Compatibility Uccx 9.0(2)SU2 with CUCM 9.1(2)SU3
Have seen the recent discussion on UCCX 9.0(2)SU2 and CUCM 9.1(2)SU2 here:
https://supportforums.cisco.com/discussion/12482146/compatibility-uccx-902su2-cucm-912su2
But with the release of CUCM 9.1(2)SU3 looking at the same compatibility matrix:
http://docwiki.cisco.com/wiki/Unified_CCX_Software_Compatibility_Matrix_for_9.0(2)_SU2
I do not see that UCCX 9.0(2)SU2 and CUCM 9.1(2)SU3 are compatible. Is this an error in the document or do we need to upgrade UCCX if we upgrade CUCM from 9.1(1) to 9.1(2)SU3?
Thanks,
Ryan
Hi Ryan-
Take a look the posts by Arundeep Nagaraj in this thread: https://supportforums.cisco.com/discussion/12439306/ucm-uccx-compatibiity
Thanks,
DJ
Similar Messages
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ITL Issue with CUCM 8.6(2a)SU3
Hi, We have a cluster of 6 nodes pub, 2 standalone ccm, 2 (ccm+tftp) and 1MoH server.
when we move a phone from one device pool to another, the phone displays "Registration Rejected:Security Error"
I tried deleting the ITL files on the phones but still no luck. with 'show itl' command, we see that the 'System Administrator Security Token' and the 'TFTP' function ITL Record parameters twice with two different certificates in all the nodes.
On another working cluster, the ITL Records of System Administrator Security Token' and the 'TFTP' function can be seen only once.
any idea why this difference? The change in CUCM happened before this issue was that we changed the DSCP enterprise parameter and reset all the phones and restarted the CCM service on all the 4 nodes.I would start by restarting the TVS and TFTP services on the servers within the cluster.
Thanks,
Tony
Please rate helpful posts! -
Is UCCX 8.5(1) compatable with CUCM 8.6
I just checked the Cisco Unified Contact Center Express (Cisco Unified CCX) Software and Hardware Compatibility Guide and CUCM 8.6 is not listed yet.
http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf
CUCM 8.6 addresses quite a few bugs specific to running CUCM on UCS was recommended to me by TAC.
If its not compatible yet, is there a timeframe when the compatibility guide will be updated to reflect CUCM 8.6 ?
Will it be with the upcoming SU2 for UCCX 8.5(1) ?
ThanksBvanbenschoten,
CUCM is not listed currently on the internal guides even for SU2. It is possible this may get updated, but it will be a while before we know this. CUCM certification generally lags the release of the product if for no other reason then it is such a quickly released product. If you would like, you can post this question back in about 2 months and hopefully we'll have some news. If I see something before then, I'll certainly let you know too.
Regards,
Robert W. Rogier
CSE -- TAC, UCC -
Hi,
I am referring the UCCX 8.5 compatibility matrix:
http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf
Per the above document, UCCX 8.5.1 SU1 is compatible with CUCM 8.6.1, it doesn't specify whether it is compatible with CUCM 8.6.1a.
Can someone shed some light, is it safe to assume that CUCM 8.6.1a is also compatible?
Thanks in advance,
inner_silenceSu's are also supported.
Here are 2 examples which will answer both your questions.
If CUCM 8.5(1)SU1 is qualified with UCCX 8.5(1), then all ES of CUCM 8.5(1)SU1 is considered qualified with UCCX 8.5(1).
If CUCM 8.5(1) is qualified with UCCX 8.5(1), then all SU of CUCM 8.5(1)SU1 is considered qualified with UCCX 8.5(1).
Hope this clarifies.
Shirish -
CU MeetingPlace Express 2.1.1.2 Compatibility with CUCM 7.1+
Afternoon all,
According to Cisco's compatibilty matrix, Cisco Unified Meeting Place Express version 2.1.1.2 is only compatbile up to CUCM 7.0.
I understand that CUMPE 2.1.1.2 is coming to end of sale life. However, if it currently works with CUCM 7.0, then surely it will work with later releases of CUCM up to 7.1(3)?
I appreciate Cisco might not support it if there is a problem, but was hoping someone else was running MPE 2.1.1.2 with CUCM 7.1(3b)
Thanks
JamieHi Jamie,
The Compatibility Tool does show support for MPE 2.1.1 with CUCM 7.1 (x)
Cisco Unified Communications Compatibility Tool
http://tools.cisco.com/ITDIT/vtgsca/VTGServlet
Here are 3 groups running this similar combo;
https://supportforums.cisco.com/message/3145002#3145002
https://supportforums.cisco.com/message/1330041#1330041
https://supportforums.cisco.com/message/3150066#3150066
Cheers!
Rob -
CCIE Collaboration - Integrating UCCX 9 with CUCM 9
Hello guys,
I just saw that video integrating UCCX 9 with CUCM 9 .hope you will like it..
http://voicebootcamp.com/index.php/free-video-labs/video/latest/uccx9
enjoy..No, it's not. You are reading it correctly. None of the UCCX 9x versions support CUCM 7x.
The last UCCX version to support CUCM 7x was 8.0(2)SU4.
Anthony Holloway
Please use the star ratings to help drive great content to the top of searches. -
Hi.
Currently we have a customer with CUCM 8.6 cluster. Customer also deployed a UCCX System with Premium seats. Now customer would like to add to that UCCX system the 5 seats that comes with the CUCM 8.6 cluster. Is that doable? I know that in order to use the 5 enhanced seats that comes with new CUCM Deployments we need to buy a part number ($2995) to be able to install the software, database, etc. But now the the customer already have a UCCX deployment.. can he use the 5 enhanced seats... upgrading them to premium first and then loading the licenses on the new system.
I'm going to open a case with licensing but I want to check here first.
Thanks in advanced,
-JoseHi Haitham,
Its is compatiable with both MCS and UCS servers. I think you need to register your PAK and contact [email protected] for mentioning about your License MAC (type show status on the CLI after you install UCCX on UCS server) to get the appropriate license.
There are limitations on the maximum IVR ports in the UCCX system.
Maximum IVR ports=2*No of Agent seat counts till you reach the hardware capacity limits say for an example say MCS 7845 seres boxes support a maximum of 300 Licensed IVR ports, after you increase your Agent seat count from 150 onwards the licensed IVR ports will not increase as it has its its max capacity in this hardware.
Coming back to your question, yes in order to increase the IVR ports you need to oreder more Agent Seat counts.
Hope it helps.
Anand
Please rate helpful posts !! -
Silent monitoring calls with CUCM 6.0 and UCCX 5.0
Dear all,
We have just integrated a 3rd party recording solution in our VoIP system, which consists of CUCM 6.0 and UCCX 5.0.
Until now we used the recording and monitoring solution of the UCCX for the agents. This worked ok.
But, as we wanted to record other people's calls and also the outgoing calls of the agents, we have created for them a Recording profile in the CCM and assign it to the agents phone.
We have also mantained the recording and monitoring active for them in the UCCX.
What is happenning now is the following:
- The 3rd party recordings works perfectly.
- The recording in the UCCX gets only one way audio.
- The silent monitoring gets only one way audio.
- If the 3rd party recording server is stopped, but the recording profile is stil active in the phones, the silent monitoring in the UCCX works.
Could any of you confirm if using a 3rd party recording solution the monitoring still works?
Is there any 3rd party silent monitoring software?
Thanks in advcance.
Best regards,
AmaiaWe've got to delimit the problem.
When we use 3rd party recording solution there are 4 RTP flows: 1 incoming to monitored/recorded phone and 3 flows outgoing.
1 incoming RTP flows
- voice of remote phone
3 outgoing RTP flows
- voice of monitored/recorded phone
- voice of monitored/recorded phone sent to 3d party recording server
- voice of remote phone sent to 3rd party recording server
When we use a network analyzer we see that the agent running in the PC (UCCX 5.0) establishes two RTP flows for the monitoring session. But we can see that one of these flows sends 3 packet RTP per 1 packet RTP sent in the other RTP flow (in the same time).
We think agent sends to monitoring device all the outoing RTP packets of the phone (that belong to three different RTP flows), instead of sending only the RTP packets belonging to voice of monitored/recorded phone.
The other flow, voice of remote phone, sounds fine.
Is there a solution to avoid this problem when we monitor with UCCX 5.0 and record with a 3rd party software at the same time?
Thanks,
Christian -
CIPC 7.0.5 compatibility with CUCM 9.1
Hi
I'm trying to figure out if Cisco IP Communicator 7.0.5.0 is compatible with CUCM 9.1
All i can find is the "Cisco Unified Communications System Release Summary Matrix for IP Telephony" but this matrix is "turned upside down" from my question.
http://www.cisco.com/en/US/docs/voice_ip_comm/uc_system/unified/communications/system/versions/IPTMtrix.html
/TonyHi Tony,
It may work with CUCM 9.1 but with some limited functionality as it has not been tested with CUCM 9.1. Please check the release notes of the CIPC 7.0.x version
http://www.cisco.com/en/US/docs/voice_ip_comm/cipc/7_0/english/release/notes/CIPC70_RN.html#wp116331
Highest callmanager version listed is 8.0
Ideally you should be using version 8.6 or higher
http://www.cisco.com/en/US/docs/voice_ip_comm/cipc/8_5/english/release_notes/CIPC_Release_Notes_8_6_chapter_00.html#CIPC_RF_S79BC94C_00
It clearly lists cucm 9.1 as supported cucm version.
HTH
Manish -
Is Unity 7.0(2) compatible with CUCM 8.6 ?
Is Unity 7.0(2) compatible with CUCM 8.6 ?
The documentation reports the compatibility with CUCM 8.5(x), but is not updated for 8.6.
Thank youThe documentation related to 8.6 may lag behind for a bit. However, if you are running or update to the latest TSP (8.4.3) on your Unity server then you're likely covered. The version of TSP is really the only compatibility factor as far as Unity/CUCM is concerned.
Hailey
Please rate helpful posts! -
Hi all experts.
Is unity 7 with domino supported with cucm 8 ?
I checked the following link and it seemed confusing
http://tools.cisco.com/ITDIT/vtgsca/VTGServlet
Kindly confirmHello -
I think this is a more clear document - http://www.cisco.com/en/US/docs/voice_ip_comm/unity/compatibility/matrix/cutspmtx.html
At minimum, you may just need to update the Unity TSP (AVSkinnyTSP) to match it with CUCM - this is most often recommended to help with MWI issues.
Sincerely,
Ginger -
Recommendation on a Pharmacy IVR system that'll work with CUCM
Hello,
I'm wondering if anyone out there has any recommendation on a pharmacy IVR system that will work well with CUCM. Was it a breeze to deploy or was it full of headaches? Any suggestions and recommendations are truly appreciated.
Thank you,
Randy C.Hello Gergely,
Thanks for taking the time to respond. .
We do have UCCX but we don't use IVR on it. This definitely would be a consideration. Do you know if UCCX IVR can work with our pharmacy's system? Say for example, a customer asks for a refill request of their medication using IVR by just entering the prescription number.
Also, what are the capabilities/limitations of UCCX IVR?
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Jabber 9.6 no voicemail tab for CUC with CUCM 9 and CUC 8.6
Hi guys,
I have Jabber 9.2 and 9.6 clients with CUCM 9.1.2, CUC 8.5.1 and Cisco IM + P 9.1.1.
We've recently updated the CUCM and CUPS to Cisco IM+P. Also we have migrated from CUPC Clients to Jabber clients. The CUC server has remained as is. I was able to get the CUPC clients to get voicemail/visual voicemail but the Jabber client doesn't even display an ico for voicemail.
The service profile for all users is set correctly to point to the CUC server and there's no errors when I go to the jabber client "show connction status".
I haven't been able to find much about this issue. I have noticed that if I set the mail store the voicemail tab appears but we are not using a mail store as the voicemail repositary. Is this now required to check voicemail? If so can someone point me to the configuration guide? We have exchange 2012. Thanks very much!
-AkinHi every body,
I have some problems but not with all the users, is only working with 3 users and the rest of them don't work, I attached the errors that I get when I try to log in Telephony Accounts in the Jabber, I used the Jabber of one User, and from here I introduce my credentials (user and password) and I can get the voice mails (only from de 3 users that is woriking fine), but when I introduce the credentials of the other user I get the error and Can't get the voicemails.
The 127.23.0.7 is the IP address of the Cisco Unity Connection.
Please Help Me I'm stuck with this! -
Cisco Jabber for Mac 9.2.1: slow to register with CUCM
Hi,
We have this specific problem on our Jabber for Mac 9.2.1 client only: the client takes 60 to 70 seconds to connect to CUCM (running 8.6.2) when it starts.
After Jabber registers with CUCM, there is no issue, it's just very slow to register.
Anybody getting the same behaviour? We have no such issue on previous versions of Jabber for Mac version 8 or Jabber for Windows version 9.
Thanks,
FabriceFor those interested, I have the answer to my own question: if one of your configured DNS servers is not responding, Jabber 9.2.1 will delay registration to the phone services. This behaviour is seen only in this specifc version of Jabber, hopefully it will be fixed in the next release. Watch out if one of your DNS server goes offilne.
Fabrice -
No ringbacktone for inbound calls with cucm 8.6
Hi,
we have this problem from many days...
we have two branches with cucm cluster(Publisher and Subscriber) at Head office and cisco untiy.The branches are connected to Head office through MPLS vpn and all the ip phones are registred to publisher located at headoffice.
our setup is like below
HO and BR2 having SIP lines and BR1 has PSTN Lines.
we implement greetings for head office and 2 branches at Headoffice Unity.
when any call comes to headoffice gateway the greetings will be played and call will be diverted to the appropriate extension.everything is fine.
But the problem is when the call comes to Branch gateway and the greetings will be played and the call gets diverted to the IP phone to which the caller dialed the extension. but the caller is not hearing the ringback tone while the extension is ringing. and the caller cannot know whether the extension is ringing or the call got disconnected.
i tried to change the " Send h225 User Information Message" in service parameters from "Use ANN for Ring Back" to H225 Info for call Progress Tone"
whenever i am changing to "H225 Info for call Progress Tone" then the branches problem getting solved but Headoffice getting the same problem.
please can anyone help............................Hi Carlo,
Thankyou for the Response...
here is the Runn config for BR1 Connected to PSTN lines....
voice-card 0
dspfarm
dsp services dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
voice class h323 1
h225 timeout tcp establish 3
interface Tunnel100
description " Tunnel JED-RYD "
bandwidth 2048
ip address 10.10.0.1 255.255.255.252
tunnel source 172.31.217.202
tunnel destination 172.31.3.18
interface FastEthernet0/0
description DAMMAM Local LAN
no ip address
duplex auto
speed auto
interface FastEthernet0/0.20
description JEDDAH Local LAN
encapsulation dot1Q 20
ip address 192.168.20.5 255.255.255.0
interface FastEthernet0/0.21
description JEDDAH VOICE VLAN
encapsulation dot1Q 21
ip address 192.168.21.5 255.255.255.0
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.21.5
interface FastEthernet0/1
ip address 172.31.217.202 255.255.255.252
duplex auto
speed auto
router eigrp 200
network 10.10.0.0 0.0.0.3
network 192.168.20.0
network 192.168.21.0
no auto-summary
router bgp 65412
no synchronization
bgp log-neighbor-changes
neighbor 172.31.217.201 remote-as 65000
no auto-summary
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.20.1
ip route 192.168.20.50 255.255.255.255 192.168.20.1
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
voice-port 0/0/0
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/1
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/2
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/3
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/2/0
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
connection plar 2022
shutdown
impedance complex2
description STC
voice-port 0/2/1
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
shutdown
impedance complex2
description STC
voice-port 0/3/0
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/1
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/2
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/3
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
sccp local FastEthernet0/0.21
sccp ccm 192.168.12.190 identifier 1 priority 1 version 5.0.1
sccp ccm 192.168.12.189 identifier 2 priority 2 version 5.0.1
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register CONFJEDRAW
associate profile 2 register TRNJED
associate profile 3 register MTPJED
switchover method immediate
switchback method immediate
switchback interval 15
dspfarm profile 2 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 2
associate application SCCP
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
shutdown
dspfarm profile 3 mtp
codec g729r8
maximum sessions software 250
associate application SCCP
shutdown
dial-peer voice 1 pots
dial-peer voice 1000 voip
description To CallManager - SBWPMPUB
destination-pattern [1-5]...
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.190
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 9001 pots
description ** 02-6140294(outgoing) **
destination-pattern [^2].T
port 0/0/1
dial-peer voice 9002 pots
description ** 02-6140295(outgoing) **
destination-pattern [^2].T
port 0/0/2
dial-peer voice 9003 pots
description ** 02-6140296(outgoing) **
destination-pattern [^2].T
port 0/0/3
dial-peer voice 9004 pots
description ** 02-6140293(outgoing) **
destination-pattern [^2].T
port 0/0/0
dial-peer voice 290 pots
incoming called-number .
direct-inward-dial
dial-peer voice 9006 pots
description ** 02-6529323(local) **
destination-pattern [^0].T
port 0/3/0
dial-peer voice 9010 pots
description ** 02-6578249(local) **
destination-pattern [^0].T
port 0/3/1
dial-peer voice 9011 pots
description "to pstn service"
shutdown
destination-pattern 0.T
port 0/3/3
dial-peer voice 9009 pots
description "to pstn service"
shutdown
destination-pattern [^0].T
port 0/3/2
dial-peer voice 9005 pots
destination-pattern .T
dial-peer voice 1001 voip
description To CallManager - Subscriber
destination-pattern [1-5]...
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.189
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 1002 voip
description " TO Unity Greetings"
destination-pattern 2050
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.190
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 1003 voip
description " TO Unity Greetings"
destination-pattern 2050
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.189
dtmf-relay h245-alphanumeric
no vad
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