The new version of Aurus RichCall product provides the co-browsing option that allows contact center agent to see the client's browser window (while talking with him on phone) and use the "pointer" to instruct him on what to do. This allows agent to better understand the context of the client's issue and solve it appropriately.

Before starting the co-browsing development we interviewed our clients interested in this option to analyze their business cases and develop exactly those features needed to fit their requirements.

You may be interested in learning what real clients want from the co-browsing option. Below are some of the requirements that we had to meet.

No Downloads

Of course no downloads for the customer. The client must be able to start the co-browsing session instantly. No apps, no Java, no browser plug-ins.

Client Controls the Action

Service reps should not be able to make mouse clicks or keyboard entries in the client's browser. The agent must have the ability to see what happens and point (see "The Pointer" below) the client what to do but not to interact with him on the same page.

The "Pointer"

Though the agent cannot interfere with customer actions, he must be able to draw the customer attention to certain areas of the page:

• "Please click this button..."
• "Here is the section with the info you need…"

We call it "the Pointer" tool – an arrow that appears in the customer browser on top of the main content.

Secure Pages Support

The co-browsing session must support secure pages (the ones that require client's login). When we interviewed our clients, only two of them reported their need to guide visitors around public pages to help them find products or other public info. The rest want to support their clients in working with the secure online self-service tools, which require user's authentication.

Starting the Co-Browsing Session by Code

The client is not required to make an online call, or to be in a chat with the agent to start the co-browsing session. Even if the client has made a regular phone call to the contact center, he must be able to enhance it with the co-browsing session if (and when) he needs it. With Aurus RichCall the agent may generate the unique 5-digits code and say it to the client who then uses it to initiate the co-browsing.

Mobile Browser Support

The co-browsing feature should support mobile browsers and provide the same functionality ("pointer", secure pages support etc).

Do you manage a Cisco Contact Center? Are you looking for a solution to get a more detailed second by second view of what is happening?

Wallboard solutions enable you to monitor your contact center activity and performance in real-time displaying calling stats, KPI info and goals against actuals. Depending on the dashboard’s purpose it may contain calls waiting, average waiting time, calls answered/dropped/abandoned, longest/average waiting time, abandon rate and lots of other indicators.

Wallboard is also an excellent way to communicate with your contact center staff – supervisors may push text messages to inform agents about important events and changes.

Most of wallboards/dashboards for Cisco UCCX / UCCE are developed by 3rd party software vendors, most of them are Cisco Solution partners.

So here is the list…

1. 2Ring DASHBOARDS & WALLBOARDS is a solution for calculating & displaying real-time data in contact centers. For every team, create a unique layout with KPIs based on contact center, ERP or ticketing system data, pictures, message tickers / marquees, youtube videos, flash and web content, and even PowerPoint slides.

2. Comstice has a great Wallboard. It shows real time stats and has different views to show: box view/ agent states/ dashboard view/ team view. And, it offers Team Voicemail.

3. Inova Contact Center Digital Signage delivers key performance metrics and important business information, along with rich, multimedia content on crystal-clear, high-definition monitors that keep your team motivated, informed and empowered to better serve your customers.

Call data awareness = agent empowerment = higher customer sat

4. Atea Systems UCCX Agent State Wallboard (UAW) allows organizations to concurrently display many different views of customer configured real time data for UCCX queues and agents on any browser capable device.

Easy config, many different views!

5. ccInfo is a powerful wallboard application for customer contact centers to visually aid supervisors and agents with real-time statistics on call traffic and handling.

6. Tenox Wallboard – free Wallboard for Cisco Unified Contact Center Express (UCCX/CCX).

It's a small .exe, is simple to set up, and can work with the old Windows platform or the new Linux/Informix combination in UCCX8.0 plus.

7. Inova Solutions Dashboards and Wallboards is a real-time reporting solutions that deliver critical metrics to your team via customized views for wallboards, dashboards and mobile devices. Inova real-time performance management solutions provide consolidated reporting across multiple systems through customized wallboards and readerboards, multimedia digital displays, web-based dashboards and desktop applications.

• "How to configure Meet-Me video conference with PIN?"
• "We need to configure Meet-Me conference with PIN…"
• "CUCM 10.x Meet-Me with Name announcement and Pin number…"

- try to search the Cisco Support Forum and you'll get dozens of similar tickets.

Yes, the built-in Meet-Me conferencing feature doesn't support PIN authentication, so here goes the list of all possible ways to set it up.

Before we start, let's agree that we only consider Cisco Unified Communications Manager Enterprise or Cisco BE 6000/7000. If you've got CUCM/CallManager Express (CME) you can play with TCL IVR scripts, but that’s definitely another story.

So you're enjoying Cisco UCM Meet-Me conferencing, but you want attendees to hear the voice prompt asking for a PIN needed to join the meeting. You’ve got 4 options:

1. Cisco Unity Express/Connection

If you have Cisco Unity deployed you can use it to achieve the Meet-Me authentication. The attendees call should be transferred via CUC and the User System Transfer Conversation should be used to authenticate the caller (a user is created on CUC). The conversation prompts the caller to sign in to CUC with his CUC ID and PIN and then transfers the call to Meet-Me conference number.

Looks like a kludge? Still it gets the job done.

2. Cisco Unified Contact Center Express (UCCX)

You can use UCCX as an audio front end to Meet-Me conferences. You may find several UCCX scripts around the web which prompt the caller for meeting ID and password and then transfer the call to the MeetMe bridge. Meeting IDs and PINs are set up by UCCX admin.

This works, but since UCCX is used for something that it’s not initially designed for, this workaround isn’t as feature-rich as you users may require and hard to maintain.

3. The "Conference Now" feature of CUCM 11

The "Conference Now" feature is introduced in Cisco UCM v.11 released in the summer of 2015. It's not the replacement for Meet-Me feature, but allows users to create their personal conference rooms protected (optionally) by the access code. The attendee has to call your conference room number, enter the access code and listen to the music until you start the meeting by joining it.

Quite promising feature but the access code stays the same until you change it, so if I participated in your meeting once, I can use it to join the next time even if I wasn’t invited. Also, no scheduler and conference control tool are available.

4. The “Conference” module of “Aurus PhoneUP” suite

The CUCM Meet-Me conferencing solution from Aurus is designed specifically for CUCM conferencing and allows you to:

• schedule Meet-Me conferences with a web-interface;
• use the MS Outlook plugin to schedule the meetings from the Outlook calendar;
• use the same phone number for meetings (the PIN entered is used to define which meeting you're joining);
• protect meetings with a randomly generated PIN (the PIN is automatically added to the meeting invitation sent to invitees);
• control the conference with the web-interface – the meeting host can see the list of participants, join a new attendee and disconnect anyone;
• start the meeting from any phone - without the necessity to use a Cisco IP phone to initiate the meetme bridge;
• control the resources of the conference bridge.

So, these are 4 options to secure your Meet-Me conferences. Which one works better for you, depends on your business requirements and Cisco products used. Hopefully, this article will help you to evaluate the pros and cons of each option in relation to your environment.

Why?

Some operations on CUCM objects could be made much easier and faster through CUCM database, for example - to get a list of devices, to add several devices to the list of devices controlled by some axl-user, etc.

How?

1. Download CURL for Windows - http://curl.haxx.se/download.html
2. Then any AXL-request can be executed from the command line:
curl.exe -k -u axluser:axlpass -H "Content-type: text/xml;" -H "SOAPAction: CUCM:DB ver=8.5" -d @axlreauest.xml https://ccm9.bcs-it.loc:8443/axl/

where:

a) axluser:axlpass - login and password of the CUCM Application User
b) https://ccm9.somedomain.loc:8443/axl/ - CUCM AXL Service address
c) axlreauest.xml - the file with the request, for example:

<soapenv:Envelope xmlns:soapenv="http://schemas.xmlsoap.org/soap/envelope/" xmlns:ns="http://www.cisco.com/AXL/API/8.5">
    <soapenv:Header/>
    <soapenv:Body>
  <ns:executeSQLUpdate sequence="?">
<sql>
insert into applicationuserdevicemap (description, fkdevice, fkapplicationuser, tkuserassociation)
values ('', 'f37738df-222b-473c-813f-7872709ab221', '1abf2126-3339-5946-e755-6d59f368a7a5', 1)
</sql>
  </ns:executeSQLUpdate>
    </soapenv:Body>
</soapenv:Envelope>

Queries to CUCM tables are performed with the executeSQLQuery. Modification of data (insert / delete / update) – with executeSQLUpdate.

What is CUCM DB?

It is actually Informix.

To find out the exact version of the database server you can perform this query:

<soapenv:Envelope xmlns:soapenv="http://schemas.xmlsoap.org/soap/envelope/" xmlns:ns="http://www.cisco.com/AXL/API/8.5">
    <soapenv:Header/>
    <soapenv:Body>
  <ns:executeSQLQuery sequence="?">
<sql>
select DBINFO('version', 'full') from systables where tabid=1
</sql>
  </ns:executeSQLQuery>
    </soapenv:Body>
</soapenv:Envelope>

Read more about Informix here: http://publib.boulder.ibm.com/infocenter/idshelp/v111/index.jsp

The structure of the database

1. CUCM 6: http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/cucm/datadict/6_0_1/dd601.pdf
2. CUCM 7: http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/cucm/datadict/7_0_1/DD_701.pdf
3. CUCM 8: http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/cucm/datadict/8_0_1/datadictionary_801.pdf
4. CUCM 9: http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/cucm/datadict/9_1_1/datadictionary_911.pdf
5. CUCM 10: http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/cucm/datadict/10_0_1/datadictionary_1001.pdf

Ask Cisco about the structure of the other databases: http://tools.cisco.com/search/results/en/us/get#q=Communications+Manager+Data+Dictionary

Reference

Here's our friend having fun on CUCM SQL quieries: http://www.ucguerrilla.com/2012/03/cucm-sql-queries-series.html

Examples of requests

Get the list of All CUCM devices
select * from device

Get the list of devices controlled by the user (returns identifiers)
select * from applicationuserdevicemap
where fkapplicationuser = 'a59e7a1c-3527-c5e4-89d8-edb3e0c10dab'

Remove a bunch of devices from the “controlled devices” list
delete from applicationuserdevicemap where fkapplicationuser = 'a59e7a1c-3527-c5e4-89d8-edb3e0c10dab'
and fkdevice in (select pkid from device where name like 'zCTIPort%' and name[9,15] > 5565400)

Add a bunch of device to the “controlled devices” list
into applicationuserdevicemap (description, fkdevice, fkapplicationuser, tkuserassociation)
select '', pkid, 'a59e7a1c-3527-c5e4-89d8-edb3e0c10dab', 1 from device where name like 'zCTIPort%' and name[9,15] > 5565402

RIS Service

To test the RIS the "/ realtimeservice / services / RisPort" path should be used in the URL instead of "/ axl".

The example of the RIS-query:

<soapenv:Envelope
     xmlns:soapenv="http://schemas.xmlsoap.org/soap/envelope/"
     xmlns:xsd="http://www.w3.org/2001/XMLSchema"
     xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance">
<soapenv:Body>

<ns1:SelectCmDevice soapenv:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/"
  xmlns:ns1="http://schemas.cisco.com/ast/soap/">
  <StateInfo xsi:type="xsd:string"/>
  <CmSelectionCriteria href="#id0"/>
</ns1:SelectCmDevice>

<multiRef id="id0" soapenc:root="0" soapenv:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/"
  xsi:type="ns2:CmSelectionCriteria"
  xmlns:soapenc="http://schemas.xmlsoap.org/soap/encoding/"
  xmlns:ns2="http://schemas.cisco.com/ast/soap/">

  <MaxReturnedDevices xsi:type="xsd:unsignedInt">200</MaxReturnedDevices>
  <Class xsi:type="xsd:string">>Phone</Class>
  <Model xsi:type="xsd:unsignedInt">255</Model>
  <NodeName xsi:type="xsd:string" xsi:nil="true"/>
  <SelectBy xsi:type="xsd:string">Name</SelectBy>

  <SelectItems soapenc:arrayType="ns2:SelectItem[1]" xsi:type="soapenc:Array">
<item href="#id1"/>
  </SelectItems>

</multiRef>

<multiRef id="id1" soapenc:root="0" soapenv:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/"
  xsi:type="ns3:SelectItem"
  xmlns:ns3="http://schemas.cisco.com/ast/soap/"
  xmlns:soapenc="http://schemas.xmlsoap.org/soap/encoding/">

  <Item xsi:type="xsd:string">test</Item>
</multiRef>

    </soapenv:Body>
</soapenv:Envelope>

Cisco Unified Communications Manager (CallManager) is the leading IP PBX in the worldwide market. Only certified professionals can deal with its rich functionality. Nevertheless, dozens of software vendors develop products that empower CUCM (Cisco Unified Communications Manager) functionality with new features.

Aurus, the official Cisco Solution Partner, invites you to learn what new features the Aurus PhoneUP product brings to Cisco IP PBX (Cisco Unified Communications Manager or Cisco BE 6000/7000).

Enterprise Phone Directory for Cisco CallManager

The "Directory" module of the PhoneUP application bundle provides the enterprise phone directory service for employees, and is an alternative to built-in CUCM phone directory. The main differences between the "Directory" module and the out-of-the-box Cisco CallManager directory are:

	CUCM	"Directory" module
Number of directories	One	Unlimited number with access control for groups of employees
Integration with external systems	AD	AD, LDAP, IBM Lotus Notes, CSV, XML, CUCM, SQL database
External contacts support	No	Yes
Caller ID	Internal numbers only	Any phone number
Incoming call details	Name / Last Name / Company	Flexible data that may contain an employee photo
Notification of missed calls via e-mail	Only with Unity	Yes
DTMF support	No	Yes
Personal directories	One (edited manually or imported from Outlook)	Unlimited number of directories synchronized with external sources
Auto-redial feature	No	Yes

The "Directory" module is integrated with Cisco Jabber. Using the standard contact search field of Cisco Jabber not only you can find an employee, but also search for client, partner and any other contact imported from external datasource. Also, when you get a call from the contact (client for example) stored in the directory Cisco Jabber will show the client's name, status and any other information from the CRM system.

Phone Call Recording (CUCM)

Cisco offers its own solution, Cisco MediaSense for call recording that captures and stores audio, forked by Cisco IP phone bridge or CUBE (Cisco Unified Border Element). But MediaSense is a recording platform that only provides basic features to manage call recordings. Cisco officially recommends using 3rd party solutions developed by technology partners that provide additional functionality.

For example, the "Record" module of PhoneUP can be integrated with Cisco Mediasense to bring you extra-features not implemented in Mediasense:

• flexible management of user access to call recordings;
• rich search and filter tools;
• call recording rules (for example, recording only external calls);
• IP phone user interface to search and play the call recording;
• playing the recording into the current phone call;
• and much more.

Attendant Console

Cisco actively promotes Cisco Jabber, the enterprise collaboration tool. At the same time, some business units need a wider range of call control features including specific ones, for example:

• top manager assistants need to monitor chief's phone lines and intercept calls during his absence;
• reception staff and contact center need a visual control of the call queues not to miss client calls or VIP calls that should not be left unanswered;
• top managers need an intuitive, time-saving interface allowing to perform basic call control actions with easy to use UI supporting drag-n-drop.

In these cases you should pay attention to the "Console" module.

Group paging via Cisco IP phones

Features of Cisco Unified Communications Manager (CUCM) and Cisco IP-phones, as well as API Cisco provides its technology partners, allow to use the IP telephony network for text and audio notifications to employee groups.

Cisco's collaboration product line doesn't include such a solution, and clients need to turn to 3rd party vendors. The "Paging" module of PhoneUP bundle supports both text and audio paging to Cisco IP phones as well as live broadcasting through speakerphones.

Features that improve the security of IP telephony network built on Cisco Unified Communications Manager (CUCM)

Cisco CallManager (CUCM) provides users with the Meet-Me conferencing feature – each conference has its phone number that needs to be dialed to join the meeting. But the "built-in" Meet-Me conferences do not provide the necessary security. Anyone who knows the number of conference room can dial it and join. To avoid this you can use the "Conference" module that works on top of CUCM conferencing feature and protects meetings with PIN.

The "Lock" module is used to lock Cisco IP phone while its owner is away to prevent abuses and frauds. In some ways this is similar to the "Extension Mobility" feature implemented in Cisco Unified Communications Manager, but is designed to improve the security. Unlike Extension Mobility a phone locked with PhoneUP:

• stays registered to CUCM;
• can receive incoming calls;
• is allowed to make outgoing calls to a limited set of directions;
• denies the access to the personal address book, call history, call recording, etc .;
• gets locked and unlocked automatically when you log in/out the PC.

XML-services for Cisco IP phones

Cisco Systems provides technology partners with rich capabilities for development of custom XML-applications for Cisco IP phones - with which users can interact via the IP phone keypad and display. The "Inform" and "Hotel" modules of the PhoneUP bundle may be considered as examples of such applications.

So, phone auto registration on CUCM. The topic is easy and most likely well-known.

A small prehistory. My colleague received a request to connect the phone for the new employee yesterday. The task is easy and repeated several times every day.

In our company auto registration on CUCM is always enabled. Yes, I know that Cisco does not recommend to keep it constantly enabled for security reasons, but so we have decided, because all the sockets in which you can plug the phone are in the closed area and outsiders do not have access to them.

My colleague, taking a new phone out of the box and turning it on to the network, instead of the expected signs 'Your current options' and numbers from the pool for auto registration, saw the following:
Registration Rejected: Security Error.

Then I began to research CUCM logs. And I saw the line in the SDL traces:

AddDevice returns "There are no free autoreg DN in the system free DN between 1010 and 1099".

It means that the auto registration process of CUCM unable to create a new DN within a predetermined range and a predetermined Partition. Why it could not do? Because these DN-s are already in the system! They can be viewed in the section Call Routing -> Directory Number.

The solution of the problem is elementary – to change the range in auto registration settings or remove unnecessary DN-s. Because the auto registration is usually used in order to do not enter the MAC address manually. After auto registration the phone is configured manually.

And now a few words about how to configure the auto registration.

• Go System -> Enterprise Parameters. Specify which protocol will be used for auto registration in the Enterprise Parameters Configuration section in the Auto Registration Phone Protocol parameter. SCCP is used by default.
• Create a Partition for auto registration. Theoretically, you don't have to do it, but it would be more correct to create it. Generally Partitions and Calling Search Space is a very powerful tool in CUCM.
• Configure Device pool (if it’s not configured).
• Go Device -> Device Settings -> Device Defaults and then specify the default device pool for all used types of phones, for example, the one that is configured previously.
• Select the server in System -> Cisco Unified CM, on which we enable auto registration. Specify the range of numbers, Partition, and External Phone Number Mask in Auto-registration Information section. The main thing is to disable Auto-registration Disabled on this Cisco Unified Communications Manager checkbox. So it is possible to set up multiple servers for auto registration. It is preferable to specify for them the different ranges of DN.
• Check CM groups in System -> Cisco Unified CM Groups. Auto registration can be enabled on the one group only! If you want to enable it on the other group, then go to the group settings and select the Auto-registration Cisco Unified Communications Manager Group item.

All appropriate services should be enabled in Cisco Unified Serviceability for auto registration process; i.e. Cisco CallManager at least, on those nodes, on which auto registration is configured.

Now a little bit more information:

• Do not try to create a DN for auto registration manually in advance! CUCM creates them itself! Otherwise, you will get the same mistake on the phone screen that my colleague got.
• Auto registration can be configured on multiple servers, but! Only one group of servers may be available for auto registration. If the group consists of several servers with enabled auto registration, then keep in mind that an automatic transition to a different server will not be set if there's no range of DN-s. For example, there are two servers - cucm1 with DN server for auto registration from 1000 to 1049 and cucm2 with DN from 1050 to 1099. If cucm1 will be listed as the primary cucm server, then phones will be registered on it. Once the phone with DN 1049 will be registered, then next phone will receive Reject. To register phones on cucm2, they need to specify it as the primary cucm. Therefore it is better to configure the auto registration on the one server only.
• If SIP is specified in Enterprise Parameters, as the default protocol for auto registration, when you try to automatically register SCCP-phone, it will start to update firmware on SIP! And it also works backwards - if the SCCP is specified as the default protocol, the SIP phone will update firmware on SCCP. If the phone knows the one protocol only (for example, CP-9951 or DX650 knows SIP only), it will be registered on SIP, even if the default protocol is Skinny.
• If CUCM cluster is in the mixed mode (a mode that allows to include the encryption of voice traffic), then the auto registration will not work for security reasons.
• And finally, Cisco Systems recommends using the auto registration only if you need to add less than 100 phones; if you need to add more, then use Bulk Administration Tool. For security reasons it is also not recommended to keep auto registration always enabled, you should enable it only as needed, so Cisco says.

That's all. I hope this article will be useful. If you have any questions, please, write them in the comments.

• Part 1 – Prerequisites
• Part 2 – the VIS role
• Part 3 – CUCM Configuration

The CUCM configuration consists of two parts: creating a trunk to VCS Control and a trunk the VIS.

In CUCM proceed to CM Administration->System->Security->SIP Trunk Security Profile, select "Non Secure SIP Trunk Profile", and click Copy.

Enter the name for the new trunk, for example 'SIP Trunk Profile CUCM video', set the Incoming Port to 5065, check 'Accept unsolicited notification' and 'Accept replaces header', and click Save.

Now proceed to Device-> Device Settings-> SIP Profile and configure the Standard SIP Profile For Cisco VCS as the screenshot shows. Depending on the CUCM version this profile may have different parameters.

Create a partition for VCS Control: Call Routing->Class of Control->Partition

And for S4B:

Create a Calling Search Space: Call Routing->Class of Control->Calling Search Space:

Create a new trunk: Device -> Trunk. Replace 'CUCM IP' with your VCS Control IP address.

After you save the trunk settings, click Reset.
Then proceed to VCS Control Configuration->Zones to create a new zone.
Replace 'CUCMIP' with your CUCM IP address.

Save the form, create a Dial Plan for calling S4B users. In this case, the domain name suffix is used as a pattern. This rule is configured in such way that if a user user1@test.com is dialed, the CUCM Video trunk is used. You can use a regular expression as a pattern.

Configuration->Dial Plans-> Search Rules

The trunk between the CUCM and VCS Control is configured, now configure the trunk between the CUCM and S4B.

Create a new Calling Search Space: Call Routing->Class of Control->Calling Search Space

Create a security profile: SIP System->Security->SIP Trunk Security Profile

Create a SIP Profile: Device->Device Settings->SIP Profile

After you save the form, click Reset.
Create a trunk to VIS: Device->Trunk, replace 'VIS IP' with the VIS IP address

Create a pattern for sending calls to the S4B trunk. It is important to specify the IPv4 pattern: you should give the full domain name, for example, domain1.com, and also select the Route Partition and SIP Trunk.

Proceed to Call Routing->SIP Route Pattern and enter your SIP domain

That's it. It's time to test.

• Part 1 – Prerequisites
• Part 2 – the VIS role
• Part 3 – CUCM Configuration

This set of articles describes how to integrate Cisco MCU with Skype for Business and make available calls from MCU to S4B users. This is a guest post that we found very useful to our audience.

Let's start with the description of the infrastructure.

MCU1-MCUn – multipoint control units – the hardware to host video audio/video conferences. It is responsible for the connection and encoding. The connection means sending video/audio stream from one endpoint to all the others. Encoding means encoding and decoding video/audio stream between the endpoints.

E1-En – video endpoints: desk endpoints, room endpoint, IP phones, software clients.

VCS Control – provides video call and session control, endpoint registration, call routing. VCS stands for Video Control Server. This is a sip-server and a controller for H.323 zones. Also used for integration with third-party applications: IP PBX, Microsoft OCS, Lync 2010, Lync 2013 (an additional license is required). B2BUA for S4B support hasn't been announced yet.

VCS Express Way – server to connect with external video endpoints. It helps the remote clients to connect from outside the firewall.

CUCM – Call Manager – the Cisco IP PBX.

ME1 – Lync 2013 mediation server used for integration with third-party telephony.

Edge1 – Lync 2013 edge server used for connecting remote clients.

FE1 – front end Lync server used for registering clients, exchanging presence statuses and messages, creating audio and video conferences.

Ei and Li – Cisco and Lync clients respectively on the internet.

VCS Control supports B2BUA role for connecting to a Lync 2013 front end server, but the separate Microsoft Interoperability option key is required. It's also possible to install the Cisco CUCILync plug-in on the Lync clients, but in our case this won't be convenient, and separate licenses are also required.

In April 2015, Microsoft released the next Lync version called Skype for Business. It has the new Video Interop Server role that enables integrating third-party videoconferencing systems with S4B users. Jeff Schertz gives a very detailed description of the new topologies in his blog. Microsoft only supports integration with CUCM starting from the version 10.5, VCS Control support was not announced. MCU and Cisco Telepresence Server support wasn't announced as well, only calls from endpoints to S4B subscribers are supported. The endpoints should be registered on CUCM, and the MCU isn't actually used in this scenario. The list of endpoints is also very limited:

• Cisco TelePresence Codecs (C40, C60, C90)
• Cisco TelePresence MX Series (MX200, MX300)
• Cisco TelePresence EX Series (EX60, EX90)
• Cisco TelePresence SX Series (SX20)

In our case, the videoconferencing system is one of the crucial business software applications, and in addition to room endpoints we need other clients to connect to meetings.

We have decided not to upgrade all the Lync servers, but to upgrade the topology only and to add the VIS role. The Cisco-S4B topology would look as follows:

The only difference with the previous topology was the VIS role with the trunk to CUCM.

The basic integration aspects are the following:

• only the calls from MCU to S4B are supported, not the other way around,
• in Lync/S4B topology a separate server has to be deployed for the VIS,
• trusted certificates are not required,
• you won't be able to create a conference with an MCU participant on the S4B side; the meeting has to hosted on the MCU,
• an S4B user won't be able to share the desktop send documents,
• the CUCM version should be 10.5 or higher.

We suppose our readers don’t need to be told what Cisco Jabber is. This article is about the corporate environment it was introduced to, where it is extremely important to support TelePresence devices, work with VDI, let mobile users stay online, quickly create complicated meetings and conferences. And also to conform to a whole lot of rules and security policies.

In June 2015, Cisco Collaboration 11 was released, and Jabber, as a part of it, has changed for the better and acquired a lot of helpful features.

In brief:

• Support of all main desktop and mobile operating systems except Linux;
• Interface unification even for Cisco TelePresence devices;
• Safe access for mobile users (an encrypted channel is created after Jabber launches);
• A p2p analogue for VDI stations without excessive transfers from terminal to server;
• Simple guest access for browser-only users.

Now let’s go into more details.

Supported devices

Cisco made Jabber client suitable almost for every device: Windows, OS X, iOS, Android, and also browsers and VDI thin clients. There’s no Linux and Windows Phone support yet. This variety provides maximum accessibility for users and allows them to use their favorite device for communications.

Licensing for a heterogeneous device park becomes easier and more intelligible.

After transition from Personal Communicator (Jabber’s predecessor) to Jabber the application design has become almost identical on all devices. You can see the same interface even on TelePresence terminals (personal and group ones) which provide high communication quality. That helps users to adjust and start using the services.

The features available in Cisco Jabber for Windows are:

• Presence – real-time availability of employees within and outside the enterprise network
• Instant messaging – including p2p chat, group chat, chat rooms
• Phonebook – contains only AD and MS Outlook contacts, but you can integrate Cisco Jabber with CRM and any enterprise DB to make other contacts available in Jabber phonebook
• Desktop sharing – the whole screen, not the certain app only
• Conferencing – voice and web meetings
• Integrated video – with media escalation
• Security features – encryption, single sign-on, enterprise policy management

Mobile clients provide a bit less functionality.

UC outside the office

Only two or three years ago it wasn’t easy to provide mobile and remote employees with access to corporate communication services. There has always been a choice between using VPN connections and restricting functionality. It may seem easy to build a VPN tunnel, but in reality it created a lot of obstacles and issues: organizational (VPN and UC are usually managed by different departments), technical (encryption matters) and user problems. The last ones could nullify all the efforts to overcome the others, because it was simply unhandy to establish a VPN connection every time you leave your office. In any case, the solution turned out to be expensive and complicated.

After the Mobile and Remote Access (MRA) function was created, everything got better. The function name speaks for itself: it allows providing mobile and remote users with a secure access to UC services. The users can use Jabber anywhere without thinking about their location or changing any settings. All they have to do is to launch the application, and Jabber will find the required server by itself. What makes this function even more attractive is that it is completely free. Software for the edge servers that provide Jabber functioning outside a corporate network costs $0, and Jabber clients don’t require a license for internal calls. Perhaps this is one of the most useful and important functions, that is implemented today in every project where Jabber is used.

Guest access

In the end of 2013 Jabber Guest solution was introduced. It allows to simply send a link to a remote party or place it online. The remote subscriber can connect from a browser or a mobile device using this link, and an employee can use the accustomed means of communication. This function is very helpful for HR, communicating with natural persons and small companies, or recruiting experts from the outside.

Cloud and VDI

Jabber can be used together with Cisco WebEx cloud services and newly created Collaboration Meeting Rooms (CMR) Cloud service which also is based on WebEx Cloud. Joint usage of Jabber and these services provides an opportunity to use the client to take part in cloud arrangements and also initiate them with a single click. The last version of Jabber introduced an opportunity to start a WebEx or CMR session directly from the chat window without planning, inviting participants, etc.

There used to be a problem with VDI, because a workstation with an operating system would be situated in a data center, and the peripheral devices (monitor, web-camera, microphone, speakers) in the office would be connected to a thin client. This caused a significant delay in voice and video transmission to/from the data center. Standard video conferencing and UC applications can’t work in these circumstances. However, a special intermediate for Jabber, installed on a thin client, allows to overcome this limitation. Audio and video data is being transmitted directly between the thin client and a remote subscriber, avoiding the workstation in data center.

Cisco Unified RTMT (Real-Time Monitoring Tool) is used to monitor various CUCM parameters, Performance Counters, and to collect Traces.

Performance Counters contain simple information on the system and devices on the system, such as number of registered phones, number of active calls, number of available conference bridge resources etc.

RTMT requires a PC running Windows or Linux and uses HTTPS and TCP to monitor the Device Status, System Perfomance, Device discovery, CTI Applications in the CUCM cluster.

Not only the CUCM admin can work with RTMT: it’s enough to include any user in the standard Standard CCM Server Monitoring group.

RTMT offers a wide range of features but we won’t review all of them here. In everyday life we need a much smaller number of counters which we will list in the next section.

Useful Performance Counters

Counters	Path	Description
General information on the system: Memory Usage CPU Usage HDD Usage	System > System summary	These parameters give an overview of the system status
The processes running on the server	System > Server > Process	To understand which process causes the high CP load
General information on CUCM: Registered Phones Call InProgress Active MGCP Ports	Call Manager > Call Manager Summary	The abrupt changes of these parameters may be a sign of some problem
Call processing activity: Call Activity Gateway Activity Trunk Activity Sip Activity	Call manager > Call Process	These graphs give an idea about the total number of calls and the gateway activity
Device information: Phone Gateway Devices H.323 Devices Media Resources SIP Trunk	Call Manager > Device	The very useful information that provides and the detailed parameters of each device. For example – device models, firmware versions, IP addresses, user association etc.
Conference Bridge resources	Stand Alone Cluster -> node name -> Cisco HW Conference Bridge Device Stand Alone Cluster -> node name -> Cisco SW Conference Bridge Device HWConferenceActive – the number of active conferences ResourceActive – the number conference participants ResourceAvailable - the number of available resources	Gives an idea about the Conference Bridge activity, but not for each conference

Alerts

RTMT supports Alerts, which are triggered under certain conditions.

RTMT includes the set of pre-installed Alerts (System > Tools > Alert > Alert Central) which provide a great benefit in terms of a quick inspection of the system.

If you see "alarm" in this list it certainly makes sense to check what caused them.

In addition to pre-installed ones, you can create your own Custom Alerts.

Let's create the Alert, which is triggered when the resources of the Hardware Conference Bridge are over. This one is useful to monitor the availability of CUCM conferencing features Meet-Me, Ad-Hoc and Conference Now (implemented in CUCM 11).

We will use the ResourceAvailable Counter mentioned in the table above.

So, to create Alert:

1. Open the appropriate counter, select it, then right-click on it and choose Set Alert / Properties
2. Select the alarm level and enter its description
3. Set the triggering condition - "Under 1".
4. The next screen allows to set up the Alert frequency. In order to keep e-mail box from being hammered we will choose no more than once per hour.
5. Next step is to configure the e-mail address for notification
6. Do not forget to configure e-mail server properties:

Secured Meet-Me conferences

The Meet-Me conferencing functionality is provided by Cisco Unified Communications Manager in such a way that it's enough to dial-in to the conference to join it. Unless you deployed the latest CUCM v11 you can’t secure your Meet-Me conferences.

The "Conference" module is developed to close this gap protecting the Meet-Me conferences:

• by PIN – you need to know the PIN to join the meeting;
• or by Caller ID – your phone number must be included into the predefined group for you to join the meeting.

In addition the "Conference" module provides the meeting scheduler functionality (web-calendar or MS Outlook plugin) and conference control tool.

Locking Cisco IP phone

The "Lock" module restricts access to IP phone functions while his owner is away. The phone may be locked either manually or automatically when user's PC is locked or turned off. When locked, the IP phone:

• denies the access to IP phone services (personal phone directory, information services, calls history and the list of missed calls, recorded calls archive and etc.);
• allows only emergency calls to be made;
• forwards all incoming calls to user's mobile.

To unlock the IP phone its owner must unlock the PC or enter the PIN using the IP phone keypad.

Extension Mobility automatic authentication

Mobile employees working in different locations use the Extension Mobility functionality for authentication on IP phones and loading their profile (phone number, settings, etc.). To authenticate the employee has to enter his User ID and PIN, which he should always remember.

A more convenient way is to use automatic authentication - when an employee logs in to PC his Extension Mobility profile is loaded automatically to the IP phone associated with this working place. When an employee leaves this location the current configuration of the device is replaced by the logout profile.

Special call control features

The "Forced Connection" feature enables top managers to get in touch with an employee even when the employee's phone is busy. When activated, the current call is put on hold and the employee gets connected to his manager.

"Monitor", "Whisper", and "Barge" features are useful, for example, in the commercial department:

• the team supervisor can silently listen to agents' calls with customers or others to ensure the high quality of service or sales is delivered;
• using the "Whisper" feature he is able to privately coach the agent without the customer hearing what he says;
• when monitoring the call a supervisor may join it to switch the call to a conference, where all three parties can hear each other.