VoIP Part 2 How to setup Call Manager Express
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
!=== 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.
!=== 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
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
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
/ 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
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
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
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
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:
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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
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.
Thanks,
D.K
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VoIP Part 1 Configuring FXS ports
VoIP Part 1 - Configuring FXS ports
We are going to start a series here explaining how to configure VoIP and Call Manager Express on Cisco Routers. This is a hot topic these days and we are hoping that you will learn how to configure Cisco's IPT products:
We are going to start a series here explaining how to configure VoIP and Call Manager Express on Cisco Routers. This is a hot topic these days and we are hoping that you will learn how to configure Cisco's IPT products:
Here we start with configs that have been available on Cisco routers for a long time, but are as useful today as ever.
We start with four analog phones connected to two routers. 1000 & 1001 are connected to R1 and 2000 & 2001 are connected to R2.
FXS ports - these are used to connect analog devices to Cisco routers and other devices. The FXS port provides power, dial tone, ring tone etc to the analog phone.
Diagram:
The first thing you need to determine is what Voice Ports the phones are connected to:
show voice port summary
R1# show voice port summary
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
====== == ========== ===== ==== ======== ======== ==
2/0/0 -- fxs-ls up dorm on-hook idle y
2/0/1 -- fxs-ls up dorm on-hook idle y
2/0/1 -- fxs-ls up dorm on-hook idle y
You can see from the above, that there are two FXS ports in R1
Then we can do the configuration:
On R1 do the following:
R1#conf t
!=== Enter global configuration mode
!=== Enter global configuration mode
R1(config)#dial-peer voice 1 pots
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. POTS is a traditional interface - FXS, FXO, E&M, PRI, E1, T1 etc
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. POTS is a traditional interface - FXS, FXO, E&M, PRI, E1, T1 etc
R1(config-dial-peer)#destination-pattern 1000
!=== This gives the analog phone a number of 1000
!=== This gives the analog phone a number of 1000
R1(config-dial-peer)#port 0/0/0
!=== Tell the router which port to push the call out of when a call is made to 1000
!=== Tell the router which port to push the call out of when a call is made to 1000
R1(config-dial-peer)#dial-peer voice 2 pots
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. POTS is a traditional interface - FXS, FXO, E&M, PRI, E1, T1 etc
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. POTS is a traditional interface - FXS, FXO, E&M, PRI, E1, T1 etc
R1(config-dial-peer)#destination-pattern 1001
!=== This gives the analog phone a number of 1001
!=== This gives the analog phone a number of 1001
R1(config-dial-peer)#port 0/0/1
!=== Tell the router which port to push the call out of when a call is made to 1001
!=== Tell the router which port to push the call out of when a call is made to 1001
This will enable the two analog phones on R1 to talk to each other. You would do something similar on R2:
R2#conf t
!=== Enter global configuration mode
!=== Enter global configuration mode
R2(config)#dial-peer voice 1 pots
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. POTS is a traditional interface - FXS, FXO, E&M, PRI, E1, T1 etc
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. POTS is a traditional interface - FXS, FXO, E&M, PRI, E1, T1 etc
R2(config-dial-peer)#destination-pattern 2000
!=== This gives the analog phone a number of 2000
!=== This gives the analog phone a number of 2000
R2(config-dial-peer)#port 0/0/0
!=== Tell the router which port to push the call out of when a call is made to 2000
!=== Tell the router which port to push the call out of when a call is made to 2000
R2(config-dial-peer)#dial-peer voice 2 pots
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. POTS is a traditional interface - FXS, FXO, E&M, PRI, E1, T1 etc
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. POTS is a traditional interface - FXS, FXO, E&M, PRI, E1, T1 etc
R2(config-dial-peer)#destination-pattern 2001
!=== This gives the analog phone a number of 2001
!=== This gives the analog phone a number of 2001
R2(config-dial-peer)#port 0/0/1
!=== Tell the router which port to push the call out of when a call is made to 2001
!=== Tell the router which port to push the call out of when a call is made to 2001
To allow the analog phones to talk to the analog phones on the other router, do the following:
R1:
R1#conf t
!=== Enter global configuration mode
!=== Enter global configuration mode
R1(config)#dial-peer voice 3 voip
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. VOIP is used when talking across an IP network
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. VOIP is used when talking across an IP network
R1(config-dial-peer)#destination-pattern 2...
!=== This tells the router that any call going to 2000 to 2999 should match this dial peer
!=== This tells the router that any call going to 2000 to 2999 should match this dial peer
R1(config-dial-peer)#session target ipv4:10.1.1.2
!=== The call that matches this dial peer should be sent to 10.1.1.2
!=== The call that matches this dial peer should be sent to 10.1.1.2
R2:
R2#conf t
!=== Enter global configuration mode
!=== Enter global configuration mode
R2(config)#dial-peer voice 3 voip
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. VOIP is used when talking across an IP network
!=== Create a dial peer. 1 is a tag which needs to be a unique number 1 to 2147483647. VOIP is used when talking across an IP network
R2(config-dial-peer)#destination-pattern 1...
!=== This tells the router that any call going to 1000 to 1999 should match this dial peer
!=== This tells the router that any call going to 1000 to 1999 should match this dial peer
R2(config-dial-peer)#session target ipv4:10.1.1.1
!=== The call that matches this dial peer should be sent to 10.1.1.1
!=== The call that matches this dial peer should be sent to 10.1.1.1
And thats it!
All the phones will now be able to successfully communicate.
We can look at the "Voice Routing Table" by typing the following command:
R1#show dial-peer voice summary
PASS
TAG TYPE ADMIN OPER PREFIX DEST-PATTERN PREF THRU SESS-TARGET PORT
1 pots up up 1000 0 0/0/0
2 pots up up 1001 0 0/0/1
3 voip up up 2... 0 syst ipv4:10.1.1.1
PASS
TAG TYPE ADMIN OPER PREFIX DEST-PATTERN PREF THRU SESS-TARGET PORT
1 pots up up 1000 0 0/0/0
2 pots up up 1001 0 0/0/1
3 voip up up 2... 0 syst ipv4:10.1.1.1
Creating dial-peers is similar to creating static routes. We are adding "voice routes" to a "voice routing table".
The four analog phones will now be able to call each other.
Thanks,
D.K,
lonetsec
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