Aurus Blog

This blog is to share our expertise in Cisco UCM, UCCX/UCCE and Cisco Telepresence.

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The Updated Paging Solution for Cisco UCM Reaches Anyone, Anywhere

The new Aurus PhoneUP 3.11, the bundle of apps for Cisco UCM, is released this week! Congrats to the R&D team!

As for me, the main killer feature of the new version is the integration of its Paging module with our Outbound solution. Finally, we can offer a paging solution that reaches employees wherever they are.

See how it works:

1. When a new message (text, audio or hybrid) is sent to the predefined group of employees, it first goes to their Cisco IP phones. Every IP phone of the group plays the audio alert, shows the text message on its display and plays the audio message through the speakerphone.

2. Then the message waits for the employee to confirm the receipt with one of the following methods:

  • pressing the “Confirmed” soft button on the IP phone;
  • entering the PIN code on the IP phone;
  • recording the voice confirmation with the IP phone;

3. Those employees who have not acknowledged the receipt are sent to Aurus Outbound along with the message content.

4. Aurus Outbound starts the pre-configured “voice drop” outbound campaign to call each employee and play the audio message.

5. After listening to the message employees confirm the receipt by sending a DTMF code or recording the voice message.

6. Once the outbound campaign is over, the PhoneUP generates the combined report and sends it to the message originator by email.

There is a number of ways you can leverage such a notification system:

  • an intruder has been identified in the building;
  • severe weather is approaching;
  • production line stops;
  • emergency calls;
  • ad hoc conference calls notifications;
  • shift changes;
  • upcoming IT system shutdown.

You’re welcome to try our new products in your lab. We’ll provide the demo licenses, docs, install video guides and tech support.

Cisco Jabber vs Spark and Acano - The Steadfast Tin Soldier

For several years Jabber has been the Cisco's UC answer to Microsoft. A little bit late but still successful one. But in 2015 Cisco gave birth to two Jabber's cousins:

     1) in the beginning of 2015 Cisco announces Cisco Spark
     2) in the end of 2015 Cisco acquired Acano

Both Spark and Acano projects have its own clients for collaboration, so this is a fruitful theme for speculation on which product will supersede/swallow another one.

Cisco Spark vs Jabber

Spark is the real Cisco vision for collaboration that fits the Trollope's vision for, so called, Workstream Communications and Collaboration (WCC).

WCC is considered as a new form of communication that comes up to take the UC's place. WCC tools include asynchronous messaging, real-time voice and video, content and context. Most of leading vendors in enterprise communication world jumped in this WCC train almost simultaneously – Cisco (Spark), Interactive Intelligence (PureCloud), Mitel (MiTeam), Unify (Circuit), Avaya (Zang) just to name a few.

But is it going to replace Jabber? Nope. And the reasons are:

  • Cisco Jabber has a large customer base, which would be easier to continue to support rather than to transition to another app;
  • Spark is a pure cloud solution which may not appeal to some verticals where everything has to be on premise;
  • the UC transition to WCC is not going to be a quick one, and until it isn't over Cisco still needs to compete with Microsoft.

So, I think for the next several years there will still be two clients – Jabber for traditional presence bubbles, buddy list, UC integration and Spark for team collaboration, workflow, persistence etc.

Currently there are business cases for both solutions even to be running simultaneously at the same company.

Acano vs Cisco Jabber

The Steadfast Tin SoldierAcano, acquired by Cisco in Nov 2015, is the best known of its truly interoperable video/audio bridge. On my opinion, this bridge was the primary Cisco's target and Acano is going to replace Cisco Telepresence Server and Conductor in Cisco's business video offering.

But besides the server component Acano provides the client app with contacts (no buddy list though), presence, very good persistent group chats, workflow and some other useful things. And that makes it another threat for Cisco Jabber.

Still I think Jabber will stand, because the primary task will be to digest the Acano's bridge, not the client. For example:

  • Cisco Telepresence Server replacement;
  • integration with Cisco Spark to power it with interop video meetings;
  • probably, integration with Cisco Webex.

So, my prediction is we'll live with Cisco Jabber and Spark and Acano at least until the next decade. Then all the platforms will be merged as well as client applications.

Co-Browsing in the Contact Center – the Details Matter

RichCall - video chat and live support for contact centersThe 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).

Video in Contact Center – Why? For Whom? How?

According to the "Global Contact Center Benchmarking Report 2015" by Dimension Data the number of video-enabled contact centers is going to rise threefold in 2016.

Who adds video-channel to the contact center and why? Guys from Aurus, a software vendor offering video chat solution for contact centers, shared with me some info about their customers.

The first question they ask everyone asking to try their software is "Why?". So, the stats is:

Now, in details…

Video to improve the company's image

One of ten companies interested in video channel is going to enable video call service on their website as a part of the company's image building. By doing this they send messages like:

  • "We're a modern company and we follow the trends in the client servicing. So you can be sure our products are also built with top-notch technologies."
  • "We're rich enough to equip our contact center with special workplaces and hire good-looking contact center agents. So you can trust us."

Sometime video call is only available for VIP clients and then the message is "We really value you and your money".

These companies pay a lot of attention to the client interface - high-quality video of the agent, branded UI and so on.

Video to higher sales

RichCall - video chat and live support for contact centers This is where video becomes optional and may be replaced with a good photo of the agent, but the features that come to the fore are:

  • co-browsing – to help the client find the right info on the website and assist him with the payment process,
  • pushing product images and videos – to convince the customer to buy,
  • text chat – to provide the client with product spec, shop addresses, agreement templates etc.

The important note is that this approach only works when the products are quite unique and the company is ready to spend as much time as required to get a new client. For example, you can talk as much as needed about your jewelry or sofa or car to persuade the client to buy. But you cannot afford that if you're, say, a travel agency – you won't spend half an hour talking about your offer to Bora Bora, knowing that after you finish the customer will start surfing the net for the cheapest option.

Video for better support

Yeh, this is the leader of "live assist" use cases. The video is not required at all and web-collaboration features become critical ones:

  • text chat supporting images and files,
  • co-browsing that works in those spaces that require user authentication,
  • app sharing with annotations,
  • screen snap shots,
  • mobile SDK (remember the Amazon MayDay hype?).

Providing remote experts with the web-collaboration tools may significantly improve the time to resolution indicator.

Another important feature is the ability to add the web-collaboration session on the fly to any phone initiated customer support call. This increases the first call resolution rate.

Cisco UCCX Wallboard/Dashboard Vendor List

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.

2Ring DASHBOARDS & WALLBOARD

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.

Comstice Wallboard

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

Inova Contact Center Digital Signage

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!

Atea Systems Wallboard

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.

CcInfo Wallboard

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.

Tenox Wallboard

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.

Inova Solutions Wallboard

Configuring Cisco Jabber 11 for iOS and Android Mobile Devices

If we have Cisco IM and Presence server and Cisco UCM in our corporate environment, as well as unhandy wi-fi Cisco 7925 telephones, which are heavy and consume the battery as fast as a Formula One car, then sooner or later we’ll think about switching to Cisco Jabber on a mobile phone.

This article tells what you need for that.

Before all experiments, make sure you have the following things:

  • Cisco Unified Communications Manager 8.6.2 or higher (preferably the latest version)
  • Cisco IM and Presence (integrated with CUCM, of course)
  • Wi-Fi Wireless Access Point, already set up by an administrator to distribute wireless internet (don’t mix it up with Wi-Fi Router, since if you have Router, your phones will be hidden behind NAT and RTP streams and it will be complicated to route correctly)
  • CUCM и IM&Presence Administrator kind enough to make us a CUCM and Presence user
  • Android 4.x or higher, better and faster (iPhone will do as well)

You can discard Cisco IM and Presence and configure Cisco Jabber for CUCM phone services only. It will be a dull and sad client (as an alternative to Jabber: what can possibly prevent you from registering an Android phone as a SIP third-party device on CUCM?), but you’ll be able to call anyway.

Proceed to Google market (or Apple Store for iPhone)

https://play.google.com/store/apps/details?id=com.cisco.im

And install the app.

But we won’t be able to run it until we perform the configuration steps in our Unified Communication infrastructure. So, led by the craving for launching Cisco Jabber on our phone, we go to the CUCM web interface and add our device.

Use the menu: Device - Phone - Add new – select "Cisco Dual Mode for Android"

Now we configure our precious, paying attention to the following aspects:

Device Name – in case of Android device it should start with BOT prefix (TCT in case of iPhone) and include name in upper case (BOT <NAME>). Allowed characters: a–z, A–Z, 0–9, (.), (_), (-). Total name length is limited to 12 characters. My device name is BOT-ATYRIN.

Description – specify or shyly conceal the phone model (or write any kind of nonsense)

Media Resource Group List – either specified explicitly or assigned through the Device Pool

Optionally specify User Hold MOH Audio Source и Network Hold MOH Audio Source

Owner – select User

Owner User ID – select yourself from the list of CUCM users

Device Security ProfileCisco Dual Mode for Android – Standard SIP Non-Security Profile

SIP ProfileStandard SIP Profile for Mobile Device

If necessary, in Product Specific Configuration Layout section you can turn video support on, and specify a list of SSID Wi-Fi access point names, separated with ( /), if you wish to connect to the specified access points only.

After that we can add a line (DN) to our device.


Associate yourself with this line.

Take notice that since Cisco Jabber 11.0.1 and CUCM 10.5(2)su2 versions you can use conversation recording and listening features!

Quoting the documentation: Silent Monitoring and Call Recording (Built-in Bridge) — In 11.0.1 Cisco Jabber for Android supports silent monitoring and call recording using Cisco Unified Communications Manager 10.5(2) su2 and later releases.

After that proceed to User Management – End User and give yourself the rights to use this device (Device Association)

Don’t forget to ensure that you are an IM and Presence user:

And have the necessary rights:

Last but not least, a bit of Cisco magic without which you may not be able to run anything:

Jabber on Android doesn’t support HOSTNAME in Android kernel before version 4.4.4, and it’s possible that the integration with Call Manager phone services won’t happen. So, you’ll only see chat and presence features.

To solve this problem, first of all it’s necessary to specify FQDN or IP address everywhere in Jabber settings.

Secondly, in System – Enterprise Parameters menu in CUCM you should fill the initially empty Organization Top Level Domain field (initially empty) with the enterprise domain value.

Now cross your fingers and pray to Cisco gods as we proceed to the most exciting part: the first launch and configuration of Cisco Jabber on your phone.

It launches, which is good enough. Let’s open “Advanced settings” and presume that we are clever enough to explore them. Fill out IM and Presence server address and press OK:

Fill out your username and password (End User credentials in CUCM, IM and Presence). If you haven’t made any mistakes, you may rejoice and shed tears of happiness:

Now you can use mobile Jabber:

All Possible Ways to Set Up PIN for CUCM Meet-Me Conference

  • "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.

Configuring Cisco Jabber for Android / iOS with a shared line

This article explains how to configure a shared line for Cisco IP phone and Cisco Jabber for mobile (Android or iOS). This will allow your employees to use their gadgets (BYOD) to receive calls made to their Cisco IP phones and make outgoing calls (the smartphone with Cisco Jabber for iOS / Android application installed must be connected to the enterprise Wi-Fi).

Note that this will require additional DLUs (for CUCM 8.x) or enhanced plus licenses (for CUCM 9.x and higher).

Cisco Jabber is supported in CUCM 8.6 and higher.

First, you will need the appropriate COP file (Cisco Options Package) for your gadget.

You can download it from the Cisco website https://software.cisco.com/download/navigator.html?mdfid=278875338&flowid=45928 for the required version of CUCM.

The step by step procedure of CUCM configuration is:

  • download the required COP file and put it on an FTP or SFTP server that is accessible from your CUCM servers,
  • sign in to CUCM Administration page,
  • go to System > Service Parameters,
  • choose your server,
  • select Cisco CallManager (Active),
  • scroll to the Clusterwide Parameters (System - Mobility) section,
  • increase the SIP Dual Mode Alert Timer value to 4500 milliseconds,
  • click Save.

If after increasing the SIP Dual Mode Alert Timer value, incoming calls are still terminated you can increase the SIP Dual Mode Alert Timer value. 4500ms is the lowest recommended value.

1. Create the appropriate SIP Profile

Device > Device Settings > SIP Profile

Create a new SIP profile or copy an existing SIP profile. Enter a suitable name for the new profile, for example, Jabber SIP Profile.

Scroll down and set the following values in the new SIP profile:

  • Timer Register Delta to 30
  • Timer Register Expires to 660
  • Timer Keep Alive Expires to 660
  • Timer Subscribe Expires to 660
  • Timer Subscribe Delta to 15

Save settings.

2. Add the user device

Verify that the device pool to which the Jabber device will be added is associated with a region that includes the support for the G.711 codec.

Follow these steps:

  • Sign in to Cisco Unified Communications Manager Administration.
  • Go to Device > Phone.
  • Click Add New.
  • From the Phone Type drop-down list, select Cisco Dual Mode for iPhone (Cisco Dual Mode for Android - for Samsung).

Click Next.

Enter the parameters of Device-Specific Information:

Enter the DeviceName. The Device Name:

  • For iPhone name must start with TCT.
  • For Android name must start with BOT.
  • Must be in upper case.
  • Can contain up to 15 characters.
  • Can include only the following characters: A to Z, 0 to 9, hyphens (-) or underscore (_).

Select Standard Dual Mode for iPhone (for iPhone) in the Phone Button Template field.

Select Standard Dual Mode for Android (for Android) in the Phone Button Template field.

Configure the following settings in order to prevent confusion for the person the user calls:

  • Media Resource Group List
  • User Hold MOH Audio Source
  • Network Hold MOH Audio Source

Select the desk phone as the primary phone, if the user has a desk phone.

Enter the parameters for Protocol Specific Information, as described below:

In the Device Security Profile drop-down list select Cisco Dual Mode for iPhone - Standard SIP Non-Secure Profile. (Or the same for Android).

In the SIP profile drop-down list select the SIP profile you just created in the Create dedicated SIP profile section.

Note: Select the SIP profile for all Cisco Dual-Mode devices that are running Jabber.

Click Save.

Click Apply Config.

Click [Line n] - Add a new DN.

Enter the phone number (DN) of this device.

Note: This can be a new DN. In this case, a desk phone with the same DN is not required.

Click Save.

Go to the end user page for the user.

Associate Standard Dual Mode device for iPhone (or for Android) that you just created for this user.

Click Save.

Querying CUCM Database from the Command Line

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’s Conferencing Options Explained

With a wide range of conferencing products offered by Cisco it may be hard to figure out what are the options and what each of them is designed for. This post is going to help you to get a sense of what is happening in the Cisco’s conferencing world.

Audio Conferencing

Cisco UCM has several conferencing features on-board:

  • Ad-Hoc - to escalate the current phone call to audio-conference,
  • Meet-Me – for permanent conferences,
  • ConferenceNow (introduced in CUCM 11) – for personal conference rooms.

All of them are audio only and use either the software conference bridge (CUCM service) or hardware conferencing resources (most often – DSP modules in Cisco ISRs). Hardware DSP modules are required to enable the transcoding feature as the software bridge only supports G711.

Since CUCM is not a conferencing platform, these three options only provide basic conferencing features, but may be enhanced with 3rd party add-ons to CUCM. For example, Aurus PhoneUP operates on top of the CUCM conferencing engine providing extra functionality like:

  • meeting scheduler,
  • PIN and Caller ID security,
  • conference control tools.

But still only audio conferencing is available. If you meetings require web collaboration and/or video you have to deploy additional Cisco products.

Web-conferencing and Collaboration

Cisco’s web-conferencing portfolio is based on the Cisco WebEx platform that provides audio/video conferencing as well as web-collaboration tools – white board, presentations, application sharing, chat and so on.

You can join the WebEx meeting over IP (PC or mobile devices) as well as by dialing the call-in number from any phone.

The WebEx platform is available in cloud (Cisco WebEx Meeting Center) or on-premises (Cisco WebEx Meeting Server).

Similar to Cisco UCM conferencing options, WebEx meetings can be:

  • started instantly,
  • scheduled,
  • always available (Cisco Collaboration Meeting Rooms).

HD-Video Conferencing

Finally, for the best user experience Cisco offers Cisco TelePresence architecture, which provides high-quality HD video conferencing.

The core components of Cisco TelePresence offer are:

  • video-endpoints - desktop, room and immersive,
  • video-conferencing bridges - Cisco TelePresence Serverand obsolete Cisco TelePresence MCU,
  • Cisco Telepresence Conductor - orchestrates the allocation of conferencing resources,
  • Cisco TelePresence Management Suite (TMS) – manages the Cisco TelePresence infrastructure providing engineers with provisioning, meeting control, resource management and meeting scheduling features.

There are also several optional components available from both Cisco and its technology partners like:

  • Cisco TelePresence Recording Server – to record conferences,
  • Cisco Expressway – to allow users outside the firewall to join the meetings,
  • Aurus U-Meet – to improve the meeting scheduling and conference control.

How to Extend CUCM (CallManager) Features

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 directoriesOneUnlimited number with access control for groups of employees
Integration with external systemsADAD, LDAP, IBM Lotus Notes, CSV, XML, CUCM, SQL database
External contacts supportNoYes
Caller IDInternal numbers onlyAny phone number
Incoming call detailsName / Last Name / CompanyFlexible data that may contain an employee photo
Notification of missed calls via e-mailOnly with UnityYes
DTMF support NoYes
Personal directoriesOne (edited manually or imported from Outlook)Unlimited number of directories synchronized with external sources
Auto-redial featureNoYes


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.

Predictive vs Progressive – Outbound Dialers Test Results

We are now beta testing our brand new math algorithm for predictive dialer to be used in Aurus Outbound product. The algorithm was recently developed by our engineers in collaboration with the Institute of Mathematics and the first test results make a killer impression.

Here are some figures…

We performed 3 synthetic tests with the following input parameters:

  • client’s time to answer: varies from 15s to 25s;
  • The success rate (percentage of calls answered by live person): 80%;
  • the number of agents involved in the campaign: 50;
  • the campaign duration: 58 min;
  • call duration range:
    • test 1: 40-60s – to emulate campaigns with short call time, for example address verification, appointment reminders etc,
    • test 2: 60-120s – for campaigns like subscription renewals, mass mailing follow-ups, “welcome calls” and so on,
    • test 3: 500-600s – for sales campaigns, surveys, market research and so on.

Agent Occupancy

The agent occupancy calculated in the tests above:

Predictive vs Progressive – Agent Occupancy

As expected, the agent occupancy in progressive campaigns depends on the average call duration – the more the talk time is the less failed calls influence the total agent utilization. However, when using predictive dialer the agent occupancy is over 90% in all tests.

Abandonment Rate

Another important metric is the call abandonment rate which indicates the percentage of “nuisance” calls – the ones where the client answers but no agent is immediately available to talk to him.

In our tests the abandonment rates are:

Predictive Dialer Abandonment Rates

The highest average abandonment rate is achieved in the 3rd test because of the highest (500-600 sec) and the most spread out (100 sec) call duration.

Having high call duration variance we may get too high abandonment rate which is unacceptable due to regulations like Ofcom and FCC (Federal Communications Commission). This can be illustrated by 2 others tests:

  • the call duration is 400-600 (variance - 200s)
  • the call duration is 400-700 (variance - 300s).
Predictive Dialer Abandonment Rates (high variance)

To avoid this our new predictive algorithm allows to setup the max abandoned rate input parameter. So, when the abandonment rate of the active campaign gets closer to the max value specified, the algorithm reduces the number of calls.

Predictive Dialer in Low-Volume Campaigns

It has been argued that predictive approach is only effective when used in high-volume (30+ agents) and long-lasting campaigns to get enough statistics for proper prediction.

We performed a couple of tests with low-volume campaign parameters and higher variance:

  • the campaign duration: 10 min (in opposite to 58 min in the tests above);
  • the number of agents: 10;
  • call duration:
    • test 1: 60-120 sec
    • test 2: 60-180 sec
Predictive vs Progressive in Low-Volume Campaigns

As we can see the effectiveness of predictive algorithm in low-volume campaigns reduced by 10%, but still it provides extra 10% of agent utilization when comparing to progressive approach.

The new predictive algorithm will be included into the nearest release of Aurus Outbound (Jan 2016), so you’re welcome to try it.

Phone Auto Registration on CUCM

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.

Implementing Unified Enterprise Directory in Heterogeneous UC Environment

Despite the fact that PhoneUP is designed for Cisco UCM its “Directory” module can be used to provide the unified enterprise phone directory with Caller ID support in heterogeneous multivendor communication infrastructure.

Here is the case study.

The holding with 15+ companies has a multivendor enterprise communications network with IP PBXs of different vendors (Cisco UCM, CME, Siemens, Asterisk etc) connected with SIP trunks:

Managing 15+ local directories were too labour-intensive and did not provide the unified contacts directory available for every employee of the holding.

The solution implemented includes:

  • Cisco Unified Communication Manager which proxies all the calls of the communications network;
  • PhoneUP “Directory” app integrated with new CUCM and all sources of employee details.

So, what are the benefits achieved?

1. Unified always up to date enterprise directory available for everyone.
PhoneUP Directory is synchronized with the employee database of each company. A set of built-in connectors allowed to integrate it with various datasources like AD, CUCM, HR software etc. The public web-interface is available for any employee and provides sort/filter/group features for fast contact search.

2. Caller ID for any IP endpoint.
Integrated with CUCM PhoneUP Directory supplements each call with the Caller ID string which is displayed on any SIP-endpoint with no matter to IP PBX that receives the call.

If you want to know more tech details about the project, contact us and we’ll share our experience with you.

Guide to Integrate Cisco MCU with Skype for Business. 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.

SIP Trunk Security Profile Configuration

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.

SIP Trunk Security Profile Configuration

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.

Standard SIP Profile For Cisco VCS

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

Partition Information

And for S4B:

Partition Information

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

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

Calling Search Space

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

SIP Trunk Security Profile

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

SIP Profile

SIP Profile

SIP Profile

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

Device Information

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.

Guide to Integrate Cisco MCU with Skype for Business. Part 2 – the VIS role


You'll need a separate server to setup the VIS role. You can use either a VM or a physical server, depending on how many calls you are planning to handle simultaneously.

Launch the setup from the disc image: \Setup\amd64\setup.exe
Once the prerequisites have been installed, the Skype for Business Server Deployment Wizard will be launched. You should select the Install Administrative Tools option.

After the installation you should launch the Skype for Business Topology Builder and download the current topology:

Topology Builder

A new section called Skype for Business 2015 will appear in the Topology Builder. You should proceed to the Video Interop Server pools folder and define a new pool:

On the first screen, enter the domain name of the VIS or the pool (if necessary):

Create a new Video Interop Server pool

Select the Front End server:

Create a new Video Interop Server pool

On the next screen, select the Edge server. Then the trunk configuration wizard will be launched.
Enter the CUCM IP address or FQDN:

Define new Video gateway

If the VIS uses several IP addresses, you can choose a specific one:

Define new Video gateway

On the next screen, set the Listening port to 5060. Keep the TLS protocol (it will be changed later):

Define new Video gateway

In properties of the created VIS, enable TCP protocol:

Edit Properties

And then select the TCP protocol in the VIS properties:

Edit Properties

Publish the topology.

After the topology was successfully published, install the Local Configuration Store, VIS role, request and install the certificates and launch the services. I won't describe these steps in details; they don't have any parameters to configure. After the services are launched, open PowerShell and enter the following command with the trunk name changed:

New-CsVideoTrunkConfiguration -Identity "Service:VideoGateway:trunk name" -GatewaySendsRtcpForActiveCalls $false -GatewaySendsRtcpForCallsOnHold $false -EnableMediaEncryptionForSipOverTls $false

Now the VIS configuration is over. The TechNet guide from Microsoft suggests creating a Dial Plan and normalization rules. This is necessary for E.164 calls, but I'm going to call using the SIP Address.

Guide to Integrate Cisco MCU with Skype for Business. Part 1 – Prerequisites


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.

CUCM On-Board Conferencing Options Overview and Restrictions

Cisco Unified Communications Manager has several native conferencing options. All of them are quite simple, “getting started” implementations. If you need more, you have to switch to one of full-featured conferencing platform described, for example, here, on Cisco’s website.

But in this post we’re going to review only on-board options shipped with CUCM.

Ad-Hoc Conference

Ad-hoc (also referred to as “instant”) conference is an impromptu conference that is not scheduled before. A point-to-point call may be escalated to an ad-hoc conference using Cisco IP phone, Cisco Jabber, or some 3rd party CTI application like attendant console.

The originator of the conference may add / remove participants, no other conference control features are available.

Meet-Me Conference

Meet-Me (also referred to as “permanent”) conferencing suggests that a range of directory numbers are allocated for exclusive use of the conference. The meet-me conference begins when the host connects. After that anyone who calls the conference number joins the conference.

Limitations:

  • the host must use a Cisco endpoint to start the conference,
  • no scheduling tool is available,
  • no authentication is available ,
  • no conference control options.

Conference Now Feature

The Conference Now feature is available on CUCM 11 and higher and is going to replace the Meet-Me option. It allows user to create their personal meeting rooms (with DN associated) protected with PIN. Anyone who calls the host’s meeting number is asked for the PIN to join the conference. The conference starts when the host joins the meeting, till that everyone receive MOH (Music On Hold) provided by basic IVR implemented in CUCM 11.

Limitations:

  • no scheduling tool,
  • the PIN is managed by host and is not generated automatically,
  • no participants list is available for the host,
  • no conference control options.

Conference Bridges

All CUCM conferencing options use the conference bridge configured in CUCM.

Software conference bridge

The CUCM software conference bridge is available out of the box and only supports G.711 codec (ALaw & ULaw). If there is a codec mismatch between the calling device and the software conference bridge, a transcoder is needed.

The software conference bridge can handle up to 128 audio streams (48 if the Cisco IP Voice Media Streaming Application service runs on the same server as the Cisco CallManager service) and supports max 100 conferences per CUCM server.

Hardware conference bridges

To obtain the transcoding feature and increase the capacity of conferencing resources you need to switch to hardware conference bridge.

For example, Cisco 1700, Cisco 2600, Cisco 2600XM, Cisco 2800, Cisco 3600, Cisco 3700, and Cisco 3800 series voice gateway routers (DSP modules are required) provide conferencing and transcoding capabilities for Cisco Unified Communications Manager.

Cisco Jabber 11 – “And I admit, it's getting better…”

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 IP Phones End-of-Sale Matrix

For the past several years Cisco experimented a lot with its IP phone portfolio. There have been several IP phone series launched after the good old 7900 and some of them only lived for a couple of years.

BTW, even in the movies of 2015 Cisco still presents 7900 (not 7800 or 8800) IP phone, check "Spy" by Paul Feig.

So It was quite difficult to follow Cisco’s brave marketing guys and to puzzle out what to sell and what is too late to sell :-). It is finally settled. Cisco’s IP phone portfolio is:

  • 7800 series – cost-effective devices for common purposes,
  • 8800 series – HD video phones for the top brass.

I have not listed:

  • 3905 – I doubt that this low-cost SIP phone without XML-services, JTAPI and AXL-control is a truly Cisco product,
  • Cisco 8945, 9951, 9971 - the end of sale announcement has not arrived yet, but I think it is a matter of several months (you're welcome to argue with me:),
  • DX650 – though this device is a phone-looking one, still it’s a representative of the DX series that belongs to the collaboration species.

Forget all other IP phones, they're no longer available:

Cisco IP Phone ModelEnd-of-Sale Date
Cisco Unified SIP Phone 3900 Series
Cisco Unified SIP Phone 3905None Announced
Cisco Unified SIP Phone 3911July 23, 2010
Cisco Unified SIP Phone 3951July 23, 2010
Cisco Unified IP Phone 6900 Series
Cisco Unified IP Phone 6911July 30, 2014
Cisco Unified IP Phone 6921July 30, 2014
Cisco Unified IP Phone 6941July 30, 2014
Cisco Unified IP Phone 6945July 30, 2014
Cisco Unified IP Phone 6961July 30, 2014
Cisco Unified IP Phone 7900 Series
Cisco Unified IP Phones 7902November 29, 2006
Cisco Unified IP Phones 7905May 22, 2006
Cisco Unified IP Phones 7906July 23, 2010
Cisco Unified IP Phones 7911February 6, 2012
Cisco Unified IP Phones 7912May 27, 2007
Cisco Unified IP Phones 7914March 31, 2009
Cisco Unified IP Phones 7915February 1, 2016
Cisco Unified IP Phones 7920June 14, 2007
Cisco Unified IP Phones 7921January 30, 2012
Cisco Unified IP Phones 7926February 1, 2016
Cisco Unified IP Phones 7931July 30, 2014
Cisco Unified IP Phones 7935November 14, 2004
Cisco Unified IP Phones 7936July 23, 2010
Cisco Unified IP Phones 7937March 31, 2014
Cisco Unified IP Phones 7940July 23, 2010
Cisco Unified IP Phones 7941January 19, 2010
Cisco Unified IP Phones 7942February 1, 2016
Cisco Unified IP Phones 7960July 23, 2010
Cisco Unified IP Phones 7961August 1, 2008
Cisco Unified IP Phones 7962February 1, 2016
Cisco Unified IP Phones 7970August 1, 2008
Cisco Unified IP Phones 7971August 1, 2008
Cisco Unified IP Phones 7985September 24, 2010
Cisco Unified IP Phone 8900 Series
Cisco Unified IP Phone 8941May 31, 2014
Cisco Unified IP Phone 8945None Announced
Cisco Unified IP Phone 8961July 9, 2015
Cisco Unified IP Phone 9900 Series
Cisco Unified IP Phone 9951None Announced
Cisco Unified IP Phone 9971None Announced

Working with Cisco Unified RTMT

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

CountersPathDescription

General information on the system:

  • Memory Usage
  • CPU Usage
  • HDD Usage

System > System summaryThese parameters give an overview of the system status
The processes running on the serverSystem > Server > ProcessTo understand which process causes the high CP load

General information on CUCM:

  • Registered Phones
  • Call InProgress
  • Active MGCP Ports

Call Manager > Call Manager SummaryThe 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 ProcessThese 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:
System > Tools > Alert > Config Email Server

That’s it.

Syslog Viewer

Syslog Viewer is the analogue of Windows Event Viewer. If something is wrong with the system it’s one of the first interface to look at.

System > Tools > Syslog Viewer

Syslog Viewer allows you to view the messages from the following logs:

  • System Logs - everything that concerns hardware and OS.
  • Application Logs – CUCM logs
  • Security Logs - user login attempts.

Trace & Log Central

Trace & Log Central collects and displays various traces and log files:

  • CUCM SDL and SDI traces
  • SDI (System Diagnostic Interface) are used for log analysis
  • SDL (Signal Distribution Layer) are mainly used when opening cases in the Cisco Technical Assistance Center (TAC).
  • CUCM application logs (for example, BAT logs)
  • System logs

Trace & Log Central works in the following modes:

  • Remote Browse - displays files directly from remote servers.
  • Collect Files – collects traces and downloads them to the PC with RTMT installed.
  • Query wizard – work with trace files containing the query string.
  • Schedule Collection - scheduled trace collection
  • Local Browse - view traces collected on the local drive
  • Collect Erash Dump
  • Real time Trace - trace view in the real-time

The Trace & Log Central can collect tons of data. You can use the following tools to simplify the SDI files analysis.

Performance Monitor and Data Logging

As already mentioned, Performance monitor contains a lot of different counters.

We have already learned to configure Alert to be triggered on the certain Counter by the right-clicking on the counter.

In addition, we can setup the counter logging – right click on the counter and select

  • Start Counter Logging

Once the logging is configured you’ll be able to view logs with Performance Log Viewer.


How We Improved Our CUCM (CallManager) Phone Directory

Hi, I’m Kirill Basikhin, international key account manager from Aurus. Here, at Aurus, we develop a bundle of applications for Cisco Unified Communications Manager. And, yes, we do use our products, because they help us work faster, smarter and easily.

This post is not going to be a promotional one, I’ll just describe the way we setup the enterprise CUCM directory with our product. Each paragraph below is a real-life daily use-case.

The important feature used in all cases below is that our phone directory is integrated with CRM, so every contact I create in CRM automatically appears in the enterprise CUCM directory with phone numbers, manager (me or one of my colleagues) name, city, products purchased (or interested in).

Incoming Calls from my Clients Get Routed Directly to Me

We’ve got UCCX installed that receives any incoming call. A simple UCCX script sends the caller’s phone number to the CUCM directory software (PhoneUP Directory). The phone directory searches its database for the client number provided and sends the response to UCCX containing the phone number of manager (me), responsible for this client. UCCX then just forwards the call to me.

So, every time my client calls Aurus he reaches me automatically. Cool? Yep!

What if the Caller is Unknown?

If UCCX does not receive the phone number to transfer the call to, it routes it to the reception (actually one of the marketing managers, we’re not a huge company). The girl receiving the incoming call gets a CAD (yeh, we still use the old-school Cisco Agent Desktop) popup with the list of employees to whom calls from the calling party were transferred most often (PhoneUP Operator).

So when my aunt from Germany calls Aurus again (she doesn’t call my mobile, its expensive) the girl receiving the call gets a list with my name only (cause my aunt called previously and asked for me). Then she just clicks my name to forward the call.

How do I Handle Calls from my Clients

When the call reaches me, my Cisco IP phone shows the client name, the city and the product(s) purchased (remember, the CUCM phone directory is synchronized with our CRM).

So I can greet the client, saying “Hi John, Kirill speaking. I can’t believe you finally tested PhoneUP!”.

After all I’m a sales man and every smile on my client’s face will finally turn to another penny in my pocket.

How do I Call my Client

Our Cisco UCM phone directory provides the search interface on Cisco IP phone, but typing the client name using the IP phone keypad makes me wanna die (though several my clients do use it!). I usually make calls in 2 ways:

  • when I work with client card in CRM, I just click the client’s phone number to call him; this is the simple click2call feature but what is REALLY useful is that our click2call supports DTMF – even if the phone number is stored unformatted (like “+1 (408) 526-7209 ext.. 4576”) the CUCM directory software calls the PSTN number, waits for the answer and then dials the DTMF;
  • when I just want to call some person, I push “Ctrl-Q” to activate the “PhoneUP Agent” and use it to search the contact and make a call, the DTMF also works.

My colleagues use Cisco Jabber and its phonebook is also integrated with CRM, but I’m old and I use the old-fashioned IM client.

How do I Lunch


When I’m quitting for the lunch, I just lock my PC. What happens then is that my Cisco IP phone automatically locks its keypad (PhoneUP Lock feature) and activates the “Forward All” mode.

So, all calls from my clients are forwarded to my mobile.

When back to my workplace I unlock my PC and my Cisco IP phone goes to the normal mode.

How do I Manage Missed Calls

At the end of working day I do not setup the Forward All, cause most of my clients do not bother themselves about the time shift.

When I’m back to the office the next day, my Cisco IP phone shows the list of missed calls (XML-service provided by PhoneUP Directory).

What is important, this list contains names of clients called. So I know who called me and I can check the CRM before calling back.

How to Improve the Enterprise Security with Cisco UC Applications

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.

Cisco IP Phone: Lacking Features

Employee, for the first time getting Cisco IP Phone, discovers all the delights of the enterprise IP telephony described in numerous articles by Cisco Systems and its partners.

Indeed, the high call quality, audio / video conferencing tools and other collaboration services (voice mail, presence indication) make enterprise communications more convenient and efficient.

However, your Cisco IP phone can be supplemented with additional features Cisco doesn't provide, which will increase the ROI of your new enterprise communications network.

Let's start.

Phone directory and Caller ID info

The alternative solution to the out of the box CUCM phone directory allows you to synchronize all company contacts (employees, clients, partners and so on) into the one, always up to date enterprise directory, that provides users with fast search, Caller ID, click to call, call control and other features.

When receiving an incoming call the display of your Cisco IP phone shows the detailed Caller ID info even when the call has come from mobile, PSTN etc.

The list of missed calls contains not just phone numbers, but also the caller info – contact or company name. This allows user to identify the really important missed calls and call them back on time.

These features of Cisco IP phone are provided by the "Directory" module of the PhoneUP bundle.

Cisco IP phone for enterprise paging

The "Paging" module allows using Cisco IP phones for group paging to enterprise employees.

The ability to send text and audio messages to the group of Cisco IP phones allows you to use your IP telephony network for employee notifications. Unlike email and IM, the message sent to IP phone won't be overlooked due to the sound alarm played by the IP phone.

Cisco IP phones can also be used for emergency announcements. With just one button you can put all phone calls on hold and broadcast your message through speakerphones with the max volume.

Pre-recorded text and audio messages can be sent manually or on schedule. The paging may also be triggered by a third-party system (for example alarm system, manufacturing process monitoring tool etc) no notify the group of employees about the incident occurred.

Special call control features

The "Priority" module of the PhoneUP suite provides several special features for managers.

The Forced Connection feature allows top manager to contact any employee even if his phone line is busy – the current call is put on hold and the employee gets connected to the manager.

Silent Monitoring is another useful feature, which helps evaluating the quality of customer service. The supervisors' interface shows the client names sales managers are talking to and allows to listen to any conversation.

Using the Whisper feature the supervisor is able to connect to any call and talk to the agent without making the client aware of the supervisors' presence. With the Barge feature the supervisor is able to add himself to the sales managers' call.

Images from video cameras on the Cisco IP phone display

Even if your Cisco IP phone doesn't support video, its display can still show video frames grabbed from the camera. For example, with just one push of a button the security officer will see who is at the door before opening it, any employee will access the video from the parking lot, the technologist will monitor the manufacturing area from anywhere in the enterprise, etc.

Frequently used info on the Cisco IP phone display

Cisco IP phones are able to display any data from the enterprise software in the real-time mode. For example:

  • exchange rates – for bank cahier,
  • KPI – for top managers,
  • agent performance data – for contact center supervisor.

The benefit of using the IP phone for this purpose is that you only need to push one button to see the data without launching the software and navigating to the certain interface.

This opportunity is provided by the "Inform" module of the PhoneUP bundle.

Cisco IP Phone as an interface to enterprise software

In some cases IP phone can be used to provide the user interface to 3rd party enterprise app. This is useful when the PC is unavailable, but the user interface is simple enough to be run on the IP phone display.

For example, employees can use Cisco IP phone to maintain records of the working time - by pressing a few buttons the employee registers the start / end of the work, and the entered data is transferred to the enterprise software.

Another example - the employee who issues goods needs to enter the order number to verify its status and configuration, confirm the issuance and print necessary documents – these operations don't require a PC, the IP phone with a large display will be enough.

System Requirements

The PhoneUP bundle integrates with Cisco Unified Communications Manager (CallManager) providing Cisco IP phones with all the features described above.

Supported IP PBX and IP phones:

IP-PBX Cisco:

  • Cisco Unified Communications Manager (CUCM) 6.X, 7.X, 8.X, 9.X,10.X;
  • Cisco BE 6000 / 7000.

Cisco IP phones and other endpoints:

  • Cisco 6900 Series (6911, 6921, 6941, 6945, 6961);
  • Cisco 7800 Series (7821, 7841, 7861);
  • Cisco 7900 Series (7906, 7911, 7914, 7915, 7916, 7921, 7925, 7931, 7937, 7941, 7942, 7945, 7961, 7962, 7965, 7970, 7971, 7975);
  • Cisco 8900 Series (8941, 8945, 8961);
  • Cisco 9900 Series (Cisco Unified IP Phone 9971, Cisco Unified IP Phone 9951);
  • Cisco IP Communicator;
  • Cisco Jabber for Windows.

CUCM Troubleshooting: Availability Issues

In this article we discuss some issues of the CUCM environment troubleshooting.

This time we will consider the aspects of CUCM availability.

So, your CUCM responds slowly or doesn’t respond at all. Why?

CUCM does not respond to endpoint requests

When the Primary CUCM slows down or even dies IP phones and/or gateways lose the registration. When taking the receiver off the hook the tone appears with delay or does not appear at all.

The most likely reasons for this are:

  • The CUCM server hung on the OS level and requires a reboot.
  • The Cisco Call Manager service hung. Also, there may be problems with the Cisco TFTP Server service responsible for the phone configuration loading. Check the status of both services (Serviceability > Tools > Control Center> Feature Services). Causes of the problems can be found in System logs .
  • High CPU load of the CUCM server. Check the CPU Usage to understand what process should be blamed.
  • A memory leak - you need to check the memory usage.

CUCM System logs

CUCM is an appliance based on Linux, and it does not provide a regular access to a full Linux console, therefore System log can only be viewed through RTMT.

We need a Syslog Viewer - the analogue of Windows Event Viewer. If something is wrong with the system it’s one of the first interfaces to look at.

System > Tools > Syslog Viewer

Syslog Viewer allows you to view the messages from the following logs:

  • System Logs – everything that concerns hardware and OS
  • Application Logs – CUCM logs
  • Security Logs – user login attempts etc.

Read more about working with RTMT here: Working with Cisco Unified RTMT

If you have problems with the Cisco Call Manager service, then the following messages can appear.

When the connection with the service is lost:

The Cisco CallManager service terminated unexpectedly. It has done this one time. The following corrective action will be taken in 60000 ms. Restart the service.

Timeout 3000 milliseconds waiting for Cisco CallManager service to connect.

If you have problems with the launch:

The service did not respond to the start or control request in a timely fashion.

High CPU

Cisco Call Manager service may stop responding because of a system resources (processor or memory) overloading.

Most often it occurs at a high CPU load.

We can monitor the CPU load in two ways:

  • CPU monitoring in CLI.
    The average CPU load can be seen by executing:
    show stats io
    show perf query class Processor

    CPU load by processes:
    show perf query counter Process "% CPU Time"
    show process load
  • CPU monitoring in RTMT
    This can be checked in the general information on the system:
    System > System summary

CUCM Administration page is not displayed

If the CUCM admin page (https:///ccmadmin) does not open:

  • First try to clear the browser’s cache
  • Check the network settings, try to ping the CUCM IP
  • If the CUCM is accessed by name, check the DNS, and whether the name resolves correctly
  • Make sure the Cisco Tomcat service is running
  • Check Firewall and Access Lists settings
  • Check the CPU load on the CUCM node

Checking Cisco Tomcat service

Use CLI to check the status of the service:

utils service list

Launch it if necessary:

utils service start Cisco Tomcat

Slow Server response

The slow server response may cause: the delay to receive the dial tone, delays in opening the admin/user web-interfaces, delays in dialing.

The possible reasons are:

  • The CUCM server and the switch have different Speed / Duplex.
    Check the settings on the server and the switch. Try to set Auto for both.
  • High CPU load or Memory leak. Check the CPU usage.
  • Also, the wrong Dial plan may cause delays in dialing.

Network Settings

The CUCM network settings may be checked with the CLI command:

show network eth0

You can change the value of the duplex or speed in CUCM by executing:

set network nic eth0

To view the switch network settings execute:

show interface fa 1/0

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