Phone PG & CTI RP PG

I want to split up my phones and cti routes. Currently I have one application user called pg_user that has both the phone devices and the cti route points being controlled by it.
I accidently deleted pg_user a while back while adding some devices to. I mean really. What developer thought it would be a good idea to put the delete button next to the save button... really....
So to protect my self from myself I want to split these two users up now.
I remember reading about doing this a long while back when I upgraded our ICM IPCC software from 7.0 to 7.5. For the life of me I can not find where that is. I have gone back through the admin guides but it just not jumping out at me.
I know I have to rerun ICMSETUP.exe and probably add another Call Manager PIM. I really dont want to just wing it though and I dont have a test bed.
Can anyone point me to the DOC or some instructions?
Thanks in advance for any direction!
Del

I found it, I found it!!!!! yea!!!!
In the cc75srnd.pdf on page 142. This is chapter 3 page 26.
Improving Failover Recovery for Customers with Large Numbers of CTI Route Points
When a Unified CCE PG fails-over, the PIM connection that was previously controlling the Unified CM
cluster is disconnected from its CTI Manager, and the duplex or redundant side of the PG will attempt
to connect it's PIM to the cluster using a different CTI Manager and Subscriber. This process requires
the new PIM connection to register for all of the devices (phones, CTI Route Points, CTI Ports, and so
forth) that are controlled by Unified CCE on the cluster. When the PIM makes these registration
requests, all of them must be confirmed by the Unified CM before the PIM can go into an active state
and process calls.
To help recover more quickly, the Unified CCE PG can have a PIM created that is dedicated to the CTI
Route Points for the customer, thus allowing this PIM to register for these devices at a rate of
approximately five per second and allowing the PIM to activate and respond to calls hitting these CTI
Route points faster than if the PIM had to wait for all of the route points, then all the agent phones, and
all the CTI ports. This dedicated CTI Route Point PIM could become active several minutes sooner and
be able to respond to new inbound calls, directing them to queuing or treatment resources while waiting
for the Agent PIM with the phones and CTI Ports to complete the registration process and become active.
This does not provide any additional scaling or other benefits for the design; the only purpose is to allow
Unified CM to have the calls on the CTI Route Points serviced faster by this dedicated PIM. It should
be used only with customers who have more than 250 Route Points because anything less does not
provide a reasonable improvement in recovery time. Additionally, only the CTI Route Points that would
be serviced by Unified CCE should be associated with this PIM, and it should have its own dedicated
CTI-Enabled JTAPI or PGUser specific to the CTI Route Point PIM.

Similar Messages

Responsible Agent after transferring phone call CTI

CRM 5.0 SP11
Using CTI, when a phone call is transferred between 2 agents we would like the responsible agent to automatically become the person that took over the interaction without them manually having to change the responsible agent partner function or forward the interaction in the Inbox.
Have been looking at the following classes but not seeing where the code it getting touched in the actual transfer between agents.
CL_CRM_AUI_ONEORDER
CL_CRM_CUCOPROCESS_IMPL
Anyone with similiar requirments.

t-code CRMC_IC_RTFRAMEWORK put a check in Disp.Mode field.

No Presence Status With 9971 Phones

CUCM - 9.1.2.10000-28
CUCM IM and Presence - 9.1.1.31900-1
9971 Load - sip9971.9-3-2-10
Have an issue with the 9971s not updating presence information on Jabber (windows and osx).
All phones are able to be controled from Jabber, ie. Dial, answer etc.
I have made sure of the following:
User is added to phone and line.
Phone is associated to with users profile.
Proper CTI permissions applied including allow control of phones supporting conferencing, CTI enabled etc.
Primary DN is selected.
All phones have CTI enabled.
Presence status is updated just fine on all other devices. If I move myself from the 9971 and control a 7965 presence status begins to function. However once moved back to 9971 presence status stops updating and Available never changes to On Call when I go off hook.
Again I can control the phone without any issues from Jabber.
This only happens to my users that are assigned 9971 phones.
Any ideas or places to look that I might have missed?

Hi,
I copy here from the docs
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/9_1_1/ccmfeat/CUCM_BK_C3E0EFA0_00_cucm-features-services-guide-91_chapter_0110010.html#CUCM_TP_SFF7026D_00
Can you check it against your scenario?
Presence Information
The presence status "on a call" is displayed for a user in the following situations:
Outgoing Calls
The user initiates a call from Cisco Jabber for Windows in Extend and Connect mode.
The user initiates a call from a device that is configured as a remote destination which is routed using Unified CM or Unified CM with Unified CM SME.
Incoming Calls
The user answers a call on a device configured as a remote destination routed using Unified CM or Unified CM with Unified CM SME.
The user answers a call routed to a device configured as a remote destination using Cisco Single Number Reach.
The presence status "on a call" is not displayed for a user in the following situations:
When the user initiates a call from a device that is configured as a remote destination; the call is not routed using Unified CM or Unified CM with Unified CM SME.
When the user answers a call on a device that is configured as a remote destination; the call is not routed using Unified CM or Unified CM with Unified CM SME.
HTH,
Christos

Automute microphone, when Jabber is controlling the Deskphone via CTI ?

Does anybody know a way on how to mute the Microphone of the Deskphone when controlling the Phone via CTI. I want to have a true human interaction with the hardphone when controlling it via CTI. Otherwise the Deskphone could be easily used as a remote Microphone.
Thanks a lot in advance
Björn

Hi Mohammed,
Thank you for your answer- I hoped for a more native way than implementing it on my own via JTAPI. I cannot imagine that i am the only one feeling uncomfortable in letting all people remotecontrol their Phones/microphones in this CTI Way. If i am sitting in a neighbour Office- i could still call my own and remotely listen to the remote office. There should be a simple way to prevent users from doing this.
Does anyone feel the same- or has a hint for me on what to do ?
thanks again.
Björn

CUPC Over VPN

We resolved a VPN issue that was preventing us to be able to log in to CUPC over VPN. I am now able to log in, I can see my buddy list and their status, however the CUPC status in the bottom of the window is listed as "Offline (No Network)".
Server Health:
Logon Server: Not Connected - Disconnected
Phone Config: Downloaded
Presence: Connected
Desk Phone: Not Connected
Softphone: Not Active
Voicemail: Connected
Secure Messaging: Not Connected - Server Unreachable
LDAP: Not Available - Server Unreachable
What could be causing some of the servers to be connected while others are disconnected? We are running Microsoft ISA VPN.

This is likely an ISA VPN configuration issue. CUPC creates separate connections to each system. For example, voicemail is an IMAP or secure IMAP connection, presence is a SIP connection, desk Phone is CTI, etc. All traffic is not tunneled through CUPS.
You will need to troubleshoot the individual protocols to understand why Microsoft's VPN product is not properly transproting them. A good place to start would be attempting telnet connections from the VPN-connected machine to the locations specified in the relavent profile on CUPS. Example: Can you telnet to your LDAP server's port as defined in CUPS?

Cisco Jabber for Windows Client SDK

Is there an SDK for Cisco Jabber for Windows client? I could only find the Web SDK. Are there no other SDKs that would let me control features of the Jabber Windows Client?

And in order to avoid TAPI limitation, they say they have started using Jabber which controls all phone functionalities through an Application User connected to the Call Manager.
Assuming they are referring to the normal Jabber for Windows/Mac/iOS/Android applications then this is wrong. Jabber - the client, not the IM&P servers which are part of the CUCM cluster (if you're familiar with Lync consider CUCM+IM&P servers a Front End Pool) - do not use an Application User at all. When a user starts Jabber, assuming SAML SSO is off, they supply an email address. This is used to find the servers via DNS SRV records, or the Mobile and Remote Access Expressway (roughly analogous to a Lync Edge pool) to tunnel through the firewall. After the email address they are prompted for their End User username and password. This is then used to access several APIs on the server, namely the User Data Service to discover what the user has. This would include what the user has configured/available to them. After this is done the Jabber client does one of three things with respect to phone functionality:
If the user has no phone devices provisioned for them - physical or software - it becomes an IM&P client only.
If the user has a software phone phone provisioned, it registers as that using SIP. In this way Jabber is the actual phone. This is called a CSF device for Jabber Windows/OS X.
If the user has a desk phone assigned to their End User account AND administrative policy is to default to desk phone control, it registers to CTI Manager on the CUCM server to control the user's physical phone.
A user can toggle between option two and three, if appropriately provisioned, but cannot have both simultaneously. In other words, Jabber is either controlling a physical phone over CTI or is a phone itself using SIP. The difference is where the audio/video media is sent from/to.
All of this happens using that human's End User credentials.
Now, if you wanted to have a server control the user's phone - either a physical one or Jabber - you would use the TSP as an interface to CTI Manager on the CUCM server and receive real-time events (e.g. the user went off-hook, the phone is ringing with this caller ID, etc.) and issue commands to that phone (e.g. make a call to X). This is done using an Application User account because the server would need to do this for multiple users simultaneously. If you also use the Super Provider concept, the CUCM administrator doesn't have to take the extra step of associating every phone to your Application User manually. Instead, you can simply issue CTI subscriptions to see any/all device activity you want.
This is all assuming you wanted to do the integration server-side to avoid having to update the Cisco TSP application on every user PC (which also limits your WPF application to machines capable of running the TSP) every time the CUCM administrator patches the servers. If you don't care about that, then you can do this all client-side on the user's PC, and have them supply their End User credentials.
Of course, you could also reconsider the Jabber SDK and just embed the make/receive call functionality directly into your application instead.

Transfer to IVR in Outbound Campaign does not work

Hi
I am configuring an outbound predictive campaign and I need when no agents are available the customer should be transfer to my IVR, but when the call is transferred it is dropped.
When I check the debugs of the gateway I found "Busy User" disconnected cause 17.
I hope you can help me with this issue.
UCCE 9.0(2)
CUCM 9.1
GW 2921
My script to Transfer to IVR is like this:
My campaign configuration is like this:
Regards.

Ok, have you configured your Dialer to Handle TR Label which comes as a response through MR PG, for Translation routing request?
look at below. how your dialer is configured to handle label? what is flow ?
Modification of Local Static Route File
The SIP Dialer installation process installs an empty template file named DNPHost in the \icm\customerInstanceName\Dialer directory. This file defines the static route mappings of a dialed number wildcard pattern to the IP address or host name with which an agent phone or CTI Route point is registered. For each static route you wish to define, enter a row in this file in the following format:
wildcard pattern, IP address or host name, description
Examples:
7?????, 10.86.227.144, transferring outbound calls to agent extensions
86! , 10.86.227.186, for CTI Route Points on CUCM node1
4?????, gambino.cisco.com, transfer to IVR campaign
For single gateway deployments, the SIP Dialer reads the static routing info from DNPHost and uses the information from the file to construct the SIP REFER message. If the SIP Dialer is setup to support a voice gateway, the SIP Dialer loads DNPHost, ensures that it has valid routing entries, and sends an alarm if the file does not exist or is invalid.
See the installed DNPHost file template for additional information about its use.
regards
Chintan

Sony Xperia J and headphones with controls

Hi All
I've just got a Sony Xperia J and am having little joy with the headphones. I've done searching and seen lots of posts about it, but I'm still not 100% sure on the situation unfortently.
The phone is CTIA. My headphones are CTIA (Altec Lansing Muzx® Ultra - MZX606). I've also tried another pair of CTIA (Skull Candy) headphones with the same result. Both work on iPhones fine, with full button functionality.
Just plug them in - nothing.
Hold down the centre button as you plug them in - Phone acknowledges them being plugged in and sound comes through them.
The next/back/play/answer etc buttons on them have no effect at all.
Why? (Feel free to get technical!). Does the Xperia J even support these buttons? I've looked at the Smart Headset and am concerned if I bought one the buttons still wouldn't work. Plus, I really don't want to buy more headphones as I already have a nice set of headphones with all the buttons I need that I've been using with my previous phone
I really want the buttons working so I can tuck it away safely on my daily commute but easily turn the volume up and down etc, as I have been doing for years on my previous phone.
Thank you!

I believe the problem might be related to voltage being used. Third party headsets that comes with music controls tends to use other voltage values compared to headsets just having a call key.
Xperia J provides 2V and if the headset requires more you might experience difficulties like this.
This might help you finding a third party headset that is fully compatible with your phone.
- Community Manager Sony Xperia Support Forum
If you're new to our forums make sure that you have read our Discussion guidelines.
If you want to get in touch with the local support team for your country please visit our contact page.

Installing sametime plugin for gaim ( Lotus Sametime users)

Hi,
I just installed sametime plugin in gaim and want to share it with all using Sametime protocol for chatting.
#sudo -s
# abs
#mkdir /var/abs/local/meanwhile
#cd /var/abs/local/meanwhile
# vi PKGBUILD
pkgname=meanwhile
pkgver=0.4.2
pkgrel=1
pkgdesc="Meanwhile Libraries"
url=""
license=""
depends=('glib2')
makedepends=()
conflicts=()
replaces=()
backup=()
install=
source=(http://keihanna.dl.sourceforge.net/sourceforge/meanwhile/meanwhile-0.4.2.tar.gz)
md5sums=()
build() {
cd $startdir/src/$pkgname-$pkgver
./configure --prefix=/usr
make || return 1
make DESTDIR=$startdir/pkg install
#makepkg
#pacman -A meanwhile-0.4.2-1.pkg.tar.gz
#mkdir /var/abs/local/gaim-meanwhile
#cd /var/abs/local/gaim-meanwhile
#vi PKGBUILD
pkgname=gaim-meanwhile
pkgver=1.2.4
pkgrel=1
pkgdesc="Gaim Meanwhile Libraries"
url=""
license=""
depends=('gaim' 'meanwhile')
makedepends=()
conflicts=()
replaces=()
backup=()
install=
source=(http://keihanna.dl.sourceforge.net/sourceforge/meanwhile/gaim-meanwhile-1.2.4.tar.gz)
md5sums=()
build() {
cd $startdir/src/$pkgname-$pkgver
./configure --prefix=/usr
make || return 1
make DESTDIR=$startdir/pkg install
#makepkg
#pacman -A gaim-meanwhile-1.2.4-1.pkg.tar.gz
Both installs like a charm.
Now start gaim as a normal user and add your sametime details.
Now isn't making packages on Arch dammmm easy.
I already love arch
Enjoy sametime connectivity.
Regards,
Abhay
PS: you may use any server to get the source tar.gz files, as per your convenience.

Markus,
I've honestly never experience this problem, but I'll assume from the image that the following is currently happening:
1) The end user does not have a soft phone.
2) The desk phone control (CTI) connection is failing to connect or find a desk phone.
I can't say right off the bat why its occuring, but should be fairly simple to find out with a problem report. Can you take a screen shot of the connection status menus?
Regards,
Jason

Script,Place call

Hi,when i use the place the call step in the script editor,and the collected string matches an internal number,the place call step rings the destination but as soon as i answer the call,the call does not get really connected and the reactive debug shows that the call goes to the end and i hear an auto generated error message.
lets say phone 2001 call to uccx and then he dialed the extension 2002,the place call step sends the call to 2002,but i see on the phone screens From CTI port 1700 and on the other phone to CTI port 1701,i don't see from 2001 to 2002.
If i use the call redirect step,then i see from 2001 to 2002 and the call works normally.I just can't use the call redirect cause there is no (no answer option) under it .So if the called number didn't answer within 10 sec ,with the call redirect i can't take an action but with the place the call step,i can.
Help will be so appreciated.
Thanks in advance.

As always excpected,you are very helpful.
Thanks man.call consult transfer works,but i have two questions:
1-why did the call fail when i use the place call step ?
2- In the consult transfer step,what are the "output digits" field for?
i used a string variable just to fill the fields,but i didn't see any digits in it when i used the debug !
Thanks in advance.

Lync 2013 to CUPS RCC - Record-Route signature is invalid

Hi all,
             I have a Lync 2013 environment and I'm trying to implement RCC with a CUPS 10 environment.
On the Lync side, I have run the following commands:
$tcpRoute = New-CsStaticRoute -TCPRoute -Destination 10.10.120.54 -Port 5060 -MatchUri cups.domain.com -ReplaceHostInRequestUri $true
Set-CsStaticRoutingConfiguration -Route @{Add=$tcpRoute}
CsTrustedApplicationpool -Identity 10.10.120.54 -Registrar LyncFE.domain.com–Site 1 –TreatAsAuthenticated $True –ThrottleAsServer $True –RequiresReplication $False
New-CsTrustedApplication -ApplicationID RCCGateway-1 –TrustedApplicationPoolFqdn 10.10.120.54 -Port 5060 -EnableTcp
Set-CsRegistrar registrar:LyncFE.domain.com -SipServerTcpPort 5060
on the test user I have set
Set “Telephony” to:
Remote call control
Set Line uri to:
tel:<ext>;phone-context=dialstring
Set Line server uri to:
sip:<username>@cups.domain.com
In the SIP logs I get the following (I have bolded the bits I think are important)
TL_INFO(TF_PROTOCOL) [0]1FDC.1670::02/19/2015-05:35:53.687.001b274e (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[1932678887] $$begin_record
Trace-Correlation-Id: 1932678887
Instance-Id: 45F
Direction: incoming
Peer: 10.10.120.54:5060
Message-Type: response
Start-Line: SIP/2.0 200 OK
From: "Test ADL"<sip:[email protected]>;tag=224a04a9f9;epid=fa015fbbdb
To: <sip:[email protected]>;tag=29e9c078-ce4a5e7
Call-ID: 669e8717f0a0487fb09aed392d84f589
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060;transport=TCP>
Via: SIP/2.0/TCP 10.70.11.70:52575;received=10.70.11.70;branch=z9hG4bKE005ADB2.742BA66DD49B1869;branched=FALSE;ms-internal-info="aaweLh0WVLdGLzdpabeIHETcyFKD5xJlVJz_tnBJ0l-7ptpit0RW23IQAA", SIP/2.0/TLS 10.70.20.36:52402;ms-received-port=52402;ms-received-cid=1A00
Record-Route: <sip:[email protected]:5060;maddr=10.10.120.54;transport=tcp;lr>,<sip:LyncFE.domain.com;transport=tcp;opaque=state:T;lr;ms-route-sig=aaF2AA9BciRLZcYZPN0XasMmn8g4LiFd4j7Fdcxg6R6uRtpit0mCXPOAAA>;tag=7E7648A7380D372597F42EAEAECC07A2
Content-Length: 190
Content-Type: application/csta+xml
Message-Body: <?xml version="1.0"?>?<RequestSystemStatusResponse xmlns="??<systemStatus>normal</systemStatus>?</RequestSystemStatusResponse">http://www.ecma-international.org/standards/ecma-323/csta/ed3">??<systemStatus>normal</systemStatus>?</RequestSystemStatusResponse>?
$$end_record
TL_ERROR(TF_SECURITY) [0]1FDC.1670::02/19/2015-05:35:53.687.001b2b66 (SIPStack,SIPAdminLog::WriteSecurityEvent:SIPAdminLog.cpp(319))[1932678887] $$begin_record
Text: Record-Route signature is invalid
Result-Code: 0xc3e93f1e SIPPROXY_E_INVALID_RECORD_ROUTE
Connection-ID: 0x1300
Peer-IP: 10.10.120.54:5060
Peer: 10.10.120.54:5060
SIP-Start-Line: SIP/2.0 200 OK
SIP-Call-ID: 669e8717f0a0487fb09aed392d84f589
SIP-CSeq: 1 INVITE
$$end_record
TL_WARN(TF_DIAG) [0]1FDC.1670::02/19/2015-05:35:53.687.001b2b74 (SIPStack,SIPAdminLog::WriteDiagnosticEvent:SIPAdminLog.cpp(805))[1932678887] $$begin_record
Severity: warning
Text: Routing error occurred; check Result-Code field for more information
Result-Code: 0xc3e93f1e SIPPROXY_E_INVALID_RECORD_ROUTE
SIP-Start-Line: SIP/2.0 200 OK
SIP-Call-ID: 669e8717f0a0487fb09aed392d84f589
SIP-CSeq: 1 INVITE
Peer: 10.10.120.54:5060
$$end_record
I have not had any luck finding the solution to this as yet - and any suggestions as to what I have done wrong would be appreciated.
Edit - apparently bold doesn't work.... sorry - it works in the editor, but not when I save the post! Please look at the ":record-route" line in the first log and the result code in the next two entries.

Hi Ben, thank you for your update.
To clarify, the
device=SEP0123456789F is an example, you have to change it with the correct Cisco IP-Phone serial number used by your user.
I've checked every step of your procedure and I've not found any issue, I've used the same procedure with a Cisco CUPS 10 and everything works fine.
Please be sure to check this on Cisco:
- the MatchURI need to be equal to the Cisco Unified Presence Domain
- on the CUCM the CTI Account (default is CtiGw) nned to have CTI controll of every phone (Standard CTI Allow Control of Phone supporting Connected Xfer and conf) and Supporting Rollover Mode
- it's important that the "Standard CTI Secure Connection" group is NOT associated to the CtiGw account (if it's associated it force the connection to TLS).
Let me know.
Regards
Luca
Luca Vitali | MCITP Lync/Exchange | snom Certified Engineer | Sonus SBC1000 Engineer

CallManager Attendant Console over WAN connection

Does anyone know if it is possible to run Attendant Console over the WAN. I have a branch office that I would like to install AC, but I cannot seem to get the Call Control layer up when I start the AC gui.
Any suggestions?

Hi,
The use of the Attendant Console outside of the central site can require a considerable amount of bandwidth if the remote sites must access large user account directories without caching them.
-Apart from IP Phone requirements , AC needs Jtapi messages communications
-Line State Updates..It depends on number of phones which change line state at a given time ..TCD server sends this as UDP packets..And how many of these are listed on the console directory/speed dial at a given
time..
Say we have 10 phones getting a new call at a time and for of these are listed on Console , we will get line state for 4 phones.
-At each login , the console connects to server and checks if userlist file has changed and if changed it downloads the same..Workaround is to have local copy of this file
But this is usually a one time step per login session.
Regarding your question, it should work, verify that no ACLs, FW or other network device is blocking traffic from|to CCM and AC console|IP Phone.
You may check this link to verify configuration is fine.
AC user password (12345)
PP|IP Phones associations
CTI|TCD started
etc
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/4_0/sys_ad/4_0_1/ccmcfg/b04ccmac.htm

Jabber Windows - no phone control with LDAP Custom filter

I am unable to control the desktop phone from the Jabber 9.1 Windows client when the CallManager LDAP Directory uses a Custom Filter.
Has anyone else experienced this?
If I set the LDAP Custom Filter to <none> and save, then Desktop Phone control works great.
If I set it to use my custom filter, then trying to enable Desktop control just gives me the spinning circle, then times out to the Red X symbol.
I do not need to resync the LDAP Directory to get the error, just enable/disable the custom filter and save.
In both cases calling from the Computer works great.
This is an On-Prem deployment with full MS-AD LDAP integration.
Versions are:
Jabber - 9.1.0 build 12296
CUPC - 8.6.4.11900-1
CUCM - 8.6.2.22900-9
I upgraded to CUCM 8.6.2 SU2 last night hoping that would fix the problem, but no luck.
The LDAP filter is one I have used in numerous other clusters with no CTI issues.
It allows me to sync to the root directory, but only import active user accounts with an entry in the ipPhone AD attribute:
(&((objectclass=user)(ipPhone=*))(!(objectclass=Computer))(!(UserAccountControl:1.2.840.113556.1.4.803:=2)))
Thanks, Randy

Hi Randy,
Have you specified this base filter in jabber-config.xml file? As per Admin Guide:
"In some cases, base filters do not return query results if you specify a closing bracket in your Cisco Jabber for Windows configuration file. For example, this issue might occur if you specify the following base filter: (&(memberOf=CN=UCFilterGroup,OU=DN))
To resolve this issue, remove the closing bracket; for example, (&(memberOf=CN=UCFilterGroup,OU=DN)"
Thanks,
Maqsood

Jabber - cannot login in Phone Accounts

Hello All,
I am currently configuring CUPS 8.6.4 to integrate with CUCM 8.6.1 and provide IM and Calls to Jabber Client (version 9.0.5). Note that I do not sync to any active directory,ldap,etc, instead the users are retrieved from CUCM. I am already able to sign into Jabber Client and have chats with other people, however I am unable to make calls. I went into the options and found that I cannot log into the Phone Account (See Attachment).
In CUCM, I have given each user the capability for CUP and CUPC (System > Licensing > Capability Assignment) and all users belongs in Standard CCM End User, Standard CTI Enabled and Standard CTI Allow Control of Phones supporting Rollover Mode. All the users have Cisco 6921 Phone and they are associated with their respective devices. I have configured SIP Trunk and activated the required services.
Please help me in this issue
Thank You

Look here:
http://www.cisco.com/en/US/docs/voice_ip_comm/cups/8_0/english/install_upgrade/deployment/guide/dgclient.html#wp1091492
These are the steps:
•Creating a Softphone Device for Each Cisco Unified Personal Communicator User
•Associating Users with Softphone Devices
•Adding Users to User Groups
•Associating a New Device with a User
•Resetting a Device
Rate this if helps!
Regards
Leonardo Santana

Jabber Cannot Control Desk Phone

I know, I know. A well-worn topic. But it's just not happening for me, and I'm new to Presence 9.1 This is a new installation. Also using CUCM 9.1 and Jabber 9.2.6. I don't even know where the Jabber error logs are.
Anyway, when I try to get Jabber to control the desk phone, it doesn't work and when I look at the connection status, I get an error stating that there is a connection error to the CTI server. That server is my CUCM Subscriber. I am LDAP integrated, but we are pointing the a global catalog. The CCMCIP service is running, the profile is created and the user is associated with it. On the CUCM user, the phone is associated to the user, the line is set up as the primary extension, and the user is in the Standard CTI and Standard CCMUSER groups. The user is configured to use IM and Presence, and the profile is selected. On the phone, the user is set as the owner and the line appearance is set to the user.
The two things I've seen that might be issues are that we are using the following LDAP filter:
(&(objectclass=user)(!(objectclass=Computer))(!(UserAccountControl:1.2.840.113556.1.4.803:=2))(|(ipphone=*)(samAccountName=adm-*)))
I didn't create this and I don't know enough about filters yet to know if this is good.
Secondly, in my past experience, the configurations for the CTI service (i.e. "cti_1.1.1.1_cti_tcp_host_synced_000" - not the real IP) have always been created automatically, but they had to be created manally for this install. I don't know if that's a function of 9.1 or if there's an integration issue.

Hey Refran,
I had the same issue when doing a LAB installation.
My problem was, that I sometimes used DNS names and sometimes IP Addresses.
After I changed everything to IP Adresses, Jabber was able to connect to the CTI Server.
How did you configure your Servers of CUCM (System--> Server) and the Integration of CUPS and CUCM?
Here a log of Jabber:
REGISTER sip:172.20.10.56 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.56:5060;branch=z9hG4bK000001b4
From: <sip:+41417xxxxx@>;tag=001d72c593ae000200004084-00002998
To: <sip:+41417xxxxx@> -------------------------------------------------------------------------------- here you should see theIP, but in my case this was missing
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 27 Jun 2013 21:33:58 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CSF/9.3.1
My recommendation for your is to test again, after changing everything to IP Adresses.
Otherwise, you can also change the hostfile of the client
Best regards
Ben

Phone PG & CTI RP PG

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