Transferring Calls- Why Skype Removed It

After using Skype for 5 years for my business, I finally have to let Skype go. I do not understand the underlying reason for Skype to remove the best feature of Skype for Business. Call transfer was the must have feature for us.<br><br>Why was it removed?<br><br>Please bring it back

This is the email repsonse I got from Skype Support when I asked why this feature was removed - especially as it is critical to business use.
We understand your concerns regarding the removal of the call transfer feature on the latest version of Skype application. We will be happy to explain this for you.
We continue to invest in the Windows Desktop platform to ensure we’re giving users all the capabilities of Skype they have come to love.
In recent versions of Skype for Windows Desktop, we have focused on providing users with features including improved video messaging and file sharing capabilities, and implementing a more intuitive way of sending and adding contact requests. We decided to remove call transfer functionality due to technical challenges in supporting this feature, which enabled us to focus on improving the quality of the user experience and develop the features that our customers enjoy the most.
We will look to readdress the possibility of an improved call forwarding function in future versions of Skype for Windows Desktop."
They obviously are not interested in the business user!
We are moving off Skype to a hosted PBX service.

Similar Messages

Why can I not make video calls using Skype?

I'm really baffled. I have a month old MacBook Air. I use Skype daily for work and personal business. Never had a problem until this evening. I went to make a video call and Skype was not linked with the camera - there was no option to make a video call and when I went into Preferences, it said I had no camera. I then used FaceTime - person I was calling also had FaceTime, so it wasn't an issue - but I've now spent an hour trying to solved the problem, including logging in and out of Skype, closing the app and reinstalling, all to no avail. The camera functions fine with FT and other apps - why will it not sync up with Skype after it's worked fine? Anyone have a clue? Thanks in advance.

See this thread:
https://discussions.apple.com/thread/5306216?tstart=0
FIX:
1. You need Time Machine
2. Go to folder /Library/CoreMediaIO/Plug-Ins/DAL/
3. Copy AppleCamera.plugin to good place (usb memory stick is the best place).
4. Go to Time machine in date that skype work fine.
5. Change AppleCamera.plugin with file from Time Machine
6. Restart system, Now skype need to work with camera.
From Apple:
MacBook Air (Mid 2013): FaceTime HD Camera might not work after OS X 10.8.5 update

I have just bought macbook air, cant make video calls on skype... why??

i have just bought macbook air, cant make video calls on skype... why??

See this thread:
https://discussions.apple.com/thread/5306216?tstart=0
FIX:
1. You need Time Machine
2. Go to folder /Library/CoreMediaIO/Plug-Ins/DAL/
3. Copy AppleCamera.plugin to good place (usb memory stick is the best place).
4. Go to Time machine in date that skype work fine.
5. Change AppleCamera.plugin with file from Time Machine
6. Restart system, Now skype need to work with camera.
If that doesnt pass,
Suggest looking here, much success has been found:
http://community.skype.com/t5/Mac/OS-X-10-8-5-broke-Video-on-MacBook-Air/td-p/18 91729/page/4

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.

Call transferring- Not transferring calls

I use call tranferring alot but my skype does not tranfer calls to my local mobile number.
Whenever skype is active, i get calls on skype. I started using the call transferring feature so that i can have my skype calls transferred on my local mobile numbers when internet connection is not available.
But my calls are not been tranferred now.
Why is this happening?

Hi,
well, Tanner has a point, it must be solved processwise.
The company I used to work for had a customer with a large UCCE contact center and they asked for a report that would show this kind of suspicious activity. UCCE has actually a way to mark these calls, unfortunately, as far as I know, there is no such thing in UCCX. The rule was simple: you can be caught only once doing that.
G.

AutoAttendant not transferring calls in CUE 8.6 to CME 9.1

**Issue is calls are not transferring once extension or directory is pressed on the phone.
I upgraded our 2811 CME 7.1/CUE 7.0 router to 2911 15.2 CME 9.1/CUE8.6. It has CME and V/k9 license enabled. I uploaded the configuration used on the 2811 to the 2911. I added the ip addresses in the ip trusted list. Calls come in and out if I point the translation list to the phone and not to the auto attendant pilot number. Incase if it was toll fraud issue I disabled it but still a no go.
Everything worked fine on the 2811. I can't understand why it would not work on the 2911 15.2. Has anyone had any issues when upgrading from 12.4 to 15.x with auto attendant not transferring calls?
I ran a trace on the Cue when placed a call to extension 114:
20.10.10.1- CME interface address
20.10.10.5- CUE interface address
659 09/12 11:43:39.418 ACCN SIPS 0 Call.transferFailed(114, RESOURCE_NOT_ACKNOWLEDGING) SIPCallContact[id=33,type=Cisco SIP Call,implId=5738D3Dse-20-10-10-5# [email protected],active=true,state=CALL_ANSWERED,inbound=true,handled=false,locale=en_US

Ok I'm starting to think its a dtmf issue. I checked to see if I call from internal to the AA and dial an extension if it works but the same issue. I ran the debug voip ccapi inout and I see consume mask is not set. What would that indicate? Could that be the problem?
This is right when I dialed extension 114
2811-TEST#
001288: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001289: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001290: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001291: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
001292: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001293: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001294: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001295: Sep 16 14:59:23.198: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
001296: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
2811-TEST#
001297: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001298: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
001299: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
2811-TEST#
At this point theres silence on the phone I see this message:
2811-TEST#
001300: Sep 16 14:59:28.914: //55/D62BA3D180C8/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=FALSE, Tone=Null,
   Tone Direction=Sum Network, Params=0x0, Call Id=55
001301: Sep 16 14:59:28.918: //56/D62BA3D180C8/CCAPI/cc_api_call_feature:
   Feature Type=50, Interface=0x49FC2B80, Call Id=56
2811-TEST#
2811-TEST#
At this point I get the message "the phone number you are trying to reach" then the call disconnects
2811-TEST#
001242: Sep 16 14:48:57.719: %VOICE_IEC-3-GW: SIP: Internal Error (ACK wait timeout): IEC=1.1.129.7.67.0 on callID 52 GUID=5D21AC143CE711E480BCEEC6B624839
Also I checked show voice iec description:
2811-TEST#show voice iec description 1.1.129.7.67.0
    IEC Version: 1
    Entity: 1 (Gateway)
    Category: 129 (Call setup timeout)
    Subsystem: 7 (SIP)
    Error: 67 (ACK wait timeout)
    Diagnostic Code: 0
2811-TEST#

CTI OS desktop client freezes after release of transferred call

Using CTI OS desktop v7.5(9), UCCE 7.5(9), Endpoint is CIPC 8.6.3 on CUCM 7.1.3
ACD call received by Agent 1 on ext 1111
Agent 1 transfers to 1112
Agent 2 answers transferred call on 1112 using answer call button.
Agent 2 disconnects using Release Call button on CTI OS desktop
Call disconnects but stays in grid with status 'Talking' and buttons on CTI OS desktop become unresponsive. CIPC is fully usable independently.
Agent 2 can reset by killing the CTIOS Desktop process and restarting.
This condition ONLY occurs as the result of an inbound transfer to Agent 2. It does not appear to happen when a call is transferred to Agent 1 in a similar scenario (Agent 1 is on a 7942)
Any thoughts as to why this occurs?
Thanks so much for any assistance!

Hi,
looks like for some reason the CIPC does not informs the CUCM about the call released event. Or it does, it's only the JTAPI gateway that misses it.
Did you try upgrading/downgrading the firmware of the CUPC?
Also, I am not quite sure that CIPC 8.x is supported. It might be, but it certainly is not on the list:
http://docwiki.cisco.com/wiki/Unified_CCE_Software_Compatibility_Matrix_for_ALL_7.x_Releases
G.

If I call my Skype-To-Go number (using my Skype re...

If I call my Skype-To-Go number (using my Skype registered CELL phone) to call a LANDLINE phone in Japan, WHY will I be charged the JAPAN MOBILE rate of $0.18/minute INSTEAD OF the JAPAN LANDLINE rate of $0.03/minute (+ VAT + connection fee)?
Solved!
Go to Solution.

Hello,
I need your STG # please.
(VAT is not charged on call rates in any location).
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.

Can I call a Skype to Go USA number from my Skype ...

Hi There,
I'm travelling to Europe (France & Spain) from India. Since Skype hasn't started the Skype to Go in India, I was thinking of going for a Skype to Go number in the USA, which my family will call from their existing Skype credit (through WiFi on iPhone). One Skype to Go number of the USA will be configured to call France and another to call Spain.Therefore, when someone calls my USA number (either through a landline in India or through Skype), it will be routed to my France/Spain local mobile number, which I will purchase on arrival. I wish to know if this (calling from Skype to the Skype to Go number) is really possible. If not, what can be the solution. Also kindly suggest any alternate route other than the USA which will be more reasonable.
One may ask why take so much trouble, and rather not call directly from Skype to the local France/Spain mobile number. The issue is that I wish to be accessible to other important people who do not use Skype, and will call from their Indian landlines to the USA (caling from India-USA on landline is quite reasonable compared to other countries). I will share my Skype to Go numbers with whom I want to be in touch when abroad.
I chose USA since India-USA on Skype/Indian landline is cheap & USA-Europe is again reasonable with unlimted packages on Skype pay as you go. Please get back to me.

smqld-gc wrote:
I want to call someone in the country Georgia? Are they able to get a Skype to Go number?
Hello,
I'm afraid not. Below is the list of countries where SkypeToGo #s are available:
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.

Ios 7 no longer displays my credit balance after I send/call, why?

I am a pay as you go customer. Since I have updated my phone to ios7, I am no longer notified of my credit balance each and every time I send a message or end a call. why is this and is there a quick fix?
Regards

Hi. I have a further update!
I called Tesco they had me call the handset techs but they couldn't help and thought it was a billing/account/notification issue so I was transfered. I then spoke to a helpful tech support person who reset the notifications (which they do centrally) and restarted the phone. No effect. The on chatting she implied that O2 was probably still checking out the 7.02 update and the compatibilities of the update. She pointed out that the carriers are often behind the Apple curve as Apple calls the shots on updates, whereas the carriers directly influence the rate of Android and Windows Phone releases. She said they could provide no more help and that I should contact Apple in case it was a known issue at their end.
So I have two live chats today with Apple USA. Awesome support. Very nice experience but no solution. They agreed with my diagnosis that the issue was either carrier settings or Apple software. We ran a suite of diagnostics and they thought it seemed the "show balance" screen was crashing or causing error. They had be restore a vanilla 7.02, which I did. I then blasted through setup avoiding restoring backup. I only added in wifi and the icloud account for running diagnostics. I did this twice, first eschewing the Tesco carrier update, no effect. The second time I accepted the carrier update, no effect.
I have just restore for the third time and re-established my data from the backup. No effect.
Had another live chat with the States as UK support is shut. They said I have done everything possible and that they was to escalate to the software engineers. However, I need to call in for that. I have pdfs of all the records and the support IDs. But to be honest the phone works in all other ways and I don't want to RMA it. I am convinced more that ever that the issue is a carrier setting issue. The settings, sim, apps and data are identical but work on an iPhone 5 and not an iPhone 5s. I just think O2 and its partners need to tweak the carrier settings or notifications somehow for the 5s. As a result I'm tempted to hold out for future updates.
Any thoughts or solutions out there?

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

Query for transferred calls

I am attempting to create a report of transferred calls from the calldetailrecord table in the CDR database using Call Manager 4.2. I have read the documentation provided for the table and it looks to me like if there is a call from 555-1212 to ext 100 and ext 100 transfers it to 101, I should see something like this:
CallingPartyNumber,FinalCalledPartyNumber
555-1212,100
100,101
555-1212,101
However, I have a few problems. First of all, I can't find anything like this pattern in over 4 months worth of data from our system, where lots of calls are transferred each day, although right now I'm just kind of combing through using different queries by time etc. Also, I have a lot of records where the CallingPartyNumber column is blank, or the OriginalCalledPartyNumber field has a weird value starting with b, like b5627368934. Also I cannot seem to determine which column would link together all of the segments of a call for a transferred call, so that I can see all transfers for one actual call (which is why I'm just trying all sorts of different queries right now). Can anyone help me out with an example query to find some transferred calls and list out all of the transfers for each call?
Thanks

Hi,
OriginalCalledPartyNumber beginning with 'b' means conference call to conference Bridge.
Transfered calls you can analyze using lastRedirectDN (redirecting number) field.
Empty callingPartyNumber means that PSTN caller has restricted number presentation.
Detailed CDR description for CCM 4.2.3 you can find at:
http://www.cisco.com/univercd/cc/td/doc/product/voice/vpdd/cdd/4_2/4_2_3/cdr423.pdf
regards,
pk

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.

Report on transferred calls

I need to report on transferred calls. Agents are using standard CAD and selecting the transfer icon. Within the transfer form, they use the softphone to dial various 800 numbers. I need to create a report that I can group by 800 numbers called using the CAD transfer option. Does this come from the call manager? We recently implemented CVP. Are there any CVP reports that would provide this information?

Hi Ryan
I'm afraid there isn't one - only the CSQ Activity reports and the CSQ Service Level Priority Summary reports seem to show the service level info, and those are summary reports rather than reports that show call lists.
You would need to get a custom report put together for this... it's an Informix SQL procedure-writing and Crystal Reports exercise to do that.
Regards
Aaron

HT1719 When I try to drag a song to my shuffle it says "this song is in iCloud and can't be transferred. Why and what is iCloud I havsn't downloades any such thing. Help me get back to the old and simple way of transferring my songs!!!~

When I try to drag a song to my shuffle it says " this song is in iCloud and can't be transfered. Why is it ? I never downloaded anything like iCloud. I just want to be able to move my music where and when I waNT TO. i HAVE BEEN UNTILL i DOWNLOADED NEWEST VERSION OF ItUNES. gET ME OUT OF THIS!!!

thats tough man, yeah exchanging it is the best option. But why do people always leave a message saying" i'll probably just by an ipod" and it's usually followed by, "even though i don't like apple or ipod" . I mean fine, if u feel like giving up on the zen micro thats fine, i personally don't care if you're going to buy an ipod. for god sakes ipod isn't even the next alternati've mp3 players, theres plenty. Are u just trying to get the creative people saying "oh man we lost another one to ipod" cuz i'm sure they don't lose sleep over u wanting to buy an ipod instead of a zen. *done ranting*
Message Edited by bleakreserve on 02-3-2005 06:34 AM

Transferring Calls- Why Skype Removed It

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