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

Similar Messages

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 external number which has auto attendant

Hi
I am a voice administrator in my company
I want to forward all of my calls to my Other location's number.
Other location has only POTS line and to reach my extension user needs to go through Auto attendant menu.
Is it possible to enter a digit pattern in call forward destination in CUCM so that it can take care of Auto attendant menu of my Other location and land on my number?
We have CUCM 8 running.
Please help!!
Ashwin

Hello Ashwin,
"Other location has only POTS line and to reach my extension user needs to go through Auto attendant menu."
What type of phone system and voice mail is providing the auto attendant?
How are the POTS lines(analog only correct?) terminated into the phone system?

Automatic call forwarding (time based) to mobile number in CUCM 9.1.1

Hi
I need to configure time based automatic call forwarding to mobile number. Can we do this from CUCM 9.1.1 ?
Example : Ex 2000 need to forward to 01111188888 in every friday after 9.00pm
Regds
$

There is one way you can still achieve it by using the Mobile connect (SNR).
In the Remote Destination you have the option to ring your mobile based on your requirement.
Please see the attached screenshot for the same.
So user need to enable the Mobile connect on friday all calls will go to the Mobile phone # (Remote destination).
Also delay before ringing should be 0 ms.
Please refer to the Mobile connect configuration guide for the same.
Kind Regards,
Raaj.

Call Forwarding with Skype Number

I have a skype number and a couple questions. I have searched the forums and did not see this question answered.
I have an unlocked iphone and travel througout a different country every month or so. Each new country means a new sim card and a new number.
My question is: Will purchasing the skype world subscription cover the costs of each country (as long as the country is covered in the plan)? It states that the subscription only covers mobiles in some countries and landlines in others. So if I purchase the unlimited world (and assuming I am in one of the countries covered) will this be all I need to receive unlimited forwarded calls to my mobile?
I hope this isnt too confusing. Thanks!

Hi, Salsy20 and Joeriba, and welcome to the Community,
Please check your call forwarding settings; Skype does not determine when to activate call forwarding based upon the number of rings; it is based upon an increment of time. For example, in the screen shot, below, the default increment is 10 seconds (circled in orange pen). You would need to determine and then experiment with how many rings = [x] seconds, or set to 0. Remember, if you make any changes to any settings, click Save!
Here is a link to the library of FAQ articles related to call forwarding:
https://support.skype.com/en/category/CALL_FORWARDING/
Regards,
Elaine
Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
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Call forwarding to wrong number

I have call forwarding set to forward calls to my cell phone. In the past few days three or four calls have been transferred instead to my home phone, which is a big nuisance, because I'm in the US at the moment and, because of the time difference, they're often waking my husband in the middle of the night in Italy.
These callers don't even know my home phone number. They always call my US Skype number. A few months ago, I had temporarily set forwarding to my home phone, but changed it back to the cell phone. I checked the settings on my account, and it's definitely set to the cell phone number. When it first happened, I first checked the account, then I asked someone to try calling my Skype number. That time it transferred correctly to my cell phone. I didn't understand how it could have happened, but thought it woul never happen again. However it's happened again twice today. Now some people don't want to risk waking my husband, or attempting to converse with him in English, so they're just not calling me at all.
If anyone could solve this mystery, I'd greatly appreciate it, and so would my husband.

Lots of SKYPE tel. numbers were wrongfully forwarded to my company's land line 626-576-9900 in the past 30 days (4/20/2015 - 5/20/2015) or so. Please test (call your SKYPE tel# from other lines to see where it ends) and call SKYPE to correct it as soon as possilbe if you own any of these numbers! I suggest everyone who owns a SKYPE number do the test. You might have missed a lots of calls you supose to receive. These numbers are as following.
281-973-2121
281-973-6874
281-973-6695
304-932-0943
301-579-5097
317-395-7485
301-579-5316
281-973-6045
281-973-6090
203-903-0092
281-973-6833
281-713-6436
317-413-9306
203-208-6591
I've chat on-line with SKYPE customer/tech. support 2 times, got 2 case#s, 1288597626 (5/5/2015) and 1288963288 (5/8/2015), however I tested 2 of the above numbers just now (10:15 AM, 5/20/2015), both are still forwarded to my company's number 626-576-9900. One of SKYPE custmer service reps. did tell me that they (SKYPE) can do nothing about it because I am not the owner of any of these numbers. Please spread the above message to any one you know who has a SKYPE number.

Call Forwarding with Online Number

Can I get skype online number and forward call to my phone in India?
As if someone calls skype online number his call can be answered at my India Mobile or Phone.
Is this possible?

I contacted Skype some months ago about this issue they said if you have a Subscription if it's covered under your subscription (for example calls to the US & Canada unlimited) and make a call on skype it will be covered.
I don't have the email or lost it somewhere as it's months old but basically that's what it says.

8330 Call Forwarding...How do I set the number of rings before calls are forwarded?

8330 Call Forwarding...How do I set the number of rings before calls are forwarded? Right now it instantly forwards my calls but I don't want that. I want to change it so that let's say that if after 5 rings I don't answer the call is forwarded. I've looked everywhere for an answer but have had no luck. Any help would be appreciated. I am with Virgin Mobile if this makes a difference.
Thank you
Message Edited by 8WDDdotcom on 08-01-2009 02:39 PM
Message Edited by 8WDDdotcom on 08-01-2009 02:44 PM

Well, call forwarding is a feature provided by your carrier -- it is not a function of the device. However it works for your carrier is how it works. Actually, from what I understand about call forwarding, it normally is an all-or-nothing circumstance...calls forward to the number you've asked them to be forwarded to, keeping you in a DND status. Regardless, though, if your carrier tells you how it works on their system, there's nothing that the device can do to override that. Unless there is some kind of 3rd party app that would do it, bypassing (somehow) the carrier network CF feature...but I've never heard of such a thing.
Occam's Razor nearly always applies when troubleshooting technology issues!
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CUPC 8.6 call forward to voicemail

Hi!
I am using Cisco Personal Communicator (CUPC) 8.6 and also CUCM 8.6. I have CUPC in Deskphone mode, connected to a 6945 IP Phone. I also have Unity Connection where my voicemail box is hosted. When I want to setup call forward to voicemail button in cupc option, it is not working. CUPC will not handle the options I setup seconds before. If I manually put in a call forward to extension number of voice mail pilot call forwarding is working. also call forwarding to my mobile is working.
I checked End User settings, IP Phone is associated to my user, also CTI controll is enabled on device and line settings. user privileges are correct. I tried it on jabber client where it works fine. I also restarted CTI and Callmanager Services on the Servers.
Does anyone has an Idea if this is a general bug in CUPC or does anyone can tell me what the problem might be?
Thanks!
René

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.

Forwarding cell phone number to skype online numbe...

Is it possible to forward cell phone number to skype online number? If so, what do I need to do to set it up? Also, would my carrier (T-Mobile) charge me minutes for forwarded calls? Thanks.
Solved!
Go to Solution.

spazby wrote:
Is it possible to forward cell phone number to skype online number? If so, what do I need to do to set it up? Also, would my carrier (T-Mobile) charge me minutes for forwarded calls? Thanks.
Maybe. Your phone should have settings to set forwarding for all calls to your T-Mobile number. If you enable that, your T-Mobile phone won't ring anymore, but you could still use your phone to make calls and for sending/receiving SMS or MMS messages.
If you are asking about forwarding calls when the mobile phone is on but you don't answer the phone, instead of those calls going to a T-Mobile voicemail system, you *may* be able to change that. T-Mobile doesn't let you do that on a prepaid phone, and that appears to be the same if you use some other prepaid services that ride on T-Mobile's network like Straight Talk. It may be possible on a postpaid plan (one where you get a monthly invoice to pay).
As for what T-Mobile would charge, you'd have to check on your plan or contact T-Mobile. Some plans have unlimited calling that includes call-forwarding, others only provide for a certain number of minutes for call-forwarded calls a month, and then there are some "plans" that make you pay for anything that is call-forwarded from your T-Mobile number. The T-Mobile web site may be able to help answer that, or a call to their customer service number will need to be made.
Good luck!
Patrick
Location/Ubicacion: Arizona USA
Time Zone/Hora Local: UTC/GMT -7
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I am not a Skype employee. No soy un empleado de Skype.

Call forwarding offnet doesn't insert 1 for non-local calls

Hello,
When users at a remote site (long distance) call over the WAN to a central site phone that is call forwarded to a number local to the central site. The remote site phone is sent through its voice gateway and the calls fails because the CallManager or gateway doesn't insert a 1.
Right now a special calling search space is configured on the central site phones that send call forwarded calls to the central site gateways.
What other ways could we accomplish the same thing. Voice translation patterns on the gateways, etc?

There are multiple places you can do digit manipulation, like you mentioned you can do it on voice translation patterns applied on dial peers on the gateway, if you have CUCM, you can also do digit manipulation on route patterns or also on route lists.
Not sure if this is what you were looking for.

Call forwarding of 2 Online Numbers

I bought 2 Online numbers. I have already set up call forwarding for 1 number to my cell (cell #1) and would like to do the same for the other number to my other cell. But it seems that I can only forward to one number (cell #1), how can I forward the second Online number to cell#2?
Thanks

CarlosD wrote:
I have two online numbers in my account. I would like the calls on one be forwarded to one cell phone and the other to a different cell phone. Is this possible? It does not seems to? Do I have to open a separate account for this?
Currently, this would require a 3rd party application using the Skype public API or SkypeKit. This function is currently not supported by the Skype clients.
You might want to do a search on:
Skype call transfer
About Me You can also use a IP Camera as your camera for Skype video Example Instructions

Can I forward my UK mobile calls to a Skype Number...

Hi
I am moving to France for 6 mths and want to forward all UK mobile calls to a Skype number. I want to know will my clients realise I am abroad? will there be a foreign ring tone and will they get charged a premium rate?
Ideally I want them to call my mobile and seamlessly be transfered to my skype number and not be charged any extra.
Do I need to buy a UK or French Skype number?
Any help of comments would be most useful.

Hi
I am moving to France for 6 mths and want to forward all UK mobile calls to a Skype number. I want to know will my clients realise I am abroad? will there be a foreign ring tone and will they get charged a premium rate?
Ideally I want them to call my mobile and seamlessly be transfered to my skype number and not be charged any extra.
Do I need to buy a UK or French Skype number?
Any help of comments would be most useful.

How to delete the number saved in CALL FORWARDING list?

how to delete the number saved in CALL FORWARDING list? seems like the number cannot be deleted in CALL FORWARDING list, i tried several times...
he thing is, here are three numbers in the call forwarding list, one of them were set up by mistake, and none of them can be deleted.......it brothers me all the time. i already turn it off, but the number list is still there...

Hello dennis130915 and welcome to the BlackBerry Support Community Forums.
The image you have uploaded is unable to be displayed.
Call Forwarding is a carrier controlled feature. Most likely, the number you see could be for the Voice Mail server.
It's advised you call your mobile service provider to further assist with this feature.
Thanks!
-HMthePirate
Come follow your BlackBerry Technical Team on twitter! @BlackBerryHelp
Be sure to click Kudos! for those who have helped you.Click Solution? for posts that have solved your issue(s)!

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

Call forward to external number(mobile)

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