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.

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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

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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

Related Documents

• Voice, Video, and Home Applications Configuration Guide, Cisco IOS Release 12.0

Cisco IOS Release 12.0(7)T

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

• Quality of Service Configuration Guide, Cisco IOS Release 12.0

• Quality of Service Command Reference, Cisco IOS Release 12.0

• Voice over IP for the Cisco AS5800 Software Configuration Guide, Cisco IOS

Release 12.0(4)XL.

Supported Platforms

• Cisco AS5800 universal access servers

• Cisco AS5300 access servers

• Cisco 2600 series routers

• Cisco 3600 series routers

Supported Standards, MIBs, and RFCs

Supported Platforms

Standards

None

MIBs

• IF-MIB

• ENTITY-MIB.my

• CISCO-ENTITY-VENDORTYPE-OID-MIB.my

• DIAL-CONTROL-MIB.my

• CISCO-DIAL-CONTROL-MIB.my

• CISCO-VOICE-DIAL-CONTROL-MIB.my

• CISCO-VOICE-IF-MIB.my

• CISCO-DSP-MGMT-MIB.my

• CISCO-MMAIL-DIAL-CONTROL-MIB.my

• CISCO-CAS-IF-MIB.my

For descriptions of supported MIBs and how to use MIBs, see the Cisco MIB web site on CCO at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.

RFCs

None

Voice over IP for the Cisco AS5800 5

Prerequisites

Before you can configure your Cisco AS5800 to use Voice over IP, you must first:

• Install a version of the Cisco IOS software that supports DSPWare 3.1.7 specific to the Cisco

AS5800 (for example, Cisco IOS Release 12.0(4)XL or Cisco IOS Release 12.0(7)T).

• Establish a working IP network. For more information about configuring IP, refer to the

“IP Overview,” “Configuring IP Addressing,” and “Configuring IP Services” chapters in the Cisco IOS 12.0 Network Protocols Configuration Guide, Part 1.

• Complete basic configuration for the AS5800. This includes, as a minimum, the following tasks:

— Configure a host name and password for the AS5800

— Configure the Fast Ethernet interface of your AS5800 so that it can be recognized as a device

— Configure the AS5800 interfaces for ISDN PRI lines

— Configure the ISDN D channels for each ISDN PRI line

— Configure the AS5800 interfaces for T1 CAS lines

— Configure the ISDN D channels for each T1 CAS PRI line

on the Ethernet LAN

For more information about any of the these configuration tasks, refer to the Cisco AS5800 Universal Access Server Software Installation and Configuration Guide, which shipped with your Cisco AS5800 and is available on the document CD-ROM.

• Install the VFC into the appropriate slot of your Cisco AS5800 universal access server. Each VFC

can hold up to 16 digital signal processor modules (DSPMs), enabling processing for up to 192 voice channels. For more information about the physical characteristics of the VFCs or DSPMs, or how to install them, refer to Installing Voice over IP Feature Cards in Cisco AS5800 Universal Access Servers document that shipped with your VFC and is available online.

• Complete your company’s dial plan.

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

Cisco IOS Release 12.0(7)T

Configuration Tasks

• Integrate your dial plan and telephony network into your existing IP network topology. Merging

your IP and telephony networks depends on your particular IP and telephony network topology. In general, we recommend the following suggestions:

— Use canonical numbers wherever possible. It is important that you avoid situations where

numbering systems are significantly different on different routers or access servers in your network.

— Make routing and dialing transparent to the user. For example, avoid secondary dial tones

from secondary switches, where possible.

— Contact your PBX vendor for instructions about how to reconfigure the appropriate PBX

interfaces.

• Configure another device in your network (preferably a Cisco 2600 or Cisco 3600 series router)

to act as a gatekeeper. The Service Provider implementation of Voice over IP is configured using 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. Unless it has a gatekeeper to interact with, it will periodically query all devices in the network, searching for a gatekeeper. For more information about configuring gatekeepers, refer to the Cisco IOS Release 12.0(3)T Service Provider Features for Voice over IP feature module.

Configuration Tasks

After you have analyzed your dial plan and decided how to integrate it into your existing IP network, you are ready to configure your network devices to support Voice over IP. The actual configuration procedure depends entirely on the topology of your voice network, but, in general, you need to complete the following tasks:

• Configuring IP Networks for Real-Time Voice Traffic

• Configuring Voice Ports

• Configuring Dial Peers

• Configuring the Cisco AS5800 as an H.323 Gateway

• Configuring the Cisco AS5800 for Interactive Voice Response

Configuring IP Networks for Real-Time Voice Traffic

You need to have a well-engineered network end-to-end when running delay-sensitive applications such as VoIP. Fine-tuning your network to adequately support VoIP involves a series of protocols and features geared toward Quality of Service (QoS). It is beyond the scope of this document to explain the specific details relating to wide-scale QoS deployment. Cisco IOS software provides many tools for enabling QoS on your backbone, such as Random Early Detection (RED), Weighted Random Early Detection (WRED), Fancy Queuing (meaning custom, priority, or weighted fair queuing), and IP Precedence. To configure your IP network for real-time voice traffic, you need to take into consideration the entire scope of your network, then select the appropriate QoS tool or tools. In addition, you must use the Cisco IOS ip cef command to ensure that Cisco Express Forwarding (CEF) is enabled.

QoS must be configured throughout your network—not just on the Cisco AS5800 devices running VoIP—to improve voice network performance. Not all QoS techniques are appropriate for all network routers. Edge routers and backbone routers in your network do not necessarily perform the

Voice over IP for the Cisco AS5800 7

Configuration Tasks

same operations; the QoS tasks they perform might also differ. To configure your IP network for real-time voice traffic, you need to consider the functions of both edge and backbone routers in your network, then select the appropriate QoS tool or tools.

In general, edge routers perform the following QoS functions:

• Packet classification

• Admission control

• Bandwidth management

• Queuing

In general, backbone routers perform the following QoS functions:

• High-speed switching and transport

• Congestion management

• Queue management

Scalable QoS solutions require cooperative edge and backbone functions.

Configuring Custom Queuing and IP RTP Reserve

Although not required, you can use the custom queuing QoS tool to fine-tune your network for real-time voice traffic. Real-time voice traffic is carried on UDP ports ranging from 16384 to 32767. Custom Queuing and other methods for identifying high priority streams should be configured for these port ranges. For more information about custom queuing, refer to the “Congestion Management” chapter in the Cisco IOS Release 12.0 Quality of Service Configuration Guide. For more information about configuring IP RTP Priority, refer to the Cisco IOS Release 12.0(5)T IP RTP Priority feature module.

Configuring Voice Ports

When an ISDN interface on the Cisco AS5800 is carrying voice data, it is referred to as a voice port.

Note A voice port was created automatically when you installed the VFC in the Cisco AS5800 and

configured an ISDN PRI group. Configuring an ISDN PRI group is part of the basic Cisco AS5800 configuration procedure. For more information, refer to the Cisco AS5800 Universal Access Server Software Installation Configuration Guide.

Signaling in Voice over IP for the AS5800 is handled by ISDN PRI group configuration. After ISDN PRI is configured for both B and D channels for both ISDN PRI lines, you need to issue the isdn incoming-voice command on the serial interface (acting as the D channel) to ensure a dial tone.

Under most circumstances, the default voice-port command values are adequate to configure voice ports to transport voice data over your existing IP network. Because of the inherent complexities involved with PBX networks, you might need specific voice-port values configured, depending on the specifications of the devices in your telephony network. For more information on specific voice-port configuration commands, refer to either the Cisco IOS Release 12.0(3)T Voice over IP for

the Cisco AS5300 feature module or the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference.

Cisco IOS Release 12.0(7)T

To configure basic ISDN parameters for Voice over IP on the Cisco AS5800, perform the following steps:

Step Command Purpose

Router(config)# isdn switch-type switch-type

Router(config)# controller T1 1/0/0 or Router(config)# controller T1 1/0/0:1

Router(config)# framing esf

Router(config)# linecode value

Router(config)# pri-group timeslots range

Router(config)# controller T1 1/0/1 or Router(config)# controller T1 1/0/0:2

Router(config)# framing esf

Router(config)# linecode value

Router(config)# pri-group timeslots range

Router(config)# interface Serial1/0/0:23 or Router(config)# interface Serial1/0/0:1:23

Router(config)# isdn incoming-voice modem

Router(config)# interface Serial1/0/1:23 or Router(config)# interface Serial1/0/0:2:23

Router(config)# isdn incoming-voice modem

Defines the telephone company’s switch type.

Enables the T1 0 controller on the T1 card and enters controller configuration mode, or

Enables the T1 1 controller on the T3 card and enters controller configuration mode.

Defines the framing characteristics.

Sets the line code type to match that of your telephone company service provider.

Configures ISDN PRI.

Enables the T1 1 on the T1 card controller and enters controller configuration mode, or

Enables the T1 2 controller on the T3 card and enters controller configuration mode.

Defines the framing characteristics.

Sets the line code type to match that of your telephone company service provider.

Configures ISDN PRI.

Configures the channel for the first ISDN PRI line on the T1 card. (The ISDN serial interface is the D channel.) or

Configures the channel for the first ISDN PRI line on the T3 card.

Enables incoming ISDN voice calls. This command has two possible keywords: data and modem. You must use the modem keyword to enable voice calls. The modem keyword represents bearer capabilities of speech.

Configures the channel for the second ISDN PRI line.or

Configures the channel for the second ISDN PRI line on the T3 card.

Configuring Voice Ports

As mentioned, under most circumstances, the default voice-port command values are adequate to configure voice ports to transport voice data over your existing IP network. If you need to configure specific voice port parameters, perform the following steps beginning in privileged EXEC mode:

Step Command Purpose

Router# configure terminal

Router(config)# voice-port {shelf/slot/port:D}|

{shelf/slot/parent:port:D}

Enters global configuration mode.

Identifies the voice port you want to configure and enters voice-port configuration mode.

Voice over IP for the Cisco AS5800 9

Configuration Tasks

Step Command Purpose

Router(config-voiceport)# cptone country

Selects the appropriate voice call progress tone for this interface.

The default for this command is us. For a list of supported countries, refer to the Multiservice Applications Command Reference.

Router(config-voiceport)# compand-type {a-law|u-law}

Router(config-voiceport)# connection {plar string|trunk

string}

Selects a companding type for this voice port.

(Optional) Specifies either the trunk connection or the private line auto ringdown (PLAR) connection. The string value specifies the destination telephone number.

Router(config-voiceport)# music-threshold number

(Optional) Specifies the threshold (in decibels) for on-hold music. Valid entries are from –70 to –30.

Router(config-voiceport)# description string

(Optional) Attaches descriptive text about this voice port connection.

Fine-Tuning ISDN Voice Ports

Depending on the specifics of your particular network, you may need to adjust voice parameters involving timing, input gain, and output attenuation for voice ports. Collectively, these commands are referred to as voice-port tuning commands.

Note In most cases, the default values for voice-port tuning commands will be sufficient.

To fine-tune ISDN voice ports, use the following commands beginning in privileged EXEC mode:

Step Command Purpose

Router# configure terminal

Router(config)# voice-port {shelf/slot/port:D} |

{shelf/slot/parent:port:D}

Router(config-voiceport)# input gain value

Router(config-voiceport)# output attenuation value

Router(config-voiceport)# echo-canel enable

Router(config-voiceport)# echo-canel coverage value

Router(config-voiceport)# non-linear

Router(config-voiceport)# playout-delay {maximum

milliseconds|nominal milliseconds}

Router(config-voiceport)# timeouts initial seconds

Enters global configuration mode.

Identifies the voice port you want to configure and enter voice-port configuration mode.

Specifies (in decibels) the amount of gain to be inserted at the receiver side of the interface. Acceptable values are from –6 to 14.

Specifies (in decibels) the amount of attenuation at the transmit side of the interface. Acceptable values are from 0 to 14.

Enables echo-cancellation of voice that is sent out the interface and received back on the same interface.

Adjusts the size (in milliseconds) of the echo-cancel. Acceptable values are 16, 24, and 32.

Enables non-linear processing, which shuts off any signal if no near-end speech is detected. (Non-linear processing is used with echo-cancellation.)

Specifies the amount of time in milliseconds configured for the playout delay buffer.

Specifies the number of seconds the system will wait for the caller to input the first digit of the dialed digits. Valid entries for this command are from 0 to 120.

Cisco IOS Release 12.0(7)T

Step Command Purpose

Router(config-voiceport)# timeouts interdigits seconds

Router(config-voiceport)# timeouts ringing

{seconds|infinity}

Router(config-voiceport)# timeouts wait-release

{seconds|infinity}

Router(config-voiceport)# translate {called

number|calling number}

For more information on specific voice-port configuration commands or additional voice-port commands, refer to either the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module or the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference..

Specifies the number of seconds the system will wait (after the caller has input the initial digit) for the caller to input a subsequent digit. Valid entries for this command are from 0 to 120.

Specifies the number of seconds the system will continue to ring the destination if there is no answer.

Specifies the wait release timeout duration in seconds.

Defines translation rules pertaining to either the called or calling numbers.

Verifying Voice Port Configuration

• Use the show voice port command to verify that the data configured is correct.

Configuring Voice Ports

• If you have not configured your device to support direct inward dial, dial in to the router and see

if you have dial tone.

• Enter DTMF digit. If the dial tone stops, you have two-way voice connectivity with the router.

Troubleshooting Tips

If you are having trouble connecting a call, and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:

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

destination, refer to the “Configuring IP” chapter in the Cisco IOS 12.0 Network Protocols Configuration Guide, Part 1.

• Check to see that the VFC has been correctly installed.

• Use the show dial-shelf command to see if the VFC is operational.

• Use the show vrm vdevices summary command to verify that you have voice devices available.

• Use the show isdn status command to view layer status information. If you receive a status

message stating that Layer 1 is deactivated, make sure the cable connection is not loose or disconnected. (This status message indicates a problem at the physical layer.)

• With T1 lines, check to see if your u-law setting is correct. With E1 lines, check to see if your

a-law setting is correct. Use the cptone command to configure both a-law or u-law values. For more information about the cptone command, refer to the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module.

• If dialing cannot occur, use the debug isdn q931 command to check the ISDN configuration.

Voice over IP for the Cisco AS5800 11

Configuration Tasks

Configuring Dial Peers

The key point to understanding how VoIP functions is to understand dial peers. Each dial peer defines the characteristics associated with a call leg, as shown in Figure 3 and Figure 4. A call leg is a discrete segment of a call connection that lies between two points in the connection. All of the call legs for a particular connection have the same connection ID.

There are two different kinds of dial peers:

• POTS—Dial peer describing the characteristics of a traditional telephony network connection.

POTS peers point to a particular voice port on a voice network device.

• VoIP—Dial peer describing the characteristics of a packet network connection. VoIP peers point

to specific VoIP devices.

An end-to-end call comprises four call legs, two from the perspective of the source access server as shown in Figure 3, and two from the perspective of the destination access server as shown in Figure 4. A dial peer is associated with each call leg. Dial peers are used to apply attributes to call legs and to identify call origin and destination. Attributes applied to a call leg include QoS, codec, VAD, and fax rate.

Figure 3 Dial Peer Call Legs from the Perspective of the Source Router

Source

Source router

Call leg for POTS

dial peer 1

Figure 4 Dial Peer Call Legs from the Perspective of the Destination Router

Destination

IP cloud

Call leg for VoIP

dial peer 2

Call leg for VoIP

dial peer 3

IP cloud

Call leg for POTS

Destination router

Destination

dial peer 4

Source

10353

10354

Inbound versus Outbound Dial Peers

Dial peers are used for both inbound and outbound call legs. It is important to remember that these terms are defined from the access server’s perspective. An inbound call leg originates outside the access server. An outbound call leg originates from the access server.

For inbound call legs, a dial peer might be associated to the calling number or the port designation. Outbound call legs always have a dial peer associated with them. The destination pattern is used to identify the outbound dial peer. The call is associated with the outbound dial peer at setup time.

Cisco IOS Release 12.0(7)T

POTS peers associate a telephone number with a particular voice port so that incoming calls for that telephone number can be received and outgoing calls can be placed. VoIP peers point to specific devices (by associating destination telephone numbers with a specific IP address) so that incoming calls can be received and outgoing calls can be placed. Both POTS and VoIP peers are needed to establish VoIP connections.

Configuring POTS Peers

POTS peers enable incoming calls to be received by a particular telephony device. To configure a POTS peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its telephone numbers, and associate it with a voice port through which calls will be established. Under most circumstances, the default values for the remaining dial peer configuration commands will be sufficient to establish connections.

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

Step Command Purpose

Router(config)# dial-peer voice

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

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

Router(config-dial-peer)# prefix string

number pots

Enters the dial peer configuration mode to configure a POTS peer. The number value of the dial-peer voice

pots command is a tag that uniquely identifies the dial peer.

Defines the telephone number associated with this POTS dial peer.

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

(Optional) Specifies the prefix for this POTS dial peer. The prefix string value is sent to the telephony interface first, before the telephone number (destination pattern) associated with this dial peer is sent.

Configuring Dial Peers

For additional POTS dial-peer configuration commands, 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.

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

+ 51 hidden pages

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