Interclustering not working in CUCM 7 & 8.5

hello,
i am having CUCM 7 at my end and CUCM 8.5 at another location. Earlier i was able to make successfull calls from either of clusters, but currently dial pattern has been change and i am not able to make calls from my CUCM cluster to another CUCM cluster.
I am having one another location and one trunk has been configured for the same and calls through that trunk are working fine.
I have just changed the dial pattern from one to another and not made any changes in trunks, route pattern,rute list or CSS. Can anyone guide here to what needs to be done to make successfull calls between two CUCMs clusters.

Hi Chris,
Customer have just changed the route pattern number 1241XXXX to 1201XXXX. In the dial number analyzer it does show that it is taking route pttern as 1201XXXX, but call is not getting connected. What can be the issue?

Similar Messages

Party Entrance Tone not working on CUCM 10.5(1.10000.7)

Party Entrance Tone not working on CUCM 10.5(1.10000.7) . This is an intermitent issue for Meet Me conference.
++On the CUCM Service Parameter Party Entrance Tone has is TRUE .
++On Directory number Party Entrance Tone is ON . However the Party entrance tone (Beep tone) not coming when Meet Me conference going on .
Gateway 3945 with IOS Version 15.3(3)M5, RELEASE SOFTWARE (fc3).
The conference Configureation as below.
sccp local GigabitEthernet0/0
sccp ccm 172.20.0.152 identifier 1 priority 1 version 7.0
sccp ccm 172.20.0.153 identifier 2 priority 2 version 7.0
sccp ccm 172.22.0.110 identifier 3 priority 3 version 7.0
sccp ccm 172.22.0.111 identifier 4 priority 4 version 7.0
sccp
sccp ccm group 1
bind interface GigabitEthernet0/0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate ccm 3 priority 3
associate ccm 4 priority 4
associate profile 2 register ASB-DR-GW04CFB
associate profile 1 register ASB-DR-GW04XCO
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729br8
codec g729r8
maximum sessions 60
associate application SCCP
dspfarm profile 2 conference
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum conference-participants 32
maximum sessions 12
associate application SCCP
Can any one tell me how the party entrance tone happening when a confernce session open.Is it generated from the hardware Confernce resource which configured on the gateway while mixing the RTP streams ? Is there anything to do with CUCM or any service in CUCM to generate the Party entrance tone ?

Hi,
It looks like the following bug
https://tools.cisco.com/bugsearch/bug/CSCup27560/?referring_site=bugquickviewredir
MeetMe Party Entrance Tones stop working
CSCup27560
Description
Symptom:
- MeetMe party entrance tones stop getting played to participants when they join/leave
- The issue will go away for a few weeks if the CCM service is restarted, but then will come back.
- When the issue comes back, it fails 100% of the time.
Conditions:
This problem would occur when we have user having a call from user in different node initiate transfer and also initiate MeetMe and dial user in different node. When transfer completes, we will have a leak in current node MeetMe. This issue should have been there in all CCM versions.
Workaround:
Restarting the CCM service will correct the problem temporarily.
You can either try the workaround or upgrade to one of the fixed versions.
HTH
Manish

MOH not working in cucm only beep sound

HI,
I have an IP Telephony set up running on CUCM 9.1.Publisher and subscriber in a single site deployment model.
Customized MOH were configured and running fine before.Now users are complaining that MOH is not working(both internal and external calls) and only beep sound is hearing when putting on hold .
I have checked the MOH server is registered and I have restart the Cisco IP Voice Media Streaming App service for both pub and sub
But when i am selecting the Sample Audio source its working fine and I can hear the cisco sound on MOH.
I have added another audio source for testing and assigned to the phones but result is same.
Multi cast allowed on the Audio source and its enabled in the MOH server.
Attached the screen shot for MOH audio source and MOH server.
Please advice how to fix this issue.

Beep sound indicates configuration issues, few things to check:
Make sure the DP assigned to the MOH server is set to use G711 between other regions, unless you enabled other codecs in service parameters
Make sure the MOH file is uploaded to all servers
Make sure the MOH server is assigned to MRG/MRGL of the devices needing to connect to it (phones, GWs, etc).
Can you try using unicast vs. multicast to see if that fixes it?
Chris

SNMPv3 Priv Credentials not working for CUCM Discovery

Hello Support Team
Im testing and practicing this week the CPC Small Demo and it seems that when I configure the SNMPv3 Priv Credentials (even with DES) for discover my CUCM, even if I configure the same priv credentials in Serviceability.
However, If I didnt configure the priv credencials, the CPC discover of CUCM works, this now presents the security risk that all the messages between the manager and agent are in plaintext ¿dont it?
I attach the screen with the configurations in CUCM and CPC, with the discovery errors.
Thanks for your help and advice.
Best Regards.

Are you allowing IMAP access into CUC? you set this in the "Voicemail Users" Class of Service on CUC, so it will apply for all users once the COS is changed.
Also you might want to set the Credentials source for voicemail service, on CUPS to "CUP"

Ringback not working when call is offered

When a call is offered to an agent the caller hears dead silence on their end. I have the Call Control Group configured to use the correct MOH source. I have verified that the file will work by changing my MOH for my device to it and it plays. I have also changed all of the CTI ports to use this file for MOH and restarted the CUCM IP Media Voice Streaming Service still without any luck. Anyone know of something I am missing? I am running CUCM 9.1.2 and UCCX 9.0.2

One common cause of silent MoH is multicast not working. CUCM sends SDP with a multicast address (because the audio source, moh server, and mrg all are set to allow it) but either the router doesn't have multicast enabled or the group isn't being propegated from CUCM to the router.
Turn off multicast on those three places, reset the moh server, and see if you get audio. If you do, you may still want to fix multicast and turn it back on since it provides far better scale.
Please remember to rate helpful responses and identify helpful or correct answers.

Standby PRI not working with voice Gateway Router & CUCM

Hi ALL ,
GOOD Day all of you .
I am facing a big problem i.e standby PRI not working with VG & CUCM , I have checked all the configuration parameter on VG & CUCM found ok but I am unable to make any call from standby link also incoming not come on the standby link . When I make a call on my Pilot no but getting busy tone .
I observer the some errors on VG like Cause i = 0x8286 - Channel unacceptable on my second PRI channel .
Please help me to reslove this proem .
Following are the PRI configuration Parameter on CUCM .
Product Specific Configuration Layout
Line Coding : HDB3
Framing : NON CRC4
Clock : External
Input Gain (-6..14 db) 0
Output Attenuation (-6..14 db) 0
Echo Cancellation Enable
Echo Cancellation Coverage (ms) 64
PRI configuration on VG
interface Serial0/0/1:15
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
BR,
SANDIPAN

Hi Craig ,
Thanks for your reply .
We are using the full 30 channel E1 PRI .
following are PRI Channel Statistics:
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 1. Layer 3 output may not apply
ISDN Se0/0/0:15, Channel [1-31]
Configured Isdn Interface (dsl) 1
   Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend)
    Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    State   : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 3 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
   Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend)
    Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    State   : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 2 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
Please find following debug error
VG_RO_01#isdn test call interface serial 0/0/0:15 09665484798
Mar 7 03:00:54.570: ISDN Se0/0/0:15 Q921: User RX <- RRp sapi=0 tei=0 nr=5
Mar 7 03:00:54.570: ISDN Se0/0/0:15 Q921: User TX -> RRp sapi=0 tei=0 nr=4
Mar 7 03:00:54.570: ISDN Se0/0/0:15 Q921: User TX -> RRf sapi=0 tei=0 nr=4
Mar 7 03:00:54.574: ISDN Se0/0/0:15 Q921: User RX <- RRf sapi=0 tei=0 nr=5
CCIL_PUNE_DR_VG_RO_01#
Mar 7 03:00:55.994: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Called num 09665484798
Mar 7 03:00:55.994: ISDN Se0/0/0:15 Q921: User TX -> INFO sapi=0 tei=0, ns=5 nr=4
Mar 7 03:00:55.994: ISDN Se0/0/0:15 Q931: SETUP pd = 8 callref = 0x0084
Bearer Capability i = 0x8890
Standard = CCITT
Transfer Capability = Unrestricted Digital
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Called Party Number i = 0x81, '09665484798'
Plan:ISDN, Type:Unknown
Mar 7 03:00:56.006: ISDN Se0/0/0:15 Q921: User RX <- RR sapi=0 tei=0 nr=6
Mar 7 03:00:56.018: ISDN Se0/0/0:15 Q921: User RX <- INFO sapi=0 tei=0, ns=4 nr=6
Mar 7 03:00:56.018: ISDN Se0/0/0:15 Q931: RELEASE_COMP pd = 8 callref = 0x8084
Cause i = 0x8286 - Channel unacceptable
Mar 7 03:00:56.022: ISDN Se0/0/0:15 Q921: User TX -> RR sapi=0 tei=0 nr=5
Mar 7 03:00:59.995: ISDN Se0/0/0:15 Q921: User TX -> INFO sapi=0 tei=0, ns=6 nr=5
Mar 7 03:00:59.995: ISDN Se0/0/0:15 Q931: SETUP pd = 8 callref = 0x0084
Bearer Capability i = 0x8890
Standard = CCITT
Transfer Capability = Unrestricted Digital
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Called Party Number i = 0x81, '09665484798'
Plan:ISDN, Type:Unknown
Mar 7 03:01:00.007: ISDN Se0/0/0:15 Q921: User RX <- RR sapi=0 tei=0 nr=7
Mar 7 03:01:00.019: ISDN Se0/0/0:15 Q921: User RX <- INFO sapi=0 tei=0, ns=5 nr=7
Mar 7 03:01:00.019: ISDN Se0/0/0:15 Q931: RELEASE_COMP pd = 8 callref = 0x8084
Cause i = 0x8286 - Channel unacceptable
Mar 7 03:01:00.019: ISDN Se0/0/0:15 Q921: User TX -> RR sapi=0 tei=0 nr=6
Mar 7 03:01:03.995: ISDN Se0/0/0:15 Q921: User TX -> INFO sapi=0 tei=0, ns=7 nr=6
Mar 7 03:01:03.995: ISDN Se0/0/0:15 Q931: RELEASE_COMP pd = 8 callref = 0x0084
Cause i = 0x80E6 - Recovery on timer expiry
Mar 7 03:01:03.995: ISDN Se0/0/0:15 **ERROR**: CCPCC_CallOrigination: SETUP timed-out (2nd T303) to NETWORK. The SETUP failed.
BR ,
SANDIPAN

CUCM - CONDUCTOR - MCU not Working (BUSY)

Hi,
I try to configure CUCM with Conductor and a single MCU behind but for some reason its not working. The call is reaching Conductor (Call History) and also MCU (SIP Event Log) but in MCU LOG i see this:
In CUCM Traces i see forbidden, but why:
Conductor:
Any suggestions?
Cheers

AFter setting the SIP trunkt o non secure its working. But its different from the deployment guide:
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/conductor/config_guide/xc3-0_docs/TelePresence-Conductor-Unified-CM-Deployment-Guide-XC3-0.pdf
Page 40
Task 27
Task 28
Here its clearly written we should set this to secure.
But if the Endpoint/Client is not secure it seems we got a 403 Forbidden.

CUCM 9 , SD BLF is not working

Hello ,
i have configured SD BLF for CIPC which is working fine , but it is not working for Cisco 7965 + 7916
Please Advice

Do yu have the Subcribe CSS on the phones and are the phones/lines using the same presence groups?
HTH
Regards,
Yosh

Cisco Jabber Softphone not working

Hi
The softphone is not working for cisco jabber windows
Attached is the logs
CUCM AND CUPS version 9.x
I found the below but is not clear to me what is the issue
2013-10-10 12:53:42,577 DEBUG [0x00001360] [tionplugin\StatusBarPhoneButton.cpp(127)] [plugin-runtime] [OnCanSelectDeskphone] - enabling deskphone menu items
2013-10-10 12:53:42,577 DEBUG [0x000011a8] [src\config\TftpHelper.cpp(114)] [csf.ecc] [ecc::TftpHelper::doRetrieveFile] - Received error: eFileNotFound downloading file using HTTP from 'http://10.255.0.11:6970/CTLSEPCSFKIKACHRISTOU.tlv', not trying TFTP following an explicit HTTP "not found" error.

Hi Chrysostomos,
I assumes your device name is CSFKIKACHRISTOU
You have added an ower id t the Device
You have allowed CTI Control for the device
You have associated the device to your user in End User Page under Device association.
Your user has is a member of both Standard CCM End Users and Standard CTI Enabled groups.
Now if you are using as softphone, the jabber client retrieves the TFTP server IP Addresses from IMP. Once it has the address, the client download the CSF config file direct from the TFTP Server.
Have you configured the TFTP Servers in IMP?
Application->Legacy Clients->Settings --- specify primary and backup tftp servers.
Exit from Jabber completely and re-login.
See how that goes.
Ben
Please rate useful posts

UCCX: Silent monitoring is not working and recorded file is also not playing

Hi,
Running UCM 8.5 cluster with UCCX 8.5.
In CSD, silent monitoring is not working.
Windows XP OS is running on supervisor desktop. when i select the agent
then the silent monitoring icon will highlighted once i click on the icon,
after 5-7 seconds i am getting an error "Silent monitoring session has
failed".
Not able to playback the recorded files:
I can able to record and the file size is 2.6mb for approximate 3min
recording.When i tried to play from supervisor record viewer it is playing
but no voice is coming. And i downloaded using play & save option and
tried to play the .wav file but i can't hear any voice.
Problem is while playing back the recorded file. I can able to see the
recorded files in the UCCX.
Please help me out!!!
Thanks & Regards,
Krishna

It's likely the same problem: CAD is not forwarding packets to CSD or the CCX recording service. Two common causes for this are:
The phone is not spanning to the PC port or does not allow PC Port VLAN Access. Change these options to true on the phone's configuration in CUCM.
The NIC of the agent's PC running CAD is not processing 802.1q-tagged Ethernet frames. It must not drop these and pass them into the Windows NDIS stack for CAD to get them. Google is your friend here; this commonly requires registry changes to make the NIC process the packets.

Outbound FAX is not working

Outbound FAX is not working. Below is the Call Flow
FAX ServeràH.323àCUCMàMGCPàGatewayàE1 PRIàPSTN
The CUCM version is 8.0.3 while the gateway IOS version is 12.4(24)T5.
Collected the following
Packet Captures between FAX server and CUCM
CUCM traces
Debugs in PSTN Gateway
Below are the observations
H.225 Setup message is seen in packet captures and CUCM traces. See snippet below
16:53:00.684 |In Message -- H225SetupMsg -- Protocol= H225Protocol|*^*^*
16:53:00.684 |Ie - H225BearerCapabilityIe -- IEData= 04 03 90 90 A5 |*^*^*
16:53:00.685 |Ie - Q931CalledPartyIe -- IEData= 70 09 A1 39 36 33 32 33 39 39 38 |*^*^*
16:53:00.685 |Ie - H225UserUserIe -- IEData= 7E 00 86 05 30 88 06 00 08 91 4A 00 05 22 80 B5 00 00 30 13 44 69 61 6C 6F 67 69 63 20 43 6F 72 70 6F 72 61 74 69 6F 6E 00 AC 15 98 05 06 B8 00 14 15 79 1B C9 3E 02 1F 3F 03 80 B9 FE 5F AC 75 00 D5 0D 98 00 07 00 AC 15 98 14 07 FB 11 00 25 13 79 1B C9 3E 02 1F 3F 03 80 B9 FE 5F AC 75 01 00 01 00 01 00 01 00 01 00 01 40 40 B5 00 00 30 14 44 69 61 6C 6F 67 69 63 20 43 6F 72 70 6F 72 61 74 69 6F 6E 10 80 01 00 |*^*^*
16:53:00.685 |MMan_Id= 0. (iep= 0 dsl= 0 sapi= 0 ces= 0 IpAddr=149815ac IpPort=2043)|*^*^*
16:53:00.685 |IsdnMsgData1= 08 02 5B CA 05 04 03 90 90 A5 70 09 A1 39 36 33 32 33 39 39 38 7E 00 86 05 30 88 06 00 08 91 4A 00 05 22 80 B5 00 00 30 13 44 69 61 6C 6F 67 69 63 20 43 6F 72 70 6F 72 61 74 69 6F 6E 00 AC 15 98 05 06 B8 00 14 15 79 1B C9 3E 02 1F 3F 03 80 B9 FE 5F AC 75 00 D5 0D 98 00 07 00 AC 15 98 14 07 FB 11 00 25 13 79 1B C9 3E 02 1F 3F 03 80 B9 FE 5F AC 75 01 00 01 00 01 00 01 00 01 00 01 40 40 B5 00 00 30 14 44 69 61 6C 6F 67 69 63 20 43 6F 72 70 6F 72 61 74 69 6F 6E 10 80 01 00 |*^*^*
16:53:00.685 |value H323-UserInformation ::= |*^*^*
16:53:00.685 |SPROCRas - {
h323-uu-pdu
    h323-message-body setup :
        protocolIdentifier { 0 0 8 2250 0 5 },
        sourceInfo
          vendor
            vendor
              t35CountryCode 181,
              t35Extension 0,
              manufacturerCode 48
            productId '4469616C6F67696320436F72706F726174 ...'H
          terminal
          mc FALSE,
          undefinedNode FALSE
        destCallSignalAddress ipAddress :
            ip 'AC159805'H,
            port 1720
        activeMC FALSE,
        conferenceID '1415791BC93E021F3F0380B9FE5FAC75'H,
        conferenceGoal create : NULL,
        callType pointToPoint : NULL,
        sourceCallSignalAddress ipAddress :
            ip 'AC159814'H,
            port 2043
        callIdentifier
          guid '2513791BC93E021F3F0380B9FE5FAC75'H
        mediaWaitForConnect FALSE,|*^*^*
16:53:00.686 |
        canOverlapSend FALSE,
        multipleCalls FALSE,
        maintainConnection FALSE,
        presentationIndicator presentationAllowed : NULL,
        screeningIndicator userProvidedVerifiedAndFailed
    nonStandardData
      nonStandardIdentifier h221NonStandard :
          t35CountryCode 181,
          t35Extension 0,
          manufacturerCode 48
      data '4469616C6F67696320436F72706F726174 ...'H
    h245Tunneling FALSE
}|*^*^*
Based on this, the CUCM is doing correct digit analysis and routes the call to MGCP gateway’s endpoint S0/SU0/DS1-0/[email protected]
Gateway sends the ISDN q931 setup message to PSTN for which it does get the Call Proceed and Progress indicators
The gateway is dropping call in Cause Code “0x80AF - Resource unavailable, unspecified”. See Snippet below
016325: *May 2 17:12:47.661: MGCP Packet received from 172.21.152.5:2427--->
CRCX 5493 S0/SU0/DS1-0/[email protected] MGCP 0.1
C: D000000003d354dd000000F500000016
X: 1
L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38
M: recvonly
R: D/[0-9ABCD*#]
Q: process,loop
<---
016383: *May 2 17:12:47.669: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x0016
        Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98381
                Exclusive, Channel 1
        Calling Party Number i = 0x0083, N/A
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0xA0, '96323998'
                Plan:Unknown, Type:National
016384: *May 2 17:12:47.757: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x8016
        Channel ID i = 0xA98381
                Exclusive, Channel 1
016385: *May 2 17:12:47.761: ISDN Se0/0/0:15 Q931: RX <- PROGRESS pd = 8 callref = 0x8016
        Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info
016386: *May 2 09:12:48.193: %C3800_ENVM-3-MFAIL_OFF: There is more than one failure with the Power System 1 or this Power System h
as been turned off.
016387: *May 2 17:12:51.669: //494/2E810F7780E1/CCAPI/cc_handle_inter_digit_timer:
   Generate inter-digit timeout CC_EV_CALL_DIGIT_END event
016388: *May 2 17:12:59.773: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x0016
        Cause i = 0x80AF - Resource unavailable, unspecified
From the Fax Utility it shows No answer at Fax Number. Also it rings the remmote fax machine twice (Actually i can hear the ring as the macine is beside me)
Please help me to solve the issue. Except this other functions are working properly. Moreover I can send internal Fax from one user to other.
Thanks and Regards,
Ashfaque.

Just a quick update, as per Cisco TAC the CUCM version 8.0.3 does not support gateway IOS version is 12.4(24)T5 (Later I found its only for conference option). So we upgraded the IOS to 15.1(1)T3. But still having the sam issue.
Thanks,
Ashfaque

Cisco 9971 Not Registering on CUCM 8.6

Hey,
Ive run into some trouble with a 9971 phone running on CUCM 8.6. The phone has been registered with the server before with its settings locked down so the access to the settings menu via the phone has been disabled. Around a few weeks later the CUCM server ran into some problems and had to be restored. Since then I am unable to get the phone to register back with the CUCM server. i get the rejected message in the phones tab. and the message Telephony service unavaliable on the phone itself.
I would factory reset the phone however as the settings were locked out from before i am unable to get into the menu and dont know another way to reset the phone, the method used with previous models (turn on hold # then press 1234...etc) does not work.
Any help would be appricated.
Regards,
Fraser

Hello Guys,
Anyone got to solve this problem? I have it on multiple 9951 phones, the factory reset process does not work as the phone apparently ignores that you are pressing the # button... As per the below link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/9_0/english/admin_guide/P567_BK_CEC6105C_00_admin-guide-8961-9951-9971_chapter_01100.html#P567_TK_PB2EC99B_00
"When the light on the Mute button and handset light strip turns                 off and all other lights (Line button, Headset button, Speakerphone button and                 Select button) stay green, press                 123456789*0# in sequence.
When you press                     1, the lights on the line buttons turn red.                     The light on the Select button flashes when a button is pressed.
If you press the buttons out of sequence, the lights on the line                     button, headset button, speakerphone button, and Select button turn green. You                     need to start over and press                     123456789*0# in sequence again."
In my situation this never happen, the line buttons turn on green and after that they all turn off and the phones displays "Phone not registered"

Cisco CP-78XX SIP Phone Pickup Not Work on CME

Hi,
I configured some SIP phones (CP-7821, CP-7841) with pickup function. Is it the Pickup / GPickup soft keys not function as the SIP phone? If yes, then I can use the FAC to access that? And I tried the FAC std. / custom as the pickup / gpickup .. both not work ... I don't know how to use the FAC on CME? As the FAC std., if I pickup local, that I should press (**3) > call?
Ref.:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmecover.html#45535
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmefacs.html#30064
This is the configuration:
CME-SIP-Phone#sh run
Building configuration...
Current configuration : 5413 bytes
! Last configuration change at 11:06:12 UTC Fri Nov 28 2014 by mtlops
version 15.4
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
service sequence-numbers
hostname CME-SIP-Phone
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.154-2.T1.bin
boot-end-marker
! card type command needed for slot/vwic-slot 0/0
enable secret 5 $XXXXXXXXXXXXXXXXXXXXXXXX
aaa new-model
aaa authentication login default local
aaa authorization console
aaa authorization exec default local
aaa session-id common
ip cef
no ipv6 cef
multilink bundle-name authenticated
stcapp feature access-code
voice-card 0
dspfarm
dsp services dspfarm
voice service pots
voice service voip
ip address trusted list
ipv4 10.118.0.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
no h225 timeout keepalive
call preserve
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 600 min 60
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice class h323 1
h225 timeout tcp establish 3
call preserve
voice class custom-cptone ABC-Company
dualtone disconnect
frequency 425
cadence 500 500
voice register pool-type 7821
description Cisco IP Phone 7821
reference-pooltype 6921
voice register pool-type 7841
description Cisco IP Phone 7841
reference-pooltype 6941
voice register global
mode cme
source-address 10.118.0.10 port 5060
timeouts interdigit 2
max-dn 200
max-pool 100
authenticate register
authenticate realm all
timezone 42
time-format 24
date-format D/M/Y
mwi stutter
mwi reg-e164
voicemail 5000
call-feature-uri pickup http://10.118.0.10/pickup
call-feature-uri gpickup http://10.118.0.10/gpickup
tftp-path flash:
file text
create profile sync 0001170446349417
ntp-server 10.118.0.10 mode unicast
ip qos dscp af11 media
ip qos dscp cs2 signal
ip qos dscp af43 video
ip qos dscp 25 service
camera
video
voice register dn 2
number 1000
pickup-call any-group
pickup-group 1
name BB Leung
label BB Leung
voice register dn 3
number 1001
pickup-call any-group
pickup-group 1
name CC Chan
label CC Chan
voice register dn 4
number 1002
pickup-call any-group
pickup-group 1
name DD Leung
label DD Leung
voice register dn 50
mwi
voice register template 1
softkeys hold Newcall Resume
softkeys idle Newcall Redial Gpickup Pickup Cfwdall DND
softkeys seized Cfwdall Endcall Redial
softkeys connected Confrn Endcall Hold Trnsfer
voice register pool 1
busy-trigger-per-button 1
id mac A8XX.XXXX.XXXX
type 7841
number 1 dn 2
template 1
dtmf-relay sip-notify
username 1001 password 112233
codec g711ulaw
no vad
voice register pool 2
busy-trigger-per-button 1
id mac 50XX.XXXX.XXXX
type 7841
number 1 dn 3
template 1
dtmf-relay sip-notify
username 1002 password 112233
codec g711ulaw
no vad
voice register pool 3
busy-trigger-per-button 1
id mac 00XX.XXXX.XXXX
type 7821
number 1 dn 4
template 1
dtmf-relay sip-notify
username 1003 password 112233
codec g711ulaw
no vad
license udi pid CISCO2921/K9 sn FHK1407F25D
license accept end user agreement
license boot c2900 technology-package uck9
hw-module pvdm 0/0
hw-module sm 1
username mtlops privilege 15 secret 5 $1$0qqx$1WGdfRW.flJrwmY7k8eUy0
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
ip address 10.118.0.10 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
interface SM1/0
no ip address
shutdown
service-module fail-open
interface SM1/1
no ip address
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 10.118.0.1
control-plane
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 7
associate application SCCP
shutdown
gatekeeper
shutdown
telephony-service
max-conferences 8 gain -6
transfer-system full-consult
fac standard
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
transport input all
scheduler allocate 20000 1000
end
CME-SIP-Phone#sh telephony-service fac
telephony-service fac standard
callfwd all **1
callfwd cancel **2
pickup local **3
pickup group **4
pickup direct **5
park **6
dnd **7
redial **8
voicemail **9
ephone-hunt join *3
ephone-hunt cancel #3
ephone-hunt hlog *4
ephone-hunt hlog-phone *5
trnsfvm *6
dpark-retrieval *0
cancel call waiting *1

VPN is not Configured prints on all phones now with the built-in VPN client if VPN isn't configured. That's normal and is just cosmetic. That should not be causing your registration issues.

AAR Not working when dn is different to ddi

I have configured AAR IN CUCM 6.X. When CAC is implemented phones in the branch office can ring some phone at HQ but not others. The phones that they can't reach have ddi's that do not match their 4 digit dn's. These phones that can't be reached over the pstn when cac kicks in. These phones also have single number reach (might be a red herring). I have external phone number mask to represent the full ddi and I have also used the AAR MASK but it still does not work.
I am testing this using a pri tester. I can ring the number over the pstn to the pri tester in normal conditions but when cac is working the branch phone just says not enough bandwidth and as expected the pri tester does not ring.
Any ideas what I am doing wrong.
dn= 1513
ddi 01189335701
aar external mask/phone number used 01189335701

Progress so far:
Got AAR to work adding translation patterns for 1513 to 5701, added this to aar partitions and new aar css including partitions. Added the mask to the aar 01189335701. This worked after adding XXXX instead of 5701 and then added 5701 to the aar mask and it worked ( bit flaky).
Configured single numner reach and now AAR does not work, any ideas

SX10 OBTP (one-button-to-push) not working

I have registered an SX10 to VCS and CUCM and the OBTP does not work from either. I see the meeting on the screen next to the clock; however, when it is time to join the meeting, there is no join button.
If I try moving around on the screen from Call to the Display name, it will for a second stop on the meeting information and then continue up to the name of the Codec. Also, if I go to the Call icon, it let's me go in to dial a room, but there is no meeting to select.
We are running TMS 14.4.0, CUCM 10.5, VCS X8.1.1, and SX10 TC7.1.4.

To add to my previous reply:
The Touch Ten is supported for all new endpoints in 7.2.0 specifically; "Network pairing of the Touch 10 is supported for the following endpoint models: SX10, SX20, SX80, MX200 G2, MX300 G2, MX700 and MX800." See page 12:
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/endpoint/software/tc7/release_notes/tc-software-release-notes-tc7.pdf

Interclustering not working in CUCM 7 & 8.5

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