International call forwarding

You can enable call forwarding to a local/international number but you would have to pay the cost of each call with credit or a subscription. Colombia Skype rates If you are forwarding to a standard Colombia landline you could utilize something like the Unlimited Latin America subscription to cover the cost.

is it possible to have international call forwarding with my usa skype number to a colombian number??
(calling my georgia skype number then going straight to a colombian number]

Similar Messages

Call forwarding to an International Number

I am working overseas and have a prepaid phone with a local number in that country. Prior to leaving the US again, can I enter the international number of my prepaid phone into the call forwarding feature on my iPhone and have my calls forwarded to that phone? I assume that the person calling my iPhone will not be charged, but will I be charged for making an international phone call each time a call is forwarded to my international phone number? Are there any other charges involved? Has anyone done this before that can tell me how well it works?

To make sure you should contact AT&T but if you set up call forwarding to an international number, you will be the one charged for the international call since your account/phone is dialing the final number.
Again, find out for sure from AT&T who gets the charge.

Call forwarding from an international Skype number

Ok so here is my setup. I live in Ottawa, Canada and I have an Australian Skype number. What I want to know is, if I put a forwarding number on my australian Skype number to a landline Canadian number and someone from Canada tries to call my Australian Skype number, what rate are they charged when they connect to my forwarded Canadian number and what rate is my Canadian number charged?

To make sure you should contact AT&T but if you set up call forwarding to an international number, you will be the one charged for the international call since your account/phone is dialing the final number.
Again, find out for sure from AT&T who gets the charge.

HTC Droid DNA - How to enable Call Forwarding in GSM mode in International Network

Hi,
I have bought HTC Droid DNA (Verizon) from ebay in India and I am able to use any 2G or 3G GSM network as it not locked by Verizon however there is no option to enable CALL Forwarding in phone settings.
Please help me and let me know how to enable call forwarding in HTC Droid DNA in GSM mode,

I don't think you can, call forwarding is enabled by Verizon here in the states. Maybe check with your carrier?

Unity 4.0 - Call Forwarding and Voice Mail

Here is the situation:
We have a DN (5301) that is not associated with a Unity mailbox but is on a 7970 phone. This extension is an "on call" number that is always forwarded to a technicans phone (local 4 digit ext or cell phone).
When a person calls 5301 I want the voicemail of the final destination to answer.
For instance if I had 5301 forwarded to 2000 - I would want 2000's voicemail to answer.
Is there a way to set this up?
Thanks in advance.
Jeff

Hi Jeff,
Sadly this cannot be changed until Unity 5.x (the ability to choose "Last Redirecting Number" in not available in any other Unity version);
Here are the Unity 5.0 release notes;
Route Forwarded Calls by the First or Last Redirecting Number
Cisco Unity supports the option of routing calls based on either the first or last redirecting number when a call is forwarded to Cisco Unity.
Note the following:
This option requires Cisco Unity-CM TSP 8.1(2) or later.
This option is not supported by integrations through PIMG units.
This option can be changed through the Advanced Settings Tool (AST), which is available in Tools Depot.
http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/release/notes/501curelnotes.html#wp507371
Call Information Exchanged by the Phone System and Cisco Unity
The phone system and Cisco Unity exchange call information to manage calls and to make the integration features possible. With each call, the following call information is typically passed between the phone system and Cisco Unity:
â¢The extension of the called party.
â¢The extension of the calling party (for internal calls) or the phone number of the calling party (if it is an external call and the phone system supports caller ID).
â¢The reason for the forward (the extension is busy, does not answer, or is set to forward all calls). There is also a reason code for Direct Calls.
Cisco Unified Communications Manager SCCP and SIP trunk integrations can also provide the following call information (the choice of first and last redirecting number is set in the Advanced Settings Tool, which is available in Tools Depot):
â¢Called number
â¢First redirecting number
â¢Last redirecting number
Note Cisco Unity can use either the first redirecting number or last redirecting number, depending on the setting in the Advanced Settings Tool, which is available in Tools Depot.
http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/design/guide/5xcudg060.html#wp1040786
If this was just a one-up type of setup you can configure a Voicemail profile (in CCM) for 2000 and apply it to 5301 that will allow this type of Call Forward to 2000's mailbox. The fact that you need this for Multiple Tech's will not work. Is there any way the Techs could use a Shared Line? Then these solutions could be adapoted.
Or in Unity set up Alternate Extensions so that User A is an Alternate Extension for User B etc. Sharing a Cisco Unity Voice Mail Box between Two or More IP Phones
Configure Alternate Extensions
Open the Unity System Administrator web page.
Navigate to the subscriber's profile. Select Subscribers > Find and Select a Subscriber > Enter Subscriber Information then click Find and click the Subscriber's name for the subscriber that owns the primary phone (2000).
When the subscriber page comes up, select the Alternate Extensions option and click Add.
Enter the alternate extension number (in this case 5301) and click the Save icon.
From this good Unity doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_configuration_example09186a008015ceec.shtml#steps
Hope this helps!
Rob

Call forward to external number(mobile)

Dears please help me on this
voice translation-rule 1
rule 1 /2837599/ /599/
rule 6 /2837596/ /596/
rule 7 /.*2837555/ /123/
2837... are my SIP DID nos
123 is my AA extn
596 and 599 is an ip phone exten
i need to transfer directly to an external no (mobile no) when i call 2837596 from outside without extension
what is the config to be done

dears , i tried it but call not forwarding please need our help
voice translation-rule 1
rule 13 /.*2837499/ /499/
ephone-dn 499 dual-line
number 499
label website
description 499
call-forward all 90504495705
corlist incoming user-international
ephone 37
device-security-mode none
video
mac-address 001E.F727.F567
ephone-template 16
username "700" password 700
type 7911
button 1:499
pin 1700

Call Forward to PSTN numbers not working

We have a CUCM version 8.6. The call farward to internal extensions are working fine. But, to the PSTN numbers it is now working. We are able to call the PSTN numbers without any issue. Can somebody help us on this?

Hi Siva,
The most likely cause for this type of issue is the CSS that is applied @ the Call Forward All level on
the DN config page. Check out the CFWDALL CSS to make sure they are set with a level with
access to PSTN numbers
Cheers!
Rob
"Your life is worth much more than gold."
- Bob Marley

CUCM 8.6 Call Forwarding to External Number Issue

Hello,
Call forwarding worked without problems, we could forward our phones to external numbers and everything was ok, when somebody called to my phone, I could got the call to my cell phone.
But now when I forward my phone to external number and try to call to my phone I get busy trigger.
We didn't change configuration or install any update.
I think its my ISP-s problem, to whom we have SIP Trunk.
I don't understand log file, so can you tell what is the problem?
Here is log:
057729XXXX is called party, cell phone number
original calling party number is 240XXXXX, but it is forwarded to 2484XXX
INVITE sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191815-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5060>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052091 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29790 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 2|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpHandler::fireSignal - SEND: index = 2, handler = 0xb2d59c98|*^*^*
17:34:18.526 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP-IP:5060|*^*^*
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1172, ISP-IP:5060)|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.536 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 358 from ISP-IP:[5060]:
[12623361,NET]
SIP/2.0 100 Trying
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/ccsip_spi_get_msg_type returned: 2 for event 1|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/context=(nil)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/gConnTab=0xf484290, addr=ISP-IP, port=5060, connid=2, transport=UDP|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Return existing connection for port 5060 connId 2|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Checking Invite Dialog|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0xb1b50c90/INVITE response with no RSEQ - disable IS_REL1XX|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.561 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
[12623362,NET]
SIP/2.0 403 Forbidden
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
Contact: <sip:ISP-IP:5060>
[12623363,NET]
ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29792 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 0|*^*^*
17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
[12623365,NET]
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0
17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
[12623366,NET]
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0
|2,100,230,1.4901099^ISP's-Other-IP^*
[12623367,NET]
ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

SIP/2.0 403 Forbidden error
If your router is sending a SIP/2.0 403 Forbidden error to the SIP server you are registered to, there is a good chance your router is blocking the incoming call due to the toll-faud prevention feature that was added to IOS version 15.1(2)T.
How to Identify if TOLLFRAUD_APP is Blocking Your Call
If the TOLLFRAUD_APP is rejecting the call, it generates a Q.850       disconnect cause value of 21, which represents ‘Call Rejected’. The       debug voip ccapi inout command can be run to       identify the cause value.
Additionally, voice iec syslog can be       enabled to further verify if the call failure is a result of the toll-fraud       prevention. This configuration, which is often handy to troubleshoot the origin       of failure from a gateway perspective, will print out that the call is being       rejected due to toll call fraud. The CCAPI and Voice IEC output is demonstrated       in this debug output:
%VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
IEC=1.1.228.3.31.0 on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
   Context=0x49EC9978
000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
   Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
The Q.850 disconnect value that is returned for blocked calls can also       be changed from the default of 21 with this command:
voice service voip
ip address trusted call-block cause
How to Return to Pre-15.1(2)T Behavior
Source IP Address Trust List
There are three ways to return to the previous behavior of voice       gateways before this trusted address toll-fraud prevention feature was       implemented. All of these configurations require that you are already running       15.1(2)T in order for you to make the configuration change.
Explicitly enable those source IP addresses from which you would like           to add to the trusted list for legitimate VoIP calls. Up to 100 entries can be           defined. This below configuration accepts calls from those host           203.0.113.100/32, as well as from the network 192.0.2.0/24. Call setups from           all other hosts are rejected. This is the recommended method from a voice           security perspective.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
Configure the router to accept incoming call setups from all source           IP addresses.
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
Disable the toll-fraud prevention application completely.
voice service voip
no ip address trusted authenticate
Two-Stage Dialing
If two-stage dialing is required, the following can be configured to       return behavior to match previous releases.
For inbound ISDN calls:
voice service pots
no direct-inward-dial isdn
For inbound FXO calls:
voice-port
secondary dialtone

Call Forward to Voicemail

I am having an issue when forwarding to another extension. I know this is probably a simple answer but I'm stumped.
I am going into the DN Configuration page for extension A. Under Call Forward, for the destination I am entering extension B for:
Forward Busy Internal
Forward Busy External
Forward No Answer Internal
Forward No Answer External
Forward No Coverage Internal
Forward No Coverage External
Forward on CTI Failure
Forward Unregistered Internal
Forward Unregistered External
And I am unchecking the Voicemail check boxes.
Then I am doing the same for extension B.
But when I have it setup this way, each extension will bounce to the other and will not go to voicemail. It will forward back and forth.
Any suggestions?

Hi,
If at least one of these phones is set to CF to VM then it will, if not, then no.
If none of your phones is set to CF to VM CUCM will not send them to VM, that is expected, if you need to ring, phone A, and if it is not answered to go to phone B, C... and so on, and send the caller to VM after you have reached all of these then use a hunt group, (the pilot can be set to CF to VM if nobody answers), if you need to ring all phones at the same time so someone can pick this up, use a hunt group with a broadcast logic.
If this is for a single user, check 'single number reach' (SNR) or mobility on CUCM.
Bottom line, there is no way to send a caller to VM if none of the phones is set to CF to VM.
HTH
Chris.

Is it possible to separate call forward unregistered & busy on a device profile?

I've got the following situation cropping up a surprising amount at our site:
User has 2 jobs within our institution. Works 3 days a week on job 1, 2 days on job 2. Each job is billed to a different cost centre, and the user doesn't want to be getting calls for job 1 when they're supposed to be working on job 2, and vice versa.
As such, we've created them 2 device profiles. When they log into the phone at the desk of whichever job they're currently working, they get prompted for which profile they want to log in to. The other device profile (if still logged in to the phone on the desk of the other job) is then forcibly logged out.
When profile 1, for job 1, is logged in, but the user is already on a call, they want incoming calls to that extension to be directed to their voicemail (i.e. set Call Forward Busy [Internal|External] to send to voicemail). They can then check voicemail and follow up on the call as soon as they're off the current one.
When profile 1/job 1 is logged *out*, i.e. they're currently working job 2, they want incoming calls to job 1's extension to be immediately diverted to a colleage within the same job 1 team.
I thought I could do this by utilising Call Forward Unregistered [Internal|External], but this does not seem to be the case. When a device profile is not logged in to a device it seems like the busy trigger just gets treated as 0 so the value of Call Forward Busy is followed. I can't see any situation where Call Forward Unregistered is ever utilised if an extension is only associated with a device profile.
Is there any way to do what I want (without massively convoluted configuration on Call Manager)? If not, do people think this is worth raising as a feature request (or bug in expected behaviour) for later versions of Call Manager?
We're currently on CUCM 8.5, FYI, in case this is something that's already been updated in version 9 or later.

Some progress on second idea. Saved a copy of universal access plist with cursor set to large then set the cursor to small again. Replacing the plist file had no effect until I went into the UA pane and changed a setting at which point it must rewrite the file and refresh because the cursor size also changed at this point.
Tried the same again and restarted Finder, no effect. Also tried altering another pref pane instead with no effect. Need a way to force the computer to look at the plist files, no idea how though. :-)

I changed my data plan from 6g to 8g while my daughter who attends college outside of the US at Toronto Canada (and we have on a international calling and international data plan) was on spring break at her grandparents house here in the US. I made the ch

I changed my data plan from 6g to 8g while my daughter who attends college outside of the US at Toronto Canada (and we have on a international calling and international data plan) was on spring break at her grandparents house here in the US. I made the change online since I had been waiting on the phone for over 10 minutes for a customer service rep to come available. Well when I made the change online since that seems to be the thing that Verizon wants it's customers to do and I didn't see all the different plans available and just did the upgrade to 8g. Next bill had over $900 in roaming charges on her phone line. I called the 1-800 number and waiting for a service rep and after 20 minutes of waiting and being put on hold was told it was the customers mistake and there was nothing they could do.Thanks for nothing. I called back after thinking about it and wondered why changing a data plan for the phones in the US would change a international call plan. Waiting over 10 minutes again between waiting for a service rep and hold for one to answer the call. Gave her all the information about it and she said she would call back. Well, 4 days later over the weekend she had nevered called back. So on the phone again for the third time and after 20 plus minutes again was told that when I did it online I click the plan that didn't include international call only the data plan. Explained that I never saw the difference in the plan packages so put on hold again and was told that they could credit $100 to my bill. Wow, thanks alot !!! We have been Verizon customers for probably atleast 12 years and this is how you treat your long term customers?

Verizon Wireless Customer Support wrote:
AHARDY454,
We definitely want to review options on what has happened. We are now connection, so you can hover over my username and send me a Direct Meesage so we can review your account information. We look forward to reviewing.
Thank you,
TonyG_VZW
Follow us on Twitter @VZWSupport
TonyG_VZW they can't exactly hover over your username unless you actually link it in your post. The generic username for all the reps just doesn't fly.

Call Forwarding / Displayed Number on Forwarding target with H.323 Gateway

Hi Community,
i´m wondering if there is sort of a simple way to get this working properly.
Scenario:
Germany, variable dial plan, no fixed NANP, and we have ClipNoScreening ;)
We use 0 for getting PSTN-dialing.
We have internal DNs, for example a 6789, 4 Digits. We use external phone number mask on our lines, 123456XXX
Our main number is 0123/456-xxx
When i call outside everything is displayed fine on the called target, +49 123/456789.
When i forward a call on my cellphone, with CFA target of 00111/222333444 (my cellphone example), and an internal colleague from within our office, is calling my office phone, everything is ALSO displayed fine.
Now here comes the BUT:
When someone calls from PSTN on my office-phone, i get displayed on my cellphone the +49 (0) 0xxxxxxxx, which means the caller number PLUS the added 0 from the gateway. Which is completely consequent and correct, since we add them on the gateway, when a call comes in, to be able to just answer directly on the office phone.
The rule on the H323 gateway:
voice translation-profile OUTGOING-VOIP
translate calling 1
translate called 2
voice translation-rule 1
rule 1 /^$.*$/ /0\1/ type subscriber unknown plan any unknown
rule 2 /^$.*$/ /00\1/ type national unknown plan any unknown
rule 3 /^$.*$/ /000\1/ type international unknown plan any unknown
voice translation-rule 2
rule 6 /4560$/ /6600/
rule 9 /^456$...$$/ /6\1/
voice translation-profile OUTGOING-POTS
translate calling 3
translate called 4
voice translation-rule 3
rule 1 /^00049/ /0/ type unknown national
rule 2 /^0/ // type unknown subscriber
rule 3 /^00/ /0/ type unknown national
rule 4 /^000/ /00/ type unknown international
voice translation-rule 4
rule 2 /^00049$.*$$/ /\1/ type unknown national
rule 3 /^000$.*$$/ /\1/ type unknown international
rule 4 /^00$.*$$/ /\1/ type unknown national
rule 5 /^0$.*$$/ /\1/ type unknown subscriber
dial-peer voice 10456 voip
translation-profile outgoing OUTGOING-VOIP
destination-pattern 456.T
progress_ind setup enable 3
modem passthrough nse codec g711ulaw
session target ipv4:<IP-OF-CUCM>
incoming called-number .
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax rate disable
fax protocol pass-through g711ulaw
no vad
no supplementary-service h225-notify cid-update
dial-peer voice 345000 pots
tone ringback alert-no-PI
translation-profile outgoing OUTGOING-POTS
destination-pattern 0.T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/0:15
forward-digits all
In case of forwarding the external call to an external device, like for example a cellphone, this is crap.
Its obvious regarding the debugs all is working as designed ;), because my phone just forwards the full calling number including the added 0, since i put in to forward the originating calling DN.
My question now:
Can i simply correct this behavior somehow, also for international calls which would the 00 get added by the gateway?
Many thanks in advance for some input,
Andreas

Hi Community,
i´m wondering if there is sort of a simple way to get this working properly.
Scenario:
Germany, variable dial plan, no fixed NANP, and we have ClipNoScreening ;)
We use 0 for getting PSTN-dialing.
We have internal DNs, for example a 6789, 4 Digits. We use external phone number mask on our lines, 123456XXX
Our main number is 0123/456-xxx
When i call outside everything is displayed fine on the called target, +49 123/456789.
When i forward a call on my cellphone, with CFA target of 00111/222333444 (my cellphone example), and an internal colleague from within our office, is calling my office phone, everything is ALSO displayed fine.
Now here comes the BUT:
When someone calls from PSTN on my office-phone, i get displayed on my cellphone the +49 (0) 0xxxxxxxx, which means the caller number PLUS the added 0 from the gateway. Which is completely consequent and correct, since we add them on the gateway, when a call comes in, to be able to just answer directly on the office phone.
The rule on the H323 gateway:
voice translation-profile OUTGOING-VOIP
translate calling 1
translate called 2
voice translation-rule 1
rule 1 /^$.*$/ /0\1/ type subscriber unknown plan any unknown
rule 2 /^$.*$/ /00\1/ type national unknown plan any unknown
rule 3 /^$.*$/ /000\1/ type international unknown plan any unknown
voice translation-rule 2
rule 6 /4560$/ /6600/
rule 9 /^456$...$$/ /6\1/
voice translation-profile OUTGOING-POTS
translate calling 3
translate called 4
voice translation-rule 3
rule 1 /^00049/ /0/ type unknown national
rule 2 /^0/ // type unknown subscriber
rule 3 /^00/ /0/ type unknown national
rule 4 /^000/ /00/ type unknown international
voice translation-rule 4
rule 2 /^00049$.*$$/ /\1/ type unknown national
rule 3 /^000$.*$$/ /\1/ type unknown international
rule 4 /^00$.*$$/ /\1/ type unknown national
rule 5 /^0$.*$$/ /\1/ type unknown subscriber
dial-peer voice 10456 voip
translation-profile outgoing OUTGOING-VOIP
destination-pattern 456.T
progress_ind setup enable 3
modem passthrough nse codec g711ulaw
session target ipv4:<IP-OF-CUCM>
incoming called-number .
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax rate disable
fax protocol pass-through g711ulaw
no vad
no supplementary-service h225-notify cid-update
dial-peer voice 345000 pots
tone ringback alert-no-PI
translation-profile outgoing OUTGOING-POTS
destination-pattern 0.T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/0:15
forward-digits all
In case of forwarding the external call to an external device, like for example a cellphone, this is crap.
Its obvious regarding the debugs all is working as designed ;), because my phone just forwards the full calling number including the added 0, since i put in to forward the originating calling DN.
My question now:
Can i simply correct this behavior somehow, also for international calls which would the 00 get added by the gateway?
Many thanks in advance for some input,
Andreas

HT4994 I have an Iphone 4 and can not forward calls. every time I put the number in the call forward field it jumps back a page and the button moves to off. can anyone assist.

I can not forward calls. I go to call forwarding, turn it on, copy and paste my phone number into the correct field (including the international dialling code).
I then tap the Call Forwarding button, which takes me back one stage and the Call Forwarding button goes to off after a couple of seconds. Anyone who can help?

Yes, a blocked call will go immediately to your voice mail.
If you would rather not have that behavior, ask your cellular carrier if you can block that person through the carrier.

Call forwarding number hacked

I am an expat living and working overseas and have a Skype call forwarding # that routes calls to my international number. Over the last 2 weeks I have received 3 phone calls from angry people across the country (Georgia, California, and Utah) claiming they've received a phone call from my Skype # claiming it's from the IRS and they owe money and to contact them immediately.
I, of course, did not make these calls and am concerned my Skype # has been "hacked" which could lead to worse identity theft. I tried to contact Skype / MSFT but the chat help discussion was a disaster and the person obviously 1] wasn't a native English speaker, and 2] did not come close to understanding my issue nor solve it.
I have no idea where to turn to look in to this issue, or how to prevent it from happening again. Would welcome any suggestions!

Deactivate the call forwarding
     *73
     press 'send' and wait for tone/message
     press 'end'
Then reactivate call forwarding
     *72
     enter # where calls are to be forwarded
     press 'send' and wait for tone/message
     press 'end'

Is there any other way to achieve per user call forward restriction other than to create multiple voice policies?

Hello,
We mentioned the environment details below:
Environment
In our PBX environment, currently a user can forward calls to any local (within a region) internal extension. But for external PSTN call forwarding, a user needs to send a request and be approved by their manager. And the forwarding restriction
is applied such that user is only allowed to forward to that particular PSTN number - to prevent toll fraud.
Moving forward to Lync, using voice policy's call forwarding and simultaneous ring PSTN usages, I can set it to allow forward and simultaneous ring to custom PSTN usage and a custom route that will only send calls to these pre-approved
external numbers.
Outcome
But in such a scenario,
sSince all the custom external allowed numbers will have to be put into a single Route match table, User A will be able to successfully
set up call forward to User B's number. (if they come to know about it somehow, that is)
rü
Route matching list will be very long due to the number of users per hubsite that has call forwarding enabled.
Questions
Is there any other way to achieve per user call forward restriction other than to create multiple voice policies ? MSPL may be ?
2. Is there a limit in the number of entries you can have on the Route pattern matching regex expression ?
Please advise. MANY THANKS.

1) I think multiple policies may be your best bet, though it's not a fun one to manage, I agree. MSPL could do it, but it would be more complex to maintain in the end. Even gateways have limitations on routes.
2) I'm not aware of a limit, though I'm not saying there's isn't one. But if you hit it, you could move to a second usage/route combo.
I'd suggest building out some PowerShell usage/route creation/organization script for this so it's not something that would need to be maintained within the GUI.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

International call forwarding

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