Kenneth M. Chipps Ph.D. Home Page



Lab 14

Configuring VOIP Using Cisco Call Manager Express

and

Cisco 2911 Router with the VIC2-2FXO Interface

Cisco 2960 POE Switch

Grandstream GXP1405 Telephones

Preliminary Discussion

In the Cisco world for organizations with 450 or fewer endpoints, such as VOIP phones, the suggested method to use to implement VOIP is Call Manager Express. This is a set of telephony functions built into the router's IOS on several of the ISR routers designed for small and medium sized organizations, as well as the branch offices of larger organizations. Support for telephony services is not part of the basic IOS package. Up through Version 12 of the Cisco IOS support for telephony services requires at least one step up from the basic IOS package, IP Base, to the IP Voice package. With Version 15 of the IOS the Unified Communications license must be activated.

One might ask why Grandstream instead of Cisco telephones. First, because these were the only hardphones I could get to work in less than a few hours. But mostly because the process required to make a Cisco hardphone work is the dumbest, most complicated, most obtuse, waste of time I have encountered in quite a while. First you have to figure out what version of the call manager you have, then what version of telephone you have, and then you have to track down the firmware files for the telephone which of course you cannot find on the Cisco website because the telephone is no longer supported just days after you bought it. Once you find all of this stuff you have to copy it to the flash memory. The telephone has to download it every time it boots and so on and so forth. All the older Cisco telephones want you to use the SCCP method instead of SIP. I could go on and on, but you get the idea. The folks at Selsius Corporation must have laughed all the way to the bank when Cisco bought this stupid system from them.

Network Layout

Here is the network we will use in this lab. The router is a 2911. The switch is a 2960 with POE ports. The telephones are Grandstream GXP1405 hardphones. This lab will work with any equipment that supports these commands.

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The router connects to the Fa0/1 port on the switch. The telephones connect to switch ports Fa0/2 and Fa0/3.

Check for an IOS Version That Supports Telephony

At the router’s privileged level check to see if the installed IOS version supports telephony service by entering

show version

It should look like this at the bottom of the output:

[pic]

The line that reads

uc uck9 Permanent uck9

is what we are looking for. This is the voice module that has been activated.

Configuring the Devices

To allow VOIP calls to travel through the network back and forth to the telephones the IP telephony functions of the devices must be enabled as they are off by default. Let's look at the configuration of each device. The configurations are shown with comments for the commands. The comments are the lines that begin with !. If you copy and paste this be sure the comments are on one line.

The MAC addresses have to be the ones from the equipment being used.

For the router:

!Enter configuration mode

enable

configure terminal

!Name the router

hostname VoiceRouter

!Activate the port connected to the switch

interface gi0/0

ip address 192.168.1.1 255.255.255.0

no shutdown

!Create the DHCP address pool for the telephones

ip dhcp excluded-address 192.168.1.1 192.168.1.5

ip dhcp pool Voice

network 192.168.1.0 255.255.255.0

option 150 ip 192.168.1.1

default-router 192.168.1.1

!Start the voice service on the router

voice service voip

!To allow communication between and within different communication methods

allow-connections h323 to h323

allow-connections h323 to SIP

allow-connections SIP to SIP

sip

!Enable the SIP Registrar

registrar server

!This enters voice register global configuration mode to set the parameters for all of the supported SIP phones

!This is the same as the telephony-service command used for SCCP telephones

voice register global

!Set the mode for provisioning the SIP phones to Cisco Unified Configuration Manager Express

mode cme

!Setup the Cisco Unified Configuration Manager Express router to receive messages from the SIP phones

!through this IP address and port

source-address 192.168.1.1 port 5060

!Set the maximum number of telephones that can be attached

max-pool 10

!Set the maximum number of directory entries per telephone

max-dn 20

exit

!Define the telephones that will be used

voice register dn 1

number 101

name 3CX1

label 101

voice register dn 2

number 102

name 3CX2

label 102

voice register dn 3

number 103

name GS1

label 103

voice register dn 4

number 104

name GS2

label 104

exit

voice register pool 1

id mac 0013.3B90.8028

number 1 dn 1

username 1111 password 1111

codec g711ulaw

voice register pool 2

id mac C8B3.7336.00A5

number 1 dn 2

username 2222 password 2222

codec g711ulaw

voice register pool 3

id mac 000B.823E.77BE

number 1 dn 3

username 3333 password 3333

voice register pool 4

id mac 000B.82E3.77C4

number 1 dn 4

username 4444 password 4444

end

Next is the switch.

Here is its configuration:

!Switch

enable

config t

int range fa0/1-5

!This command sets the ports shown above to carry voice traffic.

switchport voice vlan 1

end

You must connect the phones to one of these ports.

The last step is to setup the VOIP telephones. First, reset the telephones to the default settings by pressing these keys:

Enter key – which is the round button inside of the direction key circle

Down Arrow Key until Config is highlighted on the LCD display

Enter key – which is the round button inside of the direction key circle

Down Arrow Key until Factory Reset is highlighted on the LCD display

Press the softkey that is labeled OK

After the telephone boots, determine the IP address of each telephone by pressing the following keys:

Enter key – which is the round button inside of the direction key circle

Down Arrow Key to select Status on the LCD display

Enter key – which is the round button inside of the direction key circle

Enter key – which is the round button inside of the direction key circle

On the next screen the IP address is shown on the second line of the LCD display

Press the Left Arrow Key three times to return the display to the start point

Open a web browser. Enter the IP address of the telephone. The administrator’s password is:

admin

The only entries that need to be made are these on the Account 1 page.

For the first telephone

SIP Server: 192.168.1.1

SIP User ID: 103

Authenticate ID: 3333

Authenticate Password: 3333 – these numbers disappear from view after entry, but appear to need to be entered

For the second telephone

SIP Server: 192.168.1.1

SIP User ID: 104

Authenticate ID: 4444

Authenticate Password: 4444 – these numbers disappear from view after entry, but appear to need to be entered

The SIP Server: entry is the IP address of the router’s interface connected to the switch.

The SIP User ID: is the extension number assigned in this section of the router’s configuration

voice register dn 3

number 103

name GS1

label 103

Authenticate ID: is the username

Authenticate Password: is the password

Both from this section of the router’s configuration:

voice register pool 3

id mac 000B.823E.77BE

number 1 dn 3

username 3333 password 3333

Notice that the Authenticate Password: looks like it is blank, but it is not. This entry must be there. Now why it is hidden from the administrator is not entirely clear to me, but then neither is using the extension number for the SIP User ID: which I would have thought was the username. It only took me two days to figure all of this out, and only then with help from Grandstream.

When this is all done a line should appear on the router saying that the telephones have registered.

Call back and forth between the telephones.

Connecting to an Outside Telephone Line

To call an outside telephone number, not just call between the two extensions, an FXO port must be added to the router, and then that port must be connected to either the PSTN or to a cellular network. So this is a two part process.

First, an interface card must be installed in the router. In this example the router is a Cisco 2911. The interface card used here is the VIC2-2FXO. This card goes in an EHWIC slot. It has two FXO ports, but only one will be used. These ports are pink in color. The board looks like this:

[pic]

[pic]

Second, using a RJ-11 patch cable connect the rightside 0 port of the VIC2-2FXO to the telephone line that will be used.

[pic]

There are several ways to connect the equipment to the outside world so that calls can be made from the telephones to outside telephone numbers and to the telephones from outside lines.

The easiest method, if there is a standard analog PSTN type telephone landline connection available, is to connect the right RJ-11 port on the router’s voice card to the PSTN demarc using a standard RJ-11 cable.

If the building’s telephone system is an all VOIP system, arrange to have an analog line, such as for a fax machine, installed where the router will be.

If an analog line cannot be installed an ATA can be installed in between the router and a standard Ethernet LAN connection. As a network device with a MAC address the ATA can be assigned an IP address. The ATA will appear to the VOIP system as a VOIP telephone. The ATA will need to be registered with the VOIP system so that it can be assigned a telephone number just as is done with any VOIP telephone.

A VOIP service provider account can be created, and then the ATA used to connect to the Internet and then to the VOIP service provider.

The last method, which we will use here, is to insert a Bluetooth Cellular Gateway between the router and a cellular telephone. For example, an XLink BTTN can be used as seen below with the router in place of the telephone on the right.

[pic]

Here is the layout.

[pic]

In this type of connection attach the cellular gateway’s top RJ-11 port – DO NOT USE THE BOTTOM PORT AND DO NOT CONNECT THIS DEVICE TO AN ACTIVE PSTN ANALOG TELEPHONE DEMARC - to the right side voice port of the voice card in the router using a standard analog PSTN telephone cable. Power up the cellular gateway by connecting it to electrical power. There is no power switch. Pair the cellular gateway to a Bluetooth enabled cell phone by pressing the button with one dot beside it until the button begins to flash, then activate the Bluetooth pairing function on the cell phone. When the two devices find each other, complete the pairing as required by the cell phone. If the cell phone requires a pairing code use 0000.

On bootup the lights on the VIC2-2FXO board will brielfy light and then we should see something like this:

[pic]

And this, notice the bottom two lines:

[pic]

To activate the FXO port which connects to the PSTN demarc – which is the Bluetooth Cellular Gateway in our case, add this to the router’s configuration from the global configuration level.

voice-port 0/1/0

connection plar 103

caller-id enable

dial-peer voice 1 pots

destination-pattern 9.T

port 0/1/0

or this using the opx option to speed up the connection here in the lab

voice-port 0/1/0

connection plar opx 103

caller-id enable

dial-peer voice 1 pots

destination-pattern 9.T

port 0/1/0

These commands do the following:

voice-port 0/1/0 selects the port to which the RJ-11 cable is attached at the router end – the right hand port 0 in this case - with the other end going to the Bluetooth Cellular Gateway’s top port.

connection plar opx 103 defines the connection to this port as a plar or private line automated ringdown. Cisco explains it this way

Connection PLAR mode is a switched VoIP call. The call is setup on an as-needed basis. With connection PLAR, no bandwidth is consumed while the phone is on hook. When a phone connected to a POTS dial peer is taken off-hook, the call is automatically connected and the remote phone begins to ring.

The opx option sets the interface to not answer until the other end has answered. This way the router does not answer a call and then have nowhere to send it. However, in our case it mostly just speeds up the connection when the call is made.

103 sends all inbound calls to extension 103.

caller-id enable shows the phone number on the extension when it is called.

dial-peer voice 1 pots creates the method for outbound calls. In this case using the PSTN.

destination-pattern 9.T defines the pattern for calls. Here we dial 9 and then the 10 digit telephone number of the outside telephone we want to call.

port 0/1/0 tells these commands what port they apply to.

Test this by calling the telephone number assigned to the PSTN demarc you are using. In this example, the telephone number to call is the cell phone that is paired with the Bluetooth Cellular Gateway. The Grandstream with extension 103 should ring. Answer it and talk to the person calling.

Next, call from the extension 103 hardphone to a cell phone, but not the one tethered to the Bluetooth Cellular Gateway, when it rings answer the telephone and speak to the person on the other end.

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