I can't make a call using skype to go number

when I dial it - there is voice message saying - "the number is vacant, please call your own directory inquire service for further information".
what I am doing wrong? How should I use it? Should I call it through the country code or plain?

shalltoji wrote:
Should I call it through the country code or plain?
Hello,
You must always prefix your dial string with the international country access code (+1 for USA, +44 for UK etc.).
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.

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    allowoverlap=no
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    tlsenable=yes
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    transport=tls
    srtpcapable=yes
    encryption=yes
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    Max-Forwards: 70
    From: <sip:[email protected]>;tag=as6edf93cf
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    Call-ID: [email protected]
    CSeq: 32495 REGISTER
    User-Agent: Asterisk PBX 10.5.2
    Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com:5061", nonce="5036b5770000182c78c1d1909cfd5c74e33f033c952d240d", response="81001ceacd91b16ebb956d3c55991471"
    Expires: 120
    Contact: <sip:[email protected]:5061;transport=TLS>
    Content-Length: 0
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    From: <sip:[email protected]>;tag=as6edf93cf
    To: <sip:[email protected]>;tag=c990d13f-90f7a10d-0-55cb59a8-0
    Call-ID: [email protected]
    CSeq: 32495 REGISTER
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport=50541;received=1.2.3.4
    Expires: 45
    Contact: <sip:[email protected]:5061;transport=tls>;expires=45
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    <------------->
    --- (9 headers 0 lines) ---
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    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
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    Contact: "Scott's Office" <sip:[email protected]:5060>
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    User-Agent: Cisco/SPA504G-7.5.2b
    Content-Length: 234
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
    Supported: replaces
    Content-Type: application/sdp
    v=0
    o=- 88651316 88651316 IN IP4 192.168.1.16
    s=-
    c=IN IP4 192.168.1.16
    t=0 0
    m=audio 16484 RTP/AVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:30
    a=sendrecv
    <------------->
    --- (14 headers 12 lines) ---
    Sending to 192.168.1.16:5060 (NAT)
    Using INVITE request as basis request - [email protected]
    Found peer 'scott_office' for 'scott_office' from 192.168.1.16:5060
    == Using SIP RTP CoS mark 5
    Found RTP audio format 0
    Found RTP audio format 8
    Found RTP audio format 101
    Found audio description format PCMU for ID 0
    Found audio description format PCMA for ID 8
    Found audio description format telephone-event for ID 101
    Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
    Peer audio RTP is at port 192.168.1.16:16484
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    list_route: hop: <sip:[email protected]:5060>
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    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
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    Call-ID: [email protected]
    CSeq: 101 INVITE
    Server: Asterisk PBX 10.5.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Contact: <sip:[email protected]:5060>
    Content-Length: 0
    <------------>
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    Audio is at 9302
    Adding codec 100003 (ulaw) to SDP
    Adding codec 100004 (alaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 63.209.144.201:5061:
    INVITE sip:[email protected] SIP/2.0
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    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 10.5.2
    Date: Thu, 23 Aug 2012 23:22:34 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 370
    v=0
    o=root 1671301052 1671301052 IN IP4 192.168.1.15
    s=Asterisk PBX 10.5.2
    c=IN IP4 192.168.1.15
    t=0 0
    m=audio 9302 RTP/SAVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
    -- Called SIP/skype/+19739928881
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 100 Trying
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
    Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    Content-Length: 0
    <------------->
    --- (8 headers 0 lines) ---
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 407 Proxy Authentication Required
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Proxy-Authenticate: Digest realm="sip.skype.com", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", algorithm=MD5
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
    Content-Length: 0
    <------------->
    --- (8 headers 0 lines) ---
    set_destination: Parsing <sip:[email protected]> for address/port to send to
    set_destination: set destination to 63.209.144.201:5060
    Transmitting (NAT) to 63.209.144.201:5061:
    ACK sip:[email protected] SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 10.5.2
    Content-Length: 0
    Audio is at 9302
    Adding codec 100003 (ulaw) to SDP
    Adding codec 100004 (alaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 63.209.144.201:5061:
    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX 10.5.2
    Proxy-Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:[email protected]", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", response="6efb0e37178bae868f0a1e0ddf110e3c"
    Date: Thu, 23 Aug 2012 23:22:34 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 370
    v=0
    o=root 1671301052 1671301053 IN IP4 192.168.1.15
    s=Asterisk PBX 10.5.2
    c=IN IP4 192.168.1.15
    t=0 0
    m=audio 9302 RTP/SAVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 100 Trying
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
    Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    Content-Length: 0
    <------------->
    --- (8 headers 0 lines) ---
    Really destroying SIP dialog '[email protected]' Method: REGISTER
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 180 Ringing
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 103 INVITE
    User-Agent: SipGW 8
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
    Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    Content-Length: 0
    <------------->
    --- (9 headers 0 lines) ---
    list_route: hop: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
    -- SIP/skype-000000b1 is ringing
    <--- Transmitting (NAT) to 192.168.1.16:5060 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>;tag=as3f27fa61
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Server: Asterisk PBX 10.5.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Contact: <sip:[email protected]:5060>
    Content-Length: 0
    <------------>
    <--- SIP read from TLS:63.209.144.201:5061 --->
    SIP/2.0 408 Request Timeout
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
    Content-Length: 0
    <------------->
    --- (7 headers 0 lines) ---
    [2012-08-23 19:22:45] WARNING[17932]: chan_sip.c:20947 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '[email protected]'. Giving up.
    set_destination: Parsing <sip:[email protected]> for address/port to send to
    set_destination: set destination to 63.209.144.201:5060
    Transmitting (NAT) to 63.209.144.201:5061:
    ACK sip:[email protected]:5061;maddr=63.209.144.201;transport=tls SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
    Max-Forwards: 70
    From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
    To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
    Contact: <sip:[email protected]:5061;transport=TLS>
    Call-ID: [email protected]
    CSeq: 103 ACK
    User-Agent: Asterisk PBX 10.5.2
    Content-Length: 0
    Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
    == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [19739928881@home:2] Hangup("SIP/scott_office-000000b0", "") in new stack
    == Spawn extension (home, 19739928881, 2) exited non-zero on 'SIP/scott_office-000000b0'
    Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
    <--- Reliably Transmitting (NAT) to 192.168.1.16:5060 --->
    SIP/2.0 503 Service Unavailable
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>;tag=as3f27fa61
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Server: Asterisk PBX 10.5.2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0
    <------------>
    <--- SIP read from UDP:192.168.1.16:5060 --->
    ACK sip:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
    From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
    To: <sip:[email protected]>;tag=as3f27fa61
    Call-ID: [email protected]
    CSeq: 101 ACK
    Max-Forwards: 70
    Contact: "Scott's Office" <sip:[email protected]:5060>
    User-Agent: Cisco/SPA504G-7.5.2b
    Content-Length: 0
    <------------->
    --- (10 headers 0 lines) ---
    Really destroying SIP dialog '[email protected]' Method: INVITE
    Really destroying SIP dialog '[email protected]' Method: ACK

    I wound up calling skype support. This is the final sip.conf looks like. Hope it helps. Good luck.
    Scott
    [general]
    context=default_context
    allowguest=no
    alwaysauthreject=yes
    allowoverlap=no
    udpbindaddr=0.0.0.0
    tlsenable=yes
    tlsbinddir=0.0.0.0
    tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
    tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
    tlscipher=ALL
    tlsclientmethod=tlsv1
    tcpenable=yes
    tcpbindaddr=0.0.0.0
    transport=udp,tcp,tls
    srvlookup=yes
    dynamic_exclude_static = yes
    buggymwi=yes
    contactpermit=192.168.1.0/24
    register => tls://[email protected]
    [skype]
    type=friend
    context=from-skype
    dtmfmode=rfc2833
    host=sip.skype.com
    username=user
    fromuser=user
    secret=pass
    disallow=all
    allow=ulaw
    allow=alaw
    nat=yes
    fromdomain=sip.skype.com
    insecure=port,invite
    transport=tls
    srtpcapable=yes
    encryption=yes

  • Can't make vedio calls in skype in my nokia e 7-00

    I have tried many more times. But always it appears error massage on the display. I can't answer calls in skype also.

    @princej
    You are using version from Nokia Store?http://store.ovi.com/content/20924
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