Table of Contents

Feature Overview

Benefits
Restrictions
Related Documents

Supported Platforms
Supported Standards, MIBs, and RFCs
Prerequisites
Configuration Tasks

Configuring Voice Card and E1 Controller Settings

E1 Timing, Signaling, Framing, and Line Encoding

Verifying Voice Card and Controller Settings
Configuring Serial Interfaces
Verifying Serial Interface Configuration
Configuring Voice Ports
Verifying Voice Ports
Configuring Voice Dial Peers
Verifying Voice Dial Peers

Monitoring and Maintaining E1 Digital Packet Voice Configuration

Show Commands
Debug Commands
Reference Information

Configuration Example
Command Reference
pri-group
Glossary

Digital E1 Packet Voice Trunk Network Module Interfaces

This document describes how to configure digital E1 packet voice trunk network module interfaces on Cisco 2600 and 3600 series routers and includes the following sections:

• Feature Overview

• Supported Platforms

• Supported Standards, MIBs, and RFCs

• Prerequisites

• Configuration Tasks

• Monitoring and Maintaining E1 Digital Packet Voice Configuration

• Configuration Example

• Command Reference

• Glossary

Feature Overview

Digital E1 packet voice trunk network modules for Cisco 2600 and 3600 series routers allow enterprises or service providers, using the equipped routers as customer premises equipment, to deploy digital voice and fax relay. These modules receive constant bit-rate telephony information over E1 interfaces and can convert that information to a compressed format, so that it can be transmitted as Voice over IP (VoIP), Voice over Frame Relay (VoFR), and Voice over ATM (VoATM).

Benefits

Digital E1 packet voice trunk network modules allow Cisco 2600 and 3600 series routers to provide E1 connectivity to private branch exchanges (PBXs) or to a central office (CO). With digital E1 connectivity, Cisco 2600 and 3600 series routers can provide greater voice density for enterprise and service provider VoIP networks. A digital E1 packet voice trunk network module is a complete solution, made up of a network module with installed packet voice data modules (PVDMs), and one E1 multiflex trunk voice/WAN interface card with either one or two E1 ports.

Restrictions

The following restrictions apply to digital E1 packet voice trunk network module configuration:

• Group 4 fax is not supported.

• The high-density voice network module has one slot for a voice/WAN interface card (VWIC); VWICs supply one or two ports. Only the dual-mode (voice/WAN) multiflex trunk cards are supported in the digital E1 packet voice trunk network module, not older VICs. For more information, see the "Prerequisites" section.

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

• Common-channel signaling (CCS) and Primary Rate Interface (PRI) are not supported.

• R2 signaling is not supported.

• Voice over ATM---including AAL5 encapsulation, circuit emulation service (CES), and AAL2---is not supported for VoATM on the Cisco 2600 series router.

• Digital E1 voice is manageable through Simple Network Management Protocol (SNMP) using Release 2.0 of Cisco Voice Manager.

Related Documents

The following documents can help you understand how to install Cisco 2600 and 3600 series routers:

• Cisco 2600 Series Hardware Installation Guide

• Quick Start Guide Cisco 2600 Series Cabling and Setup

• Software Configuration Guide

• Cisco 3660 Router Cabling and Setup Quick Start Guide

• Cisco 3600 Series Hardware Installation Guide

• Cisco Network Modules Hardware Installation Guide for Cisco 3600 Series and Cisco 2600 Series Routers

The following Cisco IOS Release 12.1 documents are also helpful:

• Cisco IOS Multiservice Applications Configuration Guide

• Cisco IOS Multiservice Applications Command Reference

The following documents can help you troubleshoot ISDN, PRI, and BRI connections:

• Internetwork Troubleshooting Guide

• Cisco IOS Debug Command Reference

For information about supported hardware on a Cisco 2600 or 3600 series router, go to the following URLs:

• http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis2600/index.htm

• http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis3600/index.htm

• For the Voice over IP Quick Start Guides, go to:
  http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis3600/voice/4936vqsg.htm

Supported Platforms

This feature is supported on the following platforms:

• Cisco 2610

• Cisco 2611

• Cisco 2612

• Cisco 2613

• Cisco 2620

• Cisco 2621

• Cisco 3620

• Cisco 3640

• Cisco 3662

• Cisco 3661

Supported Standards, MIBs, and RFCs

MIBs

• CISCO-ENTITY-VENDORTYPE-OID-MIB

• OLD-CISCO-CHASSIS-MIB

• CAS_INTF_MIB

Standards

• G.711 A Law at 64,000 bps

• G.711 u Law at 64,000 bps

• G.723.1 Annex A at 5,300 bps

• G.723.1 Annex A at 6,300 bps

• G.723.1 at 5,300 bps

• G.723.1 at 6,300 bps

• G.726 at 16,000 bps

• G.726 at 24,000 bps

• G.726 at 32,000 bps

• G.728 at 16,000 bps

• G.729 at 8,000 bps

• G.729 Annex A at 8,000 bps

• G.729 Annex B at 8,000 bps

• G.729 Annex B with Annex A at 8,000 bps

RFCs

• RFC 1890

• RFC 1889

Prerequisites

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

• Obtain E1 service from your service provider or PBX.

• Install Cisco IOS Release 12.1(2)T or a later release. The minimum DRAM memory requirements to support digital E1 packet voice trunk network modules are as follows:

  • 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 might 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.

• Install one of the following high-density E1 network modules in the router chassis:

  • Single-Port 30-Channel E1 High-Density Voice Network Module (NM-HDV-1E1-30)

  • Single-Port Enhanced 30-Channel E1 High-Density Voice Network Module (NM-HDV-1E130E)

  • Dual-Port 60-Channel High-Density Voice Network Module (NM-HDV-2E1-60)

---

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

---

• Install at least one PVDM-12 in the high-density digital E1 network module if it is not already equipped. The digital E1 packet voice trunk network module contains five 72-pin SIMM sockets or banks, numbered 0 through 4, for PVDMs. Each socket can be filled with a single 72-pin PVDM. A digital E1 packet voice trunk network module can support the following numbers of channels:

  • When the digital 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, G.729 Annex B, G.723.1, G.728, and fax relay.

  • When the digital 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.

---

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

---

• Install at least one dual-mode voice/WAN interface card (VWIC) for a voice connection if a VWIC was not included with the network module. You can install one VWIC (providing one or two line interfaces) in the digital E1 packet voice trunk network module. Only the one- and two-port E1 multiflex trunk interface cards (VWIC-1MFT-E1, VWIC-2MFT-E1, VWIC-2MFT-E1-DI) are supported.

For Drop-and-Insert capability, you must install a two-port Drop-and-Insert E1 multiflex trunk voice/WAN interface card (VWIC-2MFT-E1-DI). To install a VWIC in a network module, see Cisco WAN Interface Cards Hardware Installation Guide.

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

• Establish a working IP, Frame Relay, or ATM network. For more information about configuring IP, refer to the Cisco IOS IP and IP Routing Configuration Guide for Cisco IOS Release 12.1.

• Complete your company's dial plan.

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

Cisco IOS Multiservice Applications Configuration Guide and Cisco IOS Multiservice Applications Command Reference for Cisco IOS Release 12.1 provide information about setting up voice networks.

Configuration Tasks

Complete the following tasks to configure a digital E1 packet voice trunk network module:

• Set up voice cards and E1 controllers.

• Configure serial and LAN interfaces.

• Set up voice ports.

• Configure voice dial peers.

Configuring Voice Card and E1 Controller Settings

The following steps specify codec settings for voice cards and set up E1 controllers for clocking and other E1 parameters, as well as for DS0 groups that define the channels for compressed voice and TDM groups for Drop-and-Insert capability.

E1 Timing, Signaling, Framing, and Line Encoding

With the introduction of the digital E1 packet voice trunk network modules for the Cisco 2600 and 3600series routers, you must set timing, signaling, framing, and line encoding. Digital E1 packet voice trunk network modules can connect to either a PBX (or similar telephony device) or to a central office (CO) to provide PSTN connectivity.

The differences that set E1 digital configuration apart from analog configuration are as follows:

• Timing---Analog interfaces do not require specific timing configuration. Digital E1 interfaces require not only that you set timing but that you consider the source of the timers.

• Framing---Analog interfaces do not require specific framing configuration. Digital E1 interfaces require that you configure for cyclic redundancy checking 4 (CRC-4) framing. Set the framing format to match that of the PBX or CO that connects to the digital E1 packet voice trunk network module.

• Line Encoding---Analog interfaces do not require specific line encoding configuration. Digital E1 interfaces require that you configure for high density bipolar 3 (HDB3) encoding (similar to alternative mark inversion, or AMI). Set the line encoding to match that of the PBX or CO that connects to the digital E1 packet voice trunk network module.

Timing

This section describes the five basic timing scenarios that can occur when a digital E1 packet voice trunk network module is connected to a PBX, CO, or both. In all of the examples below, the PSTN (or CO) and the PBX are interchangeable for purposes of providing or receiving clocking.

The digital E1 module has an onboard Phase-Lock Loop (PLL) chip that can either provide a clock source to both E1s or receive clocking that can drive the second E1 in the same digital E1 packet voice trunk network module. All timing commands are E1 controller configuration commands.

Single E1 Port Provides Clocking

In this scenario, the digital E1 module is the clock source for the connected device. The PLL generates the clock internally and drives the clocking on the E1 line.

Figure1: Single E1 Port Providing Clock

The following configuration sets up this clocking method:

controller E1 1/0
framing crc4
linecoding hdb3
clock source internal
pri-group timeslots 1-31

NoteGenerally, 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 provides a higher Stratum level.

Single E1 Port Receiving Clock from the Line

In this scenario, the digital E1 module receives clocking from the connected device (CO or PBX). The PLL clocking is driven by the clock reference on the receive (Rx) side of the E1 connection.

Figure2: Single E1 Receiving Clock from Line

The following configuration sets up this clocking method:

controller E1 1/0
framing crc4
linecoding b8zs
clock source line
pri-group timeslots 1-31

Dual E1s Receiving Clocking from the Line

In this scenario, the digital E1 has two reference clocks, one from the PBX and another from the CO. Because the PLL can only derive clocking from one source, this case is more complex than the two preceding examples.

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

• Looped-Time Clocking---The E1 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" the E1 so that the connected device has a viable clock and does not see slips. PBXs are not designed to accept slips on an 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 E1 port.

• Slips---These messages indicate that the E1 port is receiving clock information that is out of phase (out of synch). 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.

---

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

---

Figure3: Dual E1s Receiving Line Clocking

In this scenario, the PLL derives clocking from the CO and puts the E1 port connected to the PBX into looped-time mode. This is usually the best method because the CO provides an excellent clock source (and usually requires that it provide that source) and a PBX usually must receive clocking from the other E1 port.

The following configuration sets up this clocking method:

controller E1 1/0 << description - connected to the CO
framing crc4
linecoding hdb3
clock source line primary
pri-group timeslots 1-31
!
controller E1 1/1 << description - connected to the PBX
framing crc4
linecoding hdb3
clock source line
pri-group timeslots 1-31
 

The clock source line primary command tells the router to use this E1 port to drive the PLL. All other E1 ports configured as clock source line are then put into an implicit loop-timed mode. If the primary E1 port fails or goes down, the other E1 instead receives the clock that drives the PLL. In this configuration, E1 port 1/1 might see controlled slips, but these should not force it down. This method prevents the PBX from seeing slips.

Dual E1s---One Receives Clocking and One Provides Clocking

In this scenario, the digital E1 module receives clocking for the PLL from E1 0 and uses this clock as a reference to clock E1 1. If E1 0 fails, the PLL internally generates the clock reference to drive E1 1.

Figure4: Dual E1s---One Receiving and One Providing Clocking

The following configuration sets up this clocking method:

controller E1 1/0
framing crc4
linecoding hdb3
clock source line 
pri-group timeslots 1-31
!
controller E1 1/1
framing crc4
linecoding hdb3
clock source internal
pri-group timeslots 1-31

Dual E1s---Both Clocks from Router

In this scenario, the router generates the clock for the PLL and therefore for both E1s.

Figure5: Dual E1s---Both Clocks from Router

The following configuration sets up this clocking method:

controller E1 1/0
framing crc4
linecoding hdb3
clock source internal
pri-group timeslots 1-31
!
controller E1 1/1
framing esf
linecoding b8zs
clock source internal
pri-group timeslots 1-31

Signaling

There are three types of signaling that you should consider for digital E1:

• Channel-associated signaling (CAS)---CAS means that instead of having a specific timeslot (such as an ISDN D channel in PRI) designated to provide signaling only, signaling bits (on-hook and off-hook) are within the sixth, twelfth, eighteenth, and twenty-fourth frames of each timeslot. CAS is often called robbed-bit signaling (RBS) because it takes bits from bearer channels and uses them for signaling. CAS must be specified on both ends of the E1 link and is enabled by default on digital E1 packet voice trunk port adapters.

---

NoteDigital E1 packet voice trunk port adapters support E1 CAS. The digital E1 port adapter can support E&M wink-start, immediate-start, and delay-start signaling, as well as FXS and FXO ground-start and loop-start signaling.

---

• E&M signaling---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 because it provides better Answer Supervision (knowledge that the connected device is ready to answer the call). However, not all COs and PBXs can handle wink-start signaling. The E&M connection between the router and switch (CO or PBX) must use matching E&M signaling types otherwise calls cannot be connected properly. E&M signaling is defined with the ds0-group controller configuration command, as in the following example:

controller E1 1/0
ds0-group 1 timeslots 1-24 type e&m-wink-start

---

NoteCurrently, wink-start signaling provides only the functionality of feature-group B and not that of feature-group D, which will be supported in later releases.

---

• FXO and FXS signaling---While most digital E1 connections used for switch-to-switch (or switch-to-router) trunks are E&M connections, a digital E1 port adapter can also support FXS and FXO connections, which people normally use to provide emulated-OPX (Off-Premises eXtension) from a PBX to remote stations. As a general rule, FXO ports connect to FXS ports. 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. The FXO or FXS connection between the router and switch (CO or PBX) must use matching signaling, or calls cannot connect properly. FXS and FXO signaling are defined with the ds0-group controller configuration command, as in the following example:

controller E1 1/0
ds0-group 1 timeslots 1-24 type fxo-ground-start

or

controller E1 1/0
ds0-group 1 timeslots 1-24 type fxs-loop-start

NoteWhile some switches (CO or PBX) can specify both an inbound and outbound signaling method, Cisco VoIP gateway routers can only specify one signaling type for both inbound and outbound calls. The switch inbound and outbound signaling types must match, or calls might only work in one direction.

Framing

Digital E1 packet voice trunk port adapters support two types of framing for E1 CAS: Extended Superframe (ESF) and Super Frame (SF), also called D4 framing. The framing type of the router and switch (CO or PBX) must match. The framing controller configuration command defines E1 framing, as in the following example:

controller E1 1/0
framing esf
 

or

controller E1 1/0
framing sf

Line Encoding

Digital E1 packet voice trunk port adapters support two types of line encoding for E1 CAS: B8ZS (bipolar-8 zero substitution) and AMI (alternate mark inversion). The line encoding of the router and switch (CO or PBX) must match. The linecoding controller configuration command defines E1 framing, as in the following example:

controller E1 1/0
linecoding b8zs

or

controller E1 1/0
linecoding ami 

Verifying Voice Card and Controller Settings

To verify the configuration of voice card and controller settings, perform the following steps:

Router# show running-config
.
.
.
hostname router-alpha 
 
voice-card 1
 codec complexity high 
.
.
.
 

Step 2 Enter the privileged EXEC show controllers E1 command to display the status of E1 controllers and display information about clock sources and other settings for the E1 ports:

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
 

Configuring Serial Interfaces

The way you set up serial and LAN interfaces depends on your application. To configure VoIP, you must at least set up IP addresses for serial interfaces. When a user dials enough digits to match a configured destination pattern, the telephone number is mapped to an IP host through the dial-plan mapper. The IP host has a direct connection to either the destination telephone number or a PBX that completes the call to the configured destination pattern.

This document does not explain all possible serial interface configuration options, nor does it show LAN interface configuration. For complete information, see the Cisco IOS Release 12.1 Cisco IOS Interface Configuration Guide and the Cisco IOS Interface Command Reference.

The "Configuration Example" section shows a sample configuration. For more information about setting up voice networks, see Cisco IOS Multiservice Applications Configuration Guide for Cisco IOS Release 12.1.

NoteFor information about monitoring serial interfaces in order to trigger a busyout condition on a voice port when an interface is down, see "Configuring Voice Ports".

	Command	Purpose
Step1	Router# configure terminal	Enter global configuration mode.
Step2	Router(config)# interface serial slot/port:channel-group	Enter interface configuration mode for a serial interface that you specify by slot and port. The channel-group portion of the command is only required for channelized E1 interfaces. (For setting up channelized E1 interfaces, see Dial Services Configuration Guide for Cisco IOS Release 12.1.)
Step3	Router(config-if)# ip address ip-address mask	Assign the IP address and subnet mask to the interface.
Step4	Router(config-if)# isdn switch-type primary-qsig	Assign a switch type PRI or BRI interface, using primary-qsig for E1.
Step5	Router(config-if)# isdn protocol-emulate [user | network]	Configure the router's PRI interface to serve as either the primary QSIG slave or the QSIG master.
Step6	Router(config-if)# isdn incoming-voice [data [56 | 64] | modem [56 | 64]]	Route incoming calls to the modem and treat them as analog data, bypass the modem, or treat them as data.
Step7	Router(config-if)# fair-queue [congestive-discard-threshold [dynamic-queues | reservable-queues]]	Initiate a fair-queue for congestion control.
Step8	Router(config-if)# exit	Exit the interface.

Verifying Serial Interface Configuration

To verify serial interface configuration, enter the show interfaces serial privileged EXEC command, which displays the status of all serial interfaces or of a specific serial interface, as shown in the following example. You can use this command to check the encapsulation, IP addressing, and other settings:

Router# show interface serial0/0:0
Serial0/0:0 is up, line protocol is up 
  Hardware is QUICC Serial
  Internet address is 1.156.1.1/24
  MTU 1500 bytes, BW 1536 Kbit, DLY 20000 usec, 
     reliability 255/255, txload 1/255, rxload 1/255
  Encapsulation HDLC, loopback not set
  Keepalive not set
  Last input 00:00:00, output 00:00:00, output hang never
  Last clearing of "show interface" counters never
  Input queue: 0/75/0 (size/max/drops); Total output drops: 0
  Queueing strategy: weighted fair
  Output queue: 0/1000/64/0 (size/max total/threshold/drops) 
     Conversations  0/1/256 (active/max active/max total)
     Reserved Conversations 0/0 (allocated/max allocated)
  5 minute input rate 1000 bits/sec, 1 packets/sec
  5 minute output rate 1000 bits/sec, 1 packets/sec
     637 packets input, 64736 bytes, 0 no buffer
     Received 181 broadcasts, 0 runts, 5 giants, 0 throttles
     3617 input errors, 1506 CRC, 1646 frame, 0 overrun, 0 ignored, 0 abort
     682 packets output, 67213 bytes, 0 underruns
     0 output errors, 0 collisions, 1070 interface resets
     0 output buffer failures, 0 output buffers swapped out
     13 carrier transitions
     Timeslot(s) Used:1-24, Transmitter delay is 0 flags

Configuring Voice Ports

Follow these steps to set up voice ports to support the local and remote stations. Not all possible commands are shown here. To learn more, see Cisco IOS Multiservice Applications Configuration Guide and Cisco IOS Multiservice Applications Command Reference for Cisco IOS Release 12.1.

	Command	Purpose
Step1	Router# configure terminal	Enter global configuration mode.
Step2	Router(config)# voice-port slot/port:pri-group-number	Enter voice-port configuration mode. slot is the router location where the voice module is installed. Valid entries are from 0 to 3. port indicates the voice interface card location. Valid entries are 0 or 1. There is only one voice port per controller for QSIG. NoteThis voice-port command syntax does not apply to analog voice network modules and voice interface cards. The latter are specified using slot/subunit/port, designating the router slot for the voice network module, the location of the voice interface card in the network module, and the port on the voice interface card.
Step3	Router(config-voice-port)# busyout monitor interface interface number	(Optional) This command allows you to specify a LAN or WAN interface that will be monitored, and, when it is down, trigger a busyout (offhook) state on the voice port. This allows rerouting of calls. Busyout state for QSIG voice port implies that both the voice port and the signaling lines are down. You can enter the command repeatedly to specify as many interfaces, virtual interfaces, and subinterfaces as are required for a voice port. For example, if you enter the command three times so that three interfaces are monitored, the voice port only goes into busyout state when all three interfaces are down. When any one of the interfaces is operational, the busyout state is removed.
Step4	Router(config-voice-port)# comfort-noise	(Optional) The comfort noise parameter is enabled by default. It creates subtle background noise to fill silent gaps during calls when VAD is enabled on voice dial peers. If comfort noise is not generated, the silence can be unsettling to callers.
Step5	Router(config-voice-port)# echo-cancel enable	(Optional) This setting is enabled by default. Echo cancellation adds to the quality of voice transmissions by adjusting the echo that occurs on the interface due to impedance mismatches. Some echo is reassuring; echo over 25 milliseconds can cause problems.
Step6	Router(config-voice-port)# echo-cancel coverage {16 | 24 | 32 | 8}	(Optional) This command adjusts the echo canceller by the specified number of milliseconds; the default is 16.
Step7		Repeat Steps 2 through 6 for each DS0 group you create.
Step8	Router(config-voice-port)# exit	Exit voice-port configuration mode.
Step9	Router# compand type [a-law | u-law]	This command converts between analog and digital signals in PCM format. Specifying u-law is the North American mu-law ITU-T PCM encoding standard. Specifying a-law is the European a-law ITU-T PCM encoding standard.
Step10	Router# cp-tone	This command specifies a regional analog voice interface-related tone, ring, and cadence setting. You can also display the settings for different countries by using the question mark. For details about these commands, see the Cisco IOS Multiservice Applications Command Reference for Cisco IOS Release 12.1.

Verifying Voice Ports

To verify the voice-port configuration, enter the privileged EXEC show voice port slot/port:ds0-group command. The following sample output from the command shows explanatory information. Important command output is shown in bold.

Router# show voice port 1/0:1
 
receive 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
 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

Configuring Voice Dial Peers

Follow these steps to set up voice dial peers to support the local and remote stations. Not all possible commands are shown here. To learn more, see Cisco IOS Multiservice Applications Configuration Guide and Cisco IOS Multiservice Applications Command Reference for Cisco IOS Release 12.1.

	Command	Purpose
Step1	Router# configure terminal	Enter global configuration mode.
Step2	Router(config)# dial-peer voice number pots	Enter dial-peer configuration mode and define a local dial peer that will connect to the plain old telephone service (POTS) network. number is one or more digits identifying the dial peer. Valid entries are from 1 to 2147483647. pots indicates a peer using basic telephone service.
Step3	Router(config-dialpeer)# destination-pattern string [T]	Configure the dial peer's destination pattern so that the system can reconcile dialed digits with a telephone number. string is a series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9 and the letters A through D. The plus symbol (+) is not valid. The following special characters can be entered: The star character (*) that appears on standard touch-tone dial pads can be in any dial string but not as a leading character (for example, *650). The period (.) acts as a wildcard character. The comma (,) can be used only in prefixes and inserts a one-second pause. When the timer (T) character is included at the end of the destination pattern, the system collects dialed digits as they are entered---until the interdigit timer expires (10 seconds, by default)---or the user dials the termination of end-of-dialing key (default is #). NoteThe timer character must be a capital T.
Step4	Router(config-dialpeer)# prefix string	(Optional) Include a dial-out prefix that the system enters automatically instead of people dialing it. string is a value from 0 to 9, and you can use a comma (,) to indicate a pause. NoteThere are other digit manipulation commands available to handle such situations as prefixes for special services, ignoring some digits, and dialing in to remote PBXs as though they are local.
Step5	Router(config-dialpeer)# port slot/port:ds0-group-number	This command associates the dial peer with a specific logical interface. slot is the router location where the voice module is installed. Valid entries are from 0 to 3. port indicates the voice interface card location. Valid entries are 0 or 1. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital E1 card.
Step6	Router(config)# dial-peer voice number voip	Enter dial-peer configuration mode and define a remote VoIP dial peer. number is one or more digits identifying the dial peer. Valid entries are from 1 to 2147483647. voip indicates a VoIP peer using voice encapsulation on the IP network.
Step7	Router(config-dialpeer)# codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g728 | g729r8 [pre-ietf] | g729br8} [bytes]	The voice-card configuration codec complexity command sets the codec options that are available when you enter this command. See Step 3 of the "Configuring Voice Card and E1 Controller Settings" section. If you do not set codec complexity, g729r8 with IETF bit-ordering is used. If you set codec complexity to high, the following options are available: g711alaw---G.711 A Law 64,000 bps g711ulaw---G.711 u Law 64,000 bps g723ar53---G.723.1 Annex A 5,300 bps g723ar63---G.723.1 Annex A 6,300 bps g723r53---G.723.1 5,300 bps g723r63---G.723.1 6,300 bps g726r16---G.726 16,000 bps g726r24---G.726 24,000 bps g726r32---G.726 32,000 bps g728---G.728 16,000 bps g729r8---G.729 8,000 bps (default) g729br8---G.729 Annex B 8,000 bps If you set codec complexity to medium, the following options are valid: g711alaw---G.711 A Law 64,000 bps g711ulaw---G.711 u Law 64,000 bps g726r16---G.726 16,000 bps g726r24---G.726 24,000 bps g726r32---G.726 32,000 bps g729r8---G.729 Annex A 8,000 bps g729br8---G.729 Annex B with Annex A 8,000 bps The optional bytes parameter sets the number of voice data bytes per frame. Acceptable values are from 10 to 240 in increments of 10 (for example, 10, 20, 30, and so on). Any other value is rounded down (for example, from 236 to 230). If you specify g729r8, then the IETF (Internet Engineering Task Force) bit-ordering is used. For interoperability with a Cisco 2600, 3600, or AS5300 router running a Cisco IOS release prior to Release 12.0(5)T or12.0(4)XH, you must specify the additional keyword pre-ietf after g729r8.
Step8	Router(config-dialpeer)# vad	(Optional) This setting is enabled by default. It activates voice activity detection (VAD). VAD allows the system to reduce unnecessary voice transmissions caused by unfiltered background noise.
Step9	Router(config-dialpeer)# dtmf-relay [cisco-rtp] [h245-signal] [h245-alphanumeric]	(Optional) Dual-tone multifrequency (DTMF) describes the tone that sounds in response to a keypress on a touch-tone telephone. DTMF tones are compressed at one end of a call and decompressed at the other end. If a low-bandwidth codec, such as a G.729 or G.723, is used, the tones can sound distorted. The dtmf-relay command transports DTMF tones generated after call establishment out-of-band by using a method that transmits with greater fidelity than is possible in-band for most low-bandwidth codecs. Without DTMF relay, calls established with low-bandwidth codecs may have trouble accessing automated telephone menu systems, such as voicemail and interactive voice response (IVR) systems. A signaling method is supplied only if the remote end supports it, and the options are: Cisco proprietary (cisco-rtp), standard H.323 (h245-alphanumeric), and H.323 standard with signal duration (h245-signal).
Step10	Router(config-dialpeer)# fax-rate {2400 | 4800 | 7200 | 9600 | 12000 | 14400 | disable | voice}	(Optional) Specify the transmission speed of a fax to be sent to this dial peer. The disable keyword turns off fax transmission capability, and the voice keyword specifies the highest possible fax speed supported by the voice rate.
Step11	Router(config-dialpeer)# session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name}	Configure the IP session target for the dial peer. ipv4:destination-address indicates IP address of the dial peer. dns:host-name indicates that the domain name server will resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device. There are also wildcards available for defining domain names with the keyword by using source, destination, and dialed information in the host name. For complete command syntax information, see Cisco IOS Multiservice Applications Command Reference for Cisco IOS Release12.1.
Step12	Router(config-dialpeer)# forward-digit [all | default | extra | .. ]	Configure the interface to forward digits for voice calls.
Step13	Router(config-dialpeer)# huntstop	(Optional) Disable hunting by the interface for dial peers.
Step14	Router(config-dialpeer)# exit	Exit interface configuration.

Verifying Voice Dial Peers

Enter the privileged EXEC show dial-peer voice command. The following text is sample output from the command for a POTS dial peer. Important command output is shown in bold.

Router# show dial-peer voice 1
VoiceEncapPeer1
        tag = 1, dest-pat = \Q+14085551000',
        answer-address = \Q',
        group = 0, Admin state is up, Operation state is down
        Permission is Both,
        type = pots, prefix = \Q',
        session-target = \Q', voice-port =
        Connect Time = 0, Charged Units = 0
        Successful Calls = 0, Failed Calls = 0
        Accepted Calls = 0, Refused Calls = 0
        Last Disconnect Cause is "10"
        Last Disconnect Text is ""
        Last Setup Time = 0 
 

The following text is sample output from the show dial-peer voice command for a VoIP dial peer:

Router# show dial-peer voice 10
VoiceOverIpPeer10
        tag = 10, dest-pat = \Q',
        incall-number = \Q+14087',
        group = 0, Admin state is up, Operation state is down
        Permission is Answer, 
        type = voip, session-target = \Q',
        sess-proto = cisco, req-qos = bestEffort, 
        acc-qos = bestEffort, 
        fax-rate = voice, codec = g729r8,
        Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled, 
        Connect Time = 0, Charged Units = 0
        Successful Calls = 0, Failed Calls = 0
        Accepted Calls = 0, Refused Calls = 0
        Last Disconnect Cause is "10"
        Last Disconnect Text is ""
        Last Setup Time = 0

Monitoring and Maintaining E1 Digital Packet VoiceConfiguration

This section presents some useful commands for understanding, maintaining, and troubleshooting your configuration. Table 1 lists the debug and show commands.

Table1: Debug and Show Commands for Maintaining and Troubleshooting Your Configuration

Command	Purpose
Router# show dialplan number number	Shows which dial-peer is matched by a called number.
Router# show call active voice	Shows statistics for currently active voice calls.
Router# show call active fax	Shows statistics for currently active fax calls.
Router# show call history voice	Shows statistics on previous voice calls.
Router# show call history fax	Shows statistics on previous fax calls.
Router# show voice port	Shows the status of voice ports. See "Verifying Voice Ports".
Router# show controller E1 slot/port	Shows the status of the E1 controller. See "Verifying Voice Card and Controller Settings".
Router# show isdn status	Shows the status of an individual ISDN line.
Router# debug ccapi inout	Debugs the E1
Router# debug isdn q931	Debugs calls as they are set up and torn down on ISDN network connections (Layer 3) between the local router (user side) and the network.
Router# debug vpm all	Debugs the E1 signaling.
Router# debug vtsp all	Debugs the digits received and sent.
Router# debug voip ccapi inout	Debugs the call setup process.

Show Commands

This section illustrates some of the privileged EXEC show commands that are useful for analyzing your system. Note that important information appears in bold.

The show dialplan number command provides information about the dial peer associated with a specified dial-plan number. Notice that the dial peer is operational and that IP Precedence has been configured to the preferred setting of 5.

NoteTo pair different voice ports and telephone numbers together for troubleshooting, enter the show dialplan incall number privileged EXEC command.

Router# show dialplan number 75435
Macro Exp.: ##75435
VoiceOverIpPeer70000
        information type = voice,
        tag = 70000, destination-pattern = \Q##7....',
        answer-address = \Q', preference=0,
        group = 70000, Admin state is up, Operation state is up,
        incoming called-number = \Q', connections/maximum = 0/unlimited,
        DTMF Relay = disabled,
        application associated:
        type = voip, session-target = \Qipv4:171.68.253.18',
        technology prefix:
        settlement: disabled
        ip precedence = 5, UDP checksum = disabled,
        session-protocol = cisco, req-qos = best-effort,
        acc-qos = best-effort,
        fax-rate = 14400,   payload size =  20 bytes
        codec = g729r8,   payload size =  20 bytes,
        Expect factor = 10, Icpif = 30,signaling-type = cas,
        VAD = disabled, Poor QOV Trap = disabled,
        Connect Time = 0, Charged Units = 0,
        Successful Calls = 3, Failed Calls = 0,
        Accepted Calls = 3, Refused Calls = 0,
        Last Disconnect Cause is "10  ",
        Last Disconnect Text is "normal call clearing.",
        Last Setup Time = 344813.
Matched: ##75435   Digits: 3
Target: ipv4:171.68.253.18
 

The show call active voice command displays information about a current call:

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=171.68.235.18
RemoteUDPPort=16580
RoundTripDelay=29 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
SessionProtocol=cisco
SessionTarget=ipv4:171.68.235.18
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
 

The show call history voice command shows statistics about previous calls:

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=171.68.235.18
RemoteUDPPort=16552
RoundTripDelay=23 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
SessionProtocol=cisco
SessionTarget=ipv4:171.68.235.18
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
VAD = disabled
CoderTypeRate=g729r8
CodecBytes=20
cvVoIPCallHistoryIcpif=0
SignalingType=cas
 

The show isdn status command shows the status of ISDN calls:

Router# show isdn status
 
Global ISDN Switchtype = primary-qsig
ISDN Serial1/015 interface
        ******* Network side configuration ******* 
        dsl 0, interface ISDN Switchtype = primary-qsig
         **** Master side configuration ****
    Layer 1 Status
        ACTIVE
    Layer 2 Status
        TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
    Layer 3 Status
        24 Active Layer 3 Call(s)
    Activated dsl 0 CCBs = 24
        CCBcallid=E3C, sapi=0, ces=0, B-chan=1, calltype=VOICE
        CCBcallid=E3D, sapi=0, ces=0, B-chan=2, calltype=VOICE
        CCBcallid=E3E, sapi=0, ces=0, B-chan=3, calltype=VOICE
        CCBcallid=E3F, sapi=0, ces=0, B-chan=4, calltype=VOICE
        CCBcallid=E40, sapi=0, ces=0, B-chan=5, calltype=VOICE
        CCBcallid=E47, sapi=0, ces=0, B-chan=6, calltype=VOICE
        CCBcallid=E48, sapi=0, ces=0, B-chan=7, calltype=VOICE
        CCBcallid=E49, sapi=0, ces=0, B-chan=8, calltype=VOICE
        CCBcallid=E50, sapi=0, ces=0, B-chan=9, calltype=VOICE
        CCBcallid=E51, sapi=0, ces=0, B-chan=10, calltype=VOICE
        CCBcallid=E52, sapi=0, ces=0, B-chan=11, calltype=VOICE
        CCBcallid=E53, sapi=0, ces=0, B-chan=12, calltype=VOICE
        CCBcallid=E54, sapi=0, ces=0, B-chan=13, calltype=VOICE
        CCBcallid=E5B, sapi=0, ces=0, B-chan=14, calltype=VOICE
        CCBcallid=E5C, sapi=0, ces=0, B-chan=15, calltype=VOICE
        CCBcallid=E5D, sapi=0, ces=0, B-chan=17, calltype=VOICE
        CCBcallid=E5E, sapi=0, ces=0, B-chan=18, calltype=VOICE
        CCBcallid=E5F, sapi=0, ces=0, B-chan=19, calltype=VOICE
        CCBcallid=E60, sapi=0, ces=0, B-chan=20, calltype=VOICE
        CCBcallid=E61, sapi=0, ces=0, B-chan=21, calltype=VOICE
        CCBcallid=E62, sapi=0, ces=0, B-chan=22, calltype=VOICE
        CCBcallid=E63, sapi=0, ces=0, B-chan=23, calltype=VOICE
        CCBcallid=E64, sapi=0, ces=0, B-chan=24, calltype=VOICE
        CCBcallid=E6B, sapi=0, ces=0, B-chan=25, calltype=VOICE
    The Free Channel Mask  0xFE000000
    Total Allocated ISDN CCBs = 24
 

The show dial-peer voice summary command displays information about dial-peers that are active:

Router# show dial-peer voice summary
 
dial-peer hunt 0
  TAG TYPE   ADMIN OPER PREFIX   DEST-PATTERN    PREF SESS-TARGET    PORT
    1 pots   up    up            3                0                  1/015 
  100 voip   down  down          1                0   ipv41.2.79.7        
  200 voip   down  down          1                0   ipv41.2.79.31       
  300 vofr   up    up            1                0   Serial0/0 990       
  400 voip   down  down          1                0   ipv45.5.5.2         
 

The show voice call summary command displays a summary of all dial-peers that are active:

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            
1/015.13 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.14 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.15 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.17 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.18 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.19 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.20 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.21 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.22 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.23 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.24 g729r8    y  S_CONNECT             S_TSP_CONNECT            
1/015.25 g729r8    y  S_CONNECT             S_TSP_CONNECT            
 

The show voice dsp command displays current status of all DSP voice channels:

Router# show voice dsp
 
                                BOOT                      PAK
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
 

The show voice trace command displays a trace of all active voice transitions:

Router# show voice trace
 
1/015 1  State Transitions (state, event) -> (state, event) ...
(S_NULL, E_TSP_INFO_IND) -> (S_SETUP_INDICATED, E_TSP_INFO_IND) ->
(S_SETUP_INDICATED, E_TSP_INFO_IND) -> (S_SETUP_INDICATED, E_CC_PROCEEDING) ->
(S_SETUP_INDICATED, E_CC_ALERT) -> (S_ALERTING, E_CC_BRIDGE) ->
(S_ALERTING, E_CC_CONNECT) -> (S_CONNECT, E_CC_CAPS_IND) ->
(S_CONNECT, E_CC_CAPS_ACK) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->
(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->
(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->
(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->
(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->
(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->
(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->
(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->
(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->
(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_DSP_DTMF_DIGIT_BEGIN) ->
(S_CONNECT, E_DSP_DTMF_DIGIT) -> (S_CONNECT, E_TIMER) ->
 

The show adapi command displays information about the call distribution application programming interface (CDAPI):

Router# show cdapi
 
Registered CDAPI Applications/Stacks
====================================
 
Application TSP CDAPI Application Voice
        Application Type(s)  Voice Facility Signaling 
        Application Level    Tunnel
        Application Mode     Enbloc
 
Signaling Stack ISDN
        Interface Se1/015
 
CDAPI Message Buffers
=====================
 
Used Msg Buffers 0, Free Msg Buffers 6400
Used Raw Buffers 0, Free Raw Buffers 3200
Used Large-Raw Buffers 0, Free Large-Raw Buffers 320
2600-1#
2600-1#
2600-1#s vo call 1/015.1
1/015 1  vtsp level 0 state = S_CONNECT
 
callid 0x0EDE B01 state S_TSP_CONNECT clld 1 cllg 3456546347
2600-1# ***DSP VOICE VP_DELAY STATISTICS***
Clk Offset(ms) -383401219, Rx Delay Est(ms) 61
Rx Delay Lo Water Mark(ms) 61, Rx Delay Hi Water Mark(ms) 90
        ***DSP VOICE VP_ERROR STATISTICS***
Predict Conceal(ms) 0, Interpolate Conceal(ms) 0
Silence Conceal(ms) 0, Retroact Mem Update(ms) 0
Buf Overflow Discard(ms) 20, Talkspurt Endpoint Detect Err 0
        ***DSP VOICE RX STATISTICS***
Rx Vox/Fax Pkts 286, Rx Signal Pkts 0, Rx Comfort Pkts 0
Rx Dur(ms) 24870, Rx Vox Dur(ms) 8510, Rx Fax Dur(ms) 0
Rx Non-seq Pkts 0, Rx Bad Hdr Pkts 0
Rx Early Pkts 0, Rx Late Pkts 0
        ***DSP VOICE TX STATISTICS***
Tx Vox/Fax Pkts 826, Tx Sig Pkts 0, Tx Comfort Pkts 0
Tx Dur(ms) 24870, Tx Vox Dur(ms) 24790, Tx Fax Dur(ms) 0
        ***DSP VOICE ERROR STATISTICS***
Rx Pkt Drops(Invalid Header) 0, Tx Pkt Drops(HPI SAM Overflow) 0
        ***DSP LEVELS***
TDM Bus Levels(dBm0) Rx -12.5 from PBX/Phone, Tx -13.2 to PBX/Phone
TDM ACOM Levels(dBm0) +0.0, TDM ERL Level(dBm0) +23.5
TDM Bgd Levels(dBm0) -12.1, with activity being voice

Debug Commands

This section illustrates some of the EXEC mode debug commands that are useful when analyzing and troubleshooting your system. Note that important information appears in bold.

The debug isdn q931 command displays information about call setup and teardown of ISDN network connections (Layer 3) between the local router (user side) and the network.

The debug voip ccapi inout EXEC command traces the execution path through the call control API, which serves as the interface between the call-session application and the underlying network-specific software.

Router# debug isdn q931
Router# debug voip ccapi inout
 
001041 ISDN Se1/015 RX <-  SETUP pd = 8  callref = 0x1EC5 << the originating call
001041         Sending Complete
001041         Bearer Capability i = 0x8090A3
001041         Channel ID i = 0xA98381
001041         Calling Party Number i = 0x91, '0987654321'
001041         Calling Party SubAddr i = 0x80, 'P123'
001041         Called Party Number i = 0x91, '2312'
001041         Called Party SubAddr i = 0x80, 'P321'
001041         High Layer Compat i = 0x9181
001041         Locking Shift to Codeset 5
001041         Codeset 5 IE 0x31  i = 0x80
001041         Codeset 5 IE 0x32  i = 0x80
0010180388626431 vtsp_tsp_call_setup_ind (sdb=0x81A57008, tdm_info=0x0,
.
.
.
0029107374182399 ISDN BR1/0 TX ->  SETUP pd = 8  callref = 0x0001 << terminating call
0029105245511244         Bearer Capability i = 0x8090A3
0029103079215104         Channel ID i = 0xA98381
0029103079215104         Calling Party Number i = 0x91, '0987654321'
0029103079215104         Calling Party SubAddr i = 0x80, 'P123'
0029103079215104         Called Party Number i = 0x91, '312'
0029103079215104         Called Party SubAddr i = 0x80, 'P321'
0029103079215104         Sending Complete
0029103079215104         High Layer Compat i = 0x9181
0029103079215104         Locking Shift to Codeset 5
0029105245510852         Codeset 5 IE 0x31  i = 0x80
0029103079215104         Codeset 5 IE 0x32  i = 0x80
002925 ISDN BR1/0 RX <-  RELEASE_COMP pd = 8  callref = 0x8001
002925         Cause i = 0x8096 - Number changed 
002925         Facility i = 0x91A4053132333435
002925         User-User i = 0x08, 'USER', 0x20, 'INFORMATION'
.
.
.
003234359738368         Channel ID i = 0xA98381
003234359738368         Calling Party Number i = 0x91, '0987654321'
003234359738368         Calling Party SubAddr i = 0x80, 'P123'
003234359738368         Called Party Number i = 0x91, '312'
003234359738368         Called Party SubAddr i = 0x80, 'P321'
003234359738368         Sending Complete
003234359738368         High Layer Compat i = 0x9181
003234359738368         Locking Shift to Codeset 5
003236526034116         Codeset 5 IE 0x31  i = 0x80
003234359738368         Codeset 5 IE 0x32  i = 0x80
003209 ISDN BR1/0 RX <-  CALL_PROC pd = 8  callref = 0x8003
003209         Channel ID i = 0xA98381
003224 ISDN BR1/0 RX <-  PROGRESS pd = 8  callref = 0x8003
003224         Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have
in-band
info 

Table 2 explains the codec negotiation values that appear---in hexadecimal format--- during the capabilities exchange portion of the command output.

Table2: Codec Negotiation Values in debug voip ccapi inout

Negotiation Value in Decimal	Meaning
1	U-law PCM (g711ulaw)
2	A-law PCM (g711alaw)
3	32k ADPCM (g726r32)
4	24k ADPCM (g726r24)
5	16k ADPCM (g726r16)
6	CS-ACELP - pre-IETF (g729r8 pre-ietf)
7	Medium complexity CS-ACELP - pre-IETF (g729ar8 pre-ietf)
8	CS-ACELP with VAD (g729br8)
9	Medium complexity CS-ACELP with VAD (G.729abr8)
10	16K LD-CELP (g728)
11	G.723.1 High Rate - 6300 bps (g723r63)
12	G.723.1 High Rate with VAD - 6300 bps (g723ar63)
13	G.723.1 Low Rate - 5300 bps (g723r53)
14	G.723.1 Low Rate with VAD - 5300 bps (g723ar53)
19	CS-ACELP - IETF standard (g729r8)
20	Medium complexity CS-ACELP - IETF standard (g729ar8)

Reference Information

The information in this section helps you interpret the output from debug and show commands.

Table 3 shows Q.931 call disconnection causes. In the examples that follow, the disconnects are caused by normal call clearing.

Table3: Q.931 Call Disconnection Causes

Call Disconnection Cause Value	Meaning and Number
CC_CAUSE_UANUM = 0x1	/* unassigned number. (1) */
CC_CAUSE_NO_ROUTE = 0x3	/* no route to destination. (3) */
CC_CAUSE_NORM = 0x10	/* normal call clearing. (16) */
CC_CAUSE_BUSY = 0x11	/* user busy. (17) */
CC_CAUSE_NORS = 0x12	/* no user response. (18) */
CC_CAUSE_NOAN = 0x13	/* no user answer. (19) */
CC_CAUSE_REJECT = 0x15	/* call rejected. (21) */
CC_CAUSE_INVALID_NUMBER = 0x1C	/* invalid number. (28) */
CC_CAUSE_UNSP = 0x1F	/* normal, unspecified. (31) */
CC_CAUSE_NO_CIRCUIT = 0x22	/* no circuit. (34) */
CC_CAUSE_NO_REQ_CIRCUIT = 0x2C	/* no requested circuit. (44) */
CC_CAUSE_NO_RESOURCE = 0x2F	/* no resource. (47) */
CC_CAUSE_NOSV = 0x3F	/* service or option not available, Unspecified. (63) */
CC_CAUSE_UNINITIALIZED = 0	/* un-initialized (0) */
CC_CAUSE_UANUM = 1	/* unassigned num */
CC_CAUSE_NO_ROUTE_TO_TRANSIT_NETWORK = 2
CC_CAUSE_NO_ROUTE = 3	/* no rt to dest */
CC_CAUSE_SEND_INFO_TONE = 4
CC_CAUSE_MISDIALLED_TRUNK_PREFIX = 5
CC_CAUSE_CHANNEL_UNACCEPTABLE = 6
CC_CAUSE_CALL_AWARDED = 7
CC_CAUSE_PREEMPTION = 8
CC_CAUSE_PREEMPTION_RESERVED = 9
CC_CAUSE_NORM = 16
CC_CAUSE_BUSY = 17	/* user busy */
CC_CAUSE_NORS = 18	/* no user response*/
CC_CAUSE_NOAN = 19	/* no user answer. */
CC_CAUSE_SUBSCRIBER_ABSENT = 20
CC_CAUSE_REJECT = 21	/* call rejected. */
CC_CAUSE_NUMBER_CHANGED = 22
CC_CAUSE_NON_SELECTED_USER_CLEARING = 26
CC_CAUSE_DESTINATION_OUT_OF_ORDER = 27
CC_CAUSE_INVALID_NUMBER = 28
CC_CAUSE_FACILITY_REJECTED = 29
CC_CAUSE_RESPONSE_TO_STATUS_ENQUIRY = 30
CC_CAUSE_UNSP = 31	/* unspecified. */
CC_CAUSE_NO_CIRCUIT = 34	/* no circuit. */
CC_CAUSE_REQUESTED_VPCI_VCI_NOT_AVAILABLE = 35
CC_CAUSE_VPCI_VCI_ASSIGNMENT_FAILURE = 36
CC_CAUSE_CELL_RATE_NOT_AVAILABLE = 37
CC_CAUSE_NETWORK_OUT_OF_ORDER = 38
CC_CAUSE_PERM_FRAME_MODE_OUT_OF_SERVICE = 39
CC_CAUSE_PERM_FRAME_MODE_OPERATIONAL = 40
CC_CAUSE_TEMPORARY_FAILURE = 41
CC_CAUSE_SWITCH_CONGESTION = 42
CC_CAUSE_ACCESS_INFO_DISCARDED = 43
CC_CAUSE_NO_REQ_CIRCUIT = 44
CC_CAUSE_NO_VPCI_VCI_AVAILABLE = 45
CC_CAUSE_PRECEDENCE_CALL_BLOCKED = 46
CC_CAUSE_NO_RESOURCE = 47	/* no resource. */
CC_CAUSE_QOS_UNAVAILABLE = 49
CC_CAUSE_FACILITY_NOT_SUBCRIBED = 50
CC_CAUSE_CUG_OUTGOING_CALLS_BARRED = 53
CC_CAUSE_CUG_INCOMING_CALLS_BARRED = 55
CC_CAUSE_BEARER_CAPABILITY_NOT_AUTHORIZED = 57
CC_CAUSE_BEARER_CAPABILITY_NOT_AVAILABLE = 58
CC_CAUSE_INCONSISTENCY_IN_INFO_AND_CLASS = 62
CC_CAUSE_NOSV = 63	/* service or option * not available * unspecified. */
CC_CAUSE_BEARER_CAPABILITY_NOT_IMPLEMENTED = 65
CC_CAUSE_CHAN_TYPE_NOT_IMPLEMENTED = 66
CC_CAUSE_FACILITY_NOT_IMPLEMENTED = 69
CC_CAUSE_RESTRICTED_DIGITAL_INFO_BC_ONLY = 70
CC_CAUSE_SERVICE_NOT_IMPLEMENTED = 79
CC_CAUSE_INVALID_CALL_REF_VALUE = 81
CC_CAUSE_CHANNEL_DOES_NOT_EXIST = 82
CC_CAUSE_CALL_EXISTS_CALL_ID_IN_USE = 83
CC_CAUSE_CALL_ID_IN_USE = 84
CC_CAUSE_NO_CALL_SUSPENDED = 85
CC_CAUSE_CALL_CLEARED = 86
CC_CAUSE_USER_NOT_IN_CUG = 87
CC_CAUSE_INCOMPATIBLE_DESTINATION = 88
CC_CAUSE_NON_EXISTENT_CUG = 90
CC_CAUSE_INVALID_TRANSIT_NETWORK = 91
CC_CAUSE_AAL_PARMS_NOT_SUPPORTED = 93
CC_CAUSE_INVALID_MESSAGE = 95
CC_CAUSE_MANDATORY_IE_MISSING = 96
CC_CAUSE_MESSAGE_TYPE_NOT_IMPLEMENTED = 97
CC_CAUSE_MESSAGE_TYPE_NOT_COMPATIBLE = 98
CC_CAUSE_IE_NOT_IMPLEMENTED = 99
CC_CAUSE_INVALID_IE_CONTENTS = 100
CC_CAUSE_MESSAGE_IN_INCOMP_CALL_STATE = 101
CC_CAUSE_RECOVERY_ON_TIMER_EXPIRY = 102
CC_CAUSE_NON_IMPLEMENTED_PARAM_PASSED_ON = 103
CC_CAUSE_UNRECOGNIZED_PARAM_MSG_DISCARDED = 110
CC_CAUSE_PROTOCOL_ERROR = 111
CC_CAUSE_INTERWORKING = 127

• CC_CAP_CODEC_G711U 0x1

• CC_CAP_CODEC_G711A 0x2

• CC_CAP_CODEC_G723ar63 0x2000

• CC_CAP_CODEC_G723ar53 0x4000

• CC_CAP_CODEC_G723r63 0x100

• CC_CAP_CODEC_G723r53 0x200

• CC_CAP_CODEC_G726r16 0x10

• CC_CAP_CODEC_G729 0x4

• CC_CAP_CODEC_G729 0x8000

• CC_CAP_CODEC_G729a 0x8

• CC_CAP_CODEC_G729b 0x800

• CC_CAP_CODEC_G729ab 0x1000

• CC_CAP_FAX_NONE 0x1

• CC_CAP_FAX_VOICE 0x2

• CC_CAP_FAX_144 0x4

• CC_CAP_FAX_96 0x8

• CC_CAP_FAX_72 0x10

• CC_CAP_FAX_48 0x20

• CC_CAP_FAX_24 0x40

• CC_CAP_FAX_120 0x80

• CC_CAP_VAD_OFF 0x1

• CC_CAP_VAD_ON 0x2

Configuration Example

This section displays the configuration example of a router running a digital E1 packet voice trunk network module interface:

Router#show running-config
 
 
Building configuration...
 
Current configuration
!
version 12.1
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname 2600-1
!
!
!
memory-size iomem 10
voice-card 1
!
ip subnet-zero
no ip domain-lookup
!
frame-relay switching
isdn switch-type primary-qsig
isdn voice-call-failure 0
voice hunt user-busy
!
!
!
controller E1 1/0
 pri-group timeslots 1-31
!
controller E1 1/1
 shutdown
!
!
!
interface Ethernet0/0
 ip address 1.2.79.1 255.255.0.0
 no ip directed-broadcast
 no cdp enable
!
interface Serial0/0
 no ip address
 no ip directed-broadcast
 encapsulation frame-relay
 no ip mroute-cache
 load-interval 30
 clockrate 800000
 frame-relay traffic-shaping
 frame-relay class voice-vc
 frame-relay interface-dlci 990
  vofr data 4 call-control 5
 frame-relay intf-type dce
!
interface Ethernet0/1
 no ip address
 no ip directed-broadcast
 shutdown
 no cdp enable
!
interface Serial0/1
 ip address 5.5.5.1 255.0.0.0
 no ip directed-broadcast
 encapsulation frame-relay
 no ip mroute-cache
 clockrate 800000
 frame-relay traffic-shaping
 frame-relay class voice-data
 frame-relay interface-dlci 991
 frame-relay ip rtp header-compression
 frame-relay intf-type dce
!
interface Serial1/015
 no ip address
 no ip directed-broadcast
 ip mroute-cache
 no logging event link-status
 isdn switch-type primary-qsig
 isdn overlap-receiving
 isdn protocol-emulate network
 isdn incoming-voice voice
 no isdn T309-enable
 isdn bchan-number-order ascending
 fair-queue 64 256 0
 no cdp enable
!
router rip
 network 172.28.0.0
!
 router igrp 1
 redistribute connected
 network 1.0.0.0
!
ip default-gateway 1.2.0.1
ip classless
ip route 223.255.254.254 255.255.255.255 1.2.0.1
no ip http server
!
!
map-class frame-relay voice-vc
 no frame-relay adaptive-shaping
 frame-relay cir 512000
 frame-relay bc 512000
 frame-relay fair-queue
 frame-relay voice bandwidth 512000
 frame-relay fragment 100
!
map-class frame-relay voice-data
 no frame-relay adaptive-shaping
 frame-relay cir 512000
 frame-relay bc 1000
 frame-relay fair-queue
 frame-relay fragment 200
 frame-relay ip rtp priority 2000 16383 500
dialer-list 1 protocol ip permit
dialer-list 1 protocol ipx permit
no cdp run
!
voice-port 1/015
 compand-type a-law
!
dial-peer voice 1 pots
 destination-pattern 3
 direct-inward-dial
 port 1/015
 forward-digits all
!
dial-peer voice 100 voip
 shutdown
 destination-pattern 1
 session target ipv41.2.79.7
!
dial-peer voice 200 voip
 shutdown
 destination-pattern 1
 session target ipv41.2.79.31
!
dial-peer voice 300 vofr
 destination-pattern 1
 session target Serial0/0 990
!
dial-peer voice 400 voip
 shutdown
 destination-pattern 1
 session target ipv45.5.5.2
!
!
line con 0
 exec-timeout 0 0
 transport input none
line aux 0
line vty 0 4
 password ard
 login
!
end

Command Reference

This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.1 command references.

• pri-group

pri-group

To specify a ISDN Primary Rate interface (PRI) on a channelized T1 or E1 controller, enter the pri-group controller configuration command. To remove the ISDN-PRI configuration, enter the no form of this command.

pri-group timeslots timeslot-range

no pri-group

Syntax Description

timeslot-range

Specifies a single range of values. For T1, the allowable range is from 1 to 23. For E1, the allowable values are from 1 to 15.

Defaults

There is no ISDN-PRI group configured.

Command Modes

Controller configuration

Command History

Release	Modification
11.0	This command was introduced.
12.0(2)T	The command was introduced for the Cisco MC3810 multiservice access concentrator.
12.0(7)XK	The command was introduced for the Cisco 2600 and 3600 series.
12.1(2)T	The command was implemented in Cisco IOS Release 12.1(2)T.

Usage Guidelines

The pri-group command applies to the configuration of Voice over Frame Relay and Voice over ATM on the Cisco MC3810 multiservice access concentrator and the Cisco 2600 and 3600 series routers.

Before you enter the pri-group command, you must specify an ISDN-PRI switch type and an E1 or T1 controller.

---

NoteOnly one PRI group can be configured on a controller.

---

Examples

The following example configures ISDN-PRI on all timeslots of controller E1 on a Cisco 2600 series router:

Router(config-controller)# pri-group timeslots 1-7, 16
 
controller E1 4/0
!
controller E1 4/1
 pri-group timeslots 1-7,16
!

Related Commands

Command	Description
isdn switch-type	To configure the Cisco 2600 series router PRI interface to support QSIG signaling, enter this command.

Glossary

AAL---ATM Adaptation Layer. Service-dependent sublayer of the data link layer. The AAL accepts data from different applications and presents it to the ATM layer in the form of 48-byte ATM payload segments. AALs consist of two sublayers: convergence sublayer (CS) and segmentation and reassembly (SAR). AALs differ on the basis of the source-destination timing used, whether they use constant bit rate (CBR) or variable bit rate (VBR), and whether they are used for connection-oriented or connectionless mode data transfer. At present, the four types of AAL recommended by the ITU-T are AAL1, AAL2, AAL3/4, and AAL5.

AAL1---ATM Adaptation Layer 1. One of four AALs recommended by the ITU-T. AAL1 is used for connection-oriented, delay-sensitive services requiring constant bit rates, such as uncompressed video and other isochronous traffic.

AMI---alternate mark inversion. Line-code type used on T1 and E1 circuits. In AMI, zeros are represented by 01 during each bit cell, and ones are represented by 11 or 00, alternately, during each bit cell. AMI requires that the sending device maintain ones density. Ones density is not maintained independent of the data stream. Sometimes called binary coded alternate mark inversion.

ATM---Asynchronous Transfer Mode. International standard for cell relay in which multiple service types (such as voice, video, or data) are conveyed in fixed-length (53-byte) cells. Fixed-length cells allow cell processing to occur in hardware, thereby reducing transit delays. ATM is designed to take advantage of high-speed transmission media such as E3, SONET, and T3.

B8ZS---binary 8-zero substitution. Line-code type, used on T1 and E1 circuits, in which a special code is substituted whenever 8 consecutive zeros are sent over the link. This code is then interpreted at the remote end of the connection. This technique guarantees ones density independent of the data stream.

CAS---channel-associated signaling. Trunk signaling (for example, in a T1 line) in which control signals, such as those for synchronizing and bounding frames, are carried in the same channel along with voice and data signals.

CBR---constant bit rate. QoS class defined by the ATM Forum for ATM networks. CBR is used for connections that depend on precise clocking to ensure undistorted delivery.

CCS---common channel signaling. Trunk signaling (for example, using Primary Rate Interface) in which a control channel carries signaling for separate voice and data channels.

CES---circuit emulation service. Enables users to multiplex or concentrate multiple circuit emulation streams for voice and video with packet data on a single high-speed ATM link without a separate ATM access multiplexer.

CO---central office. Local telephone company office to which all local loops in a given area connect and in which circuit switching of subscriber lines occurs.

codec---Coder-decoder. Device that typically uses pulse code modulation to transform analog signals into a digital bit stream and digital signals back into analog.

DTMF---Dual-tone multifrequency. Use of two simultaneous voice-band tones for dialing (such as touch tone).

Drop and Insert---(also called TDM Cross-Connect) Allows DS0 channels from one T1 or E1 facility to be digitally cross-connected to DS0 channels on another T1 or E1. Using this method, channel traffic is sent between a PBX and CO PSTN switch or other telephony device, so that some PBX channels are directed for long-distance service through the PSTN while the router compresses others for interoffice VoIP calls. In addition, Drop and Insert can cross-connect a telephony switch (from the CO or PSTN) to a channel bank for external analog connectivity.

DSP---digital signal processor, same as PVDM.

E1---European digital carrier facility used for transmitting data through the telephone hierarchy. The transmission rate for E1 is 2.048 megabits per second (Mbps).

E&M---rEceive and transMit, or Ear and Mouth. Type of signaling originally developed for analog two-state voltage telephony using the ear and mouth leads; in digital telephony, uses two bits.

ESF---Extended Superframe. Framing type used on T1 circuits that consists of 24 frames of 192 bits each, with the 193rd bit providing timing and other functions. ESF is an enhanced version of SF format.

FXO---Foreign Exchange Office. A voice interface emulating a PBX trunk line to a switch or telephone equipment to a PBX extension interface.

FXS---Foreign Exchange Station. A voice interface for connecting telephone equipment, emulates the extension interface of a PBX or the subscriber interface for a switch.

IETF---Internet Engineering Task Force

ISDN---Integrated Services Digital Network. Communication protocol, offered by telephone companies, that permits telephone networks to carry data, voice, and other source traffic.

IVR---interactive voice response. Term used to describe systems that provide information in the form of recorded messages over telephone lines in response to user input in the form of spoken words or more commonly DTMF signaling. Examples include banks that allow you to check your balance from any telephone and automated stock quote systems.

packet---Logical grouping of information that includes a header containing control information and (usually) user data. Packets are most often used to refer to network layer units of data.

POTS---plain old telephone service

PVDM---packet voice data module

PSTN---Public Switched Telephone Network. General term referring to the variety of telephone networks and services in place worldwide.

QoS---quality of service. Measure of performance for a transmission system that reflects its transmission quality and service availability.

SF---Super Frame. Common framing type used on T1 circuits. SF consists of 12 frames of 192 bits each, with the 193rd bit providing error checking and other functions. SF is superseded by ESF, but is still widely used. Also called D4 framing.

SNMP---Simple Network Management Protocol. Network management protocol used almost exclusively in TCP/IP networks. SNMP provides a means to monitor and control network devices, and to manage configurations, statistics collection, performance, and security.

T1---Digital WAN carrier facility. T1 transmits DS 1-formatted data at 1.544 Mbps through the telephone switching network, using alternate mark inversion or B8ZS coding.

T1 trunk---Digital WAN carrier facility. See T1.

TDM---time-division multiplexing.

Trunk---Physical and logical connection between two switches across which network traffic travels. A backbone is composed of a number of trunks.

UNI---User-Network Interface. ATM Forum specification that defines an interoperability standard for the interface between ATM-based products (a router or an ATM switch) located in a private network and the ATM switches located within the public carrier networks. Also used to describe similar connections in Frame Relay networks.

VAD---voice activity detection.

Posted: Sun May 21 10:36:50 PDT 2000
Copyright 1989 - 2000©Cisco Systems Inc.