Voice Port Configuration Guide, Cisco IOS Release 15M&T

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 CONTENTS

CHAPTER 1 Voice Port Configuration Overview 1

Finding Feature Information 1
Voice Port Configuration Overview 1
Telephony Signaling Interfaces 3
FXS and FXO Interfaces 4
E and M Interfaces 5
Toll Fraud Prevention 6

CHAPTER 2 Configuring Analog Voice Ports 9

Finding Feature Information 9
Prerequisites for Configuring Analog Voice Ports 9
Information About Analog Voice Hardware 10
Cisco 880 Series Routers 10
Cisco 1750 Modular Router 11
Cisco2600 Cisco 3600 and Cisco 3700 Series Routers 11
Cisco MC3810 12
Basic Parameters on Analog FXO FXS or E and M Voice Ports 12
Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High-Performance
Compression Modules 13
How to Configure Analog Voice Ports 13
Configuring Basic Parameters on Analog FXO FXS or E and M Voice Ports 13
Configuring Codec Complexity on the Cisco MC3810 17

CHAPTER 3 Configuring Digital Voice Ports 19

Finding Feature Information 20
Prerequisites for Configuring Digital Voice Ports 20
Information About Digital Voice Hardware 22
Cisco 880 Series Routers 22

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Cisco 2600 Cisco 3600 and Cisco 3700 Series Routers 23

Cisco 7200 and Cisco 7500 Series Routers 24
Cisco AS5300 24
Cisco AS5350 and Cisco AS5400 Universal Gateways 25
Cisco AS5800 25
Cisco AS5850 Universal Gateway 26
Cisco Catalyst 6500 Series Switches and Cisco 7600 Series Routers 26
Cisco MC3810 27
How to Configure Digital T1 E1 Voice Ports 28
Configuring Codec Complexity on Digital T1 E1 Voice Ports 28
Changing Codec Complexity 29
Configuring the Flex Option on Codec Complexity 33
Configuring Codec Complexity 36
Configuring Controller Settings for Digital T1 E1 Voice Ports 38
Framing Formats on Digital T1 E1 Voice Ports 39
Clock Sources on Digital T1 E1 Voice Ports 40
Network Clock Timing 43
Line Coding on Digital T1 E1 Voice Ports 45
DS0 Groups on Digital T1 E1 Voice Ports 45
Configuring Basic Voice Port Parameters for Digital T1 E1 Voice Ports 50

CHAPTER 4 Fine-Tuning Analog and Digital Voice Ports 55

Finding Feature Information 55
Information About Fine-Tuning Analog and Digital Voice Ports 55
How to Configure Fine-Tuning Features for Voice Ports 56
Configuring Channel Bank Support for T1 E1 Voice Ports 56
Configuring Auto Cut-Through 60
Modifying Bit Patterns for Digital Voice Ports 61
Configuring ANI for Outbound Calling 63
Configuring Disconnect Supervision 65
Configuring FXO Supervisory Disconnect Tones 67
Configuring Timeouts Parameters 71
Changing Timing Parameters 72
Configuring the DTMF Timer 76
Configuring Comfort Noise and Music Threshold for VAD 77

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CHAPTER 5 PSTN Fallback 81

Finding Feature Information 81
Information About PSTN Fallback 82
Service Assurance Agent 82
Application of PSTN Fallback 82
Restrictions for PSTN Fallback 82
How to Configure PSTN Fallback 83
Configuring Call Fallback to Use MD5 Authentication for SAA Probes 83
Configuring Destination Monitoring without Fallback to Alternate Dial Peers 84
Configuring Call Fallback Cache Parameters 84
Configuring Call Fallback Jitter-Probe Parameters 85
Configuring Call Fallback Probe-Timeout and Weight Parameters 87
Configuring Call Fallback Threshold Parameters 88
Configuring Call Fallback Wait-Timeout 89
Configuring VoIP Alternate Path Fallback SNMP Trap 90
What to Do Next 91
Configuring Call Fallback Map Parameters 91
Configuring ICMP Pings to Monitor IP Destinations 92
Dial Peer Configuration 93
Global Configuration 94
Voice Port Configuration 95
Voice Class Configuration 96
How to Verify and Monitor the PSTN Fallback Feature 97
Verifying PSTN Fallback Configuration 97
Monitoring and Maintaining PSTN Fallback 98
What To Do Next 98

CHAPTER 6 Configuring Echo Cancellation 99

Finding Feature Information 99
Information About Echo Cancellation 99
Voice Call Transmit and Receive Paths 99
Echo Cancellation 100
Echo Canceller Operation 101
Echo Canceller Components 102

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Echo Canceller Coverage 102

ITU-T Echo Cancellation History 103
Extended G.168 Echo Canceller Features 104
Extended EC Comparison 105
Extended Echo Canceller Support by Platform 105
Extended G.168 Echo Canceller 108
How to Configure the Extended G.168 Echo Canceller 109
Restrictions for G.168 Extended Echo Canceller 109
Changing Echo Cancellers on Digital Voice Ports 110
Enabling the Extended G.168 EC in Cisco IOS Release 12.2(13)T 110
Enabling the Extended G.168 EC in Cisco IOS Release 12.2(13)ZH 111
Configuring the Extended G.168 EC on the Cisco AS5300 112
Modifying Echo Cancellation Default Settings 115
Configuration Examples for Extended G.168 Echo Cancellation 118
Enabling the Extended EC on the Cisco 1700 Series and Cisco ICS 7750 Example 118
Enabling the Extended EC Prior to Cisco IOS Release 12.3(4)XD Example 118
Enabling the Extended EC on the Cisco 7200 and Cisco 7500 Series Example 119
Enabling the Extended Echo Canceller on the Cisco AS5300 Example 119
Adjusting the Echo Canceller Size Example 120
Worst-Case Echo Return Loss Example 120

CHAPTER 7 Pulse Code Modulation (PCM) Audio Capture 121

Finding Feature Information 121
Information about PCM Audio Capture 122
PCM Audio Capture 122
How to Configure PCM Audio Capture 122
Configuring PCM Audio Capture 122
Verifying PCM Audio Capture 124
Additional References for Cisco UBE Serviceability 125
Feature Information for Pulse Code Modulation (PCM) Audio Capture 126

CHAPTER 8 Acoustic Shock Protection 127

Finding Feature Information 127
Restrictions for ASP 127
Information About ASP 128

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Acoustic Shock Protection 128

How to Configure ASP 129
Creating the Media Profile for ASP 129
Creating the Media Profile to Enable ASP 130
Configuring the Media Class at a Dial Peer Level for ASP 131
Configuring the Media Class Globally for ASP 132
Verifying ASP 133
Troubleshooting Tips 134
Configuration Examples for the Acoustic Shock Protection Feature 134
Feature Information for Acoustic Shock Protection 135

CHAPTER 9 Noise Reduction 137

Finding Feature Information 137
Prerequisites for Noise Reduction 137
Restrictions for NR 138
Information About NR 138
Noise Reduction 138
How to Configure NR 139
Creating the Media Profile for NR 139
Creating the Media Class to Enable NR 140
Configuring the Media Class at a Dial Peer Level for NR 141
Configuring the Media Class Globally for NR 142
Verifying NR 143
Troubleshooting Tips 144
Configuration Examples for the NR feature 144
Feature Information for Noise Reduction 145

CHAPTER 10 Configuring Hardware Echo Cancellation on T1 E1 Multiflex Voice WAN Interface Cards 147
Finding Feature Information 147
Prerequisites for Hardware Echo Cancellation 148
Cisco IOS Image 148
Baseboard and Daughter Card Configuration 148
Restrictions for Hardware Echo Cancellation 148
Hardware Echo Cancellation Tail Length 148
Accurate TDM ERL Readings for Echo Cancellation 148

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Sample Output of the show voice call command 148

How to Configure Hardware Echo Cancellation 150
Examples 152
show echo-cancel hardware status Example 153
show call active voice echo-canceller summary Example 153
show call active voice echo-canceller CallID Example 153

CHAPTER 11 NextPort-Based Voice Tuning and Echo Cancellation 155

Finding Feature Information 155
Prerequisites for NextPort Services 155
Information About NextPort Voice Services 156
NextPort Dual-Filter G.168 Echo Canceller 156
NextPort SPE Firmware 157
Voicecap Strings 157
Voice Tuning 157
Background Noise 158
How to Configure NextPort Services 158
Downloading NextPort SPE Firmware 158
Creating and Applying Voicecaps 160
Restrictions 160
Setting Voice Tuning Parameters with V Registers 160
Set PSTN Gains 161
Set IP Gains 161
Set Dynamic Attenuation 161
Set Comfort Noise Generation 162
Set Minimum ERL 162
Creating and Applying Voice Caps 162
Verifying Voicecap Configurations 163
Troubleshooting NextPort Voicecaps 166
Configuring the NextPort Dual-Filter G.168 Echo Canceller 167
Configuration Examples for NextPort Services 168
High ERL in the Network Example 168
Low ERL in the Network Example 169
Clipped or Squelched Speech and Low ERL in the Network Example 169
Dynamic Attenuation Example 169

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Echo Canceller Tail Coverage Example 170

Enabling NextPort Echo Canceller Control for G.711 Encoded VoIP Packets 170
Troubleshooting NextPort Echo Canceller Control for G.711 Encoded VoIP Packets 172

CHAPTER 12 Verifying Analog and Digital Voice-Port Configurations 175

Finding Feature Information 175
Information About Verifying Voice-Port Configurations 175
How to Verify Voice-Port Configurations 175
Examples 178
show voice port summary Command Examples 178
show voice port Command Examples 179
show controller Command Examples 182
show voice dsp Command Examples 183
show voice call summary Command Examples 184
show call active voice Command Example 184
show call history voice Command Example 185

CHAPTER 13 Troubleshooting Analog and Digital Voice Port Configurations 187

Troubleshooting Chart 187

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 CHAPTER 1
Voice Port Configuration Overview
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 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. In addition to the commands for basic configuration, there are
also commands that provide fine-tuning for voice quality, enable special features, and specify parameters to
match those of proprietary PBXs.
Not all voice-port commands are covered in this document. Some are described in the Cisco IOS ISDN Voice
Configuration Guide, Release 12.4 or the "Trunk Management Features" document, Cisco IOS Voice
Configuration Library, Release 12.4. The voice-port configuration commands included in this document are
fully documented in the Cisco IOS Voice Command Reference.

• Finding Feature Information, page 1

• Voice Port Configuration Overview, page 1

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

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

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Voice Port Configuration Overview

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 following illustrations show examples of how voice ports are used.
• The "Telephone to WAN" figure shows one voice port connecting a telephone to the WAN through the
router.
• The "Telephone to PSTN" figure shows one voice port connected to the PSTN and another to a telephone;
the router acts like a small PBX.
• The "PBX-to-PBX over a WAN" figure shows how two PBXs can be connected over a WAN to provide
toll bypass.

Figure 1: Telephone to WAN

Figure 2: Telephone to PSTN

Figure 3: PBX-to-PBX over a WAN

Cisco provides a variety of Cisco IOS commands for flexibility in configuring 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. The table below shows the analog and digital
voice-port connection support of the router platforms discussed in this document.

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Table 1: Analog and Digital Voice-Port Support on Cisco Platforms

Platform Analog Digital

Cisco 880 series (includes Yes Yes
IAD881B, IAD881F, C881SRST,
IAD888B, IAD888F, and
C888SRST)

Cisco 1750 Yes No

Cisco 2600 series Yes Yes

Cisco 3600 series Yes Yes

Cisco 3700 series Yea Yes

Cisco 7200 series No Yes

Cisco 7500 series No Yes

Cisco AS5300 No Yes

Cisco AS5350 No Yes

Cisco AS5400 No Yes

Cisco AS5800 No Yes

Cisco AS5850 No Yes

Cisco MC3810 Yes 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 document 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 "FXS and FXO Interfaces" and the
Telephony Signaling Interfaces. 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.

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The next three illustrations show how the different signaling interfaces are associated with different uses of
voice ports. In the "FXS Signaling Interfaces" figure, FXS signaling is used for end-user telephony equipment,
such as a telephone or fax machine. The "FXS and FXO Signaling Interfaces" figure 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 the "E&M Signaling
Interfaces" figure, 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 4: FXS Signaling Interfaces

Figure 5: FXS and FXO Signaling Interfaces

Figure 6: E and 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.

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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 operates only 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
"Configuring Disconnect Supervision" and "Configuring FXO Supervisory Disconnect Tones" sections in
the "Fine-Tuning Analog and Digital Voice Ports" chapter 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 and 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 telecommunications, 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 the
table below. Cisco voice implementation supports E&M Types I, II, III, and V.

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Table 2: EandM Wiring and Signaling Methods

E&M Type E-Lead M-Lead Signal Battery Lead Signal Ground Lead
Configuration Configuration Configuration Configuration
I Output, relay to Input, referenced to -- --
ground ground

II Output, relay to SG Input, referenced to Feed for M, Return for E,

ground connected to -48V galvanically isolated
from ground

III Output, relay to Input, referenced to Connected to -48V Connected to

ground ground ground

V Output, relay to Input, referenced to -- --

ground -48V

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 in the "Configuring Digital Voice Ports" chapter .
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 the following three types of access signaling:
• 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 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.

Toll Fraud Prevention

When a Cisco router platform is installed with a voice-capable Cisco IOS software image, appropriate features
must be enabled on the platform to prevent potential toll fraud exploitation by unauthorized users. Deploy
these features on all Cisco router Unified Communications applications that process voice calls, such as Cisco
Unified Communications Manager Express (CME), Cisco Survivable Remote Site Telephony (SRST), Cisco
Unified Border Element (UBE), Cisco IOS-based router and standalone analog and digital PBX and

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public-switched telephone network (PSTN) gateways, and Cisco contact-center VoiceXML gateways. These
features include, but are not limited to, the following:
• Disable secondary dial tone on voice ports--By default, secondary dial tone is presented on voice ports
on Cisco router gateways. Use private line automatic ringdown (PLAR) for foreign exchange office
(FXO) ports and direct-inward-dial (DID) for T1/E1 ports to prevent secondary dial tone from being
presented to inbound callers.
• Cisco router access control lists (ACLs)--Define ACLs to allow only explicitly valid sources of calls to
the router or gateway, and therefore to prevent unauthorized Session Initiation Protocol (SIP) or H.323
calls from unknown parties to be processed and connected by the router or gateway.
• Close unused SIP and H.323 ports--If either the SIP or H.323 protocol is not used in your deployment,
close the associated protocol ports. If a Cisco voice gateway has dial peers configured to route calls
outbound to the PSTN using either time division multiplex (TDM) trunks or IP, close the unused H.323
or SIP ports so that calls from unauthorized endpoints cannot connect calls. If the protocols are used
and the ports must remain open, use ACLs to limit access to legitimate sources.
• Change SIP port 5060--If SIP is actively used, consider changing the port to something other than
well-known port 5060.
• SIP registration--If SIP registration is available on SIP trunks, turn on this feature because it provides
an extra level of authentication and validation that only legitimate sources can connect calls. If it is not
available, ensure that the appropriate ACLs are in place.
• SIP Digest Authentication--If the SIP Digest Authentication feature is available for either registrations
or invites, turn this feature on because it provides an extra level of authentication and validation that
only legitimate sources can connect calls.
• Explicit incoming and outgoing dial peers--Use explicit dial peers to control the types and parameters
of calls allowed by the router, especially in IP-to-IP connections used on CME, SRST, and Cisco UBE.
Incoming dial peers offer additional control on the sources of calls, and outgoing dial peers on the
destinations. Incoming dial peers are always used for calls. If a dial peer is not explicitly defined, the
implicit dial peer 0 is used to allow all calls.
• Explicit destination patterns--Use dial peers with more granularity than .T for destination patterns to
block disallowed off-net call destinations. Use class of restriction (COR) on dial peers with specific
destination patterns to allow even more granular control of calls to different destinations on the PSTN.
• Translation rules--Use translation rules to manipulate dialed digits before calls connect to the PSTN to
provide better control over who may dial PSTN destinations. Legitimate users dial an access code and
an augmented number for PSTN for certain PSTN (for example, international) locations.
• Tcl and VoiceXML scripts--Attach a Tcl/VoiceXML script to dial peers to do database lookups or
additional off-router authorization checks to allow or deny call flows based on origination or destination
numbers. Tcl/VoiceXML scripts can also be used to add a prefix to inbound DID calls. If the prefix plus
DID matches internal extensions, then the call is completed. Otherwise, a prompt can be played to the
caller that an invalid number has been dialed.
• Host name validation--Use the "permit hostname" feature to validate initial SIP Invites that contain a
fully qualified domain name (FQDN) host name in the Request Uniform Resource Identifier (Request
URI) against a configured list of legitimate source hostnames.
• Dynamic Domain Name Service (DNS)--If you are using DNS as the "session target" on dial peers, the
actual IP address destination of call connections can vary from one call to the next. Use voice source
groups and ACLs to restrict the valid address ranges expected in DNS responses (which are used
subsequently for call setup destinations).

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For more configuration guidance, see the " Cisco IOS Unified Communications Toll Fraud Prevention " paper.

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Configuring Analog Voice Ports
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. Connections to the PSTN central office (CO) are typically made with digital
interfaces.

• Finding Feature Information, page 9

• Prerequisites for Configuring Analog Voice Ports, page 9
• Information About Analog Voice Hardware, page 10
• How to Configure Analog Voice Ports, page 13

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

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 12.4.

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Information About Analog Voice Hardware

• Install appropriate voice processing and voice interface hardware on the router. See the Information
About Analog Voice Hardware, on page 10.
• Gather the following information about the telephony connection of the voice port:
• 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

If you are connecting a voice-port interface 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.

Note The slot and port numbering of interface cards differs for each of the voice-enabled routers. For the specific
slot and port designations for your hardware platform, refer to the Cisco Interface Cards Hardware
Installation Guides. More current information may be available in the release notes for the Cisco IOS
software you are using.

Information About Analog Voice Hardware

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

Cisco 880 Series Routers

Beginning with Cisco IOS Release 12.4(15)XZ, the Cisco 880 series fixed router platforms support the
implementation of analog (FXS/DID/FXO) and digital (BRI S/T) voice ports. The IAD881B, IAD881F,
IAD888B, and IAD888F models support voice interface FXS or BRI. The IAD881F and IAD888F models
have four FXS ports and the IAD881B and IAD888B models support two ports for ISDN BRI digital voice
interface.
In the IAD881B and IAD888B models, the voice BRI interface presents an ISDN S/T interface to connect
either to an NT1 terminating an ISDN telephone network (TE-side) or to a TE user device such as an ISDN
telephone or PBX (NT-side). In the IAD881B and IAD888B models, the BRI interface is available as the
primary voice interface and is intended to be connected to a PBX (network side trunk). All the voice interfaces
are onboard though they are recognized as a 4-port FXS VIC and a 2-port BRI VIC in order to leverage
existing voice drivers.

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Cisco 1750 Modular Router

Note If the primary voice interface is FXS and the backup is BRI, then ports 0, 1, 2, and 3 are analog voice
ports, and ports 4 and 5 are digital. If the primary voice interface is BRI, then ports 1, 2, 3, and 4 are
digital.

The C881 and C888 SRST models automatically detect a failure occuring in the network and initiate a process
to auto-configure the router. This process provides call-processing backup redundancy for the IP and FXS
phones and helps to ensure that telephony capabilities stay operational. All the IP or analog phones hanging
off of a telecommuter site are controlled by the headquarters office call control (Cisco Unified CallManager
or CallManager Express). In case of a WAN failure, the telecommuter router allows all phones to re-register
to it in SRST mode and allow all inbound and outbound dialing to be routed off to the PSTN (using back up
FXO or BRI port). Upon restoration of WAN connectivity, the system automatically shifts call processing
back to the primary Cisco Unified Call Manager cluster.

Cisco 1750 Modular Router

The Cisco 1750 modular router provides 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 voice
interface card (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.

Cisco2600 Cisco 3600 and Cisco 3700 Series Routers

The Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series routers are modular, multifunction platforms
that combine dial access, routing, LAN-to-LAN services, and multiservice integration of voice, video, and
data in the same device.
Voice network modules installed in Cisco 2600 series, Cisco 3600 series, or Cisco 3700 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.
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 Routers Hardware Installation Guide
• Cisco 3600 Series Routers Hardware Installation Guide
• Cisco Network Modules Hardware Installation Guide
• Cisco Interface Cards Installation Guide

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Cisco MC3810

Cisco MC3810
To support analog voice circuits, a Cisco MC3810 must be equipped with an AVM, which supports six analog
voice ports. When you install 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, 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 .

Note For current information about supported hardware, refer to the release notes for the platform and Cisco
IOS release you are using.

Basic Parameters on Analog FXO FXS or E and M Voice Ports

This section describes commands for basic analog voice port configuration.
All the data recommended in the Prerequisites for Configuring Analog Voice Ports, on page 9 should be
gathered before you start this procedure.
If you are configuring a Cisco MC3810 that has HCMs, you should also configure the codec complexity by
performing the tasks in the Configuring Codec Complexity on the Cisco MC3810, on page 17.

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

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 this section and also in
the "Fine-Tuning Analog and Digital Voice Ports" chapter.
After the voice port has been configured, make sure that the ports are operational by performing the tasks
described in the following chapters:
• "Verifying Analog and Digital Voice-Port Configuration"

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

• "Troubleshooting Analog and Digital Voice Port Configurations"

For more information on these and other voice port commands, refer to the Cisco IOS Voice Command
Reference.

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 to determine the syntax choices that are available.

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 type 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 in dial peers.
For more information, refer to the "Dial Peer Configuration on Voice Gateway Routers" document.
Codec complexity refers to the amount of processing power that a codec compression method requires. The
greater the codec complexity, the fewer the calls that the DSP interfaces can handle. Codec complexity is
either medium or high. The default is medium. All medium-complexity codecs can also run in high-complexity
mode, but fewer (usually half as many) channels are available per DSP. The codec complexity value determines
the choice of codecs that are available in the dial peers when the codec command has been configured. For
details on the number of calls that can be handled simultaneously using each of the codec standards, refer to
the entries for the codecand codec complexity commands in the Cisco IOS Voice Command Reference.

How to Configure Analog Voice Ports

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

Perform this task to configure basic parameters:

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

SUMMARY STEPS

1. enable
2. configure terminal
3. Do one of the following:
• voice-port slot / port

4. Do one of the following:

• signal {loop-start | ground-start}

5. cptone locale
6. dial-type {dtmf | pulse}
7. operation {2-wire | 4-wire}
8. type {1 | 2 | 3 | 5}
9. Do one of the following:
• ring frequency {25 | 50}
•
•
• ring frequency {20 | 30}

10. ring number number

11. ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07 |
pattern08 | pattern09 | pattern10 | pattern11 | pattern12] | [define pulse interval]}
12. description string
13. no shutdown

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 Do one of the following: Enters voice-port configuration mode.

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

Command or Action Purpose

• voice-port slot / port Note The slash must be entered between slot and
port.
• Valid entries vary by router platform; enter the show voice port
Example: summary command for available values.
Router(config)# voice-port 1/0
Note For the Cisco 880 series platforms, the command syntax does
not include a slot number, only the port is identified. If the
Example: primary voice interface is FXS and the backup is BRI, then
ports 0, 1, 2, and 3 are analog voice ports, and ports 4 and 5
voice-port slot/subunit/port
are digital. If the primary voice interface is BRI, then ports 1,
2, 3, and 4 are digital.
Example:
Router(config)# voice-port 1/0/0

Step 4 Do one of the following: Selects the access signaling type to match that of the telephony
connection you are making.
• signal {loop-start | ground-start}
Note Configuring the signal keyword for one voice port on a Cisco
2600 or Cisco 3600 series router VIC changes the signal value
Example: for both ports on the VIC.

Router(config-voiceport)# signal
ground-start

Example:
signal {wink-start | immediate-start |
delay-dial}

Example:
Router(config-voiceport)# signal
wink-start

Step 5 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.
Example: • Cisco routers comply with the ISO 3166 locale name standards.
Router(config-voiceport)# cptone us To see valid choices, enter a question mark (?) following the
cptone command.
• The default is us.

Step 6 dial-type {dtmf | pulse} (FXO only) Specifies the dialing method for outgoing calls.

Example:
Router(config-voiceport)# dial-type dtmf

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

Command or Action Purpose

Step 7 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).
Example: • The default is 2-wire.
Router(config-voiceport)# operation
4-wire

Step 8 type {1 | 2 | 3 | 5} (E&M only) Specifies the type of E&M interface to which this voice
port is connecting. See Table 2 in the "Voice Port Configuration
Example: Overview" chapter for an explanation of E&M types.

Router(config-voiceport)# type 2 • The default is 1.

Step 9 Do one of the following: (FXS only) Selects the ring frequency, in hertz, used on the FXS
interface. This number must match the connected telephony equipment
• ring frequency {25 | 50} and may be country-dependent. If the ring frequency is not set properly,
• the attached telephony device may not ring or it may buzz.
• • The keyword default is 25 on the Cisco 1750 router, Cisco 2600
• ring frequency {20 | 30} and Cisco 3600 series routers; and 20 on the Cisco MC3810.

Example:
Router(config-voiceport)# ring frequency
50

Example:
Router(config-voiceport)# ring frequency
30

Step 10 ring number number (FXO only) Specifies the maximum number of rings to be detected
before an incoming call is answered by the router.
Example: • The default is 1.
Router(config-voiceport)# ring number 1

Step 11 ring cadence {[pattern01 | pattern02 | (FXS only) Specifies an existing pattern for ring, or defines a new one.
pattern03 | pattern04 | pattern05 | pattern06 Each pattern specifies a ring-pulse time and a ring-interval time.
| pattern07 | pattern08 | pattern09 | pattern10
| pattern11 | pattern12] | [define pulse • The default is the pattern specified by the cptone locale that has
been configured.
interval]}

Example:
Router(config-voiceport)# ring cadence
pattern01

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Command or Action Purpose

Step 12 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
Example: 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.
Router(config-voiceport)# description
255 • The default is that there is no text string (describing the voice
port) attached to the configuration.

Step 13 no shutdown Activates the voice port. If a voice port is not being used, shut the voice
port down with the shutdown command.
Example:
Router(config-voiceport)# no shutdown

Configuring Codec Complexity on the Cisco MC3810

To configure codec complexity for analog voice ports on the Cisco MC3810 using High-Performance
Compression Modules (HCMs), use the following commands:

SUMMARY STEPS

1. enable
2. show voice dsp
3. configure terminal
4. voice-card 0
5. codec complexity {high | medium}

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 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
Example: the DSP channels are in the idle state, continue to Step 3.

Router# show voice dsp

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Command or Action Purpose

Step 3 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 4 voice-card 0 Enters voice-card configuration mode and specifies voice card 0.

Example:
Router(config)# voice-card 0

Step 5 codec complexity {high | medium} Specifies codec complexity based on the codec standard being used.
This setting restricts the codecs available in dial peer configuration.
Example: All voice cards in a router must use the same codec complexity setting.

Router(config-voicecard)# codec Note If two HCMs are installed, this command configures both
complexity high HCMs at once.

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 CHAPTER 3
Configuring Digital Voice Ports
The digital voice port commands discussed in this section configure channelized T1 or E1 connections; for
information on ISDN connections, refer to the Cisco IOS ISDN Voice 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 endpoints whether a line is idle or busy, whether a
device is on-hook or off-hook, and whether a connection is being attempted. An endpoint can be a central
office (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: Foreign Exchange Service (FXS), Foreign
Exchange Office (FXO), and Ear and Mouth (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, 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 is its use of user bandwidth to perform
these signaling functions.

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 Configuring Digital Voice Ports
Finding Feature Information

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 document.

• Finding Feature Information, page 20

• Prerequisites for Configuring Digital Voice Ports, page 20
• Information About Digital Voice Hardware, page 22
• How to Configure Digital T1 E1 Voice Ports, page 28

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

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.
• Install appropriate voice processing and voice interface hardware on the router. See the Information
About Digital Voice Hardware, on page 22.
• (Cisco 2600 and Cisco 3600 series routers) For digital T1 packet voice trunk network modules, install
Cisco IOS Release 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.

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Prerequisites for Configuring Digital Voice Ports

• (Cisco 2600 and Cisco 3600 series routers) For digital E1 packet voice trunk network modules, install
Cisco IOS Release 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.
• Before you can run the IP Communications High-Density Digital Voice/Fax Network Module feature
on T1/E1 interfaces, you must install an IP Plus image (minimum) of Cisco IOS Release 12.3(7)T or a
later release.
• (Cisco MC3810 concentrators) HCMs require Cisco IOS Release 12.2(1) or a later release.
• (Cisco 7200 and Cisco 7500 series routers) For digital T1/E1 voice port adapters, install Cisco IOS
Release 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.
• Gather the following information about the telephony network connection of the voice port:
• Line interface: T1 or E1
• Signaling interface: FXO, FXS, or E&M. If the interfaces are PRI or BRI, refer to the Cisco IOS
ISDN Voice Configuration Guide, and Cisco IOS Terminal Services Configuration Guide.
• Line coding: AMI or B8ZS for T1, and AMI or HDB3 for E1
• Framing format: SF (D4) or ESF for T1, and CRC4 or no-CRC4 for E1
• Number of channels

After the controllers have been configured, the show voice port summarycommand can be used to determine
available voice port numbers. If the show voice port command and a specific port number is entered, the
default voice-port configuration for that port displays.
The following is show voice port summary sample output for a Cisco MC3810:

Router# show voice port summary

IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
====== == ========== ===== ==== ======== ======== ==
0:17 18 fxo-ls down down idle on-hook y
0:18 19 fxo-ls up dorm idle on-hook y
0:19 20 fxo-ls up dorm idle on-hook y
0:20 21 fxo-ls up dorm idle on-hook y
0:21 22 fxo-ls up dorm idle on-hook y
0:22 23 fxo-ls up dorm idle on-hook y
0:23 24 e&m-imd up dorm idle idle y

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Information About Digital Voice Hardware

Note The slot and port numbering of interface cards differs for each of the voice-enabled routers. For specific
slot and port designations, refer to the hardware installation documentation for your router platform. More
current information may be available in the release notes that accompany the Cisco IOS software you are
using.

Information About Digital Voice Hardware

Note For current information about supported hardware, refer to the release notes for the platform and Cisco
IOS release you are using.

Cisco 880 Series Routers

Beginning with Cisco IOS Release 12.4(15)XZ, the Cisco 880 series fixed router platforms support the
implementation of analog (FXS/DID/FXO) and digital (BRI S/T) voice ports. The IAD881B, IAD881F,
IAD888B, and IAD888F models support voice interface FXS or BRI. The IAD881F and IAD888F models
have four FXS ports and the IAD881B and IAD888B models support two ports for ISDN BRI digital voice
interface.
In the IAD881B and IAD888B models, the voice BRI interface presents an ISDN S/T interface to connect
either to an NT1 terminating an ISDN telephone network (TE-side) or to a TE user device such as an ISDN
telephone or PBX (NT-side). In the IAD881B and IAD888B models, the BRI interface is available as the
primary voice interface and is intended to be connected to a PBX (network side trunk). All the voice interfaces
are onboard though they are recognized as a 4-port FXS VIC and a 2-port BRI VIC in order to leverage
existing voice drivers.
The C881and C888 SRST models automatically detect a failure occuring in the network and initiate a process
to auto-configure the router. This process provides call-processing backup redundancy for the IP and FXS
phones and helps to ensure that telephony capabilities stay operational. All the IP or analog phones hanging
off of a telecommuter site are controlled by the headquarters office call control (Cisco Unified CallManager
or CallManager Express). In case of a WAN failure, the telecommuter router allows all phones to re-register
to it in SRST mode and allow all inbound and outbound dialing to be routed off to the PSTN (using back up
FXO or BRI port). Upon restoration of WAN connectivity, the system automatically shifts call processing
back to the primary Cisco Unified Call Manager cluster.

Note If the primary voice interface is FXS and the backup is BRI, then ports 0, 1, 2, and 3 are analog voice
ports, and ports 4 and 5 are digital. If the primary voice interface is BRI, then ports 1, 2, 3, and 4 are
digital.

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Cisco 2600 Cisco 3600 and Cisco 3700 Series Routers

Cisco 2600 Cisco 3600 and Cisco 3700 Series Routers

Digital voice hardware on Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series modular access routers
includes the high-density voice (HDV) network module and the multiflex trunk (MFT) voice/WAN interface
card (VWIC). When an HDV is used in conjunction with an MFT and packet voice DSP modules (PVDMs),
the HDV module is also called a digital packet voice trunk network module. The digital T1 or E1 packet voice
trunk network module supports T1 or E1 applications, including fractional use. The T1 version integrates a
fully managed DSU/CSU, and the E1 version includes a fully managed DSU. The digital T1 or E1 packet
voice trunk network module provides per-channel T1 or E1 data rates of 64 or 56 kbps for WAN services
(Frame Relay or leased line).
Digital T1 or E1 packet voice trunk network modules allow enterprises or service providers, using the
voice-equipped routers as customer premises equipment (CPE), to deploy digital voice and fax relay. These
network modules receive constant bit-rate telephony information over T1 or E1 interfaces and convert that
information to a compressed format so that it can be sent over a packet network. The digital T1 or E1 packet
voice trunk network modules can connect either to a PBX (or similar telephony device) or to a CO to provide
PSTN connectivity.
The MFT VWICs that are used in the packet voice trunk network modules are available in one- and two-port
configurations for T1 and for E1, and in two-port configurations with drop-and-insert capability for T1 and
E1. MFTs support the following kinds of traffic:
• Data. As WICs for T1 or E1 applications, including fractional data line use, the T1 version includes a
fully managed DSU/CSU, and the E1 version includes a fully managed DSU.
• Packet voice. As VWICs included with the digital T1 or E1 packet voice trunk network module to
provide connections to PBXs and COs, the MFTs enable packet voice applications.
• Multiplexed voice and data. Some two-port T1 or 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 64-kbps DS0 channels to be taken from one T1 and digitally
cross-connected to 64-kbps DS0 channels on another T1. Drop and insert, sometimes called time-division
multiplex (TDM) cross-connect, uses circuit switching rather than the digital signal processors (DSPs)
that VoIP technology employs. (Drop-and-insert is described in the "Trunk Management Features"
document.

The digital T1 or E1 packet voice trunk network module contains five 72-pin Single In-line Memory Module
(SIMM) sockets or banks, numbered 0 through 4, for PVDMs. Each socket can be filled with a single 72-pin
PVDM, and there must be at least one packet voice data module (PVDM-12) in the network module to process
voice calls. Each PVDM holds three DSPs, so with five PVDM slots populated, a total of 15 DSPs are provided.
High-complexity codecs support two simultaneous calls on each DSP, and medium-complexity codecs support
four calls on each DSP. A digital T1 or E1 packet voice trunk network module can support the following
numbers of channels:
• When the digital T1 or E1 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, G729 Annex A (E1), G.729 Annex B, G.723.1, G723.1 Annex A (T1), G.728, and fax
relay.
• When the digital T1 or E1 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.

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Cisco 7200 and Cisco 7500 Series Routers

For more information, refer to the following:

• Hardware installation documents for Cisco 2600 series
• Hardware installation documents for Cisco 3600 series
• Cisco Network Modules Hardware Installation Guide
• Cisco Interface Cards Installation Guide

Cisco 7200 and Cisco 7500 Series Routers

Cisco 7200 and Cisco 7500 series routers support multimedia routing and bridging with a wide variety of
protocols and media types. The Cisco 7000 family Versatile Interface Processor (VIP) is based on a reduced
instruction set computing (RISC) engine optimized for I/O functions. To this engine are attached one or two
port adapters or daughter boards, which provide the media-specific interfaces to the network. The network
interfaces provide connections between the routers’ peripheral component interconnect (PCI) buses and external
networks. Port adapters can be placed in any available port adapter slot, in any desired combination.
T1/E1 high-capacity digital voice port adapters for Cisco 7200 and Cisco 7500 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/E1 interfaces and can convert that information
to a compressed format for transmission as VoIP. Two types of digital voice port adapters are supported on
Cisco 7200 and Cisco 7500 series routers: two-port high-capacity (up to 48 or 120 channels of compressed
voice, depending on codec choice), and two-port moderate capacity (up to 24 or 48 channels of compressed
voice). These single-width port adapters incorporate two universal ports configurable for either T1 or E1
connection, for use with high-performance DSPs. Integrated CSU/DSUs, echo cancellation, and DS0
drop-and-insert functionality eliminate the need for external line termination devices and multiplexers.
For more information, refer to the following publications:
• Cisco 7200 VXR Installation and Configuration Guide
• Cisco 7500 Series Installation and Configuration Guide
• T1/E1 Moderate-Capacity and High-Capacity Digital Voice Port Adapter Installation and Configuration

Note For current information about supported hardware, refer to the release notes for the platform and Cisco
IOS release you are using.

Cisco AS5300
The Cisco AS5300 includes three expansion slots. One slot is for either an Octal T1/E1/PRI feature card (eight
ports) or a Quad T1/E1/PRI feature card (four ports), and the other two can be used for voice/fax or modem
feature cards. Because a single voice/fax feature card (VFC) can support up to 48 (T1) or 60 (E1) voice calls,
the Cisco AS5300 can support a total of 96 or 120 simultaneous voice calls.
Cisco AS5300 VFCs are coprocessor cards, each with a powerful reduced instruction set computing (RISC)
engine and dedicated, high-performance DSPs to ensure predictable, real-time voice processing. The design
couples this coprocessor with direct access to the Cisco AS5300 routing engine for streamlined packet
forwarding.

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Cisco AS5350 and Cisco AS5400 Universal Gateways

For more information, refer to the following publications:

• Hardware installation documents for Cisco AS5300
• Configuration documents for Cisco AS5300

Cisco AS5350 and Cisco AS5400 Universal Gateways

The Cisco AS5350 and Cisco AS5400 universal gateways are versatile data and voice communications
platforms that provide the functions of a gateway, router, and digital modems in a single modular chassis.
The gateways are intended for Internet service providers (ISPs), telecommunications carriers, and other service
providers that offer managed Internet connections, and also medium to large sites that provide both digital
and analog access to users on an enterprise network.
The cards that reside in the Cisco AS5350 and AS5400 chassis, sometimes referred to as dial feature cards
(DFCs), are of two types: trunk cards, which provide an E1, T1, or T3 interface, and universal port cards,
which host the universal DSPs that dynamically handle voice, dial, and fax calls.
For more information, refer to the following publications:
• Cisco AS5350 and AS5400 Universal Gateway Card Installation Guide
• Cisco AS5350 and AS5400 Universal Gateway Software Configuration Guide

Cisco AS5800
The Cisco AS5800 has two primary system components: the Cisco 5814 dial shelf (DS), which holds
channelized trunk cards and connects to the PSTN, and the Cisco 7206 router shelf (RS), which holds port
adapters and connects to the IP backbone.
The dial shelf acts as the access concentrator by accepting and consolidating all types of remote traffic,
including voice, dial-in analog and digital ISDN data, and industry-standard WAN and remote connection
types. The dial shelf also contains controller cards voice feature cards, modem feature cards, trunk cards, and
dial shelf interconnect cards.
One or two dial shelf controllers (DSCs) provide clock and power control to the dial shelf cards. Each DSC
contains a block of logic that is referred to as the common logic and system clocks. This block of logic can
use a variety of sources to generate the system timing, including an E1 or T1/T3 input signal from the BNC
connector on the front panel of the DSC. The configuration commands for the master clock specify the various
clock sources and a priority for each source (see the Clock Sources on Digital T1 E1 Voice Ports, on page
40).
The Cisco AS5800 voice feature card is a multi-DSP coprocessing board and software package that adds VoIP
capabilities to the Cisco AS5800 platform. The Cisco AS5800 voice feature card, when used with other cards
such as LAN/WAN and modem cards, provides a gateway for up to 192 packetized voice/fax calls and 360
data calls per card. A Cisco AS5800 can support up to 1344 voice calls in split-dial-shelf configuration with
two 7206VXR router shelves.
For more information, refer to the following publications:
• Cisco AS5800 Access Server Hardware Installation Guide
• Cisco AS5800 Operation, Administration, Maintenance, and Provisioning Guide

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Cisco AS5850 Universal Gateway

Cisco AS5850 Universal Gateway

The Cisco AS5850 is a high-density ISDN and port WAN aggregation system that provides both digital and
analog call termination. It is intended to be used in service-provider dial point-of-presence (POP) or
centralized-enterprise dial environments. The feature cards and the route switch controller (RSC) communicate
over a nonblocking interconnect that supports Fast Ethernet and full-duplex service.
The Cisco AS5850 contains ingress interfaces (CT3 and CE1/PRI) that terminate ISDN and modem calls and
break out individual calls (DS0s) from the appropriate telco services. Digital or ISDN calls are terminated on
the trunk-card HDLC controllers, and analog calls are sent to port resources on the same card or on separate
port cards. As a result, any DS0 can be mapped to any HDLC controller or port module. Unlike the Cisco
AS5800, trunk-termination and port-handling services can be performed on the same card in the same slot.
For more information, refer to the following publications:
• Cisco AS5850 Hardware Installation Guide
• Cisco AS5850 Universal Gateway Operations, Administration, Maintenance, and Provisioning Guide

Cisco Catalyst 6500 Series Switches and Cisco 7600 Series Routers
The Communication Media Module (CMM) acts as the VoIP gateway and media services module by using
Media Gateway Control Protocol (MGCP), H.323, and SIP protocols with Cisco CallManager and other call
agents. The CMM can support single or multiple Cisco CallManagers in an IP communication network.
These VoIP gateway and media services features are provided through the four different types of CMM port
adapters as shown in the table below.

Table 3: CMM Port Adapters

CMM Port Adapters Description

The 6-port T1 and E1 port adapters have onboard
• WS-SVC-CMM-6T1 digital signal processor (DSP) resources that allow
• WS-SVC-CMM-6E1 you to connect the interfaces to the public switched
telephone network (PSTN) or private branch
exchanges (PBXs) through T1/E1R2 Channel
Associated Signaling (CAS) or T1/E1 ISDN Primary
Rate Interface (PRI). The DSP resources on the port
adapters provide packetization, echo cancellation, fax
relay, tone detection and generation, concealment,
and jitter buffers.

WS-SVC-CMM-24FXS The 24-port FXS port adapter has onboard DSP

resources that allow the FXS interfaces to emulate
the central office (CO) or PBX analog trunk lines by
providing service to analog phones and fax machines,
which behave as if connected to a standard CO or
PBX line.

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Cisco MC3810

CMM Port Adapters Description

WS-SVC-CMM-ACT The ACT port adapter has DSP resources for
conferencing, transcoding, and media termination
point (MTP) services. A CMM with an ACT port
adapter supports a single conference with up to 64
participants. A single ACT port adapter supports up
to 128 audio conference ports, which can be
distributed among different conferences of two or
more parties.

For specific configuration information for the Catalyst 6500 series and Cisco 7600 series, see the following
documents:
• Cisco 6500 and 7600 series Manager Installation Guide, Release 2.1
• Cisco 6500 and 7600 series Manager User Guide, Release 2.1
• Cisco 6500 and 7600 series Manager Release Notes, Release 2.1

For specific installation and configuration information for the CMM, see the following document:
• Catalyst 6500 Series and Cisco 7600 Series CMM Installation and Verification Note
• Cisco Communication Media Module Voice Features for Catalyst 6500 Series and Cisco 7600 Series

Cisco MC3810
To support a T1 or E1 digital voice interface, the Cisco MC3810 must be equipped with a digital voice interface
card (DVM). The DVM interfaces with a digital PBX, channel bank, or video codec. It supports up to 24
channels of compressed digital voice at 8 kbps, or it can cross-connect channelized data from user equipment
directly onto the router’s trunk port for connection to a carrier network.
The DVM is available with a balanced interface using an RJ-48 connector or with an unbalanced interface
using BNC connectors.
Optional HCMs can replace standard VCMs to operate according to the voice compression coding algorithm
(codec) specified when the Cisco MC3810 is configured. The HCM2 provides 4 voice channels at high codec
complexity and 8 channels at medium complexity. The HCM6 provides 12 voice channels at high complexity
and 24 channels at medium complexity. You can install one or two HCMs in a Cisco MC3810, but an HCM
cannot be combined with a VCM in the same chassis.
For more information, refer to the following publications:
• Cisco MC3810 Multiservice Concentrator Hardware Installation Guide
• Cisco MC3810 Multiservice Concentrator Configuration Guide

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How to Configure Digital T1 E1 Voice Ports

This section describes commands for the basic configuration of digital voice ports. Make sure you have all
the data recommended in the Prerequisites for Configuring Digital Voice Ports, on page 20 before starting
these procedures.
The basic steps for configuring digital voice ports are described in the next three sections. They are grouped
by the configuration mode from which they are executed, as follows:

Configuring Codec Complexity on Digital T1 E1 Voice Ports

This section provides two configuration task tables: one for the Cisco 2600, Cisco 3600, and Cisco 3700 series
routers and the Cisco MC3810 concentrator, which use voice-card configuration mode, and the second for
the Cisco 7200 and Cisco 7500 series routers, which use DSP interface configuration mode. The task tables
can be found in the following sections:
Configuring Codec Complexity on Cisco 880 Series, Cisco 2600, Cisco 3600, Cisco 3700 Series and Cisco
MC3810:
Codec complexity refers to the amount of processing power assigned to a codec method on a voice port. On
most router platforms that support codec complexity, codec complexity is selected in voice-card configuration
mode, although it is selected in DSP interface mode on the Cisco 7200 and Cisco 7500 series. On the Cisco
880 series, Cisco 2600, Cisco 3600, Cisco 3700, Cisco 7200, and Cisco 7500 routers, codec complexity can
be configured separately for each T1/E1 digital packet voice trunk network module or port adapter. On a Cisco
MC3810, the codec complexity setting applies to both HCMs if two HCMs are installed.

Note On Cisco 2600, Cisco 3600, and Cisco 3700 series routers with digital T1/E1 packet voice trunk network
modules, codec complexity cannot be configured if DS0 or PRI groups are configured. If DS0 or PRI
groups are configured, see the Changing Codec Complexity, on page 29.

To configure codec complexity for digital voice ports on the Cisco 880 series, Cisco 2600 series, Cisco 3600
series, and Cisco 3700 series routers, and for voice ports on HCMs on the Cisco MC3810, use the following
commands:

SUMMARY STEPS

1. enable
2. show voice dsp
3. configure terminal
4. voice-card slot
5. codec complexity {high | medium}

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.

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Command or Action Purpose

• Enter your password if prompted.
Example:
Router> enable

Step 2 show voice dsp Checks the DSP voice channel activity. If any DSP voice channels are in
the busy state, codec complexity cannot be changed. When all DSP
Example: channels are in the idle state, continue to Step 2.

Router# show voice dsp

Step 3 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 4 voice-card slot Enters voice card-configuration mode for the card or cards in the slot
specified. Range is 0 to 5.
Example:
Router(config)# voice-card 0

Step 5 codec complexity {high | medium} Specifies codec complexity based on the codec standard being used. This
setting restricts the codecs available in dial peer configuration. All voice
Example: cards in a router must use the same codec complexity setting. Default is
medium.
Router(config-voicecard)# codec
complexity high Note On the Cisco MC3810, this command is valid only with one or
more HCMs installed, and voice card 0 must be specified. If two
HCMs are installed, this command configures both HCMs at
once.

Changing Codec Complexity

To change codec complexity on Cisco 880 Series, Cisco 2600 Series, Cisco 3600 Series, Cisco 3700 Series,
and Cisco MC3810 after the controller and voice ports have already been configured, use the following
commands:

Note Use the show voice dsp command to check the DSP voice channel activity. If any DSP voice channels
are in the busy state, the codec complexity cannot be changed. You must clear all calls before performing
the following task.

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SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port slot / port:ds0-group-number
4. shutdown
5. exit
6. controller {t1 | e1} slot/port
7. Do one of the following:
• no ds0-group ds0-group-number
•
•
• no pri-group timeslots timeslot-list

8. exit
9. voice-card slot
10. codec complexity {high | medium} [ecan-extended]
11. exit
12. Repeat Step 6, then continue with Step 13.
13. Do one of the following:
• ds0-group ds0-group-number timeslots timeslot - list type {e&m-immediate | e&m-delay
| e&m-wink-start | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-start}
•
•
• pri-group timeslots timeslot - list

14. exit
15. Repeat Step 3, then continue with Step 16.
16. no shutdown
17. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

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Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-port slot / port:ds0-group-number Enters voice-port configuration mode on the selected slot, port, and
DS0 group.
Example: Note The syntax of this command is platform-specific. For the
Router(config)# voice-port 1/0:23 syntax for your platform, refer to the Cisco IOS Voice
Command Reference.
Note For the Cisco 880 series platforms, the command syntax
does not include a slot number, only the port is identified.
If the primary voice interface is FXS and the backup is BRI,
then ports 0, 1, 2, and 3 are analog voice ports, and ports 4
and 5 are digital. If the primary voice interface is BRI, then
ports 1, 2, 3, and 4 are digital.
Step 4 shutdown Shuts down all voice ports assigned to the T1 interface on the voice
card.
Example:
Router(config-voiceport)# shutdown

Step 5 exit Exits voice-port configuration mode.

Example:
Router(config-voiceport)# exit

Step 6 controller {t1 | e1} slot/port Enters controller configuration mode on the T1 controller on the
selected slot and port.
Example:
Router(config)# controller t1 1/0

Step 7 Do one of the following: Removes the related DS0 groups.

• no ds0-group ds0-group-number or

• Removes the related PRI group.

•
• no pri-group timeslots timeslot-list

Example:
Router(config-controller)# no ds0-group
1

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Command or Action Purpose

Example:
Router(config-controller)# no pri-group
timeslots 1,7,9

Step 8 exit Exits controller configuration mode and returns to global configuration
mode.
Example:
Router(config-controller) exit

Step 9 voice-card slot Enters voice-card configuration mode on the specified slot.
• slot--Slot number of the voice card. Range is 0 to 6, depending
Example: on platform.
Router(config)# voice-card 1

Step 10 codec complexity {high | medium} Changes codec complexity or changes the echo canceller (EC) from
[ecan-extended] the proprietary Cisco G.165 EC to the G.168 extended EC.
• high --Supports up to six voice or fax calls per DSP module
Example: (PVDM-12), using the codecs: G.723, G.728, G.729, G.729
Router(voice-card)# codec complexity high Annex B, GSMEFR, GSMFR, fax relay, or any of the medium
ecan-extended complexity codecs.
• medium --Supports up to 12 voice or fax calls per DSP module
Example: (PVDM-12), using the codecs: G.711, G.726, G.729 Annex A,
G.729 Annex A with Annex B, and fax relay. Default value.
• ecan-extended --(Optional) Selects the G.168 extended echo
canceller. For more information, see the "How to Configure the
Extended G.168 Echo Canceller" section.

Specifying the codec complexity restricts the codecs available in

dial-peer configuration mode. All voice cards in a gateway must use
the same codec complexity.

Step 11 exit Exits voice-card configuration mode and returns to global

configuration mode.
Example:
Router(voice-card) exit

Step 12 Repeat Step 6, then continue with Step 13. --

Step 13 Do one of the following: Defines the T1 or E1 channels for use by compressed voice calls and
the signaling method that the router uses to connect to the PBX or
• ds0-group ds0-group-number timeslots CO.
timeslot - list type {e&m-immediate |
e&m-delay | e&m-wink-start | Note If you are configuring PRI groups instead of DS0 groups,
fxs-ground-start | fxs-loop-start | omit this step and proceed to Step 15.
fxo-ground-start | fxo-loop-start} or

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Command or Action Purpose

• Specifies an ISDN PRI on a channelized T1 or E1 controller.
• Note When configuring PRI groups, you must also configure the
• pri-group timeslots timeslot - list isdn switch-type command. Also, only one PRI group can
be configured on a controller.

Example:
Router(config-controller)# ds0-group 0
timeslots 1-24 type e&m-wink-start

Example:
Router(config-controller)# pri-group
timeslots 1,7,9

Step 14 exit Exits controller configuration mode and completes the process for
adding back the PRI groups or DS0 groups.
Example:
Router(config-controller)# exit

Step 15 Repeat Step 3, then continue with Step 16. --

Step 16 no shutdown Saves the controller configurations on the slot and port specified.

Example:
Router(config-controller)# no shutdown

Step 17 end Exits controller configuration mode and completes the process for
bringing the T1 controller back up.
Example:
Router(config-controller)# end

Configuring the Flex Option on Codec Complexity

The IP Communications High-Density Digital Voice/Fax Network Module feature enables the flex option for
configuring codec complexity.
On the Cisco 2600 XM, Cisco 2691, Cisco 3700 series routers, codec complexity can be configured using the
flex option for configuring codec complexity. This option allows the DSP to process up to 16 channels. In
addition to continuing support for configuring a fixed number of channels per DSP, the flex option enables
the DSP to handle a flexible number of channels. The total number of supported channels varies from 6 to
16, depending on which codec is used for a call. Therefore, the channel density varies from 6 per DSP
(high-complexity codec) to 16 per DSP (g.711 codec).
The following requirements apply to the IP Communications High-Density Digital Voice/Fax Network Module
feature.

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• When the IP Communications High-Density Digital Voice/Fax Network Module feature is used in a
Cisco CallManager network, the CCM 4.0(1) SR1 or CCM 3.3(4) release must be installed.
• Software echo cancellation is the default configuration--G.168-compliant echo cancellation is enabled
by default with a coverage of 64 milliseconds.
• Only Packet Fax/Voice DSP modules (PVDM2s) are supported on the IP Communications High-Density
Digital Voice/Fax Network Module.
• Only voice interface cards that start with VIC2 are supported in the IP Communications High-Density
Digital Voice/Fax Network Module feature except for VIC-1J1, VIC-2DID, and VIC-4FXS/DID.
• The direct inward dial (DID) feature in VIC-4FXS/DID is not supported.
• The CAMA card (VIC-2CAMA) is not supported. Any port on the VIC2-2FXO and the VIC2-4FXO
can be software configured to support analog CAMA for dedicated E-911 services (North America only).

Codec Combinations for DSP Sharing:

When network modules or PVDM2s on the motherboard are configured for DSP sharing, the codec complexity
has to match. A local resource sharing or importing from a remote network module must match its
characteristics, that is, a high-complexity network module can only share from another high-complexity
network module, whereas a flex-complexity network module can share DSPs from both high-complexity and
flex-complexity network modules. The table below summarizes the codec combinations for DSP-sharing.
Using Flex Mode
In flex mode, you can connect (or configure in the case of DS0 groups and PRI groups) more voice channels
to the module than the DSPs can accommodate. This is referred to as oversubscription. If all voice channels
should go active simultaneously, the DSPs will be oversubscribed and calls that are unable to allocate a DSP
resource will fail to connect.

Caution If you are configuring a Cisco 2600 XM router, you should not use the network-clock-participate
command for slot 1 of the router. This may cause a disruption in service to the router.

Table 4: Codec Complexity Settings for DSP Resource Sharing Between Local and Remote Sources

Local DSP Resource Remote DSP Resource

(Import) (Export)
High complexity Medium complexity Flexible complexity

High complexity Yes No No

Medium complexity Yes Yes No

Flexible complexity Yes No Yes

To enable the IP Communications Voice/Fax Network Module feature, perform this task to configure the
voice card for the flex option in codec complexity.

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SUMMARY STEPS

1. enable
2. configure terminal
3. voice-card slot
4. codec complexity flex [reservation - fixed {high | medium}]
5. voice local-bypass
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-card slot Enters voice-card configuration mode and specifies the slot location.
• For the slotargument, specify a value from 1 to 4, depending on your router.
Example:
Router(config)# voice-card 1

Step 4 codec complexity flex Specifies the flex option for codec complexity.
[reservation - fixed {high | medium}]
• flex --Up to 16 calls can be completed per DSP. The number of supported
calls varies from 6 to 16, depending on the codec used for a call. In this
Example: mode, reservation for analog VICs may be needed for certain appplications
Router(config-voicecard)# codec such as CAMA E-911 calls because oversubscription of DSPs is possible.
complexity flex If this is true, then the reservation-fixed option may be enabled. There is
no reservation by default.
• reservation-fixed--Appears as an option only when there is an analog
VIC present. Ensures that sufficient DSP resources are available to
handle a call. If you enter this keyword, then specify if the complexity
should be high or medium.

Note You cannot change codec complexity while DS0 groups are defined.
If they are already set up, perform the steps in the Changing Codec
Complexity, on page 29.
Step 5 voice local-bypass Configures local calls to bypass the DSP. This is the default.

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Command or Action Purpose

• Using this command enables intranetwork-module hairpinning (no DSPs).
Example:
Note For POTS-to-POTS calls between two network modules, hairpinning
Router(config-voicecard)# voice is not supported. If the connection manager in Cisco IOS software does
local-bypass
not automatically handle this, it might be necessary to disable
local-bypass so that DSPs are used for these calls.
Step 6 exit Exits voice-card configuration mode and returns the router to global configuration
mode.
Example:
Router(config-voicecard)# exit

Configuring Codec Complexity

On Cisco 7200 series and Cisco 7500 series routers, codec complexity is configured in the DSP interface.

Note Use the show interfaces dspfarmcommand to check the DSP voice channel activity. If any DSP voice
channels are in the busy state, the codec complexity cannot be changed. You must clear all calls before
performing the following task.

SUMMARY STEPS

1. enable
2. configure terminal
3. Do one of the following:
• dspint dspfarm slot /0
•
•
• dspint dspfarm slot / port-adapter / port

4. codec {high | medium} [ecan-extended]

5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.

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Command or Action Purpose

• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 Do one of the following: Enters DSP interface configuration mode for the Cisco 7200 series.
• dspint dspfarm slot /0 or

• Enters DSP interface configuration mode for the Cisco 7500 series.
•
• dspint dspfarm slot / port-adapter /
port

Example:
Router(config)# dspint dspfarm 2/0

Step 4 codec {high | medium} [ecan-extended] Sets the codec complexity.

• The optional ecan-extended keyword selects the G.168
Example: extended echo canceller. This keyword is supported only in
Router(config-dspfarm)# codec medium Cisco IOS Release 12.2(13)T. For more information, see the
ecan-extended "How to Configure the Extended G.168 Echo Canceller"
section.
• This command affects the choice of codecs available when
the codec command is used in dial-peer configuration mode.

Step 5 exit Exits to global configuration mode.

Example:
Router(config-dspfarm)# exit

What to Do Next
Cisco 7200 Series:
On the Cisco 7200 series, the PA-MCX-2TE1 port adapter (PA) card can be used for making voice calls. This
PA does not have any DSPs but uses the DSP resources of the PA-VXC-2TE1+ card present in another slot.
If the PA-MCX card is used, codec complexity is configured for PA-VXC, while all other echo cancellation
configurations are done for PA-MCX.

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The PA-MCX card borrows the DSP resources from the PA-VXC, PA-VXB, or PA-VXA card. If one of the
PA-VXC, PA-VXB, or PA-VXA cards has extended echo cancellation configured on the DSP interface,
extended echo cancellation is enabled for the PA-MCX card. It is recommended that you have the same codec
complexity and echo cancellation configuration on all the PA-VXC, PA-VXB, or PA-VXA cards in the router.
Cisco AS5300:
Codec support on the Cisco AS5300 is determined by the capability list on the voice feature card, which
defines the set of codecs that can be negotiated for a voice call. The capability list is created and populated
when VCWare is unbundled and DSPWare is added to VFC flash memory. The capability list does not indicate
codec preference; it simply reports the codecs that are available. The session application decides which codec
to use. Codec support is configured on dial peers rather than on voice ports; refer to the "Dial Peer Configuration
on Voice Gateway Routers" document.
Cisco AS5800:
Codec support is selected on Cisco AS5800 access servers during dial peer configuration. Refer to the "Dial
Peer Configuration on Voice Gateway Routers" document.

Configuring Controller Settings for Digital T1 E1 Voice Ports

The controller configuration for digital T1/E1 voice ports must match the line characteristics of the telephony
network connection so that voice and signaling can be transferred between them and so that logical voice
ports, or DS0 groups, may be established.
Specific line characteristics must be configured to match those of the PSTN line that is being connected to
the voice port. These are typically configured in controller configuration mode.
The figure below shows how a ds0-group command gathers some of the DS0 time slots from a T1 line into
a group that becomes a single logical voice port that can later be addressed as a single entity in voice port
configurations. Other DS0 groups for voice can be created from the remaining time slots shown in the figure,
or the time slots can be used for data or serial pass-through.

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Note All controller commands shown in the figure below, other than ds0-group, apply to all time slots in the
T1 line.

Figure 7: T1 Controller Configuration on Cisco 2600 or Cisco 3600 Series Routers

Voice port controller configuration includes setting the parameters described in the following sections:
Another controller command that might be needed, cablelength, is discussed in the Cisco IOS Interface and
Hardware Component Command Reference.

Framing Formats on Digital T1 E1 Voice Ports

The framing format parameter describes the way that bits are robbed from specific frames to be used for
signaling purposes. The controller must be configured to use the same framing format as the line from the
PBX or CO that connects to the voice port you are configuring.
Digital T1 lines use SF or ESF framing formats. SF provides two-state, continuous supervision signaling, in
which bit values of 0 are used to represent on-hook and bit values of 1 are used to represent off-hook. ESF
robs four bits instead of two, yet has little impact on voice quality. ESF is required for 64-kbps operation on
DS0 and is recommended for PRI configurations.
E1 lines can be configured for CRC4 or no cyclic redundancy check, with an optional argument for E1 lines
in Australia.

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Clock Sources on Digital T1 E1 Voice Ports

Digital T1/E1 interfaces use timers called clocks to ensure that voice packets are delivered and assembled
properly. All interfaces handling the same packets must be configured to use the same source of timing so
that packets are not lost or delivered late. The timing source that is configured can be external (from the line)
or internal to the router’s digital interface.
If the timing source is internal, timing derives from the onboard phase-lock loop (PLL) chip in the digital
voice interface. If the timing source is line (external), then timing derives from the PBX or PSTN CO to which
the voice port is connected. It is generally preferable to derive timing from the PSTN becauseits clocks are
maintained at an extremely accurate level. This is the default setting for the clocks. When two or more
controllers are configured, one should be designated as the primary clock source; it will drive the other
controllers.
The line keyword specifies that the clock source is derived from the active line rather than from the free-running
internal clock. The following rules apply to clock sourcing on the 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.

This section describes the five basic timing scenarios that can occur when a digital voice port is connected to
a PBX or CO. In all the examples that follow, the PSTN (or CO) and the PBX are interchangeable for purposes
of providing or receiving clocking.
• Single voice port providing clocking--In this scenario, the digital voice hardware is the clock source for
the connected device, as shown in the figure below. The PLL generates the clock internally and drives
the clocking on the line. 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 because CO clocking is
much more reliable. The following configuration sets up this clocking method for a digital E1 voice
port:

controller E1 1/0
framing crc4
linecoding hdb3
clock source internal
ds0-group timeslots 1-15 type e&m-wink-start

Figure 8: Single Voice Port Providing Clocking

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• Single voice port receiving internal clocking--In this scenario, the digital voice hardware receives clocking
from the connected device (CO telephony switch or PBX) (see the figure below). The PLL clocking is
driven by the clock reference on the receive (Rx) side of the digital line connection.

Figure 9: Single E1 Port Receiving Clocking from the Line

The following configuration sets up this clocking method:

controller T1 1/0
framing esf
linecoding ami
clock source line
ds0-group timeslots 1-12 type e&m-wink-start

• Dual voice ports receiving clocking from the Line--In this scenario, the digital voice port has two
reference clocks, one from the PBX and another from the CO, as shown in the figure below.

Figure 10: Dual E1 Ports Receiving Clocking from the Line

Because the PLL can derive clocking from only one source, this case is more complex than the two preceding
examples. Before looking at the details, consider the following as they pertain to the clocking method:
• • Looped-time clocking--The voice port takes the clock received on its Rx (receive) pair and
regenerates it on its Tx (transmit) pair. While the port receives clocking, the port is not driving the
PLL on the card but is "spoofing" (that is, fooling) the port so that the connected device has a
viable clock and does not see slips (that is, loss of data bits). PBXs are not designed to accept slips
on a T1 or E1 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’s voice port.
• Slips--These messages indicate that the voice port is receiving clock information that is out of
phase (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.

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Note Physical layer issues, such as bad cabling or faulty clocking references, can cause slips. Eliminate these
slips by addressing the physical layer or clock reference problems.

In the dual voice ports receiving clocking from the line scenario, the PLL derives clocking from the CO and
puts the voice port connected to the PBX into looped-time mode. This is usually the best method because the
CO provides an excellent clock source (and the PLL usually requires that the CO provide that source) and a
PBX usually must receive clocking from the other voice port.
The following configuration sets up this clocking method (controller E1 1/0 is connected to the CO; controller
E1 1/1 is connected to the PBX:

controller E1 1/0
framing crc4
linecoding hdb3
clock source line primary
ds0-group timeslots 1-15 type e&m-wink-start
!
controller E1 1/1
framing crc4
linecoding hdb3
clock source line
ds0-group timeslots 1-15 type e&m-wink-start
The clock source line primary command tells the router to use this voice port to drive the PLL. All other
voice ports configured as clock source line are then put into an implicit loop-timed mode. If the primary voice
port fails or goes down, the other voice port instead receives the clock that drives the PLL. In this configuration,
port 1/1 might see controlled slips, but these should not force it down. This method prevents the PBX from
seeing slips.

Note When two T1/E1 lines terminate on a two-port interface card, such as the VWIC-2MFT, and both controllers
are set for line clocking but the lines are not within clocking tolerance of one another, one of the controllers
is likely to experience slips. To prevent slips, ensure that the two T1 or E1 lines are within clocking
tolerance of one another, even if the lines are from different providers.

• Dual voice ports (one receives clocking and one provides clocking)--In this scenario, the digital voice
hardware receives clocking for the PLL from E1 0 and uses this clock as a reference to clock E1 1 (see
the figure below). If controller E1 0 fails, the PLL internally generates the clock reference to drive E1
1.

Figure 11: Dual E1 Ports--One Receiving and One Providing Clocking

The following configuration sets up this clocking method:

controller E1 1/0

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framing crc4
linecoding hdb3
clock source line
ds0-group timeslots 1-15 type e&m-wink-start
!
controller E1 1/1
framing crc4
linecoding hdb3
clock source internal
ds0-group timeslots 1-15 type e&m-wink-start

• Dual voice ports (router provides both clocks)--In this scenario, the router generates the clock for the
PLL and, therefore, for both voice ports (see the figure below).

Figure 12: Dual E1 Ports--Both Clocks from the Router

The following configuration sets up this clocking method:

controller E1 1/0
framing crc4
linecoding hdb3
clock source internal
ds0-group timeslots 1-15 type e&m-wink-start
!
controller E1 1/1
framing esf
linecoding b8zs
clock source internal
ds0-group timeslots 1-15 type e&m-wink-start

Network Clock Timing

Voice systems that pass digitized (pulse code modulation or PCM) speech have always relied on the clocking
signal being embedded in the received bit stream. This reliance allows connected devices to recover the clock
signal from the bit stream, and then use this recovered clock signal to ensure that data on different channels
keep the same timing relationship with other channels.
If a common clock source is not used between devices, the binary values in the bit streams may be misinterpreted
because the device samples the signal at the wrong moment. As an example, if the local timing of a receiving
device is using a slightly shorter time period than the timing of the sending device, a string of eight continuous
binary 1s may be interpreted as nine continuous 1s. If this data is then re-sent to further downstream devices
that used varying timing references, the error could be compounded. By ensuring that each device in the
network uses the same clocking signal, you can ensure the integrity of the traffic.
If timing between devices is not maintained, a condition known as clock slip can occur. Clock slip is the
repetition or deletion of a block of bits in a synchronous bit stream due to a discrepancy in the read and write
rates at a buffer.

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Slips are caused by the inability of an equipment buffer store (or other mechanisms) to accommodate differences
between the phases or frequencies of the incoming and outgoing signals in cases where the timing of the
outgoing signal is not derived from that of the incoming signal.
A T1 or E1 interface sends traffic inside repeating bit patterns called frames. Each frame is a fixed number
of bits, allowing the device to see the start and end of a frame. The receiving device also knows exactly when
to expect the end of a frame simply by counting the appropriate number of bits that have come in. Therefore,
if the timing between the sending and receiving device is not the same, the receiving device may sample the
bit stream at the wrong moment, resulting in an incorrect value being returned.
Even though Cisco IOS software can be used to control the clocking on these platforms, the default clocking
mode is effectively free running, meaning that the received clock signal from an interface is not connected to
the backplane of the router and used for internal synchronization between the rest of the router and its interfaces.
The router will use its internal clock source to pass traffic across the backplane and other interfaces.
For data applications, this clocking generally does not present a problem as a packet is buffered in internal
memory and is then copied to the transmit buffer of the destination interface. The reading and writing of
packets to memory effectively removes the need for any clock synchronization between ports.
Digital voice ports have a different issue. It would appear that unless otherwise configured, Cisco IOS software
uses the backplane (or internal) clocking to control the reading and writing of data to the DSPs. If a PCM
stream comes in on a digital voice port, it will be using the external clocking for the received bit stream.
However, this bit stream will not necessarily be using the same reference as the router backplane, meaning
the DSPs may misinterpret the data coming in from the controller.
This clocking mismatch is seen on the router’s E1 or T1 controller as a clock slip--the router is using its internal
clock source to send the traffic out the interface but the traffic coming in to the interface is using a completely
different clock reference. Eventually, the difference in the timing relationship between the transmit and receive
signal becomes so great that the controller registers a slip in the received frame.
To eliminate the problem, change the default clocking behavior through Cisco IOS configuration commands.
It is absolutely critical to set up the clocking commands properly.
Even though these commands are optional, we strongly recommend you enter them as part of your configuration
to ensure proper network clock synchronization:
network-clock-participate [slot slot-number | wic wic-slot | aim aim-slot-number network-clock-select
priority{bri | t1 | e1} slot / port
The network-clock-participate command allows the router to use the clock from the line via the specified
slot/WIC/AIM and synchronize the onboard clock to the same reference.
If multiple VWICS are installed, the commands must be repeated for each installed card. The system clocking
can be confirmed using the show network clocks command.

Caution If you are configuring a Cisco 2600 XM voice gateway with an NM-HDV2 or NM-HD-2VE installed in
slot 1, do not use the network-clock-participate slot 1 command in the configuration. In this particular
hardware scenario, the network-clock-participate slot 1 command is not necessary. If the
network-clock-participate slot 1 command is configured, voice and data connectivity on interfaces
terminating on the NM-HDV2 or NM-HD-2VE network module may fail to operate properly. Data
connectivity to peer devices may not be possible, and even loopback plug tests to the serial interface
spawned via a channel group configured on the local T1/E1 controller will fail. Voice groups such as CAS
DS0 groups and ISDN PRI groups may fail to signal properly. The T1/E1 controller may accumulate large
amounts of timing slips and Path Code Violations (PCVs) and Line Code Violations (LCVs).

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Line Coding on Digital T1 E1 Voice Ports

Digital T1/E1 interfaces require that line encoding be configured to match that of the PBX or CO that is being
connected to the voice port. Line encoding defines the type of framing used on the line.
T1 line encoding methods include AMI and B8ZS. AMI is used on older T1 circuits and references signal
transitions with a binary 1, or "mark." B8ZS, a more reliable method, is more popular and is recommended
for PRI configurations as well. B8ZS encodes a sequence of eight zeros in a unique binary sequence to detect
line-coding violations.
Supported E1 line encoding methods are AMI and HDB3, which is a form of zero-suppression line coding.

DS0 Groups on Digital T1 E1 Voice Ports

For digital voice ports, a single command, ds0-group, performs the following functions:
• Defines the T1/E1 channels for compressed voice calls.
• Automatically creates a logical voice port.

The numbering for the logical voice port created as a result of this command is controller:ds0-group-number
, where controller is defined as the platform-specific address for a particular controller. On a Cisco 3640
router, for example, ds0-group 1 timeslots 1-24 type e&m-wink automatically creates the voice port 1/0:1
when issued in the configuration mode for controller 1/0. On a Cisco MC3810 universal concentrator, when
you are in the configuration mode for controller 0, the ds0-group 1 timeslots 1-24 type e&m-winkcommand
creates logical voice port 0:1.
To map individual DS0s, define additional DS0 groups under the T1/E1 controller, specifying different time
slots. Defining additional DS0 groups also creates individual DS0 voice ports.
• Defines the emulated analog signaling method that the router uses to connect to the PBX or PSTN.

Most digital T1/E1 connections used for switch-to-switch (or switch-to-router) trunks are E&M connections,
but FXS and FXO connections are also supported. These are normally used to provide emulated-OPX
(Off-Premises eXtension) from a PBX to remote stations. FXO ports connect to FXS ports. The FXO or FXS
connection between the router and switch (CO or PBX) must use matching signaling, or calls cannot connect
properly. Either ground-start or loop-start signaling is appropriate for these connections. Ground-start provides
better disconnect supervision to detect when a remote user has hung up the telephone, but ground-start is not
available on all PBXs.
Digital ground start differs from digital E&M because the A and B bits do not track each other as they do in
digital E&M signaling (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.
E&M connections can use one of three different signaling types to acknowledge on-hook and off-hook states:
wink start, immediate-start, and delay-start. E&M wink start is usually preferred, but not all COs and PBXs
can handle wink-start signaling. The E&M connection between the router and switch (CO or PBX) must
match the CO or PBX E&M signaling type, or calls cannot be connected properly.
E&M signaling is normally 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 (robbed-bit signaling). If dial pulse dialing is used, the A and B bits are pulsed to indicate the addressing
digits. The are several further important subclasses of E&M robbed-bit signaling:

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• • E&M wink-start--Feature Group B

In the original wink start handshaking 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 handshaking 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. This wink tells the originating side that the terminating side is ready
to receive addressing digits. After receiving addressing digits, the terminating side 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. This last indication can be due
to the ultimate called endpoint’s having answered. The originating endpoint maintains an off-hook condition
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.

Note Feature Group D is supported on Cisco AS5300 platforms, and on Cisco 2600, Cisco 3600, and Cisco
7200 series with digital T1 packet voice trunk network modules. Feature Group D is not supported on E1
or analog voice ports.

To configure controller settings for digital T1/E1 voice ports, use the following commands:

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SUMMARY STEPS

1. enable
2. configure terminal
3. card type {t1 | e1} slot
4. Do one of the following:
• controller {t1 | e1} slot / port
•
• controller {t1 | e1} number
•
• controller {t1 | e1} shelf / slot / port

5. Do one of the following:

• framing {sf | esf}
•
•
• framing {crc4 | no-crc4} [australia]

6. clock source {line [primary | secondary] | internal}

7. Do one of the following:
• linecode {ami | b8zs}
•
• linecode {ami | hdb3}

8. ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd |

e&m-immediate-start|e&m-wink-start | ext-sig | fgd-eana | fxo-ground-start | fxo-loop-start |
fxs-ground-start | fxs-loop-start}
9. no shutdown

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 card type {t1 | e1} slot Defines the card as T1 or E1 and identifies the location.

Example:
Router(config)# card type t1 0

Step 4 Do one of the following: Enters controller configuration mode and specifies either T1
or E1 for the line.
• controller {t1 | e1} slot / port
• For the Cisco 2600, Cisco 3600 series, Cisco MC3810,
•
and Cisco 7200 series, identifies the slot and port.
• controller {t1 | e1} number
• For the Cisco AS5300, identifies the port number.
•
• controller {t1 | e1} shelf / slot / port • For the Cisco AS5800 and Cisco 7500 series, identifies
the shelf, slot, and port number.

Example:
Router(config)# controller t1 1/0

Example:

Example:
Router(config)# controller t1 1

Example:
or

Example:
Router(config)# controller t1 1/0/0

Step 5 Do one of the following: Selects frame type for T1 or E1 line.

• framing {sf | esf} • For T1, the frame type can be sf or esf. Default for T1
is sf.
•
• • For E!, the frame type can be crc4 or no crc4 or
• framing {crc4 | no-crc4} [australia] australia. Default for E1 is crc4.

Example:
Router(config-controller)# framing esf

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Command or Action Purpose

Example:

Example:

Example:
Router(config-controller)# framing crc4

Step 6 clock source {line [primary | secondary] | internal} Configures the clock source.
• Default is line.
Example:
• For more information about clock sources, see the
Router(config-controller)# clock source line
primary Clock Sources on Digital T1 E1 Voice Ports, on page
40.

Step 7 Do one of the following: Specifies the line encoding to use for T1 or E1 line.
• linecode {ami | b8zs} • For T1, the line encoding can be ami or b8zs. Default
for T1 is ami.
•
• linecode {ami | hdb3} • For E1, the line encoding can be ami or hdb3. Default
for E1 is hdb3.

Example:
Router(config-controller)# linecode b8zs

Example:

Example:
Router(config-controller)# linecode hdb3

Step 8 ds0-group ds0-group-number timeslots timeslot-list Defines the T1 channels for use by compressed voice calls
type {e&m-delay-dial | e&m-fgd | and the signaling method that the router uses to connect to
e&m-immediate-start|e&m-wink-start | ext-sig | the PBX or CO.
fgd-eana | fxo-ground-start | fxo-loop-start | Note This step shows the basic syntax and signaling types
fxs-ground-start | fxs-loop-start} available with the ds0-group command. For the
complete syntax, refer to the Cisco IOS Voice
Example: Command Reference.
Router(config-controller)# ds0-group 30 timeslots
0 type e&m-immediate-start

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Command or Action Purpose

Step 9 no shutdown Activates the controller.

Example:
Router(config-controller)# no shutdown

Configuring Basic Voice Port Parameters for Digital T1 E1 Voice Ports

For FXO and FXS connections the default voice-port parameter values are often adequate. However, for E&M
connections, it is important to match the characteristics of your PBX, so voice port parameters may need to
be reconfigured from their defaults.
Each voice port that you address in digital voice port configuration is one of the logical voice ports that you
created with the ds0-group command.
Companding (from compression and expansion), used in Step 6 of the following table, is the part of the PCM
process in which analog signal values are logically rounded to discrete scale-step values on a nonlinear scale.
The decimal step number is then coded in its binary equivalent prior to transmission. The process is reversed
at the receiving terminal using the same nonlinear scale.
Voice-port configuration mode allows many of the basic voice call attributes to be configured to match those
of the PSTN or PBX connection being made on this voice port.
In addition to the basic voice port parameters, there are commands that allow for the fine- tuning of the voice
port configurations or for configuration of optional features. In most cases, the default values for these
commands are sufficient for establishing voice port configurations. If it is necessary to change some of these
parameters to improve voice quality or to match parameters in proprietary PBXs to which you are connecting,
use the commands in the "Fine-Tuning Analog and Digital Voice Ports" section.
After voice port configuration, make sure the ports are operational by following the steps described in these
chapters:
For more information on voice port commands, refer to the Cisco IOS Voice Command Reference

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 to determine the syntax choices that are available.

To configure basic parameters for digital T1/E1 voice ports, use the following commands:

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 Configuring Digital Voice Ports
Configuring Controller Settings for Digital T1 E1 Voice Ports

SUMMARY STEPS

1. enable
2. configure terminal
3. Do one of the following:
• voice-port port
•
• voice-port slot / port:ds0-group-number
•
• voice-port slot / port-adapter :ds0-group-number
•
• voice-port slot / port-adapter/slot :ds0-group-number
•
• voice-port controller :{ds0-group-number | D}
•
• voice-port slot / controller :{ds0-group-number | D}
•
• voice-port shelf / slot / port:ds0-group-number

4. type {1 | 2 | 3 | 5}
5. cptone locale
6. compand-type {u-law | a-law}
7. ring frequency {25 | 50}
8. ring number number
9. ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07 |
pattern08 | pattern09 | pattern10 | pattern11 | pattern12] [define pulse interval]}
10. description string
11. no shutdown

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

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Configuring Controller Settings for Digital T1 E1 Voice Ports

Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 Do one of the following: Enters voice-port configuration mode and identifies the port to be
configured.
• voice-port port
• For the Cisco 880 series, specify the port number.
•
• voice-port slot / • For the Cisco 2600, Cisco 3600, and Cisco 3700 series, specify the
port:ds0-group-number slot, port, and DS0 group number.

• • For the Cisco 7200 series, specify the slot, port adapter,and DS0
• voice-port slot / port-adapter group number.
:ds0-group-number • For the Cisco 7500 series, specify the slot, port adapter, slot, and DS0
• group number.
• voice-port slot / port-adapter/slot • For the Cisco AS5300, specify the controller and DS0 group number
:ds0-group-number or the keyword D.
• • For the Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal
• voice-port controller gateways, specify the slot, controller, and DS0 group number or the
:{ds0-group-number | D} keyword D.

• • For the Cisco AS5800, specify the shelf, slot, port, and DS0 group
• voice-port slot / controller number.
:{ds0-group-number | D}
•
• voice-port shelf / slot /
port:ds0-group-number

Example:
Router(config)# voice-port 1:0

Example:

Example:
Router(config)# voice-port 1/1:0

Example:
Router(config)# voice-port 1/1/1:1

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Configuring Controller Settings for Digital T1 E1 Voice Ports

Command or Action Purpose

Example:
Router(config)# voice-port 1:1

Example:
Router(config)# voice-port 1/0 D

Example:
Router(config)# voice-port 1/2/0:1

Step 4 type {1 | 2 | 3 | 5} (E&M only) Specifies the type of E&M interface to which this voice port
is connected. See Table 3 in the "Voice Port Configuration Overview"
Example: chapter for an explanation of E&M types.

Router(config-voiceport)# • Default is 1.
type 1

Step 5 cptone locale Selects a two-letter locale keyword for the voice call progress tones and
other locale-specific parameters to be used on this voice port. Voice call
Example: progress tones include dial tone, busy tone, and ringback tone, which vary
with geographical region.
Router(config-voiceport)# cptone us
• Other parameters include ring cadence and compand type. Cisco
routers comply with the ISO3166 locale name standards; to see valid
choices, enter a question mark (?) following the cptone command.
• Default is us.

Step 6 compand-type {u-law | a-law} (Cisco 2600 and Cisco 3600 series routers.) Specifies the companding
standard used. This command is used in cases when the DSP is not used,
Example: such as local cross-connects, and overwrites the compand-type value set
by the cptone command.
Router(config-voiceport)# compand-type
u-law • The default for E1 is a-law.
• The default for T1 is u-law.

Note If you have a 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 compand-type a-law
command configured. However, if you request a list of commands,
the compand-type a-law command will display.

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Configuring Controller Settings for Digital T1 E1 Voice Ports

Command or Action Purpose

Step 7 ring frequency {25 | 50} (FXS only) Selects the ring frequency, in hertz, used on the FXS interface.
This number must match the connected telephony equipment, and can be
Example: country-dependent. If the ring frequency is not set properly, the attached
telephony device may not ring or it may buzz.
Router(config-voiceport)# ring
frequency 50 • Default is 25.

Step 8 ring number number (FXO only) Specifies the maximum number of rings to be detected before
an incoming call is answered by the router.
Example: • Default is 1.
Router(config-voiceport)# ring number
1

Step 9 ring cadence {[pattern01 | pattern02 | (FXS only) Specifies an existing pattern for ring, or defines a new one.
pattern03 | pattern04 | pattern05 | pattern06 Each pattern specifies a ring-pulse time and a ring-interval time. The
| pattern07 | pattern08 | pattern09 | keywords and arguments are as follows:
pattern10 | pattern11 | pattern12] [define
pulse interval]} • pattern01 through pattern12--Specifies preset ring cadence
patterns. Enter ring cadence ? to see ring pattern explanations.
Example: • define pulse interval --Specifies a user-defined pattern as follows:
Router(config-voiceport)# ring cadence • pulse is a number (1 or 2 digits from 1 to 50) specifying ring
pattern01 define 12 15 pulse (on) time in hundreds of milliseconds.
• interval is a number (1 or 2 digits from 1 to 50) specifying ring
interval (off) time in hundreds of milliseconds.

• The default is the pattern specified by the configured cptone locale

command.

Step 10 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
Example: for tracking the purpose or use of the voice port. The string argument is a
character string from 1 to 255 characters in length.
Router(config-voiceport)# description
1 • The default is that no description is attached to the configuration.

Step 11 no shutdown Activates the voice port.

Example:
Router(config-voiceport)# no shutdown

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 CHAPTER 4
Fine-Tuning Analog and Digital Voice Ports
The default parameter values for voice ports are usually sufficient for most networks. Depending on the
specifics of your particular network, however, you may need to adjust certain parameters that are configured
on voice ports. Collectively, these commands are referred to as voice port tuning commands.

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 to determine the syntax choices that are available.

• Finding Feature Information, page 55

• Information About Fine-Tuning Analog and Digital Voice Ports, page 55
• How to Configure Fine-Tuning Features for Voice Ports, page 56

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Information About Fine-Tuning Analog and Digital Voice Ports

• Channel Bank Support for T1/E1 Voice Ports--Provides support for the time-division multiplexing
(TDM) cross-connect functionality between analog voice ports and digital DS0s on the same NM-HD-2VE
using channel associated signaling (CAS).
• Auto Cut-Through--Allows you to connect to PBXs that do not provide an M-lead response.

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How to Configure Fine-Tuning Features for Voice Ports

• Modification of Bit Patterns for Digital Voice Ports--Enables commands for digital voice ports to modify
sent or received bit patterns. Different versions of E&M use different ABCD signaling bits to represent
idle and seize.
• ANI for Outbound Calling--Allows the automatic number identification (ANI) to be sent for outgoing
calls on the Cisco AS5300 (if T1 CAS is configured with the Feature Group-D (FGD)--Exchange Access
North American (FGD-EANA) signaling).
• Disconnect Supervision--Configures the router to recognize the type of signaling in use by the PBX or
PSTN switch connected to the voice port. These methods include the following:
• Battery reversal disconnect
• Battery denial disconnect
• Supervisory tone disconnect (STD)

• FXO Supervisory Disconnect Tones--Prevents an analog FXO port from remaining in an off-hook state
after an incoming call is ended. FXO supervisory disconnect tone enables interoperability with PSTN
and PBX systems whether or not they transmit supervisory tones.
• Timeouts Parameters--Modifies values for timeouts. For example, you can adjust the wait time for the
caller input of the initial digit and the subsequent digit of the dialed string. If the wait time expires before
the destination is identified, a tone sounds and the call ends.
• Timing Parameters--Changes a wide range of timing values. For example, you can specify the minimum
delay time, in milliseconds, from outgoing seizure to outdial address.
• DTMF Timer--Modifies the value for the DTMF interdigit timer.
• Comfort Noise and Music Threshold for VAD--Specifies the minimal decibel level of music played
when calls are put on hold and 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 resulting silence can fool the caller
into thinking the call is disconnected instead of being merely idle.

How to Configure Fine-Tuning Features for Voice Ports

To configure the voice port tuning features for analog and digital voice ports, complete these tasks:

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 to determine the syntax choices that are available. Full descriptions of the commands in this section
can be found in the Cisco IOS Voice Command Reference.

Configuring Channel Bank Support for T1 E1 Voice Ports

The channel bank feature provides support for the time-division multiplexing (TDM) cross-connect functionality
between analog voice ports and digital DS0s on the same NM-HD-2VE using channel associated signaling
(CAS).

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Configuring Channel Bank Support for T1 E1 Voice Ports

To establish a channel bank connection between an analog voice port and a T1 DS0, configure the connect
(voice-port) command in global configuration mode. To verify the channel bank connection, use the show
connection all command.
Restrictions for Channel Bank Support:
• The configuration for cross-connect must be on the same network module.
• A maximum of four Foreign Exchange Service (FXS) or Foreign Exchange Office (FXO) ports can be
cross-connected to a T1 interface.
• A BRI-to-PRI cross-connect cannot be configured.
• Analog-to-BRI/PRI cross-connect cannot be configured; the only connection for analog is analog-to-T1/E1
CAS (ds0-group).
• The local-bypasscommand has no effect when cross-connect is configured. It is applicable only to calls
that are hairpinned via POTS-to-POTS dial peers.
• The DS0 group must contain only one time slot. The signaling type of the DS0 group must match that
of the analog voice port.
• If the channel bank feature is used for the T1 controller, the rest of the unused DS0 group cannot be
used for fractional PRI signaling.

SUMMARY STEPS

1. enable
2. configure terminal
3. controller {t1 | e1} slot/port
4. ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd |
e&m-immediate-start | e&m-wink-start | fxs-ground-start | fxs-loop-start | fxo-ground-start |
fxo-loop-start}
5. exit
6. voice-port slot / port
7. operation {2-wire | 4-wire}
8. type {1 | 2 | 3 | 5}
9. Do one of the following:
• signal {loop-start | ground-start}
•
•
• signal {wink-start | immediate | delay-dial}

10. exit
11. connect connection-name voice-port voice-port-number {t1 | e1} controller-number ds0-group-number
12. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 controller {t1 | e1} slot/port Enters controller configuration mode and identifies the controller
type (T1 or E1) and a slot and port for configuration commands that
Example: specifically apply to the T1 or E1 interface.

Router(config)# controller t1 1/0 • Valid values for the slot and port arguments are 0 and 1.

Step 4 ds0-group ds0-group-number timeslots Defines the T1 or E1 channels for use by compressed voice calls and
timeslot-list type {e&m-delay-dial | e&m-fgd the signaling method the router uses to connect to the PBX or central
| e&m-immediate-start | e&m-wink-start | office (CO).
fxs-ground-start | fxs-loop-start |
fxo-ground-start | fxo-loop-start} • The ds0-group command automatically creates a logical voice
port.

Example: • ds0-group-number --Value from 0 to 23 that identifies the DS0

group.
Router(config-controller)# ds0-group 1
timeslots 1 type e&m-wink-start • timeslot-list --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 1 to 24; for E1,
allowable values are 1 to 31.

The signaling method selection for type depends on the connection

that you are making:
• Ear and Mouth (E&M) connects PBX trunk lines (tie lines) and
telephone equipment. The wink and delay settings both specify
confirming signals between the sending and receiving ends, or
the immediate setting stipulates no special off-hook/on-hook
signal.
• FXO connects a CO to a standard PBX interface where
permitted by local regulations.
• FXS connects basic telephone equipment and PBXs.

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Command or Action Purpose

Step 5 exit Exits controller configuration mode and returns to global configuration
mode.
Example:

Router(config-controller)# exit

Step 6 voice-port slot / port Enters voice-port configuration mode and identifies a slot and port
for configuration parameters.
Example:

Router(config)# voice-port 2/1

Step 7 operation {2-wire | 4-wire} Selects a specific cabling scheme for E&M ports:
• This command is not applicable to FXS or FXO interfaces
Example: because they are, by definition, 2-wire interfaces.
Router(config-voiceport)# operation 4-wire
• Using this command on a voice port changes the operation of
both voice ports on a VPM card. The voice port must be shut
down and then opened again for the new value to take effect.

Step 8 type {1 | 2 | 3 | 5} Specifies the E&M interface type.

Example:
Router(config-voiceport)# type 2

Step 9 Do one of the following: Defines the signal type to be used.

• signal {loop-start | ground-start}
•
•
• signal {wink-start | immediate |
delay-dial}

Example:
Router(config-voiceport)# signal
loop-start

Example:

Example:
Router(config-voiceport)# signal
wink-start

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Configuring Auto Cut-Through

Command or Action Purpose

Step 10 exit Exits voice-port configuration mode and returns to global
configuration mode.
Example:

Router(config-voiceport)# exit

Step 11 connect connection-name voice-port Creates a named connection between two voice ports associated with
voice-port-number {t1 | e1} controller-number T1 or E1 interfaces where you have already defined the groups by
ds0-group-number using the ds0-group command.

Example:

Router(config)# connect connect1 voice-port

1/1/0 t1 1/0 0

Step 12 exit Exits the current configuration session and returns to privileged EXEC
mode.
Example:

Router(config)# exit

Configuring Auto Cut-Through

The auto-cut-through command allows you to connect to PBXs that do not provide an M-lead response. To
configure auto-cut-through, complete the following task:

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port slot / port
4. auto-cut-through
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

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Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-port slot / port Enters voice-port configuration mode.

Note The syntax of this command is platform-specific. For
Example: the syntax for your platform, refer to the Cisco IOS Voice
Router(config)# voice-port 3/0 Command Reference.

Step 4 auto-cut-through (E&M only) Enables call completion on a router if a PBX does
not provide an M-lead response.
Example:
Router(config-voiceport)# auto-cut-through

Step 5 exit Exits voice-port configuration mode and completes the

configuration.
Example:
Router(config-voiceport)# exit

Modifying Bit Patterns for Digital Voice Ports

The bit modification commands for digital voice ports modify sent or received bit patterns. Different versions
of E&M use different ABCD signaling bits to represent idle and seize. For example, North American CAS
E&M represents idle as 0XXX and seize as 1XXX, where X indicates that the state of the BCD bits is ignored.
In MELCAS E&M, idle is 1101 and seize is 0101.
To manipulate bit patterns to match particular E&M schemes, use the following commands:

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port slot /port
4. condition {tx-a-bit | tx-b-bit | tx-c-bit | tx-d-bit} {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit} {on | off |
invert}
5. define {tx-bits | rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 | 0110 | 0111 | 1000 |
1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111}
6. ignore {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit}
7. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-port slot /port Enters voice-port configuration mode.

Note The syntax of this command is platform-specific. For the syntax for
Example: your platform, refer to the Cisco IOS Voice Command Reference.
Router(config)# voice-port 3/0

Step 4 condition {tx-a-bit | tx-b-bit | tx-c-bit | Manipulates sent or received bit patterns to match expected patterns on a
tx-d-bit} {rx-a-bit | rx-b-bit | rx-c-bit | connected device. Repeat the command for each transmit or receive bit to be
rx-d-bit} {on | off | invert} modified, but be careful not to destroy the information content of the bit
pattern.
Example: • The default is that the signaling format is not manipulated (for all
Router(config-voiceport)# condition transmit or receive A, B, C, and D bits).
tx-a-bit on
Note The show voice port command reports at the protocol level, and
the show controller command reports at the driver level. The driver
is not notified of any bit manipulation using the condition command.
As a result, the show controller command output does not account
for the bit conditioning.
Step 5 define {tx-bits | rx-bits} {seize | idle} (Digital E1 E&M voice ports on Cisco 2600 and Cisco 3600 series routers
{0000 | 0001 | 0010 | 0011 | 0100 | 0101 | only) Defines specific transmit or receive signaling bits to match the bit
0110 | 0111 | 1000 | 1001 | 1010 | 1011 | patterns required by a connected device for North American E&M and E&M
1100 | 1101 | 1110 | 1111} MELCAS voice signaling, if patterns different from the preset defaults are
required.
Example: • Also specifies which bits a voice port monitors and which bits it ignores,
Router(config-voiceport)# define if patterns that are different from the defaults are required.
tx-bits seize 0000
• See the define command for the default signaling patterns as defined
in American National Standards Institute (ANSI) and European
Conference of Posts and Telecommunication Administration (CEPT)
standards.

Step 6 ignore {rx-a-bit | rx-b-bit | rx-c-bit | (Digital E1 E&M voice ports on Cisco 2600 and Cisco 3600 series routers
rx-d-bit} only) Configures the voice port to ignore the specified receive bit for North
American E&M or E&M MELCAS, if patterns different from the defaults

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Configuring ANI for Outbound Calling

Command or Action Purpose

are required. See the Cisco IOS Voice Command Reference for the default
Example: signaling patterns as defined in ANSI and CEPT standards.

Router(config-voiceport)# ignore
rx-a-bit

Step 7 exit Exits voice-port configuration mode and completes the configuration.

Example:
Router(config-voiceport)# exit

Configuring ANI for Outbound Calling

On the Cisco AS5300 platform, if T1 CAS is configured with the Feature Group-D (FGD)--Exchange Access
North American (FGD-EANA) signaling, the automatic number identification (ANI) can be sent for outgoing
calls by using the calling-number outbound command.
FGD-EANA is a FGD signaling protocol of type EANA, which provides certain call services, such as emergency
(USA 911) calls. ANI is a Signaling System 7 (SS7) feature in which a series of digits, analog or digital, are
included in the call to identify the telephone number of the calling device. In other words, ANI identifies the
number of the calling party. ANI digits are used for billing purposes by Internet service providers (ISPs),
among other things. The commands in this section can be issued in voice-port or dial-peer configuration mode,
because the syntax is the same.
To configure your digital T1/E1 packet voice trunk network module to generate outbound ANI digits on a
Cisco AS5300, use the following commands:

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port slot / port
4. calling-number outbound range string1 string2
5. calling-number outbound sequence [string1] [string2] [string3] [string4] [string5]
6. calling-number outbound null
7. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.

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Command or Action Purpose

• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-port slot / port Enters voice-port configuration mode.

Note The syntax of this command is platform-specific. For the syntax
Example: for your platform, refer to the Cisco IOS Voice Command
Router(config)# voice-port 3/0 Reference.

Step 4 calling-number outbound range string1 (Cisco AS5300 only) Specifies ANI to be sent out when the T1-CAS
string2 fgd-eana command is configured as signaling type. The string1 and
string2arguments are valid E.164 telephone number strings. Both strings
Example: must be of the same length and cannot be more than 32 digits long.

Router(config-voiceport)# • Only the last four digits are used for specifying the range (string1
calling-number outbound range 3000 4000 to string2) and for generating the sequence of ANI by rotating
through the range until string2 is reached and then starting from
string1 again. If strings are fewer than four digits in length, then
entire strings are used.

Step 5 calling-number outbound sequence [string1] (Cisco AS5300 only) Specifies ANI to be sent out when the T1-CAS
[string2] [string3] [string4] [string5] fgd-eana command is configured as signaling type. This option
configures a sequence of discrete strings (string1...string5) to be passed
Example: out as ANI for successive calls using the dial peer or voice port. Limit
is five strings. All strings must be valid E.164 numbers, up to 32 digits
Router(config-voiceport)# in length.
calling-number outbound sequence 2000
3000 4000

Step 6 calling-number outbound null (Cisco AS5300 only) Suppresses ANI. No ANI is passed when this voice
port is selected.
Example:
Router(config-voiceport)#
calling-number outbound null

Step 7 exit Exits voice-port configuration mode and completes the configuration.

Example:
Router(config-voiceport)# exit

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Configuring Disconnect Supervision

Configuring Disconnect Supervision

PBX and PSTN switches use several different methods to indicate that a call should be disconnected because
one or both parties have hung up. The commands in this section are used to configure the router to recognize
the type of signaling in use by the PBX or PSTN switch connected to the voice port. These methods include
the following:
• Battery reversal disconnect
• Battery denial disconnect
• Supervisory tone disconnect (STD)

Battery reversal occurs when the connected switch changes the polarity of the line in order to indicate changes
in call state (such as off-hook or, in this case, call disconnect). This is the signaling looked for when the
battery reversal command is enabled on the voice port, which is the default configuration.
Battery denial (sometimes called power denial ) occurs when the connected switch provides a short
(approximately 600 milliseconds) interruption of line power to indicate a change in call state. This is the
signaling looked for when the supervisory disconnect command is enabled on the voice port, which is the
default configuration.
Supervisory tone disconnect occurs when the connected switch provides a special tone to indicate a change
in call state. Some PBXs and PSTN CO switches provide a 600-millisecond interruption of line power as a
supervisory disconnect, and others provide STD. This is the signal that the router is looking for when the no
supervisory disconnect command is configured on the voice port.

Note In some circumstances, you can use the FXO Disconnect Supervision feature to enable analog FXO ports
to monitor call progress tones for disconnect supervision that are returned from a PBX or from the PSTN.
For more information, see the Configuring FXO Supervisory Disconnect Tones, on page 67.

To change parameters related to disconnect supervision, use the following commands:

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port slot / port
4. no battery-reversal
5. no supervisory disconnect
6. disconnect-ack
7. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.

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Command or Action Purpose

• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-port slot / port Enters voice-port configuration mode.

Note The syntax of this command is platform-specific. For the syntax for
Example: your platform, refer to the Cisco IOS Voice Command Reference.
Router(config)# voice-port 3/0

Step 4 no battery-reversal (Analog only) Enables battery reversal. The default is that battery reversal is
enabled.
Example: • For FXO ports--Use the no battery-reversal command to configure a
Router(config-voiceport)# no loop-start voice port not to disconnect when it detects a second battery
battery-reversal reversal. The default is to disconnect when a second battery reversal is
detected.

Note This functionality is supported on Cisco 1750, Cisco 2600 series, and
Cisco 3600 series routers; only analog voice ports on VIC-2FXO cards
are able to detect battery reversal.
Also use the no battery-reversal command when a connected FXO port does
not support battery reversal detection.
• For FXS ports--Use the no battery-reversal command to configure the
voice port not to reverse battery when it connects calls. The default is to
reverse battery when a call is connected, then return to normal when the
call is over, providing positive disconnect.

See also the disconnect-ack command (Step 6).

Step 5 no supervisory disconnect (FXO only) Enables the PBX or PSTN switch to provide STD. The supervisory
disconnect command is enabled by default.
Example:
Router(config-voiceport)# no
supervisory disconnect

Step 6 disconnect-ack (FXS only) Configures the voice port to return an acknowledgment upon receipt
of a disconnect signal. The FXS port removes line power if the equipment on
Example: the FXS loop-start trunk disconnects first. This is the default.

Router(config-voiceport)# The no disconnect-ack command prevents the FXS port from responding to
disconnect-ack the on-hook disconnect with a removal of line power.

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Configuring FXO Supervisory Disconnect Tones

Command or Action Purpose

Step 7 exit Exits voice-port configuration mode and completes the configuration.

Example:
Router(config-voiceport)# exit

Configuring FXO Supervisory Disconnect Tones

If the FXO supervisory disconnect tone is configured and a detectable tone from the PSTN or PBX is detected
by the digital signal processor (DSP), the analog FXO port goes on-hook. This feature prevents an analog
FXO port from remaining in an off-hook state after an incoming call is ended. FXO supervisory disconnect
tone enables interoperability with PSTN and PBX systems whether or not they transmit supervisory tones.
To configure a voice port to detect incoming tones, you need to know the parameters of the tones expected
from the PBX or PSTN. Then create a voice class that defines the tone detection parameters, and, finally,
apply the voice class to the applicable analog FXO voice ports. This procedure configures the voice port to
go on-hook when it detects the specified tones. The parameters of the tones need to be precisely specified to
prevent unwanted disconnects because of nonsupervisory tones or noise detection.
A supervisory disconnect tone is normally a dual tone consisting of two frequencies; however, tones of only
one frequency can also be detected. Use caution if you configure voice ports to detect nondual tones, because
unwanted disconnects can result from detection of random tone frequencies. You can configure a voice port
to detect a tone with one on/off time cycle, or you can configure it to detect tones in a cadence pattern with
up to four on/off time cycles.

Note In the following procedure, the following commands were not supported until Cisco IOS Release 12.2(2)T:
freq-max-deviation, freq-max-power, freq-min-power, freq-power-twist, and freq-max-delay.

To create a voice class that defines the specific tone or tones to be detected and then apply the voice class to
the voice port, use the following commands:

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SUMMARY STEPS

1. enable
2. configure terminal
3. voice class dualtone tag
4. freq-pair tone-id frequency-1 frequency-2
5. freq-max-deviation hertz
6. freq-max-power dBmO
7. freq-min-power dBmO
8. freq-power-twist dBmO
9. freq-max-delay time
10. cadence-min-on-time time
11. cadence-max-off-time time
12. cadence-list cadence-id cycle-1-on-time cycle-1-off-time [cycle-2-on-time cycle-2-off-time]
[cycle-3-on-time cycle-3-off-time ] [cycle-4-on-time cycle-4-off-time ]
13. cadence-variation time
14. exit
15. voice-port slot / subunit / port
16. supervisory disconnect dualtone {mid-call | pre-connect} voice-class tag
17. supervisory disconnect anytone
18. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice class dualtone tag Enters voice-class configuration mode and creates a voice class
for defining one tone detection pattern. Range is 1 to 10000. The
Example: tag number must be unique on the router.

Router(config)# voice class dualtone 1 • For more information about configuring voice classes, refer
to "Dial Peer Configuration onVoice Gateway Routers".

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Command or Action Purpose

Step 4 freq-pair tone-id frequency-1 frequency-2 Specifies the two frequencies, in Hz, for a tone to be detected (or
one frequency if a nondual tone is to be detected). If the tone to
Example: be detected contains only one frequency, enter 0 for frequency-2.

Router(config-voice-class)# freq-pair 16 Note Repeat this command for each additional tone to be
300 0 specified.

Step 5 freq-max-deviation hertz Specifies the maximum frequency deviation that will be detected,
in Hz. Range is 10 to 125. Default is 10.
Example:
Router(config-voice-class)#
freq-max-deviation 10

Step 6 freq-max-power dBmO Specifies the maximum tone power that will be detected, in
dBmO. Range is 0 to 20. Default is 10.
Example:
Router(config-voice-class)# freq-max-power
20

Step 7 freq-min-power dBmO Specifies the minimum tone power that will be detected, in
dBmO. Range is 10 to 35. Default is 30.
Example:
Router(config-voice-class)# freq-min-power
35

Step 8 freq-power-twist dBmO Specifies the power difference allowed between the two
frequencies, in dBmO. Range is 0 to 15. Default is 6.
Example:
Router(config-voice-class)# freq-power-twist
15

Step 9 freq-max-delay time Specifies the timing difference allowed between the two
frequencies, in 10-millisecond increments. Range is 10 to 100
Example: (100 ms to 1 second). Default is 20 (200 ms).

Router(config-voice-class)# freq-max-delay
10

Step 10 cadence-min-on-time time Specifies the minimum tone on time that will be detected, in
10-millisecond increments. Range is 0 to 100 (0 ms to 1 second).
Example:
Router(config-voice-class)#
cadence-min-on-time 10

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Command or Action Purpose

Step 11 cadence-max-off-time time Specifies the maximum tone off time that will be detected, in
10-millisecond increments. Range is 0 to 5000 (0 ms to 50
Example: seconds).

Router(config-voice-class)#
cadence-max-off-time 2000

Step 12 cadence-list cadence-id cycle-1-on-time (Optional) Specifies a tone cadence pattern to be detected. Specify
cycle-1-off-time [cycle-2-on-time cycle-2-off-time] an on time and off time for each cycle of the cadence pattern.
[cycle-3-on-time cycle-3-off-time ] [cycle-4-on-time The arguments are as follows:
cycle-4-off-time ]
• cadence-id --Range is 1 to 10. There is no default.
Example: • cycle-N-on-time --Range is 0 to 1000 (0 ms to 10 seconds).
Router(config-voice-class)# cadence-list 1 Default is 0.
0 1000
• cycle-N-off-time --Range is 0 to 1000 (0 ms to 10 seconds).
Default is 0.

Step 13 cadence-variation time (Optional) Specifies the maximum time that the tone onset can
vary from the specified onset time and still be detected, in
Example: 10-millisecond increments. Range is 0 to 200 (0 ms to 2 seconds).
Default is 0.
Router(config-voice-class)#
cadence-variation 200

Step 14 exit Exits voice class configuration mode.

Example:
Router(config-voice-class)# exit

Step 15 voice-port slot / subunit / port Enters voice-port configuration mode.

Example:
Router(config)# voice-port 0/1/0

Step 16 supervisory disconnect dualtone {mid-call | Assigns an FXO supervisory disconnect tone voice class to the
pre-connect} voice-class tag voice port.

Example:
Router(config-voiceport)# supervisory
disconnect dualtone mid-call voice-class 1

Step 17 supervisory disconnect anytone Configures the voice port to disconnect on receipt of any tone.

Example:
Router(config-voiceport)# supervisory
disconnect anytone

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Configuring Timeouts Parameters

Command or Action Purpose

Step 18 exit Exits voice-port configuration mode and completes the
configuration.
Example:
Router(config-voiceport)# exit

Configuring Timeouts Parameters

To change timeouts parameters, use the following commands:

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port slot / port
4. timeouts call-disconnect seconds
5. timeouts initial seconds
6. timeouts interdigit seconds
7. timeouts ringing {seconds | infinity}
8. timeouts wait-release {seconds | infinity}
9. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-port slot / port Enters voice-port configuration mode.

Note The syntax of this command is platform-specific. For the syntax
Example: for your platform, refer to the Cisco IOS Voice Command
Router(config)# voice-port 3/0 Reference.

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Changing Timing Parameters

Command or Action Purpose

Step 4 timeouts call-disconnect seconds Configures the call disconnect timeout value in seconds. Range is 0 to
120. Default is 60.
Example:
Router(config-voiceport)# timeouts
call-disconnect 60

Step 5 timeouts initial seconds Sets the number of seconds that the system waits between the caller
input of the initial digit and the subsequent digit of the dialed string. If
Example: the wait time expires before the destination is identified, a tone sounds
and the call ends.
Router(config-voiceport)# timeouts
initial 10 • The seconds argument is the initial timeout duration. Range is 0
to 120. Default is 10.

Step 6 timeouts interdigit seconds Configures the number of seconds that the system waits after the caller
has input the initial digit or a subsequent digit of the dialed string. If
Example: the timeout ends before the destination is identified, a tone sounds and
the call ends. This value is important when you are using variable-length
Router(config-voiceport)# timeouts dial peer destination patterns (dial plans).
interdigit 10
• The seconds argument is the interdigit timeout wait time in
seconds. Range is 0 to 120. Default is 10.

Step 7 timeouts ringing {seconds | infinity} Specifies the duration that the voice port allows ringing to continue if
a call is not answered.
Example: • Default for secondsis 180.
Router(config-voiceport)# timeouts
ringing infinity

Step 8 timeouts wait-release {seconds | infinity} Specifies the duration that a voice port stays in the call-failure state
while the Cisco device sends a busy tone, reorder tone, or an
Example: out-of-service tone to the port.

Router(config-voiceport)# timeouts • Default for secondsis 30.

wait-release 30

Step 9 exit Exits voice-port configuration mode and completes the configuration.

Example:
Router(config-voiceport)# exit

Changing Timing Parameters

To change timing parameters, use the following commands:

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Changing Timing Parameters

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port slot / port
4. timing clear-wait milliseconds
5. timing delay-duration milliseconds
6. timing delay-start milliseconds
7. timing delay-with-integrity milliseconds
8. timing dial-pulse min-delay milliseconds
9. timing dialout-delay milliseconds
10. timing digit milliseconds
11. timing guard-out milliseconds
12. timing hookflash-out milliseconds
13. timing interdigit milliseconds
14. timing percentbreak percent
15. timing pulse pulses-per-second
16. timing pulse-digit milliseconds
17. timing pulse-interdigit milliseconds
18. timing wink-duration milliseconds
19. timing wink-wait milliseconds
20. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-port slot / port Enters voice-port configuration mode.

Note The syntax of this command is platform-specific. For the
Example: syntax for your platform, refer to the Cisco IOS Voice
Router(config)# voice-port 3/0 Command Reference.

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Changing Timing Parameters

Command or Action Purpose

Step 4 timing clear-wait milliseconds (E&M only) Specifies the minimum amount of time, in
milliseconds, between the inactive seizure signal and clearing of
Example: the call.

Router(config-voiceport)# timing clear-wait • Range is 200 to 2000. Default is 400.

200

Step 5 timing delay-duration milliseconds (E&M only) Specifies the delay signal duration for delay-dial
signaling, in milliseconds.
Example: • Range is 100 to 5000. Default is 2000.
Router(config-voiceport)# timing
delay-duration 100

Step 6 timing delay-start milliseconds (E&M only) Specifies minimum delay time, in milliseconds, from
outgoing seizure to outdial address.
Example: • Range is 20 to 2000. Default is 300.
Router(config-voiceport)# timing
delay-start milliseconds

Step 7 timing delay-with-integrity milliseconds (Cisco MC3810 E&M ports only) Specifies duration of the wink
pulse for the delay dial, in milliseconds.
Example: • Range is 0 to 5000. Default is 0.
Router(config-voiceport)# timing
delay-with-integrity 0

Step 8 timing dial-pulse min-delay milliseconds Specifies time, in milliseconds, between the generation of wink-like
pulses when the type is pulse.
Example: • Range is 0 to 5000. Default is 300 for Cisco 3600 series and
Router(config-voiceport)# timing dial-pulse 140 for Cisco MC3810.
min-delay 300

Step 9 timing dialout-delay milliseconds (Cisco MC3810 only) Specifies dial-out delay, in milliseconds,
for the sending digit or cut-through on an FXO trunk or an E&M
Example: immediate trunk.

Router(config-voiceport)# timing • Range is 100 to 5000. Default is 300.

dialout-delay 100

Step 10 timing digit milliseconds Specifies the DTMF digit signal duration in milliseconds.
• Range is 50 to 100. Default is 100.
Example:
Router(config-voiceport)# timing digit 50

Step 11 timing guard-out milliseconds (FXO ports only) Specifies the duration in milliseconds of the
guard-out period that prevents this port from seizing a remote FXS
Example: port before the remote port detects a disconnect signal.

Router(config-voiceport)# timing guard-out • Range is 300 to 3000. Default is 2000.

300

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Changing Timing Parameters

Command or Action Purpose

Note For Caller ID to work for FXO ports registered to a Cisco
Unified CM, the range in milliseconds must be between
1000 to 2000.
Step 12 timing hookflash-out milliseconds Specifies the duration, in milliseconds, of the hookflash.
• Range is 50 to 500. Default is 300.
Example:
Router(config-voiceport)# timing
hookflash-out 500

Step 13 timing interdigit milliseconds Specifies the dual-tone multifrequency (DTMF) interdigit duration,
in milliseconds.
Example: • Range is 50 to 500. Default is 100.
Router(config-voiceport)# timing interdigit
100

Step 14 timing percentbreak percent (Cisco MC3810 FXO and E&M ports only) Specifies the
percentage of the break period for the dialing pulses, if different
Example: from the default.

Router(config-voiceport)# timing • Range is 20 to 80. Default is 50.

percentbreak 20

Step 15 timing pulse pulses-per-second (FXO and E&M only) Specifies the pulse dialing rate in pulses
per second.
Example: • Range is 10 to 20. Default is 20.
Router(config-voiceport)# timing pulse 20

Step 16 timing pulse-digit milliseconds (FXO only) Configures the pulse digit signal duration.
• Range is 10 to 20. Default is 20.
Example:
Router(config-voiceport)# timing
pulse-digit 10

Step 17 timing pulse-interdigit milliseconds (FXO and E&M only) Specifies pulse dialing interdigit timing in
milliseconds.
Example: • Range is 100 to 1000. Default is 500.
Router(config-voiceport)# timing
pulse-interdigit 500

Step 18 timing wink-duration milliseconds (E&M only) Specifies maximum wink-signal duration, in
milliseconds, for a wink-start signal.
Example: • Range is 100 to 400. Default is 200.
Router(config-voiceport)# timing
wink-duration 200

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Configuring the DTMF Timer

Command or Action Purpose

Step 19 timing wink-wait milliseconds (E&M only) Specifies maximum wink-wait duration, in
milliseconds, for a wink-start signal.
Example: • Range is 100 to 5000. Default is 200.
Router(config-voiceport)# timing wink-wait
200

Step 20 exit Exits voice-port configuration mode and completes the

configuration.
Example:
Router(config-voiceport)# exit

Configuring the DTMF Timer

To configure the DTMF timer, use the following commands:

SUMMARY STEPS

1. enable
2. configure terminal
3. controller T1 number
4. ds0-group channel-number timeslots range type signaling-type dtmf dnis
5. cas-custom channel
6. dtmf timer-inter-digit milliseconds
7. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Configuring Comfort Noise and Music Threshold for VAD

Command or Action Purpose

Step 3 controller T1 number Configures a T1 controller and enters controller
configuration mode.
Example:
Router(config)# controller T1 1

Step 4 ds0-group channel-number timeslots range type Configures channelized T1 time slots, which enables a
signaling-type dtmf dnis Cisco AS5300 modem to answer and send an analog call.

Example:
Router(config-controller)# ds0-group 0 timeslots
1-4 type e&m-immediate-start dtmf dnis

Step 5 cas-custom channel Enters cas-controller configuration mode and customizes

signaling parameters for a particular E1 or T1 channel group
Example: on a channelized line.

Router(config-controller)# cas-custom 2

Step 6 dtmf timer-inter-digit milliseconds Configures the DTMF interdigit timer for a DS0 group.

Example:
Router(conf-ctrl-cas)# dtmf timer-inter-digit
100

Step 7 exit Exits cas-controller configuration mode and completes the

configuration.
Example:
Router(conf-ctrl-cas)# exit

Configuring Comfort Noise and Music Threshold for VAD

In normal voice conversations, only one person speaks at a time. Circuit-switched telephone networks dedicate
a bidirectional 64 kbps channel for the duration of each conversation, regardless of whether anyone is speaking
at the moment. This means that, in a normal voice conversation, at least 50 percent of the bandwidth is wasted
when one or both parties are silent. This figure can actually be much higher when normal pauses and breaks
in conversation are taken into account.
Packet-switched voice networks can use this "wasted" bandwidth for other purposes when voice activity
detection (VAD) is configured. VAD works by detecting the magnitude of speech in decibels and deciding
when to stop segmenting voice packets into frames. VAD has some technological problems, however, which
include the following:
• General difficulties determining when speech ends
• Clipped speech when VAD is slow to detect that speech is beginning again

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Configuring Comfort Noise and Music Threshold for VAD

• Automatic disabling of VAD when conversations take place in noisy surroundings

VAD is configured in dial peers; by default it is enabled. Two parameters associated with VAD, music threshold
and comfort noise, are configured on voice ports.
If VAD is enabled, use the following commands to adjust music threshold and comfort noise:

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag voip
4. vad [aggressive]
5. exit
6. voice vad-time milliseconds
7. voice-port slot / port
8. music-threshold number
9. comfort-noise
10. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 dial-peer voice tag voip Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 555 voip

Step 4 vad [aggressive] Enables VAD for calls using this dial peer.
Note VAD is enabled by default. Use the vad command only if
Example: you have previously disabled the feature by using the no
Router(config-dial-peer)# vad vad command.

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Command or Action Purpose

Step 5 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit

Step 6 voice vad-time milliseconds Modifies the minimum silence detection time for VAD.

Example:
Router(config)# voice vad-time 500

Step 7 voice-port slot / port Enters voice-port configuration mode.

Note The syntax of this command is platform-specific. For
Example: information, refer to the Cisco IOS Voice Command
Router(config)# voice-port 3/0 Reference.

Step 8 music-threshold number Specifies the minimal decibel level of music played when calls are
put on hold. The decibel level affects how VAD treats the music data.
Example: • Valid values range from -70 to -30. If the music threshold is set
Router(config-voiceport)# too high and VAD is configured, the remote end hears no music;
music-threshold -70 if the level is set too low, there is unnecessary voice traffic.
Default is -38.

Step 9 comfort-noise Creates subtle background noise to fill silent gaps during calls when
VAD is enabled on voice dial peers. If comfort noise is not generated,
Example: the resulting silence can fool the caller into thinking the call is
disconnected instead of being merely idle.
Router(config-voiceport)# comfort-noise
• Comfort noise is enabled by default.

Step 10 exit Exits voice-port configuration mode.

Example:
Router(config-voiceport)# exit

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 CHAPTER 5
PSTN Fallback
The PSTN Fallback feature monitors congestion in the IP network and redirects calls to the Public Switched
Telephone Network (PSTN) or rejects calls on the basis of network congestion. This feature can also use
the ICMP ping mechanism to detect loss of network connectivity and then reroute calls. The fallback subsystem
has a network traffic cache that maintains the Calculated Planning Impairment Factor (ICPIF) or delay/loss
values for various destinations. Performance is improved because each new call to a well-known destination
does not have to wait on a probe to be admitted and the value is usually cached from a previous call.
ICPIF calculates an impairment factor for every piece of equipment along the voice path and then adds them
up to get the total impairment value. Refer to International Telecommunication Union (ITU) standard G.113
for more information. The ITU assigns a value to the types of impairment, such as noise, delay, and echo.

• Finding Feature Information, page 81

• Information About PSTN Fallback, page 82
• Restrictions for PSTN Fallback, page 82
• How to Configure PSTN Fallback, page 83
• How to Verify and Monitor the PSTN Fallback Feature, page 97
• What To Do Next, page 98

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

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Information About PSTN Fallback

Information About PSTN Fallback

Service Assurance Agent

Service Assurance Agent (SAA) is a network congestion analysis mechanism that provides delay, jitter, and
packet loss information for the configured IP addresses. SAA is based on a client/server protocol defined on
the User Datagram Protocol (UDP). UDP is a connectionless transport layer protocol in the IP protocol stack.
UDP is a simple protocol that exchanges datagrams without acknowledgments or guaranteed delivery, requiring
that error processing and retransmission be handled by other protocols. The SAA probe packets go out on
randomly selected ports from the top end of the audio UDP port range.
The information that the SAA probes gather is used to calculate the ICPIF or delay/loss values that are stored
in a fallback cache, where they remain until the cache ages out or overflows. Until an entry ages out, probes
are sent periodically for that particular destination. This time interval is user configurable.
With this feature enhancement, you can also configure codes that indicate the cause of the network rejection;
for example, packets that are lost or that take too long to be transmitted. A default cause code of 49 displays
the message qos-unavail, which means Quality of Service is unavailable.

Note The Cisco SAA functionality in Cisco IOS software was formerly known as Response Time Reporter
(RTR). In the How to Configure PSTN Fallback, on page 83 section, note that the command-line interface
still uses the keyword rtr for configuring RTR probes, which are now actually the SAA probes.

Application of PSTN Fallback

The PSTN Fallback feature and enhancement provide the following benefits:
• Automatically re-routes calls when the data network is congested at the time of the call setup.
• Enables the service provider to give a reasonable guarantee about the quality of the conversation to its
Voice over IP (VoIP) users at the time of call admission.
• Provides delay, jitter, and packet loss information for the configured IP addresses.
• Caches call values from previous calls. New calls do not have to wait for probe results before they are
admitted.
• Enables a user-configurable cause code display that indicates the type of call rejection.

Restrictions for PSTN Fallback

The PSTN Fallback feature has the following restrictions:
• When detecting network congestion, the PSTN fallback feature does nothing to the existing call. It affects
only subsequent calls.
• Only a single ICPIF/delay-loss value is allowed per system.

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How to Configure PSTN Fallback

• A small additional call setup delay can be expected for the first call to a new IP destination.

Caution Configuring call fallback active in a gateway creates an SAA jitter probe against other (target) gateways
to which the calls are sent. In order for the call fallback active to work properly, the target gateways must
have the rtr responder command (in Cisco IOS releases prior to 12.3(14)T) or the ip sla monitor
responder command (in Cisco IOS Release 12.3(14)T or later) in their configurations. If one of these
commands is not included in the configuration of each target gateway, calls to the target gateway will fail.

How to Configure PSTN Fallback

Configuring Call Fallback to Use MD5 Authentication for SAA Probes

To configure call fallback to use MD5 authentication for SAA probes, use the following commands.

SUMMARY STEPS

1. enable
2. configure terminal
3. call fallback active
4. call fallback key-chain name-of-chain

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call fallback active Enables the PSTN fallback feature to alternate dial peers in
case of network congestion.
Example:
Router(config)# call fallback active

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Configuring Destination Monitoring without Fallback to Alternate Dial Peers

Command or Action Purpose

Step 4 call fallback key-chain name-of-chain Specifies the use of message digest algorithm 5 (MD5)
authentication for sending and receiving Service Assurance
Example: Agents (SAA) probes.

Router(config)# call fallback key-chain

sample

Configuring Destination Monitoring without Fallback to Alternate Dial Peers

To configure destination monitoring without fallback to alternate dial peers, use the following commands.

SUMMARY STEPS

1. enable
2. configure terminal
3. call fallback monitor

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call fallback monitor Enables the monitoring of destinations without fallback to
alternate dial peers.
Example:
Router(config)# call fallback monitor

Configuring Call Fallback Cache Parameters

To configure the call fallback cache parameters, use the following commands.

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Configuring Call Fallback Jitter-Probe Parameters

SUMMARY STEPS

1. enable
2. configure terminal
3. call fallback cache-size number
4. call fallback cache-timeout seconds
5. clear call fallback cache [ip-address]

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call fallback cache-size number Specifies the call fallback cache size.

Example:
Router(config)# call fallback cache-size 5

Step 4 call fallback cache-timeout seconds Specifies the time after which the cache entry is purged,
in seconds. Default: 600.
Example:
Router(config)# call fallback cache-timeout 300

Step 5 clear call fallback cache [ip-address] Clears the current ICPIF estimates for all IP addresses
or a specific IP address in the cache.
Example:
Router(config)# clear call fallback cache 10.1.1.1

Configuring Call Fallback Jitter-Probe Parameters

To configure call fallback jitter-probe parameters, use the following commands.

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Configuring Call Fallback Jitter-Probe Parameters

SUMMARY STEPS

1. enable
2. configure terminal
3. call fallback jitter-probe num-packets number-of-packets
4. call fallback jitter-probe precedence precedence
5. call fallback jitter-probe priority-queue

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call fallback jitter-probe num-packets Specifies the number of packets for jitter. Default: 15.
number-of-packets

Example:
Router(config)# call fallback jitter-probe
num-packets 10

Step 4 call fallback jitter-probe precedence precedence Specifies the treatment of the jitter-probe transmission. Default:
2.
Example: Specifies the differentiated services code point (dscp) packet of
or the jitter-probe transmission.
Note The call fallback jitter-probe precedence command is
mutually exclusive with the call fallback jitter-probe
Example:
dscp command. Only one of these command can be
enabled on the router. Usually, the call fallback
call fallback jitter-probe dscp jitter-probe precedence command is enabled. When the
dscp-number call fallback jitter-probe dscp command is configured,
the precedence value is replaced by the DSCP value. To
Example: disable DSCP and restore the default jitter probe
precedence value, use the no call fallback jitter-probe
Router(config)# call fallback jitter-probe dscpcommand.
precedence 2

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Configuring Call Fallback Probe-Timeout and Weight Parameters

Command or Action Purpose

Example:
or

Example:
Router(config)# call fallback jitter-probe
dscp 2

Step 5 call fallback jitter-probe priority-queue Assigns a priority to the queue for jitter probes.

Example:
Router(config)# call fallback jitter-probe
priority-queue

Configuring Call Fallback Probe-Timeout and Weight Parameters

To configure call fallback probe-timeout and weight parameters, use the following commands.

SUMMARY STEPS

1. enable
2. configure terminal
3. call fallback probe-timeout seconds
4. call fallback instantaneous-value-weight percent

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Configuring Call Fallback Threshold Parameters

Command or Action Purpose

Step 3 call fallback probe-timeout seconds Sets the timeout for an SAA probe, in seconds. Default: 30.

Example:
Router(config)# call fallback probe-timeout
20

Step 4 call fallback instantaneous-value-weight percent Configures the call fallback subsystem to take an average from
the last two probes registered in the cache for call requests:
Example: • percent --Instantaneous value weight, expressed as a
Router(config)# call fallback percentage. Range: 0 to 100. Default: 66.
instantaneous-value-weight 50

Configuring Call Fallback Threshold Parameters

To configure call fallback threshold parameters, use the following commands.

SUMMARY STEPS

1. enable
2. configure terminal
3. call fallback threshold delay delay-value loss loss-value

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call fallback threshold delay delay-value Specifies fallback threshold to use packet delay and loss values. No
loss loss-value defaults.

Example:
or

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Configuring Call Fallback Wait-Timeout

Command or Action Purpose

Note The amount of delay set by the call fallback threshold delay
Example: loss command should not be more than half the amount of the
time-to-wait value set by the call fallback wait-timeout
command; otherwise the threshold delay will not work correctly.
call fallback threshold icpif Because the default value of the call fallback wait-timeout
threshold-value command is set to 300 milliseconds, you can configure a delay
of up to 150 milliseconds for the call fallback threshold delay
Example: loss command. If you want to configure a higher threshold, the
time-to-wait delay has to be increased from its default (300
Router(config)# call fallback threshold milliseconds) using the call fallback wait-timeout command.
delay 100 loss 150
Specifies fallback threshold to use the Calculated Planning Impairment
Factor (ICPIF) threshold for network traffic.
Example:
or

Example:
Router(config)# call fallback threshold
icpif 100

Configuring Call Fallback Wait-Timeout

To configure the call fallback wait-timeout parameters, use the following commands:

SUMMARY STEPS

1. enable
2. configure terminal
3. call fallback wait-timeout milliseconds

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

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Configuring VoIP Alternate Path Fallback SNMP Trap

Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call fallback wait-timeout Configures the waiting timeout interval for a response to a probe in milliseconds.
milliseconds Default: 300 milliseconds.
Note The time-to-wait period set by the call fallback wait-timeout command
Example: should always be greater than or equal to twice the amount of the
Router(config)# call fallback threshold delay time set by the call fallback threshold delay loss
wait-timeout 200 command; otherwise the probe will fail. The delay configured by the
call fallback threshold delay loss command corresponds to a one-way
delay, whereas the time-to-wait period configured by the call fallback
wait-timeout command corresponds to a round-trip delay. The
threshold delay time should be set at half the value of the time-to-wait
value.

Configuring VoIP Alternate Path Fallback SNMP Trap

The VoIP Alternate Path Fallback SNMP Trap feature adds a Simple Network Management Protocol (SNMP)
trap generation capability. This feature is built on top of the fallback subsystem to provide an SNMP notification
trap when the fallback subsystem redirects or rejects a call because a network condition has failed to meet the
configured threshold. The SNMP trap provides VoIP management status MIB information without flooding
management systems with unnecessary messages about call status by triggering only when a call has been
redirected to the public switched telephone network (PSTN) or the alternative IP port. A call can be rejected
because of a network problem such as loss of WAN connection, delay, packet loss, or jitter. This feature
supports only VoIP signaling protocol with H.323 in this release.
This feature has to be configured on the originating gateway and the terminating gateway. To configure the
SNMP trap parameters, use the following commands:

SUMMARY STEPS

1. enable
2. configure terminal
3. call fallback active
4. snmp-server enable traps voice fallback

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.

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Configuring Call Fallback Map Parameters

Command or Action Purpose

• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 call fallback active Enables the PSTN fallback feature to alternate dial peers
in case of network congestion.
Example:
Router(config)# call fallback active

Step 4 snmp-server enable traps voice fallback Configures the SNMP trap parameters.

Example:
Router(config)# snmp-server enable traps voice
fallback

What to Do Next
Configure the rtr responder command on the terminating voice gateway. If the rtr responder is enabled on
the terminating gateway, the terminating gateway responds to the probe request when the originating gateway
sends an Response Time Report (RTR) probe to the terminating gateway to check the network conditions.

Configuring Call Fallback Map Parameters

The call fallback map command option provides a target network summary/consolidation mode. For example,
if there are four individual voice gateway routers connected together on a remote LAN via a separate
LAN-to-WAN access router, the map option allows a single probe to be sent to the single remote WAN access
router (instead of having to maintain separate probes for each of the four voice gateway routers’ IP addresses).
Because the remote access and voice gateway routers are connected together on the same remote LAN, the
probes to the access router returns similar results to probes to the individual voice gateway routers.
To configure call fallback map parameters, use the following commands.

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Configuring ICMP Pings to Monitor IP Destinations

SUMMARY STEPS

1. enable
2. configure terminal
3. Do one of the following:
• call fallback map map target ip-address address-list ip-address1 ip-address2 ... ip-address7
•
• call fallback map map target ip-address subnet ip-network netmask

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 Do one of the following: Specifies the call fallback router to keep a cache table (by IP addresses)
of distances for several destination peers sitting behind the router.
• call fallback map map target
ip-address address-list ip-address1 • map --Fallback map. Range is from 1 to 16. There is no default.
ip-address2 ... ip-address7
• target ip-address --Target IP address.
•
• ip-address1 ip-address2 ... ip-address7 --Lists the IP addresses
• call fallback map map target that are kept in the cache table. The maximum number of IP
ip-address subnet ip-network netmask addresses is seven.

Specifies the call fallback router to keep a cache table (by subnet
addresses) of distances for several destination peers sitting behind the
router.

Configuring ICMP Pings to Monitor IP Destinations

This capability is enabled to monitor the IP destinations in a VoIP network, which may not support RTR. This
monitoring is referred to as ICMP pinging. Based on the RTR or ICMP pinging, results change the operational
state of the dial-peer. The configurations described in this section also provide support for monitoring the
following session targets configured under a VoIP dial-peer:

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Configuring ICMP Pings to Monitor IP Destinations

• DNS
• IP version 4
• SIP-server
• enum

To configure call-fallback monitor probes to ping IP destinations, complete one of the following tasks:

Dial Peer Configuration

To configure dial-peer parameters to use ICMP pings to monitor IP destinations, complete this task. This
configuration applies only to VoIP dial peers.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag voip
4. call fallback [icmp-ping| rtr]
5. monitor probe {icmp-ping| rtr} [ip address]

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 dial-peer voice tag voip Enters dial peer configuration mode, specifies the method of voice
encapsulation, and defines a particular dial peer:
Example: tag --Digits that define a particular dial peer. Range is from 1 to 2147483647.
Router(config)# dial-peer voice
10 voip

Step 4 call fallback [icmp-ping| rtr] Configures dial-peer parameters for pings to IP destinations:
• icmp-ping --Uses ICMP pings to monitor the IP destinations.
Example:
• rtr --Uses RTR probes to monitor the session target and update the status
Router(config-dial-peer)# call
fallback icmp-ping of the dial peer. RTR probes are the default.

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Command or Action Purpose

Note If this call fallback icmp-ping command is not entered, the call
fallback active command in global configuration is used for
measurements. If this call fallback icmp-ping command is entered,
these values override the global configuration. One of these two
commands must be in effect before the monitor probe
icmp-pingcommand can be used. If neither of call fallback commands
is in effect, the monitor probe icmp-ping command will not work
properly.
Step 5 monitor probe {icmp-ping| rtr} [ip Enables dial-peer status changes based on the result of the probe:
address]
• icmp-ping --Uses ICMP ping as the method for the probe.
Example: • rtr --Uses RTR as the method for the probe.
Router(config-dial-peer)# monitor
probe icmp-ping ip address --IP address of the destination to be probed. If no IP address is
specified, the IP address is read from the session target.

Global Configuration
To configure global parameters to use ICMP pings to monitor IP destinations, complete this task.

SUMMARY STEPS

1. enable
2. configure terminal
3. call fallback active [icmp-ping| rtr]
4. call fallback icmp-ping [count number] [codec type] | size bytes] interval seconds [loss number] [timeout
milliseconds]

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 call fallback active [icmp-ping| rtr] Configures global parameters for pings to IP destinations:
• icmp-ping --Uses ICMP pings to monitor the IP destinations.
Example:
• rtr --Uses RTR probes to monitor the IP destinations. RTR probes are
Router(config)# call fallback active
icmp-ping the default.

Note The call fallback active icmp-ping command must be entered

before the call fallback icmp-ping command can be used. If you
do not enter this command first, the call fallback icmp ping
command will not work properly.
Step 4 call fallback icmp-ping [count number] Configures the parameters for ICMP pings:
[codec type] | size bytes] interval seconds
[loss number] [timeout milliseconds] • count --Number of ping packets to be sent to the destination IP address.
Default is 5.
Example: • codec --Codec type for deciding the ping packet size.
Router(config)# call fallback icmp • type --Acceptable codec types are g711a, g711u, g729, and g729b.
ping codec g729 interval 10 loss 10
• size --Size (in bytes) of the ping packet. Default is 32.
• interval --Time (in seconds) between ping packet sets. Default is 5.
This value should be more than the timeout value.
• loss --Threshold packet loss, expressed as a percentage. Default is 20.
• timeout --Timeout (in milliseconds) for the echo packets. Default is
500.

Voice Port Configuration

To configure voice-port parameters to use ICMP pings to monitor IP destinations, complete this task.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port slot / port
4. busyout monitor probe icmp-ping ip address [codec type | size bytes][loss percent]

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.

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Command or Action Purpose

• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-port slot / port Enters voice-port configuration mode and identifies the slot and port
where the configuration parameters take effect.
Example: Note The syntax for this command varies by platform. For more
Router(config)# voice-port 1/0 information, see the Cisco IOS Voice Command Reference

Step 4 busyout monitor probe icmp-ping ip Specifies the parameters for ICMP pings for monitoring under voice-port
address [codec type | size bytes][loss configuration:
percent]
• ip address --IP address of the destination to which the ping is sent.
Example: • codec --(Optional) Codec type for deciding the ping packet size.
Router(config-voiceport)# busyout • type --Acceptable codec types are g711a, g711u, g729, and g729b.
monitor probe 10.1.1.1 g711u loss 10
delay 2000 • size --(Optional) Size (in bytes) of the ping packet. Default is 32.

loss --(Optional) Threshold packet loss, expressed as a percentage.

Default is 20.

Voice Class Configuration

To configure voice-class parameters to use ICMP pings to monitor IP destinations, complete this task.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice class busyout tag
4. busyout monitor probe icmp-ping ip address [codec type | size bytes][loss percent]

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.

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How to Verify and Monitor the PSTN Fallback Feature

Command or Action Purpose

• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice class busyout tag Creates a voice class for local voice busyout functions:
tag --Unique identification number assigned to one voice class. Range
Example: is 1 to 10000.
Router(config)# voice class busyout 10

Step 4 busyout monitor probe icmp-ping ip address Configures the parameters for ICMP pings for monitoring under
[codec type | size bytes][loss percent] voice-port:
• ip address --IP address of the destination to which the ping is
Example: sent.
Router(config-class)# busyout monitor
probe icmp-ping 10.1.1.1 codec g729b size • codec --(Optional) Codec type for deciding the ping packet size.
32
• type --Acceptable codec types are g711a, g711u, g729, and
g729b.
• size --(Optional) Size (in bytes) of the ping packet. Default is
32.
• loss --(Optional) Threshold packet loss, expressed as a
percentage. Default is 20.

How to Verify and Monitor the PSTN Fallback Feature

Verifying PSTN Fallback Configuration

The show commands in this section can be used to display statistics and configuration parameters to verify
the operation of the PSTN Callback feature:
• show running-config --Displays the contents of the currently running configuration file to see if the
new feature is configured.
• show call history voice --Displays the call history table for voice calls and verify call fallback, call
delay, and call loss parameters.

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• show call fallback cache --Displays the current Calculated Planning Impairment Factor (ICPIF) estimates
for all IP addresses in the call fallback cache.
• show call fallback config --Displays the current configuration.
• show call fallback stats --Displays the call fallback statistics.

Monitoring and Maintaining PSTN Fallback

Use the following commands to monitor and maintain the PSTN Fallback feature:
• clear call fallback cache --Clears the current ICPIF estimates for all IP addresses in the cache.
• clear call fallback stats --Clears the call fallback statistics.
• debug call fallback detail --Displays details of VoIP call fallback.
• debug call fallback probes --Displays details of voice fallback probes.
• test call fallback probe ip-address --Tests a probe to a particular IP address and displays the ICPIF
SAA values.
• debug snmp packets --Displays information about every Simple Network Management Protocol (SNMP)
packet sent or received by the router.

What To Do Next
The Configuring ICMP Pings to Monitor IP Destinations, on page 92 describes the mechanism whereby a
dial-peer becomes temporarily disabled because of poor SAA/RTR probe results (for example, ICPIF, jitter,
or loss), or because of failure of the ICMP ping test. When this occurs, the normal alternate dial-peer selection
process (hunting) is triggered to search for an alternate dial-peer that represents an alternate route.
The global configuration voice hunt command controls whether hunting (continue to look or "hunt" for an
alternate dial-peer match) occurs, based on the specific cause code that describes why the initial dial-peer
path failed. Hunting is usually appropriate if the cause code indicates network congestion, but usually
inappropriate if the failure cause code indicates that the called user is actually busy. Even if an alternate path
is taken to reach the called user, and if the user is actually busy, the user will be busy regardless of which path
is used.
For more information about the voice hunt command, see the Cisco IOS Voice Command Reference .

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 CHAPTER 6
Configuring Echo Cancellation
Echo cancellation is a key function in packet voice. Much of the perceived quality of the connection depends
on the performance of the echo canceller. The G.168 extended echo cancellation (EC) provides an alternative
to the proprietary Cisco G.165 EC with improved performance for trunking gateway applications.
The following sections provide configuration information for echo cancellation:

• Finding Feature Information, page 99

• Information About Echo Cancellation, page 99
• How to Configure the Extended G.168 Echo Canceller, page 109
• Restrictions for G.168 Extended Echo Canceller, page 109
• Configuration Examples for Extended G.168 Echo Cancellation, page 118

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Information About Echo Cancellation

Voice Call Transmit and Receive Paths

Every voice conversation has at least two participants. From the perspective of each participant, there are two
voice paths in every call:
• Transmit path (also called the send or Tx path)--The transmit path is created when a person speaks. The
sound is transmitted from the mouth of the speaker to the ear of the listener.

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Echo Cancellation

• Receive path (also called the return or Rx path)--The receive path is created when a person hears the
conversation. The sound is received by the ear of the listener from the mouth of the speaker.

The figure below shows a simple voice call between caller A and caller B. The top line represents the Tx path
for caller A, which becomes the Rx path for caller B. The bottom line represents the Tx path for caller B,
which becomes the Rx path for caller A.

Figure 13: Transmit and Receive Paths in a Voice Network

Echo Cancellation
Echo is the sound of your own voice reverberating in the telephone receiver while you are talking. When
timed properly, echo is not a problem in the conversation; however, if the echo interval exceeds approximately
25 milliseconds (ms), it can be distracting to the speaker. In the traditional telephony network, echo is generally
caused by an impedance mismatch when the four-wire network is converted to the two-wire local loop. Echo
is controlled by echo cancellers (ECs).
A packet voice gateway, which operates between a digital packet network and the PSTN, can include both
digital (time division multiplexing [TDM]) and analog links. The analog circuit is known as the tail circuit.
It forms the tail or termination of the call from the perspective of the person experiencing the echo. The tail
circuit is everything connected to the PSTN side of a packet voice gateway--all the switches, multiplexers,
cabling, and PBXs between the voice gateway and the telephone.
The figure below shows a common voice network where echo cancellation might be used.

Figure 14: Echo Cancellation Network

An echo canceller reduces the level of echoes that leak from the Rx path (from the gateway out into the tail
circuit) into the Tx path (from the tail circuit into the gateway). From the perspective of the echo canceller in
a voice gateway, the Rx signal is a voice coming across the network from another location. The Tx signal is
a mixture of the voice call in the other location and the echo of the original voice, which comes from the tail
circuit on the initiating end and is sent to the receiving end.

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Echo Canceller Operation

Echo cancellers face into the PSTN tail circuit. They eliminate echoes in the tail circuit on its side of the
network.The echo canceller in the originating gateway looks out into the tail circuit and is responsible for
eliminating the echo signal from the initiation Tx signal and allowing a voice call to go through unimpeded.
By design, ECs are limited by the total amount of time they wait for the reflected speech to be received, which
is known as an echo tail. The echo tail is normally 32 ms.

Note Delay and jitter in the WAN do not affect the operation of the echo canceller because the tail circuit, where
the echo canceller operates, is static.

Echo cancellation is implemented in digital signal processor (DSP) firmware (DSPWare) on Cisco voice
gateways and is independent of other functions implemented in the DSP (the DSP protocol and compression
algorithm). In voice packet-based networks, ECs are built into the low-bit-rate codecs and are operated on
each DSP.
The figure below shows a typical DSP channel configured for voice processing.

Figure 15: DSP Channel Configured for Voice Processing

Echo Canceller Operation

An echo canceller removes the echo portion of the signal coming out of the tail circuit and headed into the
WAN. It does so by learning the electrical characteristics of the tail circuit and forming its own model of the
tail circuit in its memory, and creating an estimated echo signal based on the current and past Rx signal. It
subtracts the estimated echo from the actual Tx signal coming out of the tail circuit. The quality of the estimation
is continuously improved by monitoring the estimation error.
Following are descriptions of the primary measurements of relative signal levels used by echo cancellers.
They are all expressed in decibels (dB).
• Echo return loss (ERL)--Reduction in the echo level produced by the tail circuit without the use of an
echo canceller. If an Rx speech signal enters the tail circuit from the network at a level of X dB, the echo
coming back from the tail circuit into the echo canceller is X less ERL.

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Echo Canceller Components

• Echo return loss enhancement (ERLE)--Additional reduction in echo level accomplished by the echo
canceller. An echo canceller is not a perfect device; the best it can do is attenuate the level of the returning
echo. ERLE is a measure of this echo attenuation. It is the difference between the echo level arriving
from the tail circuit at the echo canceller and the level of the signal leaving the echo canceller.
• Acombined (ACOM)--Total ERL seen across the terminals of the echo canceller. ACOM is the sum of
ERL + ERLE, or the total ERL seen by the network.

For more information about the echo canceller, refer to the "Echo Analysis for Voice over IP" document.

Echo Canceller Components

A typical echo canceller includes two components: convolution processor (CP) and a nonlinear processor
(NLP).

Convolution Processor
The CP first stage captures and stores the outgoing signal toward the far-end hybrid. The CP then switches
to monitoring mode and, when the echo signal returns, estimates the level of the incoming echo signal and
subtracts the attenuated original voice signal from the echo signal.
The time required to adjust the level of attenuation needed in the original signal is called the convergence
time. Because the convergence process requires that the voice signal be stored in memory, the EC has limited
coverage of tail circuit delay, normally 64, 96, and up to 128 ms. After convergence, the CP provides about
18 dB of ERLE. Because a typical analog phone circuit provides at least 12 dB of ERL (that is, the echo path
loss between the echo canceller and the far-end hybrid), the expected permanent ERL of the converged echo
canceller is about 30 dB or greater.

Nonlinear Processor
In single-talk mode, that is, when one person is talking and the other is silent, the NLP replaces the residual
echo at the output of the echo canceller with comfort noise based on the actual background noise of the voice
path. The background noise normally changes over the course of a phone conversation, so the NLP must adapt
over time. The NLP provides an additional loss of at least 25 dB when activated. In double-talk mode, the
NLP must be deactivated because it would create a one-way voice effect by adding 25 to 30 dB of loss in
only one direction.
To completely eliminate the perception of echo, the talker echo loudness rating (TELR) should be greater
than 65 dB in all situations. To reflect this reality, ITU-T standard G.168 requires an ERL equal to or greater
than 55 dB. Segmentation local reference (SLR), receive loudness rating (RLR), and cell loss ratio (CLR)
along the echo path should allow another 10 dB to meet the expected TELR. CP, NLP and loudness ratings
(LRs) must be optimized to make sure that echo is canceled effectively.

Echo Canceller Coverage

Echo canceller coverage (also known as tail coverage or tail length) is the length of time that the echo canceller
stores its approximation of an echo in memory. It is the maximum echo delay that an echo canceller is able
to eliminate.
The echo canceller faces into a static tail circuit with input and an output. If a word enters a tail circuit, the
echo is a series of delayed and attenuated versions of that word, depending on the number of echo sources
and the delays associated with them. After a certain period of time, no signal comes out. This time period is

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ITU-T Echo Cancellation History

known as the ringing time of the tail circuit--the time required for all of the ripples to disperse. To fully
eliminate all echoes, the coverage of the echo canceller must be as long as the ringing time of the tail circuit.

ITU-T Echo Cancellation History

ITU-T standard G.164 defines the performance of echo suppressors, which are the predecessors of echo
cancellation technology. G.164 also defines the disabling of echo suppressors in the presence of 2100 Hz
tones (which precede low bit rate modems).
ITU-T standard G.165 defines echo cancellation and provides a number of objective tests that ensure a minimum
level of performance. These tests check convergence speed of the echo canceller, stability of the echo canceller
filter, performance of the nonlinear processor, and a limited amount of double talk testing. The signal used
to perform these tests is white noise. Additionally, G.165 defines the disabling of echo cancellers in the
presence of 2100 Hz signals with periodic phase reversals in order to support echo cancelling modem technology
(V.34, for example), which does not work if line echo cancellation is performed in the connection.
ITU-T standard G.168 allows more rigorous testing and satisfies more testing requirements. White noise is
replaced with a pseudospeech signal for the convergence tests. Most echo cancellation algorithms use a least
mean square (LMS) algorithm to adapt the echo cancellation filter. LMS works best with random signals, and
slows down with more correlated signals such as speech. Using the pseudospeech signal in testing provides
a more realistic portrayal of the echo cancellers performance in real use.
In Cisco IOS Release 12.3(4)XD and later releases, the G.168 EC is the default and you can no longer select
the Cisco G.165 EC on any supported platform except the Cisco AS5300. The Cisco AS5300 still supports
the Cisco G.165 EC and the extended G.168 EC. provides a summary of the Extended ITU-T G.168 Echo
Cancellation feature availability in Cisco IOS releases.

Table 5: Feature History for Extended ITU-T G.168 Echo Cancellation

Release Modification
12.2(13)T This feature was introduced.

12.2(13)ZH The extended G.168 EC became the default on the

Cisco 1700 series and the Cisco ICS 7750.

12.2(15)ZJ The extended G.168 EC became the default on the

Cisco 2600 series, Cisco 3600 series, Cisco 3700
series.
Note The extended G.168 EC is not supported on
the High-Density Analog Network Modules
(NM-HDA) and Asynchronous Interface
Module (AIM)-Voice modules on the Cisco
2600 series in this release.
12.3(1) The extended G.168 EC became the default on the
Cisco IAD2420, Cisco MC3810, and Cisco VG200.

12.3(4)T The extended G.168 EC became the default on the

Cisco 7200 series and Cisco Catalyst 4000 AGM.

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Release Modification
12.3(4)XD The G.168 extended EC became the only EC on all
voice packet platforms that support the extended
G.168 EC; the Cisco G.165 EC is no longer a
selectable option.
Note The Cisco AS5300 still supported choosing
between the Cisco G.165 EC and the
extended G.168 EC.
12.3(3) The G.148 extended EC was configurable with no
codec restriction on the Cisco AS5300.

12.3(8)XY The G.168 extended EC was supported for the

WS-SVC-CMM-6T1, WS-SVC-CMM-6E1, and
WS-SVC-CMM-24FXS port adapters on the Cisco
Communication Media Module (WS-SVC-CMM).

12.3(9) The extended G.168 EC was supported for the

NM-HDA and AIM-Voice modules on the Cisco 2600
series.

12.3(11)T The dual-filter G.168 echo canceller capability was

added to NextPort SPE firmware (SPEware) version
10.2.2 and later versions with Cisco IOS Release
12.3(11)T and later releases. See the chapter
"NextPort-Based Voice Tuning and Echo
Cancellation".

12.4(20)T Software-configurable echo cancel coverage was

extended to 80, 96, 112, and 128 ms. The default
parameter is 128 ms.

Extended G.168 Echo Canceller Features

• Configuration and reporting of extended echo path capacity and worst-case ERL
• Test mode support for manually freezing, thawing, and clearing the EC h-register
• Reporting of statistics for location of the largest reflector and the internal state of the EC
• No changes to platform--Improves platform functionality by updating the EC module through a DSPWare
upgrade and a Cisco IOS software upgrade
• Enabling and disabling of nonlinear processor--Enables and disables NLP spectrally matched comfort
noise
• ERL configuration--Can be set to three values: 0 dB, 3 dB, and 6 dB
• Expansion of EC capacity--EC capacity is expanded to 64 ms (128 ms in Release 12.4(20)T or later)

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Extended EC Comparison

Extended EC Comparison
The table below contains comparison information for G.165 and G.168 echo cancellation.

Table 6: Echo Canceller Comparison

Feature G.165 EC G.168 EC

Tail Coverage Up to 32 ms Up to 64 ms (128 ms in Release
12.4(20)T or later

Minimum ERL Greater than or equal to 6 dB Configurable to greater than or

equal to 0 dB, 3 dB, or 6 dB

Echo Suppression Up to 10 seconds Not required because of faster

convergence

Extended Echo Canceller Support by Platform

The table below lists the support for the extended G.168 EC by platform, network module, high-complexity
and medium-complexity codecs, and minimum Cisco IOS release.

Table 7: Extended Echo Canceller Algorithm Coverage by Platform

Platform Network Module High Complexity Medium Comments

Codec Complexity
Codec
Analog Digital Analog Digital

Cisco 1700 series -- 12.2(8)YN 12.2(8)YN 12.2(8)YN 12.2(8)YN Flexi6 support in

12.2(13)T 12.2(13)T 12.3(2)T 12.3(2)T Cisco IOS
Release
12.2(8)YN.

Cisco 2600 series NM-HDV -- 12.2(13)T -- 12.2(13)T Full support.

Cisco 2600XM (C549)
Cisco 3600 series
Cisco 3700 series
Cisco VG200

Cisco 2600 series NM-1V, NM-2V -- -- -- -- Not supported.

Cisco 2691 Cisco (C542)
3600 series Cisco
3700 series Cisco
VG200

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Platform Network Module High Complexity Medium Comments

Codec Complexity
Codec
Cisco 2600XM NM-HDxx 12.3(4)XD 12.3(4)XD 12.3(4)XD 12.3(4)XD --
Cisco 2691 Cisco
3640 Cisco 3660
Cisco 3700 series

Cisco 2600XM AIM-Voice -- 12.2(15)ZJ -- 12.2(15)ZJ AIM.

Cisco 2691 Cisco (C5421), 12.3(4)T 12.3(4)T
3640 Cisco 3660 AIM-Voice-30
Cisco 3700 series (C542)

Cisco 2600XM NM-HDA 12.2(15)ZJ -- 12.2(15)ZJ, 12.2(15)ZJ NM-HDA.

Cisco 2691 Cisco (C5421) 12.3(4)T 12.3(4)T 12.3(4)T Note G.728
3640 Cisco 3660 high
Cisco 3700 series complexity
is not
supported.
Cisco 2600 series NM-HDA 12.3(9) -- 12.3(9) -- --
(C5421)

Cisco 2600 series AIM-Voice -- 12.3(9) -- 12.3(9) --

(C5421)

Cisco 7200 series PA-VXx-2TE1+ -- 12.2(13)T -- 12.2(13)T PA-MCX-nTE1

and port adapters do
PA-MCX-nTE1 not have their
own DSPs, so
they use the
DSPs of
PA-VXx-2TE1+
port adapters.

Cisco 7500 series -- -- 12.2(13)T -- -- No medium

complexity.

Cisco 7600 series Communiction -- 12.3(8)XY -- -- --

Media Module 12.3(14)T
(WS-SVC-CMM)
with one of the
following port
adapters:
WS-SVC-CMM-6T1
WS-SVC-CMM-6E1

WS-SVC-CMM-24FXS 12.3(8)XY -- -- -- --
12.3(14)T

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Platform Network Module High Complexity Medium Comments

Codec Complexity
Codec
Cisco AS5300 -- -- 12.2(13)T -- -- 1-channel DSP
(restricted) on C549 with
12.3(3) extended EC, any
(unrestricted) codec
(unrestricted).

Cisco AS5350 NextPort DFC -- Digital - -- 12.3(11)T See the

Cisco AS5400 modules: DFC60 12.3(11)T "NextPort-Based
Cisco AS5850 DFC108 1 Voice Tuning
CT3_UPC 216 and Echo
UPC324 Cancellation"
chapter in this
guide.

Cisco Catalyst AGM 12.3(4)T -- -- 12.3(4)T High-complexity

4000 analog and
medium-complexity
digital is
planned.

Cisco Catalyst Cisco 6624 A002040- 00002 -- A002040- 00002 -- --

6000
Cisco 6608 -- A004040- 00002 -- A004040- 00002 --

Cisco Catalyst Communiction -- 12.3(8)XY -- -- --

6500 series Media Module 12.3(14)T
(WS-SVC-CMM)
with one of the
following port
adapters:
WS-SVC-CMM-6T1
WS-SVC-CMM-6E1

WS-SVC-CMM-24FXS 12.3(8)XY -- -- -- --
12.3(14)T

Cisco IAD2420 -- 12.2(13)T 12.2(13)T 12.3(1) mainline 12.3(1) mainline --

Cisco IAD243x VIC2-4FXO 12.3(4)XD 12.3(4)XD 12.3(4)XD 12.3(4)XD --

onboard T1

Cisco ICS 7750 -- 12.2(13)T 12.2(13)T 12.2(13)T 12.2(13)T Flexi6 support.

Cisco MC3810 HCM 549 12.2(13)T 12.2(13)T 12.3(1) mainline 12.3(1) mainline --

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Extended G.168 Echo Canceller

Extended G.168 Echo Canceller

The Extended ITU-T standard G.168 Echo Cancellation feature provides an alternative to the default proprietary
Cisco G.165 EC. Beginning in Release 12.4(20)T, the extended EC provides improved performance for
trunking gateway applications and provides a configurable tail length that supports up to 128 ms of echo
cancellation. The G.165 EC is not configurable in Cisco IOS Release 12.3(4)XD and later releases, except
on the Cisco AS5300.

Note Extended echo cancellation is configured differently depending on the version of Cisco IOS software that
you are using. If you are using Cisco IOS Release 12.3(4)XD or a later release, you do not have to use
any Cisco IOS commands to enable the Extended ITU-T standard G.168 Echo Cancellation feature because
the extended G.168 EC is the only available echo canceller. You have the option of disabling the extended
EC, but it is highly recommended that you leave it enabled.

To configure the NextPort dual-filter G.168 echo canceller, see the "NextPort-Based Voice Tuning and Echo
Cancellation" chapter in this guide.
The table below lists the Cisco IOS commands that are used for selecting the extended G.168 EC based on
your platform and Cisco IOS release.

Table 8: Cisco IOS Commands for Selecting Extended E.168 EC by Platform and Cisco IOS Release

Cisco IOS Release Cisco IOS Command

Cisco 1700 Series and Cisco ICS 7750

12.2(13)T
Router(config)# voice echo-canceller extended

12.2(13)ZH 12.2(15)ZJ 12.3(1)

Router(voice-card)# codec complexity medium

12.3(4)T and later No configuration necessary. G.168 EC enabled by

default.

Cisco 2600 Series, Cisco 3600 Series, Cisco 3700

Series, Cisco MC3810, Cisco VG200

12.2(13)T 12.2(13)ZH 12.3(1)

Router(voice-card)# codec complexity medium
ecan-extended
or
Router(voice-card)# codec complexity high
ecan-extended

12.2(15)ZJ 12.3(4)T
Router(voice-card)# codec complexity medium

12.3(4)XD and later No configuration necessary. G.168 EC enabled by

default.

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Cisco IOS Release Cisco IOS Command

Cisco 7200 Series and Cisco 7500 Series

12.2(13)T
Router(config-dspfarm)# codec complexity
medium ecan-extended

12.2(13)ZH and later No configuration necessary. G.168 EC enabled by

default.

Cisco 7600 Series with Communication Media

Modules

12.3(8)XY and later No configuration necessary. G.168 EC enabled by

default.

Cisco AS5300

12.2(13)T
Router(config)# voice echo-canceller extended
codec small codec large codec

12.3(3)
Router(config)# voice echo-canceller extended
[codec small codec large codec]

Cisco Catalyst 6500 Series with Communication

Media Modules

12.3(8)XY and later No configuration necessary. G.168 EC enabled by

default.

Cisco Catalyst 4000 AGM

12.3(4)T and later No configuration necessary. G.168 EC enabled by

default.

How to Configure the Extended G.168 Echo Canceller

Restrictions for G.168 Extended Echo Canceller

• Not all Cisco platforms that use C542 or C549 DSPs support the extended EC.
• The G.168 extended EC is not supported on the Cisco AS5300 in Cisco IOS Release 12.2(13)ZH.
• The Cisco 1700 series does not support the T1/E1 card in Cisco IOS Release 12.2(13)T.

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Changing Echo Cancellers on Digital Voice Ports

Changing Echo Cancellers on Digital Voice Ports

To use the G.168 extended EC in a release prior to Cisco IOS Release 12.3(4)XD, perform one of the following
tasks depending on your hardware platform:

Cisco 1700 or Cisco ICS 7750

Cisco 2600 Series, Cisco 3600 Series, Cisco 3700 Series, Cisco MC3810, and Cisco VG200
• Changing Codec Complexity on Cisco 2600 Series, Cisco 3600 Series, Cisco 3700 Series, and Cisco
MC3810 in the "Configuring Digital Voice Ports" chapter.

Cisco 7200 Series and Cisco 7500 Series

• Configuring Codec Complexity on Cisco 7200 Series and Cisco 7500 Series Routers in the "Configuring
Digital Voice Ports" chapter

Note See the table above for extended EC algorithm coverage by platform.

Enabling the Extended G.168 EC in Cisco IOS Release 12.2(13)T

To change codec complexity on the Cisco 1700 series and Cisco ICS 7750 and switch between the proprietary
Cisco EC and the extended G.168 EC, use the following commands.

Note You must clear all calls on the system before using the following commands. If there are active calls on
the system, the commands are ignored and a warning message is issued.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice echo-canceller extended
4. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

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Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice echo-canceller extended Enables the G.168 extended echo canceller on the Cisco 1700 series or
Cisco ICS 7750.
Example: • You do not have to shut down all the voice ports on the Cisco
Router(config)# voice echo-canceller 1700 or Cisco ICS 7750 to switch the echo canceller, but you
extended should make sure that when you switch the echo canceller, there
are no active calls on the router.
• To return to the proprietary Cisco G.165 default EC, use the no
form of the command.

Step 4 exit Exits global configuration mode.

Example:
Router(config)# exit

Enabling the Extended G.168 EC in Cisco IOS Release 12.2(13)ZH

The codec complexity medium command enables the extended echo canceller by default on the Cisco 1700
series and the Cisco ICS 7750 in Cisco IOS Release 12.2(13)ZH.

Note You must clear all calls on the system before using the following commands. If there are active calls on
the system, the commands are ignored and a warning message is issued.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-card slot
4. codec complexity medium
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables higher privilege levels, such as privileged EXEC
mode.
Example: • Enter your password if prompted.
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-card slot Enters voice card configuration mode on the specified slot.

Example:
Router(config)# voice-card 1

Step 4 codec complexity medium Enables the extended EC (default).

Example:
Router(voice-card)# codec complexity medium

Step 5 end Exits voice-card configuration mode and completes the steps
for configuring the extended EC on the Cisco 1700 series and
Example: Cisco ICS 7750.

Router(voice-card)# end

Configuring the Extended G.168 EC on the Cisco AS5300

Perform this task to enable the Extended ITU-T standard G.168 Echo Cancellation feature on the Cisco
AS5300. You must designate which EC to use on the Cisco AS5300 C542 and C549 DSPM high-complexity
platform. You can use the extended EC with codec restrictions on the C542 DSP firmware or with or without
codec restrictions on the C549 DSP firmware.

Note A firmware upgrade can be made by upgrading Cisco VCWare. For upgrade information, refer to the
Combined Version Release Notes and Compatibility Matrix for Cisco VCWare on Cisco AS5300 Universal
Access Servers/Voice Gateways .

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Before You Begin

Extended EC with Restricted Codecs on C542 or C549 DSP:
• Create a backup of your router configuration.
• Determine which codecs are required when enabling the extended echo canceller. For this information,
see the codec restriction options for the voice echo-canceller extended command.
• Specify a single small codec (g711 or g726) and a single large codec (g723, g728, GSM FR, GSM
EFR, g729, or fax-relay). Any call setup involving other codecs is rejected.
• If fax-relay is not selected as the large codec, the VoIP dial peer requires that you use the fax rate
disabled command in dial-peer configuration mode to reset the dial peer for voice calls.

• Review your existing configuration and look for all dial peers that select codecs or fax-relay specification
that are different from the codecs that you decide on. After choosing the codecs to be supported by the
extended echo canceller, either remove all dial peers with different codecs not supported by your new
configuration or modify the dial-peer codec selection by selecting a voice codec or fax-relay that is
supported by the new configuration.
• Ensure that modem relay is not configured in any of the dial-peer configurations. If modem relay is
configured, it should be disabled using the no modem relaycommand.

Extended EC with Unrestricted Codecs on C549 DSP:

• Create a backup of your router configuration.
• Ensure that modem relay is not configured in any of the dial-peer configurations. If modem relay is
configured, it should be disabled using the no modem relaycommand.

Note • The extended G.168 EC can be used only in one of the following ways on the Cisco AS5300:
◦With a restricted set of codecs with C542 or C549 DSP firmware. Two channels of voice are
supported per DSP, and full call handling capacity is supported.
◦With no restrictions on codecs with C549 DSP firmware. One channel of voice is supported
per DSP, and call handling capacity is reduced by half.

• Not all Cisco platforms that use C542 or C549 DSPs support the extended EC. Other platforms
continue to use the proprietary Cisco G.165 EC if they do not support the extended EC.

>

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SUMMARY STEPS

1. enable
2. configure terminal
3. no dial-peer voice tag voip
4. dial-peer voice tag voip
5. codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32
| g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr} [bytes payload-size]
6. exit
7. Do one of the following:
• voice echo-canceller extended
•
• voice echo-canceller extended [codec small codec large codec]

8. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 no dial-peer voice tag voip (Optional) Removes VoIP dial peers, one dial peer at a time.
• When configuring the extended EC in global configuration
Example: mode, you must remove or modify all existing VoIP dial
Router(config)# no dial-peer voice 1 voip peers before the voice echo-canceller extendedcommand is
accepted.

Step 4 dial-peer voice tag voip Enters dial-peer configuration mode so that you can modify a codec
type.
Example:
Router(config)# dial-peer voice 1 voip

Step 5 codec {g711alaw | g711ulaw | g723ar53 | g723ar63 Specifies the voice codec rate for the dial peer.
| g723r53 | g723r63 | g726r16 | g726r24 | g726r32
| g726r53 | g726r63 | g728 | g729abr8 | g729ar8 |

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Command or Action Purpose

g729br8 | g729r8 | gsmefr | gsmfr} [bytes
payload-size]

Example:
Router(config-dialpeer)# codec g711alaw

Step 6 exit Exits dial-peer configuration mode and returns to global

configuration mode.
Example:
Router(config-dialpeer)# exit

Step 7 Do one of the following: Enables the extended echo canceller with no restriction on codecs.
• voice echo-canceller extended or

• Enables the extended echo canceller with restricted codecs.

• voice echo-canceller extended [codec small • The following codec choices are valid:
codec large codec]
• Small footprint codec--G.711 or G.726.
• Large footprint codec--G.729, G.726, G.728, G.723,
Example: fax-relay, GSM FR, or GSM EFR.
Router(config)# voice echo-canceller
extended
Note The voice echo-canceller extended command enables
the extended EC without codec restrictions on a Cisco
Example: AS5300 with C549 DSP firmware. One channel of voice
is supported per DSP. Any codec is supported.
Router(config)# voice echo-canceller
extended codec small g711 large fax-relay Note The voice echo-canceller extended codec command
restricts codecs on the Cisco AS5300 with C542 and C549
DSP firmware. Two channels of voice are supported per
Example: DSP. Only specific codecs are supported.

Step 8 exit Exits global configuration mode.

Example:
Router(config)# exit

Modifying Echo Cancellation Default Settings

Perform this task to modify the default settings for echo cancellation parameters.

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SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port slot / port:ds0-group-number
4. echo-cancel enable
5. echo-cancel coverage {24 | 32 | 48 | 64 | 80 | 96 | 112 | 128}
6. echo-cancel erl worst-case [0 | 3 | 6]
7. non-linear
8. echo-cancel suppressor seconds
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables higher privilege levels, such as privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice-port slot / Enters voice-port configuration mode on the selected slot, port, and DS0 group.
port:ds0-group-number Note The syntax of this command is platform-specific. For the syntax for your
platform, refer to the Cisco IOS Voice Command Reference.
Example:
Router(config)# voice-port
1/0:0

Step 4 echo-cancel enable Enables echo cancellation.

• Echo cancellation is enabled by default. Use the no form of this command to
Example: disable echo cancellation.
Router(config-voiceport)#
echo-cancel enable • The extended G.168 EC is the default EC for all supported platforms in Cisco
IOS Release 12.3(4)XD and later releases, except the Cisco AS5300.
• On the Cisco AS5300, the Cisco G.165 EC is enabled by default with echo
suppression disabled.

Note This command is supported only when the echo-cancel coverage command
is enabled.

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Command or Action Purpose

Step 5 echo-cancel coverage {24 | 32 | 48 Adjusts the size of the echo canceller (echo path capacity coverage).
| 64 | 80 | 96 | 112 | 128}
• 128 is the default in Cisco IOS Release 12.4(20)T and later releases. Prior to
this release, the default is 64 ms.
Example:
Router(config-voiceport)# Note This command is supported only when echo cancellation is enabled. See
echo-cancel coverage 64 Step 4 of this procedure.
Step 6 echo-cancel erl worst-case [0 | 3 | (Optional) Determines worst-case ERL in dB.
6]
• Worst-case ERL is the minimum expected attenuation in the voice path. For
example, if you have a worst-case ERL of 6 (erl worst-case 6), when you
Example: speak into the phone you can expect at least 6 dB of attenuation on the signal
Router(config-voiceport)# by the time it gets back to the original source (echo). In general you do not
echo-cancel erl worst-case 6 need to change this value from 6, which is the default.

Worst-case ERL does not directly modify the inbound or outbound signals. This is
purely a configuration parameter for the EC to help it distinguish between echo and
a new signal.
Note This command is supported for the extended G.168 EC only; it is not
supported for the G.165 EC.
Step 7 non-linear (Optional) Selects nonlinear processing (residual echo suppression) in the EC, which
either shuts off any signal or mixes in comfort noise if no near-end speech is detected.
Example: Note This command is supported only when echo cancellation is enabled. See
Router(config-voiceport)# Step 4 .
non-linear
• Nonlinear processing is enabled when the extended G.168 EC is enabled. Use
the no form of this command to disable the NLP.

Step 8 echo-cancel suppressor seconds (Optional) Applies echo suppression for the number of seconds specified when using
the G.165 EC.
Example: • This command cannot be used with the extended G.168 EC in Cisco IOS
Router(config-voiceport)# Release 12.2(15)ZJ or later releases, or on NextPort (Cisco AS5350 and Cisco
echo-cancel suppressor 10 AS5400) platforms.

Note This command is required to configure the Extended ITU-T standard G.168
Echo Cancellation feature in Cisco IOS Release 12.2(13)T.
• For the AS5300, the Cisco G.165 EC is enabled by default with echo
suppression disabled. The echo suppressor can be used only on T1 DSPs when
the default Cisco G.165 EC is used.
• This command enables echo cancellation for voice that is sent out an interface
and received back on the same interface within the configured amount of time.
• This command reduces the initial echo before the echo canceller can converge.
In case of double-talk in the first number of seconds, the code automatically
disables the suppressor.

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 Configuring Echo Cancellation
Configuration Examples for Extended G.168 Echo Cancellation

Command or Action Purpose

• The echo-cancel suppressor command is visible when the G.168 extended
EC is selected but it has no effect.

Note This command is supported only when echo cancellation is enabled.

Step 9 end Exits voice-port configuration mode and completes the configuration.

Example:
Router(config-voiceport)# end

Configuration Examples for Extended G.168 Echo Cancellation

Enabling the Extended EC on the Cisco 1700 Series and Cisco ICS 7750 Example
The following example enables the G.168 extended EC on a Cisco 1700 series or a Cisco ICS 7750. The
extended EC is enabled by default when the medium keyword is used in Cisco IOS Release 12.2(13)ZH and
later.

voice-card 1
codec complexity medium

Enabling the Extended EC Prior to Cisco IOS Release 12.3(4)XD Example

The following example shows that the echo canceller has been changed from the default proprietary Cisco
EC to the extended EC in Cisco IOS releases prior to 12.3(4)XD. This example applies to the Cisco 2600
Series, Cisco 3600 Series, Cisco 3700 Series routers, and Cisco VG200.

Note The extended G.168 EC is the only EC in Cisco IOS Release 12.3(4)XD and later releases. Because it is
enabled by default, it does not display in the configuration output in Cisco IOS Release 12.3(4)XD and
later releases.

The following is example output from an originating Cisco 3640:

.
.
.
voice-card 1
codec complexity high ecan-extended
.
.
.
controller T1 1/0
framing esf

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 Configuring Echo Cancellation
Enabling the Extended EC on the Cisco 7200 and Cisco 7500 Series Example

linecode b8zs
pri-group timeslots 1-24
!
voice-port 1/0:23
.
.
.
dial-peer voice 104001 voip
destination-pattern 104001
session target ipv4:10.2.0.104
dtmf-relay cisco-rtp
codec g711alaw
fax rate 14400
fax protocol cisco
.
.
.

Enabling the Extended EC on the Cisco 7200 and Cisco 7500 Series Example
The following example changes codec complexity on a Cisco 7200 series or Cisco 7500 series:

dspint dspfarm 2/0

codec medium ecan-extended

Enabling the Extended Echo Canceller on the Cisco AS5300 Example

The following example enables the extended G.168 EC with restricted codecs on the Cisco AS5300 with
C542 or C549 DSP firmware:

!
version 12.3
no service pad
service timestamps debug datetime msec
service timestamps log uptime
no service password-encryption
service internal
!
hostname router
!
boot-start-marker
boot-end-marker
!
enable secret 5 $123
enable password temp
!
!
resource-pool disable
!
no aaa new-model
ip subnet-zero
ip rcmd rcp-enable
ip rcmd rsh-enable
ip domain name cisco.com
ip host router1 10.10.101.14
!
!
isdn switch-type primary-5ess
!
v
oice echo-canceller extended codec small g711 large fax-relay
!
!
!
fax interface-type fax-mail

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 Configuring Echo Cancellation
Enabling the Extended EC on the Cisco 7200 and Cisco 7500 Series Example

!
!
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
.
.
.

Adjusting the Echo Canceller Size Example

The following example adjusts the size of the extended EC to 64 ms on Cisco 3600 series routers:

voice-port 1/0:0
echo-cancel coverage 64

Worst-Case Echo Return Loss Example

The following example sets the worst-case echo return loss to 3:

voice-port 0:D
echo-canceller erl worst-case 3
playout-delay mode fixed
no comfort-noise

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 CHAPTER 7
Pulse Code Modulation (PCM) Audio Capture
The Pulse Code Modulation (PCM) Audio capture feature is used for debugging audio quality issues. PCM
capture refers to an existing Digital Signal Processor (DSP) feature by which the digital audio signal at
various nodes in the audio signal processing path of a voice channel may be intercepted and uploaded to the
host router using specialized DSP-to-host message packets. Cisco IOS file services allow a file containing
interleaved audio and debug data (.dat) to be created in the local file system or a remote TFTP server. This
.dat file is then decoded and deinterleaved into separate, synchronized .wav files for each of the signal
interception nodes. This feature is typically employed for capture of audio test signals in troubleshooting
specific voice issues such as echo. Signals may be captured at any or all of the defined nodes, including the
input-output nodes of an echo canceller (Rin, Sin, Sout), the Acoustic Shock Protection circuit, and the Noise
Reduction module. Additional nodes of interest will be added as new signal processing features are introduced.

• Finding Feature Information, page 121

• Information about PCM Audio Capture, page 122
• How to Configure PCM Audio Capture, page 122
• Additional References for Cisco UBE Serviceability, page 125
• Feature Information for Pulse Code Modulation (PCM) Audio Capture, page 126

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

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 Pulse Code Modulation (PCM) Audio Capture
Information about PCM Audio Capture

Information about PCM Audio Capture

PCM Audio Capture

The following are the enhancements to the PCM Audio Capture feature:
• Separate PCM capture and Banjo logger feature so that they do not share the same data (.dat) file; they
have their own data file.
• One PCM call per data file is generated dynamically. The filename contains information such as voice
port type and number, call ID, calling and called number, GUID, DSP channel number, and time stamp.
• A user on the TDM-TDM or TDM-VoIP call can dynamically enable and disable PCM capture by
entering predefined start and stop Dual Tone Multi-Frequency (DTMF) digits.
• More test points or streams can be captured.

Note PCM capture is a CPU-intensive feature, and you must not enable several PCM capture sessions while
running heavy traffic.

How to Configure PCM Audio Capture

Configuring PCM Audio Capture

SUMMARY STEPS

1. enable
2. configure terminal
3. voice pcm capture buffer number
4. voice pcm capture destination url
5. voice pcm capture on-demand-trigger
6. voice pcm capture user-trigger-string start-string stop-string stream bitmap duration call-duration
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.

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 Pulse Code Modulation (PCM) Audio Capture
Configuring PCM Audio Capture

Command or Action Purpose

• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voice pcm capture buffer number Configures the number of PCM capture buffers. The Range is from
0 to 200000. To change the PCM capture buffer size, you must first
Example: configure it with 0 and then configure it with the desired number.
Router(config)# voice pcm capture buffer
10

Step 4 voice pcm capture destination url Configures or changes the destination URL for storing captured
data.
Example:
Router(config)# voice pcm capture
destination tftp://10.10.1.2/acphan/

Step 5 voice pcm capture on-demand-trigger Configures user-triggered PCM capture.

Example:
Router(config)# voice pcm capture
on-demand-trigger

Step 6 voice pcm capture user-trigger-string start-string Changes the default user trigger PCM capture start and stop string,
stop-string stream bitmap duration call-duration stream, and duration.
• The start and stop string must have different values.
Example:
Router(config)# voice pcm capture #132 #543 • PCM stream bitmap is in hexadecimal. The range is from 1
stream ff duration 230
to FFFFFFF.

• The stream bitmap definitions are as follows:

• bit 0—Rin
• bit 1—Sin
• bit 2—Sout
• bit 3—nonNLP Sout
• bit 4—fax modem in
• bit 5—fax modem out

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Verifying PCM Audio Capture

Command or Action Purpose

• bit 6—from IP network to TDM earpiece direction: ASP
input
• bit 7—from IP network to TDM earpiece direction: ASP
output
• bit 8—NR in
• bit 9—NR out
• bit 10—from TDM mic to IP network: ASP in
• bit 11—from TDM mic to IP network: ASP out

Step 7 end Returns to privileged EXEC mode.

Example:
Router(config)# end

Verifying PCM Audio Capture

Perform this task to verify the configuration for the PCM Audio Capture feature.

SUMMARY STEPS

1. enable
2. show voice pcm capture

DETAILED STEPS

Step 1 enable

Example:
Router> enable

Enables privileged EXEC mode.

Step 2 show voice pcm capture

Example:
Router# show voice pcm capture
PCM Capture is on and is logging to URL tftp://10.10.1.2/acphan/
50198 messages sent to URL, 0 messages dropped
Message Buffer (total:inuse:free) 200000:0:200000

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Additional References for Cisco UBE Serviceability

Buffer Memory: 68000000 bytes, Message size: 340 bytes

Displays the configured PCM capture buffer and destination, number of saved messages/packets, number of dropped
messages/packets, and number of buffers allocated, both used and free.

Additional References for Cisco UBE Serviceability

Related Documents

Related Topic Document Title

Cisco IOS commands Cisco IOS Master Command List,
All Releases

Voice commands
• Cisco IOS Voice Command
Reference - A through C
• Cisco IOS Voice Command
Reference - D through I
• Cisco IOS Voice Command
Reference - K through R
• Cisco IOS Voice Command
Reference - S Commands
• Cisco IOS Voice Command
Reference - T through Z
Commands

Technical Assistance

Description Link
The Cisco Support and Documentation website http://www.cisco.com/cisco/web/support/index.html
provides online resources to download documentation,
software, and tools. Use these resources to install and
configure the software and to troubleshoot and resolve
technical issues with Cisco products and technologies.
Access to most tools on the Cisco Support and
Documentation website requires a Cisco.com user ID
and password.

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 Pulse Code Modulation (PCM) Audio Capture
Feature Information for Pulse Code Modulation (PCM) Audio Capture

Feature Information for Pulse Code Modulation (PCM) Audio

Capture
The following table provides release information about the feature or features described in this module. This
table lists only the software release that introduced support for a given feature in a given software release
train. Unless noted otherwise, subsequent releases of that software release train also support that feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Table 9: Feature Information for Pulse Code Modulation (PCM) Audio Capture

Feature Name Releases Feature Information

Pulse Code Modulation (PCM) 15.2(2)T The PCM Capture feature is used
Audio Capture for debugging audio quality issues.
In Cisco IOS Release 15.2(2)T, this
feature was implemented on the
Cisco Unified Border Element .
The following commands were
introduced or modified: show voice
pcm capture, voice pcm capture.

Pulse Code Modulation (PCM) Cisco IOS XE Release 3.6S The PCM Capture feature is used
Audio Capture for debugging audio quality issues.
In Cisco IOS XE Release 3.6S, this
feature was implemented on the
Cisco Unified Border Element
(Enterprise)
The following commands were
introduced or modified: show voice
pcm capture, voice pcm capture.

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 CHAPTER 8
Acoustic Shock Protection
Acoustic Shock Protection (ASP) is a voice circuit-breaker feature that is designed to protect users, especially
those wearing headsets, from exposure to loud, sustained, and piercing tones, such as those produced by a
fax machine. It is a workplace-safety feature for voice calls. When the tone is present at the input of the ASP
module, the audio path in the affected direction is muted to protect the listener, and a gentle alert tone is
played out for as long as the tone persists. ASP may be inserted in either or both directions of a call, that is,
applied to incoming packets to protect the ears of a listener on the Time-Division Multiplexing (TDM)
gateway, applied to incoming PSTN calls (microphone signal) to protect the ears of listeners at the other end
of the call, or applied to both simultaneously.

• Finding Feature Information, page 127

• Restrictions for ASP, page 127
• Information About ASP, page 128
• How to Configure ASP, page 129
• Configuration Examples for the Acoustic Shock Protection Feature, page 134
• Feature Information for Acoustic Shock Protection, page 135

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Restrictions for ASP

• Supported on PVDM3 only.
• Supported only on flex codec complexity.

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Information About ASP

• No support for H.32x video call, complex forking calls, and fax and modem calls.
• No support for TDM hairpin call.
• The configuration under dial peer has higher priority than the configuration at the global level.
• No support for conference calls, IP/SIP phones, and the Skinny Client Control Protocol (SCCP).
• CLI supports enabling ASP but not disabling ASP.
• No support for dynamically enabling or disabling ASP during a call.

Information About ASP

Acoustic Shock Protection

Acoustic Shock Protection (ASP) is an adaptive signal processing algorithm on the Digital Signal Processor
(DSP) that analyzes incoming audio for the presence of offending tones that might harm humans. Offending
tones include signals that are:
• Loud
• Tonal (energy concentrated around a single frequency)
• Persistent (lasts longer than a few tens of milliseconds)

If an offending tone is present, the audio path in that direction is muted temporarily, and a quiet, alerting signal
is played out to the listener side. The call is never dropped; only the audio is muted temporarily. If or when
the tone disappears from the input, the mute is removed. ASP does not disrupt low-frequency tones (below
650 Hz) such as ringback, dial, and so forth. Since ASP is designed to mute only single-frequency tones, it
allows multi-tone signals such as Dual Tone Multi-Frequency (DTMF) to pass unhindered. ASP is supported
on TDM gateways (TDM-VoIP and TDM-TDM) and on the Cisco Unified Border Element (Cisco UBE).

Note ASP is for voice calls only and not for faxes and modems.

Some of the best practices for ASP are as follows:

• Use default values
• Use ASP on dial peers where you are certain that people (not faxes) are listening.
• Do not use ASP on dial peers associated with fax machines, modems, or TTY/TDD devices. Use fax-relay
or modem-relay modes on dial peers dedicated to such devices.
• ASP is designed for deployment in situations where customers have experienced acoustic shock safety
issues. If there are issues like false triggering (for example, ASP alerts on regular voices), then you must
turn off ASP. You can choose from three detector sensitivity modes: slow, auto, or fast. Fast mode is a
highly sensitive hair-trigger. Auto mode is recommended. Slow mode lets more tone leak through, but
has better rejection of false triggers.

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 Acoustic Shock Protection
How to Configure ASP

How to Configure ASP

Creating the Media Profile for ASP

Perform this task to create a media profile to configure acoustic shock protection.

SUMMARY STEPS

1. enable
2. configure terminal
3. media profile asp tag
4. mode mode
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Device> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Device# configure terminal

Step 3 media profile asp tag Creates the media profile to configure ASP and enters media profile
configuration mode. The range for the media profile tag is from 1 to 10000.
Example:
Device(config)# media profile asp 5

Step 4 mode mode Sets the ASP sensitivity mode to preset = auto (which is default). Auto
mode provides a good tradeoff between ASP speed and false trigger
Example: rejection.
Device(cfg-mediaprofile)# mode auto
The other modes are:
• slow—Presets ASP sensitivity mode to 1. This mode provides slower
detection speed for reduced chance of false triggers.
• fast—Presets ASP sensitivity mode to 2. This mode provides faster
detection speed but higher chance of false triggers.
• expert—This mode exposes direct control of individual ASP
parameters and is recommended for test use only.

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Creating the Media Profile to Enable ASP

Command or Action Purpose

Step 5 end Returns to privileged EXEC mode.

Example:
Device(config)# end

Creating the Media Profile to Enable ASP

After the media profile is created, you must create a media class to enable acoustic shock protection. Perform
this task to create a media class.

SUMMARY STEPS

1. enable
2. configure terminal
3. media class tag
4. asp profile tag
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Device> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Device# configure terminal

Step 3 media class tag Creates the media class to enable the acoustic shock protection
feature and enters media class configuration mode. The range
Example: for the media class tag is from 1 to 10000.
Device(config)# media class 2

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Configuring the Media Class at a Dial Peer Level for ASP

Command or Action Purpose

Step 4 asp profile tag Applies the media profile to the media class. The range for the
media profile ASP tag is from 1 to 10000.
Example:
Device(cfg-mediaclass)# asp profile 200

Step 5 end Returns to privileged EXEC mode.

Example:
Device(cfg-mediaclass)# end

Configuring the Media Class at a Dial Peer Level for ASP

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag pots
4. media-class tag
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Device> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Device# configure terminal

Step 3 dial-peer voice tag pots Defines a particular dial peer and enters dial-peer voice
configuration mode. The range for the dial-peer voice tag is
Example: from 1 to 1073741823.
Device(config)# dial-peer voice 20 pots

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Configuring the Media Class Globally for ASP

Command or Action Purpose

Step 4 media-class tag Applies the media class to the specific dial peer. The range
for the media class tag number is from 1 to 10000.
Example:
Device(config-dial-peer)# media-class 2

Step 5 end Returns to privileged EXEC mode.

Example:
Device(config-dial-peer)# end

Configuring the Media Class Globally for ASP

SUMMARY STEPS

1. enable
2. configure terminal
3. media service
4. enhancement
5. tdm tag
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Device> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Device# configure terminal

Step 3 media service Enters media service configuration mode.

Example:
Device(config)# media service

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Verifying ASP

Command or Action Purpose

Step 4 enhancement Enters the submode enhance of media service.

Example:
Device(cfg-mediaservice)# enhancement

Step 5 tdm tag Applies the TDM call globally. The range for the media
class tag number is from 1 to 10000.
Example:
Device(cfg-service-enhance)# tdm 2

Step 6 end Returns to privileged EXEC mode.

Example:
Device(config-dial-peer)# end

Verifying ASP
Perform this task to verify the voice quality metrics.

SUMMARY STEPS

1. enable
2. show call active voice stats | b pid:

DETAILED STEPS

Step 1 enable

Example:
Device> enable

Enables privileged EXEC mode.

Step 2 show call active voice stats | b pid:

Example:
Device# show call active voice stats | b pid:1300
11EC : 5 09:14:25.971 PDT Thu Jul 28 2011.1 +1130 pid:1300 Answer 1300 active dur 00:01:36 tx:17/321
rx:17/321 dscp:0 media:0
DSP/TX: PK=17, SG=0, NS=1, DU=90570, VO=320
DSP/RX: PK=17, SG=0, CF=1, RX=90570, VO=320, BS=0, BP=0, LP=0, EP=0
….

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Troubleshooting Tips

DSP/DL: RT=0, ED=0

MIC Direction:
DSP/NR: NR=1, ND=0, LV=257, IN=1, PN=0, ON=0
DSP/AS: AE=1, AD=0, AV=0, AM=0, NT=0, DT=0, TT=0, TD=0, LF=0, LD=0
EAR Direction:
DSP/NR: NR=0, ND=0, LV=0, IN=0, PN=0, ON=0
DSP/AS: AE=0, AD=0, AV=0, AM=0, NT=0, DT=0, TT=0, TD=0, LF=0, LD=0
11EC : 6 09:14:25.973 PDT Thu Jul 28 2011.2 +1130 pid:2300 Originate 2300 active dur 00:01:36 tx:17/457
rx:17/321 dscp:0 media:0
Telephony call-legs: 1
SIP call-legs: 0
H323 call-legs: 1

Displays information about digital signal processing (DSP) voice quality metrics.

Troubleshooting Tips
The following commands can help troubleshoot ASP:
• debug voip hpi all
• debug voip dsmp all
• debug voip dsm all
• debug voip vtsp all
• debug vpm dsp all

Configuration Examples for the Acoustic Shock Protection

Feature
Example: Enabling ASP Globally

media profile asp 6

!
media class 1
asp profile 6
!
media service
enhancement
tdm 1

Example: Enabling ASP on a Dial Peer

media profile asp 4

!
media class 1
asp profile 4
!
dial-peer voice 2100 pots
destination-pattern 2100
incoming called-number 1100

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Feature Information for Acoustic Shock Protection

media-class 1
port 0/2/0:1
forward-digits all
dial-peer voice 1300 voip
destination-pattern 1300 session target ipv4:1.2.146.102 media-class 1

Feature Information for Acoustic Shock Protection

The following table provides release information about the feature or features described in this module. This
table lists only the software release that introduced support for a given feature in a given software release
train. Unless noted otherwise, subsequent releases of that software release train also support that feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Table 10: Feature Information for Acoustic Shock Protection

Feature Name Releases Feature Information

Acoustic Shock Protection 15.2(2)T, 15.2(3)T Acoustic Shock Protection (ASP)
is a voice circuit-breaker feature
that is designed to protect users,
especially those wearing headsets,
from exposure to loud, sustained,
and piercing tones, such as those
produced by a fax machine. It is a
workplace-safety feature for voice
calls. ASP is supported on TDM
gateways and on Cisco UBE.
The following commands were
introduced or modified: media
profile asp, media service.

Acoustic Shock Protection Cisco IOS XE Release 3.6S Acoustic Shock Protection (ASP)
is a voice circuit-breaker feature
that is designed to protect users,
especially those wearing headsets,
from exposure to loud, sustained,
and piercing tones, such as those
produced by a fax machine. It is a
workplace-safety feature for voice
calls. ASP is supported on TDM
gateways and on Cisco UBE.
In Cisco IOS XE Release 3.6S, this
feature was implemented on the
Cisco Unified Border Element
(Enterprise)
The following commands were
introduced or modified: media
profile asp, media service.

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Feature Information for Acoustic Shock Protection

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 CHAPTER 9
Noise Reduction
Noise Reduction (NR) is a voice enhancement process that improves the quality of incoming speech that
has already been corrupted with background noise; for example, a voice conference participant speaking on
a cell-phone in a car. NR works best with steady state broadband noises like engine noise but not as well
with impulsive noises like nearby chatter.

• Finding Feature Information, page 137

• Prerequisites for Noise Reduction, page 137
• Restrictions for NR, page 138
• Information About NR, page 138
• How to Configure NR, page 139
• Configuration Examples for the NR feature, page 144
• Feature Information for Noise Reduction, page 145

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for Noise Reduction

Cisco Unified Border Element
• Cisco IOS Release 15.2(2)T, or a later release must be installed and running on your Cisco Unified
Border Element.

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 Noise Reduction
Restrictions for NR

Cisco Unified Border Element (Enterprise)

• Cisco IOS XE Release 3.6S or a later release must be installed and running on your Cisco ASR 1000
Series Router.

Restrictions for NR
• Supported only on PVDM3.
• Supported only on flex codec complexity.
• No support for H.32x video call, complex forking calls, and fax and modem calls.
• No support for Time-Division Multiplexing (TDM) hairpin call.
• Configurations under POTS dial peer has higher priority over VoIP dial peer for NR.
• Configurations under the dial peer has higher priority than configurations at the global level.
• No support for conference calls, IP/SIP phones, and the Skinny Client Control Protocol (SCCP).
• CLI supports enabling NR but not disabling NR.
• No support for dynamically enabling or disabling NR during a call.

Information About NR

Noise Reduction
Noise Reduction (NR) is an adaptive signal processing algorithm on the Digital Signal Processor (DSP) that
analyzes incoming audio, extracts a fingerprint of the background noise during talker pauses, and then performs
ongoing spectral subtraction of this noise after a short training period (a few seconds). NR constantly adapts
to changes in background noises over time.
NR can affect music on hold signals by making the music quieter. NR may disrupt fax/modem/TDD devices,
although it is designed to self-disable in those cases. Use modem-relay mode for reliable fax/modem
transmission. NR is supported on TDM gateways (TDM-VoIP and TDM-TDM) and on the Cisco Unified
Border Element (Cisco UBE).
Some of the best practices for NR are as follows:
• Use default values.
• Do not use NR on dial peers associated with fax machines. Use fax or modem-relay modes for those
dial peers.
• NR, when used without dynamic user control of intensity (as is the case with gateways), must be used
at a low intensity (default or lower) since it is always on. High intensity is dramatic for demonstrations
with loud background noises, but the NR process itself will degrade “normal” calls if NR is run at high
intensity.

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How to Configure NR

How to Configure NR

Creating the Media Profile for NR

Perform this task to create a media profile to configure noise reduction parameters.

SUMMARY STEPS

1. enable
2. configure terminal
3. media profile nr tag
4. intensity level
5. noisefloor level
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Device> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Device# configure terminal

Step 3 media profile nr tag Creates the media profile to configure noise reduction parameters
and enters media profile configuration mode. The range for the
Example: media profile tag is from 1 to 10000.
Device(config)# media profile nr 2

Step 4 intensity level Configures the intensity level or depth of the noise reduction
process. The range is from 0 to 6.
Example:
Device(cfg-mediaprofile)# intensity 2

Step 5 noisefloor level Configures the noise level, in dBm, above which NR will operate.
NR will allow noises quieter than this level to pass without
Example: processing. The range is from -58 to -20.
Device(cfg-mediaprofile)# noisefloor -50

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Creating the Media Class to Enable NR

Command or Action Purpose

Step 6 end Returns to the privileged EXEC mode.

Example:
Device(config)# end

Creating the Media Class to Enable NR

After the media profile is created, you must create a media class to enable noise reduction. Perform this task
to create a media class.

SUMMARY STEPS

1. enable
2. configure terminal
3. media class tag
4. nr profile tag
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Device> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Device# configure terminal

Step 3 media class tag Creates the media class to enable the noise reduction feature
and enters media class configuration mode. The range for the
Example: media class tag is from 1 to 10000.
Device(config)# media class 2

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Configuring the Media Class at a Dial Peer Level for NR

Command or Action Purpose

Step 4 nr profile tag Applies the media profile to the media class. The range for the
media profile NR tag is from 1 to 10000.
Example:
Device(cfg-mediaclass)# nr profile 200

Step 5 end Returns to privileged EXEC mode.

Example:
Device(config)# end

Configuring the Media Class at a Dial Peer Level for NR

Perform this task to configure the media class for a dial peer.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag pots
4. media-class tag
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Device> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Device# configure terminal

Step 3 dial-peer voice tag pots Defines a particular dial peer and enters the dial-peer voice
configuration mode. The range for the dial-peer voice tag is
Example: from 1 to 1073741823.
Device(config)# dial-peer voice 20 pots

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Command or Action Purpose

Step 4 media-class tag Applies the media class to the specific dial peer. The range
for the media class tag number is from 1 to 10000.
Example:
Device(config-dial-peer)# media-class 2

Step 5 end Returns to the privileged EXEC mode.

Example:
Device(config-dial-peer)# end

Configuring the Media Class Globally for NR

Perform this task to configure a media class globally.

SUMMARY STEPS

1. enable
2. configure terminal
3. media service
4. enhancement
5. tdm tag
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Device> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Device# configure terminal

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Verifying NR

Command or Action Purpose

Step 3 media service Enters media service configuration mode.

Example:
Device(config)# media service

Step 4 enhancement Enters the submode enhance of media service.

Example:
Device(cfg-mediaservice)# enhancement

Step 5 tdm tag Applies the TDM call globally. The range for the media
class tag number is from 1 to 10000.
Example:
Device(cfg-service-enhance)# tdm 2

Step 6 end Returns to the privileged EXEC mode.

Example:
Device(config-dial-peer)# end

Verifying NR
Perform this task to verify the voice quality metrics.

SUMMARY STEPS

1. enable
2. show call active voice stats | b pid:

DETAILED STEPS

Step 1 enable

Example:
Device> enable

Enables privileged EXEC mode.

Step 2 show call active voice stats | b pid:

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Troubleshooting Tips

Example:
Device# show call active voice stats | b pid:1300
11EC : 5 09:14:25.971 PDT Thu Jul 28 2011.1 +1130 pid:1300 Answer 1300 active dur 00:01:36 tx:17/321
rx:17/321 dscp:0 media:0
DSP/TX: PK=17, SG=0, NS=1, DU=90570, VO=320
DSP/RX: PK=17, SG=0, CF=1, RX=90570, VO=320, BS=0, BP=0, LP=0, EP=0
….
DSP/DL: RT=0, ED=0
MIC Direction:
DSP/NR: NR=1, ND=0, LV=257, IN=1, PN=0, ON=0
DSP/AS: AE=1, AD=0, AV=0, AM=0, NT=0, DT=0, TT=0, TD=0, LF=0, LD=0
EAR Direction:
DSP/NR: NR=0, ND=0, LV=0, IN=0, PN=0, ON=0
DSP/AS: AE=0, AD=0, AV=0, AM=0, NT=0, DT=0, TT=0, TD=0, LF=0, LD=0
11EC : 6 09:14:25.973 PDT Thu Jul 28 2011.2 +1130 pid:2300 Originate 2300 active dur 00:01:36 tx:17/457
rx:17/321 dscp:0 media:0
Telephony call-legs: 1
SIP call-legs: 0
H323 call-legs: 1

Displays information about digital signal processing (DSP) voice quality metrics.

Troubleshooting Tips
The following commands can help troubleshoot NR:
• debug voip hpi all
• debug voip dsmp all
• debug voip dsm all
• debug voip vtsp all
• debug vpm dsp all

Configuration Examples for the NR feature

Example: Enabling NR globally

media profile nr 1
intensity 1
!
media profile nr 2
!
media profile nr 3
intensity 2
!
media profile nr 4
intensity 3
!
media profile nr 5
intensity 2

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Feature Information for Noise Reduction

!
media profile nr 7
intensity 2
!
media profile asp 6
!
media class 1
nr profile 5
asp profile 6
!
media service
enhancement
tdm 1

Example: Enabling NR on a Dial Peer

media profile nr 1
intensity 1
!
media profile nr 2
intensity 2
!
media profile nr 3
intensity 2
!
media profile asp 4
!
media class 1
nr profile 2
asp profile 4
!
dial-peer voice 2100 pots
destination-pattern 2100
incoming called-number 1100
media-class 1
port 0/2/0:1
forward-digits all

dial-peer voice 1300 voip

destination-pattern 1300
session target ipv4:1.2.146.102
media-class 1

Feature Information for Noise Reduction

The following table provides release information about the feature or features described in this module. This
table lists only the software release that introduced support for a given feature in a given software release
train. Unless noted otherwise, subsequent releases of that software release train also support that feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

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Feature Information for Noise Reduction

Table 11: Feature Information for Noise Reduction

Feature Name Releases Feature Information

Noise Reduction 15.2(2)T, Noise Reduction (NR) is a voice
enhancement or restoration process
15.2(3)T
that improves the quality of
incoming speech that has already
been corrupted with background
noise. NR is supported on TDM
gateways and on the Cisco UBE.
The following commands were
introduced or modified: intensity,
media profile nr, media service,
and noisefloor.

Noise Reduction Cisco IOS XE Release 3.6S Noise Reduction (NR) is a voice
enhancement or restoration process
that improves the quality of
incoming speech that has already
been corrupted with background
noise. NR is supported on TDM
gateways and on Cisco UBE.
In Cisco IOS XE Release 3.6S, this
feature was implemented on the
Cisco Unified Border Element
(Enterprise).
The following commands were
introduced or modified: intensity,
media profile nr, media service,
noisefloor.

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 CHAPTER 10
Configuring Hardware Echo Cancellation on T1
E1 Multiflex Voice WAN Interface Cards
The multiflex trunk (MFT) dedicated echo cancellation modules (dedicated ECAN modules) are daughter
cards that attach to the second generation multiflex voice/WAN interface cards (MFT VWIC2 family). The
dedicated ECAN modules are available in 32-channel and 64-channel configurations (EC-MFT-32 and
EC-MFT-64), which match the requirements of the 1- and 2-port T1/E1 MFT VWIC2s, respectively. This
chapter describes the configuration to enable additional echo cancellation effectiveness:
• Control of the echo canceller provided through the size of the echo cancellation buffer, ranging from
24 milliseconds (ms) to 128 ms
• Processing and memory resources to ensure robust echo canceller coverage independent from the
configuration of the echo canceller or the demand placed on the general voice DSP resources

• Finding Feature Information, page 147

• Prerequisites for Hardware Echo Cancellation, page 148
• Restrictions for Hardware Echo Cancellation, page 148
• How to Configure Hardware Echo Cancellation, page 150

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

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Prerequisites for Hardware Echo Cancellation

Prerequisites for Hardware Echo Cancellation

Cisco IOS Image

To run hardware echo cancellation on T1/E1 interfaces, you must install an IP Plus or IP Voice image
(minimum) of Cisco IOS Release 12.3(14)T or a later release.

Baseboard and Daughter Card Configuration

Hardware echo cancellation is restricted to the same baseboard voice/WAN interface card (VWIC) on which
the daughter card (EC-MFT-32 and EC-MFT-64) is installed and cannot be shared by other T1/E1 controllers.

Restrictions for Hardware Echo Cancellation

Hardware Echo Cancellation Tail Length

If you are using hardware echo cancellation, the value for tail length is set to 128 ms. This is not configurable
and cannot be changed.

Accurate TDM ERL Readings for Echo Cancellation

To ensure accurate statistics for network monitoring and troubleshooting, an estimate of the quality of the
TDM connection and the ECAN's ability to discern and cancel out echo might be necessary. To ensure accurate
readings, you must configure software-based echo cancellation by entering the echo-cancel enable type
software command (Step 6 in the procedure in How to Configure Hardware Echo Cancellation, on page 150).
If you accept the default (hardware echo cancellation) or enter the echo-cancel enable type hardware
command, the output for the show voice call command always displays "TDM ERL Level(dBm0): +6.0."
If you enter the echo-cancel enable type software command to enable software-based echo cancellation,
the show voice call command output displays accurate real-time TDM ERL measurements. The sample output
examples provided in the following sections demonstrate the difference:

Sample Output of the show voice call command

The following is sample output of software-enabled echo cancellation--hardware echo cancellation is disabled.
Note the different values for the TDM ERL levels.

Router# show voice call 0/0/0:23.1

0/0/0:23 1
vtsp level 0 state = S_CONNECT
callid 0x0001 B01 state S_TSP_CONNECT clld 9011204 cllg 9011200
Router# ***DSP VOICE TX STATISTICS***
Tx Vox/Fax Pkts: 3563, Tx Sig Pkts: 0, Tx Comfort Pkts: 4
Tx Dur(ms): 80150, Tx Vox Dur(ms): 71200, Tx Fax Dur(ms): 0
.

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Accurate TDM ERL Readings for Echo Cancellation

.
.
***DSP LEVELS***
TDM Bus Levels(dBm0): Rx -12.5 from PBX/Phone, Tx -16.4 to PBX/Phone
TDM ACOM Levels(dBm0): +27.0, TDM ERL Level(dBm0): +27.0
TDM Bgd Levels(dBm0): -84.4, with activity being silence
***DSP VOICE ERROR STATISTICS***
Rx Pkt Drops(Invalid Header): 0, Tx Pkt Drops(HPI SAM Overflow): 0
Router# show voice call 0/0/0:23.2
0/0/0:23 2
vtsp level 0 state = S_CONNECT
callid 0x0002 B02 state S_TSP_CONNECT clld 9011202 cllg 9011205
Router# ***DSP VOICE TX STATISTICS***
Tx Vox/Fax Pkts: 1800, Tx Sig Pkts: 0, Tx Comfort Pkts: 0
Tx Dur(ms): 36000, Tx Vox Dur(ms): 36000, Tx Fax Dur(ms): 0
.
.
.
***DSP LEVELS***
TDM Bus Levels(dBm0): Rx -23.5 from PBX/Phone, Tx -36.5 to PBX/Phone
TDM ACOM Levels(dBm0): +6.0, TDM ERL Level(dBm0): +6.0
TDM Bgd Levels(dBm0): +0.0, with activity being silence
***DSP VOICE ERROR STATISTICS***
Rx Pkt Drops(Invalid Header): 0, Tx Pkt Drops(HPI SAM Overflow): 0
The following is sample output showing hardware echo cancellation--note that the TDM ERL level is +6.0
in both cases.

Router# show voice call 0/0/0:23.1

0/0/0:23 1
vtsp level 0 state = S_CONNECT
callid 0x0002 B01 state S_TSP_CONNECT clld 9011204 cllg 9011200
Router#
***HARDWARE ECHO CANCELLER STATISTICS***
Echo Canceller: On Tail-length: 128ms
H-Register: Update Modem tone disable: Ignore 2100Hz tone
Worst ERL : 6dB Residual Control: Comfort noise
High level compensation: Off
Tx Power = 0.0dB Tx Avg Power = 0.0dB
Rx Power = 0.0dB Rx Avg Power = 0.0dB
ERL = 27.0dB ACOM = 0.0
3 Reflectors(Tails) = (1, 0, 0)Ms, Max Reflector = 1Ms
Ecan Status words 0x7C, 0x1001
EC Lib version: 9183.890
.
.
.
***DSP LEVELS***
TDM Bus Levels(dBm0): Rx -12.4 from PBX/Phone, Tx -15.1 to PBX/Phone
TDM ACOM Levels(dBm0): +6.0, TDM ERL Level(dBm0): +6.0
TDM Bgd Levels(dBm0): -84.4, with activity being silence
***DSP VOICE ERROR STATISTICS***
Rx Pkt Drops(Invalid Header): 0, Tx Pkt Drops(HPI SAM Overflow): 0
Router# show voice call 0/0/0:23.2
0/0/0:23 2
vtsp level 0 state = S_CONNECT
callid 0x0004 B02 state S_TSP_CONNECT clld 9011202 cllg 9011205
cmohanan-3845#
***HARDWARE ECHO CANCELLER STATISTICS***
Echo Canceller: On Tail-length: 128ms
H-Register: Update Modem tone disable: Ignore 2100Hz tone
Worst ERL : 6dB Residual Control: Comfort noise
High level compensation: Off
Tx Power = 0.0dB Tx Avg Power = 0.0dB
Rx Power = 0.0dB Rx Avg Power = 0.0dB
ERL = 6.0dB ACOM = 0.0
3 Reflectors(Tails) = (4, 0, 0)Ms, Max Reflector = 4Ms
Ecan Status words 0x7C, 0x1001
EC Lib version: 9183.890
.
.
.

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How to Configure Hardware Echo Cancellation

***DSP LEVELS***
TDM Bus Levels(dBm0): Rx -24.9 from PBX/Phone, Tx -35.7 to PBX/Phone
TDM ACOM Levels(dBm0): +6.0, TDM ERL Level(dBm0): +6.0
TDM Bgd Levels(dBm0): +0.0, with activity being silence
***DSP VOICE ERROR STATISTICS***
Rx Pkt Drops(Invalid Header): 0, Tx Pkt Drops(HPI SAM Overflow): 0

Figure 16: Sample Network Topology for the T1/E1 Multiflex Voice/WAN Interface Cards with Echo Cancellation Module

How to Configure Hardware Echo Cancellation

To configure hardware echo cancellation on T1/E1 multiflex voice/WAN interface cards, complete the
following tasks.

SUMMARY STEPS

1. enable
2. configure terminal
3. card type {e1 | t1} slot subslot
4. voice-card slot
5. voice-port {slot-number / subunit-number / port | slot / port : ds0-group-number}
6. echo-cancel enable type [hardware | software]
7. echo-cancel coverage {24| 32| 48| 64| 80| 96| 112| 128}
8. exit

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How to Configure Hardware Echo Cancellation

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 card type {e1 | t1} slot subslot Sets or changes the card type to E1 or T1.
• slot --Specifies the slot number. Range can be 0 to 6, depending on the platform.
Example:
• subslot --Specifies the VWIC slot number. Range can be 0 to 3, depending on
Router(config)# card type t1
1 0 the host module or platform.
• When the command is used for the first time, the configuration takes effect
immediately.
• A subsequent change in the card type will not take effect unless you enter the
reloadcommand or reboot the router.

Note When you are using the card type command to change the configuration of
an installed card, you must enter the no card type e1 | t1} slot subslot
command first. Then enter the card type {e1 | t1} slot subslot command for
the new configuration information.
Step 4 voice-card slot Enters voice card configuration mode.
• Specify the slot location using a value from 0 to 5.
Example:
Router(config)# voice card 1

Step 5 voice-port {slot-number / Enters voice port configuration mode and specifies the voice port.
subunit-number / port | slot / port :
ds0-group-number} • The slot-numberargument identifies the slot where the voice interface card (VIC)
is installed. Valid entries are from 0 to 3, depending on the slot in which it has
been installed.
Example:
• The subunit-number identifies the subunit on the VIC where the voice port is
Router(voice-card)# voice-port
3/0:0 located. Valid entries are 0 or 1.
• The port argument identifies the voice port number. Valid entries are 0 and 1.

or
• The slot argument is the slot in which the voice port adapter is installed. Valid
entries are from 0 to 3.

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Examples

Command or Action Purpose

• The port argument is the voice interface card location. Valid entries are 0 to 3.
• The ds0-group-numberargument indicates the defined DS0 group number. 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.

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 to determine
the syntax choices that are available.
Step 6 echo-cancel enable type Enables hardware echo cancellation.
[hardware | software]
• The hardware keyword is the default. Echo cancel coverage is hardcoded for
128 ms.
Example:
• This command is needed only to configure the software keywordto effect
router(config-voiceport)#
echo-cancel enable type software-based (DSP) echo cancellation or to restore the hardware default.
hardware
Note The hardware and software keywords are available only when the optional
hardware echo cancellation module (EC-MFT-32 or EC-MFT-64) is installed
on the multiflex VWIC.
Note If you need to obtain accurate, real-time readings for the quality of the TDM
connection and the echo canceller's ability to discern and cancel out echo,
you should enter the echo-cancel enable type software command. See
theRestrictions for Hardware Echo Cancellation, on page 148 for more
information.
Step 7 echo-cancel coverage {24| 32| 48| Adjusts the echo canceller by the specified number of milliseconds.
64| 80| 96| 112| 128}
• These coverage options are applicable only if you configured the echo-cancel
enable type software command in the previous step.
Example:
• If you configured the echo-cancel enable type hardware command in the
Router (config-voiceport) #
echo-cancel coverage 96 previous step, this value is set to 128 ms.
• Beginning with Release 12.4(20)T, the default for software echo cancellation
is 128 ms. Prior to Release 12.4(20)T, the default is 64 ms.

Step 8 exit Exits controller cofiguration mode and returns the router to privileged EXEC mode.

Example:
Router(config-voiceport)# exit

Examples
This section provides the following examples for verifying echo cancellation:

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Examples

show echo-cancel hardware status Example

The output in this section shows that hardware echo cancellation is enabled on slot 1.

Router_3725# show echo-cancel hardware status 1

VWIC HWECAN 1/0 is UP.

Software version:4.4.803 , Date:Feb 6 16:58:57 2004
Tail length:128 Tone disabler type:G.165 Fax notify: Off
Device:VWIC_8MBPS_1TIEC_TL128_MS_1P Max Channels:32
Only Port0 have Local HWECAN Connectivity.

ECAN CH ASSIGNED DSP ID VOICEPORT EC NLP COV LAW

============================================

1 yes 1/1 1/0:1.1 on off on u-Law

Total assigned channel(s):1

Total device(s) in the slot 1

show call active voice echo-canceller summary Example

The output in this section shows summary information for the hardware echo cancellation.

Router_3725# show call active voice echo-canceller summary

Call ID Port DSP/Ch Codec Ecan-type Tail Called Dial-peers

============================================

0xE71 1/0:1.1 1/1 g729r8 HW 128ms 1000 1/10

1 active call found

number of hardware ecan channels:1
number of software ecan channels:0

show call active voice echo-canceller CallID Example

The output in this section shows hardware echo canceller information for an active voice call.

Router# show call active voice echo-canceller E71

Device:VWIC HWECAN 1/0 Channel Id = 1 Tail = 128Ms

Software version:4.4.803 , Date:Feb 6 16:58:57 2004
Echo Canceller:On Tail-length:128ms
H-Register:Update Modem tone disable:Ignore 2100Hz tone
Worst ERL :6dB Residual Control:Cancel only
High level compensation:Off
Tx Power = 0.0dB Tx Avg Power = 0.0dB

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Examples

Rx Power = 0.0dB Rx Avg Power = 0.0dB

ERL = 1.0dB ACOM = 0.0
3 Reflectors(Tails) = (90, 0, 0)Ms, Max Reflector = 90Ms
Ecan Status words 0x1C, 0x00
EC Lib version:9155
More detailed syntax information about the commands used in this chapter is documented in the Cisco IOS
Voice Command Reference .

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 CHAPTER 11
NextPort-Based Voice Tuning and Echo
Cancellation
This chapter describes how to dynamically configure voice services on the NextPort-based platforms: Cisco
AS5350, Cisco AS5400, Cisco AS5400HPX, and Cisco AS5850. This chapter contains the following sections:

• Finding Feature Information, page 155

• Prerequisites for NextPort Services, page 155
• Information About NextPort Voice Services, page 156
• How to Configure NextPort Services, page 158

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Prerequisites for NextPort Services

To use the Nextport-Based Voice Tuning and Background Noise Statistics feature, you must be running
NextPort service processing element (SPE) firmware version 8.8.1 or a later version and Cisco IOS Release
12.3(4)T or a later release.
To use the NextPort dual-filter G.168 echo canceller, you must be running SPE firmware version 10.2.2 or a
later version and Cisco IOS Release 12.3(11)T or a later release.

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Information About NextPort Voice Services

Information About NextPort Voice Services

NextPort Dual-Filter G.168 Echo Canceller

The CSMV6 dial feature card (DFC) for NextPort platforms offers dual-filter G.168 echo canceller capability.
The NextPort dual-filter G.168 echo canceller (EC) improves voice quality in VoIP connections by providing
relatively less residual echo leakage, better nonlinear processing (NLP) timing, less clipping, and better comfort
noise generation (CNG) in most environments.
The dual-filter G.168 echo canceller features two concurrently operating adaptive filters (which control echo
tail coverage) and two double-talk detection functions. In addition, the comfort noise model uses "Hoth noise"
spectrum shaping to better replicate the true noise spectrum.

Note Detailed information for all Cisco IOS commands mentioned in this section can be found in the Cisco
IOS Voice Command Reference.

The NextPort dual-filter G.168 echo canceller uses the same voice-tuning (VC tune) interface for configuring
voicecap parameters as the Cisco-proprietary G.164 echo canceller. To adjust the dual-filter echo canceller,
use a voicecap or the Cisco IOS command-line interface (CLI) during configuration. You can also adjust
settings while the system is running by using the show port log and show port operational-statuscommands.
However because of the differences in internal operation of these ECs, there are some changes in the set of
available parameters for voice tuning.
See the echo-cancel coverage command for updated Cisco IOS command usage with this feature. The NextPort
dual-filter G.168 echo canceller adds the following benefits on NextPort platforms:
• Configurable parameters--Range checking that is performed on the voicecap parameters in the I960
NextPort layer has been updated. (Voicecap parameters in "raw mode" are never range-checked.)
• Up to 128 ms of echo tail coverage--Beginning with Cisco IOS Release 12.4(20)T, the NextPort dual-filter
G.168 echo canceller supports echo tails from 24-ms to 128-ms in 16-ms increments. The echo-cancel
coverage command limits the echo canceller coverage to 128-ms on NextPort platforms. For backward
compatibility, a voicecap used in "raw mode" will still configure older SPEware to settings greater than
64-ms when used with newer releases of Cisco IOS software. For situations when new SPEware is
loaded onto an older Cisco IOS release, the NextPort dual-filter G.168 echo canceller automatically sets
coverage time to 64 ms.
• Updated set of reported statistics--Text in the show voice port command output has been changed to
describe voicecap parameters and reported statistics. The show port operational-status command output
has been updated to report TX/RX mean speech level statistics.
• Power statistics (RX and TX)--These statistics average only the power that is received during signal
periods that are classified as speech.
• Unchanged configuration steps--Use voicecaps and the echo-cancel coverage command to configure
this feature. See the Voicecap Strings, on page 157.
• SPE firmware and Cisco IOS software packaging support--The SPEware that contains the dual-filter
G.168 echo canceller is field-upgradeable and can be used interchangeably with previous firmware
versions with no effect on platform call density. The new SPEware interoperates with any Cisco IOS
software release that supports voicecaps.

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NextPort SPE Firmware

Note When older Cisco IOS software releases are used, voicecaps must be used in raw mode for some parameters.
Some statistics may not be displayed or recorded properly with older software releases.

NextPort SPE Firmware

NextPort SPE firmware is software that drives the digital signal processor (DSP) portion of the NextPort dial
feature cards (DFCs). NextPort firmware is bundled with Cisco IOS software.
NextPort SPE firmware runs on the NextPort DFC60, DFC108, 1 CT3_UPC 216, and UPC324 DFCs on the
Cisco AS5350, Cisco AS5400, Cisco AS5400HPX, and Cisco AS5850 platforms. The ports on these modules
can support modem, voice, fax, and digital services and can be aggregated at any of the following levels:
• Slot level of the NextPort module
• SPE level within the NextPort module
• Individual port level

Note To use the NextPort Voice Tuning and Background Noise Statistics feature, you must use the default
bundled NextPort SPE firmware code that runs with Cisco IOS software. The NextPort-Based Voice
Tuning and Background Noise feature uses SPE firmware version 8.8.1 or a later version. The NextPort
dual-filter G.168 echo canceller uses NextPort firmware version 10.2.2, which is bundled with Cisco IOS
Release 12.3(11)T. NextPort firmware version 10.2.2 can be used with Cisco IOS Release 12.3(7)T,
12.3(10), and later releases.

For more information about NextPort SPE firmware, see the NextPort SPE Release Notes on Cisco.com.

Voicecap Strings
Additional configuration of voice services on NextPort DFCs is achieved by configuring the voice tuning
configuration capability (called voicecaps) using voicecap strings. Voicecap strings are created with the the
voicecap entry command and are applied with the voicecap configure command.

Voice Tuning
This feature allows the following parameters, among others, to be configured:
• PSTN gains--PSTN gains adjust the power levels at the PSTN side of a VoIP connection to make up
for loss plan imbalances and to ensure minimum echo return losses (ERLs) in a call. PSTN gain is
configured with the CLI rather than with voicecaps.
• IP gains--IP gains adjust IP-side levels and are applied to the signal before it is propagated through the
echo canceller. This point is also known as the reference signal.
• Dynamic attenuation--Dynamic attenuation mitigates low volume calls when attenuation has been added
on the PSTN call leg to compensate for low ERL calls.

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Background Noise

• Comfort noise generation--CNG enables and disables EC comfort noise.

• Minimum ERL--Minimum ERL switches off near-end talker clipping and poor echo canceller
performance.

Note You must have specific knowledge of the behavior of the telephone network in order to use these voicecap
capabilities.

Background Noise
The NextPort Voice Tuning and Background Noise Statistics feature reports EC background noise level, voice
activity detection (VAD) background noise level, ERL level, and Acombined (ACOM) statistics by averaging
the combined values that are computed over the duration of the call. These statistics are appended to the end
of each entry in the voice log, which you can see in the output from the show port log and show port
operational-status commands.

How to Configure NextPort Services

To configure the Nextport-Based Voice Tuning and the NextPort Dual-Filter G.168 Echo Canceller feature,
complete the following tasks:

Downloading NextPort SPE Firmware

To download NextPort SPE firmware, use the following commands:

SUMMARY STEPS

1. enable
2. configure terminal
3. spe {first slot | first slot / spe} {last slot | last slot / spe}
4. firmware location [IFS :[/]]filename
5. end
6. copy running-config startup-config

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

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Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 spe {first slot | first slot / spe} Enters SPE configuration mode and sets the range of SPEs.
{last slot | last slot / spe}
• first slot and last slot--Identifies slots for the range. For the Cisco AS5350, slot
values range from 1 to 3. For the Cisco AS5400, slot values range from 1 to 7.
Example: All ports on the specified slot are affected.
Router(config)# spe 1 1/17
• first slot / spe and last slot/spe--Identifies slots for the range. For the Cisco
AS5350, slot values range from 1 to 3. For the Cisco AS5400, slot values range
from 1 to 7. SPE values range from 1 to 17. You must include the slash mark. All
ports on the specified slot and SPE are affected.

Step 4 firmware location [IFS Downloads SPE modem code to all modems in a particular slot (that is, all modems on
:[/]]filename a feature card that contains 18 6-port modem modules).
• IFS --(Optional) Cisco IOS file specification (IFS), which can be any valid IFS
Example: on any local file system. Examples of legal specifications include:
Router(config-spe)# firmware
location flash:np.8.8.1.spe • bootflash:--Loads the firmware from a separate flash memory device.
• flash:--Loads the firmware from the flash NVRAM located within the router.
• null:--Specifies a firmware file from null: File System.
• system:/--Loads the firmware from a built-in file within the Cisco IOS image.
The optional forward slash (/) and system path must be entered with this
specification.

• filename --The firmware filename. When the filename is entered without an IFS
specification, this name defaults to the file in flash memory.
• Use the dir all-filesystems EXEC command to display legal IFSs.
• The no form of the command reverts the router back to the system-embedded
default. When the access server is booted, the firmware location command
displays the location for the firmware that is embedded in the Cisco IOS image.
If the firmware locationcommand is issued to download a firmware image from
flash and then the no version of the exact command is subsequently issued, then
the firmware location command downloads the embedded firmware in Cisco
IOS software.

Step 5 end Completes the download and exits SPE configuration mode.

Example:
Router(config-spe)# end

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Command or Action Purpose

Step 6 copy running-config Copies the configuration from running RAM into NVRAM.
startup-config
• The download occurs when the modems become available and the display shows
the SPE firmware upgrade option defined (default: busyout).
Example:
• The spe command generates NVRAM modem download and configuration file
Router# copy running-config
startup-config entries.

Note If the configuration is not saved as described in this step, download of the
firmware specified with the spe command will not occur after the next reboot.
• For detailed information on the spe command, see the following Cisco document:
"SPE and Firmware Download Enhancements".

Creating and Applying Voicecaps

Voicecaps can be used to configure any of the voice service parameters. Voicecaps are, however, primarily
used to configure only those parameters that do not have associated Cisco IOS commands.

Restrictions
• Voicecaps are configured in global configuration mode. A maximum of five voicecap entries can be
defined.
• Applying a voicecap is possible only in voice-port configuration mode. Once applied to a voice port,
the voicecap affects all calls associated with that voice port.
• To achieve the specified functionality, an SPE image capable of voice tuning must be used in conjunction
with the Cisco IOS software and module controller software.
• For backward compatibility, a voicecap used in raw mode will configure older SPEware to allow echo
canceller coverage settings greater than 64 ms when the older SPEware is used with newer releases of
Cisco IOS software. For situations when new SPEware is loaded onto an older Cisco IOS software
release, the NextPort dual-filter G.168 echo canceller automatically sets coverage time to 64 ms.

Note Voicecap parameters in raw mode are never range-checked.

For a list of available voicecap parameters and code words that are used with the NextPort dual-filter G.168
echo canceller feature, see the "NextPort-Based Voice Tuning and Echo Cancellation Guide".

Setting Voice Tuning Parameters with V Registers

The following sections contain information about voice tuning parameters with V registers:

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Setting Voice Tuning Parameters with V Registers

Note All data specified in dB is entered in the form (dB * 10). So, for example, to specify 6.0 dB, 60 must be
entered.

Set PSTN Gains

To fix poor loss plans and to ensure minimum ERLs, use PSTN gains to adjust the power levels on the PSTN
call leg. To adjust these levels, make sure that the sum of the output attenuation, the input attenuation, and
the lowest expected functional ERL is greater than the MinERL parameter setting described below. You
should balance the power levels of the far-end talker and near-end talker as seen at the echo canceller.

Note Too much attenuation may cause some calls to have low volume speech. For more information, see the
dynamic attenuation feature described in the "NextPort-Based Voice Tuning and Echo Cancellation Guide".

Note Use the Cisco IOS CLI, not voicecap indexes, to set PSTN gains.

Set IP Gains
To adjust IP-side levels that are applied to the signal before it is propagated through the echo canceller, use
IP gains. IP gains are controlled with the following V registers. The valid range for both input and output gain
is -14 dB to 14 dB.
• v261--IP output gain.
• v263--IP input gain.

Note There have been some instances where the IP-side power has been too high. Using index v263 can mitigate
this problem.

Set Dynamic Attenuation

To mitigate low volume calls when attenuation has been added on the PSTN call leg to compensate for low
ERL calls, use dynamic attenuation. Dynamic attenuation is controlled with the following V registers:
• v289--Dynamic EC Attenuation Feature Enable. Set to 1 to enable. Set to 0 to disable.
• v290--Dynamic EC Attenuation Minimum ERL Value. Valid range is from 0 dB to 60 dB.
• v291--Dynamic EC Attenuation Final Rout Gain. Set to the lowest level desired for PSTN output
attenuation. This value is usually set to 0. Valid range is from -14 dB to 6 dB.
• v292--Dynamic EC Attenuation Final Sin Gain. Set to the lowest level desired for PSTN input attenuation.
This value is usually set to 0. Valid range is from -14 dB to 6 dB.

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Set Comfort Noise Generation

During periods of far-end single talk, the echo canceller engages the non-linear processor (NLP) to suppress
residual echo. However, it will also suppress any noise signal that is coming from the near-end side. This can
lead to dead silence, which the listener may confuse with a dropped call. To overcome this condition, comfort
noise generation (CNG) is added.
To choose between NLP silence or background noise reproduction, use comfort noise generation. CNG is
controlled with the following V register:
• v294--CNG Enable. Set to 1 to enable. Set to 0 to disable.

Set Minimum ERL

The echo canceller uses the minimum ERL (MinERL) value to decide whether the incoming signal on the
PSTN call leg is an echo or a near-end talker. If this value is too high, the echo canceller will not properly
identify echo and will not adapt. If this value is too low, clipping of the near-end talker may occur.
To reduce near-end talker clipping and poor echo canceller performance, use minimum ERL. MinERL is
controlled with the following V register:
v270--Sets the level that the echo canceller expects the lowest ERL of the PSTN to be. The valid range is
from 0 dB to 20 dB. The default is 6.

Creating and Applying Voice Caps

To create and apply voice caps and voice cap entries, complete the following tasks:

SUMMARY STEPS

1. enable
2. configure terminal
3. voicecap entry name string
4. voice-port slot / port :D
5. voicecap configure name
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

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Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voicecap entry name string Creates a voicecap.

• The name argument is a word that uniquely identifies this voicecap.
Example:
• The string argument is a string of characters composed of a series of
Router(config)# voicecap entry
qualityERL v270=120 voicecap register entries, similar to a modemcap. Each entry is of the
form vINDEX=VALUE, where INDEX refers to a specific V register,
and VALUE designates the value to set that V register to.
• For more information about V_register entries, see the Setting Voice
Tuning Parameters with V Registers, on page 160.
• The example creates a simple voicecap string named "qualityERL"
with V register 270 set to 120.

Step 4 voice-port slot / port :D Enters voice-port configuration mode on the selected slot and port.

Example:
Router(config)# voice-port 3/0:D

Step 5 voicecap configure name Applies a voicecap.

• The name argument designates which of the newly created voicecaps
Example: to use on this voice port. This character value must be identical to the
Router(config-voiceport)# voicecap value entered when you created the voicecap entry. In this case, the
configure qualityERL value is "qualityERL."

Note To configure multiple voice ports, repeat Step 4 and Step 5 for
each voice port.
Step 6 exit Exits voice-port configuration mode and completes the configuration.

Example:
Router(config-voiceport)# exit

Verifying Voicecap Configurations

Use the following show commands in privileged EXEC mode to verify your configuration. Relevant fields
are shown in bold.

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SUMMARY STEPS

1. show voice port

2. show port operational-status slot / port
3. show port voice log

DETAILED STEPS

Step 1 show voice port

Use this command to display configured voicecaps, for example:

Example:
Router# show voice port
ISDN 2/0:D - 2/0:D
Type of VoicePort is ISDN
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
.
.
.
Station name None, Station number None
Translation profile (Incoming):
Translation profile (Outgoing):
Voicecap:EXAMPLE

Step 2 show port operational-status slot / port

Use this command to display background noise level information on current calls. Significant fields are shown in bold
in the following example:

Example:
Router# show port operational-status 1/0
Slot/SPE/Port -- 1/0
Service Type :Voice service
Voice Codec :G.711 u-law
Echo Canceler Length :8 ms
Echo Cancellation Control :Echo cancellation - enabled
Echo update - enabled
Non-linear processor - enabled
Echo reset coefficients - disabled
High pass filter enable - disabled
Digit detection enable :DTMF signaling - enabled
Voice activity detection :Disabled
Comfort noise generation :Generate comfort noise
Digit relay enable :OOB Digit relay - disabled
IB Digit relay - disabled
Information field size :20 ms
Playout de-jitter mode :adaptive
Encapsulation protocol :RTP
Input Gain :0.0 dB
Output Gain :0.0 dB
Tx/Rx SSRC :20/0
Current playout delay :65 ms
Min/Max playout delay :65/105 ms
Clock offset :142003 ms
Predictive concealment :0 ms
Interpolative concealment :0 ms

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Silence concealment :0 ms
Buffer overflow discards :1
End-point detection errors :0
Tx/Rx Voice packets :1337/1341
Tx/Rx signaling packets :0/0
Tx/Rx comfort noise packets :0/0
Tx/Rx duration :26745/26745 ms
Tx/Rx voice duration :0/0 ms
Out of sequence packets :0
Bad protocol headers :0
Num. of late packets :0
Num. of early packets :1
Tx/Rx Power :-87.0/-57.3 dBm
Tx/Rx Talker Level :-86.3/-57.0 dBm
TX/RX Mean Speech level :-86.3/-57.0 dBm
VAD Background noise level :6.2 dBm
ERL level :127.0 dB
ACOM level :127.0 dB
Tx/Rx current activity :silence/silence
Tx/Rx byte count :213920/214240
ECAN Background noise level :-83.4 dBm
Latest SSRC value :391643394
Number of SSRC changes :1
Number of payload violations :0

Step 3 show port voice log

Use this command to display background noise level information on completed calls. Significant fields are shown in
bold in the following example:

Example:
Router# show port voice log
Port 1/00 Events Log
*Aug 22 07:59:27.515:Voice Terminate event:
Disconnect Reason : normal call clearing (16)
Call Timer : 57 secs
Current playout delay : 65 ms
Min/Max playout delay : 65/105 ms
Clock offset : 142003 ms
Predictive concealment : 0 ms
Interpolative concealment : 0 ms
Silence concealment : 0 ms
Buffer overflow discards : 1
End-point detection errors : 0
Tx/Rx Voice packets : 2813/2816
Tx/Rx signaling packets : 0/0
Tx/Rx comfort noise packets : 0/0
Tx/Rx duration : 56260/56260 ms
Tx/Rx voice duration : 0/0 ms
Out of sequence packets : 0
Bad protocol headers : 0
Num. of late packets : 0
Num. of early packets : 1
Tx/Rx Power : -87.0/-57.3 dBm
Tx/Rx Mean Speech Level : -86.7/-57.0 dBm
Tx/Rx Talker Level : -86.3/-57.0 dBm
Average VAD Background noise level : 6.2 dBm

Average ERL level : 127.0 dB

Average ACOM level : 127.0 dB
Tx/Rx current activity : silence/silence
Tx/Rx byte count : 450080/450240
Average ECAN Background noise level: -83.4 dBm
*Aug 22 07:59:27.515:Voice SSRC change events:
Latest ssrc value : 391643394
Total ssrc changes : 1

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Troubleshooting NextPort Voicecaps

Use the following debug and showcommands in privileged EXEC mode to debug the application of a voicecap
and to check debugging output:

SUMMARY STEPS

1. debug nextport vsmgr detail

2. debug dspapi detail
3. show debug

DETAILED STEPS

Step 1 debug nextport vsmgr detail

Use this command to turn on debugging for NextPort voice services, for example:

Example:
Router# debug nextport vsmgr detail
NextPort Voice Service Manager:
NP Voice Service Manager Detail debugging is on
.
.
.

Step 2 debug dspapi detail

Use this command to turn on debugging for DSP API message event details, for example:

Example:
Router# debug dspapi detail
DSP API:
DSP API Command debugging is on
DSP API Detail debugging is on
.
.
.

Step 3 show debug

Use this command to check voicecap application debugging. The significant field in the output is highlighted in bold in
the following example:

Example:
Router# show debug
NextPort Voice Service Manager:
NP Voice Service Manager Detail debugging is on
DSP API:
DSP API Command debugging is on
DSP API Detail debugging is on
*Aug 22 08:34:47.399:dspapi [2/1:1 (4)] dsp_init

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*Aug 22 08:34:47.399:dspapi [2/1:1 (4)] dsp_voice_config_params:10 params

*Aug 22 08:34:47.399: [0] ENCAP RTP:t_ssrc=20 r_ssrc=0 t_vpxcc=0 r_vpxcc=0
ifp_payload_type=122 sid_support=19 tse_payload=101 seq_num_start=6303
redundancy=0 cc_payload_typ
Router# e=125 fax_payload_type=122 alaw_pcm_switchover=8 mulaw_pcm_switchover=0 dtmf_payload_type=121,
nte_rcv_payload_type=101dynamic_payload=0, codec=5
*Aug 22 08:34:47.399: [1] PO_JITTER:mode=2 initial=60 max=200 min=40 fax_nom=300
*Aug 22 08:34:47.399: [2] INBAND_SIG:mode=0x1 enable
*Aug 22 08:34:47.399: [3] ECHO_CANCEL:flags=0x17 echo_len=256
*Aug 22 08:34:47.399: [4] IDLE_CODE_DET:enable = 0 code=0x0 duration=6000
*Aug 22 08:34:47.403: [5] GAIN:input=0 output=0
*Aug 22 08:34:47.403: [6] CNG:
Router# 1
*Aug 22 08:34:47.403: [7] INFO_FIELD_SIZE:160
*Aug 22 08:34:47.403: [8] DIGIT_RELAY:2
*Aug 22 08:34:47.403: [9] VOICECAP:EXAMPLE
*Aug 22 08:34:47.403:dspapi [2/1:1 (4)] dsp_start_service:G711_U (5)
*Aug 22 08:34:47.403:Matched voicecap:v0=0 v1=1
*Aug 22 08:34:47.403:msg length = 0x001D
*Aug 22 08:34:47.403:session ID = 0x006D
*Aug 22 08:34:47.403: msg tag = 0x0000
*Aug 22 08:34:47.403: msg ID = 0xF201

Configuring the NextPort Dual-Filter G.168 Echo Canceller

The NextPort dual-filter G.168 echo canceller is enabled by default on NextPort platforms in Cisco IOS
Release 12.3(11)T and later releases. However, you can adjust the echo canceller tail coverage time at the
voice-port interface by completing the following tasks:

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port slot / port :D
4. echo-cancel coverage {24 | 32 | 48 | 64 | 80 | 96 | 112 | 128}
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice-port slot / port :D Enters voice-port configuration mode on the selected slot and
port.
Example:
Router(config)# voice-port 3/0:D

Step 4 echo-cancel coverage {24 | 32 | 48 | 64 | 80 | 96 | Adjusts the size of the echo canceller (EC) and selects the
112 | 128} extended EC when the Cisco default EC is present.
• Starting with Cisco IOS Release 12.4(20)T, the default
Example: coverage time and maximum possible coverage is 128 ms
Router(config-voiceport)# echo-cancel for both 8.x and 10.2.2 SPEware versions.
coverage 64

Step 5 exit Exits voice-port configuration mode and completes the

configuration.
Example:
Router(config-voiceport)# exit

Configuration Examples for NextPort Services

The following sections contain examples that could be used to optimize a NextPort-based gateway for a given
telephone network:

High ERL in the Network Example

Register v270 is used to set the limit for the minimum expected ERLs that the gateway will encounter. If the
gateway encounters ERLs that are lower than the v270 setting, the echo canceller performance will be
suboptimal.
The default setting for v270 is 6 dB. This setting should work well for usual telephone networks. However,
when the gateway is used on a well-managed telephone network with organized loss plans in place, the ERL
is often greater than 6 dB. In these cases, v270 can be raised. Making this change reduces any clipping of the
near-end signal; however, it will underperform if a low ERL is encountered.
In this example, the network is designed to have an ERL of 12 dB or greater. In this case, the following
voicecap may improve performance. Notice that the value for v270 is entered as decibels multiplied by 10.

Router> enable
Router# configure terminal
Router(config)# voicecap entry qualityERL v270=120
Router(config)# end

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Low ERL in the Network Example

Unlike the scenario described in the previous example, some telephone networks may not always produce
sufficient ERLs to meet the default setting of 6 dB. The best way to solve this problem would be to institute
better loss plans on the telephone network. However, because this is not always possible, a voicecap can be
used to alleviate the problem.
A low ERL means that the echo canceller must do a much deeper cancellation to remove sufficient echo.
Also, lowering the minimum ERL can increase the incidence of clipping. The best way to improve this situation
is to make up for the telephone network’s lack of loss plan by adding some loss in the gateway. If the lowest
ERL seen is 4 dB, adding 1 dB of output attenuation and -1 dB of input gain will ensure that the echo canceller
never sees more than a 6-dB effective ERL.
Adding this attenuation can be done by entering a voicecap, but using the CLI is the recommended approach
in this example. The following commands set the gains that are needed for this example:

Router> enable
Router# configure terminal

Router(config)# voice-port
3/0:D
Router(config-voiceport)# output attenuation 1
Router(config-voiceport)# input gain -1

Router(config-voiceport)# end

Clipped or Squelched Speech and Low ERL in the Network Example

In this example, a network in which signal level imbalance is already causing clipping to occur with a MinERL
setting of 6 dB and in which ERLs of less than 6 dB are already occurring, a dual approach must be taken.
To stop the clipping, the MinERL setting should be lowered to 12 dB. If 4-dB ERLs are occurring, 4 dB of
attenuation must be added to the input and the output to ensure that there is 12 dB of effective ERL at the
echo canceller (4 dB of real ERL, plus 4 dB of output attenuation, plus 4 dB of input attenuation equals 12
dB of effective ERL). To create these settings, the following commands are used:

Router> enable
Router# configure terminal
Router(config)# voicecap entry qualityERL v270=120
Router(config)# voice-port 3/0:D
Router(config-voiceport)# voicecap configure qualityERL
Router(config-voiceport)# output attenuation 4
Router(config-voiceport)# input gain -4
Router(config-voiceport)# end

Dynamic Attenuation Example

It is possible that worst-case settings are only required when the primary telephone circuits are all used and
an alternate carrier with a poor loss plan must be used instead. For these cases, the dynamic attenuation feature
removes the attenuation when the ERL is sufficient. In the following example, 4 dB of input and output
attenuation is to be removed when it is not necessary to ensure the minimum ERL setting. To do this, the
dynamic attenuation feature (using v289) is enabled. The required ERL must be set before attenuation is
removed (using v290), and minimum attenuation levels for the input and output (using v291 and v292) must
also be set. In this example, attenuation is set to 15 dB and then removed.

Router> enable

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Router# configure terminal

Router(config)# voicecap entry dynatten v270=120 v289=1 v290=15 v291=0 v292=0
Router(config)# voice-port 3/0:D
Router(config-voiceport)# voicecap configure dynatten
Router(config-voiceport)# output attenuation 4
Router(config-voiceport)# input gain -4
Router(config-voiceport)# end

Echo Canceller Tail Coverage Example

The following example adjusts echo canceller tail coverage to 64 ms on the Cisco AS5400:

Router(config) voice-port 1
/0:0
Router(config-voiceport)# echo-cancel coverage 64

Enabling NextPort Echo Canceller Control for G.711 Encoded VoIP Packets
This section describes how to enable NextPort echo canceller control on the Cisco AS5350, AS5400,
AS5400HPX, and AS5850 universal gateways when these gateways detect 2100 Hz tones, received in G.711
encoded VoIP packets. You can enable NextPort voicecaps to control the echo canceller from either the PSTN
or IP side of the network.

Note NextPort control over the echo canceller is possible only in G.711 codec modes. Cisco recommends that
you do not enable NextPort control over the echo canceller in conjunction with modem pass-through.

IP tone detection and NextPort control over the echo canceller is enabled using the command-line interface.
Use the following commands to enable NextPort control over the echo canceller by creating a voicecap entry
and applying it to the voice port.

SUMMARY STEPS

1. enable
2. configure terminal
3. voicecap entry name string
4. voice-port slot / port
5. voicecap configure name
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable

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Command or Action Purpose

Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

Step 3 voicecap entry name string Creates a voicecap entry.

• The name argument is a word that uniquely identifies this voicecap.
Example:
• The string argument is a series of voicecap register entries, similar to a
Router(config)# voicecap entry
npecho_ctrl v2=512 v51=32769 modemcap. Each entry is of the form vINDEX=VALUE, where INDEX
refers to a specific V register, and VALUE designates the value to which
the V register should be set.
• Example settings:
• The v2=512setting enables the 250-ms silence detection. This
setting is optional. When this setting is used in conjunction with
the v51 = 32769 setting, NextPort restores the echo canceller to
its original state after it detects the 250-ms silence.
• The v51=32769 setting enables IP side tone detection/notification
and allows NextPort to disable the NLP or the echo canceller upon
reception of 2100 Hz answer tones from the IP side. This setting
is required in Cisco IOS Release 12.3T.

Step 4 voice-port slot / port Enters voice-port configuration mode on the selected slot and port.

Example:
Router(config)# voice-port 3/0

Step 5 voicecap configure name Applies a voicecap entry to the voice port.
• The name argument designates which of the newly created voicecaps to
Example: use on this voice port. This character value must be identical to the value
Router(config-voiceport)# entered when you created the voicecap entry.
voicecap configure npecho_ctrl
Note To configure multiple voice ports, repeat Step 4 and Step 5 for each
voice port.
Step 6 exit Exits voice-port configuration mode.

Example:
Router(config-voiceport)# exit

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Troubleshooting NextPort Echo Canceller Control for G.711 Encoded VoIP Packets
You can display the EST trace messages that show the tone detections and the resultant echo operations if
you enter the debug trace module f080 0010 s / d / m command. NextPort enables and disables the NLP and
the echo canceller based on reception of 2100 Hz answer tones from the IP side or PSTN side and generates
EST trace messages for each tone detected and its echo operation. NextPort also detects the 250 ms of silence
and generates EST trace messages to indicate such detection and to indicate that the echo state has been
restored.

Router# debug trace module f080 0010 s/d/m

Use this command to display the EST trace messages. In the command, s/d/m is defined as follows:
• s = slot
• d = dfc
• m = module number

When the default configuration values for Index 51 and Index 52 are used, IP tone detection and notification
are disabled, and all existing features continue to function normally.
The following example shows EST trace messages collected from the console:

Router#
*Apr 26 21:40:51.735: 00:00:14: Port Trace Event:
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735: Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Format: ASCII
*Apr 26 21:40:51.735: Data Len : 56
*Apr 26 21:40:51.735: Data : Session 0x0144 Received Early ANS tone 0x01 from
IP side
*Apr 26 21:40:51.735: 00:00:14: Port Trace Event:
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735:
Router# Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Format: ASCII
*Apr 26 21:40:51.735: Data Len : 63
*Apr 26 21:40:51.735: Data : Session 0x0144 Received Tone Off ntf for code 0x01
from IP side
*Apr 26 21:40:51.735: 00:00:14: Port Trace Event:
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735: Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Format: ASCII
Router#*Apr 26 21:40:51.735: Data Len : 45
*Apr 26 21:40:51.735: Data : Session 0x0144 Received ANS tone 0x03 from IP
*Apr 26 21:40:51.735: 00:00:14: Port Trace Event:
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735: Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Format: ASCII
*Apr 26 21:40:51.735: Data Len : 47
*Apr 26 21:40:51.735: Data : Session 0x0144 Non-linear Processor Is Disabled
*Apr
Router# 26 21:40:51.735: 00:00:14: Port Trace Event:
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735: Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Format: ASCII
*Apr 26 21:40:51.735: Data Len : 63
*Apr 26 21:40:51.735: Data : Session 0x0144 Received Tone Off ntf for code 0x03
from IP side
*Apr 26 21:40:51.735: 00:00:14: Port Trace Event:

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*Apr 26 21:40:51.735: Port : 3/00

*Apr 26 21:40:51.735:
Router# Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Format: ASCII
*Apr 26 21:40:51.735: Data Len : 47
*Apr 26 21:40:51.735: Data : Session 0x0144 Received ANSam tone 0x07 from IP
*Apr 26 21:40:51.735: 00:00:13: Port Trace Event:
*Apr 26 21:40:51.735: Port : 3/00
*Apr 26 21:40:51.735: Address : 0x3000000
*Apr 26 21:40:51.735: Trace Event: 0x2
*Apr 26 21:40:51.735: Data Format: ASCII
*Apr 26 21:40:5
Router#1.735: Data Len : 63
*Apr 26 21:40:51.735: Data : Session 0x0144 Received Tone Off ntf for code 0x07
from IP side
*Apr 26 21:40:51.739: 00:00:13: Port Trace Event:
*Apr 26 21:40:51.739: Port : 3/00
*Apr 26 21:40:51.739: Address : 0x3000000
*Apr 26 21:40:51.739: Trace Event: 0x2
*Apr 26 21:40:51.739: Data Format: ASCII
*Apr 26 21:40:51.739: Data Len : 48
*Apr 26 21:40:51.739: Data : Session 0x0144 Received /ANSam tone 0x0f from IP
Router#*Apr 26 21:40:51.739: 00:00:13: Port Trace Event:
*Apr 26 21:40:51.739: Port : 3/00
*Apr 26 21:40:51.739: Address : 0x3000000
*Apr 26 21:40:51.739: Trace Event: 0x2
*Apr 26 21:40:51.739: Data Format: ASCII
*Apr 26 21:40:51.739: Data Len : 31
*Apr 26 21:40:51.739: Data : Session 0x0144 ECAN Is Disabled
*Apr 26 21:40:51.739: 00:00:04: Port Trace Event:
*Apr 26 21:40:51.739: Port : 3/00
*Apr 26 21:40:51.739: Address : 0x3000000
Router#*Apr 26 21:40:51.739: Trace Event: 0x2
*Apr 26 21:40:51.739: Data Format: ASCII
*Apr 26 21:40:51.739: Data Len : 63
*Apr 26 21:40:51.739: Data : Session 0x0144 Received Tone Off ntf for code 0x0f
from IP side
*Apr 26 21:46:36.431: 00:00:08: Port Trace Event:
*Apr 26 21:46:36.431: Port : 3/00
*Apr 26 21:46:36.431: Address : 0x3000000
*Apr 26 21:46:36.431: Trace Event: 0x2
*Apr 26 21:46:36.431: Data Format: ASCII
*Apr 26 21:46:36.431: Data Len : 43
*Apr 26 21:46:36.431: Data : Session 0x0144 detected 250 msec of silence
*Apr 26 21:46:36.431: 00:00:08: Port Trace Event:
*Apr 26 21:46:36.431: Port : 3/00
*Apr 26 21:46:36.431: Address : 0x3000000
*Apr 26 21:46:36.435: Trace Event: 0x2
*Apr 26 21:46:36.435: Data Format: ASCII
*Apr 26 21:46:36.435: Data Len : 41
*Apr 26 21:46:36.435: Data : Session 0x0144 Ecan State 0x0007 Restored

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 CHAPTER 12
Verifying Analog and Digital Voice-Port
Configurations
This chapter describes some basic procedures and specific CLI commands you can use to verify the set up
and configuration of the analog and digital voice ports on the routers in your voice network.

• Finding Feature Information, page 175

• Information About Verifying Voice-Port Configurations, page 175
• How to Verify Voice-Port Configurations, page 175

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest caveats and
feature information, see Bug Search Tool and the release notes for your platform and software release. To
find information about the features documented in this module, and to see a list of the releases in which each
feature is supported, see the feature information table at the end of this module.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support.
To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.

Information About Verifying Voice-Port Configurations

After configuring the voice ports on your router, there are two simple verifications you can use to check the
operation of the telephony handset. There are also eight CLI commands you can use to verify proper operation
of the configuration.

How to Verify Voice-Port Configurations

To verify the operability of the analog and digital voice-port configurations on your voice network, complete
the following tasks.

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SUMMARY STEPS

1. Check for a dial tone.

2. Check for Dual-Tone Multifrequency (DTMF) detection.
3. show voice port summary
4. show voice port
5. show running-config
6. show controller
7. show voice dsp
8. show voice call summary
9. show call active voice
10. show call history voice

DETAILED STEPS

Step 1 Check for a dial tone.

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

Step 2 Check for Dual-Tone Multifrequency (DTMF) detection.

If you have dial tone, check for DTMF detection. If the dial tone stops when you dial a digit, then the voice port is
probably configured properly.

Step 3 show voice port summary

Use this command to identify the port numbers of voice interfaces installed in your router. For examples of the output,
see the "Examples" section.

Step 4 show voice port

Use this command to verify voice-port parameter settings. Refer to the table below for the appropriate syntax for your
platform. For sample output, see the show voice port Command Examples, on page 179.

Table 12: Show Voice Port Command Syntax

Platform Voice Port Type Command Syntax

Cisco 1750 Analog show voice port [slot / port |
summary]

Cisco 2600 series Cisco 3600 series Analog show voice port [slot / port |
Cisco 3700 series summary]

Digital show voice port [slot / port :

ds0-group-number | summary]

Cisco MC3810 Analog show voice port [slot / port |

summary]

Digital show voice port [slot :

ds0-group-number | summary]

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Platform Voice Port Type Command Syntax

Cisco AS5300 Digital show voice port [controller
:{ds0-group-number | D}] [summary]

Cisco AS5350 Cisco AS5400 Cisco Digital show voice port [slot/controller
AS5850 :{ds0-group-number | D}] [summary]

Cisco AS5800 Digital show voice port {shelf / slot/port :

ds0-group-number}

Cisco 7200 series Digital show voice port {slot / port-adapter

: ds0-group-number}

Cisco 7500 series Digital show voice port {slot / port-adapter

/ slot : ds0-group-number}

Step 5 show running-config

Use this command to verify the codec complexity setting for digital T1/E1 connections. If medium complexity is specified
for the voice card, the codec complexity command is not displayed. If high complexity is specified, the codec complexity
high command is displayed. The following example shows output when high complexity is specified:

Example:
Router# show running-config
.
.
.
hostname router-alpha

voice-card 0
codec complexity high
.
.
.

Step 6 show controller

Use this command to verify that the digital T1/E1 controller is up and that no alarms have been reported, and to display
information about clock sources and other controller settings. For output examples, see the show controller Command
Examples, on page 182.

Example:
Router# show controller
{t1
| e1
}
controller-number

Step 7 show voice dsp

Use this command to display voice-channel configuration information for all DSP channels. For output examples, see
the show voice dsp Command Examples, on page 183.

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Example:
Router# show voice dsp

Step 8 show voice call summary

Use this command to verify the call status for all voice ports. For output examples, see the show voice call summary
Command Examples, on page 184.

Example:
Router# show voice call summary

Step 9 show call active voice

Use this command to display the contents of the active call table, which shows all of the calls currently connected through
the router or concentrator. For output examples, see the show call active voice Command Example, on page 184.

Example:
Router# show call active voice

Step 10 show call history voice

Use this command to display the contents of the call history table. To limit the display to the most recent calls connected
through this router, use the last keyword and define the number of calls to display with the numberargument. To limit
the display to a shortened version of the call history table, use the brief keyword. For output examples, see the show call
history voice Command Example, on page 185.

Example:
Router# show call history voice
[last
| number
| brief

Examples
This section contains output examples for the following commands on different platforms and for different
configurations:

show voice port summary Command Examples

Cisco 3640 Router Analog Voice Port

The following output is from a Cisco 3640 router:

Router# show voice port summary

IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
====== == ========== ===== ==== ======== ======== ==

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2/0/0 -- e&m-wnk up dorm idle idle y

2/0/1 -- e&m-wnk up dorm idle idle y
2/1/0 -- fxs-ls up dorm on-hook idle y
2/1/1 -- fxs-ls up dorm on-hook idle y

Cisco MC3810 Digital Voice Port

The following output is from a Cisco MC3810:

Router# show voice port summary

IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
====== == ========== ===== ==== ======== ======== ==
0:17 18 fxo-ls down down idle on-hook y
0:18 19 fxo-ls up dorm idle on-hook y
0:19 20 fxo-ls up dorm idle on-hook y
0:20 21 fxo-ls up dorm idle on-hook y
0:21 22 fxo-ls up dorm idle on-hook y
0:22 23 fxo-ls up dorm idle on-hook y
0:23 24 e&m-imd up dorm idle idle y
1/1 -- fxs-ls up dorm on-hook idle y
1/2 -- fxs-ls up dorm on-hook idle y
1/3 -- e&m-imd up dorm idle idle y
1/4 -- e&m-imd up dorm idle idle y
1/5 -- fxo-ls up dorm idle on-hook y
1/6 -- fxo-ls up dorm idle on-hook y

show voice port Command Examples

Cisco 3600 Series Router Analog EandM Voice Port

The following output is from a Cisco 3600 series router analog E&M voice port:

Router# show voice port

1/0
E&M Slot is 1, Sub-unit is 0, Port is 0
Type of VoicePort is E&M
Operation State is unknown
Administrative State is unknown
The Interface Down Failure Cause is 0
Alias is NULL
Noise Regeneration is disabled
Non Linear Processing is disabled
Music On Hold Threshold is Set to 0 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is disabled
Echo Cancel Coverage is set to 16ms
Connection Mode is Normal
Connection Number is
Initial Time Out is set to 0 s
Interdigit Time Out is set to 0 s
Analog Info Follows:
Region Tone is set for northamerica
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0

Voice card specific Info Follows:

Signal Type is wink-start
Operation Type is 2-wire
Impedance is set to 600r Ohm
E&M Type is unknown
Dial Type is dtmf
In Seizure is inactive
Out Seizure is inactive

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Digit Duration Timing is set to 0 ms

InterDigit Duration Timing is set to 0 ms
Pulse Rate Timing is set to 0 pulses/second
InterDigit Pulse Duration Timing is set to 0 ms
Clear Wait Duration Timing is set to 0 ms
Wink Wait Duration Timing is set to 0 ms
Wink Duration Timing is set to 0 ms
Delay Start Timing is set to 0 ms
Delay Duration Timing is set to 0 ms

Cisco 3600 Series Router Analog FXS Voice Port

The following output is from a Cisco 3600 series router analog Foreign Exchange Service (FXS) voice port:

Router# show voice port

1
/2
Voice port 1/2 Slot is 1, Port is 2
Type of VoicePort is FXS
Operation State is UP
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Coder Type is g729ar8
Companding Type is u-law
Voice Activity Detection is disabled
Ringing Time Out is 180 s
Wait Release Time Out is 30 s
Nominal Playout Delay is 80 milliseconds
Maximum Playout Delay is 160 milliseconds
Analog Info Follows:
Region Tone is set for northamerica
Currently processing Voice
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Analog interface A-D gain offset = -3 dB
Analog interface D-A gain offset = -3 dB
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 20 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is active
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Ring Cadence are [20 40] * 100 msec
InterDigit Pulse Duration Timing is set to 500 ms

Cisco 3600 Series Router Digital EandM Voice Port

The following output is from a Cisco 3600 series router digital E&M voice port:

Router# show voice port 1

/0:1

receEive and transMit Slot is 1, Sub-unit is 0, Port is 1

Type of VoicePort is E&M

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Operation State is DORMANT

Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Region Tone is set for US

Cisco AS5300 T1 CAS Voice Port

The following output is from a Cisco AS5300 T1 channel-associated signaling (CAS) voice port:

Router# show voice port

DS0 Group 1:0 - 1:0

Type of VoicePort is CAS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Playout-delay Mode is set to default
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 200 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Ringing Time Out is set to 180 s
Companding Type is u-law
Region Tone is set for US
Wait Release Time Out is 30 s
Station name None, Station number None

Voice card specific Info Follows:

DS0 channel specific status info:

IN OUT
PORT CH SIG-TYPE OPER STATUS STATUS TIP RING

Cisco 7200 Series Router Digital EandM Voice Port

The following output is from a Cisco 7200 series router digital E&M voice port:

Router# show voice port 1

/0:1
receEive and transMit Slot is 1, Sub-unit is 0, Port is 1 << voice-port 1/0:1
Type of VoicePort is E&M
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled

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Non Linear Processing is enabled

Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Region Tone is set for US

show controller Command Examples

Cisco 3600 Series Router T1 Controller

The following output is from a Cisco 3600 series router with a T1 controller:

Router# show controller T1 1/1/0

T1 1/0/0 is up.
Applique type is Channelized T1
Cablelength is long gain36 0db
No alarms detected.
alarm-trigger is not set
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
Data in current interval (180 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs

Cisco MC3810 E1 Controller

The following output is from a Cisco MC3810 with an E1 controller:

Router# show controller e1 1/0

E1 1/0 is up.
Applique type is Channelized E1
Cablelength is short 133
Description: E1 WIC card Alpha
No alarms detected.
Framing is CRC4, Line Code is HDB3, Clock Source is Line Primary.
Data in current interval (1 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs

Cisco AS5800 T1 Controller

The following output is from a Cisco AS5800 with a T1 controller:

Router# show controller t1 2

T1 2 is up.
No alarms detected.
Version info of slot 0: HW: 2, Firmware: 16, PLD Rev: 0

Manufacture Cookie Info:

EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x42,
Board Hardware Version 1.0, Item Number 73-2217-4,
Board Revision A0, Serial Number 06467665,
PLD/ISP Version 0.0, Manufacture Date 14-Nov-1997.

Framing is ESF, Line Code is B8ZS, Clock Source is Internal.

Data in current interval (269 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations

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0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins

0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs

show voice dsp Command Examples

Digital Voice Port on a Cisco 3640

The following output is from a Cisco 3640 router when a digital voice port is configured:

Router# show voice dsp

TYPE DSP CH CODEC VERS STATE STATE RST AI PORT TS ABORT TX/RX-PAK-CNT
==== === == ======== ==== ===== ======= === == ======= == ===== ===============
C549 010 00 g729r8 3.3 busy idle 0 0 1/015 1 0 67400/85384
01 g729r8 .8 busy idle 0 0 1/015 7 0 67566/83623
02 g729r8 busy idle 0 0 1/015 13 0 65675/81851
03 g729r8 busy idle 0 0 1/015 20 0 65530/83610
C549 011 00 g729r8 3.3 busy idle 0 0 1/015 2 0 66820/84799
01 g729r8 .8 busy idle 0 0 1/015 8 0 59028/66946
02 g729r8 busy idle 0 0 1/015 14 0 65591/81084
03 g729r8 busy idle 0 0 1/015 21 0 66336/82739
C549 012 00 g729r8 3.3 busy idle 0 0 1/015 3 0 59036/65245
01 g729r8 .8 busy idle 0 0 1/015 9 0 65826/81950
02 g729r8 busy idle 0 0 1/015 15 0 65606/80733
03 g729r8 busy idle 0 0 1/015 22 0 65577/83532
C549 013 00 g729r8 3.3 busy idle 0 0 1/015 4 0 67655/82974
01 g729r8 .8 busy idle 0 0 1/015 10 0 65647/82088
02 g729r8 busy idle 0 0 1/015 17 0 66366/80894
03 g729r8 busy idle 0 0 1/015 23 0 66339/82628
C549 014 00 g729r8 3.3 busy idle 0 0 1/015 5 0 68439/84677
01 g729r8 .8 busy idle 0 0 1/015 11 0 65664/81737
02 g729r8 busy idle 0 0 1/015 18 0 65607/81820
03 g729r8 busy idle 0 0 1/015 24 0 65589/83889
C549 015 00 g729r8 3.3 busy idle 0 0 1/015 6 0 66889/83331
01 g729r8 .8 busy idle 0 0 1/015 12 0 65690/81700
02 g729r8 busy idle 0 0 1/015 19 0 66422/82099
03 g729r8 busy idle 0 0 1/015 25 0 65566/83852
Router# show voice dsp
TYPE DSP CH CODEC VERS STATE STATE RST AI PORT TS ABORT TX/RX-PAK-CNT
==== === == ======== ==== ===== ======= === == ======= == ===== ===============
C549 007 00 {medium} 3.3 IDLE idle 0 0 1/0:1 4 0 0/0
.13
C549 008 00 {medium} 3.3 IDLE idle 0 0 1/0:1 5 0 0/0
.13
C549 009 00 {medium} 3.3 IDLE idle 0 0 1/0:1 6 0 0/0
.13
C549 010 00 {medium} 3.3 IDLE idle 0 0 1/0:1 7 0 0/0
.13
C549 011 00 {medium} 3.3 IDLE idle 0 0 1/0:1 8 0 0/0
.13
C549 012 00 {medium} 3.3 IDLE idle 0 0 1/0:1 9 0 0/0
.13
C542 001 01 g711ulaw 3.3 IDLE idle 0 0 2/0/0 0 512/519
.13
C542 002 01 g711ulaw 3.3 IDLE idle 0 0 2/0/1 0 505/502
.13
C542 003 01 g711alaw 3.3 IDLE idle 0 0 2/1/0 0 28756/28966
.13
C542 004 01 g711ulaw 3.3 IDLE idle 0 0 2/1/1 0 834/838
.13

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Examples

show voice call summary Command Examples

Cisco MC3810 Analog Voice Port

The following output is from a Cisco MC3810:

Router# show voice call summary

PORT CODEC VAD VTSP STATE VPM STATE
========= ======== === ===================== ========================
1/1 g729r8 y S_CONNECT FXSLS_CONNECT
1/2 - - - FXSLS_ONHOOK
1/3 - - - EM_ONHOOK
1/4 - - - EM_ONHOOK
1/5 - - - FXOLS_ONHOOK
1/6 - - - FXOLS_ONHOOK

Cisco 3600 Series Router Digital Voice Port

The following output is from a Cisco 3600 series router:

Router# show voice call summary

PORT CODEC VAD VTSP STATE VPM STATE
========= ======== === ===================== ========================
1/015.1 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.2 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.3 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.4 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.5 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.6 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.7 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.8 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.9 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.10 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.11 g729r8 y S_CONNECT S_TSP_CONNECT
1/015.12 g729r8 y S_CONNECT S_TSP_CONNECT

show call active voice Command Example

Cisco 7200 Series Router

The following output is from a Cisco 7200 series router:

Router# show call active voice

GENERIC:
SetupTime=94523746 ms
Index=448
PeerAddress=##73072
PeerSubAddress=
PeerId=70000
PeerIfIndex=37
LogicalIfIndex=0
ConnectTime=94524043
DisconectTime=94546241
CallOrigin=1
ChargedUnits=0
InfoType=2
TransmitPackets=6251
TransmitBytes=125020
ReceivePackets=3300
ReceiveBytes=66000
VOIP:
ConnectionId[0x142E62FB 0x5C6705AF 0x0 0x385722B0]
RemoteIPAddress=172.16.235.18

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RemoteUDPPort=16580
RoundTripDelay=29 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
SessionProtocol=cisco
SessionTarget=ipv4:172.16.235.18
OnTimeRvPlayout=63690
GapFillWithSilence=0 ms
GapFillWithPrediction=180 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=70 ms
LoWaterPlayoutDelay=30 ms
ReceiveDelay=40 ms
LostPackets=0 ms
EarlyPackets=1 ms
LatePackets=18 ms
VAD = disabled
CoderTypeRate=g729r8
CodecBytes=20
cvVoIPCallHistoryIcpif=0
SignalingType=cas

show call history voice Command Example

Cisco 7200 Series Router

The following output is from a Cisco 7200 series router:

Router# show call history voice

GENERIC:
SetupTime=94893250 ms
Index=450
PeerAddress=##52258
PeerSubAddress=
PeerId=50000
PeerIfIndex=35
LogicalIfIndex=0
DisconnectCause=10
DisconnectText=normal call clearing.
ConnectTime=94893780
DisconectTime=95015500
CallOrigin=1
ChargedUnits=0
InfoType=2
TransmitPackets=32258
TransmitBytes=645160
ReceivePackets=20061
ReceiveBytes=401220
VOIP:
ConnectionId[0x142E62FB 0x5C6705B3 0x0 0x388F851C]
RemoteIPAddress=172.16.235.18
RemoteUDPPort=16552
RoundTripDelay=23 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
SessionProtocol=cisco
SessionTarget=ipv4:172.16.235.18
OnTimeRvPlayout=398000
GapFillWithSilence=0 ms
GapFillWithPrediction=1440 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=97 ms
LoWaterPlayoutDelay=30 ms
ReceiveDelay=49 ms
LostPackets=1 ms
EarlyPackets=1 ms
LatePackets=132 ms

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VAD = disabled
CoderTypeRate=g729r8
CodecBytes=20
cvVoIPCallHistoryIcpif=0

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 CHAPTER 13
Troubleshooting Analog and Digital Voice Port
Configurations
This chapter provides information to assist you in analyzing and troubleshooting voice port problems.

• Troubleshooting Chart, page 187

Troubleshooting Chart
The table below lists some problems that you might encounter after configuring voice ports. It also provides
some suggested remedies.

Table 13: Troubleshooting Voice Port Configurations

Problem Suggested Action

No connectivity Ping the associated IP address to confirm
connectivity. If you cannot successfully ping your
destination, refer to the Cisco IOS IP Configuration
Guide.

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Troubleshooting Chart

Problem Suggested Action

No connectivity Enter the show controller t1 or show controller e1
command with the controller number for the voice
port you are troubleshooting. This will tell you:
• If the controller is up. If it is not, use the no
shutdown command to make it active.
• Whether alarms have been reported.
• What parameter values have been set for the
controller (framing, clock source, line code,
cable length). If these values do not match those
of the telephony connection you are making,
reconfigure the controller.

See the "show controller Command: Examples"

section in the "Verifying Analog and Digital
Voice-Port Configurations" chapter for output.

No connectivity Enter the show voice port command with the voice
port number that you are troubleshooting, which will
tell you:
• If the voice port is up. If it is not, use the no
shutdown command to make it active.
• What parameter values have been set for the
voice port, including default values (these do
not appear in the output for the show
running-config command). If these values do
not match those of the telephony connection
you are making, reconfigure the voice port.

See the "show voice port Command: Examples"

section in the "Verifying Analog and Digital
Voice-Port Configurations" chapter for sample output.

Telephony device buzzes or does not ring Use the show voice port command to confirm that
the ring frequency command is configured correctly.
It must match the connected telephony equipment
and may be country-dependent.

Distorted speech Use the show voice port command to confirm the
cptone keyword setting (also called region tone) is
US.
Setting a wrong cptone could result in faulty voice
reproduction during analog-to-digital or
digital-to-analog conversions.

Music on hold is not heard Reduce the configured level for the music-threshold
command.

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Troubleshooting Chart

Problem Suggested Action

Background noise is not heard Enable the comfort-noise command.

Long pauses occur in conversation; like speaking on Overall delay is probably excessive; the standard for
a walkie-talkie adequate voice quality is 150 milliseconds (ms)
one-way transit delay. Measure delay by using ping
tests at various times of the day with different network
traffic loads. If delay must be reduced, areas to
examine include propagation delay of signals between
the sending and receiving endpoints, voice encoding
delay, and the voice packetization time for various
VoIP codecs.

Jerky or choppy speech Variable delay, or jitter, is being introduced by

congestion in the packet network. Two possible
remedies are to:
• Reduce the amount of congestion in your packet
network. Pings between VoIP endpoints will
give an idea of the round-trip delay of a link,
which should never exceed 300 ms. Network
queuing and dropped packets should also be
examined.
• Increase the size of the jitter buffer with the
playout-delay command. (Refer to the Cisco
IOS Voice Troubleshooting and Monitoring
Guide.)

Clipped or fuzzy speech

• Reduce input gain. (Refer to the Cisco IOS
Voice Troubleshooting and Monitoring Guide.)
• Change the voice activity detection (VAD)
level. Sometimes VAD cuts the sound too early
and the speaker's voice is clipped. You can also
change the time that VAD waits for silence.

Clipped speech Reduce the input level at the listener’s router. (Refer
to the Cisco IOS Voice Troubleshooting and
Monitoring Guide.)

Volume too low or missed Dual-Tone Multifrequency Increase speaker’s output level or listener’s input level.
(DTMF) (Refer to the Cisco IOS Voice Troubleshooting and
Monitoring Guide.)

Echo interval is greater than 25 ms (sounds like a Configure the echo-cancel enable command and
separate voice) increase the value for the echo-cancel coverage
keyword. (See the "Configuring Echo Cancellation"
section.)

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Troubleshooting Chart

Problem Suggested Action

Too much echo Reduce the output level at the speaker’s voice port.
(Refer to the Cisco IOS Voice Troubleshooting and
Monitoring Guide.)

Note For information on test commands that force voice ports into specific states for testing refer to the Cisco
IOS Voice Troubleshooting and Monitoring Guide.

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 INDEX

A D
auto-cut-through command 60 define command 61
dial-type command 13
disconnect-ack command 65
ds0-group command 19, 45
B DSC (dial shelf controller) 25
battery reversal command 65 digital voice port clock source 25
dspint dspfarm command 36

C
E
cadence-list command 67
cadence-min-on-time command 67 E1 23
cadence-variation command 67 digital packet voice trunk network module 23
calling-number outbound command 63
calling-number outbound sequence command 63
card type command 45 F
Cisco 7200 series routers 24
digital voice port adapters 24 framing command 45
Cisco 7600 series routers 26 freq-max-delay command 67
Communication Media Modules 26 freq-max-deviation command 67
Cisco AS5300 access servers 24, 36, 63 freq-max-power command 67
codec support 36 freq-min-power command 67
FGD-EANA signaling 63 freq-pair command 67
voice/fax feature card 24 freq-power-twist command 67
Cisco MC3810 concentrators 27 FXO (foreign exchange office) 65, 67
digital voice interface card 27 Disconnect Supervision feature 65
clock source command 45 supervisory disconnect tone 67
codec complexity command 17, 28
comfort-noise command 77
compand-type command 50
condition command 61
I
connections 1, 55, 60 ignore command 61
PBX to WAN 1
PBX without M-lead response 55, 60
voice port to PSTN 1
voice port to WAN 1 L
cptone command 13, 50 linecode command 45

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 Index

M timeouts interdigit command 71

timeouts ringing command 71
music-threshold command 77 timeouts wait-release command 71
timing clear-wait command 72
timing delay-duration command 72
O timing delay-start command 72
timing delay-with-integrity command 72
operation command 13 timing dial-pulse min-delay command 72
timing dialout-delay command 72
timing digit command 72
timing guard-out command 72
P timing hookflash-out command 72
PBX switches, disconnecting 65 timing interdigit command 72
timing percentbreak command 72
timing pulse command 72
timing pulse-digit command 72
R timing pulse-interdigit command 72
ring cadence command 13, 50 timing wink-duration command 72
ring frequency command 13, 50 timing wink-wait command 72
ring number command 13, 50 type command 13, 50

S V

show call active voice command 175, 184 voice class dualtone command 67
(examples) 184 voice ports 1, 9, 12, 13, 19, 20, 28, 33, 38, 45, 50, 55, 61, 71, 72, 77, 175,
187
show call history voice command 175, 185
(examples) 185 analog 9, 12, 13, 28, 33, 55, 175, 187
show controller command 175, 182 codec complexity, configuring 13, 28, 33
(examples) 182 configuring 9, 12
show voice call summary command 175, 184 fine tuning 55
(examples) 184 troubleshooting 187
show voice dsp command 17, 28, 175, 183 verifying configuration 175
show voice port command 179 analog and digital transmission support (table) 1
(examples) 179 basic parameters 13, 28
show voice port summary command 20, 175, 178 configuring 13
(examples) 178 configuration mode 50
signal command 13 configuration overview 1
signaling techniques 4 digital 19, 20, 55, 61, 77
ground start 4 bit modifications 61
loop-start 4 configuring 19, 77
supervisory disconnect anytone command 67 fine tuning 55
supervisory disconnect dualtone command 67 requirements 20
DS0 groups on digital T1/E1 45
E1 configuration 38
T1 configuration 38
T timeouts, configuring 71
timing parameters 72
T1 38 voice activity detection 77
voice port configuration 38 voice-card command 17, 28
timeouts call-disconnect command 71 voice-port command 13, 50
timeouts initial command 71

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