Table Of Contents

Configuring Voice Ports

Voice Port Configuration Overview

Telephony Signaling Interfaces

FXS and FXO Interfaces

E&M Interfaces

Analog Voice Ports Configuration Task List

Prerequisites for Configuring Analog Voice Ports

Preparing to Configure Analog Voice Ports

Configuring Platform-Specific Analog Voice Hardware

Cisco 800 Series Routers

Cisco 1750 Modular Router

Cisco 2600 Series and Cisco 3600 Series Routers

Cisco MC3810 Multiservice Concentrator

Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High-Performance Compression Modules

Configuring Basic Parameters on Analog FXO, FXS, or E&M Voice Ports

Configuring Analog Telephone Connections on Cisco 803 and 804 Routers

Verifying Analog Telephone Connections on Cisco 803 and 804 Routers

Troubleshooting Tip for Cisco 803 and 804 Routers

Configuring Digital Voice Ports

Prerequisites for Configuring Digital Voice Ports

Preparing Information to Configure Digital Voice Ports

Platform-Specific Digital Voice Hardware

Cisco 2600 Series and Cisco 3600 Series Routers

Cisco MC3810 Multiservice Concentrator

Cisco AS5300 Universal Access Server

Cisco AS5800 Universal Access Server

Cisco 7200 and Cisco 7500 Series Routers

Configuring Basic Parameters on Digital T1/E1 Voice Ports

Configuring Codec Complexity for Digital T1/E1 Voice Ports

Configuring Controller Settings for Digital T1/E1 Voice Ports

Configuring Basic Voice Port Parameters for Digital T1/E1 Voice Ports

Fine-Tuning Analog and Digital Voice Ports

Auto Cut-Through Command

Bit Modification Commands for Digital Voice Ports

Calling Number Outbound Commands

Disconnect Supervision Commands

FXO Supervisory Disconnect Tone Commands

Timeouts Commands

Timing Commands

DTMF Timer Inter-Digit Command for Cisco AS5300 Access Servers

Voice Activity Detection Commands Related to Voice-Port Configuration Mode

Voice Quality Tuning Commands

Delay in Voice Networks

Jitter Adjustment

Echo Adjustment

Voice Level Adjustment

Verifying Analog and Digital Voice-Port Configurations

show voice port summary Command Examples

Cisco 3640 Router Analog Voice Port

Cisco MC3810 Multiservice Concentrator Digital Voice Port

show voice port Command Examples

Cisco 3600 Series Router Analog E&M Voice Port

Cisco 3600 Series Router Analog FXS Voice Port

Cisco 3600 Series Router Digital E&M Voice Port

Cisco AS5300 Universal Access Server T1 CAS Voice Port

Cisco 7200 Series Router Digital E&M Voice Port

show controller Command Examples

Cisco 3600 Series Router T1 Controller

Cisco MC3810 Multiservice Concentrator E1 Controller

Cisco AS5800 Universal Access Server T1 Controller

show voice dsp Command Examples

show voice call summary Command Examples

Cisco MC3810 Multiservice Concentrator Analog Voice Port

Cisco 3600 Series Router Digital Voice Port

show call active voice Command Example

show call history voice Command Example

Troubleshooting Analog and Digital Voice Port Configurations

Troubleshooting Chart

Voice Port Testing Commands

Detector-Related Function Tests

Loopback Function Tests

Tone Injection Tests

Relay-Related Function Tests

Fax/Voice Mode Tests

Configuring Voice Ports

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Voice ports are found at the intersections of packet-based networks and traditional telephony networks, and they facilitate the passing of voice and call signals between the two networks. Physically, voice ports connect a router or access server to a line from a circuit-switched telephony device in a PBX or the public switched telephone network (PSTN).

Basic software configuration for voice ports describes the type of connection being made and the type of signaling to take place over this connection. Additional commands provide fine-tuning for voice quality, enable special features, and specify parameters to match those of proprietary PBXs.

This chapter includes the following sections:

•Voice Port Configuration Overview

•Analog Voice Ports Configuration Task List

•Configuring Digital Voice Ports

•Fine-Tuning Analog and Digital Voice Ports

•Verifying Analog and Digital Voice-Port Configurations

•Troubleshooting Analog and Digital Voice Port Configurations

Not all voice-port commands are covered in this chapter. Some are described in the "Configuring Trunk Connections and Conditioning Features" chapter or the "Configuring ISDN Interfaces for Voice" chapter in this configuration guide. The voice-port configuration commands included in this chapter are fully documented in the Cisco IOS Voice, Video, and Fax Command Reference.

To identify the hardware platform or software image information associated with a feature in this chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to the software release notes for a specific release. For more information, see the "Identifying Supported Platforms" section in the "Using Cisco IOS Software" chapter.

Voice Port Configuration Overview

Voice ports on routers and access servers emulate physical telephony switch connections so that voice calls and their associated signaling can be transferred intact between a packet network and a circuit-switched network or device.

For a voice call to occur, certain information must be passed between the telephony devices at either end of the call, such as the devices' on-hook status, the line's availability, and whether an incoming call is trying to reach a device. This information is referred to as signaling, and to process it properly, the devices at both ends of the call segment (that is, those directly connected to each other) must use the same type of signaling.

The devices in the packet network must be configured to convey signaling information in a way that the circuit-switched network can understand. They must also be able to understand signaling information received from the circuit-switched network. This is accomplished by installing appropriate voice hardware in the router or access server and by configuring the voice ports that connect to telephony devices or the circuit-switched network.

The illustrations below show examples of voice port usage.

•In Figure 10, one voice port connects a telephone to the wide-area network (WAN) through the router.

•In Figure 11, one voice port connects to the PSTN and another to a telephone; the router acts like a small PBX.

•Figure 12 shows how two PBXs can be connected over a WAN to provide toll bypass.

Figure 10 Telephone to WAN

Figure 11 Telephone to PSTN

Figure 12 PBX-to-PBX over a WAN

Cisco provides a variety of Cisco IOS commands for flexibility in programming voice ports to match the physical attributes of the voice connections that are being made. Some of these connections are made using analog means of transmission, while others use digital transmission. Table 4 shows the analog and digital voice-port connection support of the router platforms discussed in this chapter.

Table 4 Analog and Digital Voice-port Support on Cisco Routers and Access Servers 

Platform	Analog	Digital
Cisco 803 and 804	Yes	No
Cisco 1750	Yes	No
Cisco 2600 series	Yes	Yes
Cisco 3600 series	Yes	Yes
Cisco MC3810	Yes	Yes
Cisco AS5300	No	Yes
Cisco AS5800	No	Yes
Cisco 7200 series	No	Yes
Cisco 7500 series	No	Yes

Telephony Signaling Interfaces

Voice ports on routers and access servers physically connect the router or access server to telephony devices such as telephones, fax machines, PBXs, and PSTN central office (CO) switches. These devices may use any of several types of signaling interfaces to generate information about on-hook status, ringing, and line seizure.

The router's voice-port hardware and software need to be configured to transmit and receive the same type of signaling being used by the device with which they are interfacing so that calls can be exchanged smoothly between the packet network and the circuit-switched network.

The signaling interfaces discussed in this chapter include foreign exchange office (FXO), foreign exchange station (FXS), and receive and transmit (E&M), which are types of analog interfaces. Some digital connections emulate FXO, FXS, and E&M interfaces, and they are discussed in the second half of this chapter. It is important to know which signaling method the telephony side of the connection is using, and to match the router configuration and voice interface hardware to that signaling method.

The next three illustrations show how the different signaling interfaces are associated with different uses of voice ports. In Figure 13, FXS signaling is used for end-user telephony equipment, such as a telephone or fax machine. Figure 14 shows an FXS connection to a telephone and an FXO connection to the PSTN at the far side of a WAN; this might be a telephone at a local office going over a WAN to a router at headquarters that connects to the PSTN. In Figure 15, two PBXs are connected across a WAN by E&M interfaces. This illustrates the path over a WAN between two geographically separated offices in the same company.

Figure 13 FXS Signaling Interfaces

Figure 14 FXS and FXO Signaling Interfaces

Figure 15 E&M Signaling Interfaces

FXS and FXO Interfaces

An FXS interface connects the router or access server to end-user equipment such as telephones, fax machines, or modems. The FXS interface supplies ring, voltage, and dial tone to the station and includes an RJ-11 connector for basic telephone equipment, keysets, and PBXs.

An FXO interface is used for trunk, or tie line, connections to a PSTN CO or to a PBX that does not support E&M signaling (when local telecommunications authority permits). This interface is of value for off-premise station applications. A standard RJ-11 modular telephone cable connects the FXO voice interface card to the PSTN or PBX through a telephone wall outlet.

FXO and FXS interfaces indicate on-hook or off-hook status and the seizure of telephone lines by one of two access signaling methods: loop start or ground start. The type of access signaling is determined by the type of service from the CO; standard home telephone lines use loop start, but business telephones can order ground start lines instead.

Loop-start is the more common of the access signaling techniques. When a handset is picked up (the telephone goes off-hook), this action closes the circuit that draws current from the telephone company CO and indicates a change in status, which signals the CO to provide dial tone. An incoming call is signaled from the CO to the handset by sending a signal in a standard on/off pattern, which causes the telephone to ring.

Loop-start has two disadvantages, however, that usually are not a problem on residential telephones but that become significant with the higher call volume experienced on business telephones. Loop-start signaling has no means of preventing two sides from seizing the same line simultaneously, a condition known as glare. Also, loop start signaling does not provide switch-side disconnect supervision for FXO calls. The telephony switch (the connection in the PSTN, another PBX, or key system) expects the router's FXO interface, which looks like a telephone to the switch, to hang up the calls it receives through its FXO port. However, this function is not built into the router for received calls; it only operates for calls originating from the FXO port.

Another access signaling method used by FXO and FXS interfaces to indicate on-hook or off-hook status to the CO is ground start signaling. It works by using ground and current detectors that allow the network to indicate off-hook or seizure of an incoming call independent of the ringing signal and allow for positive recognition of connects and disconnects. For this reason, ground start signaling is typically used on trunk lines between PBXs and in businesses where call volume on loop start lines can result in glare. See the "Disconnect Supervision Commands" section and "FXO Supervisory Disconnect Tone Commands" section for voice port commands that configure additional recognition of disconnect signaling.

In most cases, the default voice port command values are sufficient to configure FXO and FXS voice ports.

E&M Interfaces

Trunk circuits connect telephone switches to one another; they do not connect end-user equipment to the network. The most common form of analog trunk circuit is the E&M interface, which uses special signaling paths that are separate from the trunk's audio path to convey information about the calls. The signaling paths are known as the E-lead and the M-lead. The name E&M is thought to derive from the phrase Ear and Mouth or rEceive and transMit although it could also come from Earth and Magnet. The history of these names dates back to the days of telegraphy, when the CO side had a key that grounded the E circuit, and the other side had a sounder with an electromagnet attached to a battery. Descriptions such as Ear and Mouth were adopted to help field personnel determine the direction of a signal in a wire. E&M connections from routers to telephone switches or to PBXs are preferable to FXS/FXO connections because E&M provides better answer and disconnect supervision.

Like a serial port, an E&M interface has a data terminal equipment/data communications equipment (DTE/DCE) type of reference. In the telecommunications world, the trunking side is similar to the DCE, and is usually associated with CO functionality. The router acts as this side of the interface. The other side is referred to as the signaling side, like a DTE, and is usually a device such as a PBX. Five distinct physical configurations for the signaling part of the interface (Types I-V) use different methods to signal on-hook/off-hook status, as shown in Table 5. Cisco voice implementation supports E&M Types I, II, III, and V.

The physical E&M interface is an RJ-48 connector that connects to PBX trunk lines, which are classified as either two-wire or four-wire. This refers to whether the audio path is full duplex on one pair of wires (two-wire) or on two pair of wires (four-wire). A connection may be called a four-wire E&M circuit although it actually has six to eight physical wires. It is an analog connection although an analog E&M circuit may be emulated on a digital line. For more information on digital voice port configuration of E&M signaling, see the "DS0 Groups on Digital T1/E1 Voice Ports" section.

PBXs built by different manufacturers can indicate on-hook/off-hook status and telephone line seizure on the E&M interface by using any of three types of access signaling that are as follows:

•Immediate-start is the simplest method of E&M access signaling. The calling side seizes the line by going off-hook on its E-lead and sends address information as dual-tone multifrequency (DTMF) digits (or as dialed pulses on Cisco 2600 series routers and Cisco 3600 series routers) following a short, fixed-length pause.

•Wink-start is the most commonly used method for E&M access signaling, and is the default for E&M voice ports. Wink-start was developed to minimize glare, a condition found in immediate-start E&M, in which both ends attempt to seize a trunk at the same time. In wink-start, the calling side seizes the line by going off-hook on its E-lead, then waits for a short temporary off-hook pulse, or "wink," from the other end on its M-lead before sending address information. The switch interprets the pulse as an indication to proceed and then sends the dialed digits as DTMF or dialed pulses.

•In delay-dial signaling, the calling station seizes the line by going off-hook on its E-lead. After a timed interval, the calling side looks at the status of the called side. If the called side 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.

Table 5 E&M Wiring and Signaling Methods

E&M Type	E-Lead Configuration	M-Lead Configuration	Signal Battery Lead Configuration	Signal Ground Lead Configuration
I	Output, relay to ground	Input, referenced to ground	—	—
II	Output, relay to SG	Input, referenced to ground	Feed for M, connected to -48V	Return for E, galvanically isolated from ground
III	Output, relay to ground	Input, referenced to ground	Connected to -48V	Connected to ground
V	Output, relay to ground	Input, referenced to -48V	—	—

Analog Voice Ports Configuration Task List

Analog voice port interfaces connect routers in packet-based networks to analog two-wire or four-wire analog circuits in telephony networks. Two-wire circuits connect to analog telephone or fax devices, and four-wire circuits connect to PBXs. Typically, connections to the PSTN CO are made with digital interfaces.

This section describes how to configure analog voice ports and covers the following topics:

•Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High-Performance Compression Modules

•Configuring Basic Parameters on Analog FXO, FXS, or E&M Voice Ports

•Configuring Analog Telephone Connections on Cisco 803 and 804 Routers

Three other sections later in the chapter provide help with fine-tuning and troubleshooting:

•Fine-Tuning Analog and Digital Voice Ports

•Verifying Analog and Digital Voice-Port Configurations

•Troubleshooting Analog and Digital Voice Port Configurations

Prerequisites for Configuring Analog Voice Ports

•Obtain two- or four-wire line service from your service provider or from a PBX.

•Complete your company's dial plan.

•Establish a working telephony network based on your company's dial plan.

•Install at least one other network module or WAN interface card to provide the connection to the network LAN or WAN.

•Establish a working IP and Frame Relay or ATM network. For more information about configuring IP, refer to the Cisco IOS IP Configuration Guide, Release 12.2.

•Install appropriate voice processing and voice interface hardware on the router. See the "Configuring Platform-Specific Analog Voice Hardware" section.

Preparing to Configure Analog Voice Ports

Before configuring an analog voice port, assemble the following information about the telephony connection that the voice port will be making. If connecting to a PBX, it is important to understand the PBX's wiring scheme and timing parameters. This information should be available from your PBX vendor or the reference manuals that accompany your PBX.

•Telephony signaling interface: FXO, FXS, or E&M

•Locale code (usually the country) for call progress tones

•If FXO, type of dialing: DTMF (touch-tone) or pulse

•If FXO, type of start signal: loop-start or ground-start

•If E&M, type: I, II, III, or V

•If E&M, type of line: two-wire or four-wire

•If E&M, type of start signal: wink, immediate, delay-dial

Table 6 should help you determine which hardware and configuration instructions are appropriate for your situation. Table 7 shows slot and port numbering, which differs for each of the voice-enabled routers. More current information may be available in the release notes that accompany the Cisco IOS software you are using.

Table 6 Analog Voice Port Configurations

Telephony Signaling Interface	Router Platform	Voice Hardware Required	Section Containing Voice Port Configuration Instructions
End user: telephone or fax	Cisco 803 Cisco 804	—	"Configuring Analog Telephone Connections on Cisco 803 and 804 Routers"
FXO	Cisco 1750 Cisco 2600 series Cisco 3600 series	VIC-2FXO, VIC-2FXO-EU	"Configuring Basic Parameters on Analog FXO, FXS, or E&M Voice Ports"
	Cisco MC3810	MC3810-AVM6 MC3810-APM-FXO
FXS	Cisco 1750 Cisco 2600 series Cisco 3600 series	VIC-2FXS
	Cisco MC3810	MC3810-AVM6 MC3810-APM-FXS
E&M	Cisco 1750 Cisco 2600 series Cisco 3600 series	VIC-2E/M
	Cisco MC3810	MC3810-AVM6 MC3810-APM-EM

Table 7 Analog Voice Slot/Port Designations

Router Platform	Voice Hardware	Chassis Slot Numbers	Voice NM Slot Numbers	Voice Port Numbers
Cisco 803, 804	Analog POTS	—	—	—
Cisco 1750	Analog VIC	0 to 1	—	0 to 1
Cisco 2600 series	Voice/fax network module with two-port VIC	Varies, based on router	1	0 to 1
Cisco 3600 series	Voice/fax network module with two-port voice over interface cards (VICs)	1	3620: 0 to 1 3640: 0 to 3 3660: 1 to 6	0 to 1
Cisco MC3810	Analog voice module (AVM)	1	—	1 to 6

Configuring Platform-Specific Analog Voice Hardware

This section describes the general types of analog voice port hardware available for the router platforms included in this chapter:

•Cisco 800 Series Routers

•Cisco 1750 Modular Router

•Cisco 2600 Series and Cisco 3600 Series Routers

•Cisco MC3810 Multiservice Concentrator

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Note For current information about supported hardware, see the release notes for the platform and Cisco IOS release being used.

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Cisco 800 Series Routers

Cisco 803 and Cisco 804 routers support data and voice applications. The data applications on these routers are implemented through the ISDN port, and the voice applications are implemented with ISDN Basic Rate Interface (BRI) through the telephone ports. If a Cisco 803 or 804 router is being used, connect two devices, such as an analog touch-tone telephone, fax machine, or modem through two fixed telephone ports, the gray PHONE 1 and PHONE 2 ports that have RJ-11 connectors. Each device is connected to basic telephone services through the ISDN line.

For more information, refer to the Cisco 800 Series Routers Hardware Installation Guide.

Cisco 1750 Modular Router

The Cisco 1750 modular router provides Voice over IP (VoIP) functionality and can carry voice traffic (for example, telephone calls and faxes) over an IP network. To make a voice connection, the router must have a supported VIC installed. The Cisco 1750 router supports two slots for either WAN interface cards (WICs) or VICs and supports one VIC-only slot. For analog connections, two-port VICs are available to support FXO, FXS, and E&M signaling. VICs provide direct connections to telephone equipment (analog phones, analog fax machines, key systems, or PBXs) or to a PSTN.

For more information, refer to the Cisco 1750 Voice-over-IP Quick Start Guide.

Cisco 2600 Series and Cisco 3600 Series Routers

The Cisco 2600 and 3600 series routers are modular, multifunction platforms that combine dial access, routing, local area network-to-local area network (LAN) services, and multiservice integration of voice, video, and data in the same device.

Voice network modules installed in Cisco 2600 series or Cisco 3600 series routers convert telephone voice signals into data packets that can be transmitted over an IP network. The voice network modules have no connectors; VICs installed in the network modules provide connections to the telephone equipment or network. VICs work with existing telephone and fax equipment and are compatible with H.323 standards for audio and video conferencing.

The Cisco 2600 series router can house one network module. In the Cisco 3600 series, the Cisco 3620 router has slots for up to two network modules; the Cisco 3640 router has slots for up to four network modules; and the Cisco 3660 router has slots for up to six network modules. (Typically, one of the slots is used for LAN connectivity.)

For analog telephone connections, low-density voice/fax network modules that contain either one or two VIC slots are installed in the network module slots. Each VIC is specific to a particular telephone signaling interface (FXS, FXO, or E&M); therefore, the VIC determines the type of signaling on that module.

For more information, refer to the following:

•Cisco 2600 Series Hardware Installation Guide

•Cisco 3600 Series Hardware Installation Guide

•Cisco Network Module Hardware Installation Guide

Cisco MC3810 Multiservice Concentrator

To support analog voice circuits, a Cisco MC3810 multiservice concentrator must be equipped with an AVM, which supports six analog voice ports. By installing specific signaling modules known as analog personality modules (APMs), the analog voice ports may be equipped for the following signaling types in various combinations: FXS, FXO, and E&M. For FXS, the analog voice ports use an RJ-11 connector interface to connect to analog telephones or fax machines (two-wire) or to a key system (four-wire). For FXO, the analog voice ports use an RJ-11 physical interface to connect to a CO trunk. For E&M connections, the analog voice ports use an RJ-1CX physical interface to connect to an analog PBX (two-wire or four-wire).

Optional high-performance voice compression modules (HCMs) can replace standard voice compression modules (VCMs) to operate according to the voice compression coding algorithm (codec) specified when the Cisco MC3810 concentrator is configured. The HCM2 provides four voice channels at high codec complexity and eight channels at medium complexity. The HCM6 provides 12 voice channels at high complexity and 24 channels at medium complexity. One or two HCMs can be installed in a Cisco MC3810 multiservice concentrator, but an HCM may not be combined with a VCM in one chassis.

For more information, refer to the Cisco MC3810 Multiservice Concentrator Hardware Installation Guide.

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Note For current information about supported hardware, see the release notes for the platform and Cisco IOS release being used.

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Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High-Performance Compression Modules

The term codec stands for coder-decoder. A codec is a particular method of transforming analog voice into a digital bit stream (and vice versa) and also refers to the type of compression used. Several different codecs have been developed to perform these functions, and each one is known by the number of the International Telecommunication Union-Telecommunication Standardization Sector (ITU-T) standard in which it is defined. For example, two common codecs are the G.711 and the G.729 codecs. The various codecs use different algorithms to encode analog voice into digital bit-streams and have different bit rates, frame sizes, and coding delays associated with them. The codecs also differ in the amount of perceived voice quality they achieve. Specialized hardware and software in the digital signal processors (DSPs) perform codec transformation and compression functions, and different DSPs may offer different selections of codecs.

Select the same type of codec as the one that is used at the other end of the call. For instance, if a call was coded with a G.729 codec, it must be decoded with a G.729 codec. Codec choice is configured on dial peers. For more information, see the "Configuring Dial Plans, Dial Peers, and Digit Manipulation" chapter in this configuration guide.

Codec complexity refers to the amount of processing power that a codec compression technique requires: some require more processing power than others. Codec complexity affects call density, which is the number of calls that can take place on the DSP interfaces, which can be HCMs, port adapter DSP farms, or voice cards, depending on the type of router (in this case, the Cisco MC3810 multiservice concentrator). The greater the codec complexity, the fewer the calls that can be handled.

Codec complexity is either medium or high. The difference between medium- and high-complexity codecs is the amount of CPU power necessary to process the algorithm and, therefore, the number of voice channels that can be supported by a single DSP. All medium-complexity codecs can also be run in high-complexity mode, but fewer (usually half as many) channels will be available per DSP.

For details on the number of calls that can be handled simultaneously using each of the codec standards, refer to the entries for the codec and codec complexity commands in the Cisco IOS Voice, Video, and Fax Command Reference.

On a Cisco MC3810 concentrator, only a single codec complexity setting is used, even when two HCMs are installed. The value that is specified in this task affects the choice of codecs available when the codec dial-peer configuration command is configured. See the "Configuring Dial Plans, Dial Peers, and Digit Manipulation" chapter in this configuration guide.

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Note On the Cisco MC3810 with high-performance compression modules, check the DSP voice channel activity with the show voice dsp command. If any DSP voice channels are in the busy state, the codec complexity cannot be changed. When all the DSP channels are in the idle state, changes can be made to the codec complexity selection.

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To configure codec complexity on the Cisco MC3810 multiservice concentrator using HCMs, use the following commands beginning in privileged EXEC mode:

	Command	Purpose
Step 1	Router# show voice dsp	Checks the DSP voice channel activity. If any DSP voice channels are in the busy state, the codec complexity cannot be changed. When all the DSP channels are in the idle state, continue to Step 2.
Step 2	Router# configure terminal	Enters global configuration mode.
Step 3	Router(config)# voice-card 0	Enters voice-card configuration mode and specifies voice card 0.
Step 4	Router(config-voicecard)# codec complexity {high | medium}	(For analog voice ports) Specifies codec complexity based on the codec standard being used. This setting restricts the codecs available in dial peer configuration. All voice cards in a router must use the same codec complexity setting. The keywords are as follows: •high—Specifies two voice channels encoded in any of the following formats: G.711ulaw, G.711alaw, G.723.1(r5.3), G.723.1 Annex A(r5.3), G.723.1(r6.3), G.723.1 Annex A(r6.3), G.726(r16), G.726(r24), G.726(r32), G.728, G.729, G.729 Annex B, and fax relay. •medium—(default) Specifies four voice channels encoded in any of the following formats: G.711ulaw, G.711alaw, G.726(r16), G.726(r24), G.726(r32), G.729 Annex A, G.729 Annex B with Annex A, and fax relay. Note If two HCMs are installed, this command configures both HCMs at once.

Configuring Basic Parameters on Analog FXO, FXS, or E&M Voice Ports

This section describes commands for basic analog voice port configuration. All the data recommended in the "Preparing to Configure Analog Voice Ports" section should be gathered before starting this procedure.

If configuring a Cisco MC3810 multiservice concentrator that has HCMs, codec complexity should also be configured, following the steps in the "Configuring Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High-Performance Compression Modules" section.

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Note If you have a Cisco MC3810 multiservice concentrator or Cisco 3660 router, the compand-type a-law command must be configured on the analog ports only. The Cisco 2660, 3620, and 3640 routers do not require the configuration of th compand-type a-law command, however, if you request a list of commands, the compand-type a-law command will display.

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In addition to the basic voice port parameters described in this section, there are commands that allow voice port configurations to be fine tuned. In most cases, the default values for fine-tuning commands are sufficient for establishing FXO and FXS voice port configurations. E&M voice ports are more likely to require some configuration. If it is necessary to change some of the voice port values to improve voice quality or to match parameters on proprietary PBXs to which you are connecting, use the commands in the current section and also in the "Fine-Tuning Analog and Digital Voice Ports" section.

After the voice-port has been configured, make sure that the ports are operational by following the steps described in the following sections:

•Verifying Analog and Digital Voice-Port Configurations

•Troubleshooting Analog and Digital Voice Port Configurations

For more information on these and other voice port commands, see the Cisco IOS Voice, Video, and Fax Command Reference.

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Note The commands, keywords, and arguments that you are able to use may differ slightly from those presented here, based on your platform, Cisco IOS release, and configuration. When in doubt, use Cisco IOS command help (command ?) to determine the syntax choices that are available.

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To configure basic analog voice port parameters on Cisco 1750, Cisco 2600 series, Cisco 3600 series, and Cisco MC3810 routers, use the following commands beginning in global configuration mode:

	Command	Purpose
Step 1	Cisco 1750 and MC3810 Router(config)# voice-port slot/port Cisco 2600 and 3600 series Router(config)# voice-port slot/subunit/port	Enters voice-port configuration mode. The arguments are as follows: •slot—Specifies the number of the router slot where the voice network module is installed (Cisco 2600 and Cisco 3600 series routers) or the router slot number where the analog voice module is installed (Cisco MC3810 multiservice concentrator). •port—Indicates the voice port. Valid entries are 0 or 1. •subunit—Specifies the location of the VIC. Note The slash must be entered between slot and port. Valid entries vary by router platform; see Table 7 or enter the show voice port summary command for available values.
Step 2	FXO or FXS Router(config-voiceport)# signal {loop-start | ground-start}	Selects the access signaling type to match that of the telephony connection you are making. The keywords are as follows: •loop-start—(default) Uses a closed circuit to indicate off-hook status; used for residential loops. •ground-start—Uses ground and current detectors; preferred for PBXs and trunks.
	E&M Router(config-voiceport)# signal {wink-start | immediate-start | delay-dial}	The keywords are as follows: •wink-start—(default) Indicates that the calling side seizes the line, then waits for a short off-hook wink from the called side before proceeding. •immediate-start—Indicates that the calling side seizes the line and immediately proceeds; used for E&M tie trunk interfaces. •delay-dial—Indicates that the calling side seizes the line and waits, then checks to determine whether the called side is on-hook before proceeding; if not, it waits until the called side is on-hook before sending digits. Used for E&M tie trunk interfaces. Note Configuring the signal keyword for one voice port on a Cisco 2600 or 3600 series router VIC changes the signal value for both ports on the VIC.
Step 3	Router(config-voiceport)# cptone locale	Selects the two-letter locale for the voice call progress tones and other locale-specific parameters to be used on this voice port. Cisco routers comply with the ISO 3166 locale name standards. To see valid choices, enter a question mark (?) following the cptone command. The default is us.
Step 4	Router(config-voiceport)# dial-type {dtmf | pulse}	(FXO only) Specifies the dialing method for outgoing calls.
Step 5	Router(config-voiceport)# operation {2-wire | 4-wire}	(E&M only) Specifies the number of wires used for voice transmission at this interface (the audio path only, not the signaling path). The default is 2-wire.
Step 6	Router(config-voiceport)# type {1 | 2 | 3 | 5}	(E&M only) Specifies the type of E&M interface to which this voice port is connecting. See Table 5 for an explanation of E&M types. The default is 1.
Step 7	Cisco 1750 Router and 2600 and 3600 Series Routers Router(config-voiceport)# ring frequency {25 | 50} Cisco MC3810 Multiservice Concentrator Router(config-voiceport)# ring frequency {20 | 30}	(FXS only) Selects the ring frequency, in hertz, used on the FXS interface. This number must match the connected telephony equipment and may be country-dependent. If not set properly, the attached telephony device may not ring or it may buzz. The keyword default is 25 on the Cisco 1750 router, 2600 and 3600 series routers; and 20 on the Cisco MC3810 multiservice concentrator.
Step 8	Router(config-voiceport)# ring number number	(FXO only) Specifies the maximum number of rings to be detected before an incoming call is answered by the router. The default is 1.
Step 9	Router(config-voiceport)# ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07 | pattern08 | pattern09 | pattern10 | pattern11 | pattern12] | [define pulse interval]}	(FXS only) Specifies an existing pattern for ring, or it defines a new one. Each pattern specifies a ring-pulse time and a ring-interval time. The keywords and arguments are as follows: •pattern01 through pattern12 name pre-set ring cadence patterns. Enter ring cadence ? to see ring pattern explanations. •define pulse interval specifies a user-defined pattern: pulse is a number (one or two digits, from 1 to 50) specifying ring pulse (on) time in hundreds of milliseconds, and interval is a number (one or two digits from 1 to 50) specifying ring interval (off) time in hundreds of milliseconds. The default is the pattern specified by the cptone locale that has been configured.
Step 10	Router(config-voiceport)# description string	Attaches a text string to the configuration that describes the connection for this voice port. This description appears in various displays and is useful for tracking the purpose or use of the voice port. The string argument is a character string from 1 to 255 characters in length. The default is that there is no text string (describing the voice port) attached to the configuration.
Step 11	Router(config-voiceport)# no shutdown	Activates the voice port. If a voice port is not being used, shut the voice port down with the shutdown command.

Configuring Analog Telephone Connections on Cisco 803 and 804 Routers

Multiple devices (analog telephone, fax machine, or modem) can be connected to a Cisco 803 or 804 telephone port. The number of devices that can be connected depends on the ringer equivalent number (REN) of each device that is to be connected. (The REN can usually be found on the bottom of a device.) The REN of the router telephone port is 5, so if the REN of each device to be connected is 1, a maximum of five devices can be connected to that particular telephone port.

These routers support touch-tone analog telephones only; they do not support rotary telephones.

To configure standard features for analog telephone connections on Cisco 803 and 804 routers, use the following commands in global configuration mode:

	Command	Purpose
Step 1	Router(config)# pots country country	Specifies the country to use for country-specific default settings for physical characteristics. Enter pots country ? for a list of supported countries and the codes to enter. A default country is not defined.
Step 2	Router(config)# pots line-type {type1 | type2 | type3}	(Optional) Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router. The keywords are as follows: •type1—Specifies the resistance used for the POTS connection, typically 600 ohms. •type2—Specifies the resistance used for the POTS connection, typically 900 ohms. •type3—Specifies the resistance used for the POTS connection, typically 300/400 ohms. The default depends on the country chosen in the pots country command.
Step 3	Router(config)# pots dialing-method {overlap | enblock}	(Optional) Specifies how the router collects and sends digits dialed on connected telephones, fax machines, or modems. The keywords are as follows: •overlap—Tells the router to send each digit dialed in a separate message. •enblock—Tells the router to collect all digits dialed and to send the digits in one message. The default depends on the country chosen in the pots country command.
Step 4	Router(config)# pots disconnect-supervision {osi | reversal}	(Optional) Specifies how the router notifies the connected telephones, fax machines, or modems when the calling party has disconnect. The keywords are as follows: •osi—(open switching interval) Specifies the duration for which DC voltage applied between tip and ring conductors of a telephone port is removed. •reversal—Specifies the polarity reversal of the tip and ring conductors of a telephone port. The default depends on the country chosen in the pots country command.
Step 5	Router(config)# pots encoding {alaw | ulaw}	(Optional) Specifies the pulse code modulation (PCM) encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router. The keywords are as follows: •alaw—Specifies the ITU-T PCM encoding scheme used to represent analog voice samples as digital values. •ulaw—Specifies the North American PCM encoding scheme used to represent analog voice samples as digital values. The default depends on the country chosen in the pots country command.
Step 6	Router(config)# pots tone-source {local | remote}	(Optional) Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router. The keywords are as follows: •local—(default) Specifies that the router supplies the tones. •remote—Specifies that the telephone switch supplies the tones.
Step 7	Router(config)# pots ringing-freq {20Hz | 25Hz | 50Hz}	(Optional) Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring. The keywords are as follows: •20Hz—Indicates that connected devices ring at 20 Hz. •25Hz—Indicates that connected devices ring at 25 Hz. •50Hz—Indicates that connected devices ring at 50 Hz. The default depends on the country chosen in the pots country command.
Step 8	Router(config)# pots disconnect-time interval	(Optional) Specifies the interval at which the disconnect method is applied if connected telephones, fax machines, or modems fail to detect that a calling party has disconnected. The interval argument is the number of milliseconds of the interval and ranges from 50 to 2000. The default depends on the country chosen in the pots country command.
Step 9	Router(config)# pots silence-time seconds	(Optional) Specifies the interval of silence after a calling party disconnects. The seconds argument is the number of seconds of the interval and ranges from 0 to 10. The default depends on the country chosen in the pots country command.
Step 10	Router(config)# pots distinctive-ring-guard-time milliseconds	(Optional) Specifies the delay after which a telephone port can be rung after a previous call is disconnected. The milliseconds argument is the number of milliseconds of the delay and ranges from 0 to 1000. The default depends on the country chosen in the pots country command.

Verifying Analog Telephone Connections on Cisco 803 and 804 Routers

After configuring analog telephone connections, perform the following steps to verify proper operation:

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Step 1 Pick up the handset of an attached telephony device and check for a dial tone.

Step 2 Review the configuration using the show pots status command, which displays settings of physical characteristics and other information on telephone interfaces.

Router# show pots status

POTS Global Configuration:

 Country: United States

  Dialing Method: Overlap, Tone Source: Remote, CallerId Support: YES

  Line Type: 600 ohm, PCM Encoding: u-law, Disc Type: OSI,

  Ringing Frequency: 20Hz, Distinctive Ring Guard timer: 0 msec

  Disconnect timer: 1000 msec, Disconnect Silence timer: 5 sec

  TX Gain: 6dB, RX Loss: -6dB,

  Filter Mask: 6F

  Adaptive Cntrl Mask: 0

POTS PORT: 1

  Hook Switch Finite State Machine:

   State: On Hook, Event: 0

   Hook Switch Register: 10, Suspend Poll: 0

  CODEC Finite State Machine

   State: Idle, Event: 0

  Connection: None, Call Type: Two Party, Direction: Rx only

  Line Type: 600 ohm, PCM Encoding: u-law, Disc Type: OSI,

  Ringing Frequency: 20Hz, Distinctive Ring Guard timer: 0 msec

  Disconnect timer: 1000 msec, Disconnect Silence timer: 5 sec

  TX Gain: 6dB, RX Loss: -6dB,

  Filter Mask: 6F

  Adaptive Cntrl Mask: 0

  CODEC Registers:

   SPI Addr: 2, DSLAC Revision: 4

   SLIC Cmd: 0D, TX TS: 00, RX TS: 00

   Op Fn: 6F, Op Fn2: 00, Op Cond: 00

   AISN: 6D, ELT: B5, EPG: 32 52 00 00

   SLIC Pin Direction: 1F

  CODEC Coefficients:

   GX: A0 00

   GR: 3A A1

   Z: EA 23 2A 35 A5 9F C2 AD 3A AE 22 46 C2 F0

   B: 29 FA 8F 2A CB A9 23 92 2B 49 F5 37 1D 01

   X: AB 40 3B 9F A8 7E 22 97 36 A6 2A AE

   R: 01 11 01 90 01 90 01 90 01 90 01 90

   GZ: 60

  ADAPT B: 91 B2 8F 62 31

 CSM Finite State Machine:

   Call 0 - State: idle, Call Id: 0x0

    Active: no

   Call 1 - State: idle, Call Id: 0x0

    Active: no

   Call 2 - State: idle, Call Id: 0x0

    Active: no

 POTS PORT: 2

   Hook Switch Finite State Machine:

    State: On Hook, Event: 0

    Hook Switch Register: 20, Suspend Poll: 0

   CODEC Finite State Machine:

    State: Idle, Event: 0

    Connection: None, Call Type: Two Party, Direction: Rx only

    Line Type: 600 ohm, PCM Encoding: u-law, Disc Type: OSI,

    Ringing Frequency: 20Hz, Distinctive Ring Guard timer: 0 mse

  Disconnect timer: 1000msec,Disconnect Silence timer: 5 sec

   TX Gain: 6dB, RX Loss: -6dB,

   Filter Mask: 6F

   Adaptive Cntrl Mask: 0

 CODEC Registers:

   SPI Addr: 3, DSLAC Revision: 4

   SLIC Cmd: 0D, TX TS: 00, RX TS: 00

   Op Fn: 6F, Op Fn2: 00, Op Cond: 00

   AISN: 6D, ELT: B5, EPG: 32 52 00 00

   SLIC Pin Direction: 1F

 CODEC Coefficients:

   GX: A0 00

   GR: 3A A1

   Z: EA 23 2A 35 A5 9F C2 AD 3A AE 22 46 C2 F0

   B: 29 FA 8F 2A CB A9 23 92 2B 49 F5 37 1D 01

   X: AB 40 3B 9F A8 7E 22 97 36 A6 2A AE

   R: 01 11 01 90 01 90 01 90 01 90 01 90

   GZ: 60

  ADAPT B: 91 B2 8F 62 31

 CSM Finite State Machine:

   Call 0 - State: idle, Call Id: 0x0

    Active: no

   Call 1 - State: idle, Call Id: 0x0

    Active: no

   Call 2 - State: idle, Call Id: 0x0

    Active: no

Time Slot Control: 0

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Troubleshooting Tip for Cisco 803 and 804 Routers

Check to ensure that all cables are securely connected.

Configuring Digital Voice Ports

The digital voice port commands discussed in this section configure channelized T1 or E1 connections; for information on ISDN connections, see "Configuring ISDN Interfaces for Voice" in this configuration guide.

The T1 or E1 lines that connect a telephony network to the digital voice ports on a router or access server contain channels for voice calls; a T1 line contains 24 full-duplex channels or timeslots, and an E1 line contains 30. The signal on each channel is transmitted at 64 kbps, a standard known as digital signal 0 (DS0); the channels are known as DS0 channels. The ds0-group command creates a logical voice port (a DS0 group) from some or all of the DS0 channels, which allows you to address those channels easily, as a group, in voice-port configuration commands.

Digital voice ports are found at the intersection of a packet voice network and a digital, circuit-switched telephone network. The digital voice port interfaces that connect the router or access server to T1 or E1 lines pass voice data and signaling between the packet network and the circuit-switched network.

Signaling is the exchange of information about calls and connections between two ends of a communication path. For instance, signaling communicates to the call's end points whether a line is idle or busy, whether a device is on-hook or off-hook, and whether a connection is being attempted. An end point can be a CO switch, a PBX, a telephony device such as a telephone or fax machine, or a voice-equipped router acting as a gateway. There are two aspects to consider about signaling on digital lines: one aspect is the actual information about line and device states that is transmitted, and the second aspect is the method used to transmit the information on the digital lines.

The actual information about line and device states is communicated over digital lines using signaling methods that emulate the methods used in analog circuit-switched networks: FXS, FXO, and E&M.

The method used to transmit the information describes the way that the emulated analog signaling is transmitted over digital lines, which may be common-channel signaling (CCS) or channel-associated signaling (CAS). CCS sends signaling information down a dedicated channel and CAS takes place within the voice channel itself. This chapter describes CAS signaling, which is sometimes called robbed-bit signaling because user bandwidth is robbed by the network for signaling. A bit is taken from every sixth frame of voice data to communicate on- or off-hook status, wink, ground start, dialed digits, and other information about the call.

In addition to setting up and tearing down calls, CAS provides the receipt and capture of dialed number identification (DNIS) and automatic number identification (ANI) information, which are used to support authentication and other functions. The main disadvantage of CAS signaling is its use of user bandwidth to perform these signaling functions.

For signaling to pass between the packet network and the circuit-switched network, both networks must use the same type of signaling. The voice ports on Cisco routers and access servers can be configured to match the signaling of most COs and PBXs, as explained in this chapter.

This section discusses the following topics:

•Prerequisites for Configuring Digital Voice Ports

•Preparing Information to Configure Digital Voice Ports

•Platform-Specific Digital Voice Hardware

•Configuring Basic Parameters on Digital T1/E1 Voice Ports

Prerequisites for Configuring Digital Voice Ports

Digital T1 or E1 packet voice capability requires specific service, software, and hardware:

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

•Create your company's dial plan.

•Establish a working telephony network based on your company's dial plan.

•Establish a connection to the network LAN or WAN.

•Set up a working IP and Frame Relay or ATM network. For more information about configuring IP, refer to the Cisco IOS IP Configuration Guide, Release 12.2.

•Install appropriate voice processing and voice interface hardware on the router. See the "Platform-Specific Digital Voice Hardware" section.

•(Cisco 2600 and 3600 series routers) For digital T1 packet voice trunk network modules, install Cisco IOS Release 12.0(5)XK, 12.0(7)T, 12.2(1), or a later release. The minimum DRAM memory requirements 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 that 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.

•(Cisco 2600 and 3600 series routers) For digital E1 packet voice trunk network modules, install Cisco IOS Release 12.1(2)T, 12.2(1), or a later release. The minimum DRAM memory requirements are:

–48 MB, with one or two E1s

–64 MB, with three to eight E1s

–128 MB, with 9 to 12 E1s

For high-volume applications, the memory required may be greater than these minimum values. Support for digital E1 packet voice trunk network modules is included in Plus feature sets. The IP Plus feature set requires 16 MB of Flash memory.

•(Cisco MC3810 concentrators) HCMs require Cisco IOS Release 12.0(7)XK or 12.1(2)T, 12.2(1), or a later release.

•(Cisco 7200 and 7500 series routers) For digital T1/E1 voice port adapters, install Cisco IOS Release 12.0(5)XE, 12.0(7)T, 12.2(1), or a later release. The minimum DRAM memory requirement to support T1/E1 high-capacity digital voice port adapters is 64 MB.

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

Preparing Information to Configure Digital Voice Ports

Gather the following informat