Avaya PBX versus Skype Connect

Dears,
we have Avaya PBX ver. 5.2 (upgrade 2010) with enabled SIP server. Normally we use SIP trunks as input lines and SIP extensions.
Now we want to apply Skype Connect as a new input channel to our PBX (Czech republic, Slovakia, Hungary), but I just received answer from Avaya that Skype Connect is possible to use only un USA at this moment.... In the first phase, Avaya customers in the U.S.3 will have access to Skype Connect™, providing a SIP communications channel between Avaya communications systems and Skype.
Please help me with this case.
Thank you
Tomas Husar ([email protected])

Hello eximosr,
I not sure why Avaya is telling you that you cannot use SIP in your country, and don't know more specifics on the Avaya product you are using. 
We have Skype for SIP users all around the world using Avaya IP Office and other Avaya systems.  Half of the older systems use SIP gateway devices because they have not upgraded their systems to current releases.  That may be a solution for you also.  Please visit our:  http://www.skype.com/intl/en/business/skype-connec​t/   page to see the list of vendors for gatyeway products there.
There are many ways to use Skype for SIP and if you contact you local Telecom Consultant, they will be able to provise you with more information.
I hope this helps you solve you issue.
Thanks for using Skype and Skype Community Forums.
Regards,
Victor S.
Regards,
Victor S.
Skype Enterprise Support

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    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

  • Skype Connect and Elastix for incoming and outgoin...

    Hi,
    I ordered Skype Connect, And i want to integrate skype connect with my Elastix server to handle incoming and outgoing calls.
    I created new SIP Trunk through GUI with the following info :
    Incoming Settings
    [skype_in]
    disallow=all
    type=friend
    username=sipusername
    fromdomain=sip.skype.com
    fromuser=sipusername
    realm=sip.skype.com
    host=sip.skype.com
    dtmfmode=rfc2833
    secret=sipuserpass
    nat=yes
    insecure=invite
    qualify=yes
    allow=alaw
    allow=ulaw
    amaflags=default
    trustrpid=no
    sendrpid=yes
    context=from-trunk-sip-Skype_out
    Outgoing Settings :
    [Skype_out]
    context=from-trunk-sip-Skype_out
    Register String:
    SIPUSER:[email protected]
    Incoming calls are working properly, But outgoing calls not working, It keeps saying ( cannot-complete-as-dialed )
    Elastix log after Dial
    [Jul 17 01:01:25] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:1] ResetCDR("SIP/100-00000010", "") in new stack
    [Jul 17 01:01:25] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:2] NoCDR("SIP/100-00000010", "") in new stack
    [Jul 17 01:01:25] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:3] Progress("SIP/100-00000010", "") in new stack
    [Jul 17 01:01:25] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:4] Wait("SIP/100-00000010", "1") in new stack
    [Jul 17 01:01:26] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:5] Progress("SIP/100-00000010", "") in new stack
    [Jul 17 01:01:26] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:6] Playback("SIP/100-00000010", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    [Jul 17 01:01:26] VERBOSE[3501] file.c: -- <SIP/100-00000010> Playing 'silence/1.gsm' (language 'en')
    [Jul 17 01:01:27] VERBOSE[3501] file.c: -- <SIP/100-00000010> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
    [Jul 17 01:01:29] VERBOSE[3501] file.c: -- <SIP/100-00000010> Playing 'check-number-dial-again.gsm' (language 'en')
    [Jul 17 01:01:32] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:7] Wait("SIP/100-00000010", "1") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:8] Congestion("SIP/100-00000010", "20") in new stack
    [Jul 17 01:01:33] WARNING[3501] channel.c: Prodding channel 'SIP/100-00000010' failed
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: == Spawn extension (from-internal, 00201005566352, exited non-zero on 'SIP/100-00000010'
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [h@from-internal:1] Macro("SIP/100-00000010", "hangupcall") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000010", "1?endmixmoncheck") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,9)
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:9] NoOp("SIP/100-00000010", "End of MIXMON check") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/100-00000010", "1?nomeetmemon") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,2
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:28] NoOp("SIP/100-00000010", "End of MEETME check") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:29] GotoIf("SIP/100-00000010", "1?noautomon") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,34)
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:34] NoOp("SIP/100-00000010", "TOUCH_MONITOR_OUTPUT=") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:35] GotoIf("SIP/100-00000010", "1?noautomon2") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,41)
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:41] NoOp("SIP/100-00000010", "MONITOR_FILENAME=") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:42] GotoIf("SIP/100-00000010", "1?skiprg") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,45)
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:45] GotoIf("SIP/100-00000010", "1?skipblkvm") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,4
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:48] GotoIf("SIP/100-00000010", "1?theend") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,50)
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:50] Hangup("SIP/100-00000010", "") in new stack
    [Jul 17 01:01:33] VERBOSE[3501] app_macro.c: == Spawn extension (macro-hangupcall, s, 50) exited non-zero on 'SIP/100-00000010' in macro 'hangupcall'
    [Jul 17 01:01:33] VERBOSE[3501] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000010'
    Are there any modifications should i do in Incoming and outgoing settings to work properly .?
    Regards,

    May be the prefix is wrong. You dont have to put 00 before the country number.

  • Skype connect & VoIP Gateway

    Hi,
    Could you please any give me more information about VoIP gateway, and how to connect it with PBX and how to buy it and what is the price?
    Thanks,
    Mohamed ElDieb

    Hello ToshibaJoe,
    If your PBX doesn’t support SIP check with your manufacturer to see whether a SIP gateway, module or upgrade is available for your existing PBX, that would enable you to use Skype Connect/SIP.
    Also, it is highly recommended that a Skype Connect certified device is used, to gaurantee all features and functions work.  Following is a link to that lists all Skype Connect certified products:
    http://www.tekvizionlabs.com/3rdpartyprograms/skype/skype_verified_products.php
    Please let us know if you have further questions.
    Thank you,
    MariaA
    Skype Enterprise Support

  • Skype connect without being a business?

    Hello.
    How I do to get Skype Connect without being a business?
    Since Skype say I need to enter "my exact business name", I guess they will verify it somehow? Or should I just enter NOT_A_BUSINESS in the field?
    I want to connect my home SIP PBX with skype so I can make skype calls and receive skype calls via SIP. I only need to be able to call skype accounts and receive calls fron skype accounts, since I already have a SIP account with a operator with a accompaniying DID that I use to place/receive non-skype calls.

    Short answer you can't.
    But as i stressed in a post i deleted currently i to have had to setup a skype manager account but obviously can not proceed to be verified as a business not being one.
    Hence why i requested an experimental class or something not requiring a registered business.
    What a lot of these corporates don't realise is the legacy units no longer supported will of course windup as domestic surplus.
    Because we don't have megabucks to our name it barely raises an eyebrow.
    Especially to the equipment suppliers.
    Recently i was told by distributers that no sorry we don't support skype connect.
    Pity they didn't think the response through because had they visited the skype connect website and clicked on the company in questions banner.
    Oh dear someones telling lies.
    Most of these firms are of the opinion that unless your a technician or have a business to warrant such technology that ye humble mortal is not privy to such information.
    To those experimenting out there don't be discouraged.
    Ignore the ney sayers because they keep proving themselve's wrong half the time.
    As usual it's more red tape and bs beaurocracy impeding technology.

  • Skype Connect calls vs. local provider calls

    Hi! 
    Should I subscribe to Skype Connect, will I be able to choose which "phone line" to use for each call? Suppose I want to place via skype connect just international calls and make local calls via my local provider. Will that be possible? 
    Thanks,
    Dani

    Hello dr81to,
    I see you have a question on how to use a Skype Connect account and it's calling features. 
    In most cases, users just use Skype for all calls, sinse they are so cheap.  Calls around the world and around the block are very cheap.  But, if you wish, you can use a SIP enabled device (like a Grandstream, Cisco, Avaya, Siemens, or others...) with 2 lines and have a Skype SIP Line on one and your local PSTN line on the other. That way you can select which line you want to use.  You can also use a softphones like 3CX, or Bria to select each line too.  The choice is yours.  Have fun using Skype.
    Thank You for using Skype and the Skype Community Forums.
    Regards,
    Victor S.
    Skype Enterprise Support

  • Skype connect not connecting

    Hi there
    I have 2 separate skype connect connections, 1 into a PBX in Brisbane, the other in Wellington. They have been working happily for quite a while. As of sometime today no one can connect to them, the connection says "connecting" but it never makes a connection to the PBX and the call never completes.
    At the PBX the lines are registered and show no issues. We use these for inbound only so I don't know if outbound works but I will check. 
    Is anyone aware of any issues with Skype Connect and the best way to report it

    Thanks very much, we are using it for inbound to allow remote callers on Skype in, so I am not sure there is an alternative. It worked well for quite a while then dropped off in the last few months. 
    I don't really know the right way to get hold of them. Nothing much has worked. But an unreliable service is no good - cleaner to have nothing in terms of support and expectations

  • Skype Connect SIP trunking to Yeastar U100

    I am trying to get this service running for the first time. I signed up for a SIP profile through Skype Manager and followed Yeastar's setup instructions. Both ends indicate the regestration was successful. I can receive incoming Skype calls. I followed Yeastar's example to set up and Outgoing Route and I am attempting to call the Skype Echo Test at 001760-660-4690 as they recommend. All I ever hear is a recording saying that "All Circuiits are Busy". I have tried dialing other PSTN number and I get the same failure. Yeastar Support has look at the outgoing call msgs between me and Skype and they say what I'm sending looks right but Skype Connect is saying that the number I'm calling is invalid.
    I have been in 3 separate Chats with Skype Business support and they have not be able to give me any help as to what is going on.

    Each channel carries one phone call.  So if you have 4 channels, you can carry 4 simultaneous phone calls, in any combination of incoming or outgoing. 

  • 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)

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