CUPC Incoming Call from Hunt Group

Hi all!
Is is possible that I can see whenever someone calls me on my extension, or whenever a call comes in from a hunt group? Can I differ the incoming calls?
Regards
Rene

Are you talking about the attendant console hunt group? If yes, then you can do this by using the broadcast hunting feature. What it does is, all calls to the hunt pilot will be placed in queue and will be displayed in the Broadcast calls window within the Attendant Console. All uses can see the calling number of calls in queue and pick and chose the calls they want to answer.
Check this link for more details
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/4_1/sys_ad/4_1_3/ccmfeat/fsccmac.htm#wp1144539
Regards,
Anup

Similar Messages

2811(NM-8AM) does not answers incoming call from PC

Dear friends,
We are planning to setup an out of band management connectivity, in our test setup we have a Cisco 2811 ISR (with IOS 12.4(9)T5 ) with NM-8AM (Analog modem module) installed.
Test Setup:
1. One Analog line (PSTN) connected to PC via built-in modem
2. Second Analog line (PSTN) connected to Cisco 2811 (NM-8AM) terminated on the analog modem module
3. Both the analog lines are working fine.
Test Scenario:
1. The router should accept the incoming call from PC. And from the PC we should be able to telnet into the router via dial up connectivity (Out of band)
Test procedure:
1. The Cisco 2811 router is configured as follows:
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Dial-Up-CPE
boot-start-marker
boot-end-marker
logging buffered 16384 debugging
enable password cisco
ip cef
voice-card 0
no dspfarm
username cisco password 0 cisco
interface Loopback0
ip address 10.10.10.1 255.255.255.255
interface FastEthernet0/0
ip address 20.20.20.1 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Group-Async1
ip unnumbered Loopback0
encapsulation ppp
dialer in-band
dialer idle-timeout 600
dialer-group 1
async mode interactive
peer default ip address pool DIALIN
ppp authentication pap chap
group-range 1/0 1/7
ip local pool DIALIN 172.16.16.1 172.16.16.5
ip http server
no ip http secure-server
dialer-list 1 protocol ip permit
control-plane
line con 0
exec-timeout 0 0
line aux 0
exec-timeout 0 0
password cisco
modem InOut
transport input all
stopbits 1
speed 38400
flowcontrol hardware
line 1/0 1/7
modem InOut
transport input all
stopbits 1
flowcontrol hardware
line vty 0 4
password cisco
scheduler allocate 20000 1000
2. With the above configuration, the router is not able to answer the incoming call from the PC.
3. And from the PC we are able to dial to a PSTN number at the router side, PC gives a proper dial tone and once the number is dialed, the call is landed on the other side (router end) and it rings continuously. But, the router does not answer this incoming. The analog line at the router end is terminated on async interface 1/0.
As I am new to dialup connectivity with NM-8AM, Could some please suggest the proper configuration in our router configuration OR Are missing any commands?
Any links/references would be of great help.
Thank you
Pradeep

Hi,
I guess I'm having exactly the same setup. The integrated modem on the other end does pick up the call but soon disconnect after displaying "Waiting for carrier".
Any clue?
Thanks,
Eyad

Is it possible to pick up an incoming call from an extension that is not physically ringing?

Is there a way to pick up an incoming call from an extension that is not physically ringing?
For example if an incoming call was ringing at EXT 1019 and they were away from their desk momentarily, could the call be picked up from another extension by pressing a number sequence or something?
we are using UC560. if you could tell in detail how it can be setup(config) and how it will work.Thank you in advance.

Please rate helpful posts.
Thanks,
Alex
Here is the inof for the Call Manager Express.
Enabling Call Pickup
To enable Call Pickup features on SCCP or SIP phones, perform the following steps.
Prerequisites
•SIP phones require Cisco Unified CME 7.1 or a later version.
•The PickUp and GPickUp soft keys display by default on supported SCCP and SIP phones. If previously disabled, you must enable these soft keys with the softkeys idle command.
Restrictions
•SIP phones that do not support the PickUp and GpickUp soft keys must use feature access codes (FACs) to access these features.
•Different directory numbers with the same extension number must have the same Pickup configuration.
•A directory number can be assigned to only one pickup group.
•Pickup group numbers can vary in length, but must have unique leading digits. For example, if you configure group number 17, you cannot also configure group number 177. Otherwise a pickup in group 17 is always triggered before the user can enter the final 7 for 177.
•Calls from H.323 trunks are not supported on SIP phones.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. service directed-pickup [gpickup]
5. fac {standard | custom pickup {direct | group | local} custom-fac}
6. exit
7. ephone-dn dn-tag [dual-line | octo-line]
or
voice register dn dn-tag
8. pickup-group group-number
9. pickup-call any-group
10. end
DETAILED STEPS
Command or Action
Purpose
Step 1
enable
Example:
Router> enable
Enables privileged EXEC mode.
•Enter your password if prompted.
Step 2
configure terminal
Example:
Router# configure terminal
Enters global configuration mode.
Step 3
telephony-service
Example:
Router(config)# telephony-service
Enters telephony-service configuration mode.
Step 4
service directed-pickup [gpickup]
Example:
Router(config-telephony)# service directed-pickup gpickup
Enables Directed Call Pickup and modifies the function of the GPickUp and PickUp soft keys.
•gpickup—(Optional) Enables using the GPickUp soft key to perform Directed Call Pickup on SCCP phones. This keyword is supported in Cisco Unified CME 7.1 and later versions.
•This command determines the specific soft keys used to access different Call Pickup features on SCCP and SIP phones. For a description, see the service directed-pickup command in the Cisco Unified CME Command Reference.
Step 5
fac {standard | custom pickup {direct | group | local} custom-fac}
Example:
Router(config-telephony)# fac custom pickup group #35
Enables standard FACs or creates a custom FAC or alias for Pickup features on SCCP and SIP phones.
•standard—Enables standard FACs for all phones. Standard FAC for Park Retrieval is **10.
•custom—Creates a custom FAC for a feature.
•custom-fac—User-defined code to dial using the keypad on an IP or analog phone. Custom FAC can be up to 256 characters and contain numbers 0 to 9 and * and #.
Step 6
exit
Example:
Router(config-telephony)# exit
Returns to privileged EXEC mode.
Step 7
ephone-dn dn-tag [dual-line | octo-line]
or
voice register dn dn-tag
Example:
Router(config)# ephone-dn 20 dual-line
or
Router(config)# voice register dn 20
Enters directory number configuration mode.
Step 8
pickup-group group-number
Example:
Router(config-ephone-dn)# pickup-group 30
or
Router(config-register-dn)# pickup-group 30
Creates a pickup group and assigns the directory number to the group.
•group-number—String of up to 32 characters. Group numbers can vary in length but must have unique leading digits. For example, if there is a group number 17, there cannot also be a group number 177.
•This command can also be configured in ephone-dn-template configuration mode and applied to one or more ephone-dns. The ephone-dn configuration has priority over the template configuration.
Step 9
pickup-call any-group
Example:
Router(config-ephone-dn)# pickup-call any-group
or
Router(config-register-dn)# pickup-call any-group
Enables a phone user to pickup ringing calls on any extension belonging to a pickup group by pressing the GPickUp soft key and asterisk (*).
•The ringing extension must be configured with a pickup group using the pickup-group command.
•If this command is not configured, the user can pickup calls in other groups by pressing the GPickUp soft key and dialing the pickup group number.
Step 10
end
Example:
Router(config-ephone-dn)# end
or
Router(config-register-dn)# end
Exits configuration mode.
Examples
The following example shows the Group Pickup and Local Group Pickup features enabled with the service directed-pickup gpickup command. Extension 1005 on phone 5 and extension 1006 on phone 6 are assigned to pickup group 1.
telephony-service
load 7960-7940 P00308000500
load E61 SCCP61.8-2-2SR2S
max-ephones 100
max-dn 240
ip source-address 15.7.0.1 port 2000
service directed-pickup gpickup
cnf-file location flash:
cnf-file perphone
voicemail 8900
max-conferences 8 gain -6
call-park system application
transfer-system full-consult
fac standard
create cnf-files version-stamp 7960 Sep 25 2007 21:25:47
ephone-dn 5
number 1005
pickup-group 1
ephone-dn 6
number 1006
pickup-group 1
ephone 5
mac-address 0001.2345.6789
type 7962
button 1:5
ephone 6
mac-address 000F.F758.E70E
type 7962
button 1:6

How can i hide incoming calls from showing on my screen

How can I stop incoming calls from showing on my home screen?

If you just don't want to see the call notification on your home screen go to Settings, Notification, Phone and turn Notification Center to Off. Or you can simply select None and calls will be listed in the Notification center but you won't see a banner / alert on the home screen.
This also eliminates the Caller ID from being displayed before you answer the call.

HT201229 How can I access a list of incoming calls from a blocked number

How can I access a call log of incoming calls from a blocked number on my iphone?

Calls from blocked #'s won't go through, so they won't be listed on your device.

An incoming call from VoIP. The SPA122 generate Dial Tone after the far end hung up rather busy tone.

Hi All!
I have a problem with the SPA122 telephony adapter, uncorrectly process the subscriber signaling at the end of the call.
1) Outbound call from FXS port SPA122 . When a remote caller hangs up first , the subscriber SPA122 Reorder Tone played with a delay specified in the Reorder Delay. This circuit is working properly.
2 ) Incoming call from VoIP to the SPA122. When a remote caller hangs up first , subscriber on the FXS port of the SPA122 hears silence ~ 3-4 seconds , then SPA122 plays Dial Tone, as if he had just picked up the phone and he 's going to call . No signal lights out (Busy Tone or Reorder Tone) will not play .
Config is attached.
Model: SPA122, LAN, 2 FXS
Hardware Version: 1.0.0 Boot Version: 1.0.1 (Oct 6 2011 - 20:04:00)
Firmware Version: 1.3.2-XU (014) Jul 2 2013
Recovery Firmware: 1.0.2 (001)
WAN MAC Address: 6C:20:56:55:3A:B6
Host Name: SPA122
Domain Name: (none)
Serial Number: CCQ16450LG3
However, other VoIP terminals registered to Huawei, including older versions of the Linksys SPA2102 work in these scenarios correctly.
Where to kick it?

[2] is misconfiguration on your's side. You have CPC turned on, but no CPC capable device. Set CPC Duration to zero to turn off CPC.
By the way, wrong forum for your question. You should consider to move it to space.

Problem with incoming call from pstn

Hi,
we have a 2610XM router with CCME 3.2 and 4 bri
and we don't redirect incomming call from pstn on a internal IP Phone.
which command for redirect call ?
my isdn number (french) is 0156838050
and when we call this number with my mobile (number 0685284832), look this debug
debug isdn q931:
Mar 22 12:53:46.350: ISDN BR1/0 Q931: Ux_DLRelInd: DL_REL_IND received from L2
*Mar 22 12:53:46.378: ISDN BR1/0 Q931: RX <- SETUP pd = 8 callref = 0x73
Bearer Capability i = 0x8090A3
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0x89
Calling Party Number i = 0x2083, '685284832'
Plan:Unknown, Type:National
Called Party Number i = 0x81, '8050'
Plan:ISDN, Type:Unknown
Sending Complete
*Mar 22 12:53:46.382: %ISDN-6-LAYER2UP: Layer 2 for Interface BR1/0, TEI 64 changed to up
Router#
*Mar 22 12:53:46.415: ISDN BR1/0 Q931: TX -> CALL_PROC pd = 8 callref = 0xF3
Channel ID i = 0x89
*Mar 22 12:53:46.439: ISDN BR1/0 Q931: TX -> DISCONNECT pd = 8 callref = 0xF3
Cause i = 0x8081 - Unallocated/unassigned number
*Mar 22 12:53:46.527: ISDN BR1/0 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x73
Cause i = 0x87E4 - Invalid information element contents
Router#
thanks for your help

You have to either use a transfer pattern or a translation rule.
http://www.cisco.com/en/US/products/sw/iosswrel/ps5012/products_feature_guide_chapter09186a00801812db.html

I teach Spinning- is it possible to stop incoming calls from interrupting the music?

I teach Spinning- is it possible to stop incoming calls from interrupting the music? I have the newest iphone.

Turn on airplane mode.
That turns off the cellphone network so no-one can call you.
Don't worry they will get diverted to the answerphone.

I can't decline incoming call from my iphone display

Dear friends,
I'm using iPhone 6 but I can't decline incoming call from my iphone display. I saw YouTube video to view reject/ decline incoming call I need to press power button twice. But seems odd to me. So now what I'll do or if there is no option for that then what'll Apple do?

The Decline button will only appear if the phone is unlocked. If the phone is locked when you receive a call, the only option is to double-tap the sleep/wake button to send it directly to voicemail. If you tap the button once, it will silence the call, but it will continue the ring cycle before going to voicemail. If the screen is awake, then you will see the Decline button. Feedback to Apple goes HERE, and click on the appropriate link.

Incoming calls from telephones to my skype number drop after 3-5 seconds

Incoming calls from telephones to my skype number drop after 3-5 seconds
I spent a while trying to seriously diagnose this problem with Skype support a year ago, but didn't get anywhere; I came to the conclusion that there was something wrong with the network at my end. But it's irritating me again now, so I thought I'd check whether anyone else gets this problem. It's incoming calls only, to the skype number only. I can recieve incoming calls from Skype fine, I can make outgoing calls to Skype and to telephone numbers fine, the problem affects both Skype on my Android Mobile (when it's connected to my home WiFi) and Skype on my Windows Desktop - so, not a software configuration thing.
When making a test call from my mobile to my Skype In number, it drops at the Skype end several seconds before the mobile notices the line is dead; definitely dropping out at the Skype level, and not the telephone network level.

Hi, ScottMaggie, and welcome to the Community,
I have referred your report to those to whom I report; I am not affiliated with Skype Customer Service.
What you so well describe here, a number transfer (highlighted by me in orange type) is not something Skype has the facility to do:
None would agree to moderate between myself and the current number holder to do a transfer. If that person simply gives up the number it would be a 90 day wait with no way to guarantee I'd get the number back. They need to do a transfer as this is the only way to make it work. I have no way to contact them aside from calling the number and leaving voicemails.
In the system's simplicity, there is no flexibility.
Just out of curiosity ... and curiosity alone, have you checked the e-mail address on your account to double-check it is the correct registered e-mail address? I ask because I recently did a Skype Number renewal, and I received e-mail alerts every step of the way.
Best regards,
Elaine
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I was sitting here and my phone randomly said incoming call from my friend that is an iphone user. I answered it and nothing was there. He texted me the screen shot on his phone that said I called him, but it showed on my phone as an incoming call. Why?

I was sitting here and my phone randomly said incoming call from my friend that is an iphone user. I answered it and nothing was there. He texted me the screen shot on his phone that said I called him, but it showed on my phone as an incoming call. Why?

Heyc123 wrote:
I had bought an iphone from this guy . ... . I had try to call him but he wouldnt answer and went ghost on me . Is there another way to activate the phone without putting in his apple id
Nope.
http://support.apple.com/kb/TS4515

Intermittent Failure of incoming calls to Response Group

Hello,
We're running Lync Server 2010 and intermittently calls coming into the response group don't ring the attendant or take a long time to start ringing. We ARE seeing events in the event log pertaining to the LS Response Group Service as show below. I
haven't been able to find anything on this on TechNet or anywhere else. We are running all of the latest updates. Any help would be greatly appreciated!
Event Log
Log Name:      Lync Server
Source:        LS Response Group Service
Date:          11/22/2013 3:43:11 PM
Event ID:      31172
Task Category: (2001)
Level:         Warning
Keywords:      Classic
User:          N/A
Computer:      vm-12-04.sipdomain.net
Description:
The workflow runtime encountered an error while connecting the call.
The workflow runtime encountered a critical error.
Failure occurrences: 9, since 11/21/2013 7:42:57 AM.
The last encountered error was from a workflow having the display name: Main RSP, the URI: sip:[email protected], and the GUID: 647cc313-9ea7-46a5-a153-3e5a164b9785.
Exception: System.InvalidOperationException - Accept Call activity '_actvAcceptCall' cannot run. The Call (AudioVideoCall or InstantMessagingCall) is not in the incoming state. The current state is 'Terminated'.
Inner Exception: -
Event Xml:
<Event xmlns="http://schemas.microsoft.com/win/2004/08/events/event">
<System>
    <Provider Name="LS Response Group Service" />
    <EventID Qualifiers="34769">31172</EventID>
    <Level>3</Level>
    <Task>2001</Task>
    <Keywords>0x80000000000000</Keywords>
    <TimeCreated SystemTime="2013-11-22T21:43:11.000000000Z" />
    <EventRecordID>276664</EventRecordID>
    <Channel>Lync Server</Channel>
    <Computer>vm-12-04.sipdomain.net</Computer>
    <Security />
</System>
<EventData>
    <Data>9</Data>
    <Data>11/21/2013 7:42:57 AM</Data>
    <Data>Main RSP</Data>
    <Data>sip:[email protected]</Data>
    <Data>647cc313-9ea7-46a5-a153-3e5a164b9785</Data>
    <Data>System.InvalidOperationException - Accept Call activity '_actvAcceptCall' cannot run. The Call (AudioVideoCall or InstantMessagingCall) is not in the incoming state. The current state is 'Terminated'.</Data>
    <Data>-</Data>
</EventData>
</Event>
Calls will begin to come in again after this but it appears the service is recovering from some type of failure state.
Thanks for any help!
RWK

Did this issue happen to multiple response group?
Did this happen to calls from PSTN calls?
This problem is mostly related with network issue.
The following is a similar thread for you:
http://social.technet.microsoft.com/Forums/lync/en-US/d0e53ab2-42e7-4d79-be01-b8770d508133/calls-to-workflows-fail
For further troubleshooting. You can use Lync Logging Tool to collect trace file on Lync Front End Server.
Lisa Zheng
TechNet Community Support

CallerID not correct when calling a Hunt Group #

Not sure what happened, but when users called the 4357(Hunt Group) the caller id would show "Helpdesk and also the callers name and ext" on the caller ID of the users in the 4357 (Hunt Group List).
Now when users call the 4357 the caller id on the (Hunt Group) end just shows the "callers name and ext".
We like when it would show Helpdesk along with the user info because we know the person is actually calling the Helpdesk and not calling directly. For example, my boss knows to only pick up calls that come into the helpdesk when he is the only one here, but now he is picking them all up because it looks as though they are calling him directly.
Any suggestions?

Hi Aaron,
I have made the changes you told o make and its working now.
Thanks.
Syed Kazi.

After updating to version 5 for iPhone 4, I am unable to hear incoming call from earpiece. Speaker phone ok.

Unable to hear incoming call in earpiece. Speaker phone ok.

ok thanks for someone "Wjolsten" pointing out that i might have put on the 'ringer silence' button on on the side of the phone. ......................... boy do i feel stupid. thanks all the same for the 'tech support' from Apple, Geniuses?? i dont think so.

Inter-Trunk not route incoming calls from out

Hi,
I setup one extra gateway where I try to route part of our calls. So far I have success to route internal calls into there, but when I'm making a test call from outside that ends into "number is not used" problem.
I have:
- Route ready, elsewere the internal calls are not working.
- PSTN usage, linked to the Route
- Trunk configuration where I have selected the PSTN usage
- Incoming numbers are coming in E164 format
I have also tested the "Test-CsInterTrunkRouting" and that gives "pass":
FirstMatchingRoute : Description=;NumberPattern=^\+358123654789;Name=Test
                     Gateway;SuppressCallerId=False;AlternateCallerId=
MatchingUsage      : Test PSTN Usage
MatchingRoutes     : {Description=;NumberPattern=^\+358123654789;Name=Test
                     Gateway;SuppressCallerId=False;AlternateCallerId=}
But still, when I made a call from outsited the OCSLogger shows that mediation server try to offer call to Front-End which says only: "SIP/2.0 404 Not Found" and then bye-bye.
What is the missing magic, which made the mediation server to see alternative route? I hope it is not required that mediation server must be collocated on the Front Ends, as that one I do not have.
Any good ideas?
ps.
I'm not sure does it matter, but my Lync gives "SIP/2.0 403 Forbidden" when there is coming call from extra gateway. But as the calls into there works, then I don't see why external calls should not also work.
Petri

Could it be even so, that intra-trunk routing requires consolidated mediation server? As the call is owned by the Mediation server (stanalone), and it is trying to offer that to FE. FE reply "does not exist". Because of the standalone Mediation
server does not have the call routing engine like FE have, the call is lost.
I started to think above as Lync users are able to call to that number. So FE is able to do the routing and get calls into the correct place.
I have to say also, I have read
Ken's blog about inter-trunk routing, I have to say that I'm not so sure what he means by this: "Fortunately, in most cases, adding PSTN usages to the trunk has no effect, since there is almost always a Lync user assigned to the incoming phone
numbers". Why to add additional routing for the numbers which are already inuse? I hope it is not required, that you need to have a users ID for each number you do the inter-trunk routing?
Petri

CUPC Incoming Call from Hunt Group

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