How do I dial letters while calling with Skype?

Hi! I needed to dail several letters during a phone call, as part of a recognition serial number, but I do not know how to do that! Could anyone help me out? Thank you! I searched similar questions, but all I got is how to dial letters as part of the phone number. Unfortunately, I have to dial the specific letter instead of having the phone/Skype figure it out for me (e.g. I need to input "E" but I cannot just dial "3" to have Skype figure out and pick "E" among D, E, and F...) Anyone?? Thank you so much!!

Contact sprint for direction on how to use the calling card.

Similar Messages

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.

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 make appear the "phone with skype"-link, ...

How do I make appear the "phone with skype"-link, next to tel numbers on websites, that I visit ?

Hi, filipfirmout, and welcome to the Community,
In the Windows version, the settings to enable this are here:
Tools -> Options -> Advanced -> check in Use Skype to call to links on the web -> Save.
There is also this: http://www.skype.com/en/download-skype/click-to-call/
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!
Trustworthy information: Brian Krebs: 3 Basic Rules for Online Safety and Consumer Reports: Guide to Internet Security Online Safety Tip: Change your passwords often!

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 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/

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.

How to compare a parameter while calling a function?

Hello,
I have a powershell script with a function doing a switch process. While calling the function in the script I want to compare if the parameter is contained in the switch condition of the function.
function test{
[CmdletBinding()]
Param(
[Parameter(
Mandatory=$False,
ValueFromPipeline=$True,
ValueFromPipelineByPropertyName=$False
[string]$InputValue
$Param = switch($InputValue){
A {"string1,string2,string3"}
B {"string1,string2,string3"}
C {"string1,string2,string3"}
Return $Param}
If ((test($search)) -match ""){
$scrDynMSGArry = (test($search)).Split(",")
$NV = [PSCustomObject]@{
Param = $ParamString1 = $scrDynMSGArry[0]
String2 = $scrDynMSGArry[1]
String3 = $scrDynMSGArry[2]
Error message is: InvokeMethodOnNull for doing the split and reading the array.
I think there is a problem doing the comparison but have no idea to solve this.
Can anybody help?
Regards, Doreen

This appears to be what you are trying to do:
\_(ツ)_/
From how he is wording things, it sounds to me that if I pass $search the letter F, he wants to check to see if the switch statement contains a case for F, and if not do something. I do not think this can be done but he could always have a Default case,
returning something to indicate they need to choose something else, or the default parameters needed for whatever he is doing to work.
If you find that my post has answered your question, please mark it as the answer. If you find my post to be helpful in anyway, please click vote as helpful.
Don't Retire Technet
She please ;-)
But clayman2 is right: $search can be F, G, ... but I want to use the function "test" only if its A,B or C.
I thought if $search is F it does not enter the if part beacause the condition is not true:
If ((test($search)) -match "")
You think thats not possible?
Would it be better to compare $search with my conditions (A, B or C are defined and known) for itself and then enter the function "test"? Can I have or conditions in the if part like
If ($search -eq "A"){$scrDynMSGArry = (test($search)).Split(",")
$NV = [PSCustomObject]@{
String1 = $scrDynMSGArry[0]
String2= $scrDynMSGArry[1]
String3= $scrDynMSGArry[2]
elseif ($search -eq "B"){$scrDynMSGArry = (test($search)).Split(",")
elseif ($search -eq "C"){$scrDynMSGArry = (test($search)).Split(",")

How can I do a video call with my friend?

I am new to the tech world,61 yr.old senior citizen. I do not know anything yet, a tech connected to Skype so I could see and talk live to my friend via of video but she has an Apple i-phone so I can't yet. I was wondering if she's connected to face book can I contact her via Skype to her Facebook contact she has on her i-phone that uses Apple's version of their Skype? Since Skype is free to the Facebook users. Sorry, I can't be a volunteer. I can ask questions that alot of Seniors like me would be able to learn about this community and we could read some topics that we could use to start us off and reply what works in our circumstances and eventually maybe develop the skills that would makes us useful to the Program.
This post was transferred from its previous location to create its own new topic here; its subject and/or title has been edited to differentiate the post from other inquiries and to reflect the post's content. A link to this post appears where the post was originally added.

Dear Newbie-61,
Welcome to the Skype Community!
All calls between Skype users are free, regardless of which computer or handheld phone or tablet they use to call each other. What is important is that the Skype software must be downloaded and installed on the particular computer, laptop, tablet, or smart phone that you use. Depending upon what kind of set-up you have, you may or may not need a separate headset. Laptop computers and smart phones use the microphone and camera that comes with them.
To download Skype, please start by clicking on this link: Download Skype After you download and install Skype, sign on using the account you used to post your message on this Community.
Then. you and your friend can exchange contact invitations. This FAQ article explains how to do it:
https://support.skype.com/en/faq/FA3681/how-do-i-make-free-voice-and-video-calls-on-skype-for-window...
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!
Trustworthy information: Brian Krebs: 3 Basic Rules for Online Safety and Consumer Reports: Guide to Internet Security Online Safety Tip: Change your passwords often!

HT5661 how to set pic display while calling to contact ?

how to set pic display on calling to contact person ?

Hello HaD,
The image control is installed with either the Vision Development Module or the NI-IMAQ driver, and is not a standard component found in LabVIEW. If you have questions about the image display control or vision functions it would be better to post in the Machine Vision forum:
http://forums.ni.com/ni/board?board.id=200
In regards to your question, this menu is not available as a runtime property, and therefore the user of program will not be able to set it directly through the control. Instead, you will need to allow the user to set it programmatically in LabVIEW by use of a property node.
To create this property node, right-click on the image display icon on your program's block diagram, and select:
Create » Property Node » Palette » Palette Type.
You will then be able to change the value of this property by right-clicking on the property node icon and selecting "Change All To Write". To create a control on the front panel that the user change at runtime, right-click on the “Palette Type” input, select:
Create » Control
If you have any questions about the application architecture required to implement this design, please continue to post in the LabVIEW forum.
I hope this helps.
Best regards,
Jasper

How to listen to music while on a skype call

Can any 1 tell me how can I listen to music while on a skype call every time I answer or make a skype call it stops my music why and how can I fix it

I can't either!

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".

How to open .ogg files while browsing with a player not firefox

when selecting an .ogg link firefox opens the file in the current browser window. how can i associate this file type with a player of my choice instead. i have made this association in windows but no luck with firefox.
txs

Which application opens the file if you double-click such a file in Windows Explorer?
If the default program is Word then Firefox should offer that application as the default.
In the case of an attachment then it may not be likely that the file is send with the correct MIME type for a doc or docx file, but possibly a generic type like "application/octet-stream" that will make Firefox want to save the file.
*https://support.mozilla.org/kb/change-firefox-behavior-when-open-file

I phone 4 video call with skype

hi everybody ,
can some body tell me how can i video chat with skype ?
I have i phone 4
thank you

first of all you would ask in the iphone forum and not the appletv forum.
second of all, skype on the iphone does not support video calls.

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

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