Setup call forwarding

I can't setup call forwarding to my mphone phone. I'm getting the error that I need to use the international dialing standard. I used (with and without spaces): +1236xxxxxxx 1236xxxxxxx xxxxxxx Where xxxxxxx is my actual phone number. Is it because the 236 area code is new, or how does this international dialing format supposed to look like?

Hi Xibir,
Oh no! Call forwarding is a great feature, and I am sad to hear that your having issues activating this feature. Let's get this fixed. To set an Immediate Call Forwarding from your work phone,
• Dial *72 then the 10-digit phone number of your personal phone.
• press Send and wait for confirmation beeps/message, and
• press End.
If this does not work, please accept my follow & send us a Private message with your work mobile number. We can set this forwarding up for you.
Thanks,
PamelaF_VZW
Tweet us @vzwsuport

Similar Messages

Unable To Setup Call Forwarding - Receive Message ...

Unable To Setup Call Forwarding - Receive Message "oops the server is not responding"
have tried 3 different web browsers, chrome, ie and firefox and also tried our 3 outgoing internet lines all with different suppliers but when I go to add a call forwarding number it does nothing and I get the "oops" message.
What's going on here, I need this urgently for business!

for the call forwarding you need balance in your account and you can forward from the web browser or sky msn option also if need assistance you can connect with us

How to setup call forwarding

I am trying to setup call forwarding but unable to do as
I know it is not that much difficult but prob I am facing is
That when I try to put the number it does not give any option
To confirm which mean the setup did not complete

Hi Scooter,
There are a few ways this can be done Have a look in CUCM under
Call Routing> Directory number and see if it's set up there. You can have a DID/DN
with Call Forward set up without an actual phone by doing it via this config.
You can also check Device > CTI Route Point as the same "virtual" config can be set there as well.
Cheers!
Rob
"Show a little faith, there's magic in the night" - Springsteen

What package do I need to setup call forwarding to...

Hi I want to forward calls to my mobile when I cant answer them.
I only want to call landlines in the UK and forward calls onto my mobile.
What package would suit my needs?
Thanks

If you have UK mobile number then you would need a UK subscription that covers landlines and mobiles.
http://www.skype.com/intl/en/prices/pay-monthly/#GB
There are 2 tabs, make sure you pick the subscription from "Landlines and Mobiles" tab.
Important is that the number where you forward your calls is covered by your subscription.
Note: Unlimited Europe and Unlimited World don't cover UK mobile numbers.
Andre
If answer was helpful please mark it with Kudos and if issue is resolved mark it with solution. This will help other users find this answer more easily. Thanks in advance!

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.

Call forwarding setup having a subscription

Hi, I have a subscription and through skype desktop ?'m not able to setup the call forwarding. An error pop-up inform me that I have not enough credit!!! Yes, I have no credit, I pay for a subscription. So, I can setup the call forwarding via web. Thanks

You must have a subscription that does not include mobile phones. Check to see if that is the case if so you will need skype credit to forward the phone.
If you found a post useful, please give Kudos. If it helped to fix your issue, mark it as a solution to help others.
Thank You!

Problems with call forwarding setup

I just got an iPhone and it's great, except for one annoying problem with the call forwarding behavior.
With my old RAZR phone (also on AT&T) I had two choices when turning on forwarding, "forward immediately" or "forward when unavailable". I used the second choice and had forwarding turned on all the time. Calls would go to the cell phone first but then forward to my home number if I was on another call or didn't answer after a couple of rings. That behavior is important because I want all voice messages in the same place, on my answering machine at home, regardless of whether my cell or home number was called (so I want to deliberately bypass AT&T's voicemail system).
However the iPhone seems to only have the "immediately" behavior, which means I don't have a chance to answer a call on the iPhone before it forwards, and the iPhone doesn't even log it as a missed call. The forwarding setup is stored in AT&T's system so it seems like the capability should be there regardless of the type of phone, and this is just an arbitrary user-interface restriction with the iPhone. Anyone know how to work around this and get the "forward when unavailable" behavior?

Jason -
Thanks, this did the trick. The instructions under the section for "former AT&T Wireless customers" worked perfectly (perhaps because I am actually a former AT&T Wireless customer; I've had the number for almost 6 years, since before the merger with Cingular). Anyway, I dial 61xxxxxxxxxx# (xxx = the number to forward to), make the call, and a confirmation message appears on the screen; now the forwarding is working just the way I like it.
Thanks again!
Bob

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

CME, Call forward to CUE from CCM IP phone

I want to call forward the call from CCM IP phone to CME ephone's voicemail which setup in CUE. works okay between CME ephones. configured voice service as follows but no luck. what did I missing to implement?
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
-CCM4.1.3, configured H225 trunk. leave uncheck the MTP on the trunk device
-gatekeeper to connect between CCM and CME
-CME3.3, h323 to gatekeeper and sip to CUE
-CUE2.1
Thanks in advance,

It works by restart the CME router and have a question the sip-ua output. I have two media streams but the 2nd shows "STREAM_IDLE". I think this is for g729 connected to CCM via h323 gk. Can I get an explanation why?
CME#sh sip-ua calls
SIP UAC CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 4083132006
Called Number : 4211
Bit Flags : 0x101A0030 0x100000 0x500
CC Call ID : 95
Source IP Address (Sig ): 10.253.66.254
Destn SIP Req Addr:Port : 10.253.66.2:5060
Destn SIP Resp Addr:Port: 10.253.66.2:5060
Destination Name :
Number of Media Streams : 2
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 95
Stream Type : voice-only (0)
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 10.253.66.254:16998
Media Dest IP Addr:Port : 10.253.66.2:16904
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Media Stream 2
State of the stream : STREAM_IDLE
Stream Call ID : -1
Stream Type : voice+dtmf (1)
Negotiated Codec : No Codec (0 bytes)
Codec Payload Type : 255 (None)
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 10.253.66.254:17120
Media Dest IP Addr:Port : 0.0.0.0:0
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Number of SIP User Agent Server(UAS) calls: 0
CME#sh sccp connections
sess_id conn_id stype mode codec ripaddr rport sport
1 2 xcode sendrecv g711u 10.253.66.254 2000 16518
1 1 xcode sendrecv g729 10.253.66.254 2000 17620
Total number of active session(s) 1, and connection(s) 2

IP Phone call forward to Attendant Console - reach Attendant Console Voicemail

Hello,
I have an old Attendant Console based on CUCM7.1(5) with CUC 9.
Most of my user have voice mail box.
My issue is following: Users forward there phone to AC pilot point during lunch time or meetings.
When a call is then transfered to Attendant Console, if it is not answered, it is redirected to Unity connection and reach User Voice Mail box.
This is the good way of working for all users (to reach first redirect number mailbox), but my customer wants in the particular case of calls forwarded to standard, to reach standard voice mail box.
How can we achieve this?
If I setup a Translation Pattern on CUCM to translate a dummy number into attendant console Pilot Point number, and forward user to this dummy number, does Translation Pattern replace the first redirect number?
Thanks for your help.
Thibaut

It is likely that they are transferring a call to the operator on a line they are not logged into instead of the queue.
All calls should be routed through the queue in order to receive full CTI control of that call, I would suggest you create a speed dial with the DN of the CTI Route Point for the operators queue and then transfer the call to that instead of the operator directly, this way the call sits in the queue with any other callers.

Call forward to unity connection call handler

have the following setup:
cucm 8.6
cuc 8.6
a cti route point (DN 1000 )with forward all to unity connection call handler.
phone users who choose to have some quiet time can forward all their calls to the cti rp.
1. i need to allow only specfic DNs the ability to forward to the cti rp.
2. at times that the allowed DNs didnt choose to forward to cti rp then normal behaviour should occur - busy or unanswerd calls should go to DNs personal voice mail.
thanks

You can assign a specific partition to cti-rp and assign it to a CSS and this CSS should be assigned to those users (at call forward settings in line) only to whom you are trying to give this facility and for other users you can
or you can create a TP for it with specific partition and set the called number as per voice mail.
Suresh

Call forward to another users voicemail

Here is the scenario, i cannot find a way to accomplish this.
i am using CUCM 8.5 and Unity Connection 8.5
here is the requirement
1. User1 call forwards her phone to User2
2. A call comes in to User1, the call is forwarded to User2
3. If User2 is unavailable, the call is redirected to User2 voicemail.
The default behaviour is that if User2 does not answer, the call is redirected to the original called number (User1)
I have created forwarded routing rules in Unity connection and i can get the call to end up at User2 voicemailbox however, the user requires that this happens only when the call is forwarded manually. For example, if i call User1 and let it go to Voicemail, it will still go to User2
They want that it goes only to the User2 voicemail only in the specific circumstance that the phone is in call forward mode, not by letting it go to voicemail
Does it make sense?
I think my users are asking too much

Hi Steven,
I'll just add a note to the great tips from Hailey & Roger (+5 each!)
I thought this was an interesting question, so I tried a number of ways
to see if this could be done
The problem, as you've discovered, is that the original Forwarded CLID
is so "sticky" as to render most of the standard routing methods moot.
In my tests I was trying to route calls that come into 7001 and are CFWDALL
to 5126 to route to the 5126 mailbox without changing the Originally dialed
to last redirecting setting (as nicely noted by Hailey).
I had to put in a mid-point number that was set to CFWDALL (via PSTN)
back to the second number.
So I setup a CTI-RP on a phantom (non-DID) DN of 4241 and set the CFWDALL
to 5126. I could then CFWD from 7001 to 4241 which then routed to 5126 with
the CLID stripped off and would then give us the desired results. No other method
that I could come up with would work.
Cheers!
Rob
"Why not help one another on the way" - 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.

Support Line - Call Forward All through Hunt Pilot to cell phone

I know I may be beating a dead horse, but I've found so much help in these forums that I figured I'd give it a shot.
We have a Cisco Unified Communications Manager Business Edition and we are running into a configuration issue. What we would like to do is to create an emergency support line for our IT staff. The way we would like this to function is that a caller dials a number, CM accepts the call and dials out to our IT staff's cell phone numbers in a round-robin fashion. To avoid the caller getting dumped into the IT staff's voice mail on their cells, we would like the staff member to have to dial a number to accept the call. If there is no answer, the call rolls to the next cell number. If no one is available, the caller should be directed to Unity Connection to leave a message. Unity then will send out text and email messages to the support staff.
I know that we can use Unity to perform an assisted transfer, which will require the user to press "1" to accept the call, and we are able to get Unity to send out the notifications (text and email) when a voice message is left. The issue is with Call Manager making the outbound calls to the cell phones.
What we have attempted is to setup DNs that call forward all to the users cell numbers. These DNs have been added to a Line Group, which has a Hunt Pilot attached to it. Any time this pilot is called, we get a reorder. Using the DNA, we see that "no device is associated with the DN", which is under the DN for the first users cell forward. If we add that DN as a second line to that users IP Phone (7940), then the call into the Hunt Pilot rings that line on the IP phone, not the CFA to the cell phone.
After weeks of digging around, it seems as though CFA in a Hunt is just not possible. My boss wants official word from Cisco about this, but TAC doesn't seem to want to help due to service contract issues (which blows my mind, as we have opened several cases in the past two months for configuration related issues). Our Business Edition came with Contact Center Xpress, although we do not have the resources available to install it. If CCX will carry out this task, that might be enough to push management into getting another server to support this, but without being able to play around with it, I don't know.
If anyone has any suggestions on how to make this work, I would GREATLY appreciate the help!
Thanks in advance,
-Geoff

Hi Geoff,
Always interesting isn't it :)
Call Forward settings on individual Hunt member phones are ignored when presented a call via the Hunt feature. Here is a clip;
The concept of hunting differs from that of call forwarding. Hunting allows Cisco CallManager to extend a call to one or more lists of numbers, where each such list can specify a hunting order that is chosen from a fixed set of algorithms. When a call extends to a hunt party from these lists and the party fails to answer or is busy, hunting resumes with the next hunt party. (The next hunt party varies depending on the current hunt algorithm.) Hunting thus ignores the Call Forward No Answer (CFNA) or Call Forward Busy (CFB) settings for the attempted party. This also applies to CFWD ALL settings.
From this doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803edabe.html#wp107892
So, any Forwarding settings have to be done on the Hunt Pilot itself. These Destination settings (on the Hunt Pilot) need to be configured to go to the Unity Voicemail Pilot # or perhaps this is where you Forward out to the Cell #;
Hunt Forward Settings
Forward Hunt No Answer - When the call that is distributed through the hunt list is not answered in a specific time, this field specifies how to forward the call.
Destination This setting indicates the directory number to which calls are forwarded. (This can be the Directory Number of the Unity VM Pilot)
Forward Hunt Busy - When the call that is distributed through the hunt list is busy in a specific time, this field specifies how to forward the call.
Destination This setting indicates the directory number to which calls are forwarded. (This can be the Directory Number of the Unity VM Pilot)
Maximum Hunt Timer - Enter a value (in seconds) that specifies the maximum time for hunting.(Used in conjunction with Forward Hunt Busy)
Maybe you could leverage these Unity Connection configs to achieve your desired results. These will ensure that the Message is not left unattended;
Cascading Message Notification
Cascading message notification allows you to send notifications to a widening circle of recipients. Cisco Unity Connection continues to send notifications according to the devices you selected until the message has been saved or deleted by a recipient.
For example, to create a cascade of message notifications for your Technical Support department,
Chaining Message Notification
Message notification can be set to "chain" to a series of notification devices if an attempt to send notification to the first selected device fails. The definition of failure to a notification device is based on the options you select for retrying a device that is not answered or is busy.
http://www.cisco.com/en/US/docs/voice_ip_comm/connection/2x/user_mac/guide/2xcucmac040.html#wp1132107
http://www.cisco.com/en/US/docs/voice_ip_comm/connection/2x/user_mac_cmbe/guide/6xcucmbemac040.html#wp1132107
Hope this helps!
Rob

Setup call forwarding

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