Cisco Forward and Call return to Voicemail on CUCM 8.5 question

Hi,
I have a user A that requires the call forward when busy/no answer to another extension B. If extension B is busy/no answer, A wants call to be returned to his voicemail.
Unfortunately User B has also a call forward to another extension c, so call forward from A are forwarded to C when B is busy.
Is there any means to have the calls from A return to his voicemail when B is busy/no answer.
I would be grateful if someone can help or is it a system restrictions.

Hi
That's how it works by default.
Unity looks at the 'first redirecting' number, and uses that to allocate forwarded calls to a VM box.
So if User A forwards to User B, the first forwarding number is User A. It goes in User A's box.
That applies regardless of how many times it's forwarded, unless something happens to 'lose' the forwarding number info. That wouldn't usually happen on-system, more likely if a PRI or other trunk is traversed.
Aaron

Similar Messages

What's the meaning of Forward No Coverage at Call Forward and Call Pickup Settings

hi All,
Please, I have a CUCM and I need to understand the meaning of the follow configuratio at the line:
Forward No Coverage Internal
Forward No Coverage External
Forward on CTI Failure
Could you explain to me by a easy way?????
How can I do a test?
Thanks a lot

Actually the no coverage is not only for DNs in line groups. The way no coverage works is that you set a CF setting on any phone (doesn't need to be in a line group), and you send them to a hunt pilot. ie CFB to a hunt pilot
In the hunt pilot for the Forward Hunt No Answer/Forward Hunt Busy you choose:
Use Personal Preferences—Use this check box to enable the Call Forward No Coverage (CFNC) settings for the original called number that forwarded the call to this hunt pilot.
So, after it rings your phone, rings thru all the hunt pilot/hunt list/line group members, it goes back to the phone that routed the call to that hunt pilot, looks at the no coverage setting and does as instructed. Send it to VM, another hunt pilot, your mobile, etc.
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk

Cisco CME and Calls through SIP provider

Hello, friends.
There are Cisco (C2801-ADVENTERPRISEK9_IVS-M), Version 15.1 (4) M7.
Telephones connected to SCCP, registered SIP from the provider.
When I try to call to test number 4444 through sip in debug I see:
*Feb 10 01:51:25.317: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP XXXXXXXXXXX:5060;branch=z9hG4bK100D02077;rport=5060
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvf1OFL39Awnou/oMiaFQrf9jyybhFmf", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 01:51:25.325: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXX:5060;branch=z9hG4bK100D02077
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Date: Sun, 09 Feb 2014 21:51:25 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Cisco при этом зарегана у провайдера SIP
DC#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
Configuration:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice register global
max-dn 10
max-pool 10
voice register dn 1
number 150
voice register dn 2
number 151
voice translation-rule 9
rule 1 /^95/ //
voice translation-rule 1020
rule 1 /^.$/ /40232/
voice translation-profile outgoing
translate calling 1020
translate called 9
mgcp fax t38 ecm
mgcp profile default
dial-peer voice 2 voip
translation-profile outgoing outgoing
destination-pattern 95....
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 40232 password 7 XXXXXXXXXX realm sip.zadarma.com
authentication username 40232 password 7 XXXXXXXXXXXX realm sip.zadarma.com
registrar dns:sip.zadarma.com:5060 expires 3600
sip-server dns:sip.zadarma.com:5060
connection-reuse
host-registrar
DC#show sip-ua register status
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
150                              40001      12           no
40232                            -1         550          yes
SIP provider says cisco trying to call with the internal call number, and it is necessary in order that have an SIP provider:
Wrong Remote-Party-ID: "Vankuver" <sip:61@<my ip>>;party=calling;
Should be so sip:40232@<my ip>
Please help me!

Yes, I behind nat.
*Feb 10 18:11:53.425: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444"
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 314
v=0
o=- 2 2 IN IP4 192.168.11.14
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.11.14
t=0 0
m=audio 5724 RTP/AVP 107 0 8 101
a=alt:1 2 : gNONJ/Dj BaLJhmb/ 10.200.16.55 5724
a=alt:2 1 : DQ3e8qud c1qVrWui 192.168.11.14 5724
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
*Feb 10 18:11:53.477: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038E7FF
From: "" >;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392041513
Contact: outside ip cisco cme:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.481: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444"
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Feb 10 18:11:53.625: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP outside ip cisco cme:5060;branch=z9hG4bK1038E7FF;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:53.633: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038E7FF
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:53.637: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038F25FC
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392041513
Contact: :5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.981: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038F25FC;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:54.385: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.63.X:5060;rport=5060;branch=z9hG4bK1038F25FC
Record-Route:
From: "k40232" ;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 1942395501 1942395501 IN IP4 178.16.26.124
s=Asterisk PBX
c=IN IP4 178.16.26.124
t=0 0
m=audio 12164 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
*Feb 10 18:11:54.409: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.xxxx.xxxx:5060;branch=z9hG4bK10390E63
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:54.429: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444";tag=169E6F78-88E
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: :5060;transport=tcp>
Supported: replaces
Server: Cisco-SIPGateway/IOS-12.x
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 193
v=0
o=CiscoSystemsSIP-GW-UserAgent 149 3396 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 17190 RTP/AVP 8
c=IN IP4 92.63.108.115
a=rtpmap:8 PCMA/8000
a=ptime:20
*Feb 10 18:11:54.653: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 91.231.141.230:42294;branch=z9hG4bK-d8754z-95374017c126c928-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444";tag=169E6F78-88E
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 ACK
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

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!

Cisco 887 and wireless module

Hello there
I recently bought a cisco 887w and i'm having problem with wireless. The question is: is there any way to disable the very VERY annoying wlan-ap0 interface to boot from the main cisco IOS? Couln't find anything related with this over the internet :( I do really want to use dot11radio 0 from main router interface just like on my old cisco 877. Any help?

Nicolas thank you for your answer,
Yes, i did notice that c887w has a real ap inside. That's not too much flexible for me for the main reason that this is not intergrated like c877w as i thought. This has many disadvantages for me like i have to keep 2 different configurations, i have to login by typing service-module wlan-ap0 blah blah blah, two different managment IPs and so on. Dissapointed... but i'll get used to it : ) Fortunately they didn't do the same thing on fastethernet interface module..LOL! Hopefully on next IOS releases they do include an option to use dot11radio from main cisco IOS.
I'll have my time with this now.. I'll keep searching the cisco.com for a suitable config for me. If you have any tip for wireless bridging with the main BVIs on the router interface i'd apprieciate that : )
And for last something off topic... i noticed (not sure) that weighted early random detection (WRED.. QoS) is not supported. Is that true?
TX!!!

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

HT4993 How do I UNfoward calls. I accidently forwarded the calls to a random person's phone and they won't respond or unforward them !!! URGENT

Hi,
I called someone tonight and on accident forwarded my calls to them, how do I UNfoward my calls back to my Iphone? This is URGENT, the person will not return and calls !!!!

If you were able to turn it on, then you should be able to turn it off. Go to Settings, Phone, Call Forwarding, and turn it off.

Cisco Call Manager Express voicemail day & Night option

Hi,
anyone know how to setup a automatic night option for voicemail.
basically, at 5.30pm all incoming calls go to voicemail (a general delivery voicemail that is) without using any auto attendant?

Hi
You could set up a night service so that calls would CallFWD to voicemail after a certain time?
Details below:
night-service day Sun 12:00 07:59
night-service day Mon 17:00 07:59
night-service day Tue 17:00 07:59
night-service day Wed 17:00 07:59
night-service day Thu 17:00 07:59
night-service day Fri 17:00 12:00
night-service day Sat 12:00 12:00
ephone-dn 1 dual-line
number XXXX
pickup-group X
label Joe Bloggs
description Joe's phone
name Joe Bloggs
call-forward busy 1111
call-forward night-service 1111
call-forward noan 1111 timeout 17
huntstop channel
night-service bell
More info here..
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_n1ht.html#wp1015444
Please rate useful posts...

Hey how are you guys listen i have an iphone 4s the one i use with the H20 CARRIEr and i trying to enable the option call forwarding and when i type tho number i go back and i notice to the call forwarding it turning off as soon i back to the main menu ?

hey how are you guys listen i have an iphone 4s the one i use with the H20 CARRIEr and i trying to enable the option call forwarding and when i type tho number i go back and i notice to the call forwarding it turning off as soon i back to the main menu ?

There are a lot of posts in the forums today with people having problems with iMessage. There was also a published outage yesterday, so it's possible there are still some issues that may be impacting you both.
I would just wait it out - I'm sure it will be sorted out soon.

Call forwarding and 8.3

Since updating to 8.3, I am not longer able to activate call forwarding. I use at&t and have used this feature before, but not since upgrading to 8.3. Won't turn on in settings or dial pad. When I try to set it, forst there is no ok button. All you can do is back out of the forwarding screen. When I do, call forwarding dims out for about 2 seconds and then just shuts back off. Get an error trying to forward through the dial pad. I've called at&t and they reset my account and said they are having no issues with forwarding. Had a friend test forwarding on his iphone and at&t and his forwarding works fine, since he's still running 7.1.2.
Anyone else having this issue and/or did you get it sorted out? I need to be able to forward my calls. Help.

Hello, jtmo3.
Thank you for visiting Apple Support Communities.
I understand that you are unable to activate call forwarding on your iPhone. Depending on what carrier you have, this could be related to a provisioning issue. I would recommend checking with your carrier to make sure this feature is provisioned or not blocked.
iPhone: Understanding phone features
-Jason H.

Cisco Unified WFO - Call Recording and Quality Management with Extension Mobility agents

Hi All,
We're considering Cisco Unified WFO - Call Recording and Quality Management for a customer running UCCX 8.0, agents on multiple WAN sites, all agents using extension mobility.
The documentation I've been able to find describes three different recording methods:
Using Desktop Recording service (Endpoint) to record from an agent’s desktop.
Server Recording - Uses SPAN (not so good for remote sites)
Network Recording - Uses CUCM recording service / SIP trunk / phone's built in bridge.
Network recording or Desktop recording should be suitable for the customer but it seems that Extension mobility is not supported. Extension Mobility is not mentioned in the 8.5 installation guide, it is mentioned as ‘not supported’ in the 8.0 guide as follows:
'Server Recording and Network Recording have the following limitations:
• Extension mobility is not supported.'
Neither version's documentation specifically mention extension mobility in relation to the desktop recording method, though I realise this is a similar approach to the 'server recording' method.
So the question I have is: Is extension mobility supported in any way on version 8.0, or version 8.5 for recording? And if so which recording method(s) are supported?
Thanks,
Jonathan

Hi,
I had more luck asking questions over at the Calabrio forum - they make the software and Cisco re-brand a version of it - there is some good info on their portal (http://portal.calabrio.com), you have to register but it's fairly painless. The answer I got was:
"QM Desktop recording has always supported extention mobility as it determines the recorded user by the desktop user's login. Extention mobility was not supported for Server and Network recording until the Calabrio QM 8.6.2 release in April 2011 and will be added to Cisco QM starting with QM 8.5.2 in June 2011"
Regards,
Jonathan

Cisco Unified WFO - Call Recording and Quality Management stops recording with conferenced translator

I'm having an issue whenever one of our employee's conferences in an external translator, as soon as they bridge the customer into the call with the translator QM stops recording. I can hear the intial conversation with the customer, and then the employee put the customer on hold and call the translator service. Only when the two are bridged together the call always stops recording.
We are performing all recording on the server side, and are not using Quality Manager Desktop agents.
Any help would be much appreciated. Thanks!
-Chris

So I found the following information listed below. I don't manage the Cisco Unified CM portion of our telco system. Can we limited the ourselves to a single Codec, and would this even resolve the issue. Does this cause other issues if we didn't limit the devices that are recording to a single codec?
Recording IssuesThis topic explains how to diagnose and resolve problems that occur with contact
recordings.
Calls for devices configured for Network Recording drop when you try to conference
or transfer a call.
Symptom. When you try to transfer or conference a call and one or more devices on
the call configured for network recording, the transfer or conference fails and
parties drop off of the call.
Cause. Cisco Unified CM does not support codec changes for devices that are
configured for call recording. The codec must remain the same throughout the life
of the call. For conference calls, the conference bridge must support the codec
used before the conference completes.
Solution. Update the Cisco Unified CM configuration to ensure that no codec
changes occur for network recorded devices.

If you forward your calls to another phone and turn off yours, can the phone be tracked?

I have an iphone 5 and I want it not to be tracked but still get my calls. If I cal forward my calls to another iphone and turn off the one I dont want tracked, will it render it untrackable?

If you are seeing your device on iCloud's Find my IPhone section (even if it says offline), then it is tied to your iCloud Account, meaning you had that "Find My iPad" turned on in the iPad's Settings > iCloud.
If someone has it, they probably turned it off to prevent you from tracking it, but yes, you can use lost mode to lock up the iPad and put a message and phone number to call if found. If or when the device is turned back on, this message will be displayed on the iPad. iCloud: Use Lost Mode
Choosing the play sound option will put it in the system to play the sound when it is turned on again.

Curve w/ paint coming off return key and call key

curve w/ paint coming off return key and call key...anyone else? i don't think being 6 mo old it should look yrs old! by the way u have to replace the intermediate board...
o z
oz
in god we trust
god bless the world

Nope, mines good! IF you have time on warranty maybe a replacement?
Just a thought.
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Calling a PL/SQL Procedure from button and then return to same page

I have a button called "Process" that I want to call the database stored procedure "BUILD_WORKDAYS" with parameters P14_PROCESS_YEAR as an input parameter. I have the button on the page but is this a PL/SQL Page Process or a PL/SQL Branch? I have tried a number of combination but don't seem to be able to get the database procedure to execute. What should the syntax be?
Thanks

"I have a button called "Process" that I want to call the database stored procedure "BUILD_WORKDAYS" with parameters P14_PROCESS_YEAR as an input parameter"
Assuming you are starting with an empty Apex page:
1) Create a region of some type or other, say HTML.
2) Create a button of type "Create a button in region position" in this region. Sounds like you've already done this and called it PROCESS.
3) Create a process of type PL/SQL, On Submit After Computations and Validations.
4) Call it "Execute BUILD_WORKDAYS" (or whatever you like)
5) The PL/SQL for the process is:
BUILD_WORKDAYS(:P14_PROCESS_YEAR);
6) Specify a success and failure message.
7) In the final step of the Create Page Process wizard, select your button from the "When button pressed" select list.
8) Then create an unconditional branch branching to the same page.
That should be it.
Andy

Cisco Forward and Call return to Voicemail on CUCM 8.5 question

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