Call Forwarding with Skype Number

I have a skype number and a couple questions. I have searched the forums and did not see this question answered. 
I have an unlocked iphone and travel througout a different country every month or so. Each new country means a new sim card and a new number. 
My question is: Will purchasing the skype world subscription cover the costs of each country (as long as the country is covered in the plan)? It states that the subscription only covers mobiles in some countries and landlines in others. So if I purchase the unlimited world (and assuming I am in one of the countries covered) will this be all I need to receive unlimited forwarded calls to my mobile?
I hope this isnt too confusing. Thanks!

Hi, Salsy20 and Joeriba, and welcome to the Community,
Please check your call forwarding settings; Skype does not determine when to activate call forwarding based upon the number of rings; it is based upon an increment of time.  For example, in the screen shot, below, the default increment is 10 seconds (circled in orange pen).  You would need to determine and then experiment with how many rings = [x] seconds, or set to 0.  Remember, if you make any changes to any settings, click Save!
Here is a link to the library of FAQ articles related to call forwarding:
https://support.skype.com/en/category/CALL_FORWARDING/
Regards,
Elaine
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    Hi, 761w. and welcome to the Community,
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    Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
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    Hi, Jimliu, and welcome to the Community,
    Fabulous idea!  Unfortunately, the Skype Number feature does not provide such facility as you describe.
    Best regards,
    Elaine
    Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
    Trustworthy information: Brian Krebs: 3 Basic Rules for Online Safety and Consumer Reports: Guide to Internet Security Online Safety Tip: Change your passwords often!

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    Subject/title amended from "fault," for clarity.
    thasriq

    Hi, Thasriq, and welcome to the Community,
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    Elaine
    Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
    Trustworthy information: Brian Krebs: 3 Basic Rules for Online Safety and Consumer Reports: Guide to Internet Security Online Safety Tip: Change your passwords often!

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    Please review our Skype Fair Usage Policy
    https://support.skype.com/en/faq/FA10364/what-is-the-fair-usage-policy-for-unlimited-subscriptions?f...
    Your account may get suspended for package abuse. Once the system detected that your account is abusing any of our subscriptions such as Skype Number, you will receive an email telling you that your account has been suspended for subscription abuse. You can continue to make free Skype calls and send instant messages. However, you will be unable to continue using paid products such as Skype Credit, subscriptions and Skype Premium.
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    Did my reply answer your question? Accept it as a solution or give me a Kudo to help others, Thanks.
    If one of my replies has adequately addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others.

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    Hello, I have had my Skype number for 2 years now and never had a problem.However since end of November 2012 calls from the USA are no longer being forwarded to my cellphone in Germany.No , I have not changed my settings , why should I , since it has been working perfectly all these years.There is no way to get help from Skype and I am extremely frustrated with their service.It sucks to pay for a product that won´t work.I am a dissatisfied customer and will leave them if they do not straighten out their stuff.

    watsource wrote:
    Hello, I have been a SKYPE user for the past 5 years and everything has been working fine.  Recently, calls from telephones to my skype number are not being forwarded to my designated number.  I've tried to turn off and on forwarding feature but still not working. Calls from other skype contacts using skype calling me are forwarded but not from any landline or mobile contacts calling my skype number.
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    Patrick
    Location/Ubicacion: Arizona USA
    Time Zone/Hora Local: UTC/GMT -7
    If this message has adequately addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others.
    Si esto mensaje le ha ayudado, por favor haga clic en "Aceptar como solución". Si encuentra un mensaje útil, por favor "Da Kudos" al final del mensaje, por lo que esta información puede beneficiar a otros.
    I am not a Skype employee. No soy un empleado de Skype.

  • Call forwarding with subscription but still tellin...

    I have a subscription but when setting up call forwarding, keeps telling me that I don't have enough credit. Message reads : You need to have some Skype credit to forward calls to mobiles or land lines.
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    jaymen wrote:
    I have a subscription but when setting up call forwarding, keeps telling me that I don't have enough credit. Message reads : You need to have some Skype credit to forward calls to mobiles or land lines.
    so it doesn't let me save the forwarding set up, in my account, Skype aknowledges my subscription
    You didn't mention what subscription you have, and the number you are trying to forward to.  If you know Skype is acknowledging your subscription, it sounds like the number you are trying to forward to may not be covered by your subscription. If you list the number in a reply to this message thread, please replace the last 4 digits of the number with XXXX.  We don't need to see the full number, but everything up to the last 4 digits would be helpful in troubleshooting your issue.
    Note that many subscriptions only cover calls to landline phones, not to mobiles or other special numbers.  Skype offers subscriptions that include calling to mobiles, but not too many of them are of the "unlimited" variety due to the costs for calling mobiles in much of the world. 
    Patrick
    Location/Ubicacion: Arizona USA
    Time Zone/Hora Local: UTC/GMT -7
    If this message has adequately addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others.
    Si esto mensaje le ha ayudado, por favor haga clic en "Aceptar como solución". Si encuentra un mensaje útil, por favor "Da Kudos" al final del mensaje, por lo que esta información puede beneficiar a otros.
    I am not a Skype employee. No soy un empleado de Skype.

  • Call forwarding with airtel carrier

    Hello,
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    thanks

    Hi, Jimliu, and welcome to the Community,
    Fabulous idea!  Unfortunately, the Skype Number feature does not provide such facility as you describe.
    Best regards,
    Elaine
    Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
    Trustworthy information: Brian Krebs: 3 Basic Rules for Online Safety and Consumer Reports: Guide to Internet Security Online Safety Tip: Change your passwords often!

  • CUCM 8.6 Call Forwarding to External Number Issue

    Hello,
    Call forwarding worked without problems, we could forward our phones to external numbers and everything was ok, when somebody called to my phone, I could  got the call to my cell phone.
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    We didn't change configuration or install any update.
    I think its my ISP-s problem, to whom we have SIP Trunk.
    I don't understand log file, so can you tell what is the problem?
    Here is log:
    057729XXXX is called party, cell phone number
    original calling party number is 240XXXXX, but it is forwarded to 2484XXX
    INVITE sip:2484XXX@ISP-IP:5060 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
    From: <sip:057729XXXX@MY-CUCM>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP-IP>
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
    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
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Cisco-Guid: 0383266432-0000065536-0000191815-2219117834
    Session-Expires:  1800
    P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
    Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
    Contact: <sip:057729XXXX@MY-CUCM-IP:5060>
    Max-Forwards: 68
    Content-Type: application/sdp
    Content-Length: 215
    v=0
    o=CiscoSystemsCCM-SIP 4052091 1 IN IP4 MY-CUCM-IP
    s=SIP Call
    c=IN IP4 MY-CUCM-IP
    t=0 0
    m=audio 29790 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    |2,100,56,1.173711429^MY-CUCM-IP^MTP_3
    17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 2|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
    17:34:18.526 |EnvProcessUdpHandler::fireSignal - SEND: index = 2, handler = 0xb2d59c98|*^*^*
    17:34:18.526 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP-IP:5060|*^*^*
    17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1172, ISP-IP:5060)|*^*^*
    17:34:18.536 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
    17:34:18.536 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
    17:34:18.536 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 358 from ISP-IP:[5060]:
    [12623361,NET]
    SIP/2.0 100 Trying
    Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
    CSeq: 101 INVITE
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
    Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
    Server: CISCO-SBC/2.x
    Content-Length: 0
    |2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Info/0x0/ccsip_spi_get_msg_type returned: 2 for event 1|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Transport/0x0/context=(nil)|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Transport/0x0/gConnTab=0xf484290, addr=ISP-IP, port=5060, connid=2, transport=UDP|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Info/0x0/Return existing connection for port 5060 connId 2|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Info/0x0/Checking Invite Dialog|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/Info/0xb1b50c90/INVITE response with no RSEQ - disable IS_REL1XX|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE)  to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
    17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
    17:34:18.561 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
    17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
    [12623362,NET]
    SIP/2.0 403 Forbidden
    Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
    CSeq: 101 INVITE
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
    Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
    Server: CISCO-SBC/2.x
    Content-Length: 0
    Contact: <sip:ISP-IP:5060>
    [12623363,NET]
    ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence
    Content-Length: 0
    INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    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
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
    Session-Expires:  1800
    P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
    Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
    Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
    Max-Forwards: 68
    Content-Type: application/sdp
    Content-Length: 215
    v=0
    o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
    s=SIP Call
    c=IN IP4 MY-CUCM-IP
    t=0 0
    m=audio 29792 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    |2,100,56,1.173711431^MY-CUCM-IP^MTP_3
    17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
    17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
    17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
    17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
    17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
    17:34:18.569 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 0|*^*^*
    17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
    [12623365,NET]
    SIP/2.0 100 trying -- your call is important to us
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    CSeq: 101 INVITE
    Server: kamailio (3.3.1 (x86_64/linux))
    Content-Length: 0
    17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
    [12623366,NET]
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
    CSeq: 101 INVITE
    Reason: Q.850;cause=0;text="unknown"
    Content-Length: 0
    |2,100,230,1.4901099^ISP's-Other-IP^*
    [12623367,NET]
    ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence
    Content-Length: 0

    SIP/2.0 403 Forbidden error
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    If the TOLLFRAUD_APP is rejecting the call, it generates a Q.850       disconnect cause value of 21, which represents ‘Call Rejected’. The       debug voip ccapi inout command can be run to       identify the cause value.
    Additionally, voice iec syslog can be       enabled to further verify if the call failure is a result of the toll-fraud       prevention. This configuration, which is often handy to troubleshoot the origin       of failure from a gateway perspective, will print out that the call is being       rejected due to toll call fraud. The CCAPI and Voice IEC output is demonstrated       in this debug output:
    %VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
    IEC=1.1.228.3.31.0 on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
    000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
       Context=0x49EC9978
    000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
       >>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
    000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
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    voice service voip
    ip address trusted call-block cause
    How to Return to Pre-15.1(2)T Behavior
    Source IP Address Trust List
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    voice service voip
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      ipv4 203.0.113.100 255.255.255.255
      ipv4 192.0.2.0 255.255.255.0
    Configure the router to accept incoming call setups from all source           IP addresses.
    voice service voip
    ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
    Disable the toll-fraud prevention application completely.
    voice service voip
    no ip address trusted authenticate
    Two-Stage Dialing
    If two-stage dialing is required, the following can be configured to       return behavior to match previous releases.
    For inbound ISDN calls:
    voice service pots
    no direct-inward-dial isdn
    For inbound FXO calls:
    voice-port
    secondary dialtone

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    It's possible?
    Thanks.

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    http://www.cisco.com/en/US/partner/docs/ios/voice/fxs/configuration/guide/fxssccpsplmft.html
    HTH
    java
    if this helps, please rate
    www.cisco.com/go/pdihelpdesk

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    I'm having similar recent issues, have you had any luck solving this?

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    Hello,
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    Maxim

    Hi
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