Call forwarding to an International Number

I am working overseas and have a prepaid phone with a local number in that country. Prior to leaving the US again, can I enter the international number of my prepaid phone into the call forwarding feature on my iPhone and have my calls forwarded to that phone? I assume that the person calling my iPhone will not be charged, but will I be charged for making an international phone call each time a call is forwarded to my international phone number? Are there any other charges involved? Has anyone done this before that can tell me how well it works?

To make sure you should contact AT&T but if you set up call forwarding to an international number, you will be the one charged for the international call since your account/phone is dialing the final number.
Again, find out for sure from AT&T who gets the charge.

Similar Messages

  • Call forwarding from an international Skype number

    Ok so here is my setup. I live in Ottawa, Canada and I have an Australian Skype number. What I want to know is, if I put a forwarding number on my australian Skype number to a landline Canadian number and someone from Canada tries to call my Australian Skype number, what rate are they charged when they connect to my forwarded Canadian number and what rate is my Canadian number charged?

    To make sure you should contact AT&T but if you set up call forwarding to an international number, you will be the one charged for the international call since your account/phone is dialing the final number.
    Again, find out for sure from AT&T who gets the charge.

  • 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
    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!

  • Call forward to PSTN on cme

    Hi,
    unable to set up call forward to PSTN.
    /* Style Definitions */
    table.MsoNormalTable
    {mso-style-name:"Table Normal";
    mso-tstyle-rowband-size:0;
    mso-tstyle-colband-size:0;
    mso-style-noshow:yes;
    mso-style-priority:99;
    mso-style-qformat:yes;
    mso-style-parent:"";
    mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
    mso-para-margin:0cm;
    mso-para-margin-bottom:.0001pt;
    mso-pagination:widow-orphan;
    font-size:11.0pt;
    font-family:"Calibri","sans-serif";
    mso-ascii-font-family:Calibri;
    mso-ascii-theme-font:minor-latin;
    mso-fareast-font-family:"Times New Roman";
    mso-fareast-theme-font:minor-fareast;
    mso-hansi-font-family:Calibri;
    mso-hansi-theme-font:minor-latin;
    mso-bidi-font-family:"Times New Roman";
    mso-bidi-theme-font:minor-bidi;}
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        PSTN / IP phone ------> Calling extn on CME (which is call forwarded to another PSTN number)
    config below:
    voice service voip
    ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    supplementary-service h450.12
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
    sip
      registrar server expires max 250 min 200
      asserted-id pai
      localhost dns:XXXXX
      outbound-proxy dns:XXXXX
    dial-peer voice 100 voip
    description ** Incoming call from SIP trunk **
    session protocol sipv2
    session target sip-server
    incoming called-number .
    voice-class codec 1 
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 101 voip
    description ** Outgoinging call to SIP trunk **
    translation-profile outgoing SIPOUT
    destination-pattern 1T
    session protocol sipv2
    session target sip-server
    voice-class codec 1 
    voice-class sip dtmf-relay force rtp-nte
    voice-class sip profiles 101
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 102 voip
    description ** Outgoinging call to SIP trunk **
    destination-pattern 0[2-9].T
    session protocol sipv2
    session target sip-server
    voice-class codec 1 
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    telephony-service
    max-ephones 4
    max-dn 12
    ip source-address 192.168.100.2 port 2000
    calling-number initiator
    timeouts interdigit 5
    load 7960-7940 P00308010200
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    max-conferences 8 gain -6
    call-forward pattern .T
    call-forward system redirecting-expanded
    transfer-system full-consult dss
    transfer-pattern .T
    transfer-pattern 0.T
    create cnf-files version-stamp Jan 01 2002 00:00:00
    ephone-dn  1  dual-line
    number 4961 secondary 99474961 no-reg both
    label 4961
    name 4961
    call-forward all 021605547

    /* Style Definitions */
    table.MsoNormalTable
    {mso-style-name:"Table Normal";
    mso-tstyle-rowband-size:0;
    mso-tstyle-colband-size:0;
    mso-style-noshow:yes;
    mso-style-priority:99;
    mso-style-qformat:yes;
    mso-style-parent:"";
    mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
    mso-para-margin:0cm;
    mso-para-margin-bottom:.0001pt;
    mso-pagination:widow-orphan;
    font-size:11.0pt;
    font-family:"Calibri","sans-serif";
    mso-ascii-font-family:Calibri;
    mso-ascii-theme-font:minor-latin;
    mso-fareast-font-family:"Times New Roman";
    mso-fareast-theme-font:minor-fareast;
    mso-hansi-font-family:Calibri;
    mso-hansi-theme-font:minor-latin;}
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    sip-ua
    credentials number 99474960 username 99474960 password 7 XXXXXXXXX realm as-test.xys.net 
    authentication username 99474960 password 7 XXXXXXX 
    calling-info pstn-to-sip asserted-id number set 99474960 
    no remote-party-id 
    disable-early-media 180 
    retry invite 2
    retry register 3
    timers connect 100 
    registrar dns:as-test.xys.net expires 60  sip-server dns:as-test.xys.net 
    host-registrar

  • Stopping call forwarding on one of two numbers?

    HI,
    I have two Sype numbers, the main one for the UK and a new one that I bought a couple of months ago, a Hong Komg number.
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    Hello and welcome to the Skype Community.
    Checking your options. Canceling the HK Number wouldn't be effective until it expires on February 16th 2016, Which error message [if any] do you receive when you try to cancel call forwarding to the HK Number?
    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.

  • Call Forwarding: Change timeout before forwarding ...

    Hi,
    I've just set up call forwarding with a Skype number and I'm wondering if there is any way to change the number of times it rings before being diverted to my mobile? People are having to wait too long for me to answer and most people are hanging up before I can answer the call...

    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
    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!

  • Unable to disable Call Forwarding function on my iphone 4

    Hi,
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    - Restart iphone
    - Reset all settings
    - Called telco to erase my call forwarding details
    - Dial ##21#
    All to no avail.
    Anybody faces the same problem? Is it telco issue or iphone issue.
    Help please?

    This is a telco problem.
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    It might be a good idea to reset network settings on the iPhone in Settings > General > Reset.
    You can also eliminate the possibility of it being a setting on your phone by restoring as "new", but that's a bit extreme.

  • Call forward to external number(mobile)

    Dears please help me on this
    voice translation-rule 1
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    rule 6 /2837596/ /596/
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    123 is my AA extn
    596 and 599 is an ip phone exten
    i need to transfer directly to an external no (mobile no) when i call 2837596 from outside without extension
    what is the config to be done

        dears , i tried it but call not forwarding please need our help
    voice translation-rule 1
    rule 13 /.*2837499/ /499/
    ephone-dn  499  dual-line
    number 499
    label website
    description 499
    call-forward all 90504495705
    corlist incoming user-international
    ephone  37
    device-security-mode none
    video
    mac-address 001E.F727.F567
    ephone-template 16
    username "700" password 700
    type 7911
    button  1:499
    pin 1700

  • 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.
    But now when I forward my phone to external number and try to call to my phone I get busy trigger.
    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
    If your router is sending a SIP/2.0 403 Forbidden error to the SIP server you are registered to, there is a good chance your  router is blocking the incoming call due to the toll-faud prevention  feature that was added to IOS version 15.1(2)T.
    How to Identify if TOLLFRAUD_APP is Blocking Your Call
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    %VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
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    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:
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    000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
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    voice service voip
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    voice service voip
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      ipv4 192.0.2.0 255.255.255.0
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    voice service voip
    ip address trusted list
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    voice service voip
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    Two-Stage Dialing
    If two-stage dialing is required, the following can be configured to       return behavior to match previous releases.
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    voice service pots
    no direct-inward-dial isdn
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    voice-port
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    The image you have uploaded is unable to be displayed.
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    Thanks!
    -HMthePirate
    Come follow your BlackBerry Technical Team on twitter! @BlackBerryHelp
    Be sure to click Kudos! for those who have helped you.Click Solution? for posts that have solved your issue(s)!

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