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This document describes how to configure digital T1 packet voice trunk network modules on Cisco 2600 and 3600 routers and includes the following sections:

• Feature Overview, page 1
• Supported Platforms, page 10
• Supported Standards, MIBs, and RFCs, page 10
• Prerequisites, page 11
• Configuration Tasks, page 12
• Monitoring and Maintaining T1 Digital Packet Voice Configuration, page 25
• Configuration Examples, page 33
• Command Reference, page 44
• Glossary, page 66

Feature Overview

Digital T1 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 T1 interfaces and can convert that information to a compressed format, so that it can be transmitted as voice over IP.

Cisco IOS software configuration allows you to set up a variety of applications. Here are a few examples:

• Compressed voice over WANs
• Routing of dialed variable-length digits collected from the public switched telephone network or PBX for VoIP calls
• Support for FRF.12 fragmentation and queuing in a VoIP over Frame-Relay network
• Setup of private-line auto-ringdown (PLAR) to allow a station or DS0 to go off hook and have the call completed without dialing (especially applicable to off-premises extensions)
• Transparent trunk connections among routers
• Drop and Insert of T1 channels on a T1 trunk to allow some PBX channels to be directed to the PSTN while others are used for compressed VoIP
  
  For more information about these applications, see “Configuration Examples” on page 33.
  
  T1 digital voice over IP includes the following functionality:
• T1 Channel Associated Signaling (CAS) for the following line-signaling types:
  
  — rEceive and transMit or Ear and Mouth (E&M) immediate start
  
  — E&M wink start
  
  — E&M delay start (also called “dial repeating”)
  
  — Foreign Exchange Station (FXS) and Foreign Exchange Office (FXO) loop start
  
  — FXS and FXO ground start
• Dynamic bandwidth allocation using voice activity detection (VAD)
• Drop-and-Insert capability, allowing the interchange of time-division multiplexing (TDM) slots between the ports on a two-port T1 multiflex trunk voice/WAN interface card installed in a digital T1 packet voice trunk network module
• Support for a wide range of International Telecommunication Union (ITU-T) G-series compression specifications, including:
  
  — G.711 A Law at 64,000 bps
  
  — G.711 u Law at 64,000 bps
  
  — G.723.1 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
• Depending on codec complexity, either 30 or 60 channels of compressed voice
• High-quality voice endpoint-standard features, such as high-quality echo cancellation, silence suppression, comfort noise generation, and DTMF relay
• Group 3 fax relay
• Support for the following framing formats and line coding:
  
  — Super Frame (SF)
  
  — Extended Super Frame (ESF)
  
  — Alternate mark inversion (AMI) line coding
  
  — Binary 8-zero substitution (B8ZS) line coding

Benefits

Digital T1 packet voice trunk network modules allow Cisco 2600 and 3600 series routers to provide T1 connectivity to PBXs or to a central office (CO). With digital T1 connectivity, Cisco 2600 and 3600 series routers can provide greater voice density for enterprise and service provider VoIP networks than they could before. A digital T1 packet voice trunk network module is a complete solution, made up of a network module with installed packet voice data modules (PVDMs), and one T1 multiflex trunk voice/WAN interface card with either one or two T1 ports.

VoIP: T1 Timing, Signaling, Framing, and Line Encoding

With the introduction of the digital T1 packet voice trunk network modules for the Cisco 2600 and 3600 series routers, you must set timing, signaling, framing, and line encoding. The digital T1 packet voice trunk network modules can connect to either a PBX (or similar telephony device) or to a Central Office (CO) in order provide PSTN connectivity.

The differences that set T1 digital configuration apart from analog configuration are as follows:

• Timing. Analog interfaces do not require specific timing configuration. Digital T1 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 T1 interfaces require that you configure either SuperFrame (SF or D4 framing) or Extended SuperFrame (ESF) framing. Set the framing format to match that of the PBX or CO that connects to the digital T1 packet voice trunk network module.
• Line Encoding. Analog interfaces do not require that specific line encoding be configured. Digital T1 require that you configure either AMI (alternative mark inversion) or B8ZS (bipolar 8- zero substitution). Set the line encoding to match that of the PBX or CO that connects to the digital T1 packet voice trunk network module.

Timing

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

The digital T1 module has an on-board PLL (Phase-Lock Loop) chip that can either provide a clock source to both T1s or receive clocking that can drive the second T1. All timing commands are T1 controller configuration commands.

Single T1 Port Provides Clocking
In this scenario, the digital T1 module is the clock source for the connected device. The PLL generates the clock internally and drives the clocking on the T1 line.

Figure 1 Single T1 Port Providing Clock

The following configuration sets up this clocking method:

controller T1 1/0
framing esf
linecoding b8zs
clock source internal
ds0-group 1 timeslots 1-24 type e&m-wink

Note Generally this method is useful only when connecting to a PBX, key system or channel bank. A Cisco VoIP Gateway rarely provides clocking to the CO, because CO clocking provides a higher Stratum level.

Single T1 Port Receiving Clock from the Line
In this scenario, the digital T1 module receives clocking from the connected device (CO or PBX). The PLL clocking is driven by the clock reference on the receive (Rx) side of the T1 connection.

Figure 2 Single T1 Receiving Clock from Line

The following configuration sets up this clocking method:

controller T1 1/0
framing esf
linecoding b8zs
clock source line
ds0-group 1 timeslots 1-24 type e&m-wink

Dual T1s, Both Receive Clocking from the Line
In this scenario, the digital T1 has two reference clocks, one from the PBX and another from the CO. Since 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 underlay the clocking method:

• Looped-Time Clocking. The T1 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 T1 so that the connected device has a viable clock and does not see slips. PBXs are not designed to accept slips on a T1 line and such slips cause a PBX to drop the link into failure mode. While in looped-time mode, the router often sees slips, but because these are controlled slips, they usually do not force failures of the router’s T1 port.
• Slips. These messages indicate that the T1 port is receiving clock information that is out of phase, that is, 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.

Note Physical 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.

Figure 3 Dual T1s Receiving Line Clocking

In this scenario, the PLL derives clocking from the CO and puts the T1 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 T1.

The following configuration sets up this clocking method:

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

Dual T1s, One Receives Clocking and One Provides Clocking
In this scenario, the digital T1 module receives clocking for the PLL from T1 0 and uses this clock as a reference to clock T1 1. If T1 0 fails, the PLL internally generates the clock reference to drive T1 1.

Figure 4 Dual T1s, One Receiving and One Providing Clocking

The following configuration sets up this clocking method:

controller T1 1/0
framing esf
linecoding b8zs
clock source line
ds0-group 1 timeslots 1-24 type e&m-wink
!
controller T1 1/1
framing esf
linecoding b8zs
clock source internal
ds0-group 1 timeslots 1-24 type e&m-wink

Dual T1s, Both Clocks from Router
In this scenario, the router is “Master of the Timing Universe,” generating the clock for the PLL and therefore for both T1s.

Figure 5 Dual T1s, Both Clocks from Router

The following configuration sets up this clocking method:

controller T1 1/0
framing esf
linecoding b8sz
clock source internal
ds0-group 1 timeslots 1-24 type e&m-wink
!
controller T1 1/1
framing esf
linecoding b8zs
clock source internal
ds0-group 1 timeslots 1-24 type e&m-wink

Signaling

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

• Channel-Associated Signaling (CAS). CAS signaling means that instead of having a specific time slot (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 time slot. CAS signaling is often called robbed-bit signaling (RBS) because it takes bits from bearer channels and uses them for signaling. CAS signaling must be specified on both ends of the T1 link and is enabled by default on digital T1 packet voice trunk network modules.
  

  ---

  Note Digital T1 packet voice trunk network modules support T1 CAS at this time, but later plans are to support E1, Primary Rate Interface (PRI), R2, and Common-Channel (CCS) signaling. The digital T1 module 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 or calls are not be connected properly. E&M signaling is defined with the ds0-group controller configuration command, as in the following example:
  
  controller T1 1/0
  ds0-group 1 timeslots 1-24 type e&m-wink-start
  

  ---

  Note Currently, wink-start 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 T1 connections used for switch-to-switch (or switch-to-router) trunks are E&M connections, a digital T1 module can also support FXS and FXO connections, which people normally use to provide emulated-OPX (Off-Premise eXtensions) 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 phone, 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 are not be connected properly. FXS and FXO signaling are defined with the ds0-group controller configuration command, as in the following example:
  
  controller T1 1/0
  ds0-group 1 timeslots 1-24 type fxo-ground-start
  or
  controller T1 1/0
  ds0-group 1 timeslots 1-24 type fxs-loop-start
  

  ---

  Note While 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 may only work in one direction.

  ---

Framing

Digital T1 packet voice trunk network modules support two types of framing for T1 CAS: ESF (Extended SuperFrame) and SF (SuperFrame), also called D4 framing. The framing type of the router and switch (CO or PBX) must match. The framing controller configuration command defines T1 framing, as in the following example:

controller T1 1/0
framing esf

or
controller T1 1/0
framing sf

Line Encoding

Digital T1 packet voice trunk network modules support two types of framing for T1 CAS: B8ZS (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 T1 framing, as in the following example:

controller T1 1/0
linecoding b8zs

or
controller T1 1/0
linecoding ami

Verifying Configuration

Use the show controller privileged EXEC command to verify the proper digital T1 configuration:

router# show controller T1 1/0
T1 1/0 is up.
Applique type is Channelized T1
Cablelength is short 133
Description: Digital T1 WIC
No alarms detected.
Framing is ESF, Line Code is B8ZS, Clock Source is Line Primary.
Data in current interval (2 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

Restrictions

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

• Group 4 fax is not supported.
• The high-density voice network module has one slot for a voice/WAN interface card (VWIC); VWICs supply one or two ports. Only the dual-mode (voice/WAN) multiflex trunk cards are supported in the digital T1 packet voice trunk network module, not older VICs. For more information, see the “Prerequisites” section on page 11.
• Drop-and-Insert capability is supported only between two ports on the same multiflex card.
• Voice over Frame Relay is not supported.
• Wink-start signaling Feature-Group D is not supported.
• 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.
• Digital T1 voice is not manageable through Simple Network Management Protocol (SNMP) using existing versions of Cisco Voice Manager. Release 2.0 of Cisco Voice Manager is planned to support the feature.

Related Features and Technologies

VoIP Quality of Service

This section explains the quality issues that you should consider when building Voice over IP (VoIP) networks and offers a few tips about configuring VoIP with the appropriate Quality of Service (QoS):

• Delay. Delay is the time it takes for VoIP packets to travel between two endpoints and you should design networks to minimize this delay. However, because of the speed of network links and the processing power of intermediate devices, some delay is expected. The human ear normally accepts up to about 150 milliseconds (ms) of delay without noticing problems (the ITU's G.114 standard recommends no more than 150 ms of one-way delay). Once delay exceeds 150 ms, a conversation becomes more and more like a walkie-talkie interchange, where one person must wait for the other to stop speaking before beginning to talk. This type of delay is often evident on international long-distance calls. You can measure delay fairly easily by using ping tests at various times of the day with different network traffic loads. If network delay is excessive, reduce it before deploying VoIP networks.
• Jitter. While delay can cause unnatural starting and stopping of conversations, variable-length delays (also known as jitter) can cause a conversation to break and become unintelligible. Jitter is not usually a problem with public switched telephone network (PSTN) calls, because the bandwidth of calls is fixed. However, in VoIP networks where existing data traffic might be bursty, jitter can become an issue. Cisco voice gateways have built-in de-jitter buffering to compensate for a certain amount of jitter, but if jitter is constant on a network, identify the source and control it before deploying a VoIP network.
• Serialization. Serialization is a term that describes what happens when a router attempts to send both voice and data packets out of an interface. In general, voice packets are very small (80 to 256 bytes), while data packets can be very large (1,500 to 18,000 bytes). On relatively slow links, such as WAN connections, large data packets can take a long time to transmit onto the wire. When these large packets are mixed with smaller voice packets, the excessive transmission time can lead to both delay and jitter. You can use fragmentation to reduce the size of the data packets so that the delay and jitter also decrease.
• Bandwidth Consumption. Traditional voice conversations consume 64 Kb of network bandwidth. When this voice traffic is run though a VoIP network, it can be compressed and digitized by digital signal processors (DSPs) built into the routers. This compression can reduce the calls to sizes as small as 5.3 Kb for voice samples. Once the packets go onto the IP network, the appropriate IP/UDP/RTP headers must be added, and this can add a significant amount of bandwidth to each call (about 40 bytes per packet). Technologies such as RTP header compression, however, can reduce the IP header overhead to about 2 bytes. In addition, VAD (voice activity detection) does not send any packets unless there is active speech.

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 3661
• Cisco 3662

Supported Standards, MIBs, and RFCs

RFCs

• RFC 1890
• RFC 1889
  
  MIBs
• CISCO-ENTITY-VENDORTYPE-OID-MIB
• OLD-CISCO-CHASSIS-MIB
• CAS_INTF_MIB
  
  International Telecommunication Union (ITU-T) G-Series Codec Compression Specifications
• 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

Prerequisites

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

• Obtain T1 service from your service provider or PBX.
• Install Cisco IOS Software Release 12.0(5)XK, 12.0(7)T or a later release. The minimum DRAM memory requirements to support digital T1 packet voice trunk network modules are as follows:
  
  — 32 Mb with one or two T1s
  
  — 48 Mb with three or four T1s
  
  — 64 Mb with five to ten T1s
  
  — 128 Mb with more than ten T1s
  
  The memory required may be greater than listed above for high-volume applications.
  
  Support for digital T1 packet voice trunk network modules is included in Plus feature sets. The IP Plus feature set requires 8 Mb of flash memory; other Plus feature sets require 16 Mb.
• Install one of the following high-density T1 network modules in the router chassis:
  
  — Single-Port 24 Channel T1 High-Density Voice Network Module (NM-HDV-1T1-24)
  
  — Single-Port Enhanced 24 Channel T1 High-Density Voice Network Module (NM-HDV-1T1-24E)
  
  — Dual-Port 48 Channel High-Density Voice Network Module (NM-HDV-2T1-48)
  

  ---

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

  ---

• Install at least one packet voice data module (PVDM-12) in the high-density digital T1 network module if it is not already equipped. The digital T1 packet voice trunk network module contains five 72-pin SIMM sockets or banks, numbered 0 through 4, for PVDMs. Each socket can be filled with a single 72-pin PVDM. A digital T1 packet voice trunk network module can support the following numbers of channels:
  
  — When the digital T1 packet voice trunk network module is configured for high-complexity codec mode, up to six voice or fax calls can be completed per PVDM-12, using the following codecs: G.711, G.726, G.729, G.729 Annex B, G.723.1, G.723.1 Annex A, G.728, and fax relay.
  
  — When the digital T1 packet voice trunk network module is configured for medium-complexity codec mode, up to twelve 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 T1 packet voice trunk network module. Only the one- and two-port T1 multiflex trunk interface cards (VWIC-1MFT-T1, VWIC-2MFT-T1,
  VWIC-2MFT-T1-DI) are supported with channel-associated signaling (CAS).
  
  For Drop-and-Insert capability, you must install a two-port Drop-and-Insert T1 multiflex trunk voice/WAN interface card (VWIC-2MFT-T1-DI). To install a VWIC in a network module, see Cisco WAN Interface Cards Hardware Installation Guide.
• Install at least one other network module orWAN interface card to provide the connection to the IP LAN or WAN.
• Establish a working IP network. For more information about configuring IP, see “IP Overview,” “Configuring IP Addressing,” and “Configuring IP Services” chapters in the Cisco IOS Release 12.0 Network Protocols Configuration Guide, Part 1.
• Complete your company’s dial plan.
• Establish a working telephony network based on your company's dial plan.
  
  Voice, Video, and Home Applications Configuration Guide and Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0 provide information about setting up voice networks.

Configuration Tasks

Perform the following tasks to configure a digital T1 packet voice trunk network module:

• Set up voice cards and T1 controllers.
• Configure serial and LAN interfaces.
• Set up voice ports.
• Configure voice dial peers.

Configuring Voice Card and T1 Controller Settings

The following steps specify codec settings for voice cards and set up T1 controllers for clocking and other T1 parameters, as well as for DS0 groups that define the channels for compressed voice and TDM groups for Drop-and-Insert capability.

Verifying Voice Card and Controller Settings

To verify the configuration of voice card and controller settings, follow these steps:

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 12.0 Cisco IOS Interface Configuration Guide and the Cisco IOS Interface Command Reference.

The “Configuration Examples” section on page 33 shows a sample configuration that sets up VoIP over Frame Relay. For more information about setting up voice networks, see Voice, Video, and Home Applications Configuration Guide for Cisco IOS Release 12.0.

Note For 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” on page 20.

Verifying Serial Interface Configuration

To verify serial interface configuration, enter the privileged EXEC command showinterfaces serial, 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 Voice, Video, and Home Applications Configuration Guide and Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.

Verifying Voice Ports

Follow the procedure below to verify voice-port configuration. To learn more about these commands, see Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.

Important command output is shown in bold.

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 after the “<<“ characters:

cisco-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
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 Voice, Video, and Home Applications Configuration Guide and Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.

Verifying Voice Dial Peers

Follow the procedure below to verify dial-peer configuration. To learn more about these commands, see Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.

Important command output is shown in bold.

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

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 T1 Digital Packet Voice Configuration

This section presents some useful show and debugging commands for understanding, maintaining, and troubleshooting your configuration.

The balance of this section shows the output of the commands listed in Table 1.

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, and bold text preceded by the “<<“ characters explains the process.

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.

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

cisco-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