No call out/in with skype subscription

Hi i have bought skype number and 1year calling subscription uk/ireland. I just had connected few times to the uk landlines and received call couple of time from uk landlines but most of the time it disconnected. would you please advise what is restricting making and receiving calls to/from uk landlines. currenlt i am not able to make/receive call. thanks

I think you may need to contact customer service regarding that matter. Just open the link pasted below to see the instructions on how to get in touch with customer service. provide them with as much relevant details as possible when you contact them for faster handling of your concern.
https://support.skype.com/en/faq/FA1170/how-can-i-contact-skype-customer-service
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Similar Messages

Origin of call with Skype subscriptions

I understand the Skype subscriptions and their rates when calling TO a specific country, but does it matter where you call FROM/ the origin of the call if you have a subsription to call that particular country?
For example: If I buy a year subscription to call landlines and mobiles in the United States, for the $2.99 a month, does it matter WHERE I am calling from (such as France, London, Germany) as long as I am only calling TO the United States? Or is that rate only for people in the U.S. trying to call others in the U.S? I'm a U.S. resident studying abroad for a year and wasn't sure if the origin of the call/ where I call from, is as important as where the subsription is set up to call to.

sombrioio wrote:
I am a subscriber in CANADA for some reason the default country of origin for mey calls seems to be the United states of America. How do i Change this ver bad FAULT or default country of origin. It really pisses me off that the USA seems to have taken over the world and everything seems to be THEIRS.
When calling telephones, the people you call may see a number based on where Skype patches your call into the telephone network. When calling phones across the USA and Canada, this is often done from locations in California. Users have no control over where Skype patches calls into the telephone network, as this is done in part to minimize the costs for those calls (and how there are lower rates in general when calling with Skype, and "unlimited" subscriptions for some parts of the world).
Skype has been doing this for many years, even before its acquisition by a big US-based software company. It's not some part of a scheme for the US taking over the world. It is probably cheaper for Skype to patch calls bound for Canadian phone numbers into the phone network from wherever it is being done now (probably in California) than it would be to patch the calls into the phone network somewhere in Canada.
Patrick
Location/Ubicacion: Arizona USA
Time Zone/Hora Local: UTC/GMT -7
If this message has adequately addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others.
Si esto mensaje le ha ayudado, por favor haga clic en "Aceptar como solución". Si encuentra un mensaje útil, por favor "Da Kudos" al final del mensaje, por lo que esta información puede beneficiar a otros.
I am not a Skype employee. No soy un empleado de Skype.

How do I display call out accessories with different images?

I need some help with an app I'm developing. I'm displaying a map with an array of annotations.
I have managed to display the map & annotations successfully.
What I want to do next is have call out accessories that display different images everytime they are tapped.
Can anyone help with this? I have managed to display a right call out accessory but I'm getting stuck here & I don't know how to get my code to display images.
Any help wil be greatly appreciated!

baltwo: Thanks for the tip, I didn't know how easy it was to set up custom icons in OS X. In this case it's a difficult solution because it will have to be done by hand for every single folder, which is a problem if I'm going to have these in two places. I'm new to Automator but there doesn't seem to be a way to have it; 1) Select all the folders in a directory. 2) Copy the first image in each folder to the clipboard and have it use that image as each folder's icon.
Barney-15E: I know it sounds moronic, but yes. I've tried iPhoto but it's not a very good solution in this case. My folder structure is like this:
Root -> 282 image folders with 10 to 60 pictures in each.
All I really need is the ability to see at a glance what each image folder has without opening each individual folder. The folders are numbered sequentially (1-282 in this case) rather than by proper names, so this is the most efficient way.

How to call a 090 with skype credit?

Tried to do it but it says invalid number!
0900 1600900
that's the number i want to call. any help would be appreciated
thanks!

Hi, Wotabiros, and welcome to the Community,
As is noted in the blue box, above where we post our messages,
If you have a payment issue please contact Skype customer service as users here on the community can't help you with those. To help you find the right place for these types of questions please feel free to use the following links:
Refund requests
If you click on the blue type, this is actually a link to the instructions on how to request a refund
Regards,
Elaine
Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
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When I call a landline with skype by iMac (10.7.5) the other side hears my voice very hollow and far.

Dear mac colleagues,
I have a imac (2,7 Ghz Intel core, 4Gb internal memory) with Mac Os X 10.7.5. The problem is that the quality of my voice that people hear when I call them by skype 6.4.0.833 is bad (hollow, and far). When I call with my old macbook pro there is no problem. This is very frustrating because of a new iMac computer that powerfull I expect a lot better than this. I tried already to install skype skype again and checked the sound preferences without any result. Anybody has an idea what I could do?
greetings Guy

The following line tells me that this is not a template file, but a document created from a template called index.dwt
<html xmlns="http://www.w3.org/1999/xhtml">
This means that you saved a child document as a template file at some stage.
If you go to Modify->Templates->Detach from Template, you can remove the template structure from your document. All you then need to do is to re-save the document as a template.

Database call out webservices with JDeveloper - help

Hi all,
I am using JDeveloper 10.1.3.0 to generate java proxies for web services and deploy them to my Oracle 10g Enterprise Edition database.
I am able to create the proxy and test the web service, it works fine. When I create a deployment profile and deploy, everything succeeds, I am able to see the PL/SQL wrapper function in the database. However, when I execute the function, I get an error class name <my java classname> not found. This is a case when I execute a web service that runs on our server and requires authentication. I have provided the authentication in the setUsername and setPassword methods of the java proxy.
When I try to do the same with a stock quote web service, I do not get an error, but there is no output either. It returns nothing.
Is this an issue because of the jdk version problem? My database jvm is jdk1.4 compliant and the JDeveloper is 1.5 compliant. But I think there is somehting very basic that I am missing. Can experts please help me with this, I have been struggling for long!
Thank you very much. Any help is greatly appreciated.
Regards,
Nithya

I have still not figured this one out. Any help please?
Regards,
Nithya

Calling back home with skype to go

I know I can use this service to call from home to abroad.
If I am though away abroad on business, then can I use such a service to call back to home at a sensible rate??

Hello,
You can call from any of these countries where the STG program is offered.
Australia
Canada
Chile
Denmark
Estonia
Finland
Greece
Hungary
Ireland
Latvia
Lithuania
Luxembourg
Mexico
New Zealand
Poland
Portugal
Romania
Slovenia
Singapore
South Africa
Sweden
United Kingdom
USA
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.

Problems calling designated country with Skype Pre...

Greetings -
I am new to Skype and subscribed to Skype Premium about a month ago and designated Belgium as the country for my unlimited calls. When I try to call a number in Belgium, a window pops up and indicates that I need to purchase Skype credit in order to complete the call. I have tried to make several calls, but they do not go through and they are logged in my call history for 0:00 minutes each at a $0.00 cost. My profile has Belgium properly designated.
Any advice on how I can fix this?
Jeff
Solved!
Go to Solution.

jp1450 wrote:
Any advice on how I can fix this?
Hello Jeff,
The message you are receiving indicates that you are attempting a call not covered by your Plan. In the case of Belgium only calls to landlines are included (not cellphones).
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.

I can call out on my Skype number but when people ...

Please help. I have tried the local number, the number plus area code, and the 1 + area code + number. All ring busy all the time.

Please help. I have tried the local number, the number plus area code, and the 1 + area code + number. All ring busy all the time.

Free international 800 calling to U.S. with skype

I will be residing in Thailand this winter and want to know if I will be able to call my bank in the U.S using their toll-free number? If so, will this call be free? Also, what format should I use when dialing this 800 number from Thailand to the U.S? Thanks for any input

goonfrog6 wrote:
I will be residing in Thailand this winter and want to know if I will be able to call my bank in the U.S using their toll-free number? If so, will this call be free? Also, what format should I use when dialing this 800 number from Thailand to the U.S? Thanks for any input
If you are able to reach that number using Skype, it would be a free call.
To dial the number from Skype, you would dial +1 followed by the toll-free area code and number (total of 10 digits after the +1). Please note that Skype does not guarantee calls to all toll-free numbers in the +1 country code. Some of these numbers are actually in countries other than the USA (i.e. Canada, Bermuda, many islands in the Caribbean, plus some US territories in the Pacific), or the company/organization that has the number has limited where incoming calls can come from (those with the toll-free numbers can restrict calls to come in only from specific states and/or area codes). Skype typically patches calls to US phone numbers, including any of the toll-free area codes, into the telephone network in California. The toll-free number you want to call must be reachable by someone calling it from California in order to reach it with Skype.
You can simply test this before you leave for Thailand with Skype. If you are not able to call that number with Skype, try to see if your bank offers a non-free number you can call. Sometimes these are listed as "Call Collect" numbers or numbers specifically for use outside the USA. You'd have to purchase Skype Credit or a subscription to call the non-free numbers, but that could be an option.
Good luck!
Patrick
Location/Ubicacion: Arizona USA
Time Zone/Hora Local: UTC/GMT -7
If this message has adequately addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others.
Si esto mensaje le ha ayudado, por favor haga clic en "Aceptar como solución". Si encuentra un mensaje útil, por favor "Da Kudos" al final del mensaje, por lo que esta información puede beneficiar a otros.
I am not a Skype employee. No soy un empleado de Skype.

The telephones numbers do not let me click and call with skype in firefox 5, how do I get firefox 4 again?

Before with firefox 3 and 4 I could call phone numbers with skype directly off of the web page by clicking the phone number. This feature no longer functions in firefox 5. What do i need to do. When I go to your website I can only download 5. I would like to back up a version if I need to because I use this skype calling numbers function daily. Please help

Firefox 4.0 is no longer available. You can download to the previous secure version which is 3.6.19 from here: [http://www.mozilla.com/en-US/firefox/all-older.html Download Firefox v3.6.19]
To ensure a clean installation, please do the following: 
#Go to "Programs and Features" in Control Panel and remove "Mozilla Firefox" choosing to keep your bookmarks, customizations etc., (''don't checkmark the box''). 
#Then reboot, open Windows Explorer, navigate to C:\Program Files and delete the folder called "Mozilla Firefox". 
#Finally run the installation file you downloaded earlier. 
Your bookmarks, customizations etc., are maintained in a different location and will become available to you again once you complete the installation. 
Having said that, make a backup of your bookmarks as a precaution as follows: 
#Click the orange Firefox button, go to '''Bookmarks''', then '''Show All Bookmarks''' to open the Bookmarks Manager. 
#Click the link to '''Import and Backup '''and then click '''Export HTML '''and save the file somewhere.
You can use the same Bookmarks Manager to import the file you saved by choosing "Import HTML".

What's wrong with skype support?

Hi, I bought a subscription to call Finland, so I can call my 87yr. old mother. Worked fine for few months. This summer we took a trip to Finland to go see her. When we got back I tried calling her with Skype and her number was blocked by Skype. I contacted their support and they assured me this block would be taken away. I tried calling later againg and it was back. I now received email from Skype saying her number is blocked and I can't call it anymore with Skype???? I was told on the chat line it's because "fraudulent use" ??? My mother has not received any strange phone calls and she does not have a computer. I would sure like to talk with a supervisor, but even after requesting this on the chat all I get is this email giving me no reason why her number is suddenly blocked. This is VERY strange behavior by the Skype customer support. As a consequence of this I lost $4.99 on my subscription for not being able to make calls to my mom. I have now cancelled my subscription. Has anyone else had a similar situation?

Sorry to hear about your experience suffering from our ongoing anti-fraud activities. We take the security of user accounts and their product use very seriously which is why the number you were calling had to undergo an investigation. We've concluded that investigation and you should be able to call the number again now.
Please also check your personal messages here on the community (the envelope icon on top right) for more details in a personal message I sent you.
Follow the latest Skype Community News
↓ Did my reply answer your question? Accept it as a solution to help others, Thanks. ↓

What's wrong with my device? It does not allow me to make calls with Skype or Viber... Can someone help?

What's wrong with my device? It does not allow me to make calls with Skype or Viber... Can someone help?

My device is a Blackberry Z30. Skype and Viber just to work fine. I had to travel abroad and turn off data services while traveling outside of the US but the applications worked just fine with Wi-Fi. Suddenly they stopped working to make calls. I can still send text messages, but when I try to do a voice or video call (in the case of skype), it only says "failed call" and when I try to call using Viber I get an error message that says the I can not use the application until I complete my GMS call which is ridiculous because I'm not using the phone other than trying to place my call through Viber. The interesting thing is that when I go the permission page in my security settings everything indicates that its on but is gray out and I do not have access to change any setting. The same in both applications. The rest of the applications are accessible to change settings. I already uninstalled skype and reinstalled it and still the same problem. I'm ready to back up everything and wipe out the device to original settings.

Not given credit for subscription with skype to go...

hi, my skype to go number had expired,by mistake,I renewed,buying only 120 minutes to call the country of my subscription,I let that expired,and bought the unlimited minutes to that same country,which is supposed to have the skype to go number...that was never reactivated by Skype,and now they are telling me I need to pay again!! ???I have record of the payment which I showed to a person on the chat,and still cannot get the number reactivated...help PLEASE!!!!!

Hello,
I believe there's a small misunderstanding around Skype To Go versus subscription, but I'll gladly explain these.
Skype To Go is a number that you can call from a regular phone and Skype connects the call to your contact's phone using either Skype Credit or a subscription as payment method.
Here's a quick sketch that should explain how call is connected to your contact.
If you already have a subscription then once you set up Skype To Go you can use this subscription to cover calls to your friend. If your subscription covers your contact number then all you need to do is set up Skype To Go number
You can set up Skype To Go as shown here: http://www.skype.com/go/togo.setup
More information about Skype To Go:
https://support.skype.com/faq/FA1300
If answer was helpful please mark it with Kudos and if issue is resolved mark it with solution. This will help other users find this answer more easily. Thanks in advance!

Can't make outgoing call with Skype Connect

I have my Asterisk PBX configured with Skype Connect using SIP with TLS and SRTP. Most of my outgoing calls go through, but sometimes I can't get call out. I was able to leave asterisk console up and collect verbose and sip debug data. Can somebody help me diagnose why my calls aren't going through?
I've changed my external IP (I'm behind a NAT'd firewall) to 1.2.3.4 and my SIP profile's user ID to 11111111111111. and my domain name to test.com. If someone working for Skype needs that information they can email me and I'll send it privately.
My config:
[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://111111111111111:[email protected]
[skype]
type=friend
context=from-skype
dtmfmode=rfc2833
host=sip.skype.com
username=11111111111111
fromuser=11111111111111
secret=abcd12345
disallow=all
allow=ulaw
allow=alaw
nat=yes
fromdomain=sip.skype.com
insecure=port,invite
transport=tls
srtpcapable=yes
encryption=yes
SIP Debugging enabled
[2012-08-23 19:22:33] NOTICE[16552]: chan_sip.c:13465 sip_reregister: -- Re-registration for [email protected]
> doing dnsmgr_lookup for 'sip.skype.com'
> ast_get_srv: SRV lookup for '_sips._tcp.sip.skype.com' mapped to host 1.sip.skype.com, port 5061
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 63.209.144.201:5061:
REGISTER sip:sip.skype.com:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as6edf93cf
To: <sip:[email protected]>
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
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 200 OK
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
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[2012-08-23 19:22:33] NOTICE[17932]: chan_sip.c:21399 handle_response_register: Outbound Registration: Expiry for sip.skype.com is 45 sec (Scheduling reregistration in 30 s)
<--- SIP read from UDP:192.168.1.16:5060 --->
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
To: <sip:[email protected]>
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]
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
Looking for 19739928881 in home (domain asterisk.test.com)
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
<--- 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

No call out/in with skype subscription

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