Configuring Voice over Frame Relay

For a description of the commands used to configure Voice over Frame Relay, refer to the Cisco IOS Multiservice Applications Command Reference publication.

Voice over Frame Relay Overview

Voice over Frame Relay enables a router to carry voice traffic (for example, telephone calls and faxes) over a Frame Relay network.

This chapter describes the commands to specifically configure Voice over Frame Relay on a router. It is assumed you have already configured your Frame Relay backbone network. As part of your Frame Relay configuration, you need to configure the map class, and the Local Management Interface (LMI) among other Frame Relay functionality. For more information about Frame Relay configuration, refer to the Cisco IOS Wide-Area Networking Configuration Guide.

The Cisco VoFR implementation allows the following types of VoFR calls:

Static FRF.11 trunks
Switched VoFR calls:
- Dynamic switched calls
- Cisco-trunk (private line) calls

This section describes the setup of Cisco VoFR calls. In addition, the following functionality is described:

FRF.11-compliant speech encoding and packetization
FRF.12-based end-to-end fragmentation under Frame Relay
Permanent trunks over dynamic switched calls
Tandem switching of calls from one VoFR dial peer to another VoFR dial peer

Static FRF.11 trunks and permanent switched trunks are used to create fixed point-to-point connections, which are typically used to connect two PBXs. In this case, the VoFR system simply provides transportation of the voice connection channels, but does not provide dial-plan-based telephone call switching. This functionality is sometimes referred to as tie-line emulation. In this scenario, all telephone call switching is performed by the PBXs.

With dynamic switched calls, the VoFR system includes dial-plan information that is used to process and route the calls based on the telephone numbers dialed by the callers. The dial-plan information is contained within dial-peer entries.

Static FRF.11 Trunks

FRF.11 trunks allow for standards-based vendor interoperability by specifying the frame format and coder types to be used when sending voice traffic through a Frame Relay network; however, FRF.11 includes no specifications for end-to-end negotiation, call setup process, or any other form of communication between the Frame Relay nodes. As a result, static FRF.11 trunks must be set up by manually configuring each router within the voice trunk path with compatible parameters---a voice port and a specific subchannel on a data-link connection identifier (DLCI) are explicitly bound on each end router. Signalling information is packed and sent transparently end to end.

There is no possibility of automatic enforcement of compatible configuration parameters between the two ends of an FRF.11-based call. For example, it is possible to incorrectly configure the two ends of an FRF.11 call using different and incompatible speech compression CODECs. In this situation, the call will exist and voice packets will be sent and received, but no usable voice path will be created.

When configured, a static FRF.11 trunk remains up until the voice port or serial port is shut down, or until a network disruption occurs. The FRF.11 specification does not include any standardized methods for performing Operation, Administration, and Maintenance (OAM) functions. There is no standard protocol for detecting faults and providing rerouting of connection paths.

FRF.11 allows up to 255 subchannels to be multiplexed onto a single Frame Relay DLCI. The current implementation supports the multiplexing of a single data channel with many voice channels. (Subchannels 0 to 3 are reserved and cannot be configured either for voice or data.)

FRF.11 can only be used when an end-to-end permanent virtual circuit (PVC) is available between the voice ports at each end of the connection. At intermediate Frame Relay nodes, you must route the entire PVC; connection ID-based routing (individual Channel ID switching) is not supported. Because the entire PVC is routed, no prioritization of voice packets is possible at the intermediate Frame Relay nodes.

The connection trunk voice-port configuration command is used to establish a static FRF.11 trunk; the dial peer is configured using the session protocol frf11-trunk command, which invokes the FRF.11-compliant session protocol.

Note FRF.11 specifies that a device can pack multiple FRF.11 subframes within a single Frame Relay frame; however, the Cisco implementation of VoFR currently does not support multiple subframes within a frame.

Cisco-Switched VoFR

Cisco addresses the lack of end-to-end call parameter negotiation and call setup syntax in FRF.11 by implementing a proprietary Q.931-like session protocol running on a user-configurable channel identifier (CID) of an FRF.11-format multiplexed DLCI. The Cisco-switched VoFR protocol handles call setup and parameter negotiation for both endpoints and intermediate nodes within the (multihop) call path. The call setup mechanism originally implemented in the Cisco MC3810 is used; this mechanism can be used for either permanent switched (Cisco-trunk) or dynamic switched calls. The Cisco-switched VoFR protocol includes forwarding of the called telephone number and supports tandem switching of the call over multiple Frame Relay PVC hops.

A tandem node is an intermediate router node within the Frame Relay call path. Its purpose is to switch the frames from one PVC subchannel to another (from one VoFR dial peer onto another VoFR dial peer) as the frames traverse the network. Use of tandem router nodes also avoids the need to have complete dial-plan information present on every router.

The Cisco MC3810 also supports Voice over ATM (VoATM) and Voice over HDLC (VoHDLC). The Cisco MC3810 is able to tandem switch voice calls between VoFR, VoATM, and VoHDLC call legs. The Cisco 2600 series and 3600 series routers also support VoIP. It is not possible in this release to translate from the VoIP transport protocol to other protocols such as VoFR. As a result, a call coming in on a VoIP connection may not be (tandem) switched to a VoFR connection.

Dynamic Switched Calls

Dynamic switched calls are regular telephone calls in which the dial-plan-based call switching is performed by the Cisco router. The destination endpoint of the call is selected by the router based on the telephone number dialed and the dial-plan information contained in the dial-peer configuration entries. Contrast this implementation with permanent calls (Cisco-trunk calls), where the call endpoints are permanently fixed at configuration time.

The dial peer is configured using the session protocol cisco-switched dial-peer configuration command, which uses the Cisco proprietary session protocol.

Cisco-Trunk (Private Line) Calls

A Cisco-trunk (private line) call is basically a normal dynamic switched call of indefinite duration that uses a fixed destination telephone number and includes optional transparent end-to-end signalling. The telephone number of the destination endpoint is permanently configured into the router so that it always selects a fixed destination. Once established, either at boot-up or when configured, the call stays up until one of the voice ports or network ports is shut down, or until a network disruption occurs.

The connection trunk voice-port command is used to establish a Cisco-trunk call; the dial peer is configured using the session protocol cisco-switched command, which invokes the Cisco proprietary session protocol.

Frame Relay Fragmentation

Cisco has developed three methods of performing Frame Relay fragmentation, which are described in the following sections:

Frame Relay fragmentation can be configured in conjunction with VoFR or independently of it.

End-to-End FRF.12 Fragmentation

FRF.12 fragmentation is defined by the FRF.12 standard. The FRF.12 Implementation Agreement was developed to allow long data frames to be fragmented into smaller pieces and interleaved with real-time frames. In this way, real-time voice and non-real-time data frames can be carried together on lower speed links without causing excessive delay to the real-time traffic.

Use this fragmentation type when the PVC is not carrying voice, but is sharing the link with other PVCs that are carrying voice. The fragmentation header is only included for frames that are greater than the fragment size configured. As a result, FRF.12 is the recommended fragmentation to be used by VoIP.

Note VoIP packets should not be fragmented. However, VoIP packets can be interleaved with fragmented packets.

The Cisco 2600 series, 3600 series, and 7200 series routers and the Cisco MC3810 multiservice access concentrator support end-to-end fragmentation on a per-PVC basis. Fragmentation is configured through a map class, which can apply to one or many PVCs, depending on how the class is applied.

Frame Relay Fragmentation Using FRF.11 Annex C

When VoFR (FRF.11) and fragmentation are both configured on a PVC, the Frame Relay fragments are sent in the FRF.11 Annex C format.

This fragmentation is used when FRF.11 voice traffic is sent on the PVC, and it uses the FRF.11 Annex C format for data.

With FRF.11, all data packets contain fragmentation headers regardless of size. This form of fragmentation is not recommended for use with Voice over IP (VoIP).

Cisco Proprietary Voice Encapsulation

Cisco proprietary voice fragmentation was implemented on earlier releases of the Cisco MC3810 multiservice access concentrator. This fragmentation type is used on data packets on a PVC that is also used for voice traffic. When the vofr cisco command is configured on a DLCI and fragmentation is enabled on a map class, the Cisco 2600 series, 3600 series, and 7200 series routers can interoperate as tandem nodes (but cannot perform call termination) with Cisco MC3810 concentrators running Cisco IOS releases prior to 12.0(3)XG or 12.0(4)T.

On the Cisco 2600, 3600, and 7200 series routers, entering the vofr cisco command is the only method for configuring Cisco proprietary voice encapsulation. You must then configure a map class to enable voice traffic on the PVCs.

On the Cisco MC3810, you have two methods for configuring Cisco proprietary voice encapsulation:

Configuring the map class to support proprietary voice encapsulation by entering the vofr cisco command on the DLCI.

: When using this method, you must configure a map class to enable voice traffic.

Configuring a Frame Relay DLCI to support proprietary voice encapsulation by entering the frame-relay interface-dlci voice-encap command.

: When using this method, you can configure either an individual DLCI or configure a map class to enable voice traffic. This command is not supported on the Cisco 2600 and 3600 series routers.

These commands are mutually exclusive on the interface and each provides different advantages. The vofr cisco command uses weighted fair queueing for controlling the data flows. The frame-relay interface-dlci voice-encap provides only simple first-in, first-out (FIFO) queueing and does not provide prioritization between nonvoice (data) flows. The command does provide prioritization for voice flows.

Because the frame-relay interface-dlci voice-encap command provides only FIFO queueing, this method has less overhead than the vofr cisco command. However, the vofr cisco command provides greater control over the queueing mechanism.

Frame Relay Fragmentation Conditions and Restrictions

When Frame Relay fragmentation is configured, the following conditions and restrictions apply:

Hardware compression is not supported in this release.
Weighted fair queueing at the PVC level is the only queueing strategy that can be used.
Frame Relay traffic shaping (FRTS) must be configured to enable Frame Relay fragmentation.
VoFR frames are never fragmented, regardless of size.
When using end-to-end FRF.12 fragmentation, the VoIP packets will not include the FRF.12 header, provided the size of the VoIP packet is smaller than the fragment size configured. However, when using FRF.11 Annex C or Cisco proprietary fragmentations, VoIP packets will include the fragmentation header.
If fragments arrive out of sequence, packets are dropped.

Note Fragmentation is performed after frames are removed from the WFQ.

FRF.ll Implementation Agreement Support

VoFR provides support for different FRF.11 forum features depending on the hardware platform used (see Table 21).

Table 21: FRF.11 Support

FRF.11 Forum Features Cisco MC3810 Cisco 2600/3600 Series

Class 1 - Compliance Requirements (sec. 4.1)

Not supported

Not supported

Class 2 - Compliance Requirements (sec. 4.2)

Supported

Supported

Annex.A - Dialed Digits Transfer Syntax

Supported

Supported

Annex.B - Signalling Bit Transfer Syntax

Supported

Supported

Annex.C - Data Transfer Syntax

Supported

Supported

Annex.D - Fax Relay Transfer Syntax

Supported

Supported

Annex.E - CS-ACELP Transfer Syntax (G.729/G.729A)
- Sequence Number
- Packing Factor

Supported
Supported
Supported

Supported
Supported
Supported

Annex.F - Generic PCM/ADPCM Voice Transfer Syntax

Supported

Supported

Annex.G - G.727 Discard-Eligible E-ADPCM Voice Transfer Syntax

Not supported

Not supported

Annex.H - G.728 LD-CELP Transfer Syntax

Not supported

Supported

Annex.I - G.723.1 Dual Rate Speech Coder

Not supported

Supported

Transmission and reception of multiple subframes within a single Frame Relay frame

Not supported

Not supported

FRF.11 Forum Features	Cisco MC3810	Cisco 2600/3600 Series
Class 1 - Compliance Requirements (sec. 4.1)	Not supported	Not supported
Class 2 - Compliance Requirements (sec. 4.2)	Supported	Supported
Annex.A - Dialed Digits Transfer Syntax	Supported	Supported
Annex.B - Signalling Bit Transfer Syntax	Supported	Supported
Annex.C - Data Transfer Syntax	Supported	Supported
Annex.D - Fax Relay Transfer Syntax	Supported	Supported
Annex.E - CS-ACELP Transfer Syntax (G.729/G.729A) - Sequence Number - Packing Factor	Supported Supported Supported	Supported Supported Supported
Annex.F - Generic PCM/ADPCM Voice Transfer Syntax	Supported	Supported
Annex.G - G.727 Discard-Eligible E-ADPCM Voice Transfer Syntax	Not supported	Not supported
Annex.H - G.728 LD-CELP Transfer Syntax	Not supported	Supported
Annex.I - G.723.1 Dual Rate Speech Coder	Not supported	Supported
Transmission and reception of multiple subframes within a single Frame Relay frame	Not supported	Not supported

Prerequisites Tasks

Before you can configure your router to use VoFR, you must first perform the following tasks:

Establish a working IP network. For more information about configuring IP, refer to the "IP Overview," "Configuring IP Addressing," and "Configuring IP Services" chapters in the Cisco IOS IP and IP Routing Configuration Guide.
Configure the clock source for the Cisco MC3810 interfaces. For more information, refer to the "Configuring Synchronous Clocking on the Cisco MC3810" appendix.
Complete your company dial plan.
Establish a working telephony network based on your company dial plan:
- Integrate your dial plan and telephony network into your existing Frame Relay network topology. Make routing and/or dialing transparent to the user---for example, avoid secondary dial tones from secondary switches, where possible.
- Contact your PBX vendor for instructions about how to reconfigure the appropriate PBX interfaces.

After you have analyzed your dial plan and decided how to integrate it into your existing Frame Relay network, you are ready to configure your network devices to support VoFR.

VoFR Configuration Task List

VoFR has a set of core tasks. To configure VoFR, perform the following tasks:

Configuring Voice Ports

This section describes how to configure voice ports to support VoFR.

This section is divided into the following procedures:

Voice ports provide support for three basic voice signalling formats:

Foreign Exchange Office (FXO) interface. This interface allows a connection to be directed to the PSTN central office. The FXO interface also allows a connection to be directed to a standard PBX interface if the local telecommunications authority permits. This interface is of value for Off-Premise Extension (OPX) applications.
Foreign Exchange Station (FXS) interface. This interface allows connection for basic telephone equipment and keysets, and supplies ring, voltage, and dial tone.
"Ear and Mouth" (or "RecEive and TransMit") (E&M) interface allows connection for PBX trunk lines (tie lines). E&M is a signalling technique for 2-wire and 4-wire telephone and trunk interfaces. In addition, the Cisco MC3810 supports Mercury Exchange Limited channel associated signalling (MELCAS) for E&M voice ports.

In general, voice-port commands define the characteristics associated with a particular voice-port signalling type. Under most circumstances, the default voice-port command values are adequate to configure FXO and FXS ports to transport voice data using the Cisco MC3810. Because of the inherent complexities involved with PBX networks, E&M ports might need specific voice-port values configured, depending on the specifications of the devices in your telephony network.

Voice Ports on the Cisco 2600 and 3600 Series Routers

The Cisco 2600 and 3600 series routers provide analog and digital voice ports. The type of signalling associated with these analog voice ports depends on the interface module installed into the device. The Cisco 3600 series router supports either a 2-port or 4-port voice network module (VNM); VNMs can hold either two or four voice interface cards (VICs).

Each VIC is specific to a particular signalling type; therefore, VICs determine the type of signalling for the voice ports on that particular VNM. Even though VNMs can hold multiple VICs, each VIC on a VNM must conform to the same signalling type. For more information about the physical characteristics of VNMs and VICs or how to install them, refer to the installation document, Voice Network Module and Voice Interface Card Configuration Note, that came with your VNM.

Voice Ports on the Cisco MC3810

The Cisco MC3810 hardware features different hardware configuration options for voice ports:

Six analog voice interfaces--- The Cisco MC3810 version with the analog voice module (AVM) supports up to six analog voice personality modules (APMs) with each voice module supporting a single signalling type. Each voice personality module maps to a single analog voice port. The FXO, FXS, and E&M voice modules can be installed in any combination.
One digital voice module (DVM)--- The Cisco MC3810 version with the DVM provides support for up to 24 voice channels, one for each voice port. Depending on whether the controller is T1 or E1, different DS0 voice channels are used. The DVM supports CAS for the following types: FXO, FXS, and E&M. For E&M signalling, the DVM also supports E1 MELCAS, a standard used primarily in the United Kingdom.
With the optional BRI voice module (BVM) installed, the Cisco MC3810 multiservice access concentrator provides four ISDN BRI ports for connection to ISDN PBXs (private integrated services network exchanges, or PINXs). The BVM has four ISDN BRI ports for voice traffic. Each BRI port supports two voice channels (ISDN B channels) and one signalling channel (ISDN D channel).

Note If the BVM is installed, the speed (clock rate) of serial port 1 of the Cisco MC3810 is limited to a maximum of 192 kbps. This restriction assumes that the Multiflex Trunk (MFT) is installed in slot 3 on the Cisco MC3810. If the MFT is not installed, serial port 1 will not operate.

Table 22 lists the valid slot and port numbers for the different voice interfaces.

Table 22: Voice Interface Slot and Port Number

Interface Type Slot Valid Port Numbers

Analog voice module (AVM)

1

1-6

Digital voice module (DVM)

1

Digital T1: 1-24

Digital E1: 1-15 and 17-31

Multiflex Trunk (MFT)

0

Digital T1: 1-24

Digital E1: 1-15 and 17-31

Interface Type	Slot	Valid Port Numbers
Analog voice module (AVM)	1	1-6
Digital voice module (DVM)	1	Digital T1: 1-24 Digital E1: 1-15 and 17-31
Multiflex Trunk (MFT)	0	Digital T1: 1-24 Digital E1: 1-15 and 17-31

Note The voice-port number designations start with 1. Unlike serial port interfaces and interfaces on other Cisco products, there is no port 0 for voice ports.

Configuring FXO or FXS Voice Ports

Under most circumstances the default voice-port values are adequate for both FXO and FXS voice ports. To configure FXO and FXS voice ports, use the following commands beginning in global configuration mode:

Verifying the FXO or FXS Voice Port Configuration

You can check the validity of your voice-port configuration by performing the following tasks:

Pick up the handset of an attached telephony device and check for a dial tone.
If you have dial tone, check for DTMF detection. If the dial tone stops when you dial a digit, then the voice port is most likely configured properly.
To verify that the data configured is correct, use the show voice port command.
To verify the current status of all digital signal processor (DSP) voice channels, use the show voice dsp command.
To verify the call status for all voice ports, use the show voice call summary command .

Troubleshooting Tips

If you are having trouble connecting a call and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:

Ping the associated IP address to confirm connectivity. If you cannot successfully ping your destination, refer to the Cisco IOS IP and IP Routing Configuration Guide.
To make sure that the port is enabled, use the show voice port command. If the port is offline, use the no shutdown command.
Check the dial-peer configuration.
Check the Frame Relay configuration.
Check whether the voice network module has been correctly installed.

Fine-Tuning FXO and FXS Voice Ports

Depending on your particular network, you may need to adjust voice parameters involving timing, input gain, and output attenuation for FXO or FXS voice ports. Collectively, these commands are referred to as voice-port tuning commands.

NoteIn most cases, the default values for voice-port tuning commands will be sufficient.

To fine-tune FXO or FXS voice ports, use the following commands beginning in global configuration mode:

	Command	Purpose
Step1	For Cisco 2600 and 3600 series analog voice ports: router(config)# voice-port slot/subunit/port For Cisco 2600 and 3600 series digital voice ports: router(config)# voice-port slot/port:ds0-group For Cisco MC3810 series analog voice ports: router(config)# voice-port slot/port For Cisco MC3810 series digital voice ports: router(config)# voice-port slot:ds0-group	Identifies the voice port you want to configure and enters voice-port configuration mode.
Step2	Router(config-voiceport)# input gain value	Specifies (in decibels) the amount of gain to be inserted at the receiver side of the interface. Acceptable values are from -6 to 14.
Step3	Router(config-voiceport)# output attenuation value	Specifies (in decibels) the amount of attenuation at the transmit side of the interface. Acceptable values are from 0 to 14.
Step4	Router(config-voiceport)# echo-cancel enable	Enables echo cancellation of voice that is sent out the interface and received back on the same interface.
Step5	Router(config-voiceport)# echo-cancel coverage value	Adjust the size (in milliseconds) of the echo cancel. Acceptable values are 16, 24, and 32.
Step6	Router(config-voiceport)# timeouts initial seconds	Configures the initial timeout value. The initial timeout value specifies the number of seconds the system waits for the caller to input the first digit of the dialed digits. The default is 10 seconds.
Step7	Router(config-voiceport)# timeouts interdigit seconds	Configures the interdigit timeout value. The timeouts interdigit value specifies the number of seconds the system waits (after the caller has input the initial digit) for the caller to input a subsequent digit of the dialed digits. The default is 10 seconds.
Step8	Router(config-voiceport)# timing digit milliseconds	If the voice-port dial type is DTMF, configure the DTMF digit signal duration. The range of the DTMF digit signal duration is from 50to100 milliseconds. The default is 100.
Step9	Router(config-voiceport)# timing inter-digit milliseconds	If the voice-port dial type is DTMF, configure the DTMF interdigit signal duration. The range of the DTMF interdigit signal duration is from 50 to 500milliseconds. The default is 100.
Step10	Router(config-voiceport)# timing pulse-digit milliseconds	If the voice-port dial type is pulse, configure the pulse digit signal duration. The range of the pulse digit signal duration is from10 to 20 milliseconds. The default is 20.
Step11	Router(config-voiceport)# timing pulse-inter-digit milliseconds	If the voice-port dial type is pulse, configure the pulse interdigit signal duration. The range of the pulse interdigit signal duration is from 100 to 1000 milliseconds. The default is 500.
Step12	Router(config-voiceport)# impedance {600r \| 600c \| 900r \| 900c}	(For FXO ports only) Configures the impedance. The default is 600 ohms real.
Step13	Router(config-voiceport)# loss-plan {plan1 \| plan2 \| plan3 \| plan4 \| plan7 \| plan8 \| plan9}	(For analog ports on the Cisco MC3810 only) Specifies the loss plan for this voice port according to the signal level requirements for the DSP and the PBX. The default is plan1, which provides the following gain offset levels: FXO---A-D gain = 0 dB, D-A gain = 0 dB FXS---A-D gain = -3 dB, D-A gain = -3 dB
Step14	Router(config-voiceport)# ring cadence [on1 \| off1] [on2 \| off2] [on3\|off3] [on4 \| off4] [on5 \| off5] [on6 \| off6]	(For FXS ports only) Specifies the local ring cadence for the FXS voice port. Using this command, specify the on and off pulses for the ring. The ring cadence differs depending on the local region. The units are in 100-millisecond units.

NoteAfter you change any voice-port command, it is a good idea to cycle the port by using the shutdown and no shutdown commands.

Configuring E&M Voice Ports

Unlike FXO and FXS voice ports, the default E&M voice-port parameters most likely will not be sufficient to enable voice data transmission over your network.

NoteE&M voice-port values must match those of the PBX to which it is connected. Refer to the documentation that came with your specific PBX for the appropriate E&M voice-port configuration command values.

To configure E&M voice ports , use the following commands beginning in global configuration mode:

Command Purpose

Step1

For Cisco 2600 and 3600 series analog voice ports:

router(config)# voice-port slot/subunit/port
For Cisco 2600 and 3600 series digital voice ports:

router(config)# voice-port slot/port:ds0-group
For Cisco MC3810 series analog voice ports:

router(config)# voice-port slot/port
For Cisco MC3810 series digital voice ports:

router(config)# voice-port slot:ds0-group
Identifies the voice port you want to configure and enters voice-port configuration mode.

Step2

Router(config-voiceport)# dial-type {dtmf | pulse}
Select the appropriate dial type for out-dialing.

Step3

Router(config-voiceport)# operation {2-wire | 4-wire}
Select the appropriate cabling scheme for this voice port.

Step4

Router(config-voiceport)# type {1 | 2 | 3 | 5}
Select the appropriate E&M interface type.

Type 1 indicates the following lead configuration:

E---output, relay to ground
M---input, referenced to ground

Type 2 indicates the following lead configuration:

E---output, relay to SG
M---input, referenced to ground
SB---feed for M, connected to -48V
SG---return for E, galvanically isolated from ground

Type 3 indicates the following lead configuration:

E---output, relay to ground
M---input, referenced to ground
SB---connected to -48V
SG---connected to ground

Type 5 indicates the following lead configuration:

E---output, relay to ground
M---input, referenced to -48V.

Step5

Router(config-voiceport)# signal {wink-start | immediate | delay-dial}
Configures the signalling type for E&M voice ports. The default is wink-start.

Step6

Router(config-voiceport)# impedance {600c | 600r | 900c | complex1 | complex2}
Specifies a terminating impedance. This value must match the specifications from the telephony system to which this voice port is connected.

Step7

Router(config-voiceport)# connection {plar| tie-line|plar-opx} string
(Optional) Configures the voice-port connection mode type and the destination telephone number.

The plar value is used for PLAR connections. The tie-line value on the Cisco MC3810 is used for a tie-line connection to a PBX. The plar-opx value on the CiscoMC3810 is used for PLAR OPX, to allow the local voice port to provide a local response before the remote voice port receives an answer. If you select the plar-opx value, you must also configure the voice confirmation-tone command.

NoteIf you will be configuring Cisco-trunk permanent connections, do not configure this command now. See the "Configuring Cisco-Trunk (Private Line) Permanent Calls" section later in this chapter.

Step8

Router(config-voiceport)# compand-type {u-law | a-law}
(Digital voice ports only) Configures the companding standard used to convert between analog and digital signals in PCM systems.

Step9

Router(config-voiceport)# cptone locale
Configures the appropriate call progress tone for the local region.

The default for this command is the us keyword. For a list of supported countries, refer to the Cisco IOS Multiservice Applications Command Reference.

Step10

Router(config-voiceport)# music-threshold number
(Optional) Specifies the threshold (in decibels) for on-hold music. Valid entries are from -70 to -30.

Step11

Router(config-voiceport)# description string
(Optional) Attaches descriptive text about this voiceport connection.

Step12

Router(config-voiceport)# codec {g729r8 | g729ar8 | g726r32 | g711alaw | g711ulaw}
(Optional for Cisco MC3810 only) Configures the voice-port compression mode. The g729ar8 value is the default and is recommended.

The g729ar8 compression mode can support a maximum of 24 simultaneously active on-net voice calls and the g729r8 value can only support a maximum of 12. The g729 compression modes have a nominal data rate of 8 kbps.

NoteThe Cisco MC3810 also supports configuring the codec command on the dial peer. Because the Cisco 2600 and 3600 series routers support the codec command on the dial peer only, it is recommended you configure this command on the dial peer.

Step13

Router(config-voiceport)# vad
(Optional for Cisco MC3810 only) Enables voice activity detection (VAD).

NoteThe Cisco MC3810 also supports configuring the vad command on the dial peer as well. Because the Cisco 2600 and 3600 series routers support the vad command on the dial peer only, it is recommended you configure this command on the dial peer.

Step14

Router(config-voiceport)# voice confirmation-tone
(Optional for Cisco MC3810 only) If the voice port is configured for connection plar-opx for OPX, disable the two-beep confirmation tone that a caller hears when picking up the handset.

Verifying the E&M Voice Port Configuration

You can verify your voice-port configuration by performing the following tasks:

Pick up the handset of an attached telephony device and check for a dial tone.
If you have dial tone, check for DTMF detection. If the dial tone stops when you dial a digit, then the voice port is most likely configured properly.
To verify that the data configured is correct, use the show voice port command.
To verify the current status of all DSP voice channels on the Cisco MC3810, use the show voice dsp command.
To verify the call status for all voice ports, on the Cisco MC3810, use the show voice call summary command.

Troubleshooting Tips

If you are having trouble connecting a call and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:

Ping the associated IP address to confirm connectivity. If you cannot successfully ping your destination, refer to the Cisco IOS IP and IP Routing Configuration Guide.
To make sure that the port is enabled, use the show voice port command. If the port is offline, use the no shutdown command.
Make sure that the values pertaining to your specific PBX setup, such as timing and type, are correct.

Fine-Tuning E&M Voice Ports

Depending on your particular network, you may need to adjust voice parameters involving timing, input gain, and output attenuation for E&M voice ports. Collectively, these commands are referred to as voice-port tuning commands.

NoteIn most cases, the default values for voice-port tuning commands will be sufficient.

To fine-tune E&M voice ports , use the following commands beginning in global configuration mode:

	Command	Purpose
Step1	For Cisco 2600 and 3600 series analog voice ports: router(config)# voice-port slot/subunit/port For Cisco 2600 and 3600 series digital voice ports: router(config)# voice-port slot/port:ds0-group For Cisco MC3810 series analog voice ports: router(config)# voice-port slot/port For Cisco MC3810 series digital voice ports: router(config)# voice-port slot:ds0-group	Identifies the voice port you want to configure and enters voice-port configuration mode.
Step2	Router(config-voiceport)# input gain value	Specifies (in decibels) the amount of gain to be inserted at the receiver side of the interface. Acceptable values are from -6 to 14.
Step3	Router(config-voiceport)# output attenuation value	Specifies (in decibels) the amount of attenuation at the transmit side of the interface. Acceptable values are from 0 to 14.
Step4	Router(config-voiceport)# echo-cancel enable	Enables echo cancellation of voice that is sent out the interface and received back on the same interface.
Step5	Router(config-voiceport)# echo-cancel coverage value	Adjusts the size (in milliseconds) of the echo cancel. Acceptable values are 16, 24, and 32.
Step6	Router(config-voiceport)# non-linear	Enables nonlinear processing, which shuts off any signal if no near-end speech is detected. (Nonlinear processing is used with echo cancellation.)
Step7	Router(config-voiceport)# timeouts initial seconds	Configures the initial timeout value. The initial timeout value specifies the number of seconds the system waits for the caller to input the first digit of the dialed digits. The default is 10 seconds.
Step8	Router(config-voiceport)# timeouts interdigit seconds	Configures the interdigit timeout value. The timeouts interdigit value specifies the number of seconds the system waits (after the caller has input the initial digit) for the caller to input a subsequent digit of the dialed digits. The default is 10 seconds.
Step9	Router(config-voiceport)# timeouts wait-release {value \| infinity}	Configures the timeout value for releasing voice ports. This command limits the duration that a voice port stays in the call failure state while the Cisco MC3810 sends a busy tone, reorder tone, or out-of-service tone to the port.
Step10	Router(config-voiceport)# timing digit milliseconds	If the voice-port dial type is DTMF, configure the DTMF digit signal duration. The range of the DTMF digit signal duration is from 50to100 milliseconds. The default is 100.
Step11	Router(config-voiceport)# timing inter-digit milliseconds	If the voice-port dial type is DTMF, configure the DTMF interdigit signal duration. The range of the DTMF interdigit signal duration is from 50 to 500milliseconds. The default is 100.
Step12	Router(config-voiceport)# timing pulse-digit milliseconds	If the voice-port dial type is pulse, configure the pulse digit signal duration. The range of the pulse digit signal duration is from 10 to 20 milliseconds. The default is 20.
Step13	Router(config-voiceport)# timing pulse-inter-digit milliseconds	If the voice-port dial type is pulse, configure the pulse interdigit signal duration. The range of the pulse interdigit signal duration is from 100 to 1000milliseconds. The default is 500.
Step14	Router(config-voiceport)# timing wink-duration milliseconds	Configures the timing wink-duration value. This value sets the wink signal duration for a wink-start signal. This value applies only if the signal command is set to "wink-start." The range is from 100 to 400milliseconds and the default is 200.
Step15	Router(config-voiceport)# timing wink-wait milliseconds	Configures the timing wink-wait value. This value sets the wink wait duration for a wink-start signal. This value applies only if the signal command is set to "wink-start." The range is from 100 to 5000milliseconds and the default is 200.
Step16	Router(config-voiceport)# timing clear-wait milliseconds	Configures the timing clear-wait value. This value sets the amount of time between the inactive seizure signal and the call being cleared. The range is from 100 to 2000 milliseconds and the default is 400.
Step17	Router(config-voiceport)# timing delay-duration milliseconds	Configures the timing delay-duration value. This value sets the delay signal duration for delay dial signalling. This value applies only if the signal command is set to "delay-dial." The range is from 100 to 5000 milliseconds and the default is 140.
Step18	Router(config-voiceport)# timing delay-start milliseconds	Configures the timing delay-start value. This value sets the delay interval between the generation of the delay-start signal from incoming detection seizure. This value applies only if the signal command is set to "delay-dial." The range is from 100 to 290milliseconds and the default is 150.
Step19	Router(config-voiceport)# timing percentbreak percent	Configures the timing percent-break value. This value sets the percentage of the break period for a dialing pulse. The default is 50 percent.

NoteAfter you change any voice-port command, it is a good idea to cycle the port by using the shutdown and no shutdown commands.

Activating the Voice Port

After you have configured the voice port, you need to activate the voice port to bring it online. In fact it is a good idea to cycle the port---meaning to shut the port down and then bring it online again.

To activate a voice port, use the following command in voice-port configuration mode:

Command Purpose

Router(config-voiceport)# no shutdown
Activate the voice port.

Command	Purpose
Router(config-voiceport)# no shutdown	Activate the voice port.

To cycle a voice port, use the following commands in voice-port configuration mode:

Command Purpose

Step1

Router(config-voiceport)# shutdown
Deactivate the voice port.

Step2

Router(config-voiceport)# no shutdown
Activate the voice port.

	Command	Purpose
Step1	Router(config-voiceport)# shutdown	Deactivate the voice port.
Step2	Router(config-voiceport)# no shutdown	Activate the voice port.

NoteIf you will not use a voice port, shut it down.

Configuring Frame Relay to Support VoFR

This section describes preliminary Frame Relay configuration tasks that are necessary to support VoFR:

Configuring a Map Class to Support VoFR

Before configuring a Frame Relay DLCI for voice traffic, you must create a Frame Relay map class and configure it to support voice traffic. Configuring a Frame Relay map class is required because the voice bandwidth, fragmentation size, and traffic shaping attributes are configured on the map class. These attributes are required for sending voice traffic on the PVC.

This section is divided into the procedures described in the following sections:

A map class applies to a single DLCI or to a group of DLCIs, depending on how the class has been applied to the virtual circuit. If you have a large number of PVCs to configure, you can assign the PVCs the same traffic shaping properties without statically defining the values for each PVC. You can create multiple map classes with different variables for each map class.

NoteIf you configure the frame-relay interface-dlci voice-encap command on a CiscoMC3810, then configuring a Frame Relay map class is not required. For procedures for assigning this command to a DLCI in conjunction with configuring connection types, see the "Configuring Switched Calls on a Cisco MC3810" section and the "Configuring Cisco Trunk Permanent Calls on a Cisco MC3810" section later in this chapter.

Configuring a Frame Relay Map Class to Support Voice Traffic

To configure a Frame Relay map class to support voice traffic on a single DLCI or a group of DLCIs, use the following commands beginning in global configuration mode:

Command Purpose

Step1

Router(config)# map-class frame-relay map-class-name
Creates a map class name you will assign to a group of PVCs. The map class name must be unique.

Step2

Router(config-map-class)# frame-relay voice bandwidth bps [queue depth]
Enters the bandwidth in bits per second (bps), which will determine the number of voice calls allowed on the DLCIs where this map class is associated to. We recommend that this value be set to no higher than the minimum committed information rate (CIR) if you do not want to impact voice quality when burst is being sent. The valid range is from 8000 to 45,000,000 bps.

This command must be configured for voice calls to take place. The default for this command is 0, which disables all voice calls.

When this command is entered, a special queue is created for voice packets only so that time-sensitive voice packets have preference over data packets. The optional queue depth keyword and argument allow you to specify the size of this voice queue. The default size is 100.

For more information on determining the amount of voice bandwidth to set, see the section "Calculating Voice Bandwidth" after this procedure.

NoteThis command does not apply if you are configuring the frame-relay interface-dlci voice-encap command on the CiscoMC3810.

To configure the map class to support FRF.12 fragmentation, see the "Configuring a Frame Relay Map Class to Support FRF.12 Fragmentation" section later in this chapter. To configure the map class to support traffic shaping if you want to send both voice traffic and data traffic on the same PVC, see the "Configuring a Frame Relay Map Class for Traffic Shaping Parameters" section later in this chapter.

Calculating Voice Bandwidth

The frame-relay voice-bandwidth map-class command is used to configure how much bandwidth is reserved for voice traffic. If not enough reserved voice bandwidth remains on the PVC, then any new call attempted will be rejected.

When you calculate the amount of voice bandwidth to allocate to voice, the overall bandwidth calculation must include the voice packetization overhead and not just the raw compressed speech codec bandwidth. For VoFR voice packets, there are a total of 6 or 7 bytes total overhead per packet (including standard Frame Relay headers and flags). For subchannels (CIDs) less than number 64, the overhead is 6 bytes. For subchannels greater than or equal to number 64, the overhead is 7 bytes. Add one byte if voice sequence numbers are enabled in the voice packets.

To determine the required voice bandwidth, use the following calculation:

required_bandwidth = codec_bandwidth * (1 + overhead/payload_size)

This calculation addresses the amount of bandwidth consumed on the physical network interface. This figure does not necessarily represent the amount of connection bandwidth used within the Frame Relay network itself, which may be higher due to the overhead of switching small packets.

When using 30-millisecond (ms) duration voice packets, an approximate general rule is to add 2000 bps overhead to the raw voice compressed speech codec rate. With the 32 kbps G.726 ADPCM speech coder, a 30-ms speech frame uses 120 bytes voice payload plus 6 to 7 bytes overhead, and the overall bandwidth requirement is around 34 kbps for each call.

The codec command is configured as part of the dial peer configuration procedures in the "Configuring Dial Peers" section later in this chapter.

Configuring a Frame Relay Map Class to Support FRF.12 Fragmentation

To configure the map class to support FRF.12 fragmentation, use the following commands in map-class configuration mode:

Command Purpose

Step1

Router(config-map-class)# frame-relay fragment fragment_size
Configures Frame Relay fragmentation for the map class. The fragment_size argument defines the payload size of a fragment, and excludes the Frame Relay headers and any Frame Relay fragmentation header. The valid range is from 16 to 1600 bytes, and the default is 53.

The fragment_size argument should be less than or equal to the Message Transmission Unit (MTU) size.

Set the fragmentation size such that the largest data packet is not larger than the voice packets.

Step2

Router(config-map-class)# frame-relay fair-queue [Congestive_Discard_Threshold [Number_Dynamic_Conversation_ Queues [Number_Reservable_Conversation_Queues [Max_Buffer_Size_for_Fair_Queues]]]]
Enables WFQ for the map class.

This command applies to both fragmented and nonfragmented data. This command is equivalent to the fair-queue interface command but applies to a Frame Relay PVC.

The default Congestive_Discard_Threshold is 64. The default Number_Dynamic_Conversation_Queues is 16. The default Number_Reservable_Conversation_Queues is 2. The default for Max_Buffer_Size_for_Fair_Queues is 600.

	Command	Purpose
Step1	Router(config-map-class)# frame-relay fragment fragment_size	Configures Frame Relay fragmentation for the map class. The fragment_size argument defines the payload size of a fragment, and excludes the Frame Relay headers and any Frame Relay fragmentation header. The valid range is from 16 to 1600 bytes, and the default is 53. The fragment_size argument should be less than or equal to the Message Transmission Unit (MTU) size. Set the fragmentation size such that the largest data packet is not larger than the voice packets.
Step2	Router(config-map-class)# frame-relay fair-queue [Congestive_Discard_Threshold [Number_Dynamic_Conversation_ Queues [Number_Reservable_Conversation_Queues [Max_Buffer_Size_for_Fair_Queues]]]]	Enables WFQ for the map class. This command applies to both fragmented and nonfragmented data. This command is equivalent to the fair-queue interface command but applies to a Frame Relay PVC. The default Congestive_Discard_Threshold is 64. The default Number_Dynamic_Conversation_Queues is 16. The default Number_Reservable_Conversation_Queues is 2. The default for Max_Buffer_Size_for_Fair_Queues is 600.

NoteWhen Frame Relay fragmentation is configured, WFQ is mandatory. If a map class is configured for Frame Relay fragmentation and the queueing type on that map class is not fair-queue, the configured queueing type is automatically overridden by WFQ with the default values.

To configure the map class to support traffic shaping if you want to send both voice traffic and data traffic on the same PVC, see the next section, "Configuring a Frame Relay Map Class for Traffic Shaping Parameters."

Configuring a Frame Relay Map Class for Traffic Shaping Parameters

When you configure a Frame Relay PVC to support voice traffic, you must ensure that the carrier can accommodate the traffic rate or profile sent on the PVC. If too much traffic is sent at once, the carrier might discard frames, which causes disruptions to real-time voice traffic. The carrier might also deal with traffic bursts by queueing up the bursts and delivering them at a metered rate. Excessive queueing also causes disruption to real-time voice traffic.

To compensate for this condition, traffic shaping is required if both voice traffic and data traffic are sent over the same PVC.

NoteWhen you configure the outgoing Excess Burst (Be) size, the Committed Burst (Bc) size, and the CIR values, obtain the appropriate values from your carrier. The values configured on the router must match those of the carrier. Traffic shaping is necessary to prevent your carrier from discarding discard eligible (DE) bits on ingress or to prevent excessive burst data from affecting voice quality.

To configure a Frame Relay map class to determine the traffic shaping characteristics for one or more DLCIs, use the following commands in map-class configuration mode:

Command Purpose

Step1

Router(config-map-class)# frame-relay bc out bits
Configures the outgoing Committed Burst size for this group of PVCs. Configure the bits value to a minimum of 1000 for voice traffic.

Configure the Committed Burst size to match that of your carrier to prevent the carrier from discarding DE bits on ingress. This command is required.

Step2

Router(config-map-class)# frame-relay be out bits
Configures the outgoing Excess Burst size for this group of PVCs.

Configure the outgoing Excess Burst size to match that of your carrier to prevent the carrier from discarding DE bits on ingress. This command is required.

Step3

Router(config-map-class)# frame-relay min-cir {in | out} bps
Configures the minimum acceptable incoming or outgoing CIR for this group of PVCs.

Step4

Router(config-map-class)# frame-relay cir out bits
Configures the outgoing excess CIR for this group of PVCs.

Configure the CIR size to match your carrier to prevent the carrier from discarding DE bits on ingress. This command is required.

Step5

Router(config-map-class)# frame-relay cir in bits
(Optional) Configures the incoming CIR size for this group of PVCs.

Step6

Router(config-map-class)# frame-relay adaptive shaping becn
(Optional) Configures the adaptive traffic rate adjustment to support backward explicit congestion notification (BECN) on this group of PVCs.

	Command	Purpose
Step1	Router(config-map-class)# frame-relay bc out bits	Configures the outgoing Committed Burst size for this group of PVCs. Configure the bits value to a minimum of 1000 for voice traffic. Configure the Committed Burst size to match that of your carrier to prevent the carrier from discarding DE bits on ingress. This command is required.
Step2	Router(config-map-class)# frame-relay be out bits	Configures the outgoing Excess Burst size for this group of PVCs. Configure the outgoing Excess Burst size to match that of your carrier to prevent the carrier from discarding DE bits on ingress. This command is required.
Step3	Router(config-map-class)# frame-relay min-cir {in \| out} bps	Configures the minimum acceptable incoming or outgoing CIR for this group of PVCs.
Step4	Router(config-map-class)# frame-relay cir out bits	Configures the outgoing excess CIR for this group of PVCs. Configure the CIR size to match your carrier to prevent the carrier from discarding DE bits on ingress. This command is required.
Step5	Router(config-map-class)# frame-relay cir in bits	(Optional) Configures the incoming CIR size for this group of PVCs.
Step6	Router(config-map-class)# frame-relay adaptive shaping becn	(Optional) Configures the adaptive traffic rate adjustment to support backward explicit congestion notification (BECN) on this group of PVCs.

If your Frame Relay map class configuration is complete, see the "Verifying Your Frame Relay Configuration" section later in this chapter. To begin your dial peer configuration, see either the "Preparing to Configure Voice Dial Peers" later in this chapter, or the "Configuring Dial Peers" section later in this chapter.

Verifying Your Frame Relay Configuration

You can check the validity of your Frame Relay configuration by performing the following tasks:

To show the status of your PVCs, use the show frame-relay pvc command.
To show statistics and information on the open subchannels, use the show frame-relay vofr [[interface] [dlci[cid]]] command. (This command does not display if the vofr cisco command is entered on the Cisco MC3810.)
To show the Frame Relay fragmentation configuration, use the show frame-relay fragment [interface numbe21r [dlci]] command.
If you are using Frame Relay traffic shaping, use the show traffic-shape queue command to display the traffic shaping information. Use the queue option to display the queueing statistics.

Preparing to Configure Voice Dial Peers

After you have analyzed your dial plan and decided how to integrate it into your existing network, you are ready to configure your network devices to support VoFR. The actual configuration procedure depends entirely upon the topology of your voice network, but, in general, you need to perform the tasks described in the following sections:

TimeSaverYou might want to configure the Frame Relay dial peers in a back-to-back configuration before separating them across the Frame Relay network. You can use a back-to-back configuration to test your VoFR and dial-peer configuration to determine if you can make a voice connection. Then, when you place both peers on the network, failure to make a voice connection will isolate the cause as a network problem.

Organizing Voice Network Information

After you have configured your Frame Relay network, you should collect all of the data directly related to each dial peer by creating a peer configuration table to prepare for configuring VoFR.

Creating a Peer Configuration Table

Specific information relative to each dial peer needs to be identified before you can configure VoFR. One way to identify this information is to create a peer configuration table.

Figure 65 shows a diagram of a small voice network in which Router No. 1 connects a small sales branch office to the main office through Router No. 2. Only two devices in the sales branch office that need to be established as dial peers: a telephone and a fax machine. Router No. 2 is the primary gateway to the main office; as such, it needs to be connected to the company's PBX. Two telephones and one fax machine connected to the PBX need to be established as dial peers in the main office.

Table 23 shows the peer configuration table for the example illustrated in Figure 65.

Figure65: Sample VoFR Network

Table23: Peer Configuration Table for Sample VoFR Network

Dial Peer Extension Prefix Destination Pattern Type Voice Port Session Target

Router No. 1

1

61111

---

13107661111

POTS

1/0/0

---

2

62222

---

13107662222

POTS

1/0/1

---

10

---

---

1310767....

VOFR

---

S0/0 150

Router No. 2

11

---

---

1310766....

VOFR

---

S0/0 150

3

73333

7

1310767....

POTS

1/0/0

---

4

74444

7

1310767....

POTS

1/0/0

---

5

75555

7

1310767....

POTS

1/1

---

Dial Peer	Extension	Prefix	Destination Pattern	Type	Voice Port	Session Target
Router No. 1
1	61111	---	13107661111	POTS	1/0/0	---
2	62222	---	13107662222	POTS	1/0/1	---
10	---	---	1310767....	VOFR	---	S0/0 150
Router No. 2
11	---	---	1310766....	VOFR	---	S0/0 150
3	73333	7	1310767....	POTS	1/0/0	---
4	74444	7	1310767....	POTS	1/0/0	---
5	75555	7	1310767....	POTS	1/1	---

The dial plan shown in Table 23 lists a simple dial-peer configuration table. No configuration for forwarding digits to a PBX is shown. Additional configuration for forwarding digits is required.

Configuring Dial Peers

Dial peers describe the entities to and from which a call is established. Dial-peer configuration tasks define the address or set of addresses serviced by that dial peer and the call parameters required to establish a call to and from that dial peer.

Two different kinds of dial peers pertain to VoFR configurations:

POTS---Dial peer connected via a traditional telephony network. Voice-telephony peers point to a particular voice port on a voice-network device.
VoFR---Dial peer connected via a Frame Relay WAN backbone. VoFR dial peers point to specific voice-network devices. For VoFR, you can configure VoFR dial peers for the following call types:
- Switched calls
- Cisco-trunk permanent calls
- FRF.11 trunk calls

POTS peers associate a telephone number with a particular voice port so that incoming calls for that port can be received. VoFR peers point to specific voice-network devices (by associating destination telephone numbers with a specific Frame Relay DLCI) so that outgoing calls can be placed. Both POTS and VoFR dial peers are needed to establish VoFR connections if you want both to send and receive calls.

For tandem voice nodes, POTS dial peers are not configured.

Establishing two-way communication using VoFR requires establishing a specific voice connection between two defined endpoints. As shown in Figure 66, for outgoing calls (from the perspective of the voice-telephony dial peer 1), the voice-telephony dial peer establishes the source (the originating telephone number and voice port) of the call. The voice-network dial peer establishes the destination by associating the destination telephone number with a specific Frame Relay DLCI.

Figure66: Calls from the Perspective of Router No.1

In the example, the destination pattern 14085554000 string maps to a U.S. telephone number 555-4000, with the digit 1 plus the area code (408) preceding the number. When you configure the destination pattern, set the dial string to match the local dial conventions.

To complete the two-way communications loop, you need to configure VoFR dial peer 4 as shown in Figure 67.

Figure67: Calls from the Perspective of Router No. 2

The only exception to this example is when both POTS dial peers are connected to the same router, as shown in Figure 68. In this circumstance, you would not need to configure a VoFR dial peer. Figure 68 shows an example for switched calls only.

Figure68: Communication Between Dial Peers Sharing the Same Router

When you configure dial peers, ensure that you understand the relationship between the destination pattern and the session target. The destination pattern is the telephone number of the voice device attached to the voice port. The session target represents the route to a serial port on the peer router at the other end of the Frame Relay connection. Figure 69 and Figure 70 show the relationship between the destination pattern and the session target, as seen from the perspective of both routers in a VoFR configuration. These examples show a topology for switched calls only.

Figure69: Relationship Between Destination Pattern and Session Target from the Perspective of Router No. 1

Figure70: Relationship Between Destination Pattern and Session Target from the Perspective of Router No. 2

The following sections describe how to configure POTS peers and VoFR peers.

Configuring POTS Dial Peers

To configure a POTS dial peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its telephone number, and associate it with a voice port through which calls will be established. Under most circumstances, the default values for the remaining dial-peer configuration commands are sufficient to establish connections.

Depending on your dial plan configuration, you might need to consider how to configure voice networks with variable-length dial plans, number expansion, excess digit playout, forward digits, and default voice routes, or use hunt groups with dial-peer preferences.

NoteThe Cisco 7200 series routers only perform VoFR tandeming in Cisco IOS Release 12.1. As a result, you cannot configure POTS dial peers on the 7200 series routers.

To configure POTS dial peers, use the following commands beginning in global configuration mode:

Command Purpose

Step1

Router(config)# dial-peer voice number pots
Define a POTS peer and enters dial-peer configuration mode. All subsequent commands that you enter in dial-peer voice mode before you exit will apply to this dial peer.

The number value tag identifies the dial peer and must be unique on the router. Do not duplicate a specific tag number.

Step2

Router(config-dial-peer)# destination-pattern [+]string[T]
Configures the dial peer destination pattern.

The string is a series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are the digits 0-9 and the letters A-D. The following special characters can be entered in the string:

The plus symbol (+) can be used to indicate an E.164 standard number. On the Cisco MC3810, the plus symbol (+) is not a valid character in the string.

The star character (*) and the pound sign (#) that appear on standard touch-tone dial pads can be used in any dial string. However, these characters cannot be used as leading characters in a string (for example, *650).

The period (.) can be used as a trailing character, and is used as a wildcard character. Multiple periods as trailing characters indicate multiple wildcard digits, such as for the 789... wildcard.

The comma (,) can be used only in prefixes, and is used to insert a 1-second pause or a delay.

The timer (T) character can be used to configure variable-length dial plans.

Step3

Router(config-dial-peer)# port slot/subunit/port
or

(Cisco 2600 series and 3600 series) Associates this voice-telephony dial peer with a specific voice port.

Router(config-dial-peer)# port slot/port
(Cisco MC3810) Associates this voice-telephony dial peer with a specific voice port.

	Command	Purpose
Step1	Router(config)# dial-peer voice number pots	Define a POTS peer and enters dial-peer configuration mode. All subsequent commands that you enter in dial-peer voice mode before you exit will apply to this dial peer. The number value tag identifies the dial peer and must be unique on the router. Do not duplicate a specific tag number.
Step2	Router(config-dial-peer)# destination-pattern [+]string[T]	Configures the dial peer destination pattern. The string is a series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are the digits 0-9 and the letters A-D. The following special characters can be entered in the string: The plus symbol (+) can be used to indicate an E.164 standard number. On the Cisco MC3810, the plus symbol (+) is not a valid character in the string. The star character () and the pound sign (#) that appear on standard touch-tone dial pads can be used in any dial string. However, these characters cannot be used as leading characters in a string (for example, 650). The period (.) can be used as a trailing character, and is used as a wildcard character. Multiple periods as trailing characters indicate multiple wildcard digits, such as for the 789... wildcard. The comma (,) can be used only in prefixes, and is used to insert a 1-second pause or a delay. The timer (T) character can be used to configure variable-length dial plans.
Step3	Router(config-dial-peer)# port slot/subunit/port or	(Cisco 2600 series and 3600 series) Associates this voice-telephony dial peer with a specific voice port.
	Router(config-dial-peer)# port slot/port	(Cisco MC3810) Associates this voice-telephony dial peer with a specific voice port.

Configuring Dial Plan Options for the POTS Dial Peer

When you configure the dial plan, you have different options for how the dial plan is designed.

To configure dial plan options, use the following commands in dial-peer configuration mode:

Command Purpose

Step1

Router(config-dial-peer)# preference value
(Optional) Configures a preference for the POTS dial peer. The value is a number from 0-10 where the lower the number, the higher the preference. If POTS and voice-network (VoFR) peers are mixed in the same hunt group, POTS dial peers will be searched first, even if a voice-network peer has a higher preference number.

Step2

Router(config-dial-peer)# prefix string
(Optional) Assign the prefix of the dialed digits for the dial peer.

Valid numbers for the string argument are 0 through 9, and a comma (,). Use a comma to include a pause in the prefix

When an outgoing call is initiated to this dial peer, the prefix string value is sent to the telephony interface first, before the telephone number associated with the dial peer.

Step3

Router(config-dial-peer)# direct-inward-dial
(Optional for the Cisco 2600/3600 series only) Enables the Direct Inward Dial (DID) call treatment for the incoming called number.

Step4

Router(config-dial-peer)# incoming called-number string
(Optional for the Cisco 2600/3600 series only) Identifies the service type for a call on a router handling both voice and modem calls.

Step5

Router(config-dial-peer)# max-conn number
(Optional for the Cisco 2600/3600 series only) Specifies the maximum number of allowed connections to and from the POTS dial peer. The valid range is 1-2147483647.

Step6

Router(config-dial-peer)# forward-digits {num-digit | all | implicit}
(Optional for the Cisco MC3810 only) If you are using the forward-digits feature, configures the digit-forwarding method that will be used on the dial peer. The valid range for the number of digits forwarded (num-digit) is 0-32.

The default value is implicit, in which the exactly matched digits are not forwarded. Only digits matched by the wildcard pattern are forwarded.

To configure the next POTS dial peer, exit dial-peer configuration mode by entering exit, and repeat the previous steps. To configure VoFR dial peers, see the next section, "Configuring VoFR Dial Peers."

Configuring VoFR Dial Peers

To configure a VoFR dial peer, you need to uniquely identify the peer (by assigning it a unique tag number) and define the outgoing serial port number and the virtual circuit number.

VoFR dial peer configuration procedures are described in the following sections:

NoteOn the Cisco MC3810, you can also configure a voice class to assign idle state and out-of-service (OOS) signalling attributes to a VoFR dial peer. For more information, see the "Configuring Voice-Related Support Features" chapter.

Figure 71 shows an example of a Frame Relay network with switched calls. In the example, there are two routers (Routers No. 1 and No. 2) with telephone devices at the endpoints. In between the two endpoint routers are tandem routers (Routers A, B, and C) with switched connections in between. Standard VoFR dial peers are configured on the intermediate nodes in the Frame Relay network. Support for switched calls are configured on the dial peers on both Router No.1 and Router No2.

Figure71: Endpoint Nodes and Tandem Nodes in VoFR Network for Switched Calls

In this example, different types of VoFR dial peers must be configured on the different routers.

VoFR dial peers for switched connections must be configured between Router No. 1 and Router No.2 (the endpoints for the voice connection).

On the tandem routers (Routers A, B, and C) VoFR dial peers for switched calls must be configured for the following:

Between Router No. 1 and Router A
Between Router No. 1 and Router B
Between Router No. 2 and Router A
Between Router No. 2 and Router C
Between Router A and Router B
Between Router B and Router C

Configuring VoFR Dial Peers for Switched Calls

If you will be sending switched calls over the Frame Relay network, you must configure the VoFR dial peers to specifically support switched calls.

To configure a VoFR dial peer to support switched calls, use the following commands beginning in global configuration mode:

To configure another VoFR dial peer, exit dial-peer configuration mode and repeat Steps 1 through10.

To configure VoFR dial peers for tandem nodes, see the "Configuring VoFR Dial Peers for Tandem Nodes" later in this chapter.

For procedures on how to configure switched calls, see the "Configuring Switched Calls (User Dialed or Auto-Ringdown)" section later in this chapter. For information on configuring all call types, see the "Overview of VoFR Connection Types" later in this chapter.

Configuring VoFR Dial Peers for Cisco-Trunk (Private Line) Calls

If you will be sending Cisco-trunk (private line) calls over the Frame Relay network, you must configure the VoFR dial peers to specifically support Cisco-trunk (private line) calls. Cisco-trunk (private line) calls are permanent calls.

One key task when you configure Cisco-trunk (private line) connections is to configure the signal type for the dial peer. Depending on the router you are using, you might have several options. The signal-type dial-peer command supports the following options:

cas---Use the cas option to support North American CAS/robbed-bit signalling. This is the default signalling type.
cept---On the Cisco MC3810, use the cept option to provide a basic E1 ABCD protocol, primarily for CEPT E&M signalling. This option is primarily used for European voice networks. If this option is used with FXS or FXO voice ports, the signalling used is equivalent to MEL CAS. This option is not supported on the Cisco 2600/3600 series.
ext-signal---Use the ext-signal option in cases where some external signalling channel is being used (for example, common channel signalling), or where no signalling information is being sent at all over a permanent "dumb" voice pipe. Applications where no signalling is required include using a simple voice pipe to carry audio for a public address system.
transparent---On the Cisco MC3810 with digital voice ports, use the transparent option when the ABCD signalling bits are copied through from the T1/E1 interface "transparently" without modification or interpretation (also known as transparent FRF.11 signalling). This option allows the Cisco MC3810 to handle or transport unknown signalling protocols. On the Cisco MC3810 with analog voice ports, the transparent option does not apply. If the transparent option is sent over analog voice ports, the signalling is equivalent to the cept option. This option is not supported on the Cisco 2600 series and 3600 series in Cisco IOS Release 12.1.

The signal type normally must be configured such that the signal type selected in the dial peers on the routers at both ends of the permanent voice call are the same. When you configure a permanent connection between a T1/E1 Cisco MC3810 and an analog voice port on a Cisco 2600 or Cisco3600, the signal type normally should be set to cas, which is the default.

Cisco 2600 and Cisco 3600 analog voice ports do not support the cept or transparent signal types. The T1/E1 Cisco MC3810 can also be set to transparent, which simply passes the signalling through from the Cisco 2600/3600 without interpretation. However, when transparent is used, the CiscoMC3810 makes no assumptions regarding the on-hook and off-hook state of the call. By default, when configured using transparent, the Cisco MC3810 operates the voice path in the permanently open state so that voice packets are sent (and network bandwidth consumed) regardless of the state of the call. For more information on controlling the flow of voice packets with transparent signalling, see the "Verifying Your Frame Relay Configuration" section earlier in this chapter.

To configure a VoFR dial peer to support Cisco-trunk permanent (private line) calls, use the following commands beginning in global configuration mode:

To configure another VoFR dial peer, exit dial-peer configuration mode and repeat steps 1 through11.

To configure VoFR dial peers for tandem nodes, see the "Configuring VoFR Dial Peers for Tandem Nodes" section later in this chapter.

For procedures on how to configure Cisco-trunk permanent (private line) calls, see the "Configuring Cisco-Trunk (Private Line) Permanent Calls" section later in this chapter. For information on configuring all call types, see the "Overview of VoFR Connection Types" section later in this chapter.

Configuring VoFR Dial Peers for FRF.11 Trunk Calls

If you will be sending FRF.11 trunk calls over the Frame Relay network, you must configure the VoFR dial peers to specifically support FRF.11 trunk calls. For FRF.11 trunk calls, you must statically configure the VoFR dial peers on both sides of the FRF.11 trunk.

NoteFRF.11 trunk calls cannot be used in conjunction with dial plans.

To configure a VoFR dial peer to support FRF.11 trunk calls, use the following commands beginning in global configuration mode:

Command Purpose

Step1

Router(config)# dial-peer voice number vofr
Defines a VoFR dial peer and enters dial-peer configuration mode. All subsequent commands that you enter in dial-peer voice mode before you exit will apply to this dial peer.

The number tag value identifies the dial peer and must be unique on the router. Do not duplicate a specific tag number.

Step2

Router(config-dial-peer)# destination-pattern string
Configures the dial peer destination pattern. The same restrictions for the string listed in the POTS dial-peer configuration also apply to the VoFR destination pattern.

Step3

Router(config-dial-peer)# session protocol frf11-trunk
Configures the session protocol to support FRF.11 trunk calls.

NoteYou cannot send FRF.11 trunk calls through tandem nodes.

Step4

Router(config-dial-peer)# session target interface dlci cid
Configures the Frame Relay session target for the dial peer.

NoteThe cid value is required for FRF.11 trunk calls.

Step5

Router(config-dial-peer)# codec type bytes payload_size
Specifies the voice coder rate of speech and payload size for the dial peer. The default dial peer codec is g729r8. Note that the Cisco MC3810 is limited to a maximum of 12 calls when using g729r8; to support up to 24 calls on the Cisco MC3810, use g729ar8.

For FRF.11 trunk calls, the codec values must be set the same on both sides of the connection.

Specifying the payload size by entering the bytes value is optional. Each codec type defaults to a different payload size if you do not specify a value. To obtain a list of the default payload sizes, enter the codec command and the bytes option followed by a question mark (?).

Step6

Router(config-dial-peer)# dtmf-relay
(Optional) If the codec type is a low bit-rate codec such as g729 or g723, specifies support for DTMF relay to improve end-to-end transport of DTMF tones. DTMF tones do not always propagate reliably with low bit-rate codecs.

DTMF relay is disabled by default.

Step7

Router(config-dial-peer)# called-number termination-string
(Required for the Cisco 2600/3600 series only) Configures the termination string for the dial peer for FRF.11 trunk calls. This command is required to enable the router to establish an incoming trunk connection.

This command only applies when the session protocol command is set to frf11-trunk.

NoteAlthough this command is visible on the CiscoMC3810, the command is ignored if entered.

Step8

Router(config-dial-peer)# signal type {cas | external}
(Optional on the Cisco 2600/3600 series only) Configures the signal type when creating an FXS-FXS trunk. The default on the Cisco 2600/3600 series is cas.

Step9

Router(config-dial-peer)# no vad
(Optional) Disables VAD on the dial peer. This command is enabled by default.

Step10

Router(config-dial-peer)# sequence-numbers
(Optional) Enables the voice sequence number if required for your configuration. This command is disabled by default.

Step11

Router(config-dial-peer)# preference value
(Optional) Configures a preference for the VoFR dial peer. The value is a number from 0 to 10 where the lower the number, the higher the preference in hunt groups.

Step12

Router(config-dial-peer)# fax rate {2400 | 4800 | 7200 | 9600 | 14400 | disable | voice}
(Optional) Configures the transmission speed (in bps) at which a fax will be sent to the dial peer.

The default is voice, which specifies the highest possible transmission speed allowed by the voice rate.

To configure another VoFR dial peer, exit dial-peer configuration mode and repeat Steps 1 through12.

Repeat this procedure on the destination router on the other side of the FRF.11 trunk.

To configure VoFR dial peers for tandem nodes, see the "Configuring VoFR Dial Peers for Tandem Nodes" section later in this chapter.

For procedures on how to configure FRF.11 trunk calls, see the "Configuring FRF.11 Trunk (Private Line) Calls" section later in this chapter. For information on configuring all call types, see the 'Overview of VoFR Connection Types" section later in this chapter.

Configuring VoFR Dial Peers for Tandem Nodes

You configure standard VoFR dial peers for switched calls on the tandem routers in the network topology. Tandeming is not allowed when the call type is an FRF.11 trunk call.

You can configure VoFR dial peers for tandem routers on the Cisco MC3810 and on the Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series routers.

NoteThe Cisco 7200 series routers can only serve as tandem nodes in the VoFR network Cisco IOS Release 12.1. This is the only dial peer procedure supported on the Cisco 7200 series.

To configure VoFR dial peers on tandem routers, use the following commands beginning in global configuration mode:

Command Purpose

Step1

Router(config)# dial-peer voice number vofr
Defines a VoFR dial peer and enters dial-peer configuration mode. All subsequent commands that you enter in dial-peer voice mode before you exit will apply to this dial peer.

The number value tag identifies the dial peer and must be unique on the router. Do not duplicate a specific number tag.

Step2

Router(config-dial-peer)# destination-pattern [+]string[T]
Configures the dial peer destination pattern. The same restrictions for the string listed in the POTS dial-peer configuration also apply to the VoFR destination pattern.

Step3

Router(config-dial-peer)# session target interface dlci
Configures the Frame Relay session target for the dial peer.

Step4

Router(config-dial-peer)# preference value
(Optional) Configures a preference for the VoFR dial peer. The value argument is a number from 0 to 10 where the lower the number, the higher the preference in hunt groups.

To configure the next VoFR dial peer, exit dial-peer configuration mode by entering exit, and repeat the previous steps. On tandem nodes, at least two VoFR dial peers are required, one for each call leg in the router.

Disabling Dial-Peer Hunting on a Dial Peer (Cisco MC3810 Only)

To disable dial-peer hunting on a dial peer, enter the following commands beginning in global configuration mode:

Command Purpose

Step1

Router(config)# dial-peer voice tag {pots | vofr}
Enters dial-peer configuration mode for the specified dial peer.

Step2

Router(config-dial-peer)# huntstop
Disables dial-peer hunting on the dial peer. Once you enter this command, no further hunting will be allowed if a call fails on the specified dial peer.

To reenable dial-peer hunting on a dial peer, enter the following commands beginning in global configuration mode:

Command Purpose

Step1

Router(config)# dial-peer voice tag {pots | vofr}
Enters dial-peer configuration mode for the specified dial peer.

Step2

Router(config-dial-peer)# no huntstop
Reenables dial-peer hunting on the dial peer. Once you enter this command, no further hunting will be allowed if a call fails on the specified dial peer.

Configuring VoFR Connections

After you have configured the Frame Relay DLCI settings and you have configured your dial plan, you are ready to configure specific VoFR connections.

There are many different scenarios for VoFR connections. For information on the different connection types, see the next section, "Overview of VoFR Connection Types."

Procedures on how to configure the different connection types are described in the following sections:

In addition, special consideration is required for configuring calls for tandem nodes. For more information, see the "Configuring Connections for Tandem Nodes" section later in this chapter.

NoteUse of Cisco trunks for permanent calls (private-line) is recommended over FRF.11 trunk calls unless FRF.11 compliant standards-based interworking is required with non-Cisco devices. The Cisco trunk protocol is a superset of the FRF.11 protocol and contains Cisco proprietary extensions designed to support switched call routing and other advanced features.

Overview of VoFR Connection Types

When you configure VoFR connections, many different connection types are possible depending on the hardware platform, whether the call is to be a regular switched (user dialed or auto-ringdown) call, or whether the call is a permanent call (Cisco-trunk or FRF.11-trunk). You configure these specific connection types using combinations of several commands.

Table 24 lists the different connection types for VoFR connections supported on the Cisco 2600 and 3600 series, and the combinations of commands to enter for each call type.

Table 25 lists the different connection types for VoFR connections supported on the CiscoMC3810 series, and the combinations of commands to enter for each call type.

Table24: VoFR Connection Types Supported on the Cisco 2600/3600 Series and 7200Series

Type of Call Frame Relay DLCI interface Command to Enter Data Fragmentation Supported by vofr Command Session Protocol Command to Enter in Dial Peer Mode Voice Port Connection Command to Enter

Switched call
(user dialed or auto-ringdown) to other Cisco 2600/3600 or 7200 routers

vofr [data cid]
[call-control [cid]]¹

FRF.11 Annex C

session protocol cisco-switched²

For user dialed calls: none

For auto-ringdown calls:
connection plar destination-string

Switched call
(user dialed or auto-ringdown)
to a CiscoMC3810

vofr cisco³

Cisco proprietary⁴

session protocol cisco-switched

For user dialed calls: none

For auto-ringdown calls:
connection plar destination-string

Cisco-trunk
permanent call (private-line) to other Cisco 2600/3600 or 7200 routers

vofr data cid
call-control cid

FRF.11 Annex C

session protocol cisco-switched

connection trunk destination-string
[answer mode]

Cisco-trunk
permanent call
(private-line)
to a CiscoMC3810

vofr cisco

Cisco proprietary

session protocol cisco-switched

connection trunk destination-string
[answer mode]

FRF.11 trunk call (private-line)

vofr [data cid] [call-control cid]⁵

FRF.11 Annex C

session protocol frf11-trunk

connection trunk destination-string
[answer mode]

¹The recommended use of this command is vofr data 4 call-control 5.
²The session protocol cisco-switched option is the default setting. If you do not enter this command, the setting will still apply.
³This command consumes data CID 4 and call-control CID 5.
⁴Cisco proprietary fragmentation is based on an early draft of FRF.12, and is compatible with Cisco MC3810 concentrators running software versions prior to Cisco IOS Release 12.0(3)XG or 12.0(4)T.
⁵For FRF.111 trunk calls, the call-control option is not required. It is only required if you mix FRF.11 trunk calls with other types of voice calls on the same PVC.

Table25: VoFR Call Types Supported on the Cisco MC3810

Type of Call Frame Relay DLCI interface Command to Enter Data Fragmentation Supported by vofr Command Session Protocol Command to Enter in Dial Peer mode Voice Port Connection Command to Enter

Switched call
(user dialed or auto-ringdown)

vofr cisco¹

or

frame-relay interface-dlci dlci voice-encap size [voice-cir cir]²

Cisco proprietary³

session protocol cisco-switched

For user dialed calls: none

For auto-ringdown calls:
connection plar destination-string

For tie-line connections:
connection tie-line
destination-string

Cisco-trunk
permanent call (private line)

vofr cisco

or

frame-relay interface-dlci dlci voice-encap size [voice-cir cir]

Cisco proprietary

session protocol cisco-switched

connection trunk destination-string
[answer mode]

FRF.11 trunk call (private-line)

vofr [data cid]⁴

FRF.11 Annex C

session protocol frf11-trunk

connection trunk destination-string
[answer mode]

¹This command consumes data CID 4 and call-control CID 5.
²The voice-encap value is required, but the voice-cir value is optional.
³Cisco proprietary fragmentation is based on an early draft of FRF.12, and is compatible with Cisco MC3810 concentrators running software versions prior to Cisco IOS Release 12.0(3)XG or 12.0(4)T.
⁴If the vofr command is entered without the cisco option, only FRF.11 trunks are supported. You cannot mix FRF.11 trunk calls with other call types on the same PVC on the Cisco MC3810.

Configuring Switched Calls (User Dialed or Auto-Ringdown)

This section describes how to configure switched calls (user dialed or auto-ringdown) on the different router platforms. This section is divided into the procedures described in the following sections:

Configuring Switched Calls on Cisco 2600, 3600, and 7200 Series Routers

You can configure switched calls on Cisco 2600, 3600, and 7200 series routers. To configure switched calls on these routers, use the following commands beginning in interface configuration mode:

This configuration uses standard FRF.11 Annex C fragmentation.

Configuring Switched Calls on Cisco 2600, 3600, or 7200 Series Routers to a Cisco MC3810

On the Cisco 2600, 3600, and 7200 series routers, you can configure switched calls to a CiscoMC3810. However, the configuration is different from standard switched calls because the earlier Cisco MC3810 versions used the Cisco proprietary version of FRF.12.

NoteThe Cisco 2600 and 3600 series routers cannot terminate or initiate calls with a Cisco MC3810 running software versions prior to Cisco IOS Releases 12.0(3)XG or 12.0(4)T.

To configure switched calls to a Cisco MC3810, use the following commands beginning in interface configuration mode:

Command Purpose

Step1

Router(config-if)# frame-relay interface-dlci dlci
Configures the Frame Relay DLCI and enters DLCI configuration mode.

Step2

Router(config-if)# vofr cisco
Configures the Frame Relay DLCI to support VoFR and the Cisco proprietary fragmentation implementation.

When this command is entered, data CID 4 and call-control CID5 are automatically assigned.

If you are configuring user-dialed calls, this procedure is completed. If configuring auto-ringdown calls, proceed to the next step.

Step3

For Cisco 2600 and 3600 series analog voice ports:

router(config)# voice-port slot/subunit/port
For Cisco 2600 and 3600 series digital voice ports:

router(config)# voice-port slot/port:ds0-group
Enters voice-port configuration mode.

Step4

Router(config-voiceport)# connection plar destination-string
(Optional) For auto-ringdown calls, configures the PLAR connection, specifying the telephone number in the destination-string argument.

This configuration uses Cisco proprietary data fragmentation.

Configuring Switched Calls on a Cisco MC3810

On the Cisco MC3810 you can configure switched calls to another Cisco MC3810 or to Cisco 2600, Cisco 3600, and Cisco 7200 series routers. However, the configuration is different from switched calls on other routers because the earlier Cisco MC3810 versions used the Cisco proprietary version of FRF.12.

NoteThe Cisco 2600 and Cisco 3600 series routers cannot terminate or initiate calls with a CiscoMC3810 running software versions prior to Cisco IOS Release 12.0(3)XG or 12.0(4)T.

When configuring switched calls on a Cisco MC3810, you must enter one of the following two commands:

vofr cisco

: This command was added in Cisco IOS Release 12.0(4)T, and uses WFQ at the PVC level. Using the vofr cisco command, you cannot disable support for WFQ.

frame-relay interface-dlci dlci voice-encap size [voice-cir cir]

: The voice-encap option for the frame-relay interface-dlci command was introduced in CiscoIOS Release 11.3(1)MA and does not support WFQ. However, you also have the option to configure fancy queueing with the voice-encap option by entering the no frag-pre-queuing command and configuring the appropriate fancy queueing commands.

The procedures for using each command differ. See the appropriate procedure for the command you plan to use. These configurations both use Cisco proprietary data fragmentation.

NoteThe vofr command and the frame-relay interface-dlci voice-encap command are mutually exclusive on the same interface, so you must choose which command to use.

To configure switched calls on a Cisco MC3810 using the vofr cisco command, use the following commands beginning in interface configuration mode:

Command Purpose

Step1

Router(config-if)# frame-relay interface-dlci dlci
Configures the Frame Relay DLCI and enters DLCI configuration mode.

NoteWhen configuring switched calls on the Cisco MC3810 using the vofr cisco command, do not enter the voice-encap or voice-cir options.

Step2

Router(config-if)# vofr cisco
Configures the Frame Relay DLCI to support VoFR and the Cisco proprietary fragmentation implementation.

When this command is entered, data CID 4 and call-control CID5 are automatically assigned.

If configuring user dialed calls, this procedure is completed. If configuring auto-ringdown calls or tie-line connections on the Cisco MC3810, proceed to the next step.

Step3

For Cisco MC3810 series analog voice ports:

router(config)# voice-port slot/port
For Cisco MC3810 series digital voice ports:

router(config)# voice-port slot:ds0-group
Enters voice-port configuration mode.

Step4

Router(config-voiceport)# connection plar destination-string
or

For auto-ringdown calls, configures the PLAR connection, specifying the telephone number in the destination-string argument.

Router(config-voiceport)# connection tie-line destination-string
For tie-line calls, configures the tie-line connection, specifying the telephone number in the destination-string argument.

To configure switched calls on a Cisco MC3810 using the frame-relay interface-dlci voice-encap command, use the following commands beginning in interface configuration mode:

Command Purpose

Step1

Router(config-if)# frame-relay interface-dlci dlci voice-encap size [voice-cir cir]
Configures the Frame Relay DLCI, enters DLCI configuration mode, and specify that the DLCI will support voice traffic.

For recommended fragmentation sizes to use with the voice-encap option, see Table 26.

In this configuration, the voice-encap value is required, but the voice-cir value is optional. The default voice-cir value is the CIR configured for the Frame Relay map class

NoteDo not configure the voice-cir option to be higher than the physical link speed. If Frame Relay traffic shaping is enabled for a PVC sharing voice and data, do not configure the voice-cir option to be higher than the value set with the frame-relay mincir command.

Step2

Router(config-if)# no frag pre-queuing
(Optional) Configures the interface to disable first-come, first-served (FCFS) queueing.

By default, the frag pre-queuing command is enabled. This command supports only FCFS queueing, and performs the queueing before the data fragmentation takes place.

If you want to configure fancy queueing (WFQ, priority queueing, or custom queueing) on the interface, enter the no frag pre-queuing command. Note that after you enter the no frag pre-queuing command, you still must configure the commands to support the fancy queueing method desired.

NoteThe frag pre-queuing command is only supported on the interface if the voice-encap option for the frame-relay interface-dlci command is configured. It is not supported if the vofr cisco command is configured.

Step3

For Cisco MC3810 series analog voice ports:

router(config)# voice-port slot/port
For Cisco MC3810 series digital voice ports:

router(config)# voice-port slot:ds0-group
Enters voice-port configuration mode.

Step4

Router(config-voiceport)# connection plar destination-string
or

For auto-ringdown calls, configures the PLAR connection, specifying the telephone number in the destination-string argument.

Router(config-voiceport)# connection tie-line destination-string
For tie-line calls, configures the tie-line connection, specifying the telephone number in the destination-string argument.

Table 26 lists recommended data fragmentation sizes to use when configuring the voice-encap option for the frame-relay interface-dlci command.

Table26: Recommended Data Fragmentation Sizes

Access Rate Recommended Data Fragmentation Size¹

64 kbps

80 bytes

128 kbps

160 bytes

256 kbps

320 bytes

512 kbps

640 bytes

1536 kbps (full T1)

1600 bytes

2048 kbps (full E1)

1600 bytes

¹The data fragmentation size is based on back-to-back Frame Relay. If you are sending traffic through an IGX node with standard Frame Relay, subtract 6 bytes from the recommended data fragmentation size.

NoteWhen you configure the data fragmentation size, use the slower access rate of either the local or remote device to calculate which data fragmentation size to use. If you configure a data fragmentation size too high for either the local or remote device, the access rate will become throttled because the slower device cannot handle the larger data fragmentation size. For example, if the access rate at the local device is 512kbps and the access rate of the remote device is 256 kbps, configure the data segmentation size based on the slower 256-kbps access rate.

Configuring Cisco-Trunk (Private Line) Permanent Calls

This section describes how to configure Cisco-trunk (private line) calls on the different router platforms. This section is divided into the procedures described in the following sections:

Configuring Cisco Trunk Permanent Calls on Cisco 2600, 3600, and 7200 Series Routers

You can configure Cisco trunk permanent calls on Cisco 2600, 3600, and 7200 series routers. To configure Cisco trunk permanent calls on these routers, use the following commands beginning in interface configuration mode:

Command Purpose

Step1

Router(config-if)# frame-relay interface-dlci dlci
Configures the Frame Relay DLCI and enters DLCI configuration mode.

Step2

Router(config-if)# vofr [data cid] [call-control [cid]]
Configures the Frame Relay DLCI to support VoFR.

NoteWhen the vofr command is used, all subchannels on the DLCI are configured for FRF.11 encapsulation. If the vofr command is entered without any keywords or arguments, the data subchannel will be CID 4 and there will be no call-control subchannel.

If configuring tandem calls, this step ends your configuration.

Step3

For Cisco 2600 and 3600 series analog voice ports:

router(config)# voice-port slot/subunit/port
For Cisco 2600 and 3600 series digital voice ports:

router(config)# voice-port slot/port:ds0-group
Enters voice-port configuration mode.

Step4

Router(config-voiceport)# connection trunk destination-string [answer-mode]
For private line calls, configures the trunk connection, specifying the telephone number in the destination-string argument.

When you configure Cisco trunk permanent calls, one side must be the call initiator (master) and the other side is normally the call answerer (slave). By default, the voice operates in master mode. Enter the answer-mode keyword to specify that the voice port should operate in slave mode.

Step5

Router(config-voiceport)# shutdown
Shuts down the voice port.

Step6

Router(config-voiceport)# no shutdown
Reactivates the voice port to enable the trunk connection to take effect.

This configuration uses standard FRF.11 Annex C fragmentation.

NoteEvery time you enter the connection trunk or no connection trunk command, you must toggle the voice port (by entering shutdown, then no shutdown) for the changes to take effect.

Configuring Cisco Trunk Permanent Calls on Cisco 2600 and 3600 Routers to a Cisco MC3810

To configure Cisco trunk permanent calls to a Cisco MC3810 from a 2600 series or 3600 series router, use the following commands beginning in interface configuration mode:

Command Purpose

Step1

Router(config-if)# frame-relay interface-dlci dlci
Configures the Frame Relay DLCI and enters DLCI configuration mode.

Step2

Router(config-if)# vofr cisco
Configures the Frame Relay DLCI to support VoFR and the Cisco proprietary data implementation.

When this command is entered, data CID 4 and call-control CID5 are automatically assigned.

Step3

For Cisco 2600 and 3600 series analog voice ports:

router(config)# voice-port slot/subunit/port
For Cisco 2600 and 3600 series digital voice ports:

router(config)# voice-port slot/port:ds0-group
Enters voice-port configuration mode.

Step4

Router(config-voiceport)# connection trunk destination-string [answer-mode]
For private line calls, configures the trunk connection, specifying the telephone number in the destination-string argument.

When you configure Cisco trunk permanent calls, one side must be the call initiator (master) and the other side is normally the call answerer (slave). By default, the voice operates in master mode. Enter the answer-mode keyword to specify that the voice port should operate in slave mode.

Step5

Router(config-voiceport)# shutdown
Shuts down the voice port.

Step6

Router(config-voiceport)# no shutdown
Reactivates the voice port to enable the trunk connection to take effect.

This configuration uses Cisco proprietary data fragmentation.

NoteEvery time you enter the connection trunk or no connection trunk command, you must toggle the voice port (by entering shutdown, then no shutdown) for the changes to take effect.

Configuring Cisco Trunk Permanent Calls on a Cisco MC3810

When configuring Cisco trunk permanent calls on a Cisco MC3810 interface, you must enter one of the following two commands:

vofr cisco

: This command was added in Cisco IOS Release 12.0(3)XG, and uses WFQ at the PVC level. Using the vofr cisco command, you cannot disable support for WFQ.

frame-relay interface-dlci dlci voice-encap size [voice-cir cir]

: The voice-encap option for the frame-relay interface-dlci command was introduced in CiscoIOS Release 11.3(1)MA and does not support WFQ. However, you also have the option to configure fancy queueing with the voice-encap option by entering the no frag-pre-queuing command and configuring the appropriate fancy queueing commands.

The procedures for using each command differ. See the appropriate procedure for the command you plan to use. These configurations both use Cisco proprietary data fragmentation.

NoteThe vofr command and the frame-relay interface-dlci voice-encap command are mutually exclusive on the same interface, so you must choose which command to use.

To configure Cisco trunk permanent calls on a Cisco MC3810 using the vofr cisco command, use the following commands from interface configuration mode:

Command Purpose

Step1

Router(config-if)# frame-relay interface-dlci dlci
Configures the Frame Relay DLCI and enters DLCI configuration mode.

NoteWhen configuring Cisco trunk calls on the CiscoMC3810 using the vofr cisco command, do not enter the voice-encap or voice-cir options.

Step2

Router(config-if)# vofr cisco
Configures the Frame Relay DLCI to support VoFR and the Cisco proprietary data implementation.

When this command is entered, data CID 4 and call-control CID5 are automatically assigned.

Step3

For Cisco MC3810 series analog voice ports:

router(config)# voice-port slot/port
For Cisco MC3810 series digital voice ports:

router(config)# voice-port slot:ds0-group
Enters voice-port configuration mode.

Step4

Router(config-voiceport)# connection trunk destination-string [answer-mode]
For private line calls, configures the trunk connection, specifying the telephone number in the destination-string.

When configuring Cisco trunk permanent calls, one side must be the call initiator (master) and the other side is normally the call answerer (slave). By default, the voice operates in master mode. Enter the answer-mode keyword to specify that the voice port should operate in slave mode.

Step5

Router(config-voiceport)# shutdown
Shuts down the voice port.

Step6

Router(config-voiceport)# no shutdown
Reactivates the voice port to enable the trunk connection to take effect.

NoteEvery time you enter the connection trunk or no connection trunk command, you must toggle the voice port (by entering shutdown, then no shutdown) for the changes to take effect.

To configure Cisco trunk permanent calls on a Cisco MC3810 using the frame-relay interface-dlci voice-encap command, use the following commands beginning in interface configuration mode:

Command Purpose

Step1

Router(config-if)# frame-relay interface-dlci dlci voice-encap size [voice-cir cir]
Configures the Frame Relay DLCI, enters DLCI configuration mode, and specify that the DLCI will support voice traffic.

For recommended fragmentation sizes to use with the voice-encap option, see Table 26.

In this configuration, the voice-encap value is required, but the voice-cir value is optional. The default voice-cir value is the CIR configured for the Frame Relay map class

NoteDo not configure the voice-cir option to be higher than the physical link speed. If Frame Relay traffic shaping is enabled for a PVC sharing voice and data, do not configure the voice-cir option to be higher than the value set with the frame-relay mincir command.

Step2

Router(config-if)# no frag pre-queuing
(Optional) Enters this command if you want to disable FCFS queueing on the interface.

By default, the frag pre-queuing command is enabled. This command supports only FCFS queueing, and performs the queueing before the data fragmentation takes place.

If you want to configure fancy queueing (WFQ, priority queueing, or custom queueing) on the interface, enter the no frag pre-queuing command. Note that after you enter the no frag pre-queuing command, you still must configure the commands to support the fancy queueing method desired.

NoteThe frag pre-queuing command is only supported on the interface if the voice-encap option for the frame-relay interface-dlci command is configured. It is not supported if the vofr cisco command is configured.

Step3

For Cisco MC3810 series analog voice ports:

router(config)# voice-port slot/port
For Cisco MC3810 series digital voice ports:

router(config)# voice-port slot:ds0-group
Enters voice-port configuration mode.

Step4

Router(config-voiceport)# connection trunk destination-string [answer-mode]
For private line calls, configures the trunk connection, specifying the telephone number in the destination-string argument.

When you configure Cisco trunk permanent calls, one side must be the call initiator (master) and the other side is normally the call answerer (slave). By default, the voice port is the master. Enter the answer-mode keyword to specify that the voice port will be the slave.

Step5

Router(config-voiceport)# shutdown
Shuts down the voice port.

Step6

Router(config-voiceport)# no shutdown
Reactivates the voice port to enable the trunk connection to take effect.

NoteEvery time you enter the connection trunk or no connection trunk command, you must toggle the voice port (by entering shutdown, then no shutdown) for the changes to take effect.

Configuring FRF.11 Trunk (Private Line) Calls

On the Cisco MC3810 and on Cisco 2600 and 3600 series routers, you can configure FRF.11 trunk calls to a second router.

You cannot configure FRF.11 trunk calls for tandem VoFR configurations.

NoteThis configuration requires that the session protocol dial-peer configuration command be set to frf11-trunk. See the "Configuring VoFR Dial Peers" section.

To configure FRF.11 trunk (private line) calls, use the following commands beginning in interface configuration mode:

Command Purpose

Step1

Router(config-if)# frame-relay interface-dlci dlci
Configures the Frame Relay DLCI and enters DLCI configuration mode.

Step2

Router(config-if)# vofr [data cid] [call-control cid]
or

(Cisco 2600 and Cisco 3600) Configures the Frame Relay DLCI to support VoFR and to optionally enter the data and call-control CIDs.

Router(config-if)# vofr [data cid]
(Cisco MC3810) Configures the Frame Relay DLCI to support VoFR and to optionally enter the data CID. The call-control option is not supported on the CiscoMC3810.

NoteWhen the vofr command is used, all subchannels on the DLCI are configured for FRF.11 encapsulation except the data subchannel. If the vofr command is entered without any keywords or arguments, the data subchannel will be CID 4 and there will be no call-control subchannel.

Step3

For Cisco 2600 and 3600 series analog voice ports:

router(config)# voice-port slot/subunit/port
For Cisco 2600 and 3600 series digital voice ports:

router(config)# voice-port slot/port:ds0-group
For Cisco MC3810 series analog voice ports:

router(config)# voice-port slot/port
For Cisco MC3810 series digital voice ports:

router(config)# voice-port slot:ds0-group
Enters voice-port configuration mode.

Step4

Router(config-voiceport)# connection trunk destination-string [answer-mode]
For private line calls, configures the trunk connection, specifying the telephone number in the destination-string argument.

When you configure FRF.11 trunk calls, one side must be the call initiator (master) and the other side is normally the call answerer (slave). By default, the voice port is the master. Enter the answer-mode keyword to specify that the voice port will be the slave.

Step5

Router(config-voiceport)# shutdown
Shuts down the voice port.

Step6

Router(config-voiceport)# no shutdown
Reactivates the voice port to enable the FRF.11 trunk connection to take effect.

This configuration uses FRF.11 Annex C data fragmentation.

NoteEvery time you enter the connection trunk or no connection trunk command, you must toggle the voice port (by entering shutdown, then no shutdown) for the changes to take effect.

Configuring Connections for Tandem Nodes

Tandeming is the switching of an incoming VoFR call on a Frame Relay DLCI to an outgoing VoFR enabled DLCI. Tandeming works for switched calls and Cisco-trunk permanent calls only. You cannot tandem FRF.11 trunk calls over a multihop network.

Tandeming is supported on the Cisco MC3810 and on Cisco 2600, 3600, and 7200 series routers. The Cisco 7200 series can only act as a tandem router in a VoFR network.

NoteWhen creating voice networks with a mixture of router types, the Cisco MC3810 must be running Cisco IOS Release 12.0(3)XG, 12.0(4)T, or later to act as a tandem node.

Depending on which router is used as the end node and which router is used as the tandem node, you must use the correct Frame Relay PVC type when configuring your connections. Table 27 shows the different combinations of routers that can serve as end nodes and tandem nodes, and the Frame Relay PVC type required.

Table27: VoFR End Node and Tandem Node Combinations Supported

End Node(s) Tandem Node vofr Command to Enter for the Frame Relay DLCI

Cisco 2600/3600

Cisco 2600, Cisco 3600, or Cisco7200

vofr call-control

Cisco 2600/3600 and CiscoMC3810

Cisco MC3810

vofr cisco

Cisco MC3810

Cisco 2600, Cisco 3600, or Cisco7200

vofr cisco

When you configure a tandem node, you must configure two VoFR dial peers, one for each tandem connection.

Verifying Your Voice Connections

Verify that the voice connection for switched calls is working by performing the following tasks:

Pick up the handset on a telephone connected to the configuration and verify that you can get a dial tone.
Make a call from the local telephone to a configured dial peer and verify that the call attempt is successful.

Verify that the voice connection for FXO-FXS trunk calls from a telephone to a remote PBX is working by performing the following tasks:

Pick up the telephone and listen to hear the dial tone from the remote PBX.
Dial digits so that the remote PBX routes the call.

Verify your dial-peer and voice-port configuration by performing the following tasks:

If you have relatively few dial peers configured, you can use the show dial-peer voice command to verify that the data configured is correct. On the CiscoMC3810, use the show dial-peer voice summary command.
To show the status of the voice ports, use the show voice port command.
To show the call status for all voice ports, use the show call active voice [brief]. On the CiscoMC3810, use the show voice call command.
To show the current status of all DSP voice channels on the Cisco MC3810, use the show voice dsp command.

Verify your VoFR configuration on the DLCI by performing the following tasks:

To show the VoFR configuration, use the show frame-relay vofr [interface [dlci [cid]]] command. This command is not supported on the Cisco MC3810 when the vofr cisco or the frame-relay interface-dlci dlci voice-encap command is configured.
To show the status of Cisco-trunk permanent calls (private-line) on the Cisco MC3810, use the show voice permanent command.

Troubleshooting Tips

If you are having trouble connecting a call, you can try to resolve the problem by performing the following tasks:

If no calls are going through, make sure the frame-relay voice bandwidth command is configured.
If you have VoFR configured on a PVC and are experiencing problems with data connectivity on that PVC, make sure the frame-relay fragment command has been configured.
If data is not being sent but fragmentation is configured, make sure that FRTS is turned on.
If you suspect the problem is with the dial plan or the dial peers, use the show dial-plan number dial string command to display which dial peers are used when a specific number is called.
If you have problems connecting an FRF.11 trunk call, make sure the session protocol dial-peer command is set to frf11-trunk.
If configuring FRF.11 trunk calls on the Cisco 2600/3600, verify that the called-number vofr dial-peer command is configured, and that its number matches the destination pattern of the corresponding POTS dial peer.
Be sure the voice port, serial port, or the T1/E1 controller are set to no shutdown.
Be sure to toggle the voice port (by first entering shutdown, then no shutdown), every time you enter the connection trunk or no connection trunk commands.

Monitoring and Maintaining Your VoFR Configuration

To monitor and maintain your VoFR configuration, use the following commands in EXEC mode:

:

Command Purpose

Router#show call active voice [brief]
(Cisco 2600/3600 series only) Displays the active call table.

Router#show call history voice [last number] | [brief]
(Cisco 2600/3600 series only) Displays the call history table.

Router#show call history voice record
(Cisco MC3810 only) Displays the call history table.

Router#show dial-peer voice
Displays configuration information and call statistics for dial peers.

Router#show frame-relay fragment
Displays information about the Frame Relay fragmentation taking place in your Cisco router.

NoteOn the Cisco MC3810, if the frame-relay interface-dlci voice-encap command is configured, information about that PVC is not displayed.

Router#show frame-relay pvc
Displays statistics about PVCs for Frame Relay interfaces.

Router#show frame-relay vofr
Displays information about the FRF.11 subchannels being used on VoFR DLCIs.

NoteOn the Cisco MC3810, if the frame-relay interface-dlci voice-encap command is configured, information about that PVC is not displayed.

Router#show interfaces serial
Displays information about a serial interface.

Router#show traffic-shape queue
Displays information about the elements queued at a particular time at the VC (DLCI) level.

Router#show voice call
(Cisco MC3810 only) Displays the call status for all voice ports.

Router#show voice permanent-call
Displays information about the permanent calls on a voice interface.

Dial Planning and Dial-Peer Digit Manipulation Options

Depending on the configuration, you may need to consider different strategies for how you configure the dial plan and dial-peer digit manipulation, and how you configure the dial peers. Dial peer digit-manipulation tasks that you may need to consider are described in the following sections:

NoteThese concepts also apply to VoATM.

Excess Digits Playout

Excess digits are defined as received digits that are beyond the length of the destination pattern on a terminating router. A terminating router will forward excess digits to the telephony interface. For example, if the digits "123456789" are matched on a terminating router with a destination pattern of "1.....," the "6789" are excess digits and will be forwarded.

A router that is originating a call will only collect digits up to the length of a defined destination pattern. When a number is dialed that is longer than the destination pattern, after the last digit in the destination pattern is dialed, the call is immediately placed, and the additional digits are not collected by the router.

For example, if the digits "123456789" are dialed on an originating router with a destination pattern of 1...., then the digits "6789" are not collected. The call is placed immediately after the digit "5" is dialed. The additional digits "6789" are not collected, but are passed through the audio path.

In configurations where a phone on an FXS port on the router is connected to the PSTN through an FXO port on the same router, additional steps are necessary to enable a call to go through. Because the uncollected excess digits are passed through the audio path, the excess digits might not go through if the audio path is not ready when the excess digit is dialed. When dialing a destination pattern in this situation, pause and wait for a second dial tone before dialing the excess digits. This will give the audio path time to accept the excess digits.

Forward Digits and Voice Default Routes

Voice default routing for fixed-length dial plans can be used with larger PBX configurations to forward digits to router telephony interfaces.

Figure 72 shows an example of routing voice calls through a PBX using forward digits. In the configuration, the Cisco MC3810-t and Cisco MC3810-T are tandem nodes, and are required to support forwarding digits so that calls from Cisco MC3810s 0, 5, or 9 can make a call to extension "8208."

Figure72: Routing Voice Calls Among Cisco MC3810s Through a PBX

On the two tandem nodes, the forward-digits command is required. This command specifies to forward all digits matched with the destination "8..." to the appropriate port. For a call from Cisco MC3810-0 to reach extension 8208, the call must first terminate at Cisco MC3810-T, which plays out the digits "8208" to the voice port connected to the PBX. The PBX then routes the voice call to Cisco MC3810-t. On the tandem nodes, although the forward-digits all command is used, the forward-digits 4 command can also be used in this example.

The following are the dial-peer configurations on each Cisco MC3810 required for this configuration:

Cisco MC3810-t Configuration

dial-peer voice 8 pots
  destination-pattern 8208
  session-target s0 1
dial-peer voice 1000 pots
  destination-pattern 8...
  forward-digits all
  port 1/1
dial-peer voice 9999 pots
  destination-pattern ....
  forward-digits all
  port 1/1

Cisco MC3810-T Configuration

dial-peer voice 1 vofr
  destination-pattern 8200
  session-target s0 1
dial-peer voice 6 vofr
  destination-pattern 8209
  session-target s0 6
dial-peer voice 10 vofr
  destination-pattern 8209
  session-target s0 10
dial-peer voice 1 pots
  destination-pattern 8...
  forward-digits all
  port 1/1

Cisco MC3810-0 Configuration

dial-peer voice 1 pots
  destination-pattern 8200
  port 1/1
dial-peer voice 1000 vofr
  destination-pattern8...
  session-target s0 1
For Cisco MC3810-5:
dial-peer voice 5 pots
  destination-pattern 8205
  port 1/1
dial-peer voice 1000 vofr
  destination-pattern 8...
  session-target s0-1

Cisco MC3810-9 Configuration

dial-peer voice 9 pots
  destination-pattern 8209
  port 1/1
dial-peer voice 1000 vofr
  destination-pattern 8...
  session-target s0 1

The concept of voice default routes is also shown in Figure 72 and in the configuration. In the example, the configurations for the destination dial plans "8..." in both the tandem nodes are voice default routes because all voice calls dialed that start with 8 followed by three digits will either match on 8208 or end up with 8...., which is the last resort voice route used by Cisco MC3810-t if no other route is discovered.

Hunt Groups and Preference Configuration

When configuring the destination pattern for voice-telephony dial peers (such as to a PBX), if you configure each dial peer with a different destination pattern, then you may have limitations on the availability of calls getting through. Because you configure each dial peer with a different destination pattern, the destination is a single DS0 time slot on the voice connection to the PBX. If that time slot gets busy, then a call attempt to the dial peer mapped to that specific DS0 time slot will fail.

For example, in the following dial-peer configuration, four POTS dial peers are configured, all with a different destination pattern on the same PBX:

dial-peer voice 1 pots
destination-pattern 3001
port 1/1
 
dial-peer voice 2 pots
destination-pattern 3002
port 1/2
 
dial-peer voice 3 pots
destination-pattern 3003
port 1/3
 
dial-peer voice 4 pots
destination-pattern 3004
port 1/4

Because each dial peer has a different destination pattern configured, no backup is available if the DS0 time slot mapped to the dial peer is busy with another call.

However, the router supports the concept of hunt groups, in which you configure a group of dial peers on the same PBX with the same destination pattern. With a hunt group, if a call attempt is made to a dial peer on a specific DS0 time slot, if that time slot is busy, the router hunts for another time slot on that channel until an available time slot is found. In this case, each dial peer is configured using the same destination pattern of 3000, forming a dial pool to that destination pattern.

To provide specific dial peers in the pool with a preference over other dial peers, you can configure the preference order for each dial peer using the preference command. The following is an example of the dial-peer configuration with all dial peers having the same destination pattern, but different preference orders:

dial-peer voice 1 pots
destination pattern 3000
port 1/1
preference 0
 
dial-peer voice 2 pots
destination pattern 3000
port 1/2
preference 1
 
dial-peer voice 3 pots
destination pattern 3000
port 1/3
preference 2
 
dial-peer voice 4 pots
destination pattern 3000
port 1/4
preference 3

Note that when the preference order is set, the lower the preference number, the higher the priority. The highest priority is given to the dial peer with preference order 0.

You can also set the preference order on the network side, for voice network dial peers. However, you cannot mix the preference orders for POTS dial peers (local telephone devices) and voice-network peers (devices across the WAN backbone). You can set a separate preference order for each dial-peer type, but the preference order will not work on both at the same time. For example, you can configure preference order 0, 1, and 2 for voice telephony dial peers, and you can configure a separate preference order 0, 1, and 2 for the voice network dial peers, but the two preference orders are separate. The system only resolves the preference among dial peers of the same type; it does not resolve preferences between the two separate preference order lists.

The hunt group feature hunts for dial peers in the following order:

1. Dial peers are first hunted based on the priority set using the preference dial-peer command.

2. Dial peers are then hunted using the criteria of the longest number match. For example, if you have one dial peer set for 345.... and a second dial peer set for 3456789, the hunt group would first select the dial peer 3456789 because it has the longest match of the two dial peers.

3. Local POTS dial peers are hunted first before voice network peers.

4. Dial peers are then hunted in the order they were defined on the router.

NoteIf POTS and voice network peers are mixed in the same hunt group, the POTS dial peers must have priority over the voice-network peers.

Variable-Length Dial Plans

In most voice configurations, fixed-length dialing plans, in which all the dial-peer destination patterns have the same length, are sufficient because the telephone number strings are all the same length. However, in some voice network configuration, variable-length dial plans are required, especially if the network crosses national boundaries and the telephone number strings are of different lengths.

If you enter the "T" timer character in the destination pattern, the router can be configured to accept a fixed-length dial string, and then wait for additional dialed digits. For example, the following dial-peer configuration shows how the T character can be set to allow variable-length dial strings.

dial peer voice 1 pots
destination-pattern 2222T
port 1/1

In this example, the router accepts the digits 2222, and then waits for an unspecified number of dialed digits. If digits continue to be entered before the interdigit timeout expires, then the router will gather up to 31 additional digits. Once the interdigit timeout expires, the router places the call. The interdigit timeout is set by the timeouts inter-digit voice-port command.

The interdigit timeout timer can be immediately terminated when you enter the "#" character. If the # character is entered while the router is waiting for the additional digits, the # character is treated as an end-dial accelerator. The # character is not treated as an actual digit in the destination pattern and is not sent as part of the dialed string across the network.

However, if the # character is entered before the router is waiting for the additional digits (meaning before the T character is entered in the destination pattern), then the # character is treated as a dialed digit. For example, if a destination pattern is configured with the string 222...T, then the digits 2222####1234567 can be gathered, but the digits 2222###1234#67 cannot be gathered because the final # character is treated as a terminator.

The default value for the interdigit timeout is 1 second. If the duration is not changed, using the T timer adds 10 seconds to each call setup time because the call is not attempted until the timer expires (unless the # character is used as a terminator). Because of this dependency, if you use variable-length dial plans, the voice-port interdigit timeout should be reduced to reduce the call setup time.

VoFR Configuration Examples

This section provides specific configuration examples for different VoFR connections and call type scenarios. This section includes the following examples:

NoteIn the examples, some commands are shown with a lowercase letter in boldface. These letters indicate command settings that must match on the different routers. For example, the frame-relay cir s value indicates that the committed information rate "s" must match on the routers as shown.

The examples do not provide complete configurations, but show the required commands to configure VoFR.

Two Routers Using Frame Relay Fragmentation

Figure 73 shows an example of Frame Relay fragmentation between two routers.

Figure73: Two Routers Using Frame Relay Fragmentation

Router A (Cisco 3600) Router B (Cisco 3600)

hostname 3600A ! interface serial 0/0 ip address xxx.xxx.xxx 255.255.255.0 frame-relay traffic shaping ! frame-relay interface-dlci 100 class toto ! map-class frame-relay toto encapsulation frame-relay frame-relay cir s frame-relay bc u frame-relay fragment y
hostname 3600B ! interface serial 0/0 ip address xxx.xxx.xxx 255.255.255.0 frame-relay traffic shaping frame-relay class toto frame-relay interface-dlci 100 ! map-class frame-relay toto encapsulation frame-relay frame-relay cir s frame-relay bc u frame-relay fragment y

This configuration uses FRF.12 fragmentation.

Two Routers Using a VoFR PVC

Figure 74 shows an example of two routers with connections using a VoFR PVC.

Figure74: Two Routers Using a VoFR PVC

Router A (Cisco 3600) Router B (Cisco 3600)

hostname 3600A ! interface serial 0/0 frame-relay traffic shaping ! frame-relay interface-dlci 100 vofr data z class toto ! map-class frame-relay toto frame-relay voice-bandwidth t frame-relay min-cir x frame-relay cir s frame-relay bc u frame-relay fragment y
hostname 3600B ! interface serial 0/0 frame-relay traffic shaping frame-relay class toto ! frame-relay interface-dlci 100 vofr data z ! map-class frame-relay toto frame-relay voice-bandwidth t frame-relay min-cir x frame-relay cir s frame-relay bc u frame-relay fragment y

This configuration uses FRF.11 Annex C fragmentation.

Router Using a VoFR PVC to a Cisco MC3810

Figure 75 shows an example of a Cisco 3600 series router with connections to a Cisco MC3810 using a VoFR PVC. In this example, the VoFR interface on the Cisco MC3810 is configured using the vofr cisco command.

NoteWhen a Cisco MC3810 is on a VoFR network, the configuration for connections to and from the Cisco MC3810 is slightly different than for other routers that support VoFR.

Figure75: Router Using a VoFR PVC to a Cisco MC3810

Router A (Cisco 3600) Router B (Cisco 3600)

hostname 3600A ! interface serial 0/0 ip address xxx.xxx.xxx 255.255.255.0 frame-relay traffic shaping ! frame-relay interface-dlci 100 vofr cisco class toto ! map-class frame-relay toto frame-relay voice-bandwidth t frame-relay min-cir x frame-relay cir s frame-relay bc u frame-relay fragment y
hostname mc3810B ! interface serial 0 ip address xxx.xxx.xxx 255.255.255.0 frame-relay traffic shaping frame-relay class toto ! frame-relay interface-dlci 100 vofr cisco ! map-class frame-relay toto frame-relay voice-bandwidth t frame-relay min-cir x frame-relay cir s frame-relay bc u frame-relay fragment y

This configuration uses FRF.11 Annex C fragmentation.

Cisco MC3810 Concentrators Using Different VoFR Configuration Methods

On the Cisco MC3810, there are two different configuration methods for configuring VoFR on an interface. You can either use the vofr cisco command or the frame-relay interface-dlci dlci voice-encap size command. The first configuration method uses the vofr cisco command, which automatically provides support for WFQ and matches the configuration method supported on the Cisco 2600 and Cisco 3600 series routers.

The second configuration method uses the frame-relay interface-dlci voice-encap command, which does not automatically provide support for any queueing mechanism. However, if you enter the no frag-pre-queuing command, you can then statically configure either WFQ, priority queueing, or custom queueing on the interface. This second configuration method can be used to configure VoFR connections to Cisco 2600 and Cisco 3600 series routers.

You must choose one configuration method, because these commands are mutually exclusive on an interface.

Figure 76 shows an example of a VoFR PVC between two Cisco MC3810 concentrators, with each concentrator using a different VoFR configuration method. In this example, the VoFR PVC on the first Cisco MC3810 is configured using the vofr cisco command; on the second concentrator, the VoFR PVC is configured using the frame-relay interface-dlci voice-encap command.

Figure76: VoFR PVC Between Two Cisco MC3810 Concentrators Using Different Configuration Methods

Router A (Cisco 3600) Router B (Cisco 3600)

hostname mc3810A ! interface serial 0 ip address xxx.xxx.xxx 255.255.255.0 frame-relay traffic shaping ! frame-relay interface-dlci 100
vofr cisco class toto ! map-class frame-relay toto frame-relay min-cir in x frame-relay cir s frame-relay bc u frame-relay fragment y frame-relay voice-bandwidth t
hostname mc3810B ! interface serial 0 ip address xxx.xxx.xxx 255.255.255.0 frame-relay traffic shaping frame-relay class toto ! frame-relay interface-dlci 100 voice-encap 320 no frag-pre-queuing fair-queue 64 512 18 ! map-class frame-relay toto frame-relay min-cir in x frame-relay cir s frame-relay bc u

This configuration uses FRF.11 Annex C fragmentation.

Cisco-Trunk (Private Line) Calls Between Two Routers

Figure 77 shows an example of VoFR Cisco-trunk (private line) calls between two routers.

Figure77: Permanent Switched Calls Between Two Routers

Router A (Cisco 3600) Router B (Cisco 3600)

hostname 3600A ! interface serial 0/0 ip address xxx.xxx.xxx 255.255.255.0 encapsulation frame-relay frame-relay traffic shaping frame-relay interface-dlci 100 class voice vofr cisco ! map-class frame-relay voice frame relay cir s frame relay bc u frame relay voice bandwidth v frame-relay min-cir x frame-relay fragment y ! voice-port 2/0/0 connection trunk 6001 answer-mode ! dial-peer voice 1 pots destination pattern 7001 port 2/0/0 dial-peer voice 2 vofr codec x bytes y destination-pattern 6001 session protocol cisco-switched session target Sn 100
hostname mc3810B ! interface serial 0 ip address xxx.xxx.xxx 255.255.255.0 encapsulation frame-relay frame-relay traffic shaping frame-relay interface-dlci 100 class voice vofr cisco ! map-class frame-relay voice frame relay cir s frame relay bc u frame relay voice bandwidth v frame-relay min-cir x frame-relay fragment y ! voice-port 1/5 connection trunk 7001 ! dial-peer voice 2 pots destination pattern 6001 port 1/5 ! dial-peer voice 4 vofr codec x bytes y destination-pattern 7001 session protocol cisco-switched session target Sn 100

FRF.11 Trunk Calls Between Two Routers

Figure 78 shows an example of FRF.11 trunk calls configured between two routers.

Figure78: FRF.11 Trunk Calls Between Two Routers

Router A (Cisco 3600) Router B (Cisco 3600)

hostname 3600A ! interface serial 0/0 ip address xxx.xxx.xxx 255.255.255.0 encapsulation frame-relay frame-relay traffic shaping frame-relay interface-dlci 100 class voice vofr data 4 ! map-class frame-relay voice frame-relay cir s frame-relay min-cir in x frame-relay bc u frame-relay voice bandwidth v frame-relay fragment y ! voice-port 2/0/0 connection trunk 6001 ! dial-peer voice 1 pots destination-pattern 7001 port 2/0/0 ! dial-peer voice 2 vofr codec x bytes y destination-pattern 6001 session protocol frf11-trunk session target Sn 100 d called-number 7001 dtmf-relay vad
hostname mc3810B ! interface serial 0 ip address xxx.xxx.xxx 255.255.255.0 encapsulation frame-relay frame-relay traffic shaping frame-relay interface-dlci 100 class voice vofr data 4 ! map-class frame-relay voice frame-relay cir s frame-relay min-cir in x frame-relay bc u frame-relay voice bandwidth v frame-relay fragment y ! voice-port 1/5 connection trunk 7001 ! dial-peer voice 2 pots destination-pattern 6001 port 1/5 ! dial-peer voice 4 vofr codec x bytes y destination-pattern 7001 session protocol frf11-trunk session target Sn 100 d dtmf-relay vad

Tandem Configuration with Three Routers for Switched Calls

Figure 79 shows an example of a tandem configuration with two Cisco 3600 routers as endpoints, and a third Cisco 3600 as a tandem node.

Figure79: Tandem Configuration with Three Routers for Switched Calls

Router A (Cisco 3600) Endpoint Router C (Cisco 3600) Tandem Node Router B (Cisco 3600) Endpoint

hostname 3600A ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr data 4 call-control 5 ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 pots destination-pattern 1001 port 1/0/0 ! dial-peer voice 2 vofr destination-pattern 2... session target serial 0/0 100 ! voice-port 1/0/0
hostname3600C
! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr data 4 call-control 5 ! interface serial 0/1 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 200 class voice vofr ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 vofr destination-pattern 1... session target serial 0/0 100 ! dial-peer voice 2 vofr destination-pattern 2... session target serial 0/1 200
hostname3600B ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr data 4 call-control 5 ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 posts destination-pattern 2001 port 1/0/0 ! dial-peer voice 2 vofr destination-pattern 1... session target serial 0/0 200 ! voice-port 1/0/0

Tandem Configuration with a Cisco MC3810 Tandem Node for Switched Calls

Figure 80 shows an example of a tandem configuration with a Cisco MC3810 acting as a tandem node.

NoteWhen a Cisco MC3810 is on a VoFR network, the configuration for connections to and from the Cisco MC3810 is slightly different than for other routers that support VoFR. The vofr cisco command is required for those connections.

Figure80: Tandem Configuration with a Cisco MC3810 Tandem Node for Switched Calls

Router A (Cisco 3600) Endpoint Router C (Cisco MC3810) Tandem Node Router B (Cisco 3600) Endpoint

hostname 3600A ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr data 4 call-control 5 ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 pots destination-pattern 1001 port 1/0/0 ! dial-peer voice 2 vofr destination-pattern 2... session target serial 0/0 100 ! voice-port 1/0/0
hostname mc3810C ! interface serial 0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr cisco ! interface serial 1 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 200 class voice vofr cisco ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 vofr destination-pattern 1... session target serial 0/0 100 ! dial-peer voice 2 vofr destination-pattern 2... session target serial 0/1 200
hostname3600B ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr data 4 call-control 5 ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 posts destination-pattern 2001 port 1/0/0 ! dial-peer voice 2 vofr destination-pattern 1... session target serial 0/0 200 ! voice-port 1/0/0

Tandem Configuration with a Cisco MC3810 Endpoint Node for Switched Calls

Figure 81 shows an example of a tandem configuration with a Cisco MC3810 acting as an endpoint node.

Figure81: Tandem Configuration with a Cisco MC3810 Endpoint Node

Router A (Cisco 2600) Endpoint Router C (Cisco 3600) Tandem Node Router B (Cisco MC3810) Endpoint

hostname 2600A ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr data 4 call-control 5 ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 pots destination-pattern 1001 port 1/0/0 ! dial-peer voice 2 vofr destination-pattern 2001 session target serial 0/0 100 ! voice-port 1/0/0
hostname 3600C ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr data 4 call-control 5 ! interface serial 0/1 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 200 class voice vofr cisco ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 vofr destination-pattern 1001 session target serial 0/0 100 ! dial-peer voice 2 vofr destination-pattern 2001 session target serial 0/1 200
hostname mc3810B ! interface serial 0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 200 class voice vofr cisco ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 pots destination-pattern 2001 port 1/1 ! dial-peer voice 2 vofr destination-pattern 1001 session target serial 0 200 ! voice-port 1/1

Tandem Configuration with a Cisco MC3810 Endpoint Node for Cisco-Trunk (Private Line) Calls

Figure 82 shows an example of a tandem configuration with a Cisco MC3810 acting as an endpoint node for Cisco-trunk (private line) calls. When a Cisco MC3810 is on a VoFR network, the configuration for connections to and from the Cisco MC3810 is slightly different than for other routers that support VoFR. The vofr cisco command is required for those connections.

Figure82: Tandem Configuration with a Cisco MC3810 Endpoint Node for Permanent Switched Call

Router A (Cisco 2600) Endpoint Router C (Cisco 3600) Tandem Node Router B (Cisco MC3810) Endpoint

hostname 2600A ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr data 4 call-control 5 ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 pots destination-pattern 1001A port 1/0/0 ! dial-peer voice 2 vofr destination-pattern 2... session target serial 0/0 100 ! voice-port 1/0/0 connection trunk 2001A answer-mode
hostname 3600C ! interface serial 0/0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr data 4 call-control 5 ! interface serial 0/1 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 200 class voice vofr cisco ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 vofr destination-pattern 1001 session target serial 0/0 100 ! dial-peer voice 2 vofr destination-pattern 2001 session target serial 0/1 200
hostname mc3810B ! interface serial 0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 200 class voice vofr cisco ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 pots destination-pattern 2001A port 1/1 ! dial-peer voice 2 vofr destination-pattern 1... session target serial 0 200 ! voice-port 1/1 connection trunk 1001A

Tandem Configuration with All Cisco MC3810 Concentrators for Switched Calls

Figure 83 shows an example of a tandem configuration with Cisco MC3810 concentrators as both endpoint and tandem nodes.

NoteWhen a Cisco MC3810 is on a VoFR network, the configuration for connections to and from the Cisco MC3810 is slightly different than for other routers that support VoFR. Either the vofr cisco command or the frame-relay interface-dlci voice-encap command is required for these connections on the Cisco MC3810.

Figure83: Tandem Configuration with all Cisco MC3810 Concentrators

Router A (Cisco MC3810) Endpoint Router C (Cisco MC3810) Tandem Node Router B (Cisco MC3810) Endpoint

hostname mc3810A ! interface serial 0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr cisco ! map-class frame-relay voice frame-relay cir a frame-relay bc b frame-relay voice bandwidth c frame-relay min-cir t ! dial-peer voice 1 pots destination-pattern 1001 port 1/1 ! dial-peer voice 2 vofr destination-pattern 2... session target serial 0 100 ! voice-port 1/1
hostname mc3810C ! interface serial 0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr cisco ! interface serial 1 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 200 class voice vofr cisco ! map-class frame-relay voice frame-relay cir a frame-relay bc b frame-relay voice bandwidth c frame-relay min-cir t ! dial-peer voice 1 vofr destination-pattern 1... session target serial 0 100 ! dial-peer voice 2 vofr destination-pattern 2... session target serial 1 200
hostname mc3810B ! interface serial 0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 200 class voice vofr cisco ! map-class frame-relay voice frame-relay cir a frame-relay min-cir t frame-relay bc b frame-relay voice bandwidth c frame-relay fragment d ! dial-peer voice 1 pots destination-pattern 2001 port 1/1 ! dial-peer voice 2 vofr destination-pattern 1... session target serial 0 200 ! voice-port 1/1

Cisco-Trunk Call with Hunt Groups

Figure 84 shows an example of a Cisco-trunk (private line) call with hunt groups configured. In this example, the two routers are in master-slave mode with a backup path. RouterB is configured as a slave and Router A is configured as the master. The master makes periodic attempts to establish the trunk until the trunk is established. Two dial peers match the destination string configured in the voice port, but because one dial peer has a higher preference, the call setup is attempted through that dial peer. If the call setup fails, the master can continue attempting call setups using the next available dial peer. After all dial peers are exhausted, the master can continue following the list cyclically by starting again from the dial peer with the highest preference.

Figure84: Cisco Trunk (Private Line) Call with Hunt Groups

Router A (Cisco 3600) Router B (Cisco 3600)

hostname mc3810A ! interface serial 0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr cisco ! interface serial 1 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 200 class voice vofr cisco ! map-class frame-relay voice frame-relay cir a frame-relay bc b frame-relay voice bandwidth c frame-relay min-cir t ! dial-peer voice 1 pots destination-pattern 1001A port 1/1 ! dial-peer voice 100 vofr destination-pattern 2... session target serial 0 100 preference 1 ! dial-peer voice 200 vofr destination-pattern 2... session target serial1 200 preference 2 ! voice-port 1/1 connection trunk 2005A description FXO port
hostname mc3810B ! interface serial 0 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 100 class voice vofr cisco ! interface serial 1 encapsulation frame-relay frame-relay traffic-shaping frame-relay interface-dlci 200 class voice vofr cisco ! map-class frame-relay voice frame-relay cir a frame-relay bc b frame-relay voice bandwidth c frame-relay min-cir t ! dial-peer voice 1 pots destination-pattern 2001A port 1/1 ! dial-peer voice 100 vofr destination-pattern 1... session target serial0 100 preference 1 ! dial-peer voice 200 vofr destination-pattern 1... session target serial 1 200 preference 2 ! voice-port 1/1 description FXS port connection trunk 1001A answer-mode

	Command	Purpose
Step1	Router(config-dial-peer)# preference value	(Optional) Configures a preference for the POTS dial peer. The value is a number from 0-10 where the lower the number, the higher the preference. If POTS and voice-network (VoFR) peers are mixed in the same hunt group, POTS dial peers will be searched first, even if a voice-network peer has a higher preference number.
Step2	Router(config-dial-peer)# prefix string	(Optional) Assign the prefix of the dialed digits for the dial peer. Valid numbers for the string argument are 0 through 9, and a comma (,). Use a comma to include a pause in the prefix When an outgoing call is initiated to this dial peer, the prefix string value is sent to the telephony interface first, before the telephone number associated with the dial peer.
Step3	Router(config-dial-peer)# direct-inward-dial	(Optional for the Cisco 2600/3600 series only) Enables the Direct Inward Dial (DID) call treatment for the incoming called number.
Step4	Router(config-dial-peer)# incoming called-number string	(Optional for the Cisco 2600/3600 series only) Identifies the service type for a call on a router handling both voice and modem calls.
Step5	Router(config-dial-peer)# max-conn number	(Optional for the Cisco 2600/3600 series only) Specifies the maximum number of allowed connections to and from the POTS dial peer. The valid range is 1-2147483647.
Step6	Router(config-dial-peer)# forward-digits {num-digit \| all \| implicit}	(Optional for the Cisco MC3810 only) If you are using the forward-digits feature, configures the digit-forwarding method that will be used on the dial peer. The valid range for the number of digits forwarded (num-digit) is 0-32. The default value is implicit, in which the exactly matched digits are not forwarded. Only digits matched by the wildcard pattern are forwarded.

	Command	Purpose
Step1	Router(config)# dial-peer voice number vofr	Defines a VoFR dial peer and enters dial-peer configuration mode. All subsequent commands that you enter in dial-peer voice mode before you exit will apply to this dial peer. The number value tag identifies the dial peer and must be unique on the router. Do not duplicate a specific number tag.
Step2	Router(config-dial-peer)# destination-pattern [+]string[T]	Configures the dial peer destination pattern. The same restrictions for the string listed in the POTS dial-peer configuration also apply to the VoFR destination pattern.
Step3	Router(config-dial-peer)# session target interface dlci	Configures the Frame Relay session target for the dial peer.
Step4	Router(config-dial-peer)# preference value	(Optional) Configures a preference for the VoFR dial peer. The value argument is a number from 0 to 10 where the lower the number, the higher the preference in hunt groups.

	Command	Purpose
Step1	Router(config)# dial-peer voice tag {pots \| vofr}	Enters dial-peer configuration mode for the specified dial peer.
Step2	Router(config-dial-peer)# huntstop	Disables dial-peer hunting on the dial peer. Once you enter this command, no further hunting will be allowed if a call fails on the specified dial peer.

Type of Call	Frame Relay DLCI interface Command to Enter	Data Fragmentation Supported by vofr Command	Session Protocol Command to Enter in Dial Peer Mode	Voice Port Connection Command to Enter
Switched call (user dialed or auto-ringdown) to other Cisco 2600/3600 or 7200 routers	vofr [data cid] [call-control [cid]]¹	FRF.11 Annex C	session protocol cisco-switched²	For user dialed calls: none For auto-ringdown calls: connection plar destination-string
Switched call (user dialed or auto-ringdown) to a CiscoMC3810	vofr cisco³	Cisco proprietary⁴	session protocol cisco-switched	For user dialed calls: none For auto-ringdown calls: connection plar destination-string
Cisco-trunk permanent call (private-line) to other Cisco 2600/3600 or 7200 routers	vofr data cid call-control cid	FRF.11 Annex C	session protocol cisco-switched	connection trunk destination-string [answer mode]
Cisco-trunk permanent call (private-line) to a CiscoMC3810	vofr cisco	Cisco proprietary	session protocol cisco-switched	connection trunk destination-string [answer mode]
FRF.11 trunk call (private-line)	vofr [data cid] [call-control cid]⁵	FRF.11 Annex C	session protocol frf11-trunk	connection trunk destination-string [answer mode]

	Command	Purpose
Step1	Router(config-if)# frame-relay interface-dlci dlci	Configures the Frame Relay DLCI and enters DLCI configuration mode.
Step2	Router(config-if)# vofr cisco	Configures the Frame Relay DLCI to support VoFR and the Cisco proprietary fragmentation implementation. When this command is entered, data CID 4 and call-control CID5 are automatically assigned. If you are configuring user-dialed calls, this procedure is completed. If configuring auto-ringdown calls, proceed to the next step.
Step3	For Cisco 2600 and 3600 series analog voice ports: router(config)# voice-port slot/subunit/port For Cisco 2600 and 3600 series digital voice ports: router(config)# voice-port slot/port:ds0-group	Enters voice-port configuration mode.
Step4	Router(config-voiceport)# connection plar destination-string	(Optional) For auto-ringdown calls, configures the PLAR connection, specifying the telephone number in the destination-string argument.

Access Rate	Recommended Data Fragmentation Size¹
64 kbps	80 bytes
128 kbps	160 bytes
256 kbps	320 bytes
512 kbps	640 bytes
1536 kbps (full T1)	1600 bytes
2048 kbps (full E1)	1600 bytes

End Node(s)	Tandem Node	vofr Command to Enter for the Frame Relay DLCI
Cisco 2600/3600	Cisco 2600, Cisco 3600, or Cisco7200	vofr call-control
Cisco 2600/3600 and CiscoMC3810	Cisco MC3810	vofr cisco
Cisco MC3810	Cisco 2600, Cisco 3600, or Cisco7200	vofr cisco

Table of Contents

Configuring Voice over Frame Relay