[Solved] bitlbee/skype connection trouble

Hi all, been trying to solve this problem for a while now but to no avail, there doesn't appear to be much out there that offers any hints for where to look. I'm trying to get bitlbee, irssi and Skype working together, but every time I try to connect my Skype account, I just get connection time-outs or messages telling me it's unable to connect:
22:15 -!- mode/&bitlbee [-C] by root
22:15 <@root> Password accepted, settings and accounts loaded
22:15 <@root> Trying to get all accounts connected...
22:15 <@root> skype - Logging in: Connecting
22:15 -!- mode/&bitlbee [+C] by root
22:17 <@root> skype - Login error: Connection timeout
22:17 <@root> skype - Logging in: Signing off..
22:17 <@root> skype - Logging in: Reconnecting in 5 seconds...
22:25 -!- mode/&bitlbee [-C] by root
22:25 <@root> Password accepted, settings and accounts loaded
22:25 <@root> Trying to get all accounts connected...
22:25 <@root> skype - Logging in: Connecting
22:25 <@root> skype - Login error: Could not connect to server
22:25 <@root> skype - Logging in: Signing off..
22:25 <@root> skype - Logging in: Reconnecting in 5 seconds..
22:25 -!- mode/&bitlbee [+C] by root
22:25 <@root> skype - Logging in: Connecting
22:25 <@root> skype - Login error: Could not connect to server
22:25 <@root> skype - Logging in: Signing off..
22:25 <@root> skype - Logging in: Reconnecting in 15 seconds..
22:26 <@root> skype - Logging in: Connecting
22:26 <@root> skype - Login error: Could not connect to server
22:26 <@root> skype - Logging in: Signing off..
22:26 <@root> skype - Logging in: Reconnecting in 45 seconds..
Now, I have skyped set up and running on the machine, skype4py is authorised in the Public API tab of Skype, using the default port etc. but whether I connect from a remote machine or the same one, it just times out. Strangely, when I try to run skyped -l for logging, it just shows me the help message. As far as I can tell, it has executed just fine as it gives no errors. The username is correct, as is the password I'm using. I've generated the necessary certificates, as per this guide.
$ skyped
skyped is started on port 2727, pid: 635
I can't find any log files to check for this so I'm a little bit lost as to what to do next. I can use bitlbee to connect to MSN no problem, so I don't think that's the issue. I imagine it's to do with skyped, but I don't really know how to check, so if anyone could provide any advice on the matter I would appreciate it. Even just a pointer in the right direction would help immensely. Hopefully one of my fellow Archers has this working.
Last edited by JHeaton (2012-04-01 15:16:36)

Hi pshevtsov, thank you for the reply. I checked and although skyped was starting Skype, it was not continuing to run so I tried it with --nofork as you said and I got this error message (this also appears to be the error shown in the latest comment on the AUR page for skype4py).
[joel@glitch ~]$ skyped -n
Traceback (most recent call last):
  File "/usr/bin/skyped", line 120, in skype_idle_handler
    skype.skype.SendCommand(c)
  File "/usr/lib/python2.7/site-packages/Skype4Py/skype.py", line 778, in SendCommand
    self._Api.send_command(Command)
  File "/usr/lib/python2.7/site-packages/Skype4Py/api/posix_dbus.py", line 207, in send_command
    result = self.skype_out.Invoke(cmd)
AttributeError: 'NoneType' object has no attribute 'Invoke'
Exiting.
[joel@glitch ~]$
I don't really know enough Python to work out how to sort this issue out and can't seem to find anything on the Sourceforge project page in terms of troubleshooting.

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    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 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)

    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.

  • Skype connectivity with lync online plan1

    hi ,recently I purchased lync online plan1 to enable skype connectivity with lync but not able to add any contact from lync client as I am using lync 2010 client at windows pc and lync 2011 client at mac PC
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    Hi skype user,
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    For Windows, you could install the Lync 2013 Basic.
    http://www.microsoft.com/en-us/download/details.aspx?id=35451
    For Mac OS , you could install the latest update “October 2014 update for Lync for Mac 2011 14.0.10”
    http://www.microsoft.com/en-us/download/details.aspx?id=36517
    If it still does not work, you could post the question on Office365 forum for assistance. Thank you for your understanding.
    http://community.office365.com/en-us/f/166.aspx
    Best regards,
    Eric

  • 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 
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     destination-pattern 2...
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     voice-class codec 1
     dtmf-relay h245-alphanumeric
     no vad

  • Skype connect

    Hi,
    I'm new to skype so I don't understand completely how it works,  me and my son who lives in Northern Virginia and I live in VA. Beach, we are tying to set-up skype with video on the computer, well I am able to see my self on my camera, but when I try to connect he gets a message that says person can only receive IM messages.

    Hi,
    I have a problem with skype connection, for more than a week now. Every time when I sign in to my account I get a warning message "Skype home is unavailable at the moment. Check back later to see you news and alerts" I can't make any kind of calls and still my credits are enought what can i do? It like Iam always of line why???

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    i have configured Lync edge with single public IP without reverse proxy and my external users can connect it.. now my next task is to provide the lync-skype connectivity for which i need to configure federation services.. i have an extra Public IP and
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    Hi babarmunir,
    Is there any update on the federation issue?
    Just for your reference, here is another case explains the SRV record for Lync federation:
    http://social.technet.microsoft.com/Forums/lync/en-US/7a6ce1c5-5c0b-45c2-9c1d-732446743fee/federation-srv-records
    Kent Huang
    TechNet Community Support

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