interface Serial0
 ip address 10.10.10.1 255.255.255.252
 encapsulation frame-relay IETF
 frame-relay traffic-shaping
 frame-relay class fr1
 frame-relay interface-dlci 68 voice-encap 80
 frame-relay lmi-type ansi

router eigrp 100
  network 10.10.10.0

ip classless

map-class frame-relay fr1
 frame-relay adaptive-shaping becn
 frame-relay cir 56000
 frame-relay bc 1000
 frame-relay mincir 16000

dial-peer voice 1 pots
 forward-digits all
 destination-pattern 2..
 port 1/1

dial-peer voice 2 pots
 forward-digits all
 destination-pattern 2..
 port 1/2

dial-peer voice 3 pots
 forward-digits all
 destination-pattern 2..
 port 1/3

dial-peer voice 4 pots
 forward-digits all
 destination-pattern 2..
 port 1/4

dial-peer voice 5 pots
 forward-digits all
 destination-pattern 2..
 port 1/5

dial-peer voice 6 pots
 forward-digits all
 destination-pattern 2..
 port 1/6

dial-peer voice 7 vofr
 destination-pattern 4..
 session target Serial0 68

Explanation:

The above configs need to be entered into all 3810's. The differences in the rest of the configs depend on what kinds of voice-ports are installed in your 3810. Ports come in flavors of FXS, FXO, E&M, and DVM T1 ports. The kind of port you have is determined by what kind of voice equipment you're connecting to the 3810. FXS ports are used for connecting telephone handsets directly into the 3810 while the remaining port-types are used for connecting into telephone PBX or Key-System units. You'll generally order ports to match ports already present on the voice equipment.

The following list explains what each line in the above config means:

• The Frame Relay commands under the Serial interface control how the voice packets are sent down the Frame Relay circuit. They are:
  • encapsulation frame-relay IETF is the same config used on all Frame Relay circuits. It tells the interface to encpasulate all packets within Frame Relay frames.

  • frame-relay traffic-shaping applies Traffic Shaping to the interface. The 3810 will work without this command, but performance will quickly degrade. Without it, the router will spit data down the pipe as it arrives, with no effort made to avoid traffic congestion. Turning on Traffic Shaping makes the interface aware of the bandwidth limits being used by this particular Frame Relay circuit, making the router aware of the bandwidth CIR and burst limits. These values are defined by the map-class, a few lines down.

  • frame-relay class fr1 applies the map-class "fr1" to this interface. A map-class is basically just a script, or batch-file, with a bunch of sequential commands entered, and is defined below with the "map-class" command. "fr1" is just a name identifying the particular script. Multiple scripts can be defined, named "larry", "curly", and "moe", each one applied to a different interface.

  • frame-relay interface-dlci 68 voice-encap 80 assigns this interface to the appropriate circuit's DLCI. The DLCI will have been provided by the telephone company and, in this example, is DLCI number 68. The "voice-encap" command after this defines how the router applies data segmentation to voice-packets: how they will be sliced and diced before routing them across the circuit. The number refers to the desired packet size, in bytes. Valid sizes must fall within the range of 80 to 1600 bytes. This command is *required* for voice traffic to be routed. If you forget to enter it, data will be routed OK, but voice won't go nowhere. Values must match on both ends of the circuit. You might want to try entering the whole line without the voice-encap option when troubleshooting a circuit, since mismatched values will cause both voice and data to stop working.

    Set the segmentation-size according to the size of the Frame Relay circuit. 80 bytes is a good size for 56K and 64K circuits. 160 bytes is a good size for 128K circuits. Cisco has a guide that shows the recommended encap-size for various sized circuits, here.

  • frame-relay lmi-type ansi defines the type of LMI protocol used by this Frame Relay circuit. It's the same command as is used on all Frame Relay circuits.

• router eigrp 100 turns on the routing protocol. The numbers below it define what subnets are advertised by the routing protocol. If you're using Static Routes then replace this with your static routing configs.

• ip classless turns on classless routing, which should be entered on all routers by default.

• map-class frame-relay fr1 is the script used by Traffic Shaping. "fr1" is the name of the script, and can be named anything. The commands under it are the defintions that allow the router to intelligently detect traffic conditions on the circuit, like congestion and available bandwidth. They are:
  • frame-relay adaptive-shaping becn makes the router aware of Frame Relay "Backwards Explicit Congestion Notification", in which the upstream Frame Relay switch sets the "BECN bit" on a frame telling the sending device (your router) to slow down and pace the data transmission due to circuit congestion. Without this command the router would not be aware of any congestion and would continue hapilly sending data down the circuit, requiring the TCP stacks on the end hosts to request retransmissions down an unreliable circuit.

  • frame-relay cir 56000 tells the router what the incoming Committed Information Rate (CIR) is for this circuit. Set it according to your circuit's bandwidth.

  • frame-relay bc 1000 tells the router what the outgoing Committed Information Rate (CIR) is for this circuit. Generally this should be lower than the outgoing CIR. Cisco recommends this value, but doesn't explain why...

  • frame-relay mincir 16000 defines the minimum acceptable CIR for this circuit. If available bandwidth falls below this limit, data won't be sent.
    * Cisco has a guide for creating Traffic Shaping map-classes here.

• dial-peer voice 1 pots maps a phone number to a destination port. Dial-peers are sort of like static routes. All ports need to have a phone number mapped to it. Assuming all of the telephone extensions in your office all begin with 2, and are three digits in length, you would create a seperate dial-peer statement for each port. A dial-peer also needs to be created that points to the phone numbers at the other end of the circuit. Each dial-peer is numbered uniquely from all others.

  Thus the first line of a dial-peer contains the sequence-number, followed by the type of mapping. Your choices are "pots" for pointing to a local port, and "vofr" for pointing to a Frame Relay circuit. If you hit the "?" key after the dial-peer number, you'll see three choices; "pots", "vofr", and "VoFR". The last two are indentical so you can use either one

• forward-digits all causes all digits to be forwarded out the port. Without this line, the first digit of the string is dropped.

• destination-pattern 2.. defines the phone number. Each dot is a wild-card and will match any digit. Thus, 2.. will match any three-digit telephone extension beginning with a 2. A 4-digit extension beginning with a 2 would have a "2" followed by three digits

• port 1/1 points the phone number to the local port.

At least one additional dial-peer needs to be configured, mapping the remote phone number to the other side of the circuit:

• dial-peer voice 7 vofr defines a dial-peer for the Voice Over Frame Relay type. Whereas a pots dial-peer points to a local port, a vofr dial-peer points over a Frame Relay port. A dial-peer pointing over an ATM circuit would be defined as "voATM".

• destination-pattern 4.. defines the remote phone number. In this example all of the remote telephone extensions are three digits that begin with 4. Digits sent over a Frame Relay circuit are sent with all digits intact by default, therefore you do not need to use the "forward-digits" command.

• session target Serial0 68 points the phone number down the appropriate circuit. In this case, the digits are sent down DLCI 68 on interface Serial 0.

• If there are different remote extensions, each one beginning with a different initial digit, you'll need to create a seperate vofr dial-peer, each one listing the appropriate digit and number of wild cards, with each dial-peer pointing down the same circuit. Dial-peers are the only way the router knows where to send digits, so if you dial a number and the call doesn't go through, make sure there's a dial-peer configured for the number you're calling. Once the digits reach the router on the other end, that router will look through it's own dial-peers to decide what local port to send the call out.

Voice Ports:

The other configurations required are the voice-port configs. The voice-ports are the ports that are used to connect to the voice equipment. If the voice-ports are anything other than a Digital T1 port, there are as many voice-ports as there are physical ports. If you have a Digital T1 port there will be 24 voice-ports in the config.

The settings used on the voice-ports are dependent on the signalling and cabling used by the voice equipment. The following issues need to be addressed before connecting voice equipment to each type of port:

• FXS ports. These are "Station Ports", which are used to connect directly to analog telephone-handsets and fax machines. As there are 6 ports on the 3810, you can connect a maximum of 6 telephones to the router in this way. FXS ports are meant for small office environments, and generally don't require any special configs on the voice-ports. Just plug them in and define your dial-peers, and they should work. The cables used to connect to the ports are standard phone cables.

• FXO ports. These are "Office Ports", which are used to connect into telephone key-systems and PBXs. Each line will support one telephone call, so even though you may be connecting to a key-system or PBX with hundreds of phones, only 6 concurrent calls can be routed at once. This is usually acceptable in a small office environment. FXO ports usually require defining the type of cabling used and the off-hook signalling used. The cables used to connect to the ports are straight-thru cables, like Ethernet.
  Configure each of the voice-ports like this:
  voice-port 1/1
   operation 4-wire
   type 2
   signal wink-start

  • There will be as many voice-port lines as there are physical ports, numbered 1/1 through 1/6, assuming all 6 ports are being used. You've gotta type each one.

  • The line operation 4-wire defines whether the cabling used is 2-wire or 4-wire.

  • The line type 2 refers to the kind of of E&M signalling to use. There are 5 varieties, and the 3810 uses Type 2 E&M signalling by default. The telephone switch needs to use the same kind.

  • The line signal wink-start refers to another type of signalling used by telephone switches. Your options are "wink-start", "immediate", and "delay-dial". Just make sure the 3810 and the telephone switch are using the same thing.
    • The main issue to keep in mind is that the 3810 uses wink-start signalling by default, which most PBX's and telephone key-systems support. Make sure that the telephone switch sets it's "wink-start windowing size" to at least 200 milliseconds. This defines how long the telephone switch will wait for the 3810 to send it digits. Many telephone switches have their windowing size set to a shorter value and the 3810 isn't quick enough to send digits at anything quicker than 200 milliseconds.

• E&M ports. These are "Ear and Mouth" ports, and refer to the type of signalling used to talk to the telephone equipment. These are used to connect to a telephone key-system or a PBX, and is simply an alternative to FXO ports. They also require defining the type of cabling used and signalling parameters. The cables used to connect to the ports use a specific pin-out, like this.
  Configure each of the voice-ports according to the example above, for FXO ports.
  Also, if you're using wink-start, use the same rule for windowing size as described above for FXO ports.

• DVM T1 ports. If you are connecting to a PBX you will usually need a Digital T1 port, which is labelled DVM (Digital Voice Module) on the 3810. This is not the Serial 0 port, which is used for T1 connections also. Serial 0 is where you plug in the Frame Relay circuit. The DVM port is where you plug in the cable to the PBX. As it's a T1, there are 24 logical channels, and therefore the 3810 can route up to 24 concurrent calls at once. You'll need to define the signalling type on the Controller port. The type of cabling is a rolled-T1 cable.
  First, configure the T1 Controller port like this:
  controller T1 0
   framing esf
   linecode b8zs
   mode cas
   voice-group 1 timeslots 1-24 type e&m-wink-start

  • The line controller T1 0 refers to the Controller port for your T1. This lets you define settings for the whole pipe, independent of each logical channel.

  • The line framing esf refers to the signalling type required. ESF is the most common type used. Make sure it matches the signalling required by the T1 port on the telephone switch.

  • The line linecode b8zs is another kind of signalling used on T1's. Again, make sure it matches what's required by the telephone switch.

  • The line mode cas turns on Channel Associated Signalling. This is required for the port to handle voice traffic.

  • The line beginning with voice-group 1 defines how many logical channels will be used by the port. If you are using less than 24 channels with the telephone switch, replace the "24" with the number of the last channel you'll be using. This is followed by the signalling used, with wink-start being the default signalling type used.

    Next, configure each of the voice-ports like this:

    voice-port 1/1
     timing answer-winkwidth 0

  • There will be 24 voice-ports, numbered 1/1 through 1/24. You've gotta type each one.

  • The line timing answer-winkwidth 0 ensures that the router exchanges digits with the telephone switch using wink-start as quickly as possible. Not required, but a good idea.