0008db5d (SIPStack,CSIPMessage::ParseBufferChain:SIPMessage.cpp(694))( 0000005F03D806F0 ) Start Line:  INVITE sip:7804636201;[email protected];user=phone SIP/2.0
    TL_INFO(TF_PROTOCOL) [0]411C.2DE8::02/24/2015-17:23:42.269.0009106f (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[3706963737] $$begin_record
    Trace-Correlation-Id: 3706963737
    Instance-Id: 2F91
    Direction: outgoing
    Peer: lync2013.network.caedm.ca:5070
    Message-Type: request
    Start-Line: INVITE sip:[email protected]:5070;user=phone;maddr=lync2013.network.caedm.ca SIP/2.0
    From: "Joel Smith" <sip:[email protected]>;tag=2ksjs48fxg;epid=000413774E0401
    To: <sip:7804636201;[email protected];user=phone>
    Call-ID: 3faa35f677ef48719b27c796251b0519
    CSeq: 1 INVITE
    Contact: <sip:[email protected];opaque=user:epid:cO0WSS9wCFqUnP0dpEh6uQAA;gruu>;reg-id=1
    Via: SIP/2.0/TLS 192.168.100.17:55489;branch=z9hG4bKE036484A.405BD23C943B158E;branched=TRUE
    Via: SIP/2.0/TLS 192.168.1.201:51470;branch=z9hG4bK-fdh7rhbbvsri;rport;ms-received-port=51470;ms-received-cid=600
    Record-Route: <sip:Lync2013.network.caedm.ca:5061;transport=tls;opaque=state:T;lr>;tag=B39FB8145D545F357B2737F43833CEB4
    Max-Forwards: 69
    Content-Length: 3563
    Content-Type: multipart/alternative;boundary="next_part_u00iwyrezkkuxf3d"
    P-Asserted-Identity: "Joel Smith"<tel:+17808092404;ext=2404>
    Message-Body: --next_part_u00iwyrezkkuxf3d
    Content-Type: application/sdp
    Content-Transfer-Encoding: 7bit
    Content-Dis; handling=optional; ms-proxy-2007fallback
    TL_INFO(TF_DIAG) [0]411C.2DE8::02/24/2015-17:23:42.270.000915ec (SIPStack,SIPAdminLog::WriteDiagnosticEvent:SIPAdminLog.cpp(802))[3706963737] $$begin_record
    Severity: information
    Text: Routed a locally generated response
    SIP-Start-Line: SIP/2.0 100 Trying
    SIP-Call-ID: 3faa35f677ef48719b27c796251b0519
    SIP-CSeq: 1 INVITE
    Peer: 192.168.1.201:51470
    Data: destination="[email protected]"
    $$end_record
    TL_INFO(TF_PROTOCOL) [0]411C.2DE8::02/24/2015-17:23:42.274.000928d4 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[3706963737] $$begin_record
    Trace-Correlation-Id: 3706963737
    Instance-Id: 2F93
    Direction: outgoing;source="local"
    Peer: 192.168.1.201:51470
    Message-Type: response
    Start-Line: SIP/2.0 101 Progress Report
    From: "Joel Smith" <sip:[email protected]>;tag=2ksjs48fxg;epid=000413774E0401
    To: <sip:7804636201;[email protected];user=phone>
    Call-ID: 3faa35f677ef48719b27c796251b0519
    CSeq: 1 INVITE
    Via: SIP/2.0/TLS 192.168.1.201:51470;branch=z9hG4bK-fdh7rhbbvsri;rport;ms-received-port=51470;ms-received-cid=600
    Content-Length: 0
    ms-diagnostics: 12006;reason="Trying next hop";source="LYNC2013.NETWORK.CAEDM.CA";PhoneUsage="Long Distance";PhoneRoute="LocalRoute";Gateway="208.68.17.53";appName="OutboundRouting"
    $$end_record
    TL_INFO(TF_PROTOCOL) [1]411C.2DE8::02/24/2015-17:23:42.488.000930bc (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[741182734] $$begin_record
    Trace-Correlation-Id: 741182734
    Instance-Id: 2F96
    Direction: incoming
    Peer: lync2013.network.caedm.ca:5070
    Message-Type: response
    Start-Line: SIP/2.0 183 Session Progress
    FROM: "Joel Smith"<sip:[email protected]>;tag=2ksjs48fxg;epid=000413774E0401
    TO: <sip:7804636201;[email protected];user=phone>;tag=d265bdc1c8;epid=0A24894D6D
    CALL-ID:  3faa35f677ef48719b27c796251b0519
    CSEQ: 1 INVITE
    CONTACT:  <sip:[email protected];gruu;opaque=srvr:MediationServer:0wzNMLUTNFKXO5KjW1mbdQAA>;isGateway
    VIA:  SIP/2.0/TLS 192.168.100.17:55489;branch=z9hG4bKE036484A.405BD23C943B158E;branched=TRUE,SIP/2.0/TLS 192.168.1.201:51470;branch=z9hG4bK-fdh7rhbbvsri;rport;ms-received-port=51470;ms-received-cid=600
    RECORD-ROUTE:  <sip:Lync2013.network.caedm.ca:5061;transport=tls;opaque=state:T;lr>;tag=B39FB8145D545F357B2737F43833CEB4
    CONTENT-LENGTH:  1388
    CONTENT-TYPE:  application/sdp
    TL_VERBOSE(TF_NETWORK) [0]411C.2DE8::02/24/2015-17:23:51.369.00098f6b (SIPStack,CRecvContext::CreateIncomingRequest:RecvContext.cpp(920))[3030787245]( 0000005F01E739D0 ) creating SIP_MID_CANCEL request
    TL_VERBOSE(TF_PARSE) [0]411C.2DE8::02/24/2015-17:23:51.369.00098f90 (SIPStack,CSIPMessage::ParseBufferChain:SIPMessage.cpp(694))( 0000005F03D7E2E0 ) Start Line:  CANCEL sip:7804636201;[email protected];user=phone SIP/2.0
    TL_VERBOSE(TF_PARSE) [0]411C.2DE8::02/24/2015-17:23:51.369.00099054 (SIPStack,CSIPMessage::ParseNextHeader:SIPMessage.cpp(1532))( 0000005F03D7E2E0 ) Found Header:  Reason: SIP;cause=488;text="Media Connectivity Failure"
    TL_INFO(TF_PROTOCOL) [0]411C.2DE8::02/24/2015-17:23:51.369.000990c6 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[3706963737] $$begin_record
    Trace-Correlation-Id: 3706963737
    Instance-Id: 2FA0
    Direction: incoming
    Peer: 192.168.1.201:51470
    Message-Type: request
    Start-Line: CANCEL sip:7804636201;[email protected];user=phone SIP/2.0
    From: "Joel Smith" <sip:[email protected]>;tag=2ksjs48fxg;epid=000413774E0401
    To: <sip:7804636201;[email protected];user=phone>
    Call-ID:  3faa35f677ef48719b27c796251b0519
    CSeq: 1 CANCEL
    Via:  SIP/2.0/TLS 192.168.1.201:51470;branch=z9hG4bK-fdh7rhbbvsri;rport
    Max-Forwards:  70
    Content-Length:  0
    $$end_record
    I thought it might be a timeout issue, so I tried following these steps located here:
    http://ipfone.hu/lync-mediation-server-cancel-problem/ After rebooting the server no changes were noticed.
    I also checked out this website
    http://blog.insidelync.com/2013/04/sip-trunking-101-with-lync-server-2013/ regarding disabling the check box "enable outbound routing failover timeout". Doing that had no effect.
    Any other ideas would be appreciated.

    Hi,
    yes I see the config file is very simple and standard.
    So the issue with snom on branch sites is random, it's correct?
    From what I read in your answer, sometimes you can establish a correct communication between a snom and the called number +17804636201.
    Have you tried to collect a network capture on a snom at branch location?
    Do you have some other version of snom phone (300, 710, 821) to test?
    Do you have some LPE ip-phone (Polycom CX600 o HP4110-4120) to test?
    Regards
    Luca
    Luca Vitali | MCITP Lync/Exchange | snom Certified Engineer | Sonus SBC1000 Engineer

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