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To your success
David Bombal

Extension mobility in Cisco Unified CME 4.2 and later versions provides the benefit of phone mobility for end users. CME still supports 240 IP Phones.

The way this works is very similar to CallManger.

Overview:

Extension mobility allows phone users to temporarily access a physical phone other than their own phone and utilize their personal settings, such as directory number, speed-dial lists, and services, as if the phone is their own desk phone. The phone user can make and receive calls on that phone using the same personal directory number as is on their own desk phone. This is very useful for hot desking.

Each IP phone that is enabled for extension mobility is configured with a logout profile. This profile determines the default appearance of a phone that is enabled for extension mobility when there is no phone user logged into that phone. Minimally, the logout profile allows calls to emergency services such as 911. A single logout profile can be applied to multiple phones.

After an IP phone that is enabled for extension mobility boots up, the Services button on the phone is configured with a login service URL hosted by Cisco Unified CME that points to the extension mobility Login page.

A phone user logs in to a Cisco Unified IP phone that is enabled for extension mobility by pressing the Services button or a Unified CCX agent can log in using a Unified CCX Cisco Agent Desktop. User authentication and authorization is performed by Cisco Unified CME. If the login is successful, Cisco Unified CME retrieves the appropriate user profile, based on user name and password match, and replaces the phone's logout profile with the user profile.

After the phone user is logged in, the service URL points to a logout URL hosted by Cisco Unified CME to provide a logout prompt on the phone. Logging into a different device automatically closes the first session and start a new session on the new device. When a phone user is not logged in to any phone, incoming calls to the phone user's directory number are sent to the phone user's voice mailbox.

For button appearance, extension mobility associates directory numbers, then speed-dial numbers in the logout profile or user profile to phone buttons in a sequence. If the profile contains more numbers than there are buttons on the physical phone to which the profile is downloaded, the remaining numbers in the profile are ignored.
 

Configuring Extension Mobility:

Step 1: Configure a Logout Profile for an IP Phone
You need to create a logout profile to define the default appearance (when no one is logged in) for a phone that is enabled for extension mobility:

           configure terminal
           !=== Enter global configuration mode

           voice logout-profile <tag>
           !=== Enter voice logout profile configuration mode. This creates as logout profile
           to that defines the default settings of a phone when no one is logged in

           username <username> password <password>
           !=== Create username/password to be used by a TAPI phone device to login in

           number <number> type <type>
           !=== Telephone number to be displayed on the phone.
           !=== Example - number 1000 type silent-ring

           speed-dial <speed-tag> number [label label]
           !=== Speed dial number and label
           !=== Example - speed-dial 1 1001

           pin <pin>
           !=== PIN to be used by a phone user to disable call blocking


Step 2: Enable an IP Phone for Extension Mobility
To enable Extension Mobility on a phone in CME do the following:

           configure terminal
           !=== Enter global configuration mode

 ephone <phone-tag>
!=== Setup a phone with tag from 1 to what is supported on the router

 mac-address <mac-address>
 !=== Type in the phones MAC address

type <phone-type>
!=== Specify the type of phone like 7960, 7940 etc

logout-profile <profile-tag>
!=== Specify the log out profile on this phone when no one is logged in and enable Extension Mobility on the phone

 

Step 3: Configure a User Profile

The last step is to create the user profiles:
 

           configure terminal
           !=== Enter global configuration mode

voice user-profile <profile-tag>
!=== Create a user profile for a specific user

name <username> password <password>
!=== username & password of the user

          number <number> type <type>
          !=== Telephone number to be displayed on the phone when a user is logged in
          !=== Example - number 1002 type silent-ring

           speed-dial <speed-tag> number [label label]
           !=== Speed dial number and label
           !=== Example - speed-dial 2 1003

           pin <pin>
           !=== PIN to be used by a phone user to disable call blocking when logged in
 

So, the one big feature that was missing from CME is now available - Extension Mobility on a single router.

Note: Extension mobility across multiple CME routers is not supported.

Uses of DSPs:
DSPs are used for the following functions:

• Voice Termination
• Hardware Conferencing
• Transcoding
• Media Termination Points

We will discuss all of these uses in this and upcoming newsletters.

Before discussing each function, it is important to understand codec complexity.

Codec Complexity:
The two codecs we will concentrate on here are G711 and G729 as they are the codecs supported by Cisco's Call Manager IP Phones. For a brief overview of codecs, please go here: Codec Overview

Codecs are generally put into two categories - medium and high complexity. The complexity of a codec is determined by the amount of processing the DSP needs to do to create the packeterized voice.

Medium Complexity Codecs: These codecs use less resources to produce packeterized voice than high complexity codecs:

High Complexity Codecs: These codecs comsume a lot of resources to produce packeterized voice.

Flex Mode: Originally, you had to dedicate DSPs to either medium or high complexity. The good news is that on the new C5510 chipsets (more common these days) the requirement to specify the codec complexity at configuration time has been eliminated.

We now have a mode called flex mode which allows the DSPs on the fly to support either medium or high complexity. A DSP in flex mode accepts a call of any supported codec type, as long as it has available processing power. The overhead of each call is tracked dynamically via a calculation of processing power in millions of instructions per second (MIPS). Cisco IOS performs a MIPS calculation for each call received and subtracts MIPS credits from its budget whenever a new call is initiated.

In flex mode, note the following:

• 15 MIPS per call for G.711alaw/ulaw
• 30 MIPS per call for G.729/G.729a/G.729b/G.729ab

PVDMS:

High-Density Packet Voice digital signal processor (DSP) Modules (PVDM2) enable Cisco Integrated Services Routers (ISRs) to provide the functions listed above. PVDMS basically house the DSPS.

The high-density packet voice DSP modules are available in five versions: PVDM2-8, PVDM2-16, PVDM2-32, PVDM2-48, and PVDM2-64.

Here are some PVDMS - top and bottom views:

Top Bottom PVDM2-64     PVDM2-48     PVDM2-32     PVDM2-8 and PVDM2-16       Below is a table showing the number of DSPs per PVDM and how many calls can be made per DSP:   Name Description1 Number of DSPs Maximum Channels in G.711 Maximum Channels in High Complexity Codecs Maximum Channels in Medium Complexity Codecs PVDM2-8 8-Channel Packet Fax/Voice DSP Module 1* 8 4 4 PVDM2-16 16-Channel Packet Fax/Voice DSP Module 1** 16 6 8 PVDM2-32 32-Channel Packet Fax/Voice DSP Module 2 32 12 16 PVDM2-48 48-Channel Packet Fax/Voice DSP Module 3 48 18 24 PVDM2-64 64-Channel Packet Fax/Voice DSP Module 4 64 24 32

A PVDM2 connects to the host through 80-pin single in-line memory module (SIMM) slots. The module is field insertable and removable. Below an example shows how PVDM2 is plugged into PVDM2 SIMM slots on the Cisco high-density digital voice network modules.
 

Voice Termination:

When a router receives a call from the PSTN on a PRI, BRI or FXO port and the call is going to an IP phone, the router will require DSPs to terminate the VoIP call and PSTN call. The DSPs will be used to convert the traditional voice call into a packeterized VoIP call and vice versa.

Thus, it applies to a call that has two call legs, one leg on a time-division multiplexing (TDM)
interface and the second leg on a Voice over IP (VoIP) connection. The TDM leg must be terminated by hardware that performs coding/decoding and packetization of the stream. This termination function is performed by digital signal processor (DSP) resources residing in the same hardware module, blade, or platform.

The number of supported calls depends on the computational complexity of the codec used for a call and also on the complexity mode configured on the DSP. Cisco IOS enables you to configure a complexity mode on the hardware module. Some hardware platforms have two complexity modes, medium complexity and high complexity, while other hardware platforms have medium and high complexity as well as flex mode.

Calulating the required number of DSPs (G.711 Voice Termination):

I think that the easiest way to calculate the number of DSPs required in a router is to use the DSP Calculator on Cisco's website.

Here is a demonstration of how to use the DSP calculator on a 2811 with a PRI:

Step 1:

Choose your Router & IOS

Click Next

Step 2:

Choose the Voice Interface Cards (VICs) that this router is going to house.

Also choose the number of channels to enable and the VoIP codec that is going to be used.

In this example we have done the following:

1) 1 x PRI (E1)

2) Enable all 30 channels

3) We are only going to use G711 calls

Click Submit

Step 3:

RESULT!

In the output below, you can see that the minimum DSPs required for this configuration is 2 which requires either of the following:

1 X PVDM2-32 or 2 X PVDM2-16

However, it is better (if it can be afforded) to put more DSPs in, as they could be used for more capacity later. Don't forget that if the codec is changed to G729 is will require more processing power and the extra DSPs could also be used for hardware conferencing or transcoding.

CLI Output:

G729 example (Voice Termination):

Step 1:

Choose your Router & IOS

Click Next

Step 2:

Choose the Voice Interface Cards (VICs) that this router is going to house.

Also choose the number of channels to enable and the VoIP codec that is going to be used.

In this example we have done the following:

1) 1 x PRI (E1)

2) Enable all 30 channels

3) We are only going to use G729 calls

Click Submit

Step 3:

RESULT!

In the output below, you can see that the minimum DSPs required for this configuration is now 4 which requires either of the following:

one PVDM2-64
or
two PVDM2-32
or
one PVDM2-16 + one PVDM2-48

Next newsletter:

We will discuss how to configure DSPs to perform transcoding.

Resources and References:

DSP Calculator

Call Manager Express (CME) is a router based VoIP solution currently supporting up to 240 phones. It provides many of the features of Call Manager for a small business.

CME is based on router IOS. So no need to buy extra hardware - you require a version of IOS that supports CME on a voice enabled Cisco router. You will also need to download the phone firmware and GUI files into flash on the router. CME can be configured directly using the IOS without any other files. However, the phones will use whatever firmware they already have which may not give you the functionality you require.

To get your phones to use the firmware that is current with your version of CME, you will need to extract the basic.tar (or zip) into flash which will provide you with the relevant firmware and GUI files. CME GUI will not work without these files and neither will the Cisco Unity Express (CUE) GUI if you have one installed.

For example, you should extract cme-basic-4.0.0.1.tar into flash for instance for my IOS (c2800nm-ipvoicek9-mz.124-9.T.bin). Refer to the Compatibility matrix for latest cme software versions.

The 7970s and 7961s for example require more files than a 7960 (see compatibility matrix above)

On of the easiest way to set a basic system is to run the telephony-service setup wizard. This will allow you to set your IP phones and get them registered with the router. More complicated configurations will be discussed in the coming months.

In this example, I am going to configure 5 phones on my CME system.

Basic CME Setup:

           R1#conf t
           !=== Enter global configuration mode

           R1(config)#telephony-service setup
           !=== Start the CME wizard. If you have previously entered the telephony-service
           command, you will have to remove it before running this command. Be careful - removing
           the telephony service will remove config from you router.

The following questions will be asked and your responses will determine the configuration that is generated.

           Do you want to setup DHCP service for your IP phones? [yes/no]:yes

           IP network for telephony-service DHCP Pool:10.1.1.0
           Subnet mask for DHCP network :255.255.255.0
           TFTP Server IP address (Option 150) :10.1.1.1
           Default Router for DHCP Pool :10.1.1.1          

           Do you want to start telephony-service setup? [yes/no]: yes

           Enter the IP source address for Cisco CallManager Express: 10.1.1.1
           Enter the Skinny Port for Cisco CallManager Express: [2000]:2000

           How many IP phones do you want to configure : [0]: 5

           Do you want dual-line extensions assigned to phones? [yes for dual-line
            / no for single-line]:yes

           What language do you want on IP phones?
           0 English
           1 French
           2 German
           3 Russian
           4 Spanish
           5 Italian
           6 Dutch
           7 Norwegian
           8 Portuguese
           9 Danish
           10 Swedish
           [0]:0

           Which Call Progress tone set do you want on IP phones :
           0 United States
           1 France
           2 Germany
           3 Russia
           4 Spain
           5 Italy
           6 Netherlands
           7 Norway
           8 Portugal
           9 UK
           10 Denmark
           11 Switzerland
           12 Sweden
           13 Austria
           14 Canada
           [0]:9

           What is the first extension number you want to configure :[0]:1000

           Do you have Direct-Inward-Dial service for all your phones? [yes/no]:yes

           !=== If you answer yes to the previous question, you see the following prompt:
           Enter the full E.164 number for the first phone: 02071231000

           Do you want to forward calls to a voice message service? [yes/no]:yes

           !=== If you answer yes to the previous question, you see the following prompt:
           Enter the extension or pilot number of the voice message service:1999

           Call forward No Answer Timeout: [18]:10

           Do you wish to change any of the above information? [yes/no]:no

The router will automatically generate the required configuration. All you will now need to do is plug the IP phones into the network and they will auto register.

Watch the video to see the configuration working:

	DEMO of CME setup on live routers

The router will automatically be configured with config as follows:

telephony-service
max-ephones 10
max-dn 10
ip source-address 10.1.1.1 port 2000
auto assign 1 to 10
network-locale GB
create cnf-files version-stamp Jan 01 2002 00:00:00
dialplan-pattern 1 020712311.. extension-length 3
voicemail 199
max-conferences 8 gain -6
transfer-system full-consult
!
!
ephone-dn 1 dual-line
number 100
call-forward busy 1000

ephone 1

I will be explaining these commands and many more in upcoming newsletters, but lets introduce the concept of Ephones and Ephone-dns...

Ephones & Ephone-dns explained:

An ephone is the the way you configure a physical phone/handset in CME. The physical phone is matched to the configuration by its MAC address.

An ephone-dn is a telephone number/directory number/line on an ephone. You can have multiple ephone-dns on a single ephone depending on the model of phone. A 7960 for instance could have 6 ephone-dns on it.

The IP Phones will now be able to call each other once they are plugged in and auto register.

This is only a basic setup. I will continue discussing the options in CME in the coming months as well as giving detailed information on the commands used and created.

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