Cisco Systems AS5800 User Manual 3

Voice over IP for the Cisco AS5800

The Voice over IP for the Cisco AS5800 feature adds Voice over IP carrier-class gateway functionality to the Cisco AS5800 platform. This document contains the following sections:

• Feature Overview, page 1

• Supported Platforms, page 5

• Supported Standards, MIBs, and RFCs, page 5

• Prerequisites, page 6

• Configuration Tasks, page 7

• Configuration Example, page 19

• Command Reference, page 21

Feature Overview

Voice over IP (VoIP) enables a Cisco AS5800 universal access server to provide voice and fax traffic, such as telephone calls and faxes, over an IP network. There are basically two different environments in which VoIP can be deployed: enterprise and service provider. Different strategies have been developed for deploying VoIP in both of these environments. The Cisco AS5800 universal access server can be configured for deployment in either an enterprise or a service provider environment but, because of the extensive capabilities of the Cisco AS5800 universal access server, it is more likely that it will function as a carrier class gateway in a service provider environment. This document, then, describes how to configure the Cisco AS5800 universal access server to act as a carrier class gateway in your VoIP network. To configure the Cisco AS5800 universal access server to perform in an enterprise environment, refer to the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module. The configuration steps for both the Cisco AS5300 access server and the Cisco AS5800 universal access server for an enterprise environment are identical.

Voice over IP in either the service provider or enterprise environment is primarily a software feature; however, to use this feature on the Cisco AS5800, you must install a VoIP feature card (VFC). The VFC uses the Cisco AS5800’s T1/E1 and T3 Public Switched Telephone Network (PSTN) interfaces and local-area network (LAN) or wide-area network (WAN) routing capabilities to provide up to a 192 ports or channels (per VFC card) for VoIP packetized voice traffic.

Voice over IP for the Cisco AS5800 1

Feature Overview

Benefits

Two-Stage-Dial Toll Bypass

With Voice over IP on the Cisco AS5800, you can leverage your network’s WAN infrastructure to offer long distance toll bypass services. Toll bypass occurs in two stages. For example, customers can be assigned an account number and a Personal Identification Number (PIN). When a user dials a local number or a 1-800-Internet Telephone Service Provider (ITSP) number, she connects to the local VoIP point of presence. She is then prompted by the Interactive Voice Response (IVR) to input her account and PIN numbers. Following authentication, a second dial tone allows her to enter an E.164 destination telephone number.

The local gatekeeper maps the E.164 destination telephone number to an IP address of a remote-zone gatekeeper, which then selects a gateway to terminate the call. The gateway encodes the call, encapsulates it in Real Time Protocol (RTP) packets and routes it over the WAN to the remote gateway. The remote gateway decodes the call and delivers it to the receiver.

For information about configuring IVR, refer to the Cisco IOS Release 12.0(7)T Configuring Interactive Voice Response for Cisco Access Platforms feature module.

Figure 1 illustrates this benefit.

Cisco IOS Release 12.0(7)T

Figure 1 Two-Stage Dial Toll Bypass

Benefits

Gatekeeper

Zone A

Cisco

gatekeeper

RAS RAS

Cisco

gateway

Cisco

IP/PSTN

gateway

PRI PRI

Fax

server

QoS WAN

H.225/H.245 RTP

PRI

PSTN

RAS

gatekeeper

Cisco

gateway

PRI

Digital sound

voice mail

Gatekeeper

Zone B

Cisco

IP/PSTN

gateway

PSTN Voice-Traffic and Fax-Traffic Off load

Carriers can leverage their WAN infrastructure to off load voice and fax traffic from their congested PSTN networks by using the Cisco AS5800 as a carrier class voice gateway. In this application, PSTN traffic designated to be off-loaded is forwarded to a tandem switch connected to the Cisco AS5800 gateway. The AS5800 gateway then encapsulates the off-loaded PSTN traffic into RTP streams and routes it across the WAN.

The signaling interface between the PSTN and the Cisco AS5800 can be either Common Channel Signaling (CCS), with SS7 terminated by the VCO-4K service point or Channel Associated Signaling (CAS), configured in Direct Inward Dial (DID) mode. Figure 2 illustrates this application.

13342

Voice over IP for the Cisco AS5800 3

Feature Overview

Figure 2 VoIP Used as a PSTN Gateway to Off load Voice Traffic and Fax Traffic

Local Exchange

Carrier

T1 ISDN

PRI

Cisco AS5800

Universally Accessible Voice-Mail and Fax-Mail Services

VoIP on the Cisco AS5800 can be used to leverage the technology prefixes feature. Gateways (with voice/fax feature cards) that are connected to the voice-mail and fax-mail servers can be identified by gatekeepers based on a prefix prepended to an E.164 telephone number.

Additional Benefits

VoIP on the Cisco AS5800 can be used to provide the following additional benefits:

WAN

Local Exchange

Carrier

T1 ISDN

PRI

Cisco AS5800

• Remote PBX presence over WANs

• POTS-Internet telephony gateways

30744

Restrictions

To run Voice over IP on the Cisco AS5800, the AS5800 must have a version of the Cisco IOS software installed that supports DSDWare 3.1.7 (for example, Cisco IOS Release 12.0(4)XL or Cisco IOS Release 12.0(7)T).

Related Features and Technologies

• Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module

• Cisco IOS Release 12.0(3)T Service Provider Features for Voice over IP feature module

• Cisco IOS Release 12.0(5)T IP RTP Priority feature module

• Cisco IOS Release 12.0(7)T Configuring Interactive Voice Response for Cisco Access Platforms

feature module

Outbound Dialing on POTS Peers

When a router receives a voice call, it selects an outbound dial peer by comparing the called number (the full E.164 telephone number) in the call information with the number configured as the destination pattern for the POTS peer. The router then strips out the left-justified numbers corresponding to the destination pattern matching the called number. If you have configured a prefix, the prefix will be put in front of the remaining numbers, creating a dial string, which the router will then dial. If all numbers in the destination pattern are stripped-out, the user will receive (depending on the attached equipment) a dial tone.

For example, suppose there is a voice call whose E.164 called number is 1 310 767-2222. If you configure a destination-pattern of “1310767” and a prefix of “9,” the router will strip out “1310767” from the E.164 telephone number, leaving the extension number of “2222.” It will then append the

Voice over IP for the Cisco AS5800 13

Configuration Tasks

prefix, “9,” to the front of the remaining numbers, so that the actual numbers dialed is “9, 2222.” The comma in this example means that the router will pause for one second between dialing the “9” and the “2” to allow for a secondary dial tone.

Direct Inward Dial for POTS Peers

Direct inward dial (DID) is used to determine how the called number is treated for incoming POTS call legs. As shown in Figure 5, incoming means from the perspective of the router. In this case, it is the call leg coming into the access server to be forwarded through to the appropriate destination pattern.

Figure 5 Incoming and Outgoing POTS Call Legs

PBX

Unless otherwise configured, when a call arrives on the access server, the server presents a dial tone to the caller and collects digits until it can identify the destination dial peer. After the dial peer is identified, the call is forwarded through the next call leg to the destination.

There are cases where it might be necessary for the server to use the called-number (DNIS) to find a dial peer for the outgoing call leg—for example, if the switch connecting the call to the server has already collected the digits. DID enables the server to match the called-number with a dial peer and then directly place the outbound call. With DID, the server does not present a dial tone to the caller and does not collect digits; it forwards the call directly to the configured destination.

To use DID and incoming called-number, a dial peer must be associated with the incoming call leg. Before doing this, it helps if you understand the logic behind the algorithm used to associate the incoming call leg with the dial peer.

The algorithm used to associate incoming call legs with dial peers uses three inputs (which are derived from signaling and interface information associated with the call) and four defined dial peer elements. The three signaling inputs are:

Cisco AS5800

Incoming

call leg

cloud

Outgoing

call leg

Cisco AS5800

PBX

22356

• Called-number (DNIS)—Set of numbers representing the destination, which is derived from the

ISDN setup message or CAS DNIS.

Cisco IOS Release 12.0(7)T

• Calling-number (ANI)—Set of numbers representing the origin, which is derived from the ISDN

setup message or CAS DNIS.

• Voice port—The voice port carrying the call.

The four defined dial peer elements are:

• Destination pattern—A pattern representing the phone numbers to which the peer can connect.

• Answer address—A pattern representing the phone numbers from which the peer can connect.

• Incoming called-number—A pattern representing the phone numbers that associate an incoming

call leg to a peer based on the called-number or DNIS.

• Port—The port through which calls to this peer are placed.

Using the elements, the algorithm is as follows:

For all peers where call type (VoIP versus POTS) match dial peer type: if the type is matched, associate the called number with the incoming called-number

else if the type is matched, associate calling-number with answer-address else if the type is matched, associate calling-number with destination-pattern else if the type is matched, associate voice port to port

This algorithm shows that if a value is not configured for answer-address, the origin address is used because, in most cases, the origin address and answer-address are the same.

To configure a POTS dial peer for direct inward dial, use the following commands beginning in global configuration mode:

Step Command Purpose

Router(config)# dial-peer voice number pots

Router(config-dial-peer)# direct-inward-dial

Note Direct inward dial is configured for the calling POTS dial peer.

Enters the dial peer configuration mode to configure a POTS peer.

Specifies direct inward dial for this POTS peer.

Configuring Dial Peers

Distinguishing Voice and Modem Calls on the Cisco AS5800

When the Cisco AS5800 is handling both modem and voice calls, it needs to be able to identify the service type of the call—that is, whether or not the incoming call to the server is a modem or a voice call. When the access server handles only modem calls, the service type identification is handled through modem pools. Modem pools associate calls with modem resources based on the called-number (DNIS). In a mixed environment, where the server receives both modem and voice calls, you need to identify the service type of a call by using the incoming called-number command.

Without this, the server attempts to resolve whether an incoming call is a modem or voice call based on the interface over which the call comes. If the call comes in over an interface associated with a modem pool, the call is assumed to be a modem call; if a call comes in over a voice port associated with a dial peer, the call is assumed to be a voice call.

It helps to understand the logic behind the algorithm the system uses to distinguish voice and modem calls. The algorithm is as follows:

If the called-number matches a number from the modem pool,

handle the call as a modem call

If the called-number matches a configured dial peer incoming called number,

handle the call as a voice call

Else handle the call as a modem call by default modem pool

If there is no called-number information configured, call classification is handled as follows:

If the interface matches the interface configured for the modem pool,

handle the call as a modem call.

If the voice port matches the one configured as the dial peer port,

handle the call as a voice call

Else handle the call as a modem call by default modem pool

To identify the service type of a call to be voice, use the following commands beginning in global configuration mode:

Voice over IP for the Cisco AS5800 15

Configuration Tasks

Step Command Purpose

Router(config)# dial-peer voice

Router(config-dial-peer)# incoming called-number

number

number pots

Enter the dial peer configuration mode to configure a POTS peer.

Specify direct inward dial for this POTS peer.

Configuring VoIP Peers

VoIP peers enable outgoing calls to be made from a particular telephony device. To configure a VoIP peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its destination telephone number and destination IP address. As with POTS peers, under most circumstances, the default values for the remaining dial peer configuration commands will be adequate to establish connections.

To configure a VoIP peer, use the following commands beginning in global configuration mode:

Step Command Purpose

Router(config)# dial-peer voice number voip

Router(config-dial-peer)# destination-pattern string

Router(config-dial-peer)# tech-prefix number

Router(config-dial-peer)# session-target {ipv4:destination-address | dns:[$s$.|$d$.|$e$.|$u$.]

host-name|loopback:rtp|loopback:compressed| loopback:unompressed|ras}

Enters the dial peer configuration mode to configure a VoIP peer. The number value of the dial-peer voice voip command is a tag that uniquely identifies the dial peer.

Defines the destination telephone number associated with this VoIP dial peer.

Specifies that a particular technology prefix be prepended to the destination patter of this dial peer.

Specifies a destination IP address for this dial peer.

For additional VoIP dial peer configuration options, refer to the “Voice-Related Commands” section of the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference, the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module, and the Cisco IOS Release

12.0(3)T Service Provider Features for Voice over IP feature module.

Verifying Dial Peer Configuration

If you have relatively few dial peers configured, you can use the show dial-peer voice command

•

to verify that the data configured is correct. Use this command to display a specific dial peer or to display all configured dial peers.

• Use the show dialplan number command to show the dial peer to which a particular number

(destination pattern) resolves.

Troubleshooting Tips

• Ping the associated IP address to confirm connectivity. If you cannot successfully ping your

destination, refer to the chapter, “Configuring IP,” in the Cisco IOS 11.3 Network Protocols Configuration Guide, Part 1.

• Use the show dial-peer voice command to verify that the operational status of the dial peer is up.

Cisco IOS Release 12.0(7)T

Configuring the Cisco AS5800 as an H.323 Gateway

• Use the show dialplan number command on the local and remote routers to verify that the data

is configured correctly on both.

• If you have configured number expansion, use the show num-exp command to check that the

partial number on the local router maps to the correct full E.164 telephone number on the remote router.

• If you have configured a CODEC value, there can be a problem if both VoIP dial peers on either

side of the connection have incompatible CODEC values. Make sure that both VoIP peers have been configured with the same CODEC value.

• Use the debug voip ccani inout command to verify the output string the router dials is correct.

• Use the debug cch323 rtp command to check RTP packet transport.

• Use the debug cch323 h245 command to check logical channel negotiation.

• Use the debug cch323 h225 command to check the call setup.

Configuring the Cisco AS5800 as an H.323 Gateway

The Service Provider implementation of Voice over IP uses both gatekeepers and gateways. Because of the extensive capabilities of the Cisco AS5800 universal access server, it is likely that it will function as a carrier class gateway in a Service Provider environment. The final step in configuring the Cisco AS5800 for Voice over IP functionality is to configure one of its interfaces as a gateway interface. You can use either an interface that is connected to the gatekeeper or a loopback interface for the gateway interface. The interface that is connected to the gatekeeper is usually a LAN interface—Fast Ethernet, Ethernet, FDDI, or Token Ring.

To configure a gateway interface, perform the following steps beginning in the global configuration mode:

Step Command Purpose

3 Configure the interface. This step will vary, depending on

Router(config)# gateway

Router(config)# ip cef

Router(config)# int fa0

Router(config-if)# h323-gateway voip interface

Router(config-if)# h323-gateway voip id gatekeeper-id {ipaddr ip-address [port-number]|multicast}

Router(config-if)# h323-gateway voip h323-id

interface-id

Router(config-if) h323-gateway voip tech-prefix prefix

Enables the gateway.

Enables Cisco Express Routing.

the interface you select as being the interface connected to the gatekeeper. For the purposes of this procedure, a Fast Ethernet interface is used.

Enters configuration mode for the configured Fast Ethernet interface connected to the gatekeeper.

Identifies this interface as a VoIP gateway interface.

Defines the name and location of the gatekeeper for this gateway.

Defines the H.323 name of the gateway, identifying this gateway to its associated gatekeeper.

Defines the technology prefix that the gateway will register with the gatekeeper.

For more information about configuring gateways and gatekeepers, refer to the Cisco IOS Release

12.0(3)T Service Provider Features for Voice over IP feature module.

Voice over IP for the Cisco AS5800 17

Configuration Tasks

Verifying Gateway Interface Configuration

Use the show gateway command to find the current registration information and status of the gateway.

Configuring the Cisco AS5800 for Interactive Voice Response

The Interactive Voice Response (IVR) Service Provider application provides IVR capabilities using Tool Command Language (TCL) scripts. For example, an IVR script is played when a caller receives a voice-prompt instruction to enter a specific type of information, such as a PIN. After playing the voice prompt, the IVR application collects the predetermined number of touch tones (digit collection) and forwards the collected digits to a server for storage and retrieval. Call records can be kept, and a variety of accounting functions performed.

Available IVR Scripts

The following is a description of the available IVR scripts:

• fax_hop_on_1—Collects digits from the redialer, such as account number and destination

number. When placing the call to the H.323 network, the set of fields configured in the call information structure are entered, destination, and account.

• clid_authen—Authenticates the call with Automatic Number Identification (ANI) and Dialed

• clid_authen_npw—Same as clid_authen, but uses a null password when authenticating, rather

• clid_authen_collect—Authenticates the call with ANI and DNIS and collects the destination

• clid_authen_col_npw—Same as clid_authen_collect, but uses a null password and does not

• clid_col_npw_3—Same as clid_authen_col_npw except if authentication with the digits

Configuring IVR

To use IVR with scripts, you need to configure the inbound POTS dial peer to support IVR, as well as enable IVR functionality by using the call application global configuration. To configure IVR, use the following commands beginning in the global configuration mode:

Number Identification Service (DNIS), collects the destination data, and makes the call.

than DNIS.

data, but if authentication fails, it collects the account and password.

use or collect DNIS.

collected (account and PIN number) failed, the script clid_authen_col_npw just played a failure message (auth_failed.au) and then hung up. This script, clid_col_npw_3 allows two failures, then plays the retry audio file (auth_retry.au) and collects the account and PIN numbers again

The caller can interrupt the message by entering digits for the account number which will trigger the prompt to enter the PIN number. If authentication fails the third time, the script plays the audio file auth_fail_final.au, then hangs up.

Step Command Purpose

Router (config)# call application voice name

Router(config)# dial-peer voice number pots

Router(config-dial-peer)# application name

Cisco IOS Release 12.0(7)T

Creates and then calls the application that interacts with the IVR feature.

Enters the dial peer configuration mode to configure a POTS peer.

Selects an IVR session application for the dial peer to use.

Step Command Purpose

Router(config-dial-peer)# destination-pattern string

Router(config-dial-peer)# port shelf/slot/port:D

Router(config-dial-peer)# prefix string

Defines the telephone number associated with this POTS dial peer.

Associates this POTS dial peer with a specific logical dial interface.

For more information about configuring IVR, refer to the Cisco IOS Release 12.0(7)T Configuring Interactive Voice Response for Cisco Access Platforms feature module.

Verifying IVR Configuration

• If you have relatively few dial peers configured, you can use the show dial-peer voice command

to verify that the data configured is correct. Use this command to display a specific dial peer or to display all configured dial peers.

• Use the show running configuration command to show all configured parameters relating to

IVR.

Configuration Example

The following configuration example shows an abbreviated configuration using a Cisco 2600 router and a CiscoAS5800 universal access server as gateways and a Cisco 3600 router as a gatekeeper. Figure 6 shows the network diagram for this particular scenario.

Figure 6 AS5800 Universal Access Server Acting as a Gateway

Cisco 2600

10BASE-T

AS5800 VoIP

H.323 gateway

5000

Catalyst

5000

Cisco 3640 gatekeeper

100BASE-T

10BASE-T

NT Server

Cisco CallManage

10BASE-T

30460

Voice over IP for the Cisco AS5800 19

Configuration Example

Configuring the Cisco 3640 as a Gatekeeper

! Configure the Ethernet interface to be used at the gatekeeper interface. interface Ethernet0/1

ip address 172.30.00.00 255.255.255.0 no ip directed-broadcast no logging event link-status

no keepalive ! ! Configure the gatekeeper interface and enable the interface. gatekeeper

zone local gk3.gg-dn1 gg-dn1 173.50.00.00

zone prefix gk3.gg-dn1 21*

gw-type-prefix 9#* gw ipaddr 173.60.0.0 1720

gw-type-prefix 6#* gw ipaddr 173.60.0.199 1720

no use-proxy gk3.gg-dn1 default inbound-to terminal

no shutdown !

Configuring the Cisco 2600 as a Gateway

! Configure POTS and VoIP dial peers. dial-peer voice 88 voip

destination-pattern 11111

tech-prefix 9#

session ras ! dial-peer voice 11 pots

incoming called-number 11111

destination-pattern 6#12345 port 1/1/1 prefix 12345 ! ! Configure the gateway interface. interface Ethernet0/0 ip address 173.60.0.199 255.255.255.0 no ip directed-broadcast no ip mroute-cache no logging event link-status no keepalive no cdp enabled h323-gateway voip interface h323-gateway voip id gk3.gg-dn1 ipaddr 173.30.0.0 1719 h323-gateway voip h323-id gw6@gg-dn1 h323-gateway voip tech-prefix 6# !

Configuring the Cisco AS5800 as a Gateway

! Configure the T1 controller. (This configuration is for a T3 card.) controller T1 1/0/0:1

framing esf linecode b8zs

pri-group timeslots 1-24 ! ! Configure POTS and VoIP dial peers. dial-peer voice 11111 pots

incoming called-number 12345

destination-pattern 9#11111

direct-inward-dial

port 1/0/0:1:D

prefix 11111

Cisco IOS Release 12.0(7)T

Command Reference

! dial-peer voice 12345 voip

destination-pattern 12345

tech-prefix 6#

session target ras ! ! Enable gateway functionality. gateway ! ! Enable Cisco Express Forwarding. ip cef ! ! Configure and enable the gateway interface. interface FastEthernet0/3/0

ip address 173.60.0.0.255.255.255.0

no ip directed-broadcast

no keepalive

full-duplex

no cdp enable

h323-gateway voip interface

h323-gateway voip id gk3.gg-dn1 ipaddr 173.30.0.0 1719

h323-gateway voip h323-id gw3@gg-dn1

h323-gateway voip tech-prefix 9# ! ! Configure the serial interface.(This configuration is for a T3 serial interface.) interface Serial1/0/0:1:23

no ip address

no ip directed-broadcast

ip mroute-cache

isdn switch-type primary-5ess

isdn incoming-voice modem

no cdp enable

Command Reference

This section documents new or modified commands. All other commands used with this feature are documented in one of the following Cisco IOS documentation:

• Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference

• Cisco IOS Release 12.0 Dial Solutions Command Reference

• Cisco IOS Release 12.0(3)T Voice over IP for the AS5300 feature module

• Cisco IOS Release 12.0(3)T Service Provider Features for Voice over IP feature module

• Cisco IOS Release 12.0(7)T Configuring Interactive Voice Response for Cisco Access Platforms

feature module

New Commands

• dtmf-relay

• show vrm vdevice

• show vrm active_calls

• test vrm busyout

• test vrm reset

• test vrm unbusyout

Voice over IP for the Cisco AS5800 21

Command Reference

Modified Commands

• codec

• port

• show csm

• show voice port

• voice-port

In Cisco IOS Release 12.0(1)T or later, you can search and filter the output for show and more commands. This functionality is useful when you need to sort through large amounts of output, or if you want to exclude output that you do not need to see.

To use this functionality, enter a show or more command followed by the “pipe” character (|), one of the keywords begin, include, or exclude, and an expression that you want to search or filter on:

command | {begin | include | exclude} regular-expression

Following is an example of the show atm vc command in which you want the command output to begin with the first line where the expression “PeakRate” appears:

show atm vc | begin PeakRate

For more information on the search and filter functionality, refer to the Cisco IOS Release 12.0(1)T feature module titled CLI String Search.

Cisco IOS Release 12.0(7)T

codec

Syntax Description

codec

To specify the voice coder rate of speech for a dial peer, use the codec dial-peer configuration command. To restore the default voice coder rate of speech value, use the no form of this command.

codec {g711alaw | g711ulaw | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 |

g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmfr}

no codec

g711alaw G.711 A-Law 64000 bits per second (bps).

g711ulaw G.711 u-Law 64000 bps.

g723r53 G.723.1 5300 bps.

g723r63 G.723.1 6300 bps.

g726r16 G.726 16000 bps.

Defaults

Command Modes

g726r24 G.726 24000 bps.

g726r32 G.726 32000 bps.

g728 G.728 16000 bps.

g729abr8 G.729 ANNEX-A & B 8000 bps.

g729ar8 G.729 ANNEX-A 8000 bps.

g729br8 G.729 ANNEX-B 8000 bps.

g729r8 G.729 8000 bps.

gsmfr GSMFR 13200 bps.

g729r8.

Dial-peer configuration

Voice over IP for the Cisco AS5800 23

Command Reference

Command History

Usage Guidelines

Examples

Release Modification

11.3(1)T This command was introduced.

11.3(3)T Support for Cisco 2600 series routers was added.

12.0(3)T Support for the Cisco AS5300 access server was added.

12.0(7)T Additional voice coder rates of speech were added.

For toll quality, use the g711alaw or g711ulaw values. These values provide high-quality voice transmission but use a significant amount of bandwidth. For almost toll quality (and a significant savings in bandwidth), use the g729r8 value.

If codec values for the VoIP peers of a connection do not match, the call will fail.

This command is only applicable to VoIP peers.

The following example configures a voice coder rate that provides toll quality but uses a relatively high amount of bandwidth:

dial-peer voice 10 voip

codec g711alaw

Related Commands

Command Description

dtmf-relay Specifies how an H.323 gateway relays DTMF tones between telephony

interfaces and an IP network.

Cisco IOS Release 12.0(7)T

dtmf-relay

Syntax Description

dtmf-relay

To specify how an H.323 gateway relays dual tone multifrequency (DTMF) tones between telephony interfaces and an IP network, use the dtmf-relay dial-peer configuration command. To remove all signaling options and transmit the DTMF tones as part of the audio stream, use the no form of this command.

dtmf-relay [cisco-rtp] [h245-alphanumeric] [h245-signal]

no dtmf-relay

cisco-rtp (Optional) Forwards DTMF tones by using RTP protocol with a

Cisco proprietary payload type.

h245-alphanumeric (Optional) Forwards DTMF tones by using the H.245

“alphanumeric” User Input Indication method. Supports tones 0-9, *, #, and A-D.

h245-signal (Optional) Forwards DTMF tones by using the H.245 “signal”

User Input Indication method. Supports tones 0-9, *, #, and A-D.

Defaults

Command Modes

Command History

Usage Guidelines

No default behavior or values.

Dial-peer configuration

Release Modification

12.0(7)T This command was introduced.

DTMF is the tone generated when you press a digit on a touch-tone phone. This tone is compressed at one end of a call; when the tone is decompressed at the other end, it can become distorted, depending on the codec used. The DTMF relay feature transports DTMF tones generated after call establishment out of band using a standard H.323 out-of-band method and a proprietary RTP-based mechanism.

The gateway sends DTMF tones in the format you specify only if the remote device supports it. If the remote device supports multiple formats, the gateway chooses the format based on the following priority:

• cisco-rtp (highest priority)

• none, meaning that the DTMF is sent in-band

Voice over IP for the Cisco AS5800 25

Command Reference

Examples

The principal advantage of the dtmf-relay command is that it transmits DTMF tones with greater fidelity than is possible in-band for most low-bandwidth CODECs, such as G.729 and G.723. Without the use of DTMF relay, calls established with low-bandwidth CODECs may have trouble accessing automated DTMF-based systems, such as voice-mail, menu-based ACD systems, and automated banking systems.

Note The cisco-rtp option of the dtmf-relay command is a proprietary Cisco implementation and

only operates between two Cisco AS5800 universal access servers running Cisco IOS Release

12.0(2)XH, or between Cisco AS5800 universal access servers or Cisco 2600 or 3600 modular access routers running Cisco IOS Release 12.0(2)XH or later releases. Otherwise, the DTMF relay feature does not function, and the gateway sends DTMF tones in-band.

The following example configures DTMF relay with the cisco-rtp option when sending DTMF tones to dial-peer 103:

5800# configure terminal 5800(config)# dial-peer voice 103 voip 5800(config-dial-peer)# dtmf-relay cisco-rtp 5800(config-dial-peer)# end 5800#

Related Commands

The next example configures the gateway to send DTMF in-band (the default) when sending DTMF tones to dial-peer 103:

5800# configure terminal 5800(config)# dial-peer voice 103 voip 5800(config-dial-peer)# no dtmf-relay 5800(config-dial-peer)# end

Command Description

codec

Specifies the voice coder rate of speech for a dial peer.

Cisco IOS Release 12.0(7)T

port

To associate a dial peer with a specific voice port, use the port dial peer configuration command. To cancel this association, use the no form of this command.

Cisco 2600/3600 Series Router

port slot/subunit/port

no port

Cisco MC3810

port slot/port

no port

Cisco AS5300 Access Server

port controller number:D

no port

Syntax Description

Cisco AS5800 Access Server

port {shelf/slot/port:D} | {shelf/slot/parent:port:D}

no port

controller number:D Specifies the T1 or E1 controller; :D indicates the D channel associated

with ISDN PRI. Valid entries for the controller number variable is 0 to 3.

shelf/slot/port:D Specifies the T1 or E1 controller on the T1 card; :D indicates the

D-channel associated with ISDN PRI. Valid entries for the shelf variable is 0 to 9999. Valid entries for the slot variable is 0 to 11. Valid entries for the port variable is 0 to 11.

shelf/slot/parent:port:D Specifies the T1 controller on the T3 card; :D indicates the D-channel

associated with ISDN PRI. Valid entries for the shelf variable is 0 to

9999. Valid entries for the slot variable is 0 to 11. Valid entries for the

port variable is 1 to 28. The value for the parent variable is always 0.

port Specifies the voice port number. Valid entries are 0 or 1.

slot Specifies the slot number where the voice interface card is installed.

Valid entries are 0 or 1.

Default

subunit Specifies the subunit on the voice interface card in the router where the

voice port is located. Valid entries are 0 or 1.

No port is configured.

Voice over IP for the Cisco AS5800 27

Command Reference

Command Mode

Command History

Usage Guidelines

Dial-peer configuration

Release Modification

11.3(1)T This command was introduced (Cisco 3600 series router).

11.3(3)T Port-specific values for the Cisco 2600 were added.

11.3 MA Port-specific values for the Cisco MC3810 were added.

12.0(3)T Port-specific values for the Cisco AS5300 were added.

12.0(7)T Port-specific values for the Cisco AS5800 were added.

This command is used for calls incoming from a telephony interface to select an incoming dial peer and for calls coming from the VoIP network to match a port with the selected outgoing dial peer.

This command applies only to POTS peers.

Example

The following example associates a Cisco 3600 series router POTS dial peer 10 with voice port 1, which is located on subunit 0, and accessed through port 0:

dial-peer voice 10 pots

port 1/0/0

The following example associates a Cisco MC3810 POTS dial peer 10 with voice port 0, which is located in slot 1:

dial-peer voice 10 pots

port 1/0

The following example associates a Cisco AS5300 POTS dial peer 10 with voice port 0:D:

dial-peer voice 10 pots

port 0:D

The following example associates a Cisco AS5800 POTS dial peer 10 with voice port 1/0/0:D (T1 card):

dial-peer voice 10 pots

port 1/0/0:D

The following example associates a Cisco AS5800 POTS dial peer 10 with voice port 1/0/0:1:D (T3 card):

dial-peer voice 10 pots

port 1/0/0:1:D

Cisco IOS Release 12.0(7)T

show csm

Syntax Description

show csm

To display the call switching module (CSM) statistics for a particular or all DSP channels or for a specific modem or DSP channel, use the show csm privileged EXEC command.

Cisco AS5300 Access Server

show csm {modem [slot/port | modem-group-number] | voice [slot/dspm/dsp/dsp-channel]}

Cisco AS5800 Universal Access Server

show csm voice [shelf/slot/port]

modem Specifies CSM call statistics for modems.

voice Specifies CSM call statistics for DSP channels.

slot/port (Optional) Specifies the location (and thereby the identity) of a specific

modem.

Defaults

Command Modes

Command History

modem-group-number (Optional) Displays configuration for the dial peer identified by the

argument number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767.

slot/dspm/dsp/dsp-channel (Optional) Identifies the location of a particular DSP channel.

shelf/slot/port (Optional) Identifies the location of the voice interface card.

No default behavior or values.

Privileged EXEC

Release Modification

11.3 NA This command was introduced.

12.0(3)T Port-specific values for the Cisco AS5300 were added.

12.0(7)T Port-specific values for the Cisco AS5800 were added.

Usage Guidelines

This command shows the information related to CSM, which includes the DSP channel, the start time of the call, the end time of the call, and the channel on the controller used by the call.

Voice over IP for the Cisco AS5800 29

Command Reference

Examples

Use the show csm modem command to display the CSM call statistic information for a specific modem, for a group of modems, or for all modems. If a slot/port argument is specified, then CSM call statistics are displayed for the specified modem. If the modem-group-number argument is specified, the CSM call statistics for all of the modems associated with that modem group are displayed. If no keyword is specified, CSM call statistics for all modems on the AS5300 are displayed.

Use the show csm voice command to display CSM statistics for a particular DSP channel. If the slot/dspm/dsp/dsp-channel or shelf/slot/port argument is specified, the CSM call statistics for calls using the identified DSP channel will be displayed. If no argument is specified, all CSM call statistics for all DSP channels will be displayed.

The following is sample output from the Cisco AS5300 for the show csm voice command:

Router# show csm voice 2/4/4/0 slot 2, dspm 4, dsp 4, dsp channel 0, slot 2, port 56, tone, device_status(0x0002): VDEV_STATUS_ACTIVE_CALL.

csm_state(0x0406)=CSM_OC6_CONNECTED, csm_event_proc=0x600E2678, current call thru PRI line invalid_event_count=0, wdt_timeout_count=0 wdt_timestamp_started is not activated wait_for_dialing:False, wait_for_bchan:False pri_chnl=TDM_PRI_STREAM(s0, u0, c22), tdm_chnl=TDM_DSP_STREAM(s2, c27) dchan_idb_start_index=0, dchan_idb_index=0, call_id=0xA003, bchan_num=22 csm_event=CSM_EVENT_ISDN_CONNECTED, cause=0x0000 ring_no_answer=0, ic_failure=0, ic_complete=0 dial_failure=0, oc_failure=0, oc_complete=3 oc_busy=0, oc_no_dial_tone=0, oc_dial_timeout=0 remote_link_disc=0, stat_busyout=0 oobp_failure=0 call_duration_started=00:06:53, call_duration_ended=00:00:00, total_call_duration=00:00:44 The calling party phone number = 408 The called party phone number = 5271086 total_free_rbs_timeslot = 0, total_busy_rbs_timeslot = 0, total_dynamic_busy_rbs_timeslot = 0, total_static_busy_rbs_timeslot = 0, total_sw56_rbs_timeslot = 0, total_sw56_rbs_static_bo_ts = 0, total_free_isdn_channels = 21, total_busy_isdn_channels = 0,total_auto_busy_isdn_channels = 0, min_free_device_threshold = 0

Cisco IOS Release 12.0(7)T

The following is sample output from the Cisco AS5800 for the show csm voice command:

5800# show csm voice 1/8/19 shelf 1, slot 8, port 19 VDEV_INFO:slot 8, port 19 vdev_status(0x00000401):VDEV_STATUS_ACTIVE_CALL.VDEV_STATUS_HASLOCK. csm_state(0x00000406)=CSM_OC6_CONNECTED, csm_event_proc=0x60868B8C, current call thru PRI line invalid_event_count=0, wdt_timeout_count=0 watchdog timer is not activated wait_for_bchan:False pri_chnl=(T1 1/0/0:22), vdev_chnl=(s8, c19) start_chan_p=0, chan_p=62436D58, call_id=0x800D, bchan_num=22 The calling party phone number = The called party phone number = 7511 ring_no_answer=0, ic_failure=0, ic_complete=0 dial_failure=0, oc_failure=0, oc_complete=1 oc_busy=0, oc_no_dial_tone=0, oc_dial_timeout=0 remote_link_disc=0, busyout=0, modem_reset=0 call_duration_started=3d16h, call_duration_ended=00:00:00, total_call_duration=00:00:00

Table 1 explains the fields contained in both of these examples.

show csm

Table 1 show csm voice Field Descriptions

Field Description

slot Indicates the slot where the VFC resides.

shelf/slot/port Specifies the T1 or E1 controller.

dspm/dsp/dsp channel Indicates which DSP channel is engaged in this call.

dsp Indicates the DSP through which this call is established.

slot/port This is the logical port number for the device. This is equivalent to the DSP

channel number. The port number is derived from:

(max_number_of_dsp_channels per dspm=12) * the dspm # (0-based) + (max_number_of_dsp_channels per dsp=2) * the dsp # (0-based) + the dsp channel number (0-based).

tone Indicates which signalling tone is being used (DTMF, MF, R2). This only

applies to CAS calls. Possible values are:

—mf

—dtmf

— r2-compelled

— r2-semi-compelled

— r2-non-compelled

Voice over IP for the Cisco AS5800 31

Command Reference

Table 1 show csm voice Field Descriptions (continued)

Field Description

device_status The status of the device. Possible values are:

— VDEV_STATUS_UNLOCKED—Device is unlocked (meaning that it is

available for new calls).

— VDEV_STATUS_ACTIVE_WDT—Device is allocated for a call and the

watchdog timer is set to time the connection response from the central office.

— VDEV_STATUS_ACTIVE_CALL—Device is engaged in an active,

connected call.

— VDEV_STATUS_BUSYOUT_REQ—Device is requested to busyout;

does not apply to voice devices.

— VDEV_STATUS_BAD—Device is marked as bad and not usable for

processing calls.

— VDEV_STATUS_BACK2BACK_TEST—Modem is performing

back-to-back testing (for modem calls only).

— VDEV_STATUS_RESET—Modem needs to be reset (for modem only).

— VDEV_STATUS_DOWNLOAD_FILE—Modem is downloading a file

(for modem only).

— VDEV_STATUS_DOWNLOAD_FAIL—Modem has failed during

downloading a file (for modem only).

— VDEV_STATUS_SHUTDOWN—Modem is not powered up (for modem

only).

— VDEV_STATUS_BUSY—Modem is busy (for modem only).

— VDEV_STATUS_DOWNLOAD_REQ—Modem is requesting

connection (for modem only).

Cisco IOS Release 12.0(7)T

show csm

Table 1 show csm voice Field Descriptions (continued)

Field Description

csm_state CSM call state of the current call (PRI line) associated with this device. Possible

values are:

— CSM_IDLE_STATE—Device is idle.

— CSM_IC_STATE—A device has been assigned to an incoming call.

— CSM_IC1_COLLECT_ADDR_INFO—A device has been selected to

perform ANI/DNIS address collection for this call. ANI/DNIS address information collection is in progress. The ANI/DNIS is used to decide whether the call should be processed by a modem or a voice DSP.

— CSM_IC2_RINGING—The device assigned to this incoming call has

been told to get ready for the call.

— CSM_IC3_WAIT_FOR_SWITCH_OVER—A new device is selected to

take over this incoming call from the device collecting the ANI/DNIS address information.

— CSM_IC4_WAIT_FOR_CARRIER—This call is waiting for the

CONNECT message from the carrier.

— CSM_IC5_CONNECTED—This incoming call is connected to the

central office.

— CSM_IC6_DISCONNECTING—This incoming call is waiting for a

DISCONNECT message from the VTSP module to complete the disconnect process.

— CSM_OC_STATE —An outgoing call is initiated.

— CSM_OC1_REQUEST_DIGIT—The device is requesting the first digit

for the dial-out number.

— CSM_OC2_COLLECT_1ST_DIGIT—The first digit for the dial-out

number has been collected.

— CSM_OC3_COLLECT_ALL_DIGIT—All the digits for the dial-out

number have been collected.

— CSM_OC4_DIALING—This call is waiting for a dsx0 (B channel) to be

available for dialing out.

— CSM_OC5_WAIT_FOR_CARRIER—This (outgoing) call is waiting for

the central office to connect.

— CSM_OC6_CONNECTED—This (outgoing) call is connected.

— CSM_OC7_BUSY_ERROR—A busy tone has been sent to the device

(for VoIP call, no busy tone is sent; just a DISCONNECT INDICATION message is sent to the VTSP module) and this call is waiting for a DISCONNECT message from the VTSP module (or ONHOOK message from the modem) to complete the disconnect process.

— CSM_OC8_DISCONNECTING—The central office has disconnected

this (outgoing) call and the call is waiting for a DISCONNECT message from the VTSP module to complete the disconnect process.

csm_state: invalid_event_count= Number of invalid events received by the CSM state machine.

wdt_timeout_count= Number of times the watchdog timer is activated for this call.

wdt_timestamp_started Indicates whether the watchdog timer is activated for this call.

wait_for_dialing: Indicates whether this (outgoing) call is waiting for a free digit collector to

become available to dial out the outgoing digits.

Voice over IP for the Cisco AS5800 33

Command Reference

Table 1 show csm voice Field Descriptions (continued)

Field Description

wait_for_bchan: Indicates whether this (outgoing) call is waiting for a B channel to send the call

out on.

pri_chnl= Indicates which type of TDM stream is used for the PRI connection. For PRI

and CAS calls, it will always be TDM_PRI_STREAM.

tdm_chnl= Indicates which type of TDM stream is used for the connection to the device

used to process this call. In the case of a VoIP call, this will always be set to TDM_DSP_STREAM.

dchan_idb_start_index= First index to use when searching for the next IDB of a free D channel.

dchan_idb_index= Index of the currently available IDB of a free D channel.

csm_event= Event just passed to the CSM state machine.

cause Event cause.

ring_no_answer= Number of times call failed because there was no response.

ic_failure= Number of failed incoming calls.

ic_complete= Number of successful incoming calls.

dial_failure= Number of times the connection failed because there was no dial tone.

oc_failure= Number of failed outgoing calls.

oc_complete= Number of successful outgoing calls.

oc_busy= Number of outgoing calls where the connection failed because there was a busy

signal.

oc_no_dial_tone= Number of outgoing calls where the connection failed because there was no dial

tone.

oc_dial_timeout= Number of outgoing calls where the connection failed because the timeout value

was exceeded.

call_duration_started= Indicates the start of this call.

call_duration_ended= Indicates the end of this call.

total_call_duration= Indicates the duration of this call.

The calling party phone number = Calling party number as given to CSM by ISDN.

The called party phone number = Called party number as given to CSM by ISDN.

total_free_rbs_timeslot = Total number of free RBS (CAS) timeslots available for the whole system.

total_busy_rbs_timeslot = Total number of RBS (CAS) timeslots that have been busied out. This includes

both dynamically and statically busied out RBS timeslots.

total_dynamic_busy_rbs_ timeslot =

total_static_busy_rbs_timeslot = Total number of RBS (CAS) timeslots that have been statically busied out (that

total_free_isdn_channels = Total number of free ISDN channels.

total_busy_isdn_channels = Total number of busy ISDN channels.

total_auto_busy_isdn_channels = Total number of ISDN channels that are automatically busied out.

Total number of RBS (CAS) timeslots that have been dynamically busied out.

is, they are busied out using the CLI command)

Cisco IOS Release 12.0(7)T

Related Commands

show csm

Command Description

show call active voice Displays the Voice over IP active call table.

show call history voice Displays the Voice over IP call history table.

show num-exp Displays how the number expansions are configured in Voice over IP.

show voice port Displays configuration information about a specific voice port.

Voice over IP for the Cisco AS5800 35

Command Reference

show voice port

To display configuration information about a specific voice port, use the show voice port privileged EXEC command.

Cisco 2600/3600 Series Router

show voice port slot-number/subunit-number/port

Cisco MC3810

show voice port [slot/port] [summary]

Cisco AS5300 Access Router

show voice port controller number:D

Cisco AS5800 Universal Access Router

show voice port {shelf/slot/port:D} | {shelf/slot/parent:port:D}

Syntax Description

For the Cisco 2600/3600 series:

slot-number Slot number in the Cisco router where the voice interface card is

installed. Valid entries are from 0 to 3, depending on the slot where it has been installed.

subunit-number Subunit on the voice interface card where the voice port is located. Valid

entries are 0 or 1.

port Voice port number. Valid entries are 0 or 1.

Cisco IOS Release 12.0(7)T

show voice port

For the Cisco MC3810:

slot/port (Optional) Displays information for only the voice port you specify with

the slot/port designation.

The slot variable specifies the slot number in the Cisco router where the voice interface card is installed. The only valid entry is 1.

The port variable specifies the voice port number. Valid ranges are as follows:

Analog voice ports: from 1 to 6.

Digital voice port:

Digital T1: from 1 to 24.

Digital E1: from 1 to 15, and from 17 to 31.

summary (Optional) Display a summary of all voice ports.

For the Cisco AS5300 Access Server:

controller number Specifies the T1 or E1 controller.

Command Mode

:D Indicates the D channel associated with ISDN PRI.

For the Cisco AS5800 Universal Access Server:

shelf/slot/port Specifies the T1 or E1 controller on the T1 card.Valid entries for the shelf

variable is 0 to 9999. Valid entries for the slot variable is 0 to 11. Valid entries for the port value is 0 to 11.

shelf/slot/parent:port Specifies the T1 controller on the T3 card. Valid entries for the shelf

variable is 0 to 9999. Valid entries for the slot variable is 0 to 11. Valid entries for the port variable is 1 to 28. The value for the parent variable is always 0.

:D Indicates the D channel associated with ISDN PRI.

Privileged EXEC

Voice over IP for the Cisco AS5800 37

Command Reference

Command History

Usage Guidelines

Examples

Release Modification

11.3(1)T This command was introduced.

11.3 MA Port-specific values for the Cisco MC3810 were added.

12.0(3)T Port-specific values for the Cisco AS5300 were added.

12.0(7)T Port-specific values for the Cisco AS5800 were added.

This command applies to Voice over IP, Voice over Frame Relay, Voice over ATM, and Voice over HDLC.

Use the show voice port privileged EXEC command to display configuration and voice interface card-specific information about a specific port.

The following is sample output from the show voice port command for an E&M voice port on the Cisco 3600 series:

router# show voice port 1/0/0 E&M Slot is 1, Sub-unit is 0, Port is 0 Type of VoicePort is E&M Operation State is unknown Administrative State is unknown The Interface Down Failure Cause is 0 Alias is NULL Noise Regeneration is disabled Non Linear Processing is disabled Music On Hold Threshold is Set to 0 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is disabled Echo Cancel Coverage is set to 16ms Connection Mode is Normal Connection Number is Initial Time Out is set to 0 s Interdigit Time Out is set to 0 s Analog Info Follows: Region Tone is set for northamerica Currently processing none Maintenance Mode Set to None (not in mtc mode) Number of signaling protocol errors are 0

Voice card specific Info Follows: Signal Type is wink-start Operation Type is 2-wire Impedance is set to 600r Ohm E&M Type is unknown Dial Type is dtmf In Seizure is inactive Out Seizure is inactive Digit Duration Timing is set to 0 ms InterDigit Duration Timing is set to 0 ms Pulse Rate Timing is set to 0 pulses/second InterDigit Pulse Duration Timing is set to 0 ms Clear Wait Duration Timing is set to 0 ms Wink Wait Duration Timing is set to 0 ms

Cisco IOS Release 12.0(7)T

show voice port

Wink Duration Timing is set to 0 ms Delay Start Timing is set to 0 ms Delay Duration Timing is set to 0 ms

The following is sample output from the show voice port command for an FXS voice port on the Cisco 3600 series:

router# show voice port 1/0/0 Foreign Exchange Station 1/0/0 Slot is 1, Sub-unit is 0, Port is 0 Type of VoicePort is FXS Operation State is DORMANT Administrative State is UP The Interface Down Failure Cause is 0 Alias is NULL Noise Regeneration is enabled Non Linear Processing is enabled Music On Hold Threshold is Set to 0 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 16ms Connection Mode is Normal Connection Number is Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Analog Info Follows: Region Tone is set for northamerica Currently processing none Maintenance Mode Set to None (not in mtc mode) Number of signaling protocol errors are 0 Voice card specific Info Follows: Signal Type is loopStart Ring Frequency is 25 Hz Hook Status is On Hook Ring Active Status is inactive Ring Ground Status is inactive Tip Ground Status is inactive Digit Duration Timing is set to 100 ms InterDigit Duration Timing is set to 100 ms Hook Flash Duration Timing is set to 600 ms

The following is sample output from the show voice port command for an FXS voice port on the Cisco MC3810:

router# show voice port 1/2 Voice port 1/2 Slot is 1, Port is 2 Type of VoicePort is FXS Operation State is UP Administrative State is UP No Interface Down Failure Description is not set Noise Regeneration is enabled Non Linear Processing is enabled In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 8 ms Connection Mode is normal Connection Number is not set Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Coder Type is g729ar8 Companding Type is u-law

Voice over IP for the Cisco AS5800 39

Command Reference

Voice Activity Detection is disabled Ringing Time Out is 180 s Wait Release Time Out is 30 s Nominal Playout Delay is 80 milliseconds Maximum Playout Delay is 160 milliseconds

Analog Info Follows: Region Tone is set for northamerica Currently processing Voice Maintenance Mode Set to None (not in mtc mode) Number of signaling protocol errors are 0 Impedance is set to 600r Ohm Analog interface A-D gain offset = -3 dB Analog interface D-A gain offset = -3 dB Voice card specific Info Follows: Signal Type is loopStart Ring Frequency is 20 Hz Hook Status is On Hook Ring Active Status is inactive Ring Ground Status is inactive Tip Ground Status is active Digit Duration Timing is set to 100 ms InterDigit Duration Timing is set to 100 ms Ring Cadence are [20 40] * 100 msec InterDigit Pulse Duration Timing is set to 500 ms

The following is sample output from the show voice port summary command for all voice ports on a Cisco MC3810 with an analog voice module (AVM):

router# show voice port summary

IN OUT ECHO PORT SIG-TYPE ADMIN OPER IN-STATUS OUT-STATUS CODEC VAD GAIN ATTN CANCEL 1/1 fxs-ls up up on-hook idle 729a n 0 0 y 1/2 fxs-ls up up on-hook idle 729a n 0 0 y 1/3 e&m-wnk up up idle idle 729a n 0 0 y 1/4 e&m-wnk up up idle idle 729a n 0 0 y 1/5 fxo-ls up up idle on-hook 729a n 0 0 y 1/6 fxo-ls up up idle on-hook 729a n 0 0 y

Table 2 explains the fields in the sample output.

Table 2 show voice port Field Descriptions

Field Description

Administrative State Administrative state of the voice port.

Alias User-supplied alias for this voice port.

Analog interface A-D gain offset Offset of the gain for analog-to-digital conversion.

Analog interface D-A gain offset Offset of the gain for digital-to-analog conversion.

Clear Wait Duration Timing Time of inactive seizure signal to declare call cleared.

Coder Type Voice compression mode used.

Companding Type Companding standard used to convert between analog and digital signals in

PCM systems.

Connection Mode Connection mode of the interface.

Connection Number Full E.164 telephone number used to establish a connection with the trunk or

PLAR mode.

Cisco IOS Release 12.0(7)T

show voice port

Table 2 show voice port Field Descriptions (continued)

Field Description

Currently Processing Type of call currently being processed: none, voice, or fax.

Delay Duration Timing Maximum delay signal duration for delay dial signaling.

Delay Start Timing Timing of generation of delayed start signal from detection of incoming

seizure.

Description Description of the voice port.

Dial Type Out-dialing type of the voice port.

Digit Duration Timing DTMF digit duration in milliseconds.

E&M Type Type of E&M interface.

Echo Cancel Coverage Echo cancel coverage for this port.

Echo Cancellation Whether or not echo cancellation is enabled for this port.

Hook Flash Duration Timing Maximum length of hook flash signal.

Hook Status Hook status of the FXO/FXS interface.

Impedance Configured terminating impedance for the E&M interface.

In Gain Amount of gain inserted at the receiver side of the interface.

In Seizure Incoming seizure state of the E&M interface.

Initial Time Out Amount of time the system waits for an initial input digit from the caller.

InterDigit Duration Timing DTMF interdigit duration in milliseconds.

InterDigit Pulse Duration Timing Pulse dialing interdigit timing in milliseconds.

Interdigit Time Out Amount of time the system waits for a subsequent input digit from the caller.

Maintenance Mode Maintenance mode of the voice port.

Maximum Playout Delay The amount of time before the Cisco MC3810 DSP starts to discard voice

packets from the DSP buffer.

Music On Hold Threshold Configured music-on-hold threshold value for this interface.

Noise Regeneration Whether or not background noise should be played to fill silent gaps if VAD is

activated.

Nominal Playout Delay The amount of time the Cisco MC3810 DSP waits before starting to play out

the voice packets from the DSP buffer.

Non-Linear Processing Whether or not non-linear processing is enabled for this port.

Number of signaling protocol errors Number of signaling protocol errors.

Operations State Operation state of the port.

Operation Type Operation of the E&M signal: two-wire or four-wire.

Out Attenuation Amount of attenuation inserted at the transmit side of the interface.

Out Seizure Outgoing seizure state of the E&M interface.

Port Port number for this interface associated with the voice interface card.

Pulse Rate Timing Pulse dialing rate in pulses per second (pps).

Region Tone Configured regional tone for this interface.

Ring Active Status Ring active indication.

Ring Cadence Configured ring cadence for this interface.

Ring Frequency Configured ring frequency for this interface.

Voice over IP for the Cisco AS5800 41

Command Reference

Table 2 show voice port Field Descriptions (continued)

Field Description

Ring Ground Status Ring ground indication.

Ringing Time Out Ringing time out duration.

Signal Type Type of signaling for a voice port: loop-start, ground-start, wink-start,

immediate, and delay-dial.

Slot Slot used in the voice interface card for this port.

Sub-unit Subunit used in the voice interface card for this port.

Tip Ground Status Tip ground indication.

Type of VoicePort Type of voice port: FXO, FXS, and E&M.

The Interface Down Failure Cause Text string describing why the interface is down,

Voice Activity Detection Whether Voice Activity Detection is enabled or disabled.

Wait Release Time Out The time that a voice port stays in the call-failure state while the Cisco

MC3810 sends a busy tone, reorder tone, or an out-of-service tone to the port.

Wink Duration Timing Maximum wink duration for wink start signaling.

Wink Wait Duration Timing Maximum wink wait duration for wink start signaling.

The following is sample output from the Cisco AS5800 for the show voice port command:

5800# show voice port 1/0/0:D ISDN 1/0/0:D Type of VoicePort is ISDN Operation State is DORMANT Administrative State is UP No Interface Down Failure Description is "" 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 16 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

Table 3 explains the fields in the sample output.

Table 3 show voice port Field Descriptions for the Cisco AS5800

Field Description

Type of VoicePort Indicates the voice port type.

Operational State Operational state of the voice port.

Administrative State Administrative state of the voice port.

Clear Wait Duration Timing Time of inactive seizure signal to declare call cleared.

Currently Processing Type of call currently being processed: none, voice, or fax.

Cisco IOS Release 12.0(7)T

show voice port

Table 3 show voice port Field Descriptions for the Cisco AS5800 (continued)

Field Description

Operations State Operation state of the port.

Operation Type Operation of the E&M signal: two-wire or four-wire.

Noise Regeneration Whether or not background noise should be played to fill silent gaps if VAD is

activated.

Non-Linear Processing Whether or not non-linear processing is enabled for this port.

Music-On-Hold Threshold Configured music-on-hold threshold value for this interface.

In Gain Amount of gain inserted at the receiver side of the interface.

Out Attenuation Amount of attenuation inserted at the transmit side of the interface.

Pulse Rate Timing Pulse dialing rate in pulses per second (pps).

Echo Cancellation Whether or not echo cancellation is enabled for this port.

Echo Cancel Coverage Echo Cancel Coverage for this port.

Connection Mode Connection mode of the interface.

Connection Number Full E.164 telephone number used to establish a connection with the trunk or

PLAR mode.

Initial Time Out Amount of time the system waits for an initial input digit from the caller.

Interdigit Time Out Amount of time the system waits for a subsequent input digit from the caller.

Regional Tone Configured regional tone for this interface.

Related Commands

Command Description

show call active voice Displays the Voice over IP active call table.

show call history voice Displays the Voice over IP call history table.

show dial-peer voice Displays configuration information for dial peers.

show voice port Displays configuration information about a specific voice port.

Voice over IP for the Cisco AS5800 43

Command Reference

show vrm active_calls

To display active-only voice calls either for a specific VFC or all VFCs, use the

show vrm active_calls privileged EXEC command.

show vrm active_calls {dial-shelf-slot-number | all}

Syntax Description

dial shelf slot number Slot number of the dial shelf. Valid number is 0 to 13.

all Lists all active calls for VFC slots.

Defaults

No default behavior or values.

Command Modes

Privileged EXEC

Command History

Usage Guidelines

Examples

Release Modification

12.0(7)T This command was introduced.

Use the show vrm active_calls to display active-only voice calls either for a specific VFC or all VFCs. Each active call occupies a block of information describing the call. This information provides basically the same information as the show vrm vdevice command.

The following is sample output from the show vrm active_calls command specifying dial shelf slot number:

5800# show vrm active_calls 6 slot = 6 virtual voice dev (tag) = 61 channel id = 2 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 241 Resource (vdev_common) status = 401 means :active others tot ingress data = 24 tot ingress control = 1308 tot ingress data drops = 0 tot ingress control drops = 0 tot egress data = 22051 tot egress control = 1304 tot egress data drops = 0 tot egress control drops = 0

Cisco IOS Release 12.0(7)T

show vrm active_calls

slot = 6 virtual voice dev (tag) = 40 channel id = 2 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 157 Resource (vdev_common) status = 401 means :active others

Table 4 explains the fields in the sample output.

Table 4 show vrm vdevice Field Descriptions

Field Description

slot Slot where voice card is installed.

virtual voice dev (tag) Identification number of the virtual voice device.

channel id Identification number of the channel associated with this virtual voice device.

capability list map Bitmaps for the codec supported on that DSP channel. Available values are:

• CC_CAP_CODEC_G711U: 0x1

• CC_CAP_CODEC_G711A: 0x2

• CC_CAP_CODEC_G729IETF: 0x4

• CC_CAP_CODEC_G729a: 0x8

• CC_CAP_CODEC_G726r16: 0x10

• CC_CAP_CODEC_G726r24: 0x20

• CC_CAP_CODEC_G726r32: 0x40

• CC_CAP_CODEC_G728: 0x80

• CC_CAP_CODEC_G723r63: 0x100

• CC_CAP_CODEC_G723r53: 0x200

• CC_CAP_CODEC_GSM: 0x400

• CC_CAP_CODEC_G729b: 0x800

• CC_CAP_CODEC_G729ab: 0x1000

• CC_CAP_CODEC_G723ar63: 0x2000

• CC_CAP_CODEC_G723ar53: 0x4000

• CC_CAP_CODEC_G729: 0x8000

last/current codec loaded/used Indicates the last codec loaded or used.

TDM timeslot Time division multiplexing timeslot.

Resource (vdev_common) status Current status of the VFC.

tot ingress data Total amount of data (number of packets) sent from the PSTN side of the

connection to the VoIP side of the connection.

tot ingress control Total number of control packets sent from the PSTN side of the connection to

the VoIP side of the connection.

tot ingress data drops Total number of data packets dropped from the PSTN side of the connection

to the VoIP side of the connection.

tot ingress control drops Total number of control packets dropped from the PSTN side of the

connection to the VoIP side of the connection.

tot egress data Total amount of data (number of packets) sent from the VoIP side of the

connection to the PSTN side of the connection.

Voice over IP for the Cisco AS5800 45

Command Reference

Related Commands

Table 4 show vrm vdevice Field Descriptions (continued)

Field Description

tot egress control Total number of control packets sent from the VoIP side of the connection to

the PSTN side of the connection.

tot egress data drops Total number of data packets dropped from the VoIP side of the connection to

the PSTN side of the connection.

tot egress control drops Total number of control packets dropped from the VoIP side of the connection

to the PSTN side of the connection.

Command Description

show vrm vdevices Displays detailed information for a specific DSP or a brief summary display for

all VFCs.

Cisco IOS Release 12.0(7)T

show vrm vdevices

To display detailed information for a specific DSP or a brief summary display for all VFCs, use the

show vrm vdevices privileged EXEC command.

show vrm vdevices {{vfc-slot-number | voice-device-number} | summary}

Syntax Description

vfc-slot-number Slot number of the VFC. Valid number is 0 to 11.

voice-device-number DSP number. Valid number is 1 to 96.

summary List synopsis of voice feature card DSP mappings, capabilities, and

Defaults

No default behavior or values.

show vrm vdevices

resource states.

Command Modes

Command History

Usage Guidelines

Privileged EXEC

Release Modification

12.0(7)T This command was introduced.

Use the show vrm vdevice to display detailed information for a specific DSP or a brief summary display for all VFCs. The display provides information on the number of channels, channels per DSP, bitmap of DSPMs, version numbers, and so on. This information is useful in monitoring the current state of your VFCs.

The display for a specific DSP provides information on the codec that each channel is using, if active, or last used and if the channel is not currently transmitting cells. It also displays the state of the resource. In most cases, if there is an active call on that channel, the resource should be marked active. If the resource is marked as reset and/or bad, this may be an indication of a response loss for the VFC on a reset request. If this condition persists, you might experience a problem with the communication link between the router shelf and the VFC.

Voice over IP for the Cisco AS5800 47

Command Reference

Examples

The following is sample output from the show vrm vdevice command specifying dial shelf slot number and DSP number. In this particular example, the call is active so the statistics displayed are for this active call. If no calls are currently active on the device, the statistics would be for the previous (or last active) call.

5800# show vrm vdevices 6 1 slot = 6 virtual voice dev (tag) = 1 channel id = 1 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 0 Resource (vdev_common) status = 401 means :active others tot ingress data = 101 tot ingress control = 1194 tot ingress data drops = 0 tot ingress control drops = 0 tot egress data = 39722 tot egress control = 1209 tot egress data drops = 0 tot egress control drops = 0

slot = 6 virtual voice dev (tag) = 1 channel id = 2 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 1 Resource (vdev_common) status = 401 means :active others tot ingress data = 21 tot ingress control = 1167 tot ingress data drops = 0 tot ingress control drops = 0 tot egress data = 19476 tot egress control = 1163 tot egress data drops = 0 tot egress control drops = 0

Table 5 explains the fields in the sample output.

Table 5 show vrm vdevice Field Descriptions

Field Description

slot Slot where voice card is installed.

virtual voice dev (tag) Identification number of the virtual voice device.

channel id Identification number of the channel associated with this virtual voice device.

Cisco IOS Release 12.0(7)T

show vrm vdevices

Table 5 show vrm vdevice Field Descriptions (continued)

Field Description

capability list map Bitmaps for the codec supported on that DSP channel. Available values are:

• CC_CAP_CODEC_G711U: 0x1

• CC_CAP_CODEC_G711A: 0x2

• CC_CAP_CODEC_G729IETF: 0x4

• CC_CAP_CODEC_G729a: 0x8

• CC_CAP_CODEC_G726r16: 0x10

• CC_CAP_CODEC_G726r24: 0x20

• CC_CAP_CODEC_G726r32: 0x40

• CC_CAP_CODEC_G728: 0x80

• CC_CAP_CODEC_G723r63: 0x100

• CC_CAP_CODEC_G723r53: 0x200

• CC_CAP_CODEC_GSM: 0x400

• CC_CAP_CODEC_G729b: 0x800

• CC_CAP_CODEC_G729ab: 0x1000

• CC_CAP_CODEC_G723ar63: 0x2000

• CC_CAP_CODEC_G723ar53: 0x4000

• CC_CAP_CODEC_G729: 0x8000

last/current codec loaded/used Indicates the last codec loaded or used.

TDM timeslot Time division multiplexing timeslot.

Resource (vdev_common) status Current status of the VFC. Possible field values are:

• FREE = 0x0000

• ACTIVE_CALL = 0x0001

• BUSYOUT_REQ = 0x0002

• BAD = 0x0004

• BACK2BACK_TEST = 0x0008

• RESET = 0x0010

• DOWNLOAD_FILE = 0x0020

• DOWNLOAD_FAIL = 0x0040

• SHUTDOWN = 0x0080

• BUSY = 0x0100

• OIR = 0x0200

• HASLOCK = 0x0400 /* vdev_pool has locked port */

• DOWNLOAD_REQ = 0x0800

• RECOVERY_REQ = 0x1000

• NEGOTIATED = 0x2000

• OOS = 0x4000

tot ingress data Total amount of data (number of packets) sent from the PSTN side of the

connection to the VoIP side of the connection.

tot ingress control Total number of control packets sent from the PSTN side of the connection to

the VoIP side of the connection.

Voice over IP for the Cisco AS5800 49

Command Reference

Table 5 show vrm vdevice Field Descriptions (continued)

Field Description

tot ingress data drops Total number of data packets dropped from the PSTN side of the connection

to the VoIP side of the connection.

tot ingress control drops Total number of control packets dropped from the PSTN side of the

connection to the VoIP side of the connection.

tot egress data Total amount of data (number of packets) sent from the VoIP side of the

connection to the PSTN side of the connection.

tot egress control Total number of control packets sent from the VoIP side of the connection to

the PSTN side of the connection.

tot egress data drops Total number of data packets dropped from the VoIP side of the connection to

the PSTN side of the connection.

tot egress control drops Total number of control packets dropped from the VoIP side of the connection

to the PSTN side of the connection.

The following is sample output from the show vrm devices command specifying a summary list. In the Voice Device Mapping area, the C_Ac column indicates number of active calls for a specific DSP. If there are any non zero numbers under the C_Rst and/or C_Bad column, this indicates a reset request was sent but it was lost; this could mean a faulty DSP.

5800# show vrm vdevices summary *********************************************************** ******summary of voice devices for all voice cards********* ***********************************************************

slot = 6 major ver = 0 minor ver = 1 core type used = 2 number of modules = 16 number of voice devices (DSPs) = 96 chans per vdevice = 2 tot chans = 192 tot active calls = 178 module presense bit map = FFFF tdm mode = 1 num_of_tdm_timeslots = 384 auto recovery is on

number of default voice file (core type images) = 2 file 0 maj ver = 0 min ver = 0 core_type = 1 trough size = 2880 slop value = 0 built-in codec bitmap = 0 loadable codec bitmap = 0 fax codec bitmap = 0

file 1 maj ver = 3 min ver = 1 core_type = 2 trough size = 2880 slop value = 1440 built-in codec bitmap = 40B loadable codec bitmap = BFC fax codec bitmap = 7E

-------------------Voice Device Mapping-----------------------Logical Device (Tag) Module# DSP# C_Ac C_Busy C_Rst C_Bad

--------------------------------------------------------------1 1 1 2 0 0 0 2 1 2 2 0 0 0 3 1 3 2 0 0 0 4 1 4 2 0 0 0 5 1 5 2 0 0 0 6 1 6 2 0 0 0 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 7 2 1 2 0 0 0 8 2 2 2 0 0 0 9 2 3 2 0 0 0 10 2 4 1 0 0 0 11 2 5 2 0 0 0 12 2 6 1 0 0 0 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++

Cisco IOS Release 12.0(7)T

show vrm vdevices

<information deleted> +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 91 16 1 2 0 0 0 92 16 2 2 0 0 0 93 16 3 1 0 0 0 94 16 4 2 0 0 0 95 16 5 2 0 0 0 96 16 6 2 0 0 0 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++

Total active call channels = 178 Total busied out channels = 0 Total channels in reset = 0 Total bad channels = 0 Note :Channels could be in multiple states

Table 6 explains the fields in the sample output.

Table 6 show vrm vdevice summary Field Descriptions

Field Description

slot Slot number where VFC is installed.

major ver Major version of firmware running on VFC.

minor ver Minor version of firmware running on VFC.

core type used Type of DSPware in use. Possible field values are:

• 1 = UBL (boot loader)

• 2 = high complexity core

• 3 = medium complexity core

• 4 = low complexity core

• 255 = invalid.

number of modules Number of modules on the VFC. Maximum number possible is 16.

number of voice devices (DSP)s Number of possible DSPs. Maximum number is 96.

chans per vdevice Number of channels (meaning calls) each DSP can handle.

tot chans Total number of channels.

tot active calls Total number of active calls on this VFC.

module presense bit map Indicates a 16-bit bitmap, each bit representing a module.

tdm mode Time division multiplex bus mode. Possibe field values are:

• 0 = VFC is in classic mode

• 1 = VFC is in plus mode.

This field should always be 1.

num_of_tdm_timeslots Total number of calls that can be handled by the VFC.

auto recovery Indicates whether auto recovery is enabled. When autorecovery is enabled,

the VRM will try to recover a DSP by resetting it if, for some reason, the DSP stops responding.

number of default voice file (core type images)

maj ver Major version of the DSPware in use.

min ver Minor version of the DSPware in use.

Number of DSPware files in use.

Voice over IP for the Cisco AS5800 51

Command Reference

Table 6 show vrm vdevice summary Field Descriptions (continued)

Field Description

core type Type of DSPware in use: Possible field values are:

• 1 = boot loader

• 2 = high complexity core

• 3 = medium complexity core

• 4 = low complexity core

trough size This value indirectly represents the complexity of the DSPware in use.

slop value This value indirectly represents the complexity of the DSPware in use.

built-in codec bitmap Represents the bitmap of the codec built into the DSP firmware. Possible field

values are:

• CC_CAP_CODEC_G711U 0x0001

• CC_CAP_CODEC_G711A 0x0002

• CC_CAP_CODEC_G729IETF 0x0004

• CC_CAP_CODEC_G729a 0x0008

• CC_CAP_CODEC_G726r16 0x0010

• CC_CAP_CODEC_G726r24 0x0020

• CC_CAP_CODEC_G726r32 0x0040

• CC_CAP_CODEC_G728 0x0080

• CC_CAP_CODEC_G723r63 0x0100

• CC_CAP_CODEC_G723r53 0x0200

• CC_CAP_CODEC_GSM 0x0400

• CC_CAP_CODEC_G729b 0x0800

• CC_CAP_CODEC_G729ab 0x1000

• CC_CAP_CODEC_G723ar63 0x2000

• CC_CAP_CODEC_G723ar53 0x4000

• CC_CAP_CODEC_G729 0x8000

Cisco IOS Release 12.0(7)T

show vrm vdevices

Table 6 show vrm vdevice summary Field Descriptions (continued)

Field Description

loadable codec bitmap Represents the loadable codec bitmap for the loadable CODECs. Possible

field values are:

• CC_CAP_CODEC_G711U = 0x0001

• CC_CAP_CODEC_G711A = 0x0002

• CC_CAP_CODEC_G729IETF = 0x0004

• CC_CAP_CODEC_G729a = 0x0008

• CC_CAP_CODEC_G726r16 = 0x0010

• CC_CAP_CODEC_G726r24 = 0x0020

• CC_CAP_CODEC_G726r32 = 0x0040

• CC_CAP_CODEC_G728 = 0x0080

• CC_CAP_CODEC_G723r63 = 0x0100

• CC_CAP_CODEC_G723r53 = 0x0200

• CC_CAP_CODEC_GSM = 0x0400

• CC_CAP_CODEC_G729b = 0x0800

• CC_CAP_CODEC_G729ab = 0x1000

• CC_CAP_CODEC_G723ar63 = 0x2000

• CC_CAP_CODEC_G723ar53 = 0x4000

• CC_CAP_CODEC_G729 = 0x8000

fax codec bitmap Represents the fax codec bitmap. Possible field values are:

• FAX_NONE = 0x1

• FAX_VOICE = 0x2

• FAX_144 = 0x4

• FAX_96 = 0x8

• FAX_72 = 0x10

• FAX_48 = 0x20

• FAX_24 = 0x40

Logical Device (Tag) Tag number or the DSP number on that VFC.

Module # Number identifying the module associated with a specific logical device.

DSP# Number identifying the DSP on the VFC.

C_Ac Number of active calls on identified DSP.

C_Busy Number of busied-out channels associated with identified DSP.

C_Rst Number of channels in the reset state associated with identified DSP.

C_Bad Number of defective (“bad”) channels associated with identified DSP.

Total active call channels Total number of active calls.

Total busied out channels Total number of busied-out channels.

Total channels in reset Total number of channels in reset state.

Total bad channels Total number of defective channels.

Voice over IP for the Cisco AS5800 53

Command Reference

Related Commands

Command Description

show vrm active_calls Displays active-only voice calls either for a specific VFC or all VFCs.

Cisco IOS Release 12.0(7)T

test vrm busyout

To busyout a specific DSP or channels on a specific DSP, use the test vrm busyout privileged EXEC command.

test vrm busyout slot-number {first-dsp-number {last-dsp-number | {channel number}} | all

Syntax Description

slot-number Number identifing the slot where the VFC is installed. Values for this

first-dsp-number Specifies the first DSP in a range to be busied out. Each VFC holds 96

last-dsp-number Specifies the last DSP in a range to be busied out. Each VFC holds 96

channel (Optional) Specifies that a certain channel on the specified DSPs will be

number Indicates the channel to be busied out. Values are 1 or 2.

test vrm busyout

field are 0 to 11.

DSPs, so the value for this argument is 1 to 96.

busied out.

Defaults

Command Modes

Command History

Usage Guidelines

all Indicates that all 96 DSPs on the VFC installed in the defined slot will be

busied out.

No default behavior or values.

Privileged EXEC

Release Modification

12.0(7)T This command was introduced.

Use the test vrm busyout command to busy out either one specific DSP or a range of DSPs on a specific VFC. In addition, you can use this comand to busyout a particular channel on a specified DSP or range of DSPs. To restore the activity of the busied-out DSP(s), use the test vrm unbusyout command.

Examples

The following example busies out all of the DSPs and associated channels for the VFC located in slot 4:

router# test vrm busyout 4 all

Voice over IP for the Cisco AS5800 55

Command Reference

Related Commands

The following example busied out all of the channels from DSP1 to DSP3 for the VFC located in slot 4:

router# test vrm busyout 4 1 3

The following example busies out only channel 2 of DSP1 for the VFC located in slot 4:

router# test vrm busyout 4 1 channel 2

Command Description

test vrm unbusyout Restores activity to a busied-out DSP or busied-out channels on a DSP.

Cisco IOS Release 12.0(7)T

test vrm reset

To reset a particular DSP, use the test vrm reset privileged EXEC command.

Syntax Description

slot-number Number identifing the slot where the VFC is installed.

dsp-number Number identifying the DSP to be reset.

Defaults

No default behavior or values.

Command Modes

Privileged EXEC

test vrm reset

test vrm reset {slot-number dsp-number}

Command History

Usage Guidelines

Examples

Release Modification

12.0(7)T This command was introduced.

Use the test vrm reset command to send a hard reset command to an identified DSP. When this command is used, any active calls on all channels associated with this DSP are dropped. Under most circumstances, you will never need to use this command.

The following example resets DSP 4 on the VFC installed in slot 2:

router# test vrm reset 4 2

Resetting voice device may termiate active calls [confirm}

Reset command sent to voice card 4 for voice device 2.

Voice over IP for the Cisco AS5800 57

Command Reference

test vrm unbusyout

To restore activity to a busied-out DSP or busied-out channels on a DSP, use the test vrm unbusyout privileged EXEC command.

test vrm unbusyout slot-number {first-dsp-number {last-dsp-number | {channel number}} | all

Syntax Description

slot-number Number identifing the slot where the VFC is installed. Values for this

first-dsp-number Specifies the first DSP in a range to be restored. Each VFC holds 96

last-dsp-number Specifies the last DSP in a range to be restored. Each VFC holds 96

channel (Optional) Specifies that a certain channel on the specified DSPs will be

number Indicates the channel to be restored. Values are 1 or 2.

field are 0 to 11.

DSPs, so the value for this argument is 1 to 96.

restored.

Defaults

Command Modes

Command History

Usage Guidelines

all Indicates that all 96 DSPs on the VFC installed in the defined slot will be

restored.

No default behavior or values.

Privileged EXEC

Release Modification

12.0(7)T This command was introduced.

Use the test vrm unbusyout command to restore either one specific DSP or a range of DSPs on a specific VFC. In addition, you can use this comand to restore a particular channel on a specified DSP or range of DSPs. To busy out a DSP (or range of DSPs) or to busy out a particular channel, use the test vrm busyout command.

Examples

Cisco IOS Release 12.0(7)T

The following example restores the activity of all of the DSPs and associated channels for the VFC located in slot 4:

router# test vrm unbusyout 4 all

Related Commands

test vrm unbusyout

The following example restores the activity of all the channels on the DSP from DSP1 to DSP3 for the VFC located in slot 4:

router# test vrm unbusyout 4 1 3

The following example restores the activity of only channel 2 of DSP1 for the VFC located in slot 4:

router# test vrm unbusyout 4 1 channel 2

Command Description

test vrm busyout Busies out a DSP or busies out channels on a DSP.

Voice over IP for the Cisco AS5800 59

Command Reference

voice-port

To enter the voice-port configuration mode, use the voice-port global configuration command.

Cisco 2600/3600 Series Router

voice-port slot-number/subunit-number/port

Cisco MC3810

voice-port [slot/port] [summary]

Cisco AS5300 Access Router

voice-port controller number:D

Cisco AS5800 Universal Access Router

voice-port {shelf/slot/port:D} | {shelf/slot/parent:port:D}

Syntax Description

For the Cisco 2600/3600 series:

slot-number Slot number in the Cisco router where the voice interface card is

installed. Valid entries are from 0 to 3, depending on the slot where it has been installed.

subunit-number Subunit on the voice interface card where the voice port is located. Valid

entries are 0 or 1.

port Voice port number. Valid entries are 0 or 1.

For the Cisco MC3810:

slot/port (Optional) Displays information for only the voice port you specify with

the slot/port designation.

The slot variable specifies the slot number in the Cisco router where the voice interface card is installed. The only valid entry is 1.

The port variable specifies the voice port number. Valid ranges are as follows:

Analog voice ports: from 1 to 6.

Cisco IOS Release 12.0(7)T

Digital voice port:

Digital T1: from 1 to 24.

Digital E1: from 1 to 15, and from 17 to 31.

summary (Optional) Display a summary of all voice ports.

voice-port

For the Cisco AS5300 Access Server:

controller number Specifies the T1 or E1 controller.

:D Indicates the D channel associated with ISDN PRI.

For the Cisco AS5800 Universal Access Server:

shelf/slot/port Specifies the T1 or E1 controller on the T1 card. Valid entries for the

shelf variable is 0 to 9999. Valid entries for the slot value is 0 to 11. Valid entries for the port variable is 0 to 11.

shelf/slot/parent:port Specifies the T1 controller on the T3 card. Valid entries for the shelf

variable is 0 to 9999. Valid entries for the slot variable is 0 to 11. Valid entries for the port variable is 1 to 28. The value for the parent variable is always 0.

:D Indicates the D channel associated with ISDN PRI.

Defaults

Command Modes

Command History

Usage Guidelines

No default behavior or values.

Global configuration

Release Modification

11.3(1)T This command was introduced.

11.3(3)T Support for Cisco 2600 series routers was added.

12.0(3)T Support for the Cisco AS5300 Access Server was added.

12.0(7)T Support for the Cisco AS5800 Access Server was added.

Use the voice-port global configuration command to switch to the voice-port configuration mode from the global configuration mode. Use the exit command to exit the voice-port configuration mode and return to the global configuration mode.

Examples

The following example accesses the voice-port configuration mode for port 0, located on subunit 0 on a voice interface card installed in slot 1 for the Cisco 3600 series:

configure terminal

voice-port 1/0/0

Voice over IP for the Cisco AS5800 61

Command Reference

Related Commands

The following example accesses the voice-port configuration mode for digital voice port 24 on a Cisco MC3810 with a DVM installed:

configure terminal

voice-port 1/24

The following example accesses the voice-port configuration mode for the Cisco AS5300:

configure terminal

voice-port 1:D

The following example accesses the voice-port configuration mode for the Cisco AS5800 (T1 card):

configure terminal

voice-port 1/0/0:D

The following example accesses the voice-port configuration mode for the Cisco AS5800 (T3 card):

configure terminal

voice-port 1/0/0:1:D

Command Description

dial-peer voice Enters dial-peer configuration mode and specifies a tag number for a dial peer.

Cisco IOS Release 12.0(7)T

Debug Commands

This section documents new or modified debug commands. All other commands used with this feature are documented in one of the follwing Cisco IOS documentation:

• Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference

• Cisco IOS Release 12.0 Dial Soluions Command Reference

• Cisco IOS Release 12.0(3)T Voice over IP for the AS5300 feature module

• Cisco IOS Release 12.0(3)T Service Provider Features for Voice over IP feature module

New Debug Commands

• debug vrm control

• debug vrm error

• debug inout

Debug Commands

Voice over IP for the Cisco AS5800 63

Debug Commands

debug vrm control

To display all control messages sent to and received from the DSP, use the debug vrm control privileged EXEC command. To stop displaying DSP-specific control messages, use the no form of this command.

[no] debug vrm control

Syntax Description

There are no arguments or keywords used in this command.

Defaults

No default behavior or values.

Command History

Release Modification

12.0(7)T This command was introduced.

Examples

The following example displays DSP-specific control messages going to the VRM:

*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size C *Nov 22 19:17:49.351: content : 0 0 0 1 0 8 0 1 0 4B 0 0 0 0 0

*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size 14 *Nov 22 19:17:49.351: content : 0 0 0 1 0 10 0 1 0 4A 0 1 0 0 0

*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size 1C *Nov 22 19:17:49.351: content : 0 0 0 1 0 18 0 1 0 5C 0 2 0 2 0

*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size 16 *Nov 22 19:17:49.351: content : 0 0 0 1 0 12 0 1 0 4C 0 3 0 1 0

*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size E *Nov 22 19:17:49.351: content : 0 0 0 1 0 A 0 1 0 42 0 4 0 0 0

*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size 10 *Nov 22 19:17:49.351: content : 0 0 0 1 0 C 0 1 0 5B 0 5 0 0 0

*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size E *Nov 22 19:17:49.351: content : 0 0 0 1 0 A 0 1 0 4E 0 6 FF DA 0

*Nov 22 19:17:51.995: SEND CONTROL slot 4 tag 1 size C *Nov 22 19:17:51.995: content : 0 0 0 1 0 8 0 1 0 44 0 7 FF DA 0

*Nov 22 19:17:51.995: SEND CONTROL slot 4 tag 1 size C *Nov 22 19:17:51.995: content : 0 0 0 1 0 8 0 1 0 47 0 8 FF DA 0

Cisco IOS Release 12.0(7)T

*Nov 22 19:17:51.995: SEND CONTROL slot 4 tag 1 size C *Nov 22 19:17:51.995: content : 0 0 0 1 0 8 0 1 0 44 0 9 FF DA 0

*Nov 22 19:17:51.995: SEND CONTROL slot 4 tag 1 size 1C *Nov 22 19:17:51.995: content : 0 0 0 1 0 18 0 1 0 5C 0 A 0 2 0

*Nov 22 19:17:51.995: SEND CONTROL slot 4 tag 1 size 1C

debug vrm control

*Nov 22 19:17:51.995: content : 0 0 0 1 0 18 0 1 0 49 0 B 0 1 0

*Nov 22 19:17:54.815: SEND CONTROL slot 4 tag 1 size E *Nov 22 19:17:54.815: content : 0 0 0 1 0 A 0 1 0 53 0 1 0 0 0

*Nov 22 19:17:54.815: SEND CONTROL slot 4 tag 1 size E *Nov 22 19:17:54.815: content : 0 0 0 1 0 A 0 1 0 54 0 1 0 0 0

*Nov 22 19:17:54.815: SEND CONTROL slot 4 tag 1 size E *Nov 22 19:17:54.815: content : 0 0 0 1 0 A 0 1 0 57 0 1 0 0 0

*Nov 22 19:17:54.827: nip_voice_service_cb : Msg from DS slot 4 cmd = 196. *Nov 22 19:17:54.827: RECEIVED CONTROL slot 4 tag 1 size 1C *Nov 22 19:17:54.827: content : 0 0 0 1 0 18 0 1 0 C4 0 1 8F EA 9B *Nov 22 19:17:54.827: DSP msg 196 received *Nov 22 19:17:54.827: nip_voice_service_cb : Msg from DS slot 4 cmd = 197. *Nov 22 19:17:54.827: RECEIVED CONTROL slot 4 tag 1 size 24 *Nov 22 19:17:54.827: content : 0 0 0 1 0 20 0 1 0 C5 0 1 0 0 0 *Nov 22 19:17:54.827: DSP msg 197 received *Nov 22 19:17:54.827: nip_voice_service_cb : Msg from DS slot 4 cmd = 200. *Nov 22 19:17:54.827: RECEIVED CONTROL slot 4 tag 1 size 34 *Nov 22 19:17:54.827: content : 0 0 0 1 0 30 0 1 0 C8 0 1 0 0 0 *Nov 22 19:17:54.827: DSP msg 200 received

*Nov 22 19:17:58.539: SEND CONTROL slot 4 tag 1 size E *Nov 22 19:17:58.539: content : 0 0 0 1 0 A 0 1 0 53 0 1 0 0 0

*Nov 22 19:17:58.539: SEND CONTROL slot 4 tag 1 size E *Nov 22 19:17:58.539: content : 0 0 0 1 0 A 0 1 0 54 0 1 0 0 0

*Nov 22 19:17:58.539: SEND CONTROL slot 4 tag 1 size E *Nov 22 19:17:58.539: content : 0 0 0 1 0 A 0 1 0 57 0 1 0 0 0

*Nov 22 19:17:58.551: nip_voice_service_cb : Msg from DS slot 4 cmd = 196. *Nov 22 19:17:58.555: RECEIVED CONTROL slot 4 tag 1 size 1C *Nov 22 19:17:58.555: content : 0 0 0 1 0 18 0 1 0 C4 0 1 8F EA 9B *Nov 22 19:17:58.555: DSP msg 196 received *Nov 22 19:17:58.555: nip_voice_service_cb : Msg from DS slot 4 cmd = 197. *Nov 22 19:17:58.555: RECEIVED CONTROL slot 4 tag 1 size 24 *Nov 22 19:17:58.555: content : 0 0 0 1 0 20 0 1 0 C5 0 1 0 0 0 *Nov 22 19:17:58.555: DSP msg 197 received *Nov 22 19:17:58.555: nip_voice_service_cb : Msg from DS slot 4 cmd = 200. *Nov 22 19:17:58.555: RECEIVED CONTROL slot 4 tag 1 size 34 *Nov 22 19:17:58.555: content : 0 0 0 1 0 30 0 1 0 C8 0 1 0 0 0 *Nov 22 19:17:58.555: DSP msg 200 received *Nov 22 19:18:02.127: SEND CONTROL slot 4 tag 1 size C *Nov 22 19:18:02.127: content : 0 0 0 1 0 8 0 1 0 47 0 C 0 0 0

Format of the Send messages is as follows:

SEND CONTROL slot <slot#> tag <tag#> size <size> content : <x x x x> <x x> <x x> <x x> <x x> <x x x>

tag# len chan msg proc rtp_header

Format for the Receive messages is as follows:

nip_voice_service_cb : Msg from DS slot <slot#> cmd = <msg>. RECEIVED CONTROL slot <slot#> tag <tag#> size <size> content : 0 0 0 1 0 18 0 1 0 C4 0 1 8F EA 9B content : <x x x x> <x x> <x x> <x x> <x x> <x x x>

DSP msg <msg> received

tag# len chan msg proc rtp_header

Voice over IP for the Cisco AS5800 65

Debug Commands

Related Commands

Table 7 describes the fields in previous example.

Table 7 debug vrm control Field Descriptions

Field Description

tag# DSP number.

len Length of the packet from the RTP header (the next two bytes).

chan Channel number (the next two bytes).

msg Message ID number (the next two bytes).

proc Process ID (the next two bytes).

rtp_header First three bytes of the RTP header.

Command Description

debug vrm error Displays debug messages for all DSP-specific error messages going to the voice

resource manager (VRM).

debug vrm inout Displays debug messages for all DSP-specific messages going to and coming

from the voice resource manager (VRM).

Cisco IOS Release 12.0(7)T

debug vrm error

To display all DSP-specific error messages going to the voice resource manager (VRM), use the debug vrm error privileged EXEC command. To stop displaying DSP-specific error messages, use the no form of this command.

[no] debug vrm error

Syntax Description

There are no arguments or keywords used in this command.

Defaults

No default behavior or values.

Command History

Release Modification

12.0(7)T This command was introduced.

debug vrm error

Examples

The following examples show some possible outputs from the debug vrm error command, displaying DS_specific error messages.

This example shows that an error occurred when sending data from the DSP to IP network (ingress direction):

- vrm_vtsp_send_ingress_data : fs_input failed

This error message shows that an error occurred when sending control message from the DSP to VTSP:

- vrm_vtsp_send_ingress_control : failed

This error message shows that there is no voice card present and a voice call is attempted:

- vrm_vtsp: No Voice Card ready yet.

This error message shows that no free resource is available, and a voice call is attempted:

- vrm_vtsp_open : vdev_common not available

This error message shows that there is already an active call on this channel, so abort:

- vrm_vtsp_open : vchan_instance already in use ABORT OPEN

The following messages show that the VTSP did a “dirty close” on a particular channel. “Dirty close” means that the DSP did not respond to the VTSP's request for the final statistics of the call.

- vrm_vtsp_open : cdb->dsp_info not NULL Abort OPEN

- vrm_vtsp_close failure no vtsp_cdb_ptr

- vrm_vtsp_close: without a dsp_info!

- vrm_vtsp_close : dirty close on tag <tag#> channel <chan#>

The following error mesaage describes the status of the DSP (virtual device):

- vrm_vtsp_close : vdev freed not locked. Status <value>

Voice over IP for the Cisco AS5800 67

Debug Commands

Possibe status values are as follows:

• ACTIVE_CALL = 0x0001

• BUSYOUT_REQ = 0x0002

• BAD = 0x0004

• BACK2BACK_TEST = 0x0008

• RESET = 0x0010

• DOWNLOAD_FILE = 0x0020

• DOWNLOAD_FAIL = 0x0040

• SHUTDOWN = 0x0080

• BUSY = 0x0100

• OIR = 0x0200

• HASLOCK = 0x0400 /* vdev_pool has locked port */

• DOWNLOAD_REQ = 0x0800

• RECOVERY_REQ = 0x1000

• NEGOTIATED = 0x2000

• OOS = 0x4000

The following error message shows that a "set_codec" command was issued, but the codec was not supported by the DSP:

- VTSP_FAIL: codec <value> not supported

Possible codec values are as follows:

• 0 = voipCodecG729,

• 1 = voipCodecG729a,

• 2 = voipCodecG726r16,

• 3 = voipCodecG726r24,

• 4 = voipCodecG726r32,

• 5 = voipCodecG711ulaw,

• 6 = voipCodecG711Alaw,

• 7 = voipCodecG728,

• 8 = voipCodecG723r63,

• 9 = voipCodecG723r53,

• 10 = voipCodecGSM,

Cisco IOS Release 12.0(7)T

• 11 = voipCodecG729b,

• 12 = voipCodecG729ab,

• 13 = voipCodecG723ar63,

• 14 = voipCodecG723ar53,

• 15 = voipCodecG729IETF

debug vrm error

This error message shows that there is no buffer left in the pool for the VTSP to send a message to the DSP. <Number> int his output referst o the number of times the VRM ran out of buffer space.

- vrm_vtsp_get_packet: no buffers <number>

This error message notifies the VRM of a DSP alarm:

- vrm_vtsp_indicate_alarm : alarm_type <value> slot <slot#> tag <tag#> chan <chan#>

Possible values for the alarm are as follows:

• FATAL_ERROR = 0x01

• MEMORY_ERROR = 0x02

• BUFFER_ERROR = 0x04

• DOWNLOAD_ERROR = 0x08

• CHECKSUM_ERROR = 0x10

This eror message shows that the DSP sent a defective message:

- vrm msg offset too big tag <tag#> vchan <chan#>

Table 8 expains the field contained in the previous example.

Table 8 debug vrm error Field Descriptions

Field Description

slot# Slot in the Cisco AS5800 where the VFC is installed.

tag# DSP number. Possible values for this field are 1 to 96.

chan# Channel number. Possible values for this field are 1 and 2.

This error message indicates that an alarm message was received from the VFC/DSP and was successfully sent to the VTSP:

- vrm_msg_process_alarm_msg for <slot#>.<tag#>.<chan#> , state=<value>

Possible state values are as follows:

• 0 = RESET

• 1 = ADMINDOWN

• 2 = CORE_READY

• 3 = CODEC_READY

• 4 = VOICE_IDLE

• 5 = FAX_IDLE

• 6 = VOICE_READY

• 7 = FAX_READY

• 8= DTMF_READY

Voice over IP for the Cisco AS5800 69

Debug Commands

Related Commands

Command Description

debug vrm control Displays debug messages for all DSP-specific control messages going to the

voice resource manager (VRM).

debug vrm inout Displays debug messages for all DSP-specific messages going to and coming

from the voice resource manager (VRM).

Cisco IOS Release 12.0(7)T

debug vrm inout

To display debug messages for all DSP-specific messages going to and coming from the voice resource manager (VRM), use the debug vrm inout privileged EXEC command. To stop displaying DSP-specific messages, use the no form of this command.

[no] debug vrm inout

Syntax Description

There are no arguments or keywords used in this command.

Defaults

No default behavior or values.

Command History

Release Modification

12.0(7)T This command was introduced.

debug vrm inout

Examples

Related Commands

The following example displays DSP-specific messages going to the VRM when a call is made:

*Jun 17 13:02:41.495:vrm_vtsp_open :vtsp_cdb_ptr 623D2170 *Jun 17 13:02:41.495:vrm_vtsp_open :VTSP_SUCCESS *Jun 17 13:02:41.535:vrm_vtsp_get_capabilities :vtsp_cdb_ptr 623D2170 *Jun 17 13:02:41.535:vrm_vtsp_get_capabilities :vtsp_cdb_ptr 623D2170 *Jun 17 13:02:41.535:vrm_vtsp_set_codec :vtsp_cdb_ptr 623D2170 new_codec 5 *Jun 17 13:02:41.535:VTSP_SUCCESS:Codec 5 was loaded already. *Jun 17 13:02:41.767:vrm_vtsp_set_codec :vtsp_cdb_ptr 623D2170 new_codec 5 *Jun 17 13:02:41.767:VTSP_SUCCESS:Codec 5 was loaded already.

The following example displays DSP-specific messages going to the VRM when a call is complete:

*Jun 17 13:02:49.119:vrm_vtsp_close :vtsp_cdb_ptr 623D2170 *Jun 17 13:02:49.119:vrm_vtsp_close :0x2 close OK

Command Description

debug vrm control Displays debug messages for all DSP-specific control messages going to the

voice resource manager (VRM).

debug vrm error Displays debug messages for all DSP-specific error messages going to the voice

resource manager (VRM).

Voice over IP for the Cisco AS5800 71

Glossary

AAA—Authentication, Authorization, and Accounting. AAA is a suite of network security services that provide the primary framework through which access control can be set up on your Cisco router or access server.

ACOM—Term used in G.165, “General Characteristics of International Telephone Connections and International Telephone Circuits: Echo Cancellers.” ACOM is the combined loss achieved by the echo canceller, which is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.

a-law—A voice compression technique commonly used in Europe.

ANI—Answer Number Indication. The calling number (number of calling party).

ARQ—Admission request.

Call leg—A logical connection between the router and either a telephony endpoint over a bearer

channel, or another endpoint using a session protocol.

CAS—Channel Associated Signaling. In E1 applications, timeslot 16 is used to transmit CAS information. Each frame’s timeslot 16 carries signaling information (ABCD bits) for two of the B channel timeslots.

CIR—Committed Information Rate. The average rate of information transfer a subscriber (for example, the network administrator) has stipulated for a Frame Relay PVC.

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 signals. In Voice over IP, it specifies the voice coder rate of speech for a dial peer.

Data Link Connection Identifier (DLCI)—Frame Relay virtual circuit number corresponding to a particular destination. The DLCI is part of the Frame Relay header and is usually 10 bits long.

Dial peer—An addressable call endpoint. In Voice over IP, there are two kinds of dial peers: POTS and VoIP.

DNS—Domain Name System used to address translation to convert H.323 IDs, URLs, or e-mail IDs to IP addresses. DNS is also used to assist in the location of remote gatekeepers and to reverse-map raw IP addresses to host names of administrative domains.

DNIS—Dialed number identification service. The destination number.

DS0—A 64-Kbps channel on an E1 or T1 WAN interface.

DSP—Digital Signal Processor.

DTMF—Dual tone multifrequency. Use of two simultaneous voice-band tones for dial (such as

touch tone).

E.164—The international public telecommunications numbering plan. A standard set by ITU-T which addresses telephone numbers.

E1—Wide-area digital transmission scheme. E1 is the European equivalent of a T1 line. The E1’s higher clock rate (2.048 MHz) allows for 32 64-Kbps channels, which include one channel for framing and one channel for D-channel information.

Cisco IOS Release 12.0(7)T

E&M—Ear and mouth RBS signaling.

Endpoint—An H.323 terminal or gateway. An endpoint can call and be called. It generates and/or

terminates the information stream.

FIFO—First-in, first-out. In data communication, FIFO refers to a buffering scheme where the first byte of data entering the buffer is the first byte retrieved by the CPU. In telephony, FIFO refers to a queuing scheme where the first calls received are the first calls processed.

Gatekeeper—A gatekeeper maintains a registry of devices in the multimedia network. The devices

The gatekeeper is an H.323 entity on the LAN that provides address translation and control access to the LAN for H.323 terminals and gateways. The gatekeeper may provide other services to the H.323 terminals and gateways, such as bandwidth management and locating gateways.

Gateway—A gateway allows H.323 terminals to communicate with non-H.323 terminals by converting protocols. A gateway is the point at which a circuit-switched call is encoded and repackaged into IP packets.

An H.323 gateway is an endpoint on the LAN that provides real-time two-way communications between H.323 terminals on the LAN and other ITU-T terminals in the WAN, or to another H.323 gateway.

H.323—An International Telecommunication Union (ITU-T) standard that describes packet-based video, audio, and data conferencing. H.323 is an umbrella standard that describes the architecture of the conferencing system, and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its actual protocol.

Glossary

H.323 RAS—Registration, admission, and status. The RAS signaling function performs registration, admissions, bandwidth changes, status and disengage procedures between the VoIP gateway and the gatekeeper.

HSRP—Hot Standby Routing Protocol. HSRP is a Cisco proprietary protocol which provides a redundancy mechanism when more than one router is connected to the same segment/subnet of an Ethernet/FDDI/Token Ring network.

ISDN—Integrated Services Digital Network. ISDN is a communications protocol, offered by telephone companies, that permits telephone networks to carry data, voice, and other traffic.

ITU-T—Telecommunication standardization sector of ITU.

IVR—Integrated voice response. A software feature that allows the use of one of several interactive

voice response scripts during the call processing functionality.

LEC—Local exchange carrier.

LRQ—Location request.

MCU—Multipoint control unit

mu-law—a-law—A voice compression technique commonly used in North America.

Multicast—A process of transmitting PDUs from one source to many destinations. The actual

mechanism (that is, IP multicast, multi-unicast, etc.) for this process may be different for LAN technologies.

Multilink PPP—Multilink Point-to-Point Protocol. This protocol is a method of splitting, recombining, and sequencing datagrams across multiple logical data links.

Multipoint-unicast—A process of transferring Protocol Data Units (PDUs) where an endpoint sends more than one copy of a media stream to different endpoints. This may be necessary in networks which do not support multicast.

node—An H.323 entity that uses RAS to communicate with the gatekeeper. For example, an endpoint such as a terminal, proxy, or gateway.

PDU—Protocol Data Units. Used by bridges to transfer connectivity information.

Voice over IP for the Cisco AS5800 73

Glossary

PBX—Private Branch Exchange. Privately-owned central switching office.

PLAR—Private Line Auto Ringdown. This type of service results in a call attempt to some

particular remote endpoint when the local extension is taken off-key.

POTS—Plain Old Telephone Service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the Public Switched Telephone Network.

POTS dial peer—Dial peer connected via a traditional telephony network. POTS peers point to a particular voice-port on a voice network device.

PRI—Primary Rate Interface. PRI is an ISDN interface to primary rate access. Primary rate access consists of a single 64 Kbps D channel plus 23 T1 or 30 E1 B channels for voice or data.

PSTN—Public Switched Telephone Network. PSTN refers to the local telephone company.

PVC—Permanent Virtual Circuit.

QoS—Quality of Service, which refers to the measure of service quality provided to the user.

RAS—Registration, Admission, and Status Protocol. This is the protocol that is used between

endpoints and the gatekeeper to perform management functions.

RBS—Robbed Bit Signaling

RRQ—Registration request.

RSVP—Resource Reservation Protocol. This protocol supports the reservation of resources across

an IP network.

T1—Digital WAN carrier facility. T1 transmits DS-1 formatted data at 1.544 Mbps through the telephone-switching network, using AMI or B8ZS coding. T1 is the North American equivalent of an E1 line.

TCL—Tool Command Language. An interpreted script language developed by Dr. John Ousterhout of the University of California, Berkeley, and now developed and maintained by Sun Microsystems Laboratories.

U-law—A companding technique commonly used in North America. U-law is standardized as a 64-Kbps codec in ITU-T G.711.

SPI—Service provider interface.

TDM—Time division multiplexing. Technique in which information from multiple channels can be

allocated bandwidth on a single wire based on preassigned time slots. Bandwidth is allocated to each channel regardless of whether the station has data to transmit.

Vo IP —Voice over IP. The ability to carry normal telephone-style voice over an IP-based internet with POTS-like functionality, reliability, and voice quality. VoIP is a blanket term which generally refers to Cisco’s standards based (H.323, etc.) approach to IP voice traffic.

VoIP dial peer—Dial peer connected via a packet network; in the case of Voice over IP, this is an IP network. VoIP peers point to specific VoIP devices.

VTSP—Voice telephony service provider.

Cisco IOS Release 12.0(7)T

Zone—A collection of all terminals (tx), gateways (GW), and Multipoint Control Units (MCU) managed by a single gatekeeper (GK). A Zone includes at least one terminal, and may or may not include gateways or MCUs. A Zone has only one gatekeeper. A Zone may be independent of LAN topology and may be comprised of multiple LAN segments which are connected using routes or other devices.

Cisco Systems AS5800 User Manual 3

Specifications and Main Features

Frequently Asked Questions

User Manual

Outbound Dialing on POTS Peers

Direct Inward Dial for POTS Peers

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

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Related Commands