First Time Making A Call With Skype

I just bought Skype credit to start making calls for the first time. In the account section I was asked to check my mobile phone for the information code and enter it below. But I need help for what my mobile phone information code is.
James Kwegir Ampiah

Frequently asked questions about Apple ID - http://support.apple.com/kb/HE37 --> Can I change the answers to the security questions for my Apple ID? --> Yes. You can change the answers to the security questions provided when you originally signed up for your Apple ID. Go to My Apple ID (http://appleid.apple.com/) and click Manage your account.
Some Solutions for Resetting Forgotten Security Questions - https://discussions.apple.com/docs/DOC-4551
Forgotten security questions - https://discussions.apple.com/message/18402551 and https://discussions.apple.com/message/18625296
More involved forgotten question issues - https://discussions.apple.com/thread/3961813
John Galt's tips (09&11/2012) - https://discussions.apple.com/message/19809294 and https://discussions.apple.com/message/20229239
If none of the above work, contact iTunes Support at http://www.apple.com/support/itunes/contact/ and follow the instructions to report the issue to the iTunes Store.

Similar Messages

First time making a purchase with this new mac...

when I type in my password, it prompts me to ask my security questions, but i can't remember one of them,any help?
thanks

Frequently asked questions about Apple ID - http://support.apple.com/kb/HE37 --> Can I change the answers to the security questions for my Apple ID? --> Yes. You can change the answers to the security questions provided when you originally signed up for your Apple ID. Go to My Apple ID (http://appleid.apple.com/) and click Manage your account.
Some Solutions for Resetting Forgotten Security Questions - https://discussions.apple.com/docs/DOC-4551
Forgotten security questions - https://discussions.apple.com/message/18402551 and https://discussions.apple.com/message/18625296
More involved forgotten question issues - https://discussions.apple.com/thread/3961813
John Galt's tips (09&11/2012) - https://discussions.apple.com/message/19809294 and https://discussions.apple.com/message/20229239
If none of the above work, contact iTunes Support at http://www.apple.com/support/itunes/contact/ and follow the instructions to report the issue to the iTunes Store.

HT204380 help. tried for first time to use FaceTime with another iPhone user & it would not work. "FaceTime failed". we are both on wi-fi networks & iTunes accounts are set up correctly.

help. tried for first time to use FaceTime with another iPhone user & call refused to connect. Call would dial, allow us to accept call, then error message "FaceTime failed" would appear & call disconnect. we are both on wi-fi networks, both phones have updated software & both enabled FaceTime with iTunes logged in. don't know what else to do to get FaceTime to connect. Someone please help.

Thanks Helen for the response. At least you can get some of the contacts to work. Absolutely none of mine will. Feel like my phone is cheating me out something it should do. Think I'm going to just have to dump all the Apple products in general because this phone hasn't worked right since I got it.

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.

HT202213 Home sharing problem: I have never set up home sharing but I feel I understand it. I recently (last 2 months or so) deauthorized, first time ever, all computers with my iTunes account. I am trying to set up home sharing but get iTunes says "You

Home sharing problem: I have never set up home sharing but I feel I the concept and how it works. I recently (last 2 months or so) deauthorized, first time ever, all computers with my iTunes account. I have 3 computers authorized. I am trying to set up home sharing but get iTunes says:
"Home Sharing could not be activated because this computer is not authorized for the Apple Id "#########@###.com". Would you like to authorize now?" I click "authorize" and get the next error " You cannont authorize more than 5 computers. You have already authorized 5 computers with this Apple ID. To authorize this computer you must first deauthorize one of the other computers."
Can someone help or shed more light on my problem?
Thanks,
Richard

Home Sharing is designed to work on your local network not across the internet/cloud.
Stuff is accessed under the Computers column where your local iTunes library on a local computer would appear.
Home Sharing would share your iTunes content (i.e. stuff stored in itunes on the computer, not in the cloud) with AppleTV or an iPad etc on the SAME network.
AppleTV2 will not be able to see itunes content on the work computer over the internet. It's not designed to. if the work computer was on the home network it would.
iCloud is in it's infancy and is not a mature product - iTunes TV Show purchases appear on AppleTV, but currently music does not unless you are subscribed to iTunes Match. I find this rather odd to be honest, along with the inability to buy music on AppleTV2. Movies purchased in iTunes are not authorised for iCloud viewing currently either.
Maybe it has something to do with iTunes Match 'getting in the way' - i think they assume you'll use that whereas you really want to be able to access Purchased music from the cloud without subscribing to itunes Match which is overkill for some.
AC

I just bought a Shuffle for my wife. I have a Touch. When I plugged the Shuffle in for the first time it sync'ed with all my Touch stuff and filled up. I have created a playlist that I would like to install on the Shuffle. How do I keep the Shuffle from s

I have a Touch. It sync's automaticly when I plug it in. I like this.
I bought a Shuffle for my wife today. I want to put a playlist on it. When I plugged it in for the first time, it sync'ed with all my Touch stuff and is now full.
How do I empty the Shuffle.
How can I plug the Shuffle in if it automaticly fills up with my Touch stuff?
I have windows xp.
Thanks,
Jeff

Some apps that you download on your iPhone may possibly be to new for the iPod touch 2nd gen. I have a iPhone 5 and a iPod touch 2nd generation myself. If you plug his ipod into iTunes and restore it from a backup of your iPhone than it will download the paid apps and music that are compatible with his ipod. Hope this helps.

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.

I have a Mac os x 10.58. Please could someone illustrate exactly what settings I should have on my computer to make video calls with Skype?

I have a mac OS X 10.58 Please could someone illustrate exactly what settings I should have on my computer to make video calls with Skype? Thanks.

Then all you need is an external compatible webcam with your Mac Mini. Skype is currently compatible with your Mac and operating system
http://www.skype.com/intl/en-us/get-skype/on-your-computer/macosx/

Good day my question is , does anyone of you experiencing when making a call with facetime the phone shuts off and restart

good day my question is , does anyone of you experiencing when making a call with facetime the phone shuts off and restart

See this veeeeeeeeeery long thread.
https://discussions.apple.com/thread/3404857
It's an issue with hundreds of phones. Some claim it's software related, some hardware, some say it's a mixture of both.
Some people have had temporary success with quitting the Phone.app / restoring as a new phone / restoring from an iCloud backup / disabling Siri / toggling speakerphone on & off...
Unfortunately all of the above only been temporary, with the problem recurring again after a few calls, or a few days etc.
So far the only solution that seems permanent has been to replace the phone. But even so, it's the luck of the draw to recieve a functional phone, and not another faulty one.
I'm completely annoyed as there is no Apple Store near me...... and Apple haven't even acknowledged the issue, despite the above thread being 34 pages long, as of writing.

Address Book – 'Call with Skype' launches VMWare Fusion

Clicking 'Call with Skype' against a number in Address Book launches Fusion and starts up XP. I had the Windows version of Skype installed in XP under Fusion but have removed it using the correct Add/Remove Software process. Clicking 'Call with Skype' once again starts the dialling from the Mac version of Skype. Weird and unwanted..... Any ideas to prevent this?

Not a problem afterall, as it turns out. Only shame on me for being a newb.
Though during my puzzle-solving process came across the skype-plugins for the AB. They are located in /library/Address Book Plug-Ins/ folder and are not automatically removed, when uninstalling Skype.
Names: SkypeABDialer.bundle and SkypeABSMS.bundle.

Remove 'Call with Skype' option in Contacts.app.

Hi,
I recently upgraded to OS X Yosemite from Mavericks. Before upgrading I uninstalled Skype. After upgrading, in Contacts.app, there exists two options 'Call with Skype' and 'Send SMS with Skype'. How do i remove it?

Found the answer.
1. I had installed Skype when I was using OS X Lion.
2. In Lion, Contacts.app was known as Address Book.app
3. So went to the folder '~/Library/Address Book Plug-Ins' and deleted the 'SkypeABDialer.bundle' & 'SkypeABSMS.bundle' files.
4. Problem Solved.
Thanks for the help anyways.

I just purchased an ibook for my mac for the first time and it started with two pages then switched to one with notes and i can't change it back. Anyone else having this problem?

I just purchased an ibook for my mac for the first time and it started with two pages then switched to one with notes and i can't change it back. Anyone else having this problem?

Up the top where the three buttons are (red yellow green) are three images. Click on the third image that looks like a notepad (not the first which is a library book), and that should get rid of 'Notes'. To read using two pages make the window bigger.

Making calls with Skype Number

First I was wondering if you need wifi to use a Skype Number?
Secondly, how much would it cost to make out-going calls using the Skype Number (would I need Skype credits)?
Thank you!
Solved!
Go to Solution.

Hi, 761w. and welcome to the Community,
Let us start by clearing up some confusion: a Skype Number is for incoming calls only. The sole purpose of having a Skype Number is to permit people who do not use Skype to call you using a real land line number, so you can receive their call on Skype. More about Skype Numbers here: Skype number
So, a Skype Number is not related in any way to calls placed to fixed or land lines or mobile numbers. You are correct; you will need to purchase credit or a subscription to cover the cost of these calls.
You are welcome!
Best 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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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

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".

First Time Making A Call With Skype

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