SKYPE CONNECT Trunk to CUCM.

im interested in creating a sip trunk to skype connect. im new to voice but i think i can get it done.
is a cube absolutely necessary or the trunk can be created directly to cucm.
also can a regular cisco router with access to the internet work just as well as a cube ?
my current system has a voice gateway  E1 to the PSTN.. is it possible to give this voicerouter access to the internet and use to connect to skype instead of the cube?

hey anas.. i have set up the skype trunk.. and calls seem to be going out my problem is incoming calls i have a skype id .. this is the matching incoming dial peer.. dial-peer voice 6 voip
 description incoming skype
 translation-profile incoming IncomingSkype
 preference 1
 destination-pattern 13478093543
 session target ipv4:192.168.xxx.x (cucm)
 incoming called-number 13478093543
 voice-class codec 1
 dtmf-relay h245-alphanumeric.
the translation profile changes the calling number to  a 4 digit number 
a number which matches this dial peer which is already on the working system
dial-peer voice 10 voip
 description Codec Match
 destination-pattern 2...
 session target ipv4:192.168.xxx.x (cucm)
 voice-class codec 1
 dtmf-relay h245-alphanumeric
 no vad

Similar Messages

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  • Avaya PBX versus Skype Connect

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  • Can't make outgoing call with Skype Connect

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    To: <sip:[email protected]>;tag=c990d13f-90f7a10d-0-55cb59a8-0
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    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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    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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    Call-ID: [email protected]
    CSeq: 101 INVITE
    Max-Forwards: 70
    Contact: "Scott's Office" <sip:[email protected]:5060>
    Expires: 240
    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]
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    == 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>
    <--- Transmitting (NAT) to 192.168.1.16:5060 --->
    SIP/2.0 100 Trying
    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]>
    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
    <------------>
    -- Executing [19739928881@home:1] Dial("SIP/scott_office-000000b0", "SIP/skype/+19739928881") in new stack
    == Using SIP RTP CoS mark 5
    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=z9hG4bK1c3f16ee;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: 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
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    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
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    Content-Length: 0
    <------------->
    --- (8 headers 0 lines) ---
    <--- SIP read from TLS:63.209.144.201:5061 --->
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    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

  • Third Party Phone over SIP Trunk with CUCM 9.x

    Hi all,
    I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
    I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
    Cisco Phone: INVITE sip.60xxxx%23@ipadress
    Third Party SIP Phone:  INVITE sip:[email protected]
    It seems the Cisco phones gets some extra configured the Third Party ones dont...
    Thanks in advance for any help.
    //Per

    Thanks for the answer
    Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
    When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
    When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty.  The termination Cause Code is that the number requested is Unallocated/Unassigned..
    In other words something is missing to get CUCM to strip 60 from the Polycom phones dialed number and send it towards the SIP trunk like it does when the Cisco phones call it.
    Unfortunatley i dont have the meens to attach the trace...
    Thanks again for any help/advice
    With regards, Per.

  • Mitel System With Trunk To CUCM DTMF Not Recognized.

    Hello,
    Not sure if this is correct forum but maybe someone has seen this. Overview is that DTMF is not being recognized by CVP from the greeting. Call flow is such.
    Mitel 5000 --siptrunk---CUCM9---siptrunk---CVP9
    Currently the DTMF type on all trunks is RFC2833. And verified the Mitel is RFC as well. I think the problem is that for payload Mitel is sending 96. But on the Cisco CUCM the trunks are using the default of 101 for both the trunk to Mitel and CVP. Here is the invite in from Mitel you can see it sends payload of 96.
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.254.20.32:5060;branch=z9hG4bK2312043656-10065
    Max-Forwards: 70
    Allow: NOTIFY,REGISTER,REFER,SUBSCRIBE,INVITE,ACK,OPTIONS,CANCEL,BYE
    User-Agent: Mitel-5000-ICP-5.1.0.56
    P-Asserted-Identity: "Laura B." <sip:[email protected]>
    From: "Laura B." <sip:[email protected]:5060>;tag=Mitel-5000_2312043669-10065
    To: 72799 <sip:[email protected]:5060>
    Call-ID: 2312043591-10065
    CSeq: 1 INVITE
    Contact: "Laura B." <sip:[email protected]:5060>
    Content-Type: application/sdp
    Content-Length: 266
    v=0
    o=Mitel-5000-ICP 169457268 1404250460 IN IP4 10.254.20.31
    s=SIP Call
    c=IN IP4 10.254.20.31
    t=0 0
    m=audio 6770 RTP/AVP 0 8 96
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:96 telephone-event/8000
    a=ptime:20
    a=maxptime:30
    a=cdsc:1 image udptl t38
     5:16 PM
    here is 200 ok
     5:16 PM
     5:16 PM
    Jay Schulze:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.254.20.32:5060;branch=z9hG4bK2312043656-10065
    From: "." <sip:[email protected]:5060>;tag=Mitel-5000_2312043669-10065
    To: 72799 <sip:[email protected]:5060>;tag=10585346~dfbf10b3-6c69-4443-852f-cbf609935a6f-42551743
    Date: Tue, 01 Jul 2014 21:34:24 GMT
    Call-ID: 2312043591-10065
    CSeq: 1 INVITE
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    Allow-Events: presence, kpml
    Supported: replaces
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    P-Preferred-Identity: <sip:[email protected]>
    Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060>
    Content-Type: application/sdp
    Content-Length: 238
    v=0
    o=CiscoSystemsCCM-SIP 10585346 1 IN IP4 10.38.246.136
    s=SIP Call
    c=IN IP4 10.38.246.166
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 23956 RTP/AVP 0 96
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=rtpmap:96 telephone-event/8000
    a=fmtp:96 0-15
    This is the invite over to CVP. Still sending 96. I'm pretty sure I can fix this by setting the payload type to 96 on both trunks on CUCM. I was able to test with it afterhours. After reading through Mitel docs there is no way to change payload type on their side. The question is really would there be anything else this could maybe break on the CVP side?

    Well that did not work. What I found was it was going through 96 all the way to CVP. However when CVP returned a label to the VXML. The VXML sent back the 200 OK with 101.
    If I changed the the rtp payload-type nte 96 on the VXML dial-peer it did work. However it broke any other call flow to CVP from being able to recognize DTMF. 
    I wonder if this is some type of bug on VXML. Because it was my understanding by having the command 'asymmetric payload full' under SIP. It would pass the payload type from one call leg to the other.

  • Skype connect, outbound failing

    I recently setup skype connect for product testing on an NEC SV8100.
    I can recieve call ok.
    When i dial out the other end rings, but no RTP audio, then it gives engaged.. I still get charged for each call !!
    Done a wireshark trace :
    No.     Time        Source                Destination           Protocol Length Info
          1 0.000000    172.10.0.1            63.209.144.201        SIP/SDP  934    Request: INVITE sip:[email protected], with session description
    Frame 1: 934 bytes on wire (7472 bits), 934 bytes captured (7472 bits)
    Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
    Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
    User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
    Session Initiation Protocol
    No.     Time        Source                Destination           Protocol Length Info
          2 0.261931    63.209.144.201        172.10.0.1            SIP      521    Status: 407 Proxy Authentication Required
    Frame 2: 521 bytes on wire (4168 bits), 521 bytes captured (4168 bits)
    Ethernet II, Src: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f), Dst: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff)
    Internet Protocol Version 4, Src: 63.209.144.201 (63.209.144.201), Dst: 172.10.0.1 (172.10.0.1)
    User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
    Session Initiation Protocol
    No.     Time        Source                Destination           Protocol Length Info
          3 0.328555    172.10.0.1            63.209.144.201        SIP      455    Request: ACK sip:[email protected]
    Frame 3: 455 bytes on wire (3640 bits), 455 bytes captured (3640 bits)
    Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
    Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
    User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
    Session Initiation Protocol
    No.     Time        Source                Destination           Protocol Length Info
          4 0.457122    172.10.0.1            63.209.144.201        SIP/SDP  1175   Request: INVITE sip:[email protected], with session description
    Frame 4: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
    Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
    Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
    User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
    Session Initiation Protocol
    No.     Time        Source                Destination           Protocol Length Info
          5 0.999458    172.10.0.1            63.209.144.201        SIP/SDP  1175   Request: INVITE sip:[email protected], with session description
    Frame 5: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
    Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
    Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
    User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
    Session Initiation Protocol
    No.     Time        Source                Destination           Protocol Length Info
          6 2.017736    172.10.0.1            63.209.144.201        SIP/SDP  1175   Request: INVITE sip:[email protected], with session description
    Frame 6: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
    Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
    Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
    User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
    Session Initiation Protocol
    No.     Time        Source                Destination           Protocol Length Info
          7 4.000270    172.10.0.1            63.209.144.201        SIP/SDP  1175   Request: INVITE sip:[email protected], with session description
    Frame 7: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
    Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
    Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
    User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
    Session Initiation Protocol
    No.     Time        Source                Destination           Protocol Length Info
          8 8.019283    172.10.0.1            63.209.144.201        SIP/SDP  1175   Request: INVITE sip:[email protected], with session description
    Frame 8: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
    Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
    Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
    User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
    Session Initiation Protocol
    No.     Time        Source                Destination           Protocol Length Info
          9 11.947857   63.209.144.201        172.10.0.1            UDP      214    Source port: 26998  Destination port: 10020
    Frame 9: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)
    Ethernet II, Src: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f), Dst: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff)
    Internet Protocol Version 4, Src: 63.209.144.201 (63.209.144.201), Dst: 172.10.0.1 (172.10.0.1)
    User Datagram Protocol, Src Port: 26998 (2699, Dst Port: 10020 (10020)
    Data (172 bytes)
    No.     Time        Source                Destination           Protocol Length Info
         10 11.948226   172.10.0.1            63.209.144.201        ICMP     70     Destination unreachable (Port unreachable)
    Frame 10: 70 bytes on wire (560 bits), 70 bytes captured (560 bits)
    Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
    Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
    Internet Control Message Protocol
    No.     Time        Source                Destination           Protocol Length Info
         11 11.967964   63.209.144.201        172.10.0.1            UDP      214    Source port: 26998  Destination port: 10020
    Frame 11: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)
    Ethernet II, Src: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f), Dst: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff)
    Internet Protocol Version 4, Src: 63.209.144.201 (63.209.144.201), Dst: 172.10.0.1 (172.10.0.1)
    User Datagram Protocol, Src Port: 26998 (2699, Dst Port: 10020 (10020)
    Data (172 bytes)

    Try to reset all Skype settings.
    Quit Skype or use Windows Task Manager to kill any Skype.exe process. Go to Windows Start and in the Search/Run box type %appdata% and then press Enter or click the OK button. The Windows File Explorer will pop up. There locate a folder named “Skype”. Rename this folder to something different, e.g. Skype_old.
    If you are on the latest Skype 6.5/6.6 version, then do also this:
    Go to Windows Start and in the Search/Run box type %temp%\skype and then press Enter or click the OK button. Delete the DbTemp folder.
    Restart Skype.
    N.B. If needed, you will still be able to re-establish your call and chat history. All data is still saved in the Skype_old folder.

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    It is simple. For me, my country etc - there are many much better much cheaper services, and Skype can't offer nothing to beat them. But my client wants to use Skype Connect, and I have to solve his problems.
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