CCM 9.X New Features

Pause in Speed Dial
Users can configure speed dials with FAC, CMC and post connect DTMF
Comma accepted in speed dial as delimiter and pause

Feature allows two methods of configuration:
-Method 1: Using comma as a pause and also as a delimiter
-Method 2: Dialstring/FAC/CMC/Post connect digits with no commas
Method 1: Command Delimiter for Pause
-Comma used to delineate dial string, FAC, CMC, and post connect digits
-For post connect digits, commas insert a 2 second delay
-Commas may be duplicated to create longer delays
-Preferred method for non-CUPC devices
Method 2: No Comma
-All digits to be used for dial string, FAC, CMC and post call digits entered as one string
-Once a digit string has been matched, CUCM moves on to next digit string
-Can be used on SCCP and SIP phones, but required for CUPC
Pause in Speed Dial Examples
-Will dial 914085551212, after connect, wait 8 seconds to dial 123456
-FAC for International Calls. Will dial 90114455612323# with FAC of 2244
-Will dial 91408551212, with a FAC of 6534 and CMC of 5656, wait 6 seconds, the dial the DTMF digits 9933
-Will dial 914085551212 with a FAC of 6534 and CMC of 5656, then immediately after connect, dial 9933
New Service Parameter allows configuration of interdigit delay
If the speed dial FAC or CMC is wrong
-Method 1: Call disconnects and an error is displayed
-Method 2: phone displays an error and allows user to manually enter information
Pause in Speed Dial Caveats
-Dial string is truncated in the calls history list (only dialed number)
-Feature may not work with CUPC client and variable length/overlapping dialplans (no comma delineation)
-This feature is not supported SRST

Codec Preference
Pre CUCM 9.0
-Administrator could only eliminate codecs (based on Maximum Audio Bit Rate)
-Could not prioritize G.711alaw over G.711ulaw, or G.729 codecs
With CUCM 9.0
-System default codec preference same as earlier versions
-Allow administrator to deterministically specify codec order
-Allow codec selection based on received offer
-Custom Codec list applied globally or on a GW/Trunk Level
-Can be applied to: SIP, MGCP, SCCP, H323 and EMCC

Codecs preference still choose by Regions

For SIP Devices/Trunk, can specify “Accept Codec Preference in received Offer”
Can change codec selection for EMCC logged in devices
Codec Preference Caveats
A common Codec Preference List must be the same on all clusters when using the following features:
-Extension Mobility Cross Cluster
-H323 Inter Cluster Trunks

Biggest challenge will be unexpected codec

-Check “Accept Audio Codec Preferences in Received Offer” settings
-Check at Device level and system level
When using non-pass through MTP, codec negotiated hop-by-hop

Native Call Queuing

Enables Hunt Pilot to queue callers

-Allow for redirection of calls based on different queue criteria
-Allow agents to participate in multiple queues

  • Auto logout and call re-queue if agent does not answer
  • Longest waiting call in all queues will be delivered first
  • No ‘post call’ time or agent greeting options
  • On phone ‘Queue Status’ display

Cisco Extend and Connect

What is the existing limitation?

  • Using CTI (webex connect or CUCILync), user can monitor a calls, but not control the call
  • No enterprise features for non-CUCM registered devices
  • Cannot hold/resume, transfer, conference or park
  • Remote devices ring and can be answered, but not mid-call features

What is Cisco Extend and Connect?

  • A new device type, CTI Remote Device that represents all remote destinations for a user
  • Anchors enterprise calls on the CTI Remote Device
  • Allows a CTI application (like Jabber) 3rd party control of the remote connection to enable enterprise call features

Examples of a deployment scenario
Contact Center agent working from home

  • Low bandwidth at house, VOIP not an option (hard phone or soft client) and cell phone is not an option
  • Extend connect sends call to home phone and CAD agent allows enterprise features needed for contact center agents

Use Cisco Unified Communications with legacy PBX

  • Customer has PBX under contract and not ready to move phones
  • Customer wants UC for IM, Chat and messaging, but phones on PBX
  • Extend Connect enables Jabber deployment for UC, but enterprise control of PBX phone (as remote device for Jabber)

New End User Webpages

CUCM 9.0 now has two types of end-user’s webpages

  • One type of page is for core Users with one phone and one line
  • The other page will be for users with multiple phones with one or more lines on each device

New User Page UI targeted towards core users
Cisco Mobility Updates
Simultaneous Ring in previous versions of CUCM

  • CUCM 7.0 introduced the parameter “Reroute Remote Destination Calls to Enterprise Number”
  • Calls direct to cell would ignore time of day settings and call the cell
  • Calls would be anchor on the enterprise phone….but the line would not ring

New features in CUCM 9.0:

  • Added “Ring All Shared Lines” service parameter
  • Uses Boolean Setting
  • True – all lines (including other remote destinations) ring
  • False – only the dialed number (remote destination) rings
  • Default and existing behavior is False

Single Number Reach Voicemail
The Problem:

  • When a call is extended to a SNR destination, CUCM cannot determine if the call was answered by the user or VM
  • Based on “Answer Too Soon”
  • Time based mechanism is unreliable and requires tweaking for each service provider

New Solution

  • CUCM 9.0 introduces a new parameter called “Single Number Reach Voicemail Policy”
  • Can be either Timer Controlled or User Controlled
  • Timer Controlled uses existing “Answer Too Soon” timer
  • User Controlled requires the user to send a signal (DTMF) to accept the call

Hunt Pilot Connected Number Display
Hunt pilot DN display in previous versions
-Calls to a hunt pilot display the DN of the hunt pilot as the connected party ID

  • Applies to both MGCP and SIP trunks

Hunt pilot DN display in CUCM 9.0

  • This feature allows the connection to be updated with the answering party’s DN as the Called Party ID
  • Applied on the Hunt Pilot Configuration page
  • SIP: PAI and Remote PartyID are updated
  • MGCP/H323: Connected Number sent to update the Called Party ID

RTCP Support

  • RTCP provides out-of-band statistics and control info for RTP
  • RTP sent on even port and RTCP is send over next higher odd port
  • RTCP is supported between phones directly

RTCP not supported by:

  • Trusted Relay Point (TRP)
  • RSVP Agent
  • DTMF Translator
  • Passthru MTP

CUCM 9.0 RTCP New features:

  • CUCM 9.0 supports RTCP through MTP in pass thru mode
  • In non-pass thru mode, RTCP will still be blocked
  • Only valid for SIP to SIP calls

BRI G.Clear

  • CUCM v7.0 (1) first introduced G.Clear support for MGCP PRI
  • G.Clear required for tandem ISDN bearer circuits in VOIP network

New features:

  • CUCM 9.0 expands support for G.Clear to BRI interfaces
  • Supported on MGCP BRI interface
  • Supports G.Clear over SIP trunk with Early Offer and G.Clear

Security and OS Updates

  • Red Hat Enterprise Linux 5.0 v7.0.2
  • Host rename/reIP simplified (3 less steps to complete)

Optimized CLI commands:

  • Utils dbreplication stop/dropadmindb/reset
  • Utils dbreplication forcedatasyncsub
  • Utils dbreplication status replicate
  • Utils dbreplication runtimestate

Upgrade paths

  • L2 upgrade from 8.6(1) and later to 9.0(1)
  • Refresh Upgrade for 8.x (prior to 8.5), 7.1(5) and 6.1(5)

Security Feature Update
CTL Client Update

  • Single installer for all Windows versions
  • Supports Windows 7 (32 and 64 bit), Windows XP and Windows Vista

Updates to AXIS 2.0 (support .NET clients)

Assured Services for SIP Line side devices

  • MLPP support for 99xx/89xx SIP phones and 3rd party SIP Phone
  • TLS connections for 3rd party SIP phones

LDAP Enhancements
Custom User Fields

  • Existing LDAP agreements sync 13 default attributes
  • LDAP agreements will allow 5 Custom User fields
  • Custom User Fields are common across all sync agreements
  • Custom User Fields updated on 1 agreement are synched across all agreements
  • Attribute will be validated at save time
  • Error message thrown when saving and the attribute does not exist

LDAP and Manual User Support
Prior to CUCM 9.0

  • Enabling LDAP sync would prohibit adding local users
  • End user to be used by CUCM must be defined on AD and synched
  • Extra users could trigger extra CAL’s on the MS AD

With CUCM 9.0

  • Administrator can have both LDAP sync users and locally defined users
  • Ability to modify local users and roles assigned to LDAP users
  • Deleting LDAP synch will mark users synced for deletion (garbage collection)
  • Administrator can convert an LDAP user to a local user
  • User status field is used to differentiate between the Local user and LDAP Synchronized users

To convert LDAP synchronized user to the local user account:

  • Check the box Convert User Account and Save changes
  • After a user is converted to local CUCM user all the fields become editable

CUCM IM and Presence

Beginning with release 9.0, CUCM and CUP will start integration to be one product

  • Includes common release and upgrade process
  • Centralize administration
  • Simplify licensing, now included as part of CUCM user licensing
  • Deprecating IP Phone Messenger (IPPM) and CUPC 7.0

Through CUCM IM and Presence administration screens, configure UC Services for clients

UC Services that can be defined:

  • Voice Mail
  • Visual Voice Mail
  • Conferencing
  • Directory
  • IM
  • Presence
  • CTI

UC Services are used to build a UC Service Profile

UC Service Profiles assigned to users:

  • Licensing for the feature handled at the user level
  • Home cluster specified in the user page

When migrating to CUCM 9.0, existing service profiles and configuration in CUP will be migrated

  • CUCM IM and Presence uses Templates and Layouts to speed up user creation
  • BAT/AXL have been updated for CUCM/CUCM IM and Presence



Cisco Presence Integration with CCM 7.x,8.x

Cisco Presence Integration with CCM

Summary steps of integrating Cisco call manager with Cisco presence

Step#1: Enable presence globally on Cisco Call manager

By default presence subscription is disabling on CCM.

System>Service parameter>Cisco Call Manager>

Search for “Inter-presence” key word and set “Allow Subscription”

Step#2: Create SIP trunk Security Profile in CCM

Special setting is required for SIP trunk which runs from CCM to Presence.

Copy “non Secure SIP Trunk Profile” to “Presence non-secure SIP trunk Profile”

Modify below parameters:

  1. Device security mode: Non-Secure
  2. Incoming Transport type: TCP+UDP
  3. Outgoing Transport Type: TCP
  4. IncomingPort 5060 (untick Enable digest authentication)
  5. Enable application Level Authentication UNTICK
  6. Accept Presence Subscription TICK
  7. Accept Out-of-Dialogue REFER TICK
  8. Accept Unsoliciliated Notification TICK
  9. Accept Replace header TICK
  10. Transforms security status UNTICK
  11. Save

Step#3: Add a SIP trunk now from CCM to Presence


Protocol = SIP

Fill below:

  1. Device Name : PRESENCE-TRUNK
  2. Description : blah
  3. Device Pool : DP_HQ
  4. Common Dev conf : None
  5. call classification : On-Net
  6. Media resource Grp : MRG_HQ
  7. Location : HQ_LOC
  8. AAR GROUP : HQ_AARG (if not using AAR leave empty)
  9. Packet Capture mode : None
  10. Packet Capture duration: 0
  11. MTP required : TICK
  12. Retry Video call as audio : TICK
  13. SIP information – Destination Add:
  14. DST is a SRV: UNTICK
  15. Destination port : 5060
  16. SIP PROFILE : Presence non-secure SIP trunk Profile
  17. Save

Step#4: Make your IP Phone presence capable

  1. Register a phone 2001 name it HQ-Phone1
  2. Create end user “test” and associate HQ-Phone1/2001 with the “test” user
  3. Make sure test user is a part of “Standard CCM End User” and “standard CTI enable”
  4. Make sure Primary extension “2001” is selected when you create the above “test” user

Also Make Physical phone DN2001 has “test” user associated with it. This is the last option in line 2001’s setting before “save” button. If this has not been done and you run presence diagnostic it will keep telling you that “No line appearance existed in CCM.

Step#5: Add an application user for IPPM and MOC CTI ports

This will be used by Presence server to initiate IP Phone services:

A) Go to > User Management>Application User>

  1. User ID : IPPM
  2. pass : blah
  3. Presence Group : Standard
  4. Select the Controlled devices for this feature
  5. Groups : Standard CCM End User
  6. Save

Repeat above “A” steps for MOC_USER as well. MOC_USER will be used by MOC CTI user in Presence. All users who want presence using Microsoft MOC client will be associated to this user.

Make sure all “accept” tick boxes are TICKED on MOC_USER.

B) Go to > SYSTEM>Application Server> Add NEW

Add Presence server IP address here i.e.

Step#6: Create IP Phone service URL

Go to> Device>Device Settings> IP Phone Service

  1. Service Name : IP PhoneMSG
  2. ASCII Service Name : IP PhoneMSG
  3. Service Description : Blah
  4. Service URL :
  5. Service Category : XML Service
  6. Service Type : Standard IP Phone Service
  7. Blank
  8. Blank
  9. Enable : TICK
  10. Save

****Then subscribe above service to HQ phone1/2001*****

Step#7: Enable presence licensing for each user

Go to> System>License>Capability Assignment>

Then Find the end user you want to assign the presence license.

Tick the user and hit <Bulk Assignment>

A new pop up window will come. Tick both check-boxes in that and save.

  1. Enable CUP – TICK
  2. Enable CUPC – TICK

That s all we needed to do on Call Manager. Now Jump on the Presence BOX.

Step#8: Presence box general configuration:

After installing basic presence, you’ll see presence post install setup screen on your web browser by typing presence Server IP address on your browser and supplying credentials to the login screen.

So you’ll see “Post Install Setup” screen with below options:

  1. CUCM Publisher IP address : (default, not changeable)
  2. AXL User : Administrator
  3. Axl password: blah…
  4. Confirm password : blah <then hit the “NEXT”>
  5. Security password : blah (whatever you supplied during installation)
  6. Then hit the “CONFIRM” (Ignore the warning)

Finally you will get 3 options:

A) Home B) Status C) TOPOLOGY

  1. Click on “HOME” you’ll see you are in a new home i.e. presence main admin page.

Step#9: Upload License and Activate presence Services

  1. First upload the license if you haven’t done that so far.
  2. GO to > Cisco Unified Serviceability>>Tools>Activate services
    Activate all services; it will take 2-3 minutes.

Step#10: Configure Presence

Jump straight on Presence Admin page>>Diagnostic>System Troubleshooter

Pay attention to RED crossed balls and yellow exclamation! Signs and fix them one by one.

  1. Under Presence Engine: Click on FIX under “no communication presence” this will take you to add presence gateway:

Add NEW>

Presence Gateway type : CUCM
description : blah

Presence Gateway: ← CCM IP

Double check the settings under below menus:

  1. SYSTEM> CCM Publisher : Check all parameter under this
  2. SYSTEM> Application Listener>Default class SIP TCP Listener (make sure its what you have defined in the SIP trunk on CCM – transport method TCP or UDP, both should have the same protocol/port) we are using:
    Protocol = TCP
    PORT = 5060
    Add NEW> description=blah/all address pattern=all

Step#11: Tune the Presence Engine’s Service parameter (same as we do with CCM)

SYSTEM>> Service Parameter>Select active CUPS Server> Select Presence Engine

SYSTEM>> Service Parameter>Select active CUPS Server> Select UP SIP Proxy(ver 7)


  1. Search “Proxy Domain” and set it to : domain name (Ex.
  2. Search “Transport Preferred Order” and set it to : TCP/UDP/TLS

Step#12: Configure IP Phone Messenger on Presence server

Application>IP Phone> Setting

  1. IPPM Application Status : ON
  2. Application user Name : IPPMSG (created in step 3A)
  3. Application Password: blah…
  4. confirm password : Blah
  5. Max Instant message : 25 default
  6. Subscription timeout : 3400 default   (3600 in ver 7)
  7. Publish timeout : 3600 default

Hit “SAVE”

Step#13: Select a SIP trunk between Presence to CCM

Tell presence which SIP trunk should be used for pumping calls to CCM.


  1. CUP CVP Support : UNTICK
  2. MAX Contact List Size : 200
  3. Enable Instant messaging : TICK
  4. Enable SIP Publish on CUCM TICK
  5. CUCM SIP Publish Trunk : <Select_Your_Trunk><– A MUST

Don’t forget to save after above. Above SIP trunk will be automatically listed in above “5”. This we is the one we created on CCM.

Step#14: Set TFTP address for IP COMMUNICATOR Clients

Application>Unified IP Personal Communicator>Settings

  1. Proxy Listener : Default Cisco SIP proxy TCP Listener
  2. Primary TFTP : (CCM pub tftp)
  3. Backup TFTP : (sub tftp) or whatever

LDAP – if you are using LDAP put LDAP parameters there. Else disable it.

Step#15: For MOC client define CTI Gateway

Application>>CUCM CTI Gateway>Settings

  1. Application Status : ON
  2. Application Username : MOC_USER (make sure its created on CCM as app user)
  3. Application Password : blah
  4. Confirmed Password : blah
  5. CUCM Address : (CCM address)

Now time to run the Presence troubleshooter again. This will tell you what’s remaining and how to fix it. Once those are done, activate the presence and other services. Still remaining:

  1. MOC integration
  2. Creating users and testing presence
  3. Voicemail integration with Presence



 Step 1  Power ON the Secondary Server and start the services.

Step 2  Upload HA license to Publisher.

System > License Information > Add License.

Step 3  Add new server to the cluster.

System > Server>Add New.

Step 4  Select Network   Deployment Type – WAN/LAN

System > System Parameters.

Step 5  Select Failover Server.

Cisco Unified CCX Serviceability>Tools>Service parameters>ALPHAUCCX>Cisco AMC Service>Failover collector>

Step 6  Check the DB replication status for the data stores.

             Cisco Unified CCX Serviceability>Tools > Data store  Control Center > Replication Servers  

Step 7  Create CTI ports for Secondary Server.

Step 8  Reboot the Servers.


Cisco Nexus 1000V Webinar Training Video’s

Nexus 1000V Webinar Training Videos
Nexus 1000V Family Overview and Update
Virtual Services (VSG, vWAAS, vNAM)
Virtual Security Gateway Introduction
Journey to the Cloud w/ N1KV: vCloud Director & Long Distance vMotion
Secure VDI with Nexus1000V & VSG
Nexus 1000V v1.4 New Features and Installation/Upgrade Overview
Nexus1010 Overview & Best Practices
Virtual Security Gateway Technical Overview
Nexus 1000V Key Features Overview
Nexus 1000V Troubleshooting
Nexus 1000V Web Site
Nexus 1000V Webinars
Nexus 1000V and VMware vCloud Director Whitepaper
Nexus 1000V and VMware View Cisco Validated Design
Cisco Virtual Experience Infrastructure Configuration Guide (including Nexus 1000V)
Virtualize DMZ Whitepaper
Nexus 1000V Community
Cisco Nexus 1000V Virtual Switch Software Demo Video
VMware and Cisco Accelerate Virtual Machine Deployments
Cisco Nexus 1010 Virtual Services Appliance
Cisco Nexus 1000V Network Analysis Module Virtual Service Blade 4.2.1N
Cisco Nexus 1000V Series Switches
Cisco Nexus 1000V Series Switches Video Data Sheet
Cisco Virtual Network Management Center
Cisco Virtual Security Gateway for Cisco Nexus 1000V Series Switches
Cisco Virtualization Experience Infrastructure (VXI) Configuration Guide
Cisco Nexus 1000V MIB Quick Reference
Cisco Nexus 1000V Series Switches Deployment Guide Version 2
Best Practices in Deploying Cisco Nexus 1000V Series Switches on Cisco UCS B and C Series Cisco UCS Manager Servers
Cisco Nexus 1000V Command Reference, Release 4.2(1)SV1(4)
Cisco Nexus 1000V Installation Screencasts
The following screencasts are available to assist you in the installation of Release 4.2(1) SV1(4) of the Cisco Nexus 1000V.
Understanding Cisco Nexus 1000V and VMware Software Version Compatibility
Video This screencast introduces and explains the use of the Host Software Version Compatibility table in the Cisco Nexus 1000V and VMware Compatibility Informationdocument provided for each release. Given the VMware host ESX or ESXi software version, the compatibility table shows the appropriate Virtual Ethernet Module (VEM) installation files and the minimum required software versions of the VMware vCenter Server, the vCenter Update Manager, and the vSphere CLI. The screencast also shows how to determine the software version of various components and how to download update software.
Installing a Redundant Pair of VSMs from an OVA file Using the Installer Application
Video This screencast shows how to install a redundant pair of VSMs in an HA configuration from an OVA file using the Installer Application.
Installing the VEM using the VUM [two parts]
VideoVideo Part 1 of the screencast shows you how to install the VEM on an ESX host using the VUM.Part 2 of the screencast shows you how to troubleshoot and verify the installation.
Installing the VEM using the vSphere CLI
Video This screencast shows how to install the VEM on an ESX host using the VMware vSphere CLI.
Cisco Nexus 1000V Upgrade Screencasts
The following screencasts are available to assist you in upgrading the Cisco Nexus 1000V to Release 4.2(1) SV1(4). We recommend that you view these screencasts and perform the procedures in the order shown.
Understanding Cisco Nexus 1000V and VMware Software Version Compatibility
Video This screencast introduces and explains the use of the Host Software Version Compatibility table in the Cisco Nexus 1000V and VMware Compatibility Informationdocument provided for each release. Given the VMware host ESX or ESXi software version, the compatibility table shows the appropriate Virtual Ethernet Module (VEM) installation files and the minimum required software versions of the VMware vCenter Server, the vCenter Update Manager, and the vSphere CLI. The screencast also shows how to determine the software version of various components and how to download update software.
Understanding the Upgrade Process for the Cisco Nexus 1000V Release 4.2(1) SV1(4)
Video This screencast provides an overview of the steps required to update your VMware infrastructure for a non-disruptive upgrade, and to update your VEMs and VSMs to Cisco Nexus 1000V Release 4.2(1) SV1(4).
Upgrading the VMware vCenter Update Manager to Release 4.0 Update 1 Patch 2
Video This screencast shows how to upgrade the VMware vCenter Update Manager (VUM) to the minimum required version for the upgrade to Cisco Nexus 1000V Release 4.2(1) SV1(4).
Upgrading the Cisco Nexus 1000V from 4.0(4) SV1(3, 3a, or 3b) to 4.2(1) SV1(4) [three parts]
This three-part video shows you how to upgrade your Virtual Ethernet Modules (VEMs) and Virtual Supervisor Modules (VSMs) to Cisco Nexus 1000V Release 4.2(1)SV1(4).
Video Part 1 of the screencast describes the process, lists the prerequisites, and shows how to download the required software.
Video Part 2 of the screencast shows you how to upgrade the VEMs.
Video Part 3 of the screencast shows you how to upgrade the Virtual Supervisor Modules (VSMs).
Upgrading the Cisco Nexus 1000V from VMware Release 4.0 to 4.1 [three parts]
This screencast shows how to upgrade your vCenter Server, vCenter Update Manager, and vSphere CLI from VMware Release 4.0 to Release 4.1.
Video Part 1 describes how to check software compatibility and how to upgrade the vCenter Server.
Video Part 2 describes how to upgrade the VUM.
Video Part 3 describes how to upgrade the hosts.

CCNP Voice free Training

Cisco Quick Learning Module
Introducing Voice over IP
CTT-TAC: Introduction to Basic Analog Voice over IP 
CTT-TAC: Basic Analog-to-Digital Voice over IP
CTT-TAC: Analog Voice Internetworking with E&M Signaling 
CTT-TAC: Basic Analog-to-Digital Voice over IP 
Understanding Gateway Dial Peers
Understanding Dial Plans
Implementing Features in Cisco Unified Communications Manager
Configuring Presence-Enabled Speed Dials and Call Lists
Implementing Partitions and Calling Search Spaces
Implementing AAR for Locations-Based CAC in Cisco Unified Communications Manager 6.0
Examining Remote Site Redundancy Options
Implementing TEHO with PSTN Backup in Cisco Unified Communications Manager v6.0
Cisco Unified Communications Express as Survivable Remote Site Telephony
Gathering Information for Troubleshooting
Troubleshooting Common Gateway Registration Issues
Troubleshooting Common Cisco Unity Integration Issues
Introduction to Modular QoS CLI
Congestion Management Configuring CBWFQ and LLQ
Congestion Avoidance Introducing RED and WRED

All About Extension mobility


This document describes the common problems in Extension Mobility.

Error :- Host not found



  • Check that the Cisco Tomcat service is running by choosing Cisco Unified Serviceability > Tools > Control Center—Network Services


  •                          If you have changed the ip address on service URL then click on “Update subscriptions” (Device > Device Settings > Phone Services >IP Phone Services Configuration).and resubscribe each phone to which the wrong service was subscribed.




Error:- You can’t see the EM feature after hitting the services button




-Verify that you have configured the Extension Mobility service

-Verify the service URL is correct

– Start/Restart the EM services on each node you are running.




Error:- You can’t log in/out of the EM feature but you can see it after pressing the services button



This error comes when you haven’t enabled the extension mobility , subscribed the phones/device profiles to the service as needed and haven’t associated user to a device profile.






Error:- To set up speed dials and other services from your phone, please goto https://x.x.x.x:8443/ccmuser/


The above error comes when you haven’t subscribed the phone or device profile to the EM profile. Once this is done you should be able to see the EM profile and log in correctly.

Error:-After a user logs out and the phone reverts to the default device profile, the user finds that the phone services are no longer available.


1. Check the Enterprise Parameters to make sure that the Synchronization Between Auto Device Profile and Phone Configuration is set to True.

2. Subscribe the phone to the Cisco Extension Mobility service.



Error:-After performing a login or logout, the user finds that the phone resets instead of restarting.


  • Locale change may provide the basis for reset.


  • If the User Locale that is associated with the login user or profile is not the same as the locale or device, after a successful login, the phone will perform a restart that is followed by a reset. This occurs because the phone configuration file is being rebuilt.

Error[201]-Authentication error


The user should check that the correct UserID and PIN were entered; the user should check with the system administrator that the UserID and PIN are correct.

Error [22]-Dev.logon disabled


Make sure that you have chosen “Enable Extension Mobility” check box on the Phone Configuration window.

Error [205]-User Profile Absent


Make sure that you have associated a Device Profile to the user.

Error [208]-EMService Conn. error


Verify that the Cisco Extension Mobility service is running by choosing Cisco Unified Serviceability > Tools > Control Center—Feature Services

Error [25]-User logged in elsewhere


Check whether the user is logged in to another phone. If multiple logins need to be allowed, ensure the Multiple Login Behavior service parameter is set to Multiple Logins Allowed

Error:- Http Error [503]



If you get this error when Services button is pressed, check that the Cisco Communications Manager Cisco IP Phone Services service.

  • If you get this error when you select Extension Mobility service, check that the Cisco Extension Mobility Application service is running by choosing Cisco Unified Serviceability > Tools > Control Center—Network Services.



Error:- [202]-Blank userid or pin


Enter a valid userid and PIN.

Error:- [26]- Busy, please try again


  • Check whether the number of concurrent login/logout requests is greater than the Maximum Concurrent requests service parameter. If so, lower the number of concurrent requests.


  • To verify the number of concurrent login/logout requests, use Cisco Unified Communications Manager Cisco Unified Real-Time Monitoring Tool to view the Requests In Progress counter in the Extension Mobility object.

Error:-[6]-Database Error



  • Check whether a large number of requests exists
  • If large number of requests exists, the Requests In Progress counter in the Extension Mobility object counter specifies a high value. If the requests are rejected due to large number of concurrent requests, the Requests Throttled counter also specifies a high value.

Error:- [207]-Device Name Empty


Error:- XML Error [4] Parse Error


Check that the URL that is configured for Cisco Extension Mobility is correct and there should be no space in between.

Error:- 8945 phone does not show EM service

Resolution:-   Set service provisioning to default or internal. Refer Bug CSCtx70127

Distributed Voice and VXML Gateway Design with CVP SIP Deployments


This document discusses solutions to distributed VXML and Voice GW deployments when Ingress-GW and VXML-GW are NOT residing on same router in SIP call flow scenarios.


When the VoiceXML and voice gateway functions reside at the same branch location but on separate devices, an IPCC design engineer should make sure that the VRU leg is sent to the local VoiceXML GW. There are two ways to ensure that the calls are handled within the branch and not sent across the WAN to a different VoiceXML gateway.

Two possible solutions

  • Configure Unified ICM with multiple Customers instances, one per location
    • Since this is not a scalable solution and also cannot be used if VRU is Type10 so we will not discuss it here.
  • Configure Unified CVP with setTransferLabel or SigDigits feature
    • If a customer has VoiceXML Gateway and Voice Gateway configured on the same router, then you should use CVP Call Server’s “setTransferLabel” mechanism which applies only to co-resident VoiceXML and Voice Gateway configurations.
    • If a customer has VoiceXML Gateway and Voice Gateway configured on different routers, then you should use CVP Call Server’s “SigDigits” mechanishm which applies to distributed VoiceXML and Voice Gateway configurations.

Unified CVP SigDigits Feature

The SigDigits feature in Unified CVP allows you to use the dial plan on the SIP Proxy to route calls to the correct site. When the call arrives at an ingress gateway, the gateway will prepend digits before sending the call to Unified CVP. Those prepended digits are unique to that site from a dial-plan perspective. When the call arrives at Unified CVP, Unified CVP will strip the prepended digits and store them in memory, resulting in the original DID on which the call arrived. Unified CVP then notifies Unified ICM of the call arrival using the original DID, which matches a Dialed Number in Unified ICM.




Call arrives at the Toronto voice gateway (TOR-V-GW) with a DID of 416-431-1000. The Toronto Voice Gateway will add 821 as a site code and will send it to central SIP Proxy server in Chicago. The Chicago SIP Proxy will send it to Chicago CVP server. So Chicago CVP Server will receive 821.416-431-1000. On the Chicago CVP server we have “SigDigits=3” configured. Chicago CVP server will strip 821 and will send 416-431-1000 as a dialed number to ICM


Unified ICM Returns a Label

When Unified ICM returns a label to Unified CVP in order to transfer the call to a VoiceXML gateway for IVR treatment or to transfer the call to an agent phone, Unified CVP will prepend the digits that it stored in memory before initiating the transfer. The dial plan in the SIP Proxy must be configured with the prepended digits in such a way to ensure that calls with a certain prepended digit string are sent to specific VoiceXML gateways.




ICM returns 416-431-2000 as label to queue the call at the edge Toronto VoiceXML gateway (TOR-VXML-GW). ICM will add 821 and will send the call to SIP Proxy server. Chicago SIP Proxy server will receive 821.416-431-2000. Based on the 821 site code, it will send call to TOR-VXML-GW for queuing.


How to turn on SipDigits Feature

To turn on this feature and specify how many significant digits should be stripped is to change the sip.SigDigits = X field in file in all CVP Call Servers. Where X is the number of digits to be stripped.


SIP RingTone and Error Services

There is one caveat that SigDigits feature does not cover the 91919191 (RingTone) and 92929292 (Error) DNs for those services. What this means is that once the agent becomes available, CVP needs to pull the call that is queued at the edge on the VXML-GW and connect the caller to the agent. During this transition, CVP needs to play a ringback tone to the PSTN caller. CVP does that by initiating a SIP call (CVP sends a SIP invite) towards the VXML-GW by dialing 91919191 to play a rintone back.


VXML-GW matches this 91919191 on an incoming SIP VoIP dial-peer and sends ringback.wav file to the ingress Voice GW. This file is being stored in the VXML-GW and is benign sent to the ingress GW. So there will be g711 traffic will flow for some duration.