CUCM 7 Blackberry call forwarding

Hello,
I have CUCM 7. I wish to call forward to a blackberry device. When this is attempted the ISDN origin and destination cause codes are 4. This states a special information tone is to be generated. Does anyone know a solution for this ?
Many thanks
Stephen
Cause No. 4 - send special information tone [Q.850]
This cause indicates that the called party cannot be reached for reasons that are of a long
term nature and that the special information tone should be returned to the calling party.

Hi, thanks for your response.
We need a SIP route pattern because the 3rd party server at the end of the SIP trunk uses SIP URI dialling when it talks back to the CallManager.
The requirements are that the call must first go over the SIP trunk to the 3rd party SIP server. If there is no response, then an alternate destination must be tried by the CallManager. This can be another SIP trunk to a second SIP server or it can be an DN internal to the CallManager.
Route groups can't be used because if a SIP trunk in configured and then assigned to a route group, the same SIP trunk can not then be referenced in a SIP route pattern.
In other words for communication to work with the 3rd party SIP server,
Step 1: The SIP trunk must be referenced directly by a route pattern.
Step 2: The SIP trunk must be referenced by a SIP route pattern.
We won't need route groups if the second destination is internal to the CallManager.
However, how do we hunt with 1 - Route pattern to SIP Trunk and then 2 - Internal DN?

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

CUCM 8.6 - Call Forward All Doesn't Work With DN In Hunt Group

Hello,
I'm trying to make a hunt group that's sole purpose is to dial out to our Help Desk member's cell phone (on-call after hours). The idea is that we want the hunt group to broadcast the call to the on-call engineers. Whoever picks up first takes the call.
At least that's the idea now. They may want to change that to - ring primary on-call cell phone, if no answer ring secondary on-call engineer.
The target cell phones will change every week and I wanted to give the supervisor a means of changing the numbers through the ccmuser page.
I created two DNs and placed them in a hunt group, each with call forward all to a cell phone number.
It seems that the hunt group ignores the call forward all setting and that is by design (https://supportforums.cisco.com/document/9126/cisco-callmanager-call-forward-all-cfa-does-not-work-if-line-hunt-group).
Does anyone else have any suggestions on how to accomplish this?

That's correct, what you're seeing is WAD, this topic has been discussed plenty of times before, the only option you have is SNR

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.

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.

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)!

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 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?

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

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

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. :-)

CCME Call Forward from one Hunt Group to another Hunt Group Failure

Hi I have a couple of hungroups in Cisco Call Manager Express. I am trying to configure a Call Forward no answer from one hunt group to another. Does anybody know if this is possible? If so, is there a config available? Here is my config, but it is not working.
Thanks,
Derek
voice hunt-group 20 parallel
final 2290
list 2201,2251,2252
timeout 30
pilot 2209

Hi Derek,
This is supported with the restriction shown below;
final number
Router(config-voice-hunt-group)# final 8888
Defines the last extension in a voice hunt group.
If a final number in one hunt group is configured as a pilot number of another hunt group, the pilot number of the first hunt group cannot be configured as a final number in any other hunt group.
From;
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmecover.html#72373
Cheers!
Rob

Problem with call forwarding and "call divert"

I have been receiving calls lately and a dialogue box pops up and says "fowarded call" with a button that says "dismiss" - I'm not sure why or how to disable this. in my settings I have call forwarding turned off. Also when I make calls I sometimes see messages saying something about call forwarding active or active diverts. Again I have call forwarding turned off and I see no settings for divert. How can I fix this?
Thanks.

i am having a similar problem. I have a Pearl 8100 with T-Mobile. I've had it for almost two years (since Jan. '07). About four weeks ago, this problem started. I have a landline office phone through AT&T, and when I leave the office, I forward this land line to my Pearl. When someone calls my office after hours, on nights and weekends, the problem is that instead of the person's name and phone number showing up on my caller ID, my Pearl shows that I am calling myself from my own cell phone! I don't get "Private Number" like you get, the caller ID shows that I am calling myself!
I've spent six or seven hours with T-Mobile tech support, with five different tech people over a period of several days and several phone calls, and they can't figure it out; their final and "best" answer is that it's a blackberry problem, that the issue has been addressed and identified as a "known issue" and a "trouble ticket" has been created. I contacted RIM and they directed me to their archived troubleshooting and to this forum, so I still don't have an answer. And T-M had no answer when I told them that this feature worked just fine from Jan. '07 through Aug. '08. T-Mobile offered to put me into a different device, the T-Mobile wing, which by fair assessment, looks like a piece of junk, and it's window's mobile, which is even worse. I rejected this option.
I'm hoping someone can solve this. One friend of mine thought that it's a network issue, any issue with caller ID would be with the network, not the phone or its software...this makes some sense to me...and one of the techs at TM thought that this could be the issue as well. As T-M upgrades its networks to 3G, these bugs are popping out. who knows, I'm sticking with this Pearl and 8100 for a little while longer, b/c I am waiting to see if an answer shows up somewhere on one of the forums, but I hate not knowing who's calling before I answer the phone!

CUCM 7 Blackberry call forwarding

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