Cisco IOS Multiservice Applications Configuration Guide, Release 12.1 - Configuring Voice Ports for Voice over IP [Cisco IOS Software Releases 12.1 Mainline]

Table Of Contents

Configuring Voice Ports for Voice over IP

Configuring Analog Voice Ports

Configuring FXO or FXS Voice Ports

Fine-Tuning FXO and FXS Voice Ports

Verifying FXO and FXS Voice Port Configuration

Troubleshooting Tips

Configuring E&M Voice Ports

Fine-Tuning E&M Voice Ports

Verifying E&M Voice Port Configuration

Troubleshooting Tips

Configuring Digital Voice Ports

Configuring Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers

Timing

Framing

Line Encoding

Restrictions

Prerequisites

Configuring Voice Card and T1 Controller Settings

Configuring Voice Port Parameters

Configuring 1- and 2-Port T1/E1 Multiflex VWICs on Cisco 2600 and 3600 Series Routers

Restrictions

Prerequisites

Configuring Voice Cards and DS0s

Configuring E1 or T1 Controllers

Configuring Drop-and-Insert

Configuring Voice Ports Parameters

Configuring ISDN BRI VoIP for Cisco 2600 and 3600 Series VICs

Prerequisites

Configuring BRI Interfaces

Configuring T1/E1 High-Capacity Digital Voice Port Adapters for Cisco 7200 Series Routers

Restrictions

Prerequisites

Configuring the DSPfarm Interface

Configuring Card Type and T1 Controller Settings

Configuring Card Type and E1 Controller Settings

Configuring Voice Ports

Configuring ISDN PRI Voice Ports

Configuring Voice Ports

Configuring E1 R2 Signalling for VoIP

Verifying E1 R2 Signalling Configuration

Troubleshooting Tips

Configuring T1 CAS

T1 CAS Signalling Systems

Channelized T1 Robbed-Bit Features

Verifying T1 CAS Configuration

Troubleshooting Tip

Configuring Busyout Monitor for VoIP

Activating the Voice Port

Voice Port Configuration Examples

Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers Configuration Examples

Routed Digits—Switched VoIP Calls

FRF.12—Switched VoIP Calls

Routing Calls Through an H.323 Gatekeeper

PLAR Configuration—Switched VoIP Calls

Connection Trunk Configuration—Permanent VoIP Calls

Drop-and-Insert Sample Configuration

1- and 2-Port T1/E1 Multiflex VWICs on Cisco 2600 and 3600 Series Routers Configuration Examples

Drop-and-Insert with VoIP and PSTN Services

Drop-and-Insert with Data and PSTN Services

T1 Configuration

E1 Configuration

Drop-and-Insert with PSTN, Data, and VoIP Services

Cisco 3600 Series and Cisco 2600 Series ISDN BRI Configuration Examples

Router A: Connection to a PBX

Router B: Connection to PSTN

Configuring VoIP for E1 R2 Signalling Example

Configuring VoIP for T1-CAS Example

T1/E1 High-Capacity Digital Voice Port Adapters for the Cisco 7200 Series Configuration Examples

Routed Digits—Switched VoIP Calls

FRF.12—Switched VoIP Calls

Routing Calls Through an H.323 Gatekeeper

PLAR Configuration—Switched VoIP Calls

Connection Trunk Configuration—Permanent VoIP Calls

Drop-and-Insert Sample Configuration

Busyout Monitor Configuration Example

Configuring Voice Ports for Voice over IP

VoIP supports both analog and digital telephony connections. The connection supported (and the associated signalling, whether analog or digital) depends on the type of VNM or VFC installed in your Cisco router or access server.

This chapter shows you how to configure voice ports for Voice over IP. This chapter contains the following sections:

•Configuring Analog Voice Ports

•Configuring Digital Voice Ports

•Voice Port Configuration Examples

For a complete description of the commands used in this chapter, refer to the Cisco IOS Multiservice Applications Command Reference publication. To locate documentation of other commands that appear in this chapter, use the command reference master index or search online.

Configuring Analog Voice Ports

Analog voice signalling in VoIP is sent via an analog voice port. Analog voice ports support three basic voice signalling types:

•FXO. Foreign Exchange Office interface. The FXO interface is an RJ-11 connector that allows a connection to be directed at the PSTN central office (or to a standard PBX interface, if the local telecommunications authority permits). This interface is of value for off-premises extension applications.

•FXS. The Foreign Exchange Station interface. This interface is an RJ-11 connector that allows connection for basic telephone equipment, keysets, PBXs, and supplies ring, voltage, and dial tone.

•E&M. The "ear and mouth" interface (or "recEive and transMit") interface. This interface is an RJ-48 connector that allows connection for PBX trunk lines (tie lines). It is a signalling technique for 2-wire and 4-wire telephone and trunk interfaces.

The VMN or VFC installed in your Cisco device determines the type of analog signalling a voice port sends.

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

To configure the analog voice ports in Voice over IP, perform the following tasks:

•Configuring FXO or FXS Voice Ports

•Fine-Tuning FXO and FXS Voice Ports

•Configuring E&M Voice Ports

•Fine-Tuning E&M 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. If you need to change the default configuration for these voice ports, use the following commands beginning in privileged EXEC mode:

Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# voice-port slot

Identifies the voice port you want to configure and enters voice-port configuration mode.

Note The syntax of the voice-port command is specific to Cisco hardware platforms. For information on how to configure this command for your specific device, refer to the voice-port command documentation in the Cisco IOS Multiservice Applications Command Reference publication.

Step 3

Router(config-voiceport)# dial-type {dtmf | pulse}

(For FXO ports only) Selects the appropriate dial type for out-dialing, either touchtone (DTMF) or pulse.

Step 4

Router(config-voiceport)# signal {loop-start | ground-start}

Selects the appropriate signal type for this interface. With the loop-start keyword, only one side of a connection can hang up. (The default signalling type is loop-start.) With ground-start signalling, both sides of a connection can place calls and hang up.

Step 5

Router(config-voiceport)# cptone country

Selects the appropriate voice call progress tone for this interface.

For a list of supported countries, refer to the Cisco IOS Multiservice Applications Command Reference publication.

Step 6

Router(config-voiceport)# ring frequency {25 | 50}

(For FXS ports only) Selects the appropriate ring frequency (in Hertz) specific to the equipment attached to this voice port.

Step 7

Router(config-voiceport)# ring number number

(For FXO ports only) Specifies the maximum number of rings to be detected before answering a call.

Step 8

Router(config-voice-port)# connection {plar | trunk} string

(Optional) Sets up a connection mode for the voice port.

The plar keyword specifies a private line, automatic ring down (PLAR) connection, which rings a remote telephone when the dial peer goes off hook.

The trunk keyword specifies a straight tie-line connection to a PBX.

The string argument specifies the remote telephone number or significant start digits of the number.

Step 9

Router(config-voiceport)# music-threshold number

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

Step 10

Router(config-voiceport)# description string

(Optional) Attaches descriptive text about this voice-port connection.

Step 11

Router(config-voiceport)# comfort-noise

(Optional) Specifies that background noise will be generated.

Step 12

Router(config-voiceport)# no shutdown

Activates the voice port.

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

Fine-Tuning FXO and FXS Voice Ports

Depending on the specifics of 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.

Note In 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 privileged EXEC mode:

:

Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# voice-port slot

Identifies the voice port you want to configure and enters voice-port configuration mode.

Note The syntax of the voice-port command is specific to Cisco hardware platforms. For information on how to configure this command for your specific device, refer to the voice-port command documentation in the Cisco IOS Multiservice Applications Command Reference publication.

Step 3

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.

Step 4

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.

Step 5

Router(config-voiceport)# echo-canel enable

Enables echo-cancellation of voice that is sent out the interface and received back on the same interface.

Step 6

Router(config-voiceport)# echo-canel coverage value

Adjusts the size (in milliseconds) of the echo-cancel. Acceptable values are 16, 24, and 32.

Step 7

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.)

Step 8

Router(config-voiceport)# timeouts initial seconds

Specifies the number of seconds the system will wait for the caller to input the first digit of the dialed digits. Valid entries for this command are from 0 to 120.

Step 9

Router(config-voiceport)# timeouts interdigits seconds

Specifies the number of seconds the system will wait (after the caller has input the initial digit) for the caller to input a subsequent digit. Valid entries for this command are from 0 to 120.

Step 10

Router(config-voiceport)# timing digits milliseconds

If the voice-port dial type is DTMF, configures the DTMF digit signal duration. The range of the DTMF digit signal duration is from 50 to 100. The default is 100.

Step 11

Router(config-voiceport)# timing inter-digits milliseconds

If the voice-port dial type is DTMF, configures the DTMF interdigit signal duration. The range of the DTMF interdigit signal duration is from 50 to 500. The default is 100.

Step 12

Router(config-voiceport)# timing pulse digit milliseconds

(FXO ports only) If the voice-port dial type is pulse, configures the pulse digit signal duration. The range of the pulse digit signal duration is from 10 to 20. The default is 20.

Step 13

Router(config-voiceport)# timing pulse-inter-digit milliseconds

(FXO ports only) If the voice-port dial type is pulse, configures the pulse interdigit signal duration. The range of the pulse interdigit signal duration is from 100 to 1000. The default is 500.

Step 14

Router(config-voiceport)# no shutdown

Activates the voice port.

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

Verifying FXO and 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.

•Use the show voice-port command to verify that the data configured is correct.

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.

•Use the show voice-port command to make sure that the port is enabled. If the port is offline, use the no shutdown command.

•If you have configured E&M interfaces, make sure that the values pertaining to your specific PBX setup, such as timing or type, are correct.

•Check that the VNM has been correctly installed. For more information, refer to the installation document, Voice Network Module and Voice Interface Card Configuration Note, that came with your VNM.

Configuring E&M Voice Ports

Unlike with 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 IP network. E&M voice-port values must match those specified by the particular PBX device 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 privileged EXEC mode:

Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# voice-port slot

Identifies the voice port you want to configure and enters voice-port configuration mode.

Note The syntax of the voice-port command is specific to Cisco hardware platforms. For information on how to configure this command for your specific device, refer to the voice-port command documentation in the Cisco IOS Multiservice Applications Command Reference publication.

Step 3

Router(config-voiceport)# dial-type {dtmf | pulse}

Selects the appropriate dial type for out-dialing, either touchtone (DTMF) or pulse.

Step 4

Router(config-voiceport)# signal {wink-start | immediate | delay-dial}

Selects the appropriate signal type for this interface. The wink-start keyword indicates that the calling side seizes the line by going off-hook on its E lead, then waits for a short off-hook "wink" indication on its M lead from the called side before sending address information as DTMF digits.

The immediate keyword indicates that the calling side seizes the line by going off-hook on its E lead and sends address information as DTMF digits. Immediate signalling is used for E&M tie trunk interfaces.

The delay-dial keyword indicates that the calling side seizes the line by going off-hook on its E lead. After a timing interval, the calling side looks at the supervision from the called side. If the supervision is on-hook, the calling side starts sending information as DTMF digits; otherwise the calling side waits until the called side goes on-hook and then starts sending address information. Delay-dial signalling is used for E&M tie trunk interfaces.

Step 5

Router(config-voiceport)# cptone country

Selects the appropriate voice call progress tone for this interface.

For a list of supported countries, refer to the Cisco IOS Multiservice Applications Command Reference publication.

Step 6

Router(config-voiceport)# operation {2-wire | 4-wire}

Selects the appropriate cabling scheme for this voice port.

Step 7

Router(config-voiceport)# type {1 | 2 | 3 | 5}

Selects 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

Step 8

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.

Step 9

Router(config-voice-port)# connection {plar | trunk} string

(Optional) Sets up a connection mode for the voice port.

The plar keyword specifies a PLAR connection, which rings a remote telephone when the dial peer goes off-hook.

The trunk keyword specifies a straight tie-line connection to a PBX.

The string argument specifies the remote telephone number or significant start digits of the number.

Step 10

Router(config-voiceport)# music-threshold number

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

Step 11

Router(config-voiceport)# description string

(Optional) Attaches descriptive text about this voice-port connection.

Step 12

Router(config-voiceport)# comfort-noise

(Optional) Specifies that background noise will be generated.

Step 13

Router(config-voiceport)# no shutdown

Activates the voice port.

:

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

Fine-Tuning E&M Voice Ports

Depending on the specifics of your particular network, you may need to adjust (or fine-tune) voice parameters involving timing, input gain, and output attenuation for E&M voice ports.

Note In 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 privileged EXEC mode:

Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# voice-port slot

Identifies the voice port you want to configure and enters voice-port configuration mode.

Note The syntax of the voice-port command is specific to Cisco hardware platforms. For information on how to configure this command for your specific device, refer to the voice-port command documentation in the Cisco IOS Multiservice Applications Command Reference publication.

Step 3

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 for the value argument are from -6 to 14.

Step 4

Router(config-voiceport)# output attenuation value

Specifies (in decibels) the amount of attenuation at the transmit side of the interface. Acceptable values for the value argument are from 0 to 14.

Step 5

Router(config-voiceport)# echo-cancel enable

Enables echo-cancellation of voice that is sent out the interface and received back on the same interface.

Step 6

Router(config-voiceport)# echo-cancel coverage value

Adjusts the size (in milliseconds) of the echo-cancel. Acceptable values for the value argument are 16, 24, and 32.

Step 7

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.)

Step 8

Router(config-voiceport)# timeouts initial seconds

Specifies the number of seconds the system will wait for the caller to input the first digit of the dialed digits. Valid entries for the seconds argument are from 0 to 120.

Step 9

Router(config-voiceport)# timeouts interdigit seconds

Specifies the number of seconds the system will wait (after the caller has input the initial digit) for the caller to input a subsequent digit. Valid entries for the seconds argument are from 0 to 120.

Step 10

Router(config-voiceport)# timing clear-wait milliseconds

Specifies the minimum amount of time between the inactive seizure signal and the call being cleared. Valid entries for the milliseconds argument are from 200 to 2000 milliseconds.

Step 11

Router(config-voiceport)# timing delay-duration milliseconds

Specifies the delay signal duration for delay dial signalling. Valid entries for the milliseconds arguments are from 100 to 5000 milliseconds.

Step 12

Router(config-voiceport)# timing delay-start milliseconds

Specifies the minimum delay time from outgoing seizure to outdial address. Valid entries for the milliseconds argument are from 20 to 2000 milliseconds.

Step 13

Router(config-voiceport)# timing dial-pulse min-delay milliseconds

Specifies the time between generation of wink-like pulses. Valid entries for the milliseconds argument are from 0 to 5000 milliseconds.

Step 14

Router(config-voiceport)# timing digit milliseconds

If the voice-port dial type is DTMF, configures the DTMF digit signal duration. Valid entries for the milliseconds argument are from 50 to 100 milliseconds.

Step 15

Router(config-voiceport)# timing inter-digit milliseconds

If the voice-port dial type is DTMF, specifies the DTMF interdigit duration. Valid entries for the milliseconds argument are from 50 to 500 milliseconds.

Step 16

Router(config-voiceport)# timing pulse pulse-per-second

If the voice-port dial type is pulse, specifies the pulse dialing rate. Valid entries for the pulse-per-second argument are from 10 to 20 pulses per second.

Step 17

Router(config-voiceport)# timing pulse-inter-digit milliseconds

If the voice-port dial type is pulse, specifies the pulse dialing interdigit timing. Valid entries for the milliseconds argument are 100 to 1000 milliseconds.

Step 18

Router(config-voiceport)# timing wink-duration milliseconds

Specifies the maximum wink signal duration. Valid entries for the milliseconds argument are from 100 to 400 milliseconds.

Step 19

Router(config-voiceport)# timing wink-wait milliseconds

Specifies the maximum wink-wait duration for a wink start signal. Valid entries for the milliseconds argument are from 100 to 5000 milliseconds.

Step 20

Router(config-voiceport)# no shutdown

Activates the voice port.

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

Verifying E&M 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.

•Use the show voice-port command to verify that the data configured is correct.

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.

•Use the show voice-port command to make sure that the port is enabled. If the port is offline, use the no shutdown command.

•If you have configured E&M interfaces, make sure that the values pertaining to your specific PBX setup, such as timing or type, are correct.

•Check that the VNM has been correctly installed. For more information, refer to the installation document that came with your VNM.

Configuring Digital Voice Ports

When a digital interface on a Cisco access server or router is carrying voice data, it is referred to as a digital voice port. Cisco offers a variety of options for sending digital voice signals, depending on the specific Cisco router or access server.

The following sections include tasks for configuring digital voice port types:

•Configuring Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers

•Configuring 1- and 2-Port T1/E1 Multiflex VWICs on Cisco 2600 and 3600 Series Routers

•Configuring ISDN BRI VoIP for Cisco 2600 and 3600 Series VICs

•Configuring T1/E1 High-Capacity Digital Voice Port Adapters for Cisco 7200 Series Routers

•Configuring ISDN PRI Voice Ports

•Configuring E1 R2 Signalling for VoIP

•Configuring T1 CAS

•Configuring Busyout Monitor for VoIP

Configuring Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers

Digital T1 packet voice trunk network modules for Cisco 2600 and 3600 series routers allow enterprises or service providers, using the equipped routers as CPE, to deploy digital voice and fax relay. These modules receive constant bit-rate telephony information over T1 interfaces and can convert that information to a compressed format, so that it can be sent as VoIP. The digital T1 packet voice trunk network modules can connect to either a PBX (or similar telephony device) or to a central office (CO) in order provide PSTN connectivity.

T1 digital VoIP includes the following functionality:

•T1 channel associated signalling (CAS) for the following line-signalling types:

–E&M immediate start

–E&M wink start

–E&M delay start (also called "dial repeating")

–FXS and FXO loop start

–FXS and FXO ground start

•Dynamic bandwidth allocation using voice activity detection (VAD)

•Drop-and-insert capability, allowing the interchange of time-division multiplexing (TDM) slots between the ports on a two-port T1 multiflex trunk voice/WAN interface card installed in a digital T1 packet voice trunk network module

•Support for a wide range of ITU-T G-series compression specifications

•Depending on codec complexity, either 30 or 60 channels of compressed voice

•High-quality voice endpoint-standard features, such as high-quality echo cancellation, silence suppression, comfort noise generation, and DTMF relay

•Group 3 fax relay

•Support for the following framing formats and line coding:

–Super frame (SF)

–Extended super frame (ESF)

–Alternate mark inversion (AMI) line coding

–Binary 8-zero substitution (B8ZS) line coding

You must set timing, signalling, framing, and line encoding as follows:

•Timing. Digital T1 interfaces require not only that you set timing but also that you consider the source of the timers.

•Signalling. Digital T1 interfaces require that you specify a signalling type. The following signalling types are available:

–CAS

–E&M

–FXO and FXS

•Framing. Digital T1 interfaces require that you configure either SF or D4 framing or ESF framing. Set the framing format to match that of the PBX or CO that connects to the digital T1 packet voice trunk network module.

•Line Encoding. Digital T1 require that you configure either AMI or B8ZS. Set the line encoding to match that of the PBX or CO that connects to the digital T1 packet voice trunk network module.

Timing

This section describes the five basic timing scenarios that can occur when a digital T1 packet voice trunk network module is connected to a PBX, CO, or both. In all of the following examples, the PSTN (or CO) and the PBX are interchangeable for the purposes of providing or receiving clocking.

The digital T1 module has an on-board Phase-Lock Loop (PLL) chip that can either provide a clock source to both T1 lines or receive clocking that can drive the second T1 line. All timing commands are T1 controller configuration commands.

Single T1 Port Provides Clocking

In this scenario, the digital T1 module is the clock source for the connected device. The PLL generates the clock internally and drives the clocking on the T1 line. Figure 13 shows how the single T1 port provides clocking for the PBX.

Figure 13 Single T1 Port Providing Clock

The following configuration sets up this clocking method:
controller T1 1/0
 framing esf
 linecoding b8zs
 clock source internal
 ds0-group 1 timeslots 1-24 type e&m-wink
Note Generally this method is useful only when connecting to a PBX, key system, or channel bank. A Cisco VoIP Gateway rarely provides clocking to the CO.

Single T1 Port Receiving Clock from the Line

In this scenario, the digital T1 module receives clocking from the connected device (CO or PBX). The PLL clocking is driven by the clock reference on the receive (Rx) side of the T1 connection. Figure 14 shows how the single T1 port receives clocking from the line.

Figure 14 Single T1 Receiving Clock from the Line

The following configuration sets up this clocking method:
controller T1 1/0
 framing esf
 linecoding b8zs
 clock source line
 ds0-group 1 timeslots 1-24 type e&m-wink
Dual T1 Ports, Both Receiving Clock from the Line

In this scenario, the digital T1 has two reference clocks, one from the PBX and another from the CO. Because the PLL can only derive clocking from one source, this case is more complex than the two preceding examples.

Before looking at the details, consider two important concepts that underlie the clocking method:

•Looped-time clocking. The T1 port takes the clock received on its Rx pair and regenerates it on its transmit (Tx) pair. While the port receives clocking, the port is not driving the PLL on the card but is "spoofing" the T1 so that the connected device has a viable clock and does not see slips. PBXs are not designed to accept slips on a T1 line and such slips cause a PBX to drop the link into failure mode. While in looped-time mode, the router often sees slips, but because these are controlled slips, they usually do not force failures of the router T1 port.

•Slips. Slip messages indicate that the T1 port is receiving clock information that is out of phase, that is, out of synchronization. Because the router has only a single PLL, it can experience controlled slips while it receives clocking from two different time sources. The router can usually handle controlled slips because its single PLL architecture anticipates them.

Note Physical layer issues, such as bad cabling or faulty clocking references, can also cause slips.

Figure 15 shows how the dual T1 ports receive clocking from the line.

Figure 15 Dual T1 Ports Receiving Line Clocking

As shown in Figure 15, the PLL derives clocking from the CO and puts the T1 port connected to the PBX into looped-time mode. This is usually best because the CO provides an excellent clock source (and usually requires that it provide that source) and a PBX usually must receive clocking from the other T1.

The following configuration sets up this clocking method:
! The following T1 port is connected to the CO.
controller T1 1/0 
 framing esf
 linecoding b8zs
 clock source line primary
 ds0-group 1 timeslots 1-24 type e&m-wink
!
! The following T1 port is connected to the PBX.
controller T1 1/1 
 framing esf
 linecoding b8zs
 clock source line
 ds0-group 1 timeslots 1-24 type e&m-wink
The clock source line primary command tells the router to use this T1 port to drive the PLL. All other T1 ports configured as clock source line are then put into an implicit loop-timed mode. If the primary T1 port fails or goes down, the other T1 instead receives the clock that drives the PLL. In this configuration, T1 1/1 may see controlled slips, but these slips should not force the line down. This method prevents the PBX from seeing slips.

Dual T1s, One Receiving Clock and One Providing Clock

In this scenario, the digital T1 module receives clocking for the PLL from T1 0 and uses this clock as a reference to clock T1 1. If T1 0 fails, the PLL internally generates the clock reference to drive T1 1.

Figure 16 shows dual T1 ports where one T1 port receives clocking from the line and one T1 port provides clocking.

Figure 16 Dual T1s, One Receiving and One Providing Clocking

The following configuration sets up this clocking method:
controller T1 1/0
 framing esf
 linecoding b8zs
 clock source line 
 ds0-group 1 timeslots 1-24 type e&m-wink
!
controller T1 1/1
 framing esf
 linecoding b8zs
 clock source internal
 ds0-group 1 timeslots 1-24 type e&m-wink 
Dual T1s, Both Receiving Clock from the Router

In this scenario, the router generates the clock for the PLL and therefore for both T1s.

Figure 17 shows how dual T1 ports both receive clocking from the router.

Figure 17 Dual T1s, Both Clocks from Router

The following configuration sets up this clocking method:
controller T1 1/0
 framing esf
 linecoding b8sz
 clock source internal
 ds0-group 1 timeslots 1-24 type e&m-wink
!
controller T1 1/1
 framing esf
 linecoding b8zs
 clock source internal
 ds0-group 1 timeslots 1-24 type e&m-wink
Signalling

There are three types of signalling that you should consider for digital T1:

•CAS. CAS signalling means that instead of having a specific time slot (such as an ISDN D channel in PRI) designated to provide signalling only, signalling bits (on-hook and off-hook) are within the 6th, 12th, 18th, and 24th frames of each time slot. CAS signalling is often called robbed-bit signalling (RBS) because it takes bits from bearer channels and uses them for signalling. CAS signalling must be specified on both ends of the T1 link and is enabled by default on digital T1 packet voice trunk network modules.

Note Digital T1 packet voice trunk network modules support T1 CAS for this Cisco IOS release. Future Cisco IOS releases will support E1, PRI, R2, and common channel signalling (CCS). The digital T1 module can support E&M wink-start, immediate-start, and delay-start signalling, and FXS and FXO ground-start and loop-start signalling.

•E&M signalling. E&M connections can use one of three different signalling types to acknowledge on-hook and off-hook states: wink-start, immediate-start, and delay-start. E&M wink-start is usually preferred because it provides better Answer Supervision (knowledge that the connected device is ready to answer the call). However, not all COs and PBXs can handle wink-start signalling. The E&M connection between the router and switch (CO or PBX) must use matching E&M signalling types or calls will not be connected properly. E&M signalling is defined with the ds0-group controller configuration command, as in the following example:
controller T1 1/0
 ds0-group 1 timeslots 1-24 type e&m-wink-start
Note Currently, wink-start signalling provides only the functionality of Feature Group B and not that of Feature Group D, which will be supported in later Cisco IOS releases.

•FXO and FXS signalling. Although most digital T1 connections used for switch-to-switch (or switch-to-router) trunks are E&M connections, a digital T1 module can also support FXS and FXO connections, which normally is used to provide emulated-OPX (off-premises extensions) from a PBX to remote stations. As a general rule, FXO ports connect to FXS ports. Either ground-start or loop-start signalling is appropriate for these connections. Ground-start signalling provides better disconnect supervision to detect when a remote user has hung up the phone, but ground-start signalling is not available on all PBXs. The FXO or FXS connection between the router and switch (CO or PBX) must use matching signalling or calls will not be connected properly. FXS and FXO signalling are defined with the ds0-group controller configuration command, as in the following examples:
controller T1 1/0
 ds0-group 1 timeslots 1-24 type fxo-ground-start
or
controller T1 1/0
 ds0-group 1 timeslots 1-24 type fxs-loop-start
Note Although some switches (CO or PBX) can specify both an inbound and outbound signalling method, Cisco VoIP gateway routers can only specify one signalling type for both inbound and outbound calls. The switch inbound and outbound signalling types must match, or calls may only work in one direction.

Framing

Digital T1 packet voice trunk network modules support two types of framing for T1 CAS: ESF and SF (also called D4 framing). The framing type of the router and switch (CO or PBX) must match. The framing controller configuration command defines T1 framing, as in the following examples:
controller T1 1/0
 framing esf
or
controller T1 1/0
 framing sf
Line Encoding

Digital T1 packet voice trunk network modules support two types of framing for T1 CAS: B8ZS and AMI. The line encoding of the router and switch (CO or PBX) must match. The linecoding controller configuration command defines T1 framing, as in the following examples:
controller T1 1/0
 linecoding b8zs
or
controller T1 1/0
 linecoding ami 
Restrictions

The following restrictions apply to digital T1 packet voice trunk network module configuration:

•Group 4 fax is not supported.

•The high-density voice network module has one slot for a voice/WAN interface card (VWIC); VWICs supply one or two ports. Only the dual-mode (voice/WAN) multiflex trunk cards are supported in the digital T1 packet voice trunk network module, not older voice interface cards (VICs).

•Drop-and-insert capability is supported only between two ports on the same multiflex card.

•Voice over Frame Relay is not supported.

•Wink-start signalling Feature Group D is not supported.

•CCS and PRI are not supported.

•R2 signalling is not supported.

•Voice over ATM—including ATM Adaptation Layer 5 (AAL5) encapsulation, circuit emulation services (CES), and AAL2—is not supported.

•Digital T1 voice is not manageable through Simple Network Management Protocol (SNMP) using existing versions of Cisco Voice Manager. Release 2.0 of Cisco Voice Manager is planned to support the feature.

Prerequisites

Digital T1 packet voice requires specific service, software, and hardware as follows:

•Obtain T1 service from your service provider or PBX.

•Install Cisco IOS Release 12.0(5)XK, 12.0(7)T, or a later release. The minimum DRAM memory requirements to support digital T1 packet voice trunk network modules are as follows:

–32 MB with one or two T1 lines

–48 MB with three or four T1 lines

–64 MB with five to ten T1 lines

–128 MB with more than ten T1 lines

The memory required for high-volume applications may be greater than listed.

Support for digital T1 packet voice trunk network modules is included in Plus feature sets. The IP Plus feature set requires 8 MB of Flash memory; other Plus feature sets require 16 MB.

•Install one of the following high-density T1 network modules in the router chassis:

–Single-Port 24 Channel T1 High-Density Voice Network Module (NM-HDV-1T1-24)

–Single-Port Enhanced 24 Channel T1 High-Density Voice Network Module (NM-HDV-1T1-24E)

–Dual-Port 48 Channel High-Density Voice Network Module (NM-HDV-2T1-48)

Note You can install one module in a Cisco 2600 series router or a Cisco 3620 router. A Cisco 3640 router can support three modules, and you can install as many as six modules in a Cisco 3660 router.

•Install at least one packet voice data module (PVDM-12) in the high-density digital T1 network module if it is not already equipped. The digital T1 packet voice trunk network module contains five 72-pin SIMM sockets or banks, numbered 0 through 4, for PVDMs. Each socket can be filled with a single 72-pin PVDM. A digital T1 packet voice trunk network module can support the following numbers of channels:

–When the digital T1 packet voice trunk network module is configured for high-complexity codec mode, up to six voice or fax calls can be completed per PVDM-12, using the following codecs: G.711, G.726, G.729, G.729 Annex B, G.723.1, G.723.1 Annex A, G.728, and fax relay.

–When the digital T1 packet voice trunk network module is configured for medium-complexity codec mode, up to 12 voice or fax calls can be completed per PVDM-12, using the following codecs: G.711, G.726, G.729 Annex A, G.729 Annex B with Annex A, and fax relay.

Note Each PVDM holds three DSPs. With five PVDM slots populated, a total of 15 DSPs are provided. High-complexity codecs support two simultaneous calls on each DSP, while medium-complexity codecs support four calls on each DSP.

•Install and configure at least one dual-mode VWIC for a voice connection if a VWIC was not included with the network module. You can install one VWIC (providing one or two line interfaces) in the digital T1 packet voice trunk network module. Only the one- and two-port T1 multiflex trunk interface cards (VWIC-1MFT-T1, VWIC-2MFT-T1, and VWIC-2MFT-T1-DI) are supported with CAS.

For drop-and-insert capability, you must install a two-port drop-and-insert T1 multiflex trunk VWIC (VWIC-2MFT-T1-DI). To install a VWIC in a network module, refer toCisco WAN Interface Cards Hardware Installation Guide.

•Install and configure at least one other network module or WIC to provide the connection to the IP LAN or WAN.

Configuring Voice Card and T1 Controller Settings

To specify codec settings for voice cards and set up T1 controllers for clocking and other T1 parameters, and for DS0 groups that define the channels for compressed voice and TDM groups for drop-and-insert capability, use the following commands beginning in privileged EXEC mode:
Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# voice-card slot

Enters voice card interface configuration mode and specifies the slot location by using a value from 0 to 5, depending upon your router.

Step 3

Router(config-voice-ca)# codec complexity {high | medium}

Specifies the codec complexity based on the codec standard you are using. High-complexity codecs support lower call density than do medium-complexity codecs. The number of channels supported is based on the number of PVDMs installed and the codec complexity. Here are guidelines:

•When the digital T1 packet voice trunk network module is configured for high-complexity codec mode, up to six voice or fax calls can be completed per PVDM-12, using the following codecs: G.711, G.726, G.729, G.729 Annex B, G.723.1, G.723.1 Annex A, G.728, and fax relay.

•When the digital T1 packet voice trunk network module is configured for medium-complexity codec mode, up to 12 voice or fax calls can be completed per PVDM-12, using the following codecs: G.711, G.726, G.729 Annex A, G.729 Annex B with Annex A, and fax relay.

All voice cards in a router must use the same codec complexity setting.

The keyword that you specify for codec complexity command affects the choice of codecs available using the codec dial-peer configuration command. For more information about applying codecs to dial peers, see the "Configuring Dial Peers" section later in this chapter.

Note You cannot change codec complexity while DS0 groups are defined. If they are already set up, use the no ds0-group command before resetting the codec complexity.

Step 4

Router(config)# controller T1 slot/port

Enters controller configuration mode for the T1 controller at the specified slot/port location. Valid values for the slot and port arguments are 0 and 1.

Step 5

Router(config-controller)# clock source {line [primary] | internal}

Configures controller T1 1/0 to specify the clock source. The line keyword specifies that the clock source is derived from the active line—rather than from the free-running internal clock. This is the default setting and is generally more reliable. These rules apply to clock sourcing on the T1 controller ports:

•When both ports are set to line clocking with no primary specification, port 0 is the default primary clock source and port 1 is the default secondary clock source.

•When both ports are set to line and one port is set as the primary clock source, the other port is by default the backup or secondary source and is loop-timed.

•If one port is set to clock source line or clock source line primary and the other is set to clock source internal, the internal port recovers clock from the clock source line port if the clock source line port is up. If it is down, then the internal port generates its own clock.

•If both ports are set to clock source internal, there is only one clock source—internal.

Step 6

Router(config-controller)# framing {sf | esf}

Sets the framing according to your service provider instructions. Use the sf keyword to select SF format and the esf keyword to select the ESF format.

Step 7

Router(config-controller)# linecode {b8zs | ami}

Sets the line encoding according to the instructions given by your service provider. Use the b8zs keyword to select B8ZS encoding, which encodes a sequence of eight zeros in a unique binary sequence to detect line-coding violations. Use the ami keyword to select AMI encoding, which represents zeros using a 01 during each bit cell, and ones are represented by 11 or 00, alternately, during each bit cell. AMI requires that the sending device maintain ones density. Ones density is not maintained independent of the data stream.
Step 8
Router(config-controller)# cablelength long {gain26 | gain36}{-15db | -22.5db | -7.5db | 0db}
or

cablelength short {133 | 266 | 399 | 533 | 655}
(T1 interfaces only) Sets the cable length. The cable length setting must conform to the actual cable length you are using. For example, if you attempt to enter the cablelength short command on a long-haul T1 link, the command is rejected.

To set a cable length longer than 655 feet for a T1 link, enter the cablelength long command. The keywords are as follows:

•gain26 specifies the decibel pulse gain at 26. This is the default pulse gain.

•gain36 specifies the decibel pulse gain at 36.

•-15db specifies the decibel pulse rate at -15 decibels.

•-22.5db specifies the decibel pulse rate at -22.5 decibels.

•-7.5db specifies the decibel pulse rate at -7.5 decibels.

•0db specifies the decibel pulse rate at 0 decibels. This is the default pulse rate.

To set a cable length 655 feet or less for a T1 link, enter the cablelength short command. There is no default for cablelength short. The keywords are as follows:

•133 specifies a cable length from 0 to 133 feet.

•266 specifies a cable length from 134 to 266 feet.

•399 specifies a cable length from 267 to 399 feet.

•533 specifies a cable length from 400 to 533 feet.

•655 specifies a cable length from 534 to 655 feet.

If you do not set the cable length, the system defaults to a setting of cablelength long gain26 0db.
Step 9

Router(config-controller)# ds0-group ds0-group-no timeslots timeslot-list type {e&m-immediate | e&m-delay | e&m-wink | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-start}

Defines the T1 channels for use by compressed voice calls and the signalling method the router uses to connect to the PBX or CO. You should set up DS0 groups after you have specified codec complexity in voice-card configuration. If you modify the codec complexity command parameters, you must first remove any existing DS0 groups, then reinstate them after the change to the codec complexity.

The ds0-group-no argument is a value from 0 to 23 that identifies the DS0 group.

Note The ds0-group command automatically creates a logical voice port that is numbered as follows: slot/port:ds0-group-no. Although only one voice port is created, applicable calls are routed to any channel in the group.

The timeslot-list argument is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of time slots. For T1, allowable values are from 1 to 24. To map individual DS0 time slots, define additional groups. The system maps additional voice ports for each defined group.

The signalling method selection for the type keyword depends on the connection that you are making:

•The E&M interface allows connection for PBX trunk lines (tie lines) and telephone equipment. The wink and delay settings each specify confirming signals between the sending and receiving ends, whereas the immediate setting stipulates no special off-hook/on-hook signals.

•The FXO interface is for connection of a CO to a standard PBX interface where permitted by local regulations; the interface is often used for OPXs.

•The FXS interface allows connection of basic telephone equipment and PBXs.

Step 10

Router(config-controller)# tdm-group tdm-group-no timeslots timeslot-list type [e&m | fxs [loop-start | ground-start] fxo [loop-start | ground-start]]

(Optional) Defines TDM channel groups for the drop-and-insert (also called TDM Cross-Connect) function with a two-port T1 multiflex trunk interface card.

The tdm-group-no argument specifies a value from 0 to 23 that identifies the channel group.

The timeslot-list argument is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of time slots. For T1, allowable values are from 1 to 24.

The signalling method selection for the type keyword depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line.

Note The group numbers for controller groups must be unique. For example, a TDM group should not have the same ID number as a DS0 group.

Step 11

Router(config-controller)# no shutdown

Activates the controller.

Step 12

Router(config-controller)# exit

Exits controller configuration mode.

Step 13

Router(config)# connect id T1 slot/port tdm-group-no-1 T1 slot/port tdm-group-no-2

(Optional) Sets up the connection between two T1 TDM groups of time slots on the trunk interfaces—for the drop-and-insert capability.

The id argument is a name for the connection.

Identify each T1 controller by its slot/port location. Valid values for the slot and port arguments are 0 and 1.

The tdm-group-no-1 and tdm-group-no-2 arguments identify the TDM group numbers (from 0 to 23) on the specified controller.
Repeat Steps 2 and 3 for each voice card.

Repeat Steps 4 through 12 for each controller.

Configuring Voice Port Parameters

To configure voice port parameters, use the following commands, beginning in global configuration mode:

Command

Purpose

Step 1

Router(config)# voice-port slot/port:ds0-group-no

Enters voice-port configuration mode.

The slot argument is the router location where the voice module is installed. Valid entries are from 0 to 3.

The port argument indicates the VIC location. Valid entries are 0 or 1.

Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1 card.

Step 2

Router(config-voice-port)# busyout monitor interface interface number

(Optional) Specifies a LAN or WAN interface that will be monitored, and, when it is down, trigger a busyout (off-hook) state on the voice port. This allows rerouting of calls. For example, if you specify Serial 1/0 as the interface and number, the voice port sends a busyout signal when that interface is down. You can issue the command repeatedly to specify as many interfaces, virtual interfaces, and subinterfaces as are required for a voice port.

If you issue the command three times so that three interfaces are monitored, the voice port only goes into busyout state when all three interfaces are down. When any one of the interfaces is operational, the busyout state is removed.

Step 3

Router(config-voice-port)# comfort-noise

(Optional) Enables comfort noise. (This parameter is enabled by default.) It creates subtle background noise to fill silent gaps during calls when VAD is enabled on voice dial peers. If comfort noise is not generated, the silence can be unsettling to callers.

Step 4

Router(config-voice-port)# echo-cancel enable

(Optional) Enables echo cancellation. (This setting is enabled by default.) Echo cancellation adds to the quality of voice transmissions by adjusting the echo that occurs on the interface due to impedance mismatches. Some echo is reassuring; echo over 25 milliseconds long can cause problems.

Step 5

Router(config-voice-port)# echo-cancel coverage {16 | 24 |32 | 8}

(Optional) Adjusts the echo canceller by the specified number of milliseconds; the default is 16.

Step 6

Router(config-voice-port)# connection {plar |trunk} string

(Optional) Sets up a connection mode for the voice port.

The plar keyword specifies a private line auto ring down (PLAR) connection, which rings a remote telephone when the dial peer goes off hook.

The trunk keyword specifies a straight tie-line connection to a PBX.

The string argument specifies the remote telephone number or significant start digits of the number.

Step 7

Router(config-voice-port)# timeouts interdigit seconds

(Optional) Sets the number of seconds the system waits—after the caller has input the initial digit—for a subsequent digit of the dialed string. If the timeout ends before the destination is identified, a tone sounds and the call ends. The default value is 10 seconds, and the timeout can be set from 0 to 120 seconds.

Note Changes to the default for this command normally are not required.

Step 8

Router(config-voice-port)# exit

Exits voice-port configuration mode.

Repeat Steps 2 through 8 for each DS0 group you create.

Verifying Digital T1 Packet VTNM Configuration

You can check the validity of your digital T1 packet VTNM configuration by performing the following tasks:

•To verify the voice-port configuration, use the show voice port command.

•To display the current voice-card setting, use the show running-config command.

•To display information about clock sources and other settings for the T1 ports, use the show controllers t1 command.

•To display the status of T1 or E1 TDM controller groups and how they are set up, use the show connection all command.

Configuring 1- and 2-Port T1/E1 Multiflex VWICs on Cisco 2600 and 3600 Series Routers

Cisco T1/E1 Multiflex VWICs support voice and data applications in Cisco 2600 and 3600 series routers. The VWICs offer WIC and VIC functionality in a variety of applications for enterprises and for service providers that supply CPE.

Figure 18 shows how T1/E1 Multiflex VWIC are used where VWIC ports are assigned to a PBX and a CO in an network environment where there is no WAN connectivity.

Figure 18 T1/E1 Multiflex VWIC Applications, VWIC Ports Assigned to PBX and CO (No WAN Connectivity)

Multiflex VWICs support the following applications:

•Data. As WICs for T1/E1 applications, including fractional use, the T1 version integrates a fully managed data service unit/channel service unit (DSU/CSU), and the E1 version includes a fully managed DSU.

•Packet voice. As VICs included with the digital T1 packet voice trunk network module to provide T1 connections to PBXs and COs, the T1 VWICs enable packet VoIP applications.

•Multiplexed voice and data. 2-port T1/E1 VWICs can provide drop-and-insert multiplexing services with integrated DSU/CSUs. For example, when used with a digital T1 packet voice trunk network module, drop-and-insert allows you to take 64-Kb DS0 channels from one T1 and digitally cross-connect them to 64-Kb DS0 channels on another T1. Drop-and-insert, sometimes called TDM cross-connect, uses circuit switching and does not use the DSPs that VoIP technology employs.

The following multiflex VWICs are available:

•1-port T1 Multiflex Trunk Interface (VWIC-1MFT-T1)

•1-port E1 Multiflex Trunk Interface (VWIC-1MFT-E1)

•2-port T1 Multiflex Trunk Interface (VWIC-2MFT-T1)

•2-port E1 Multiflex Trunk Interface (VWIC-2MFT-E1)

•2-port T1 Multiflex Trunk Interface with drop-and-insert (VWIC-2MFT-T1-DI)

•2-port E1 Multiflex Trunk Interface with drop-and-insert (VWIC-2MFT-E1-DI)

Multiflex VWIC features include the following:

•Drop-and-Insert capabilities that allow individual 64-Kb DS0 channels to be transparently passed, uncompressed, between two ports on the same multiflex VWIC without passing through a DSP. For example:

–By using this method, you can send the channel traffic between a PBX and CO or other telephony device.

–Drop-and-insert can cross-connect a telephony switch (from the CO or PSTN) to a channel bank to provide external analog connectivity.

Note T1/E1 channels can be used either for drop-and-insert or VoIP, but not both.

•Physical-layer alarm forwarding feature between the ports on 2-port cards

•T1/E1 or fractional T1/E1 network interfaces

•Per-channel T1/E1 data rates of 64 or 56 kbps for WAN services (Frame Relay or leased line)

Restrictions

The following restrictions apply to T1/E1 multiflex VWIC configurations:

•On Cisco 3660 platforms, multiflex VWICs are supported only when they are installed in a digital T1 packet voice trunk network module.

•On all Cisco 2600 and 3600 platforms, digital T1 packet voice trunk network modules only support T1 multiflex VWICs.

•E1 VWICs are not supported on Cisco 3660 platforms.

•Cisco 3620 and 3640 combination network modules allow the installation of either a 1-port VWIC or a 2-port drop-and-insert VWIC.

•Drop-and-insert capability is supported only between two ports on the same multiflex card.

•When installed in a Cisco 2600 chassis slot, DSP resources for packet voice are not available to the multiflex VWICs with drop-and-insert.

Prerequisites

T1/E1 multiflex VWICs require the following specific service, software, and hardware:

•Obtain T1 or E1 service from your service provider.

•Install Cisco IOS Release 12.0(5)XK, 12.0(7)T, or a later release.

•If you are installing multiflex VWICs in a digital T1 packet voice trunk network module, refer to the "Configuring Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers" section earlier in this chapter.

Note You can install one digital T1 packet voice trunk network module in a Cisco 2600 series router or a Cisco 3620 router. A Cisco 3640 router can support three modules, and you can install as many as six modules in a Cisco 3660 router.

•Install the T1 or E1 multiflex VWIC by following the instructions in Cisco 2600 and 3600 Series WAN Interface Cards Hardware Installation Guide.

•If you are using drop-and-insert with a digital T1 packet voice trunk network module, install at least one other network module or WIC to provide the connection to the IP LAN or WAN.

Configuring Voice Cards and DS0s

If you are configuring T1 multiflex VWICs installed in digital T1 packet voice trunk network modules for voice, use the following commands beginning in privileged EXEC mode:

Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# voice-card slot

Enters voice card interface configuration mode. The slot argument specifies the card location by using a value from 0 to 5, depending upon your router.

Step 3

Router(config-voice-ca)# codec complexity {high | medium}

Specifies the codec complexity based on the codec standard you are using. High-complexity codecs support lower call density than do medium-complexity codecs. The number of channels supported is based on the number of PVDMs installed and the codec complexity. Here are guidelines:

•In high-complexity codec mode, up to six voice or fax calls can be completed per PVDM-12, using the following codecs: G.711, G.726, G.729, G.729 Annex B, G.723.1, G.723.1 Annex A, G.728, and fax relay.

•In medium-complexity codec mode, up to 12 voice or fax calls can be completed per PVDM-12, using the following codecs: G.711, G.726, G.729 Annex A, G.729 Annex B with Annex A, and fax relay.

All voice cards in a router must use the same codec complexity setting.

The keyword that you specify for codec complexity affects the choice of codecs available using the codec dial-peer configuration command. For more information about applying codecs to dial peers, see the "Configuring Dial Peers" section later in this chapter.

You cannot change codec complexity while DS0 groups are defined. If they are already set up, use the no ds0-group command before resetting the codec complexity.

Step 4

Router(config)# controller T1 slot/port

Enters controller configuration mode for the VWIC. Valid values for the slot argument are 0 through 5 and for the port argument are 0 and 1.

Step 5

Router(config-controller)# ds0-group ds0-group-no timeslots timeslot-list type {e&m-immediate | e&m-delay |e&m-wink | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-start}

(Voice only) Defines the T1 channels for use by compressed voice calls and the signalling method the router uses to connect to the PBX or CO. Set up DS0 groups after you have specified codec complexity in voice-card interface configuration mode. If you modify the codec complexity command parameters, you must first remove any existing DS0 groups, then reinstate them after the change to the codec complexity.

The ds0-group-no argument is a value from 0 to 23 that identifies the DS0 group.

Note The ds0-group command automatically creates a logical voice port that is numbered as follows: slot/port:ds0-group-no. Although only one voice port is created, applicable calls are routed to any channel in the group.

The timeslot-list argument is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen to indicate a range of time slots. For T1, allowable values are from 1 to 24. To map individual DS0 time slots, define additional groups. The system maps additional voice ports for each defined group.

The signalling method selection for the type keyword depends on the connection that you are making:

•The E&M interface allows connection for PBX trunk lines (tie lines) and telephone equipment. The wink and delay settings each specify confirming signals between the sending and receiving ends, whereas the immediate setting stipulates no special off-hook/on-hook signals.

•The FXO interface is for connection of a CO to a standard PBX interface where permitted by local regulations; the interface is often used for OPXs.

•The FXS interface allows connection of basic telephone equipment and PBXs.

Configuring E1 or T1 Controllers

To configure T1 and E1 controllers, use the following commands beginning in global configuration mode:
Command

Purpose
Step 1
Router(config)# controller {T1 | E1} slot/port
Enters controller configuration mode for the T1 or E1 controller at the specified slot/port location.
Step 2

Router(config-controller)# loopback {diagnostic | local {payload | line}| remote {iboc | esf {payload | line}}

(Optional) Generates a local loopback test at the line or payload level, or a remote loopback.

Step 3

Router(config-controller)# clock source {line [primary] | internal}

Specifies the clock source. The line keyword specifies that the clock source is derived from the active line—rather than from the free-running internal clock. This is the default setting and is generally more reliable. These rules apply to clock sourcing:

•When both ports are set to line clocking with no primary specification, port 0 is the default primary clock source and port 1 is the default secondary clock source.

•When both ports are set to line and one port is set as the primary clock source, the other port is by default the backup or secondary source and is loop-timed.

•If one port is set to clock source line or clock source line primary and the other is set to clock source internal, the internal port recovers clock from the clock source line port if the clock source line port is up. If it is down, then the internal port generates its own clock.

•If both ports are set to clock source internal, there is only one clock source—internal.

Step 4

Router(config-controller)# framing {sf | esf}
or

Router(config-controller)# framing {crc4 | no-crc4} [australia]

Sets the framing to SF or ESF format, according to service provider requirements.

Sets the framing to cyclic redundancy check 4 (CRC4) or no CRC4, according to service provider requirements. The australia optional keyword specifies Australian Layer 1 Homologation for E1 framing.

Step 5

Router(config-controller)# linecode {b8zs | ami | hdb3}

Sets the line encoding according to your service provider's instructions. Use the b8zs keyword to specify B8ZS line encoding. B8ZS, available only for T1 lines, encodes a sequence of eight zeros in a unique binary sequence to detect line coding violations.

Use the ami keyword to specify AMI encoding. AMI, available for T1 or E1 lines, represents zeros using a 01 during each bit cell, and ones are represented by 11 or 00, alternately, during each bit cell. AMI requires that the sending device maintain ones density. Ones density is not maintained independent of the data stream.

For E1, set the line-coding to either AMI or high-density bipolar 3 (HDB3), which is the default.

Step 6

Router(config-controller)# line-termination {75-ohm | 120-ohm}

(E1 only) Enters a line-termination value. This command specifies the impedance (amount of wire resistance and reactivity to current) for the E1 termination. Impedance levels are maintained to avoid data corruption over long-distance links.

Specify the 120-ohm keyword to match the balanced 120-ohm interface. This is the default.

Specify the 75-ohm keyword for an unbalanced BNC 75-ohm interface.

Step 7

Router(config-if)# fdl {att | ansi | both}

(T1 interfaces only) Sets the Facility Data Link (FDL) exchange standard for the CSU controllers. The FDL is a 4-kbps channel used with the ESF framing format to provide out-of-band messaging for error-checking on a T1 link.

You typically leave this setting at the default, ansi, which follows the American National Standards Institute (ANSI) T1.403 standard for extended superframe facilities data-link exchange support. Changing it allows improved management in some cases but can cause problems if your setting is not compatible with that of your service provider.

Use the att keyword to select the AT&T TR54016 standard for ESF facilities data-link exchange support.

Use the both keyword to enable both of the described standards.
Step 8
Router(config-controller)# cablelength long {gain26 | gain36} {-15db | -22.5db | -7.5db | 0db}
or

cablelength short {133 | 266 | 399 | 533 | 655}
(T1 interfaces only) Sets the cable length. The cable length setting must conform to the actual cable length you are using. For example, if you attempt to enter the cablelength short command on a long-haul T1 link, the command is rejected.

To set a cable length longer than 655 feet for a T1 link, enter the cablelength long command. The keywords are as follows:

•gain26 specifies the decibel pulse gain at 26. This is the default pulse gain.

•gain36 specifies the decibel pulse gain at 36.

•-15db specifies the decibel pulse rate at -15.

•-22.5db specifies the decibel pulse rate at -22.5.

•-7.5db specifies the decibel pulse rate at -7.5.

•0db specifies the decibel pulse rate at 0. This is the default pulse rate.

To set a cable length 655 feet or less for a T1 link, enter the cablelength short command. There is no default for cablelength short. The keywords are as follows

•133 specifies a cable length from 0 to 133 feet.

•266 specifies a cable length from 134 to 266 feet.

•399 specifies a cable length from 267 to 399 feet.

•533 specifies a cable length from 400 to 533 feet.

•655 specifies a cable length from 534 to 655 feet.

If you do not set the cable length, the system defaults to a setting of cablelength long gain26 0db.
Repeat the steps following Step 4 for each controller.

Configuring Drop-and-Insert

(Optional) To set up drop-and-insert, use the following commands beginning in controller configuration mode:

Command

Purpose

Step 1

Router(config-controller)# tdm-group tdm-group-no timeslots timeslot-list type [e&m | fxs [loop-start | ground-start] fxo [loop-start | ground-start]

Sets up TDM channel groups for the drop-and-insert function with a 2-port multiflex VWIC.

The tdm-group-no argument identifies the TDM group and is a value from 0 to 23 for T1 and from 0 to 30 for E1.

The timeslot-list argument indicates a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. The valid range is from 1 to 24 for T1. For E1, the range is from 1 to 31.

The signalling method selection for the type keyword depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line.

Note The group numbers for controller groups must be unique. For example, a TDM group should not have the same ID number as a DS0 group or channel group.

Step 2

Router(config-controller)# channel-group channel-group-no timeslots timeslot-list [speed [48 | 56 | 64 ]]

(Optional) Sets up channel groups for WAN data services with a 2-port multiflex drop-and-insert VWIC.

The channel-group-no argument identified the channel group and is a value from 0 to 23 for T1 and from 0 to 30 for E1; because there can be only one channel group on a 1- or 2-port multiflex VWIC, 0 is always the value.

The timeslot-list argument indicates a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. The valid range is from 1 to 24 for T1. For E1, the range is from 1 to 31.

The optional speed keyword defaults to 56 kbps for T1 and 64 kbps for E1.

Note Although the CLI displays 48 as a speed option, it is not supported.

Step 3

Router(config-controller)# no shutdown

Activates the controller.

Step 4

Router(config-controller)# exit

Exits controller configuration mode.

Step 5

Router(config)# connect id {T1 | E1} slot/port-1 tdm-group-no-1 {T1 | E1} slot/port-2 tdm-group-no-2

Sets up the connection between two T1 or E1 TDM groups of time slots on the WVIC—for drop-and-insert.

Use the id argument to define a name for the connection.

Use the slot/port argument to identify each controller by its location.

Use the tdm-group-no-1 and tdm-group-no-2 arguments to identify the TDM group numbers (from 0 to 23 or 30) on the specified controller.

Configuring Voice Ports Parameters

To configure voice port parameters to support local and remote stations, use the following commands beginning in global configuration mode:

Command

Purpose

Step 1

Router(config)# voice-port slot/port:ds0-group-no

Enters voice-port configuration mode.

The slot argument identifies the router location where the voice module is installed. Valid entries are from 0 to 3.

The port argument indicates the multiflex VWIC location. Valid entries are 0 or 1.

Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1 card.

Step 2

Router(config-voice-port)# busyout monitor interface interface number

(Optional) Specifies a LAN or WAN interface that will be monitored, and, when it is down, trigger a busyout (off-hook) state on the voice port. This allows rerouting of calls. For example, if you specify Serial 1/0 as the interface and number, the voice port sends a busyout signal when that interface is down. You can issue the command repeatedly to specify as many interfaces, virtual interfaces, and subinterfaces as are required for a voice port.

For example, if you issue the command three times so that three interfaces are monitored, the voice port only goes into busyout state when all three interfaces are down. When any one of the interfaces is operational, the busyout state is removed.

Step 3

Router(config-voice-port)# comfort-noise

(Optional) Enables comfort noise. (This parameter is enabled by default.) It creates subtle background noise to fill silent gaps during calls when VAD is enabled on voice dial peers. If comfort noise is not generated, the silence can be unsettling to callers.

Step 4

Router(config-voice-port)# echo-cancel enable

(Optional) Enables echo cancellation. (This setting is enabled by default.) Echo cancellation adds to the quality of voice transmissions by adjusting the echo that occurs on the interface due to impedance mismatches. Some echo is reassuring; echo over 25 milliseconds long can cause problems.

Step 5

Router(config-voice-port)# echo-cancel coverage {16 | 24 |32 | 8}

(Optional) Adjusts the echo canceller by the specified number of milliseconds; the default is 16.

Step 6

Router(config-voice-port)# connection {plar | trunk} string

(Optional) Sets up a connection mode for the voice port.

The plar keyword specifies a PLAR connection, which rings a remote telephone when the dial peer goes off hook.

The trunk keyword specifies a straight tie-line connection to a PBX.

The string argument specifies the remote telephone number or significant start digits of the number.

Repeat Steps 1 through 8 for each DS0 group you create.

Verifying T1/E1 Multiflex Voice/WAN Interface Configuration

You can check the validity of your digital T1/EI multiflex interface configuration by performing the following tasks:

•To display the current voice-card setting, use the show running-config command. If no codec complexity is shown, the default of medium complexity is set.

•To display the status of T1 or E1 controllers and display information about clock sources and other settings for the ports, use the show controllers t1/e1 command.

•To display the status of T1 or E1 TDM controller groups and how they are set up, use the show connection all command.

•To verify the voice-port configuration, use the show voice port.

Configuring ISDN BRI VoIP for Cisco 2600 and 3600 Series VICs

VoIP enables the Cisco 2600 and Cisco 3600 series of modular routers to carry voice traffic simultaneously with data traffic over an IP network. VoIP is primarily a software feature, supporting both voice and fax calls. Support for the ISDN BRI signalling type allows a Cisco 2600 or Cisco 3600 series router to provide voice access connectivity to either an ISDN telephone network or a digital interface on PBX and key communications system. The voice or data also crosses an IP network to which the router connects, allowing branch offices and enterprises to route incoming PSTN ISDN BRI calls over an IP network or send outgoing digital fax and voice calls via an IP network.

ISDN BRI VoIP offers direct ISDN network connectivity and connectivity to the digital interfaces of PBX and key communications systems. Prior to the introduction of this feature, VoIP was available only for FXS connection to a POTS telephone or other telephony equipment, FXO for connection to a POTS PBX or key system, or E&M for 2-wire and 4-wire telephone and trunk interfaces—typically used to connect remote calls from an IP network to a PBX.

ISDN BRI VoIP provides the following toll-saving benefits for enterprises and branch offices:

•ISDN BRI network connectivity, particularly critical in areas where this is the standard provider offering

•Use of digital terminal equipment such as digital telephones and fax machines

•Off-premises ISDN BRI dialing in to an IP network

Figure 19 shows a home-office user dialing directly in to a local router via the PSTN, and reaching headquarters through an IP network, saving the cost of a long-distance call. In another example, Figure 19 shows how an extension at headquarters makes a fax or voice call to a branch office in a different area code using a corporate IP network only.

Figure 19 Applications for ISDN BRI Voice over IP

Prerequisites

Before you can configure your Cisco 2600 or Cisco 3600 series router for VoIP on a BRI interface, you must perform the following tasks:

•Obtain BRI service from your telecommunications provider. The BRI line must be provisioned at the switch to support voice calls.

•Install and configure at least one network module or WIC to provide the connection to the IP LAN or WAN.

•Install a 2-slot VNM (NM-2V) into the appropriate slot of your Cisco router. A 1-slot VNM (NM-1V) does not provide use of all four BRI VIC slots. At least one other network module or WIC must be installed in the router to provide the connection to the IP LAN or WAN. For information on installation or the physical characteristics of your VNM, refer to the installation document, Voice Network Module and Voice Interface Card Configuration Note, that came with your VNM.

•Install a 2-port BRI VIC (VIC-2BRI-S/T-TE) into Slot 0, the first slot of the VNM. Slot 1 of the VNM should remain empty. Each of the two ports of a BRI VIC can carry two voice calls, one over each ISDN B channel, for a total of four calls per BRI VIC.

Configuring BRI Interfaces

To configure BRI interfaces, use the following commands beginning in privileged EXEC mode:
Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.
Step 2
Router(config)# isdn switch-type switch-type
Configures the global ISDN switch type to match the service provider switch type. For a list of keywords, see Table 8.
Step 3

Router(config)# interface bri slot/port

Enters interface configuration mode to configure parameters for the specified interface.

The slot argument specifies the location of the VNM in the router.

The port argument specifies the location of the BRI VIC in the VNM. Valid values are 0 or 1.

Step 4

Router(config-if)# no ip address

Specifies that there is no IP address for this interface.

Step 5

Router(config-if)# no ip-directed broadcast

Disables the translation of directed broadcast to physical broadcasts.

Step 6

Router(config-if)# isdn switch-type switch-type

(Optional) Configures the interface ISDN switch type to match the service provider switch type. The interface ISDN switch type overrides the global ISDN switch type on the interface.

For a list of switch type keywords, see Table 8.

Step 7

Router(config-if)# isdn spid1 spid-number [ldn]

Specifies a SPID and local directory number for the B1 channel. Currently, only the DMS-100 and NI-1 switch types require SPIDs. Although the Lucent 5ESS switch type might support a SPID, we recommend that you set up that ISDN service without SPIDs.

Step 8

Router(config-if)# isdn spid2 spid-number [ldn]

Specifies a SPID and local directory number for the B2 channel.

Step 9

Router(config-if)# isdn twait-disable

(Optional) Delays a National ISDN BRI switch a random time before activating the Layer 2 interface when the switch starts up. Use this command when the ISDN switch type is basic-nil.

Step 10

Router(config-if)# isdn incoming-voice modem

Configures the port for incoming voice calls.
Table 8 lists the available switch type keywords.

Table 8 ISDN Switch Types

Country

ISDN Switch Type

Description

Australia

basic-ts013

Australian TS013 switches

Europe

basic-1tr6

German 1TR6 ISDN switches

basic-nwnet3

Norwegian NET3 ISDN switches (phase 1)

basic-net3

NET3 ISDN switches (United Kingdom and others)

vn2

French VN2 ISDN switches

vn3

French VN3 ISDN switches

Japan

ntt

Japanese NTT ISDN switches

New Zealand

basic-nznet3

New Zealand NET3 switches

North America

basic-5ess

Lucent Technologies basic rate switches

basic-dms100

NT DMS-100 basic rate switches

basic-ni1

National ISDN-1 switches

Verify ISDN BRI Configuration

You can check the validity of your ISDN BRI configuration by performing the following tasks:

•To show the current configuration running on the terminal, use the show running-config command.

•To display information about the physical attributes of the ISDN BRI B and D channels, use the show interfaces bri command.

Configuring T1/E1 High-Capacity Digital Voice Port Adapters for Cisco 7200 Series Routers

T1/E1 high-capacity digital voice port adapters for Cisco 7200 series routers allow enterprises or service providers, using the equipped routers as CPE, to deploy digital voice and fax relay. These port adapters receive constant bit-rate telephony information over T1 interfaces and can convert that information to a compressed format and be sent as VoIP.

T1/E1 digital voice over IP includes the following functionality:

•T1 CAS for the following line-signalling types:

–E&M immediate start

–E&M wink start

–E&M delay start (also called "dial repeating")

–FXS and FXO loop start

–FXS and FXO ground start

•Dynamic bandwidth allocation using VAD

•Drop-and-insert capability, allowing the interchange of TDM slots between the ports on a 2-port digital T1/E1 voice port adapter

•Support for a wide range of ITU-T G-series compression specifications, including:

–G.711 a-law at 64,000 bps

–G.711 u-law at 64,000 bps

–G.723

–G.726 at 16,000 bps

–G.726 at 24,000 bps

–G.726 at 32,000 bps

–G.728 at 16,000 bps

–G.729 at 8000 bps

–G.729 Annex A at 8000 bps

–G.729 Annex B at 8000 bps

–G.729 Annex B with Annex A at 8000 bps

•48 channels of compressed voice

•High-quality voice endpoint-standard features, such as high-quality echo cancellation, silence suppression, comfort noise generation, and DTMF relay

•Group 3 fax relay

•Support for the following T1 framing formats and line coding:

–SF

–ESF

–AMI line coding

–B8ZS line coding

•Support for the following E1 framing formats and line coding:

–CRC4

–No CRC4

–Line-code type (HDB3)

Restrictions

The following restrictions apply to digital T1/E1 voice port adapter configuration:

•Group 4 fax is not supported.

•Wink-start signalling Feature Group D is not supported.

•CCS is not supported.

•Voice over ATM—including AAL5 encapsulation, CES, and AAL2—is not supported.

•Digital T1/E1 voice is not manageable through SNMP using existing versions of Cisco Voice Manager. Release 2.0 of Cisco Voice Manager is planned to support the feature.

Prerequisites

Digital T1/E1 voice requires specific service, software, and hardware as follows:

•Obtain T1/E1 service from your service provider or PBX.

•Install Cisco IOS Release 12.0(5)XE, 12.0(6)T, or a later release. The minimum DRAM memory requirements to support T1/E1 high capacity digital voice port adapters is 64 MB.

The memory required for high-volume applications may be greater than listed.

Support for T1/E1 high-capacity digital voice port adapters is included in Plus feature sets. The IP Plus feature set requires 16 MB of Flash memory.

•Install the following high-density T1 or E1 port adapter in the router chassis: Single-Port 30 Channel T1/E1 High-Density Voice Port Adapter (PA-VXC-2TE1).

•Install at least one other LAN/WAN port adapter to provide the connection to the IP LAN or WAN.

Configuring the DSPfarm Interface

To configure a DSPfarm interface, use the following commands beginning in global configuration mode:

Command

Purpose

Step 1

Router(config)# dspinterface dspfarm slot/port

Opens DSPfarm interface configuration mode to configure the DSP interface.

Step 2

Router(config-dspfarm)# codec {high | medium | low} 1-30

Specifies the codec complexity based on the codec standard you are using. High-complexity codecs support lower call density than do medium-complexity codecs. For example:

•When the digital T1/E1 voice port adapter is configured for high-complexity codec mode, each DSP can support up to two calls using the following codecs: G.711, G.726, G.729, G.729 Annex B, G.723.1, G.723.1 Annex A, G.728, and fax relay.

•When the digital T1/E1 voice port adapter is configured for medium-complexity codec mode, each DSP can support up to six calls using the following codecs: G.711, G.726, G.729 Annex A, G.729 Annex B with Annex A, and fax relay

The keyword that you specify for codec affects the choice of codecs available using the codec dial-peer configuration command.

Note You cannot change codec complexity while DS0 groups are defined. If they are already set up, use the no ds0-group command before resetting the codec complexity.

Step 3

Router(config-dspfarm)# no shutdown

Enables the interface.

Configuring Card Type and T1 Controller Settings

To specify codec settings for card types and set up T1 controllers for clocking and other T1 parameters, and for DS0 groups that define the channels for compressed voice and TDM groups for drop-and-insert capability, use the following commands beginning in privileged EXEC mode:
Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# card type {t1/e1} slot

Enters T1 card type and specifies the slot location. Valid entries for the slot argument are 0 to 5, depending upon your router.

Step 3

Router(config)# controller T1 slot/port

Enters controller configuration mode for the T1 controller at the specified slot/port location. Valid values for the slot and port arguments are 0 and 1.

Step 4

Router(config-controller)# clock source {line [primary] | internal}

Configures controller T1 1/0 to specify the clock source. The line keyword specifies that the clock source is derived from the active line—rather than from the free-running internal clock. This is the default setting and is generally more reliable. These rules apply to clock sourcing on the T1 controller ports:

•When both ports are set to line clocking with no primary specification, port 0 is the default primary clock source and port 1 is the default secondary clock source.

•When both ports are set to line and one port is set as the primary clock source, the other port is by default the backup or secondary source and is loop-timed.

•If one port is set to clock source line or clock source line primary and the other is set to clock source internal, the internal port recovers clock from the clock source line port if the clock source line port is up. If it is down, then the internal port generates its own clock.

•If both ports are set to clock source internal, there is only one clock source—internal.

Step 5

Router(config-controller)# framing {sf | esf}

Sets the framing according to the instructions from your service provider. Use the esf keyword to select the ESF framing format or the sf keyword for the SF framing format.

Step 6

Router(config-controller)# linecode {b8zs | ami}

Sets the line encoding according to the instructions from your service provider. Use the b8zs keyword to specify B8ZS line encoding, which encodes a sequence of eight zeros in a unique binary sequence to detect line-coding violations. Use the ami keyword to specify AMI line encoding, which represents zeros using a 01 during each bit cell, and ones are represented by 11 or 00, alternately, during each bit cell. AMI requires that the sending device maintain ones density. Ones density is not maintained independent of the data stream.
Step 7
Router(config-controller)# cablelength long {-15db | -22.5db | -7.5db | 0db}
or

cablelength short {110ft | 220ft | 330ft | 440ft | 550ft | 600ft}
(T1/E1 interfaces only) Configures the cable length. The cable length setting must conform to the actual cable length you are using. For example, if you attempt to enter the cablelength short command on a long-haul T1 link, the command is rejected.

To set a cable length longer than 600 feet for a T1 link, use the cablelength long command. The keywords are as follows:

•-15db specifies the decibel pulse level at -15 dB.

•-22.5db specifies the decibel pulse level at -22.5 dB.

•-7.5db specifies the decibel pulse level at -7.5 dB.

•0db specifies the decibel pulse level at 0 dB. This is the default pulse rate.

To set a cable length 600 feet or less for a T1 link, use the cablelength short command. There is no default for cablelength short. The keywords are as follows:

•110ft specifies a cable length from 0 to 110 feet.

•220ft specifies a cable length from 111 to 220 feet.

•330ft specifies a cable length from 221 to 330 feet.

•440ft specifies a cable length from 331 to 440 feet.

•550ft specifies a cable length from 441 to 550 feet.

•600ft specifies a cable length from 551 to 600 feet.

If you do not set the cable length, the system defaults to a setting of cablelength long 0db.
Step 8

Router(config-controller)# ds0-group ds0-group-no timeslots timeslot-list type {e&m-immediate | e&m-delay | e&m-wink | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-start}

Defines the T1 channels for use by compressed voice calls and the signalling method the router uses to connect to the PBX or CO. You should set up DS0 groups after you have specified codec complexity in voice-card configuration mode. If you modify the codec complexity command parameters, you must first remove any existing DS0 groups, then reinstate them after the change to the codec complexity.

The ds0-group-no argument identifies the DS0 group and is a value from 0 to 23.

Note The ds0-group command automatically creates a logical voice port that is numbered as follows: slot/port:ds0-group-no. Although only one voice port is created, applicable calls are routed to any channel in the group.

The timeslot-list argument indicates a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. For T1 or E1, allowable values are from 1 to 24. To map individual DS0 time slots, define additional groups. The system maps additional voice ports for each defined group.

The signalling method selection for the type keyword depends on the connection that you are making:

•The E&M interface allows connection for PBX trunk lines (tie lines) and telephone equipment. The wink and delay settings each specify confirming signals between the sending and receiving ends, whereas the immediate setting stipulates no special off-hook/on-hook signals.

•The FXO interface is for connection of a CO to a standard PBX interface where permitted by local regulations; the interface is often used for OTXs.

•The FXS interface allows connection of basic telephone equipment and PBXs.

Step 9

Router(config-controller)# tdm-group tdm-group-no timeslots timeslot-list type [e&m | fxs [loop-start | ground-start] fxo [loop-start | ground-start]]

(Optional) Configures TDM channel groups for the drop-and-insert (also called TDM Cross-Connect) function with a 2-port T1 multiflex trunk interface card.

The tdm-group-no argument identifies the channel group and is a value from 1 to 31.

The timeslot-list argument indicates a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. For T1, allowable values are from 1 to 24.

The signalling method selection for the type keyword depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line.

Note The group numbers for controller groups must be unique. For example, a TDM group should not have the same ID number as a DS0 group.

Step 10

Router(config-controller)# no shutdown

Activates the controller.

Step 11

Router(config-controller)# exit

Exits controller configuration mode.

Step 12

Router(config)# connect id T1 slot/port tdm-group-no-1 T1 slot/port tdm-group-no-2

(Optional) Sets up the connection between two T1 TDM groups of time slots on the trunk interfaces—for the drop-and-insert capability.

The id argument specifies the name for the connection.

The slot/port argument identifies each T1/E1 controller by its location. Valid values for slot and port are 0 and 1.

The tdm-group-no-1 and tdm-group-no-2 arguments identify the TDM group numbers (from 1 to 31) on the specified controller.
Repeat Steps 2 and 3 for each card type.

Repeat Steps 4 through 12 for each controller.

Configuring Card Type and E1 Controller Settings

To specify codec settings for card types and set up E1 controllers for clocking and other E1 parameters, as well as for DS0 groups that define the channels for compressed voice and TDM groups for drop-and-insert capability, use the following commands beginning in privileged EXEC mode:

Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# card type {t1/e1} slot

Enters E1 card type and specifies the slot location by using a value from 0 to 5, depending upon your router.

Step 3

Router(config-voice-ca)# codec {high | medium | low} 1-30

Specifies the codec complexity based on the codec standard you are using. High-complexity codecs support lower call density than do medium-complexity codecs.

For example:

•When the digital E1 voice port adapter is configured for high-complexity codec mode, each DSP can support up to two calls using the following codecs: G.711, G.726, G.729, G.729 Annex B, G.723.1, G.723.1 Annex A, G.728, and fax relay.

•When the digital E1 voice port adapter is configured for medium-complexity codec mode, each DSP can support up to six calls using the following codecs: G.711, G.726, G.729 Annex A, G.729 Annex B with Annex A, and fax relay

The keyword that you specify for codec affects the choice of codecs available using the codec dial-peer configuration command.

Note You cannot change codec complexity while DS0 groups are defined. If they are already set up, use the no ds0-group command before resetting the codec complexity.

Step 4

Router(config)# controller E1 slot/port

Enters controller configuration mode for the E1 controller at the specified slot/port location. Valid values for the slot and port arguments are 0 and 1.

Step 5

Router(config-controller)# clock source {line [primary] | internal}

Configures controller E1 1/0 to specify the clock source. The line keyword specifies that the clock source is derived from the active line—rather than from the free-running internal clock. This is the default setting and is generally more reliable. These rules apply to clock sourcing on the E1 controller ports:

•When both ports are set to line clocking with no primary specification, port 0 is the default primary clock source and port 1 is the default secondary clock source.

•When both ports are set to line and one port is set as the primary clock source, the other port is by default the backup or secondary source and is loop-timed.

•If one port is set to clock source line or clock source line primary and the other is set to clock source internal, the internal port recovers clock from the clock source line port if the clock source line port is up. If it is down, then the internal port generates its own clock.

•If both ports are set to clock source internal, there is only one clock source—internal.

Step 6

Router(config-controller)# framing {crc4 | no crc4}

Sets the framing according to the instructions from your service provider. Choose CRC4 format or No CRC4 format.

Step 7

Router(config-controller)# linecode {hdb3}

Sets the line encoding according to the instructions from your service provider.

Step 8

Router(config-controller)# ds0-group ds0-group-no timeslots timeslot-list type {e&m-immediate | e&m-delay | e&m-wink | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-start}

Defines the E1 channels for use by compressed voice calls and the signalling method the router uses to connect to the PBX or CO. You should set up DS0 groups after you have specified codec complexity in voice-card configuration. If you modify the codec complexity command parameters, you must first remove any existing DS0 groups, then reinstate them after the change to the codec complexity.

The ds0-group-no argument identifies the DS0 group and is a value from 0 to 23.

Note The ds0-group command automatically creates a logical voice port that is numbered as follows: slot/port:ds0-group-no. Although only one voice port is created, applicable calls are routed to any channel in the group.

The timeslot-list argument indicate a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. For T1 or E1, allowable values are from 1 to 24. To map individual DS0 time slots, define additional groups. The system maps additional voice ports for each defined group.

The signalling method selection for the type keyword depends on the connection that you are making:

•The E&M interface allows connection for PBX trunk lines (tie lines) and telephone equipment. The wink and delay settings each specify confirming signals between the sending and receiving ends, whereas the immediate setting stipulates no special off-hook/on-hook signals.

•The FXO interface is for connection of a CO to a standard PBX interface where permitted by local regulations; the interface is often used for OPXs.

•The FXS interface allows connection of basic telephone equipment and PBXs.

Step 9

Router(config-controller)# tdm-group tdm-group-no timeslots timeslot-list type [e&m | fxs [loop-start | ground-start] fxo [loop-start | ground-start]]

(Optional) Configures TDM channel groups for the drop-and-insert (also called TDM Cross-Connect) function with a 2-port T1/E1 multiflex trunk interface card.

The tdm-group-no argument identifies the channel group and is a value from 1 to 31.

The timeslot-list argument indicates a range of time slots and is a single number, numbers separated by commas, or a pair of numbers separated by a hyphen. For T1 or E1, allowable values are from 1 to 24.

The signalling method selection for the type keyword depends on the connection that you are making. The fxs and fxo options allow you to specify a ground-start or loop-start line.

Note The group numbers for controller groups must be unique. For example, a TDM group should not have the same ID number as a DS0 group.

Step 10

Router(config-controller)# no shutdown

Activates the controller.

Step 11

Router(config-controller)# exit

Exits controller configuration mode.

Step 12

Router(config)# connect id E1 slot/port tdm-group-no-1 E1 slot/port tdm-group-no-2

(Optional) Sets up the connection between two T1/E1 TDM groups of time slots on the trunk interfaces for the drop-and-insert capability.

The id argument specifies a name for the connection.

The slot/port argument identifies each E1 controller by its location. Valid values for slot and port are 0 and 1.

The tdm-group-no-1 and tdm-group-no-2 arguments identify the TDM group numbers (from 1 to 31) on the specified controller.

Repeat Steps 2 and 3 for each card type.

Repeat Steps 4 through 12 for each controller.

Configuring Voice Ports

To set up voice ports to support the local and remote stations, use the following commands beginning in privileged EXEC mode:

Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# voice-port slot/port:ds0-group-no

Enters voice-port configuration mode.

The slot argument is the router location where the voice port adapter is installed. Valid entries are from 0 to 3.

The port argument indicates the VIC location. Valid entries are 0 or 1.

Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card.

Step 3

Router(config-voice-port)# busyout monitor interface interface number

(Optional) Specifies a LAN or WAN interface that will be monitored, and, when it is down, trigger a busyout (off-hook) state on the voice port. This allows rerouting of calls. For example, if you specify Serial 1/0 as the interface and number, the voice port sends a busyout signal when that interface is down. You can issue the command repeatedly to specify as many interfaces, virtual interfaces, and subinterfaces as are required for a voice port.

For example, if you issue the command three times so that three interfaces are monitored, the voice port only goes into busyout state when all three interfaces are down. When any one of the interfaces is operational, the busyout state is removed.

Step 4

Router(config-voice-port)# comfort-noise

(Optional) Creates subtle background noise to fill silent gaps during calls when VAD is enabled on voice dial peers. (This parameter is enabled by default.) If comfort noise is not generated, the resulting silence can fool the caller into thinking the call is disconnected instead of being merely idle.

Step 5

Router(config-voice-port)# echo-cancel enable

(Optional) Enables echo cancellation. (This setting is enabled by default.) Echo cancellation adds to the quality of voice transmissions by adjusting the echo that occurs on the interface due to impedance mismatches. Some echo is reassuring; echo over 25 milliseconds long can cause problems.

Step 6

Router(config-voice-port)# echo-cancel coverage {16 | 24 |32 | 8}

(Optional) Adjusts the echo canceller by the specified number of milliseconds; the default is 16.

Step 7

Router(config-voice-port)# connection {plar |trunk} string

(Optional) Sets up a connection mode for the voice port.

The plar keyword specifies a PLAR connection, which rings a remote telephone when the dial peer goes off-hook.

The trunk keyword specifies a straight tie-line connection to a PBX.

The string argument specifies the remote telephone number or significant start digits of the number.

Step 8

Router(config-voice-port)# timeouts interdigit seconds

(Optional) Sets the number of seconds the system waits—after the caller has input the initial digit—for a subsequent digit of the dialed string. If the timeout ends before the destination is identified, a tone sounds and the call ends. The default value is 10 seconds, and the timeout can be set from 0 to 120 seconds.

Note Changes to the default for this command normally are not required.

Step 9

Router(config-voice-port)# exit

Exits voice-port configuration mode.

Repeat Steps 2 through 9 for each DS0 group you create

Verifying T1/E1 High-Capacity Digital Voice Port Adapters Configuration

You can check the validity of your T1/E1 high-capacity digital voice port configuration by performing the following tasks:

•To display the current voice-card setting, use the show running-config command. If no codec complexity is shown, the default of medium complexity is set.

•To display the status of T1 or E1 controllers and displays information about clock sources and other settings for the T1/E1 ports, use the show controllers t1 command.

•To verify the voice-port configuration, use the show voice port command.

Configuring ISDN PRI Voice Ports

With ISDN PRI, signalling in VoIP for the Cisco AS5300 and AS5800 is handled by ISDN PRI group configuration. After ISDN PRI has been configured for both B and D channels for both ISDN PRI lines, you need to enter the isdn incoming-voice command on the serial interface (acting as the D channel) to ensure a dial tone.

Under most circumstances, the default voice-port command values are adequate to configure voice ports to transport voice data over your existing IP network. Because of the inherent complexities involved with PBX networks, you might need specific voice-port values configured, depending on the specifications of the devices in your telephony network.

To configure basic ISDN PRI parameters for the Cisco AS5300 or Cisco AS5800 access servers, use the following commands beginning in global configuration mode:

Command

Purpose

Step 1

Router(config)# isdn switch-type switch-type

Defines the telephone company switch type.

Step 2

Router(config)# controller T1 1/0/0
or

Router(config)# controller T1 1/0/0:1

Enables the T1 0 controller on the T1 card and enters controller configuration mode.

or

Enables the T1 1 controller on the T3 card and enters controller configuration mode.

Step 3

Router(config-controller)# framing esf

Defines the framing characteristics.

Step 4

Router(config-controller)# linecode value

Sets the line-code type to match that of your telephone company service provider.

Step 5

Router(config-controller)# pri-group timeslots range

Configures ISDN PRI.

Step 6

Router(config-controller)# controller T1 1/0/1
or

Router(config-controller)# controller T1 1/0/0:2
or

Router(config-controller)# controller T1 0

Enables the T1 1 on the T1 card controller (Cisco AS5800).

or

Enables the T1 2 controller on the T3 card (Cisco AS5800).

or

Enables the T1 0 controller (Cisco AS5300).

Step 7

Router(config-controller)# framing esf

Defines the framing characteristics.

Step 8

Router(config-controller)# linecode value

Sets the line-code type to match that of your telephone company service provider.

Step 9

Router(config-controller)# pri-group timeslots range

Configures ISDN PRI.

Step 10

Router(config-controller)# exit

Exits controller configuration mode.

Step 11

Router(config)# interface Serial1/0/0:23
or

Router(config)# interface Serial1/0/0:1:23
or

Router(config)# interface Serial0:23

Configures the channel for the first ISDN PRI line on the T1 card. (The ISDN serial interface is the D channel.) (Cisco AS5800)

or

Configures the channel for the first ISDN PRI line on the T3 card. (The serial interface is the D channel.) (Cisco AS5800)

or

Configures the channel for the first ISDN PRI line. (The serial interface is the D channel.) (Cisco AS5300)

Step 12

Router(config-if)# isdn incoming-voice modem

Enables incoming ISDN voice calls. This command has two possible keywords: data and modem. You must use the modem keyword to enable voice calls. The modem keyword represents bearer capabilities of speech.

Step 13

Router(config-if)# interface Serial1/0/1:23
or

Router(config-if)# interface Serial1/0/0:2:23
or

Router(config-if)# interface Serial1:23

Configures the channel for the second ISDN PRI line on the T1 card (Cisco AS5800).

or

Configures the channel for the second ISDN PRI line on the T3 card (Cisco AS5800).

or

Configures the channel for the second ISDN PRI line (Cisco AS5300).

Step 14

Router(config-if)# isdn incoming-voice modem

Enables incoming ISDN voice calls. This command has two possible keywords: data and modem. You must use the modem keyword to enable voice calls. The modem keyword represents bearer capabilities of speech.

Step 15

Router(config-if)# exit

Exits interface configuration mode.

Configuring Voice Ports

As mentioned, under most circumstances, the default voice-port command values are adequate to configure voice ports to transport voice data over your existing IP network. To configure specific voice port parameters, use the following commands beginning in privileged EXEC mode:
Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# voice-port {shelf/slot/port:D} | {shelf/slot/parent:port:D}
or

Router(config)# voice-port controller-number:D

Identifies the voice port you want to configure and enters voice-port configuration mode (Cisco AS5800)

or

Identifies the voice port you want to configure and enters voice-port configuration mode (Cisco AS5300).

Step 3

Router(config-voiceport)# cptone country

Selects the appropriate voice call progress tone for this interface.

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

Step 4

Router(config-voiceport)# compand-type {a-law | u-law}

Selects a companding type for this voice port.

Step 5

Router(config-voiceport)# connection {plar string | trunk string}

(Optional) Specifies either the trunk connection or the PLAR connection. The string argument specifies the destination telephone number.

Step 6

Router(config-voiceport)# music-threshold number

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

Step 7

Router(config-voiceport)# description string

(Optional) Attaches descriptive text about this voice-port connection.

Step 8

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.

Step 9

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.

Step 10

Router(config-voiceport)# echo-canel enable

Enables echo cancellation of voice that is sent out the interface and received back on the same interface.
Step 11
Router(config-voiceport)# echo-canel coverage value
Adjusts the size (in milliseconds) of the echo cancellation. Acceptable values are 16, 24, and 32.
Step 12
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.)
Step 13

Router(config-voiceport)# playout-delay {maximum milliseconds | nominal milliseconds}

Specifies the amount of time in milliseconds configured for the playout delay buffer.

Step 14

Router(config-voiceport)# timeouts initial seconds

Specifies the number of seconds the system will wait for the caller to input the first digit of the dialed digits. Valid entries for this command are from 0 to 120.

Step 15

Router(config-voiceport)# timeouts interdigits seconds

Specifies the number of seconds the system will wait (after the caller has input the initial digit) for the caller to input a subsequent digit. Valid entries for this command are from 0 to 120.

Step 16

Router(config-voiceport)# timeouts ringing {seconds | infinity}

Specifies the number of seconds the system will continue to ring the destination if there is no answer.

Step 17

Router(config-voiceport)# timeouts wait-release {seconds | infinity}

Specifies the wait release timeout duration in seconds.

Step 18

Router(config-voiceport)# translate {called number | calling number}

Defines translation rules pertaining to either the called or calling numbers.

Step 19

Router(config-voiceport)# exit

Exits voice-port configuration mode.
For more information on specific voice-port configuration commands or additional voice-port commands, refer to the Cisco IOS Multiservice Applications Command Reference publication.

Verifying ISDN PRI Configuration

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

•To verify that the data configured is correct, use the show voice port command.

•If you have not configured your device to support DID, dial in to the router and learn if you have dial tone.

•Enter a DTMF digit. If the dial tone stops, you have two-way voice connectivity with the router.

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 "Configuring IP" chapter in the Cisco IOS IP and IP Routing Configuration Guide publication.

•Determine if the VFC has been correctly installed. For more information, refer to Installing Voice-over-IP Feature Cards in Cisco AS5300 Universal Access Servers, which came with your VNM.

•To learn if the VFC is operational, use the show vfc slot number command.

•To view layer status information, use the show isdn status command. If you receive a status message stating that Layer 1 is deactivated, make sure the cable connection is not loose or disconnected. (This status message indicates a problem at the physical layer.)

•With T1 lines, determine if your a-law setting is correct. With E1 lines, determine if your u-law setting is correct. To configure both a-law or u-law values, use the cptone command. For more information about the cptone command, refer to the Cisco IOS Multiservice Applications Command Reference publication.

•If dialing cannot occur, use the debug isdn q931 command to check the ISDN configuration.

Configuring E1 R2 Signalling for VoIP

The VoIP VNM for the Cisco AS5300 supports E1 R2 signalling and ISDN PRI. R2 signalling is an international signalling standard that is common to channelized E1 networks. However, there is no single signalling standard for R2. The ITU-T Q.400-Q.490 recommendation defines R2, but a number of countries and geographic regions implement R2 in entirely different ways. Cisco addresses this lack of standards by supporting many localized implementations of R2 signalling in its Cisco IOS software.

The Cisco E1 R2 signalling default is ITU, which supports the technology used in the following countries: Denmark, Finland, Germany, Russia (ITU variant), Hong Kong (ITU variant), and South Africa (ITU variant). The expression "ITU variant" means there are multiple R2 signalling types in the specified country, but Cisco supports the ITU variant.

Cisco also supports specific local variants of E1 R2 signalling in the following regions, countries, and corporations:

•Argentina

•Australia

•Brazil

•China

•Colombia

•Costa Rica

•East Europe (includes Croatia, Russia, and the Slovak Republic)

•Ecuador ITU

•Ecuador LME

•Greece

•Guatemala

•Hong Kong (China variant)

•Indonesia

•Israel

•Korea

•Malaysia

•Mexico (Telmex corporation)

•Mexico (Telnor corporation)

•New Zealand

•Paraguay

•Peru

•Philippines

•Saudi Arabia

•Singapore

•South Africa (Panaftel variant )

•Thailand

•Uruguay

•Venezuela

•Vietnam

Of the local variants listed, the following local variants have been verified:

•Argentina

•Brazil

•China

•Mexico (Telmax)

•Singapore

•Thailand

R2 signalling is channelized E1 signalling used in Europe, Asia, and South America. It is equivalent to channelized T1 signalling in North America. There are two types of R2 signalling: line signalling and interregister signalling. R2 line signalling includes R2 digital, R2 analog, and R2 pulse. R2 interregister signalling includes R2 compelled, R2 noncompelled, and R2 semicompelled. These signalling types are configured using the cas-group command.

Many countries and regions have their own E1 R2 variant specifications, which supplement the ITU-T Q.400-Q.490 recommendation for R2 signalling. Unique E1 R2 signalling parameters for specific countries and regions are set by entering the cas-custom command followed by the country command.

The Cisco implementation of R2 signalling has dialed number identification service (DNIS) support turned on by default. If you enable the automatic number identification (ani) option, the collection of DNIS information is still performed. Specifying the ani option does not disable DNIS collection. DNIS is the number being called. ANI is the number of the caller. For example, if you are configuring router A to call router B, then the DNIS number is assigned to router B; the ANI number is assigned to router A. ANI is similar to caller ID.

To configure E1 R2 signalling, use the following commands beginning in global configuration mode:
Command

Purpose

Step 1

Router(config)# controller e1 number

Specifies the E1 controller that you want to configure with R2 signalling.

Step 2

Router(config-controller)# cas-group channel timeslots range type {r2-analog | r2-digital | r2-pulse} [dtmf | r2-compelled [ani] | r2-non-compelled [ani] | r2-semi-compelled [ani]]

Configures R2 CAS on the E1 controller. For a complete description of the available R2 options, refer to the cas-group (controller e1) command in the Cisco IOS Dial Services Command Reference publication.

Step 3

Router(config-controller)# cas-custom channel

Enters cas-custom configuration mode. In this mode, you can localize E1 R2 signalling parameters, such as specific R2 country settings for Hong Kong.

For the customization to take effect, the number used for the channel argument in the cas-custom command must match the channel number specified by the cas-group command.

Step 4

Router(config-controller)# country name use-default

Specifies the local country, region, or corporation specification to use with R2 signalling. Replace the name argument with one of the supported country names. Refer to the cas-custom command in the Cisco IOS Dial Services Command Reference publication for the list of supported regions, countries, or corporation specifications.

We strongly recommend that you include the use-defaults option, which engages the default settings for a specific country. The default setting for all countries is ITU.
Step 5
ani-digits
answer-signal
caller-digits
category
default
dnis-digits
invert-abcd
ka
kd
metering
nc-congestion
unused-abcd
request-category
(Optional) Further customizes the R2 signalling parameters. Some switch types require you to fine-tune your R2 settings. Do not tamper with these commands unless you fully understand the requirements of your switch.

For nearly all network scenarios, the country name use-defaults command fully configures the local settings for your country. You should not need to perform Step 5.

Refer to the cas-custom command in the Cisco IOS Dial Services Command Reference publication for more information about each signalling command.
Step 6

Router(config-controller)# exit

Exits controller configuration mode.

Step 7

Router(config)# voice-port controller-number:channel-number

Enters voice-port configuration mode for the specified voice port.

Step 8

Router(config-voice-port)# cptone country-code

Defines the country-specific pulse code modulation (PCM) encoding and tones. The PCM encoding type must match the country code defined by the cas-custom command.

Step 9

Router(config-voice-port)# exit

Exits voice-port configuration mode.

Step 10

Router(config)# exit

Exits global configuration mode.
The E1 R2 signalling type (whether ITU, ITU variant, or local variant as defined by the cas-custom command) needs to match the appropriate PCM encoding type as defined by the cptone command. For countries for which a cptone value has not yet been defined, you can try the following:

•If the country uses a-law E1 R2 signalling, use the GB value for the cptone command.

•If the country uses u-law E1 R2 signalling, use the US value for the cptone command.

For more information about configuring R2 signalling, refer to the Cisco IOS Dial Services Configuration Guide: Terminal Services and the Cisco IOS Dial Services Configuration Guide: Network Services publications.

Verifying E1 R2 Signalling Configuration

You can check the validity of your E1 R2 signalling configuration by performing the following tasks:

•To view the status for all controllers, use the show controller e1 command. To view the status for a particular controllers, use the show controller e1 command.

•To check the robbed-bit signalling status of each channel, use the debug serial interface command and the show controller e1 command.

Troubleshooting Tips

If the connection does not come up, check for the following:

•Loose wires, splices, connectors, shorts, bridge taps, and grounds

•Backward send and receive

•Mismatched framing types (for example, CRC-4 versus no-CRC-4)

•Send and receive pair separation (crosstalk)

•Faulty line cards or repeaters

•Noisy lines (for example, power and crosstalk)

If you see errors on the line or the line is going up and down, check for the following:

•Mismatched line codes—for example, HDB3 versus AMI

•Receive level

•Frame slips due to poor clocking plan

Configuring T1 CAS

CAS is the transmission of signalling information within the voice channel. Various types of CAS signalling are available in the T1 world. The most common forms of CAS signalling are loop-start, ground-start, and E&M. The main disadvantage of CAS signalling is its use of user bandwidth to perform signalling functions. CAS signalling is often referred to as robbed-bit signalling because user bandwidth is being "robbed" by the network for other purposes. In addition to receiving and placing calls, CAS signalling processes the receipt of DNIS and automatic number identification (ANI) information, which is used to support authentication and other functions.

The service provider application for T1 CAS includes connectivity to the public network using T1 CAS from the Cisco AS5300 to the end office switch. In this configuration, the Cisco AS5300 captures the dialed-number or called-party number information and passes it along to the upper level applications for interactive voice response (IVR) script selection, modem pooling, and other applications. Service providers also require access to calling party number, ANI, for user identification, for billing account number, and in the future, for more complicated call routing.

Service providers that implement VoIP include traditional voice carriers, new voice and data carriers, and existing ISPs. Some of these service providers might use subscriber side lines for their VoIP connectivity to the PSTN; others might use tandem-type service provider connections.

T1 CAS Signalling Systems

Voice over IP for the AS5300 supports the following T1 CAS signalling systems:

•E&M. E&M signalling is typically used for trunks. It is normally the only way that a CO switch can provide two-way dialing with DID. In all the E&M protocols, off-hook is indicated by A = B = 1, and on-hook is indicated by A = B = 0. If dial pulse dialing is used, the A and B bits are pulsed to indicate the addressing digits. There are several further important subclasses of E&M robbed-bit signalling:

–E&M wink start—Feature Group B

In the original Wink Start protocol, the terminating side responds to an off-hook from the originating side with a short wink (transition from on-hook to off-hook and back again). This wink tells the originating side that the terminating side is ready to receive addressing digits. After receiving addressing digits, the terminating side then goes off-hook for the duration of the call. The originating endpoint maintains off-hook for the duration of the call.

–E&M wink start—Feature Group D

In Feature Group D Wink Start with Wink Acknowledge protocol, the terminating side responds to an off-hook from the originating side with a short wink (transition from on-hook to off-hook and back again) just as in the original Wink Start protocol. This wink tells the originating side that the terminating side is ready to receive addressing digits. After receiving addressing digits, the terminating side then provides another wink (called an acknowledgment wink) that tells the originating side that the terminating side has received the dialed digits. The terminating side then goes off-hook to indicate connection when the ultimate called endpoint has answered. The originating endpoint maintains off-hook for the duration of the call.

–E&M immediate start

In the Immediate Start protocol, the originating side does not wait for a wink before sending addressing information. After receiving addressing digits, the terminating side then goes off-hook for the duration of the call. The originating endpoint maintains off-hook for the duration of the call.

•Ground start/FXS—Ground-start signalling was developed to aid in resolving glare when two sides of a connection tried to go off-hook at the same time. Two sides of the connection simultaneously going off-hook creates a problem with loop-start signalling because the only way an incoming call from the network was recognized by the CPE using loop-start was to ring the phone. The 6-second ring cycle left a substantial amount of time for glare to occur. Ground-start signalling eliminates this problem by providing an immediate seizure indication from the network to the CPE device. This indication tells the CPE device that a particular channel has an incoming call on it. Ground Start is different than E&M in that the A and B bits do not track each other (that is, A is not necessarily equal to B). When the CO delivers a call, it "seizes" a channel (goes off-hook) by setting the A bit to 0. The CO equipment also simulates ringing by toggling the B bit. The terminating equipment goes off-hook when it is ready to answer the call. Digits are usually not delivered for incoming calls.

Channelized T1 Robbed-Bit Features

ISPs can provide switched 56-kbps access to their customers using the Cisco AS5300. The subset of T1 CAS (robbed bit) supported features are as follows:

•Supervisory: line side:

–fxs-loop-start

–fxs-ground-start

–sas-loop-start

–sas-ground-start

–Modified R1

•Supervisory: trunk side:

–e&m-fgb

–e&m-fgd

–e&m-immediate-start

•Informational: line side:

–DTMF

•Informational: trunk side:

–DTMF

–MF

To configure T1 CAS for VoIP on the Cisco AS5300, use the following commands beginning in privileged EXEC mode:
Command

Purpose
Step 1
Router# configure terminal
Enters global configuration mode.
Step 2

Router(config)# controller t1 number

Enters controller configuration mode to configure your controller port. The controller ports are labeled 0 through 3 on the Quad T1/PRI and E1/PRI cards.

Step 3

Router(config-controller)# framing {sf | esf}

Specifies the framing type designated by your telephone company.

Step 4

Router(config-controller)# clock source line primary

Configures the primary PRI clock source. Configure other lines as secondary or internal clock sources. Note that only one PRI can be clock source primary and one PRI can be clock source secondary.

Step 5

Router(config-controller)# linecode {ami | b8zs | hdb3}

Specifies the line-code type designated by your telephone company.

Step 6

Router(config-controller)# cas-group channel timeslots range type signal

Configures all channels for E&M, FXS, and SAS analog signalling. Enter 1-24 for T1. If E1, enter 1-31.

Signalling types for the signal argument include e&m-fgb, e&m-fgd, e&m-immediate-start, fxs-ground-start, fxs-loop-start, sas-ground-start, and sas-loop-start.

You must use the same type of signalling that your CO uses.

For E1 using the Anadigicom converter, use cas e&m-fgb signalling.

Step 7

Router(config-controller)# controller t1 number

Enters controller configuration mode to configure the
second controller port (there are a total of four controller ports). The controller ports are labeled 0 through 3 on the Quad T1/PRI and E1/PRI cards.

Step 8

Router(config-controller)# framing {sf | esf}

Specifies the framing type designated by your telephone company.

Step 9

Router(config-controller)# clock source line secondary

Configures the secondary PRI clock source. Note that only one PRI can be clock source primary and one PRI can be clock source secondary.

Step 10

Router(config-controller)# linecode {ami | b8zs | hdb3}

Specifies the line-code type designated by your telephone company.

Step 11

Router(config-controller)# cas-group channel timeslots range type signal

Configures all channels for E&M, FXS, and SAS analog signalling. Enter 1-24 for T1. If E1, enter 1-31.

Signalling types for the signal argument include e&m-fgb, e&m-fgd, e&m-immediate-start, fxs-ground-start, fxs-loop-start, sas-ground-start, and sas-loop-start.

You must use the same type of signalling that your CO uses.

For E1 using the Anadigicom converter, use cas e&m-fgb signalling.

Step 12

Router(config-controller)# controller t1 number

Enters controller configuration mode to configure the third controller port (there are a total of four controller ports). The controller ports are labeled 0 through 3 on the Quad T1/PRI and E1/PRI cards.

Step 13

Router(config-controller)# framing {sf | esf}

Specifies the framing type designated by your telephone company.

Step 14

Router(config-controller)# clock source line internal

Configures the internal PRI clock source. Note that only one PRI can be clock source primary and one PRI can be clock source secondary. All other controller ports use an internal PRI clock source.
Step 15
Router(config-controller)# linecode {ami | b8zs | hdb3}
Specifies the line-code type designated by your telephone company.
Step 16

Router(config-controller)# cas-group channel timeslots range type signal

Configures all channels for E&M, FXS, and SAS analog signalling. Enter 1-24 for T1. If E1, enter 1-31.

Signalling types for the signal argument include e&m-fgb, e&m-fgd, e&m-immediate-start, fxs-ground-start, fxs-loop-start, sas-ground-start, and sas-loop-start.

You must use the same type of signalling that your CO uses.

For E1 using the Anadigicom converter, use cas e&m-fgb signalling.
Repeat Steps 12 through 16 to configure the last controller.

Verifying T1 CAS Configuration

You can check the validity of your T1 CAS configuration by entering the show controller t1 or show controller e1 command and specify the port number.

Troubleshooting Tip

Make sure the show controller t1 output is not reporting alarms or violations.

Configuring Busyout Monitor for VoIP

The Busyout Monitor feature is one aspect of Call Admission Control (CAC) that allows network administrators to use both a data network and the PSTN to provide the best possible quality for VoIP calls. Although voice calls are routed across the data network whenever possible to take advantage of the cost savings provided by integrated applications, the Busyout Monitor allows network administrators to provide voice services through the PSTN in the event of a network interface failure.

If a locally connected LAN or WAN interface on a VoIP gateway fails, it busies out voice ports, which means that a connected PBX or key system reroutes the call through the local PSTN.

The Busyout Monitor CAC feature provides the following benefits:

•Before the Busyout Monitor feature, there was no logical connection between the LAN/WAN interfaces of a Cisco 2600 or 3600 series VoIP gateway and the directly connected voice ports, although most PBXs and key systems can reroute a call when the primary path is busy or out of service. If one or more interfaces failed, the PBX or key system continued to accept calls that could not be completed and people placing these calls did not know that the call path failed. The Busyout Monitor feature takes advantage of the rerouting capabilities of private communications systems.

•Because a network administrator can define Busyout Monitor port by port, the feature allows freedom in choosing the level of monitoring for VoIP calls.

•Tracks any directly connected main interface, subinterface, or virtual interface (for example, dialer, virtual template, and so on) but does not monitor the status of remote devices.

•Monitors multiple locally connected LAN/WAN interfaces for each port, so that a network administrator can take advantage of multiple IP paths before rerouting calls to the PSTN. If the Busyout Monitor feature is checking multiple LAN/WAN interfaces for a single voice port, all of those interfaces must fail before the feature busies out the voice port.

Busyout Monitor has the following restriction: Busyout Monitor monitors only locally connected LAN/WAN interfaces and does not monitor the status of remote devices. The feature cannot determine the status of the end-to-end path.

Note In some cases, for example, in a VoIP over Frame Relay environment, you can use the Frame Relay PVC end-to-end keepalive feature to track the end-to-end path and thereby busy out a port when its corresponding PVC is down. For more information about Frame Relay keepalive, refer to the Cisco IOS Wide-Area Networking Command Reference and the Cisco IOS Wide-Area Networking Configuration Guide publications.

To configure Busyout Monitor, use the following commands beginning in privileged EXEC mode:

Command

Purpose

Step 1

Router# configure terminal

Enters global configuration mode.

Step 2

Router(config)# voice-port slot/port:ds0-group-no
or

Router(config)# voice-port controller-number:D
or

Router(config)# voice-port {shelf/slot/port:D} | {shelf/slot/parent:port:D}
or

Router(config)# voice-port slot/port:ds0-group-no

Enters voice-port configuration mode (Cisco 2600/3600 series).

or

Enters voice-port configuration mode (Cisco AS5300).

or

Enters voice-port configuration mode (Cisco AS5800).

or

Enters voice-port configuration mode (Cisco 7200 series).

Note The syntax of the voice-port command is specific to Cisco hardware platforms.

Step 3

Router(config-voice-port)# busyout monitor interface interface number

(Optional) Allows you to specify a LAN or WAN interface that will be monitored, and, when it is down, triggers a busyout (off-hook) state on the voice port. This allows rerouting of calls. For example, if you specify Serial 1/0 as the interface and number, the voice port sends a busyout signal when that interface is down. You can issue the command repeatedly to specify as many interfaces, virtual interfaces, and subinterfaces as are required for a voice port.

For example, if you issue the command three times so that three interfaces are monitored, the voice port only goes into busyout state when all three interfaces are down. When any one of the interfaces is operational, the busyout state is removed.

Step 4

Router(config-voice-port)# exit

Exits voice-port configuration mode.

Note Repeat this procedure for each DS0 group that you create.

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

Activates the voice port.

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

Voice Port Configuration Examples

This section contains the following configuration examples:

•Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers Configuration Examples

•1- and 2-Port T1/E1 Multiflex VWICs on Cisco 2600 and 3600 Series Routers Configuration Examples

•Cisco 3600 Series and Cisco 2600 Series ISDN BRI Configuration Examples

•T1/E1 High-Capacity Digital Voice Port Adapters for the Cisco 7200 Series Configuration Examples

•Busyout Monitor Configuration Example

Digital T1 Packet Voice Trunk Network Modules on Cisco 2600 and 3600 Series Routers Configuration Examples

This section includes the following configuration examples:

•Routed digits. Shows how to set up a router to collect digits from the PBX/PSTN or from a phone and route the VoIP call based on the digits received.

•FRF.12. Shows how to configure a Cisco 2600 or 3600 router to support FRF.12 fragmentation and queueing in a VoIP over Frame Relay network.

•Gatekeeper. Shows how to configure a Cisco 2600 or 3600 series router to route VoIP calls by using an H.323 gatekeeper.

•PLAR. Shows how to set up a Cisco 2600 or 2600 series router for PLAR.

•Trunk connection. Shows how to configure a Cisco 2600 or 3600 router for a transparent trunk connection.

•Variable-length digits. Shows how to configure a Cisco 2600 or 3600 router to collect variable-length strings of digits from either a PBX/PSTN or a telephone and route the VoIP call based on the digits received.

•Drop-and-insert. Shows how to configure a Cisco 2600 or 3600 router with a 2-port drop-and-insert T1 multiflex trunk voice/WIC (VWIC-2MFT-T1-DI) and a digital T1 packet VNM so that individual DS0 channels are transparently passed between T1 ports without going through a DSP. For example, this allows the directing of some PBX channels to the PSTN for long-distance service, while other channels are compressed for VoIP calls between interoffice sites.

These examples are not necessarily complete configurations. They are designed to illustrate specific tips and techniques, and only the relevant portions of the configurations are shown. Each configuration includes a brief introduction, side-by-side configurations for routers at either end, and explanations of key points.

Routed Digits—Switched VoIP Calls

Figure 20 shows how to set up a Cisco 2600 or 3600 router to collect digits from either a PBX/PSTN or a phone and route a VoIP call based on the digits received. The commands used in the configurations are explained inline. Only relevant sections of the configuration are shown. The example assumes that the IP portion of the network is already in place.

Figure 20 Sample Configuration: Routed Digits
Alpha Router

Beta Router
hostname router-alpha
!
voice-card 1
 codec complexity high
!
dial-peer voice 1 voip
 codec g723r53
 fax-rate 14400
 destination-pattern 5....
 session target ipv4:192.168.100.1
!
dial-peer voice 2 pots
 destination-pattern 4.... 
 prefix 4 
 port 1/0:1
!
controller T1 1/0
 framing esf
 linecode b8zs
 clock source line
 ds0-group 1 timeslots 1-24 type e&m-wink
!
interface serial 0/0
 ip address 192.168.100.2 255.255.255.0
hostname router-beta
 !
 voice-card 1
  codec complexity high
 !
 dial-peer voice 1 voip
  codec g723r53
  fax-rate 14400
  destination-pattern 4....
  session-target ipv4:192.168.100.2
 !
 dial-peer voice 2 pots
  destination-pattern 5....
  prefix 5
  port 1/0:1
 !
 controller T1 1/0
  framing esf
  linecode b8zs
  clock source internal 
  ds0-group 1 timeslot 1-24 type e&m-wink
 !
 interface s0/0
  ip address 192.168.100.1 255.255.255.0
In this configuration, the PBX seizes the T1 to the router, which expects to collect digits from the PBX. Upon collecting those digits, the router tries to match a dial peer to route the call. If the router receives the correct digits, it routes the call according to the configuration of the dial peer.

Here are some key points for consideration:

•The codec complexity high command tells the router which types of codecs can be used on this voice card—either high or medium. High-complexity codecs permits only two calls for each DSP (6 for each PVDM-12). The codecs supported under high complexity are G.711, G.726, G.729, G.729 Annex B, G.728, G.723.1, G.723.1 Annex A, and fax relay. The default is medium complexity, which allows G.711, G.726, G.729 Annex A, G.729 Annex A with Annex B, and fax relay. Medium-complexity codecs permit 4 calls for each DSP—a total of 12 for each PVDM-12. All T1 cards in a router must have the same complexity. To change the codec complexity, first remove any configured DS0 group from the T1 controller and then reapply it after the change is complete.

•The ds0-group 1 timeslots 1-24 type e&m-wink command performs the following functions:

–Defines the T1 channels for compressed voice calls.

–Defines the signalling method that the router uses to connect to the PBX or PSTN.

–Automatically creates a voice-port 1/0:1. The numbering for this voice-port is slot/port:ds0-group no. In this configuration, all calls to "4...." or "5...." are routed to any DS0 time slot, although only 1/0:1 is shown. To map individual DS0s, define additional DS0 groups under the T1 controller. Mapping additional DS0 groups creates individual DS0 voice ports.

•The dial-peer voice commands define the dialing plan within the router. They specify both the remote phone numbers (voip or vofr) and the locally connected phone numbers (pots). The digits in the destination pattern can either be complete numbers or partial numbers with wildcard digits, represented by ".". Each "." represents an individual digit for collection.

FRF.12—Switched VoIP Calls

Figure 21 shows how to configure a Cisco 2600 or 3600 router to support FRF.12 fragmentation and queueing in a VoIP over Frame Relay network. FRF.12 is a Frame Relay Forum standard mechanism for fragmenting data packets. This fragmentation helps eliminate the delays that occur when sending voice and data over the same network—large data packets can delay smaller voice packets from being sent into the IP network. FRF.12 is also supported on the MC3810 and 7200 routers, which can be used as tandem nodes for VoIP networks.

Note This example shows VoIP over Frame Relay, which is not the same as VoFR. For more information about VoFR, see the Cisco IOS Release12.0(4)T feature module Voice over Frame Relay Using FRF.11 and FRF.12.

Figure 21 Sample Configuration: FRF.12 Switched VoIP Calls

The following configuration fragments both the IP and IPX data traffic to 80 bytes, allowing the VoIP traffic to be only minimally delayed on the network. The FRF.12 setup also traffic-shapes the output traffic rate to match the provisioned CIR from the Frame Relay carrier. Matching the provisioned CIR from the Frame Relay carrier ensures that traffic is not dropped or delayed within the Frame Relay network.

Here are some key points for consideration:

•The frame-relay traffic-shaping command enables Frame Relay traffic shaping (FRTS) on the main interface. Enable it if FRTS will be used on subinterfaces.

•The class cisco_frf12 command tells the interface to use the parameters for FRTS defined in the map class called cisco_frf12.

•The map-class cisco_frf12 grouping of commands defines the rules for FRTS. If per-interface/subinterface parameters must differ, define multiple map classes per router.

•The frame-relay fragment 80 command defines the size of the data or voice packets that FRF.12 fragments. Set the size to about the size of the voice packets or slightly larger. A general rule is 80 bytes for each DS0 of WAN bandwidth. With large quantities of bandwidth and small data frames, the fragment size may need to remain small.

•The frame-relay fair-queue command enables WFQ on a per-PVC basis to ensure that voice traffic gets priority over data traffic.
Alpha Router

Beta Router
hostname router-alpha
!
ipx routing
! 
voice-card 1
 codec complexity high
!
controller T1 1/0
 framing esf 
 linecode b8zs
 clock source line 
 ds0-group 1 timeslot 1-24 type e&m-wink
!
dial-peer voice 1 voip
 dtmf-relay  h245-alpha
 codec g723r53
 destination-pattern 5....
 session target ipv4:192.168.100.2
!
dial-peer voice 2 pots
 destination-pattern 4....  
 prefix 4
 port 1/0:1
!
interface serial 0/0
 encapsulation frame-relay
 frame-relay traffic-shaping
! 
interface serial 0/0.1 point-to-point
 ip address 192.168.100.1 255.255.255.0
 ipx network ABCD
 frame-relay interface-dlci 100
 class cisco_frf12
!
map-class frame-relay cisco_frf12
frame-relay voice bandwidth 42000
frame-relay fragment 80
no frame-relay adaptive-shaping
frame-relay cir 32000
frame-relay bc 1000
frame-relay mincir 64000
frame-relay fair-queue
hostname router-beta
!
ipx routing
!
voice-card 1
 codec complexity high
!
controller T1 1/0
 framing esf
 linecode b8zs
 clock source line
 ds0-group 1 timeslot 1-24 type e&m-wink
!
dial-peer voice 1 voip
 dtmf-relay h245-alpha
 codec g723r53
 destination-pattern 4....
 session target ipv4:192.168.100.2
!
dial-peer voice 2 pots
 destination-pattern 5....
 prefix 5
 port 1/0:1
!
interface serial 0/0
 encapsulation frame-relay
 frame-relay traffic-shaping
!
interface serial 0/0.1 point-to-point
 ip address 192.168.100.2 255.255.255.0
 ipx network ABCD
 frame-relay interface-dlci 101
 class cisco_frf12
!
map-class frame-relay cisco_frf12
frame-relay voice bandwidth 42000
frame-relay fragment 80
no frame-relay adaptive-shaping
frame-relay cir 64000
frame-relay bc 1000
frame-relay mincir 64000
frame-relay fair-queue
Routing Calls Through an H.323 Gatekeeper

Note With the introduction of Cisco IOS Release 12.0(5)T and subsequent releases, Cisco VoIP gateways support H.323v2 (H.323 Version 2), which is backwards compatible with systems running H.323 Version 1. However, H.323 Version 2 features do not interoperate with H.323 Version 1 features in Cisco IOS releases prior to 11.3(9)NA or 12.0(3)T. Earlier Cisco IOS versions contain H.323 Version 1 software that does not support protocol messages with an H.323 Version 2 protocol identifier. All systems must be running either Cisco IOS Release 11.3(9)NA and later or Cisco IOS Release 12.0(3)T and later releases to interoperate with H.323 Version 2. Gateway Resource Availability Indication (RAI) messages are currently not supported on the Cisco 2600 and 3600 series. (These are messages that are sent to the Gatekeeper to inform it about the status of a Gateway DSP or DS0 availability.)

Figure 22 shows how to configure a Cisco 2600 or 3600 series router to route VoIP calls through an H.323 gatekeeper. This setup shows calls being routed from a gateway in Zone-Alpha, through the gatekeeper, to a gateway in Zone-Beta.

Figure 22 Sample Configuration: Routing Calls Through an H.323 Gatekeeper
Gatekeeper
hostname router-gatekeeper
!
gatekeeper
zone local alpha alpha.com
zone local beta beta.com
no use-proxy alpha.com remote-zone beta.com
no use-proxy beta.com remote-zone alpha.com
zone prefix router-alpha 4....
zone prefix router-beta 5....
no shutdown
!
interface ethernet 0/0
ip address 10.1.1.3 255.255.255.0
Alpha Router

Beta Router
hostname router-alpha
!
voice-card 1
!
controller T1 1/0
 framing esf 
 linecode b8zs
 clock source internal
 ds0-group 1 timeslot 1-24 type e&m-wink
!
voice-port 1/0:1
!
dial-peer voice 1 voip
 dtmf-relay h245-alpha
 destination-pattern 5....
 tech-prefix 1#
 session target ras
!
dial-peer voice 2 pots 
 destination-pattern 4....
 prefix 4
 port 1/0:1
!
gateway 
!
interface ethernet 0/0
 ip address 10.1.1.1 255.255.255.0
 h323-gateway voip interface
 h323-gateway voip id alpha ipaddr 10.1.1.3 
1719
 h323-gateway voip h323-id  
router-alpha@alpha.com
 h323-gateway voip tech-prefix 1#
hostname router-beta
!
voice-card 1
!
controller T1 1/0
 framing esf
 linecode b8zs
 clock source line
 ds0-group 1 timeslot 10-24 type e&m-wink
!
voice-port 1/0:1
!
dial-peer voice 1 voip
 dtmf-relay h245-alpha
 destination-pattern 4....
 tech-prefix 1#
 session target ras
!
dial-peer voice 2 pots
 destination-pattern 5....
 prefix 5
 port 1/0:1
!
gateway
!
interface ethernet 0/0
 ip address 10.1.1.2 255.255.255.0
 h323-gateway voip interface
 h323-gateway voip id beta ipaddr 10.1.1.3 
1719
 h323-gateway voip h323-id 
router-beta@beta.com
 h323-gateway voip tech-prefix 1#
Here are some key points for consideration:

•The session target ras command tells the router to route through the gatekeeper. RAS is the communication that occurs between an H.323 gateway and the gatekeeper.

•The gateway command tells the router to use RAS to register with the gatekeeper.

•The gatekeeper command tells the router to act as a gatekeeper and respond to calls made through RAS from H.323 gateways and H.323 clients.

PLAR Configuration—Switched VoIP Calls

Figure 23 shows how to set up a Cisco 2600 or 3600 series router for a PLAR. PLAR is used to allow a station or DS0 to go off hook, and—without the user dialing digits—have a call completed to the far end. PLAR can also provide dial tone from a remote PBX for off-premises applications.

In this configuration, the phones off router Beta go off hook and receive dial tone from the PBX connected to router Alpha. From there, users can dial any digits in to the PBX as if their stations are directly connected to it.

Figure 23 Sample Configuration: PLAR

Here are some key points for consideration:

•The configuration includes the dtmf-relay command because the users will send DTMF digits to the PBX over the VoIP network, and the router must not compress these digits. The command ensures that the router sends the digits out-of-band, so that they are not distorted.

•The connection plar command configures the PLAR connection. The router uses the digits that follow the command internally to send the call to a dial peer—the user does not dial these digits.

•Voice port 1/0:2 is created by DS0 group 2, as shown in the last digit of the specification. Each DS0 group creates a separate voice port, which allows the definition of individual DS0s on the digital T1 card.
Alpha Router

Beta Router
hostname router-alpha
!
voice-card 1
!
!
controller T1 1/0
 framing esf
 linecode b8zs
 ds0-group 1 timeslot 1 type fxo-loop
 ds0-group 2 timeslot 2 type fxo-loop
!
dial-peer voice 1 voip
 dtmf-relay  h245-alpha
 codec g729a
 destination-pattern 2..
 session target ipv4:192.168.100.2 
!
dial-peer voice 2 pots
 destination-pattern 101 
 port 1/0:1
!
dial-peer voice 3 pots
 destination-pattern 102
 port 1/0:2
!
voice-port 1/0:1
 connection plar 201
!
voice-port 1/0:2
 connection plar 202
!
interface s0/0
 ip address 192.168.100.1 255.255.255.0
hostname router-beta
!
dial-peer voice 1 voip
 destination-pattern 1..
 dtmf-relay h245-alpha
 codec g729a
 session target ipv4:192.168.100.1
!
dial-peer voice 2 pots
 destination-pattern 201
 port 1/1
!
!
dial-peer voice 3 pots
 destination-pattern 202
 port 1/2
!
voice-port 1/1
!
!
voice-port 1 / 2
!
!
interface serial 0/0
 ip address 192.168.100.2 255.255.255.0
Connection Trunk Configuration—Permanent VoIP Calls

Figure 24 shows how to configure a Cisco 2600 or 3600 router for a trunk connection. A trunk connection is like a "wire" between the two routers. It is a transparent connection, so it allows features such as hookflash (also called switchhook flash) or "hoot n' holler" (point-to-point) to pass. This type of trunk configuration can also be used for OPXs that require rollover to a centralized voice-mail system when the user does not answer.

A trunk connection can only be used between E&M ports or with FXO-to-FXS connections.

Figure 24 Sample Configuration: Connection Trunk Permanent VoIP Calls
Alpha Router

Beta Router
hostname router-alpha
!
voice-card 1
!
controller T1 1/0
 framing esf 
 linecode b8zs
 ds0-group 1 timeslot 1 type e&m-wink
 ds0-group 2 timeslot 2 type e&m-wink
 clock source line
!
voice-port 1/0:1
 connection trunk 1111 
!
voice-port 1/0:2 
 connection trunk 1112
!
dial-peer voice 1 voip
 dtmf-relay h245-alpha
 codec g729a
 destination-pattern 111.
 session target ipv4:192.168.100.2
!
dial-peer voice 2 pots
 destination-pattern 2221
port 1/0:1
!
dial-peer voice 3 pots
 destination-pattern 2222
 port 1/0:2
!
interface serial 0/0
 ip address 192.168.100.1 255.255.255.0
hostname router-beta
!
voice-card 1 
!
controller T1 1/0
 framing esf
 linecode b8zs
 ds0-group 1 timeslot 1 type e&m-wink
 ds0-group 2 timeslot 2 type e&m-wink
 clock source line
!
voice-port 1/0:1
 connection trunk 2221
!
voice-port 1/0:2
 connection trunk 2222
!
dial-peer voice 1 voip
 dtmf-relay h245-alpha
 codec g729a
 destination-pattern 222.
 session target ipv4:192.168.100.1
!
dial-peer voice 2 pots
 destination-pattern 1111
 port 1/0:1
!
dial-peer voice 3 pots
 destination-pattern 1112
 port 1/0:2
!
interface serial 0/0
 ip address 192.168.100.2 255.255.255.0
In this configuration, a permanent and transparent path is set up between individual DS0s on each router. It passes dial tone from the remote PBX and passes DTMF digits out of band.

The connection trunk command establishes the permanent trunk connection between the routers. The digits after the command are passed internally within the router to match a dial peer so that the call can be set up.

Drop-and-Insert Sample Configuration

Figure 25 shows an example of drop-and-insert. Drop-and-insert technology is one way to integrate old PBX technologies with VoIP. It allows you to take 64-kbps DS0 channels from one T1 and digitally cross-connect them to 64-kbps DS0 channels on another T1. Drop-and-insert is sometimes called TDM cross-connect.

Drop-and-insert allows individual 64-kbps DS0 channels to be transparently passed, uncompressed, between T1 ports without passing through a DSP. Using this method, the channel traffic is sent between a PBX and CO switch (PSTN) or other telephony device, allowing the use, for example, of some PBX channels for long-distance service through the PSTN while the router compresses others for interoffice VoIP calls. In addition, drop-and-insert can cross-connect a telephony switch (from the CO or PSTN) to a channel bank to provide external analog connectivity.

Note the following design requirements:

•On the Cisco 2600, 3620, and 3640 platforms, drop-and-insert is only permitted between T1 ports on the same multiflex trunk module (MFT). The MFT module can either be in a standalone WIC slot or integrated into a digital T1 packet voice trunk network module VWIC slot.

•When the MFT module is installed in the VWIC slot of a digital T1 packet voice trunk network module, it does not allow the T1 ports to provide WAN connectivity (for example, Frame Relay, PPP, and so on) in addition to voice and drop-and-insert.

•WAN and drop-and-insert capabilities are supported when the MFT is in a standalone WIC slot.

Figure 25 Sample Configuration: Drop-and-Insert

The following configuration example shows how to configure drop-and-insert.
Router RTR-A

Router RTR-B
hostname RTR-A
!
voice-card 1
  codec complexity high
!
controller T1 1/0
 clock source line
 framing esf
 linecoding b8zs
 ds0-group 1 timeslots 1-12 type e&m-wink
 tdm-group 2 timeslots 13-24 type e&m
!
controller T1 1/1
 clock source line primary
 framing esf
 linecoding b8zs
 tdm-group 3 timeslots 13-24 type e&m
!
voice-port 1/0:1
!
dial-peer voice 1 voip
 destination-pattern 4....
 codec g723r63
 dtmf-relay h245-alpha
 session target ipv4:192.168.100.2
!
dial-peer voice 2 pots
 destination-pattern 5....
 prefix 5 
 port 1/0:1
!
interface serial 0/0
 encapsulation ppp
 ip address 192.168.100.1 255.255.255.0
!
connect tdm1 T1 1/0 2 T1 1/1 3
hostname RTR-B
!
voice-card 1
 codec complexity high
!
controller T1 1/0
 clock source line
 framing esf
 linecoding b8zs
 ds0-group 1 timeslots 1-12 type e&m-wink
 tdm-group 2 timeslots 13-24 type e&m
!
controller T1 1/1
 clock source line primary
 framing esf
 linecoding b8zs
 tdm-group 3 timeslots 13-24 type e&m
!
voice-port 1/0:1
!
dial-peer voice 1 voip
 destination-pattern 5....
 codec g723r63
 dtmf-relay h245-alpha
 session target ipv4:192.168.100.1
!
dial-peer voice 2 pots
 destination-pattern 4....
 prefix 4
 port 1/0:1
!
interface serial 0/0
 encapsulation ppp
 ip address 192.168.100.2 255.255.255.0
!
connect tdm1 T1 1/0 2 T1 1/1 3 
Here are some key points for consideration:

•The tdm-group 2 timeslots 13-24 type e&m command defines drop-and-insert by setting up the time slots from each T1 that will be used in the digital cross-connect. The type keyword is optional, but its use is specific to the drop-and-insert feature. For example:

–If you include the type keyword with a signalling type, the drop-and-insert digital cross-connect ensures that the specified signalling (on-hook and off-hook) is passed between the DS0s. It also uses the signalling bits to signal busyout if one of the T1s goes down.

–If you do not use the type keyword, the drop-and-insert cross-connect is clear channel and does not interpret any signalling.

•The connect tdm1 T1 1/0 2 T1 1/1 3 command activates the drop-and-insert digital cross-connect between the T1s. The tdm1 portion of the command is just a name for the cross-connect, and the name can be any word, number, or series of letters.

•You can verify drop-and-insert connections by using the show connect command.

1- and 2-Port T1/E1 Multiflex VWICs on Cisco 2600 and 3600 Series Routers Configuration Examples

This section includes three sample configurations to illustrate different scenarios:

•Drop-and-insert where PSTN and VoIP services are provided through the same service provider line.

•Drop-and-insert where PSTN and data services are provided through the same service provider line.

•Drop-and-insert where PSTN, data, and VoIP services are provided through the same service provider line.

Drop-and-insert technology is one way to integrate old PBX technologies with VoIP. It allows you to take 64-kbps DS0 channels from one T1 or E1 and digitally cross-connect them to 64-kbps DS0 channels on another T1 or E1.

Drop-and-insert allows individual 64-kbps DS0 channels to be transparently passed, uncompressed, between T1/E1 ports without DSP processing. Channel traffic is sent between a PBX and CO switch or other telephony device, allowing the use, for example, of some PBX channels for long-distance service through the PSTN while the router compresses others for interoffice VoIP calls. In addition, drop-and-insert can cross-connect a telephony switch (from the CO or PSTN) to a channel bank to provide external analog connectivity.

Keep the following considerations in mind:

•Drop-and-insert works only between ports on the same multiflex VWIC.

•The VWIC can either be in a standalone WIC slot on a Cisco 2600 series router, integrated into a digital T1 packet voice trunk network module VWIC slot, or installed in a Cisco 3600 series 2-port network module (NM-1E2W, NM-2E2W, and NM-1E1R2W).

•When the VWIC module is installed in the VWIC slot of a digital T1 packet voice trunk network module, the T1 ports do not provide WAN connectivity (for example, Frame Relay, PPP, and so on) in addition to voice and drop-and-insert.

•WAN and drop-and-insert capabilities are supported when the VWIC is in a chassis WIC slot on a Cisco 2600 series router.

Drop-and-Insert with VoIP and PSTN Services

Figure 26 shows drop-and-insert when a 2-port multiflex VWIC is installed in a digital T1 packet voice trunk network module VWIC slot and VoIP is configured. WAN connections must be provided by separate links.

Figure 26 Sample Configuration: Drop-and-Insert with VoIP and PSTN Services

The following configuration shows the configuration for drop-and-insert when a 2-port Multiflex VWIC is installed in a digital T1 packet voice trunk network module VWIC slot and VoIP is configured.
Router RTR-A

Router RTR-B
hostname RTR-A
!
voice-card 1
  codec complexity high
!
controller T1 1/0
framing esf
linecoding b8zs
ds0-group 1 timeslots 1-12 type e&m-wink
tdm-group 2 timeslots 13-24 type e&m
!
controller T1 1/1
framing esf
linecoding b8zs
clock source line primary
tdm-group 3 timeslots 13-24 type e&m
!
voice-port 1/0:1
!
dial-peer voice 1 voip
destination-pattern 4...
codec g723r63
dtmf-relay h245-alpha
session target ipv4:209.165.200.253
session target ipv4:209.165.200.252
!
dial-peer voice 2 pots
destination-pattern 5...
prefix 5 
port 1/0:1
!
interface serial 0/0
encapsulation ppp
ip address 209.165.200.252 255.255.255.224
!
connect tdm1 T1 1/0 2 T1 1/1 3
hostname RTR-B
!
voice-card 1
  codec complexity high
!
controller T1 1/0
framing esf
linecoding b8zs
ds0-group 1 timeslots 1-12 type e&m-wink
tdm-group 2 timeslots 13-24 type e&m
!
controller T1 1/1
framing esf
linecoding b8zs
clock source line primary
tdm-group 3 timeslots 13-24 type e&m
!
voice-port 1/0:1
!
dial-peer voice 1 voip
destination-pattern 5....
codec g723r63
dtmf-relay h245-alpha
!
!
!
dial-peer voice 2 pots
destination-pattern 4....
prefix 4
port 1/0:1
!
interface serial 0/0
encapsulation ppp
ip address 209.165.200.253 255.255.255.224
!
connect tdm1 T1 1/0 2 T1 1/1 3 
Clock Sources

In this example, two clock sources are available on each router multiflex VWIC ports: one from the PBX and one from the PSTN CO. However, the clock sources must be the same, so the system adjusts to this need.

The primary keyword of the clock source command, applied to T1 1/1, means that the PSTN is providing the clock source. The T1 1/0 port connected to the PBX is automatically put into looped-time mode, which means that the port takes the clocking received on its Rx (receive) pair and regenerates it back on its Tx (transmit) pair. While it is receiving clocking, it does not drive the on-board clock. It is "spoofing" the port so that the connected PBX does not detect clocking that is out of synchronization, which is indicated by slips. The router detects the slips as controlled and does not force the port to fail.

Additional Considerations

Here are some additional key points for consideration:

•The tdm-group 2 timeslots 13-24 type e&m command defines drop-and-insert by setting up the time slots from each T1 that will be used in the digital cross-connect. The type keyword is optional, but its use is specific to the drop-and-insert feature. For example:

–If you include the type keyword with a signalling type, the drop-and-insert digital cross-connect ensures that the specified signalling (on-hook and off-hook) is passed between the DS0s. It also uses the signalling bits to signal busyout if one of the T1s goes down.

–If you do not use the type keyword, the drop-and-insert cross-connect is clear channel and does not interpret any signalling.

•The connect tdm1 T1 1/0 2 T1 1/1 3 command activates the drop-and-insert digital cross-connect between the T1s. The tdm1 portion of the command is just a name for the cross-connect, and the name can be any word, number, or series of letters.

•You can verify drop-and-insert connections by using the show connection command.

Drop-and-Insert with Data and PSTN Services

Figure 27 shows configuration for drop-and-insert when a 2-port Multiflex VWIC is installed in a Cisco 2600 series chassis slot or in a WIC slot of a Cisco 3600 series network module. Frame Relay data and PSTN voice calls travel between the PBXs, but no VoIP or VoIP over Frame Relay information is carried.

Figure 27 Sample Configuration: Drop-and-Insert with Data and PSTN Voice Services

Clock Sources

As in the previous example, two clock sources are available on each router multiflex VWIC ports: one from the PBX and one from the PSTN CO. However, the clock sources must be the same, so the system adjusts to this need.

The primary clock source is T1 or E1 1/0, connected to the PSTN, and its clock is a reference for T1 or E1 1/1. If T1 1/0 fails, the clock source to drive T1 or E1 1/1 is generated from the line to the PBX.

Additional Considerations

The channel-group 0 command is configured in such a way that the service provider can send Frame Relay Link Management Interface (LMI) information on T1 channels 13 through 24 (17 through 31 on E1) for Frame Relay data services. This command automatically creates interface serial 1/0:0.

Interface serial 1/0:0 is where all WAN and Layer 3 protocol details are configured, for example, Frame Relay encapsulation or IP addresses.

T1 Configuration
Router RTR-A

Router RTR-B
hostname RTR-A
!
controller T1 1/0
framing esf
linecoding b8zs
clock source line primary
tdm-group 1 timeslots 1-12 
channel-group 0 timeslots 13-24
!
controller T1 1/1
framing esf
linecoding b8zs
clock source line
tdm-group 2 timeslots 1-12
!
interface serial 1/0:0
encapsulation frame-relay
!
interface serial 1/0:1.1
ip address 209.165.200.252 255.255.255.224
frame-relay interface-dlci 100 br
!
interface ethernet 0
ip address 209.165.200.250 255.255.255.224
!
router eigrp 1
network 209.165.200.224
!
connect tdm1 T1 1/0 1 T1 1/1 2
hostname RTR-B
!
controller T1 1/0
framing esf
linecoding b8zs
clock source line primary
tdm-group 1 timeslots 1-12
channel-group 0 timeslots 13-24
!
controller T1 1/1
framing esf
linecoding b8zs
clock source line
tdm-group 2 timeslots 1-12
!
interface serial 1/0:0
encapsulation frame-relay
!
interface serial 1/0:1.1
ip address 209.165.200.253 255.255.255.224
frame-relay interface-dlci 100 br
!
interface ethernet 0
ip address 209.165.201.1 255.255.255.224
!
router eigrp 1
network 209.165.200.224
network 209.165.201.0
!
connect tdm1 T1 1/0 1 T1 1/1 2
E1 Configuration
Router RTR-A

Router RTR-B
hostname RTR-A
!
controller E1 1/0
framing crc4
linecoding hdb3
clock source line primary
tdm-group 1 timeslots 1-15 
channel-group 0 timeslots 17-31
!
controller E1 1/1
framing crc4
linecoding hdb3
clock source line
tdm-group 2 timeslots 1-15
!
interface serial 1/0:0
encapsulation frame-relay
!
interface serial 1/0:1.1
ip address 209.165.200.252 255.255.255.224
frame-relay interface-dlci 100 br
!
interface ethernet 0
ip address 209.165.200.250 255.255.255.224
!
router eigrp 1
network 209.165.200.224
!
connect tdm1 T1 1/0 1 T1 1/1 2
hostname RTR-B
!
controller E1 1/0
framing crc4
linecoding hdb3
clock source line primary
tdm-group 1 timeslots 1-15
channel-group 0 timeslots 17-31
!
controller E1 1/1
framing crc4
linecoding hdb3
clock source line
tdm-group 2 timeslots 1-15
!
interface serial 1/0:0
encapsulation frame-relay
!
interface serial 1/0:1.1
ip address 209.165.200.253 255.255.255.224
frame-relay interface-dlci 100 br
!
interface ethernet 0
ip address 209.165.201.1 255.255.255.224
!
router eigrp 1
network 209.165.200.224
network 209.165.201.0
!
connect tdm1 T1 1/0 1 T1 1/1 2
Drop-and-Insert with PSTN, Data, and VoIP Services

Figure 28 shows how to use some T1 channels for passing voice from the PSTN to the PBX, and some channels for data services that also pass VoIP traffic. This setup requires both a digital T1 packet voice trunk network module with a multiflex VWIC installed and a separate multiflex VWIC.

Figure 28 Sample Configuration: Drop-and-Insert with PSTN, Data, and VoIP Services

Clock Sources

The primary clock source is T1 1/0, and its clock is a reference for T1 1/1. If T1 1/0 fails, the clock source to drive T1 1/1 is generated internally.
Router RTR-A

Router RTR-B
hostname RTR-A
!
controller T1 1/0
description - NM-HDV connected to PBX
framing esf
linecoding b8zs
clock source internal 
tdm-group 1 timeslots 1-12 
ds0-group 2 timeslots 13-24 type e&m-wink
!
controller T1 1/1
description - xconnect to VWIC T1
framing esf
linecoding b8zs
clock source line
tdm-group 2 timeslots 1-12
!
controller T1 2/0
description - connected to TELCO WAN
framing esf
linecoding b8zs
channel-group 0 timeslots 13-24 
tdm-group 3 timeslots 1-12
clock source line
!
controller T1 2/1
description - xconnect to NM-HDV
framing esf
linecoding b8zs
clock source internal
tdm-group 4 timeslots 1-12
!
voice-port 1/0:2
!
interface serial 2/0:0
encapsulation frame-relay
!
interface serial 1/0:0.1
ip address 209.165.200.252 255.255.255.224
frame-relay interface-dlci 100 br
!
interface ethernet 0
ip address 209.165.200.250 255.255.255.224
!
router eigrp 1
network 209.165.200.224
!
dial-peer voice 1 voip
destination-pattern 5...
session target ipv4:209.165.200.253
!
dial-peer voice 2 pots
destination-pattern 4...
prefix 4
prefix 5
port 1/0:2
port 1/0:2
!
connect tdm1 T1 1/0 1 T1 1/1 2
connect tdm2 T1 2/0 3 T1 2/1 4
hostname RTR-B
!
controller T1 1/0
description -  NM-HDV connected to PBX
framing esf
linecoding b8zs
clock source internal
tdm-group 1 timeslots 1-12
!
controller T1 1/1
description - xconnect to VWIC T1
framing esf
linecoding b8zs
clock source line
tdm-group 2 timeslots 1-12
!
!
controller T1 2/0
description - connected to TELCO WAN
framing esf
linecoding b8zs
channel-group 0 timeslots 13-24
tdm-group 3 timeslots 1-12
clock source line
!
controller T1 2/1
description - xconnect NM-HDV
framing esf
linecoding b8zs
clock source internal
tdm-group 4 timeslots 1-12
!
voice-port 1/0:2
!
interface serial 2/0:0
encapsulation frame-relay
!
interface serial 1/0:0.1
ip address 209.165.200.253 255.255.255.0
frame-relay interface-dlci 100 br
!
interface ethernet 0
ip address 209.165.201.1 255.255.255.224
!
router eigrp 1
network 209.165.200.224
network 209.165.201.0
!
dial-peer voice 1 voip
destination-pattern 4...
session target ipv4:209.165.200.252
!
dial-peer voice 2 pots
destination-pattern 5...
!
connect tdm1 T1 1/0 1 T1 1/1 2
connect tdm2 T1 2/0 3 T1 2/1 4
Additional Considerations

The following connections are made by using channels 1 through 12 from the service provider:

•The channels are brought into the multiflex VWIC that is not installed in the digital T1 packet voice trunk network module.

•These 12 channels cross-connect to the other multiflex VWIC port.

•From there, an external T1 crossover cable cross-connects the channels to the first T1 port on the digital T1 packet voice trunk network module.

•The 12 channels cross-connect to the other T1 port on the digital T1 packet voice trunk network module and out to the connected PBX.

Channels 13 through 24 pass Frame Relay LMI from the service provider for data services, and the channels terminate on the multiflex VWIC channel group. This serial interface is used for data traffic from the Ethernet, and VoIP traffic that originates on channels 13 through 24 from the PBX connected to the digital T1 packet voice trunk network module.

Cisco 3600 Series and Cisco 2600 Series ISDN BRI Configuration Examples

The configuration examples included in this section correspond to the topology shown in Figure 29. The routers each include a BRI VIC and a 2-slot voice network module, along with other VICs and modules that are included for the sake of completeness. Router A is connected to a PBX through the BRI VIC and is connected to Router B by a serial Ethernet interface. Router B includes a BRI VIC for connection to the PSTN, in order to process voice calls from off-premises terminal equipment.

Figure 29 Configuration Topology

Router A: Connection to a PBX

The following example illustrates the configuration of a Cisco 3640 router for connection to a BRI VIC accessing a PBX:
vicbri_3640_s1#show running config
Building configuration...
Current configuration:
!
version 12.0
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname vicbri_3640_s1
!
logging buffered 200000 debugging
!
ip subnet-zero
ip host keyer 223.255.254.254
!
isdn switch-type basic-ni
!
!
The following commands configure the ports on VICs. The last four specified ports are for FXO and E&M VICs.
voice-port 1/0/0
!
voice-port 1/0/1
!
voice-port 2/0/0
!
voice-port 2/0/1
!
voice-port 3/0/0
 operation 4-wire
 type 2
!
voice-port 3/0/1
 operation 4-wire
 type 2
!
voice-port 3/1/0
 input gain 10
 connection plar 39019
!
voice-port 3/1/1
 input gain 10
 connection plar 39020
The following commands configure dial peers to specify where incoming VoIP calls should be directed. In the first example, calls received with a starting digit of 5 are sent to the PBX via the BRI VIC.
dial-peer voice 10 pots
 destination-pattern 5..... 
 port 1/1/0
!
This command sets up a local BRI connection:
dial-peer voice 11 pots 
 destination-pattern 66002
 port 1/0/0
!
In this example, calls with a starting digit of 9 are PSTN calls that are routed over IP:
dial-peer voice 13 voip 
 destination-pattern 9.......
 session target ipv4:12.0.0.2
!
This command sets up an FXS connection over IP to the other router:
dial-peer voice 12 voip (calls to other router with FXS - go over IP)
 destination-pattern 7....
 session target ipv4:12.0.0.2
!
The following global configuration commands define how to expand an extension number into a particular destination pattern:
num-exp 8 9529399
num-exp 1 550950
num-exp 2 76002
The following commands configure the Ethernet and serial interfaces:
interface Ethernet0/0
 ip address 1.14.122.10 255.255.0.0
 ip helper-address 223.255.254.254
 no ip directed-broadcast
!
interface Serial0/0
 ip address 3.0.0.2 255.0.0.0
 no ip directed-broadcast
 no ip mroute-cache
 no keepalive
 no fair-queue
!
interface Ethernet0/1
 ip address 11.0.0.1 255.0.0.0
 no ip directed-broadcast
!
interface Serial0/1
 ip address 14.0.0.1 255.0.0.0
 no ip directed-broadcast
 no keepalive
 shutdown                             
 no fair-queue
 clockrate 2000000
The following commands configure the BRI interfaces:
interface BRI1/0
 no ip address
 no ip directed-broadcast
 isdn switch-type basic-ni1
 isdn twait-disable
 isdn spid1 14085552121010 5552121
 isdn spid2 14085552122010 5552122
 isdn incoming-voice modem
!
interface BRI1/1
 no ip address
 no ip directed-broadcast
 isdn switch-type basic-ni1
 isdn twait-disable
 isdn spid1 14085556362010 5556362
 isdn spid2 14085556364010 5556364
 isdn incoming-voice modem
!
interface BRI2/0
 no ip address
 no ip directed-broadcast
 isdn switch-type basic-ni1
 isdn twait-disable
 isdn spid1 14085555711010 5555711
 isdn spid2 14085555712010 5555712
 isdn incoming-voice modem
!
interface BRI2/1
 no ip address
 no ip directed-broadcast
 isdn switch-type basic-ni1
 isdn twait-disable
 isdn spid1 14085555162010 5555162
 isdn spid2 14085555163010 5555163
 isdn incoming-voice modem
!
ip default-gateway 1.14.0.1
ip classless
ip route 2.0.0.0 255.0.0.0 Ethernet0/1
ip route 2.0.0.0 255.0.0.0 Serial0/1
ip route 223.255.254.254 255.255.255.255 Ethernet0/0
!
!
!
line con 0
 exec-timeout 0 0
 transport input none
line aux 0
line vty 0 4
 login
!
end
vicbri_3640_s1#  
Router B: Connection to PSTN

The following example illustrates the configuration of a Cisco 2600 series router for connection to a BRI VIC accessing an ISDN telephone network:
vicbri_2600_s2#sh run
Building configuration...
Current configuration:
!
version 12.0
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname vicbri_2600_s2
!
logging buffered 200000 debugging
!
ip subnet-zero
!
isdn switch-type basic-ni
!
!
The following commands configure the ports on VICs:
voice-port 1/0/0
!
voice-port 1/0/1
!
The following commands configure dial peers to specify where incoming VoIP calls should be directed. In the first example, a local FXS connection is made to Router A.
dial-peer voice 22 voip
 destination-pattern 6....
 session target ipv4:12.0.0.1
!
This command sets up a connection to the PSTN via a BRI VIC:
dial-peer voice 23 pots
 destination-pattern 9....
 port 1/1/0
!
This command sets up a local BRI connection:
dial-peer voice 24 pots
 destination-pattern 76003
 port 1/0/0
!
This command sets up a connection to a PBX via Router A:
!
dial-peer voice 26 voip
 destination-pattern 5....
 session target ipv4:12.0.0.1
!
The following commands configure the Ethernet and serial interfaces:
interface Ethernet0/0
 ip address 1.14.122.11 255.255.0.0
 no ip directed-broadcast
!
interface Serial0/0
 ip address 2.0.0.1 255.0.0.0
 no ip directed-broadcast
 no keepalive
!
interface Ethernet0/1
 ip address 11.0.0.2 255.0.0.0
 no ip directed-broadcast
!
interface Serial0/1
 ip address 14.0.0.2 255.0.0.0
 no ip directed-broadcast
 no keepalive
 no fair-queue
The following commands configure the BRI interfaces. Note that only one BRI VIC is installed in a VNM.
!
interface BRI1/0
 no ip address
 no ip directed-broadcast
 isdn switch-type basic-ni1
 isdn twait-disable
 isdn spid1 14085551111 5551111
 isdn spid2 14085551112 5551112
 isdn incoming-voice modem
interface BRI1/1
 no ip address
 no ip directed-broadcast
 isdn switch-type basic-ni1
 isdn twait-disable
 isdn spid1 14085552111 5552111
 isdn spid2 14085552112 5552112
 isdn incoming-voice modem
!
ip classless
ip route 3.0.0.0 255.0.0.0 Ethernet0/1
ip route 3.0.0.0 255.0.0.0 Serial0/1
ip route 223.255.254.0 255.255.255.0 Ethernet0/0
!
!
!
line con 0
 exec-timeout 0 0
 transport input none
line aux 0
line vty 0 4
 login
!
end
vicbri_2600_s2#               
Configuring VoIP for E1 R2 Signalling Example

The following example configures R2 signalling and customizes R2 parameters on controller E1 2 of a Cisco AS5300. In most cases, the same R2 signalling type is configured on each E1 controller.
! Specify the E1 controller that you want to configure with R2 signalling. A controller
! informs the access server how to distribute or provision individual time slots for a
! connected channelized E1 line. You must configure one E1 controller for each E1 line.
! Configure channel associated signalling. The signalling type forwarded by the
! connecting telco switch must match the signalling configured on the Cisco AS5300.
! The country code is ITU by default.
!
controller E1 0
 framing NO-CRC4
 cas-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
 cas-custom 0
!
controller E1 1
 framing NO-CRC4
 clock source line primary
 cas-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled
!
! Customize some of the E1 R2 signalling parameters with the cas-custom channel
! controller configuration command. This example specifies the default R2 settings for
! Brazil.
!
 cas-custom 0
  country brazil use-defaults
  metering
  category 2
  answer-signal group-b 1
!
controller E1 2
!
controller E1 3
!
! Configure voice port parameters. Be sure that the cptone command value is compatible
! with the country code defined by the cas-custom command. In this example, because
! ITU has no specific cptone value defined and uses aLaw E1 R2 signalling, the GB
! cptone command value is used.
!
voice-port 0:0
 cptone GB
!
voice-port 1:0
 cptone BR
 description Brasil Tone
!
! Define the parameters associated with the VoIP dial peer.
!
dial-peer voice 101 voip
 destination-pattern +500..
 session target ipv4:172.14.25.1
!
! Define the parameters associated POTS dial peer.
!
dial-peer voice 8221 pots
 destination-pattern 011822...
 direct-inward-dial
 port 0:0
!
! Configure LAN interfaces.
!
interface Ethernet0
 ip address 172.13.103.33 255.255.0.0
 no ip directed-broadcast
 no ip mroute-cache
 load-interval 30
 no cdp enable
!
interface FastEthernet0
 ip address 173.14.25.100 255.255.0.0
 no ip directed-broadcast
 bandwidth 1000000
 load-interval 30
 duplex full
 hold-queue 75 in
!
no ip classless
ip route 223.255.254.253 255.255.255.255 Ethernet0
!
!
line con 0
 exec-timeout 0 0
 logging synchronous level all
 transport input none
 escape-character BREAK
line aux 0
 rotary 1
 transport preferred none
 transport input all
 flowcontrol hardware
line vty 0 4
 exec-timeout 60 0
 password lab
 login
!
end
Note We strongly recommend that you specify your country default settings. To display a list of supported countries, enter the cas-custom country ? command. The default setting for all countries is ITU.

Configuring VoIP for T1-CAS Example

The following example configures T1 CAS parameters on a Cisco AS5300:
! Enter global configuration mode.
config terminal
! Enter controller configuration mode to configure your controller port. The controller 
ports are labeled 0 through 3 on the Quad T1/PRI and E1/PRI cards.
 controller t1 0
! Enter your telco's framing type.
  framing esf
! Enter the clock source for the line. Configure other lines as clock source secondary
! or internal. Note that only one PRI can be clock source primary and one PRI can be
! clock source secondary
  clock source line primary
! Enter your telco's line code type.
  linecode b8zs
! Configure all channels for E&M, FXS, and SAS analog signalling. Enter 1-24 for T1. 
! If E1, enter 1-31.
! Signalling types include e&m-fgb, e&m-fgd, e&m-immediate-start, fxs-ground-start,
! fxs-loop-start, sas-ground-start, and sas-loop-start.
! You must use the same type of signalling that your central office uses. 
! For E1 using the Anadigicom converter, use cas e&m-fgb signalling.
  cas-group 1 timeslots 1-24  type e&m-fgb dtmf dnis
! Configure each additional controller (there are four). In this example, the
! controller number is 1, instead of 0. The clock source is secondary, instead of
! primary. The cas-group is 2, instead of 1
 controller t1 1
  framing esf
  linecode b8Zs
  clock source line secondary
  cas-group 2 timeslots 1-24 type e&m-fgb
! Configure each additional controller.
 controller T1 2
  clock source internal
  cas-group 0 timeslots 1-24 type e&m-fgd mf ani-dnis
 controller T1 3
  clock source internal
! Enter the dial peer configuration mode to configure a POTS peer.
! Specify destination pattern for this POTS peer.
 dial-peer voice 3070 pots
  destination-pattern +30...
  port 0:1
  prefix 30
! Specify destination pattern, and direct inward dial for each POTS peer.
 dial-peer voice 4080 pots
  destination-pattern +40...
  direct-inward-dial
  port 1:2 
  prefix 40
! Specify the destination pattern and the direct inward dial for the dial peer.
 dial-peer voice 1050 pots
  destination-pattern +10...
  direct-inward-dial
  prefix 50
! Specify the destination pattern and the direct inward dial for the dial peer.
 dial-peer voice 2060 pots
  destination-pattern +20...
  direct-inward-dial
  prefix 60
 dial-peer voice 5050 voip
  answer-address 10...
  destination-pattern +50...
  end
 end
T1/E1 High-Capacity Digital Voice Port Adapters for the Cisco 7200 Series Configuration Examples

This section includes the following configuration examples:

•Routed digits. Shows how to set up a router to collect digits from the PBX/PSTN or from a phone and route the VoIP call based on the digits received.

•FRF.12. Shows how to configure a Cisco 7200 series router to support FRF.12 fragmentation and queueing in a VoIP over Frame-Relay network.

•Gatekeeper. Shows how to configure a Cisco 7200 series router to route VoIP calls by using an H.323 gatekeeper.

•PLAR. Shows how to set up a Cisco 7200 series router for PLAR.

•Trunk connection. Shows how to configure a Cisco 7200 series router for a transparent trunk connection.

•Variable-length digits. Shows how to configure a Cisco 7200 series router to collect variable-length strings of digits PBX/PSTN or phone and route the VoIP call based on the digits received.

•Drop-and-insert. Shows how to configure a Cisco 7200 series router with a 2-port drop-and-insert T1 multiflex trunk voice/WIC (VWIC-2MFT-T1-DI) and a digital T1 high-capacity voice port adapter so that individual DS0 channels are transparently passed between T1 ports without going through a DSP. For example, this allows the directing of some PBX channels to the PSTN for long-distance service, while other channels are compressed for VoIP calls between interoffice sites.

These examples are not necessarily complete configurations. They are designed to illustrate specific tips and techniques, and only the relevant portions of the configurations are shown. Each configuration includes a brief introduction, side-by-side configurations for routers at either end, and explanations of key points.

Routed Digits—Switched VoIP Calls

Figure 30 shows how to set up a Cisco 7200 series router to collect digits from either a PBX/PSTN or a telephone and route a VoIP call based on the digits received. The commands used in the configurations are explained inline. Only relevant sections of the configuration are shown. The example assumes that the IP portion of the network is already in place.

Figure 30 Sample Configuration: Routed Digits
Alpha Router

Beta Router
hostname router-alpha
!
voice-card 1
 codec high
!
dial-peer voice 1 voip
 codec g723r53
 fax-rate 14400
 destination-pattern 5....
 session target  ipv4:192.168.100.1
!
dial-peer voice 2 pots
 destination-pattern 4.... 
 prefix 4 
 port 1/0:1
!
controller T1 1/0
 framing esf
 linecode b8zs
 clock source line
 ds0-group 1 timeslots 1-24 type e&m-wink
!
interface serial 0/0
 ip address 192.168.100.2 255.255.255.0
hostname router-beta
!
voice-card 1
 codec high
!
dial-peer voice 1 voip
 codec g723r53
 fax-rate 14400
 destination-pattern 4....
 session-target ipv4:192.168.100.2
!
dial-peer voice 2 pots
 destination-pattern 5....
 prefix 5
 port 1/0:1
!
controller T1 1/0
 framing esf
 linecode b8zs
 clock source internal 
 ds0-group 1 timeslot 1-24 type e&m-wink
!
interface s0/0
 ip address 192.168.100.1 255.255.255.0
In this configuration, the PBX seizes the T1/E1 to the router, which expects to collect digits from the PBX. Upon collecting those digits, the router tries to match a dial peer to route the call. If the router receives the correct digits, it routes the call according to the configuration of the dial peer.

Here are some key points for consideration:

•The codec command tells the router which types of codecs that can be used on this card type—either high or medium. High-complexity codecs permit only two calls for each DSP. The codecs supported under high complexity are G.711, G.726, G.729, G.729 Annex B, G.728, G.723.1, G.723.1 Annex A, and fax relay. The default is medium complexity, which allows G.711, G.726, G.729 Annex A, G.729 Annex A with Annex B, and fax relay. Medium-complexity codecs permit four calls for each DSP. To change the codec complexity, first remove any configured DS0 group from the T1/E1 controller and then reapply it after the change is complete.

•The ds0-group 1 timeslots 1-24 type e&m-wink command performs the following functions:

–Defines the T1/E1 channels for compressed voice calls.

–Defines the signalling method that the router uses to connect to the PBX or PSTN.

–Automatically creates a voice-port 1/0:1. The numbering for this voice-port is slot/port:ds0-group no. In this configuration, all calls to "4...." or "5...." are routed to any DS0 time slot, although only 1/0:1 is shown. To map individual DS0s, define additional DS0 groups under the T1/E1 controller. Defining additional DS0 groups create individual DS0 voice ports.

•The dial-peer voice commands define the dialing plan within the router. They specify both the remote phone numbers (voip or vofr) and the locally connected phone numbers (pots). The digits in the destination pattern can either be complete numbers or partial numbers with wildcard digits, represented by ".". Each "." represents an individual digit for collection.

FRF.12—Switched VoIP Calls

Figure 31 shows how to configure a Cisco 7200 series router to support FRF.12 fragmentation and queueing in a VoIP over Frame Relay network. FRF.12 is a Frame Relay Forum standard mechanism for fragmenting data packets. This fragmentation helps eliminate the delays that occur when sending voice and data over the same network—large data packets can delay smaller voice packets from being sent into the IP network. FRF.12 is also supported on the Cisco MC3810 and Cisco 7200 routers, which can be used as tandem nodes for VoIP networks.

Note This example shows VoIP over Frame Relay, which is not the same as VoFR. For more information about VoFR, see the "Configuring Voice over Frame Relay."

Figure 31 Sample Configuration: FRF.12 Switched VoIP Calls

The following configuration fragments both the IP and IPX data traffic to 80 bytes, allowing the VoIP traffic to be only minimally delayed on the network. The FRF.12 setup also traffic-shapes the output traffic rate to match the provisioned CIR from the Frame Relay carrier. This ensures that traffic is not dropped or delayed within the Frame Relay network.

Here are some key points for consideration:

•The frame-relay traffic-shaping command enables FRTS on the main interface. Enable it if FRTS will be used on subinterfaces.

•The class cisco_frf12 command tells the interface to use the parameters for FRTS defined in the map class called cisco_frf12.

•The map-class cisco_frf12 grouping of commands defines the rules for FRTS. If per-interface/subinterface parameters must differ, define multiple map classes per router.

•The frame-relay fragment 80 command defines the size of the data or voice packets that FRF.12 fragments. Set the size to about the size of the voice packets or slightly larger. A general rule is 80 bytes for each DS0 of WAN bandwidth. With large quantities of bandwidth and small data frames, the fragment size may need to remain small.

•The frame-relay fair-queue command enables WFQ on a per-PVC basis to ensure that voice traffic gets priority over data traffic.
Alpha Router

Beta Router
hostname router-alpha
!
ipx routing
! 
card type t1 1
!
dspint DSPfarm 1/0 codec high L30
!
controller T1 1/0
 framing esf 
 linecode b8zs
 clock source line 
 ds0-group 1 timeslot 1-24 type e&m-wink
!
dial-peer voice 1 voip
 dtmf-relay  h245-alpha
 codec g723r53
 destination-pattern 5....
 session target ipv4:192.168.100.2
!
dial-peer voice 2 pots
 destination-pattern 4....  
 prefix 4
 port 1/0:1
!
interface serial 0/0
 encapsulation frame-relay
 frame-relay traffic-shaping
! 
interface serial 0/0.1       point-to-point
 ip address 192.168.100.1 255.255.255.0
 ipx network ABCD
 frame-relay interface-dlci 100
 class cisco_frf12
!
map-class frame-relay cisco_frf12
frame-relay voice bandwidth 42000
frame-relay fragment 80
no frame-relay adaptive-shaping
frame-relay cir 32000
frame-relay bc 1000
frame-relay mincir 64000
frame-relay fair-queue
hostname router-beta
!
ipx routing
!
card type t1 1
 codec high 
!
dspint DSPfarm 1/0 codec high L30
controller T1 1/0
 framing esf
 linecode b8zs
 clock source line
 ds0-group 1 timeslot 1-24 type e&m-wink
!
dial-peer voice 1 voip
 dtmf-relay h245-alpha
 codec g723r53
 destination-pattern 4....
 session target ipv4:192.168.100.2
!
dial-peer voice 2 pots
 destination-pattern 5....
 prefix 5
 port 1/0:1
!
interface serial 0/0
 encapsulation frame-relay
 frame-relay traffic-shaping
!
interface serial 0/0.1 point-to-point
 ip address 192.168.100.2 255.255.255.0
 ipx network ABCD
 frame-relay interface-dlci 101
 class cisco_frf12
!
map-class frame-relay cisco_frf12
frame-relay voice bandwidth 42000
frame-relay fragment 80
no frame-relay adaptive-shaping
frame-relay cir 64000
frame-relay bc 1000
frame-relay mincir 64000
frame-relay fair-queue
Routing Calls Through an H.323 Gatekeeper

Note With the introduction of Cisco IOS Release 12.0(5)T and subsequent releases, Cisco VoIP gateways support H.323v2 (H.323 Version 2), which is backwards compatible with systems running H.323 Version 1. However, H.323 Version 2 features do not interoperate with H.323 Version 1 features in Cisco IOS releases prior to 11.3(9)NA or 12.0(3)T. Earlier Cisco IOS versions contain H.323 Version 1 software that does not support protocol messages with an H.323 Version 2 protocol identifier. All systems must be running either Cisco IOS Release 11.3(9)NA and later or Cisco IOS Release 12.0(3)T and later releases to interoperate with H.323 Version 2. Gateway Resource Availability Indication (RAI) messages are currently not supported on the Cisco 7200 series. (These are messages that are sent to the Gatekeeper to inform it about the status of a Gateway DSP or DS0 availability.)

Figure 32 shows how to configure a Cisco 7200 series router to route VoIP calls through an H.323 gatekeeper. This setup shows calls being routed from a gateway in Zone-Alpha, through the gatekeeper, to a gateway in Zone-Beta.

Figure 32 Sample Configuration: Routing Calls Through an H.323 Gatekeeper
Gatekeeper
hostname router-gatekeeper
!
gatekeeper
zone local alpha alpha.com
zone local beta beta.com
no use-proxy alpha.com remote-zone beta.com
no use-proxy beta.com remote-zone alpha.com
zone prefix router-alpha 4....
zone prefix router-beta 5....
no shutdown
!
interface ethernet 0/0
ip address 10.1.1.3 255.255.255.0
Alpha Router

Beta Router
hostname router-alpha
!
card type t1 1
!
dspint DSPfarm 1/0
<