Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2 - Configuring Trunk Connections and Conditioning Features [Cisco IOS Software Releases 12.2 Mainline]

Table Of Contents

Configuring Trunk Connections and Conditioning Features

Trunking Overview

Simulated Lines and Trunks

Trunk Conditioning Signaling Attributes

Congestion Monitoring and Management Features

T1/E1 Alarm Conditioning

PSTN Fallback

Calculated Impairment Planning Factor

Service Assurance Agent

Busyout

Local Voice Busyout

Advanced Voice Busyout

Busyout Monitor

Trunk Management Prerequisite Tasks

Configuring Trunk-Conditioning Signaling Attributes

Assigning Trunk-Conditioning Attributes to Network Dial Peers

Assigning Voice Classes to Voice Ports

Verifying the Signaling Attributes and Trunk Conditioning

Configuring Trunk Connections

Configuring PLAR (Switched) Connections

Configuring Trunk/Tie-Line Connections

Configuring PLAR-OPX Connections

Configuring T1/E1 Alarm Generation Parameters

Verifying Alarm-Generation Parameters

Configuring PSTN Fallback

Configuring Fallback to Alternate Dial Peers

Configuring Destination Monitoring without Fallback to Alternate Dial Peers

Configuring Call Fallback Cache Parameters

Configuring Call Fallback Jitter-Probe Parameters

Configuring Call Fallback Probe-Timeout and Weight Parameters

Configuring Call Fallback Threshold Parameters

Configuring Call Fallback Map Parameters

Verifying PSTN Fallback Configuration

Troubleshooting Tips

Monitoring and Maintaining PSTN Fallback

Configuring Local Voice Busyout

Configuring the Busyout Trigger Event

Configuring Busyout of Voice Ports

Configuring a Voice Port to Monitor the Link to a Remote Interface

Configuring a Busyout Monitoring Voice Class

Trunk Connections and Conditioning Configuration Examples

Trunk Conditioning Configuration Example

Voice Class for VoFR and VoATM Dial Peers Configuration Example

Voice Class for Voice Ports Configuration Example

Voice Class for Default Signaling Patterns Configuration Example

Voice Class for Specified Signaling Patterns Configuration Example

PLAR (Switched Calls) Configuration Example

Permanent Trunks Configuration Example

Congestion Monitoring and Management Configuration Examples

Configuring PSTN Fallback for VoIP over Frame Relay Example

Configuring PSTN Fallback for VoIP over MLP Example

Local Voice Busyout Configuration Examples

Alarm Trigger for Busyout of Voice Ports Configuration Example

Configuring Trunk Connections and Conditioning Features

This chapter describes trunk connections and conditioning features for the Cisco 2600 and 3600 series routers and MC3810 multiservice concentrators. The features include trunk conditioning, tie-line simulation, T1/E1 alarms, Public Switched Telephone Network (PSTN) fallback, and busyout. This chapter contains:

•Trunking Overview

•Trunk Conditioning Signaling Attributes

•Congestion Monitoring and Management Features

•Trunk Management Prerequisite Tasks

•Trunk Management Configuration Tasks List

•Configuring T1/E1 Alarm Generation Parameters

•Trunk Connections and Conditioning Configuration Examples

•Congestion Monitoring and Management Configuration Examples

For a complete description of the commands in this chapter, refer to the Cisco IOS Voice, Video, and Fax Command Reference. To locate documentation of other commands that appear in this chapter, use the command reference master index or search online.

To identify the hardware platform or software image information associated with a feature in this chapter, use the Feature Navigator on Cisco.com to search for information about the feature or refer to the software release notes for a specific release. For more information, see the "Identifying Supported Platforms" section in the "Using Cisco IOS Software" chapter.

Trunking Overview

A trunk is a communication line between two switching systems—the switching equipment in a central office (CO) and PBX. It is a physical and logical point-to-point connection with a permanent wire over which network traffic travels. A backbone is composed of a number of trunks.

Voice over IP (VoIP) simulates trunk connections. The simulated connections occur between PBXs that are connected to Cisco routers or access servers on each side of the network

In Figure 103, two PBXs are connected to a router using a simulated trunk and a recEive and transMit (E&M) voice port. In this case, a permanent, non-switched connection transparently connects the two PBXs.

Figure 103 Simulated Trunk Connection

Simulated Lines and Trunks

Simulated lines and trunks enable a telephone user at one location to dial an access code to access a PBX at another location. A second dial tone can be heard coming from the remote PBX. There are two types of simulated connections—switched and permanent—that can be configured for both analog and digital systems. The connections are created with the Cisco connection command.

The connection trunk command creates a permanent call that is connected as soon as the routers on each end are booted (see Figure 104). Permanent calls pass limited telephony signaling and operate without collecting digits or requiring changes to the overall dial plan.

Figure 104 Connection Trunk Configuration

The calls simulate a permanent tie-line between two PBXs. Both ends must be configured and have compatible voice-port signaling that is:

•E&M to E&M

•Foreign Exchange Office (FXO) to Foreign Exchange Station (FXS)

The signaling cannot be FXO to ground start.

When a switched call is configured (see Figure 105), the user can make a call without dialing any digits. The telephony signaling, such as hookflash, is not passed. The call will not roll over to voice mail if the remote telephone does not answer and digits from an attached telephony device are not collected.

Figure 105 Connection Private-Line Auto Ringback (PLAR) Configuration

The switched call configuration works with any type of voice port (E&M, FXO, or FXS) and can be used without any effect on an existing dial plan. It is commonly used to connect PBXs in which the remote devices appear to be physical extensions. The PBX provides dial tone to the extensions, not the router.

The connection tie-line command creates a switched call between two stations or PBXs that bypasses the switch. The connection plar-opx command creates a call that is similar to a switched call. The connection does not take place between the PBX and the local router until the far-end FXS device answers. This enables the PBX to provide centralized voice mail or attendant services when the remote device does not answer.

Trunk Conditioning Signaling Attributes

Trunk conditioning signaling attributes apply to permanent point-to-point voice connections (private lines and tie-lines) created using the connection trunk command. This feature provides the following capabilities:

•Creation of voice classes.

•Specific signaling attributes in each voice class.

•Signaling attributes in the voice class for Voice over Frame Relay (VoFR) and Voice over Asynchronous Transfer Mode (VoATM) dial peers.

Trunk conditioning enables greater control over Cisco private-line calls that are sent over Frame Relay or ATM networks. When private-line or tie-line calls are sent between two PBXs, fault indications are sent to the sending PBX. If the call fails, the PBX is able to select an alternate path to route the calls. Selecting an alternate path applies to analog connections or digital T1/E1 using channel-associated signaling (CAS)/robbed-bit ABCD signaling. It does not cover common channel signaling (CCS).

When T1/E1 CAS is carried in transparent pass-through mode for arbitrary, unknown, or unsupported CAS protocols, it is necessary to define on-hook/idle patterns so that the domain specific part (DSP)/signaling code can sense the idle call state and shut off the flow of voice packets when no active call is in progress. This mode provides an additional idle bandwidth-saving mechanism for those cases when Voice Activity Detection (VAD) is not desired.

Note Cisco MC3810 series concentrators support additional trunk-conditioning features that specify timing, signaling, and transmission options. The features provide enhanced control over call rerouting in cases of trunk failure and increased bandwidth availability due to suppression of voice packets on Out-of-Service (OOS) trunks.

Congestion Monitoring and Management Features

Congestion monitoring of permanent and switched calls is performed with these features: T1/E1 alarm conditioning, PSTN fallback, and busyout functionality including busyout monitoring. These features provides the following capabilities:

•Signaling and suppression of voice traffic for idle or OOS network trunks.

•Busyout of the ports interfacing with a local PBX.

An OOS condition can be signalled using an ABCD bit pattern that is different from the busy or seized state. The difference enables the PBX to differentiate between OOS and congestion.

T1/E1 Alarm Conditioning

Alarm conditioning provides status monitoring on T1/E1 PBX voice interfaces for simulated lines and trunks created using the connection command. It supports operation with CAS, but does not support CCS.

A T1/E1 alarm can be triggered by events detected through the monitoring of a specified set of voice ports within a T1/E1 trunk. A monitored set includes a defined voice port that has a specified DS0 group or groups and configured for one of the following:

•End-to-end connection of permanent virtual circuits (PVCs)

•Busyout of switched virtual circuits (SVCs), where the busyout state is initiated using the busyout monitor command.

When all the monitored voice ports on a T1/E1 trunk are OOS (PVCs are OOS and SVCs are busied out), a T1/E1 Alarm Indication Signal (AIS) is generated on the T1/E1 trunk connected to the PBX or PSTN.

Note Voice ports busied out by the busyout forced command do not trigger a T1/E1 alarm.

PSTN Fallback

PSTN fallback monitors congestion in the IP network and redirects calls to the PSTN or reject calls based on the network congestion. PSTN fallback is supported on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators. For information concerning Voice over IP (VoIP), Voice over ATM (ATM), Calculated Impairment Planning Factor (ICPIF), and Service Assurance Agent (SAA), see the following:

•Cisco IOS Multiservice Applications Configuration Guide

•Cisco IOS Multiservice Applications Command Reference

•Configuring Voice over ATM for the Cisco MC3810

•Voice over ATM on Cisco 3600 Series Routers

•Managing Voice Quality with Cisco Voice Manager (CVM) and Telemate

•Monitoring the Router and Network

PSTN fallback can re-routed calls to an alternate IP destination or to the PSTN if the IP network is found unsuitable for voice traffic at that time. The user defines the congestion thresholds based on the configured network. This functionality enables the service provider to give a reasonable guarantee about the quality of the conversation to their VoIP users at the time of call admission.

Note PSTN fallback does not ensure that a VoIP call is protected from the effects of congestion. This is the function of the other Quality of Service (QoS) mechanisms such as IP Real-Time Transport Protocol (RTP) priority or low latency queueing (LLQ).

PSTN fallback includes the following features:

•Offers flexibility to define the congestion thresholds based on the network by:

–Defining a threshold based on ICPIF, which is derived as part of International Telecommunication Union (ITU) G.113.

–Defining a threshold based solely on packet delay and loss measurements.

•Uses SAA probes to provide packet delay, jitter, and loss information for the relevant IP addresses. Based on the packet loss, delay, and jitter encountered by these probes, an ICPIF or delay/loss value is calculated. See "Service Assurance Agent" section.

•Supports calls of any codec. Only G.729 and G.711 have accurately simulated probes. Calls of all other codecs are emulated by a G.711 probe.

The fallback subsystem has a network traffic cache that maintains the ICPIF or delay/loss values for various destinations. The subsystem helps performance, because new calls to a well-known destination do not have to wait on a probe. The value is usually cached from a previous call.

Once the ICPIF or delay/loss values are calculated and stored, the values remain until the cache ages out or overflows. Until an entry ages out, probes are sent periodically for that destination. The time interval is user configurable. In the following example, it is assumed that call fallback active is enabled and an ICPIF threshold is defined. The call control would be similar if loss and delay thresholds were defined.

Step 1 A call comes into the router. The IP address of the destination is checked against the configured maps to see if it should be sent to another router, such as a backhaul router, or to an alternate dial peer. If it should be sent to another router, the IP address for the fallback subsystem is replaced with the target router. If it should be sent to an alternate dial peer, the router matches that dial peer and obtains the destination information (codec, IP address, and so on).

Note The change is made in the destination address of the probing address. The destination for the actual call is not changed.

Step 2 The router calls the fallback subsystem to look up the specified destination in its network traffic cache. If the ICPIF value exists and is current, then the router uses that value to decide whether to permit the call into the VoIP network. If the router determines that the network congestion is below the configured threshold (by looking at the value from the probe or a cached value), then the call is connected. Otherwise, the router checks the next dial-peer match again in the same way. Eventually, if all the VoIP dial peers are deemed unsuitable, then the call is hairpinned to the PSTN by virtue of a configured POTS dial peer (for analog or digital interfaces). If no PSTN dial peer is present, a fast-busy is sent to the PBX (in case of digital interfaces).

Note It is not possible to signal a fast-busy to some interfaces.

Step 3 The fallback subsystem continues probing in the background periodically (period time is configured by the call fallback probe-timeout command), so that the network congestion information is available when there is a call request. The first call for a particular dial peer may be delayed while the router calculates the congestion information for that destination.

If the timeout threshold is set and the router has not received calls for a particular destination after the threshold expires, then the router removes that destination's traffic information from the cache.

Calculated Impairment Planning Factor

ICPIF calculates an impairment factor for every piece of equipment along the voice path and adds the values to get the total impairment. The ITU assigns the different types of impairments, such as noise, delay, and echo.

The ICPIF handling has been introduced for compatibility with Cisco H.323. Part of ICPIF includes a concept of Total Impairment Value that is a function of loss of packets, delay of packets, and codecs used based on the round-trip reports from SAA. For this feature, all codecs are classified as 729 class codecs or 711 class codecs.

Service Assurance Agent

SAA is a network congestion analysis mechanism. SAA provides delay, jitter, and packet loss information for the configured IP addresses. SAA is based on a client-server protocol defined on UDP. It has an Message Digest 5 (MD5), which is a message authentication algorithm in SNMP v.2. MD5 verifies the integrity of the communication, authenticates the origin, and checks for timeliness.

SAA uses UDP port (port 1976) for sending the SAA control message to the terminating gateway. The SAA probe packets go out on randomly selected ports from the top end of the audio UDP port range (16384 - 32767).

The port pair (RTP & Real-Time Transport Control Protocol [RTCP] port) is selected, and by default SAA for call fallback uses the RTCP port (odd number) to avoid going into the priority queue, if enabled. If fallback is configured to use the priority queue, the RTP port (even number) is selected. The audio UDP port range must be included in the priority queue for fallback priority queueing to work.

Busyout

Three busyout conditions are discussed in the following sections:

•Local Voice Busyout

•Advanced Voice Busyout

•Busyout Monitor

Local Voice Busyout

Local voice busyout is designed to busy out trunks assigned to PVCs so that the PBX does not seize the circuit. Local voice busyout enables the PBX to route a call based on the actual availability of trunks. Local voice busyout enables the following:

•A group of voice ports to be marked busy if a link is broken.

•Specific voice ports in a PVC application to be marked busy under specified conditions.

When ports are marked busy, a call is forced back to the originating equipment (typically a PBX) that reroutes the call over an alternate path. This action ensures that a caller does not experience "dead air" resulting from a connection that never terminates.

The local voice busyout feature provides a way to busy out a voice port if a monitored network interface changes state. When a monitored interface changes to a specified state—to OOS or in-service—the voice port presents a seized/busyout condition to the attached PBX or other customer premises equipment (CPE). The PBX or other CPE can then attempt to select an alternate route.

Local voice busyout is different from busy-back. Busy-back refers to the signal sent from within the network to the calling party that indicates a busy (or congested) state anywhere along the route, up to and including the condition of the called party.

Note Local voice busyout is supported on analog and digital voice ports using CAS, but not on BRI Voice Modules (BVMs).

Advanced Voice Busyout

Advanced voice busyout monitors links to remote and IP-addressable interfaces and uses an SAA probe signal for VoIP. Voice classes are configured to simplify and speed up the configuration of voice busyout on multiple voice ports. SAA probe monitoring of remote interfaces is intended for use with VoIP, VoFR, and VoATM networks.

Busyout Monitor

Busyout monitor is one aspect of Call Admission Control (CAC) that uses a data network and the PSTN to provide the best possible quality and cost savings for VoIP calls. Busyout monitor CAC functionality also provides the following:

•Logical connections between LAN/WAN interfaces of routers in a VoIP gateway with directly connected voice ports.

•Port-by-port definition.

•Tracking of any directly connected main interface, subinterface, or virtual interface without monitoring the status of remote devices.

Trunk Management Prerequisite Tasks

Before configuring the trunk connections and conditioning features, the one of the following must be configured:

•VoFR using FRF.11

•VoATM

•VoIP

•Voice ports

Before configuring the congestion-monitoring features, the following requirements must be met:

•Alarm conditioning requires Cisco IOS Release 12.1(3)T or later. The following must also be configured:

–VoFR or VoATM, including plain old telephone service (POTS) and network dial peers

–Voice ports, including busyout and trunk conditioning

–DS0 groups

•PSTN fallback requires that VoIP be configured.

•Voice busyout and SAA probe enhancements required that the following configuration tasks be completed:

–VoFR or VoATM, including POTS and network dial peers

–Voice ports

–VoIP network

–Call fallback on the local router

–SAA responder on the target (far-end) router

Note Trunk Management Configuration Tasks List

This section includes procedures for configuring the following trunk management features:

•Configuring Trunk-Conditioning Signaling Attributes

•Assigning Trunk-Conditioning Attributes to Network Dial Peers

•Assigning Voice Classes to Voice Ports

•Configuring Trunk Connections

–Configuring PLAR (Switched) Connections

–Configuring Trunk/Tie-Line Connections

–Configuring PLAR-OPX Connections

•Configuring T1/E1 Alarm Generation Parameters

•Configuring PSTN Fallback

•Configuring Local Voice Busyout

Configuring Trunk-Conditioning Signaling Attributes

Different trunk-conditioning signaling attributes may be required to match the characteristics of the different PBXs to which the router connects. For this reason, trunk-conditioning attributes are configured by creating a voice class for each set of attributes required. The trunk-conditioning attributes are configured for the voice class and the voice class is assigned to one or more dial peers.

A voice class must be configured and assigned to at least one dial peer before the trunk conditioning signaling attributes take effect.

Note This configuration supports the North America CAS Protocol and applies only to Cisco private-line or FRF.11 trunk calls. It does not apply to digital T1/E1 trunks using CCS.

To create a voice class and define the trunk-conditioning attributes, use the following commands beginning in global configuration mode:
Command

Purpose
Step 1
Router(config)# voice class permanent tag
Creates a voice class. The tag number range is from 1 to 10000, and it must be unique on the router.

Note The voice-class command in dial-peer configuration mode is entered with a hyphen. The voice class command in global configuration mode is entered without the hyphen.
Step 2
Router(config-voice-class)# signal keepalive seconds
(Optional) Defines the keepalive signaling packet interval. The seconds range is from 1 to 65535; the default is 5.
Step 3
Router(config-voice-class)# {no-action | idle-only | 
oos-only | both}
(Optional) Sets the signaling pattern (when the far-end keepalive message is lost or when AIS is received from the far end). The keywords are as follows:

•no-action—Sends no signaling pattern.

•idle-only or oos-only—Sends only one signaling pattern.

•both—Restores the default (both signaling patterns are sent).

Note The no form of the command restores the default also.
Step 4
Router(config-voice-class)# signal pattern 
{idle receive | idle transmit | oos receive | 
oos transmit} bit-pattern
(Optional) Overrides the default values for the idle and receive OOS patterns or configures OOS transmit signaling patterns. The keywords and argument are as follows:

•idle receive—Defines the signaling pattern for an idle message from the network and the signaling pattern to be sent to the PBX if the network trunk is OOS and signal sequence oos idle-only or signal sequence oos are configured. The defaults are:

–For near-end E&M—0000 (for T1) or 0001 (for E1)

–For near-end FXO loop start—0101

–For near-end FXO ground start—1111

–For near-end FXS—0101

–For near-end MELCAS—1101
•idle transmit—Defines the signaling pattern for an idle message from the PBX. The defaults are:

–For near-end E&M—0000

–For near-end FXO—0101

–For near-end FXS loop start—0101

–For near-end FXS ground start—1111

–For near-end MELCAS—1101
•oos receive—Defines the OOS signaling pattern to be sent to the PBX if the network trunk is OOS and signal sequence oos oos-only or signal sequence oos are configured. The defaults are:

–For near-end E&M—1111

–For near-end FXO loop start—1111

–For near-end FXO ground start—0000

–For near-end FXS loop start—1111

–For near-end FXS ground start—0101

–For near-end MELCAS—1111

•oos transmit—Defines the signaling pattern for an OOS message from the PBX. There are no default signaling patterns defined.

•bit-pattern—Defines the ABCD bit pattern. Valid values are from 0000 to 1111.

The receive signal pattern comes from the data network side to the PBX. The transmit signal pattern comes from the PBX to the data network side. The range for all options is from 0000 to 1111.

Repeat the command entry for each signal pattern required.
Step 5
Router(config-voice-class)# signal timing oos 
timeout {seconds | disabled}
(Optional) Changes the timeout period for asserting a receive OOS pattern to the PBX when signaling packets are lost. This action changes the delay time before a busyout is sent to the PBX. The keyword and argument are as follows:

•seconds—Defines the delay duration between the loss of signaling packets and the beginning of the OOS state. The range is from 1 to 65535. The default is 30.

•disabled—Deactivates the detection of packet loss. If no signaling packets are received from the network, the router does not send an OOS pattern to the PBX and it continues sending voice packets. Use this option to disable busyout to the PBX.
Step 6
Router(config-voice-class)# signal timing oos 
restart seconds
(Optional) Configures permanent voice connections to be restarted after the trunk has been OOS for a specified time. The default is no signal timing OOS pattern parameters are configured.

Note This command has no effect if signal timing oos timeout is set to disabled.
Step 7
Router(config-voice-class)# signal timing oos 
slave-standby seconds
(Optional) Configures a slave port to return to its initial standby state after the trunk has been OOS for a specified time. The default is no signal timing OOS pattern parameters are configured.

Note This command has no effect if signal timing oos timeout is set to disabled.
Step 8
Router(config-voice-class)# signal timing oos 
{suppress-all | suppress-voice} seconds
(Optional) Configures the router or concentrator to stop sending voice packets or voice and signaling packets to the network if it detects a transmit OOS signaling pattern from the PBX for a specified time. The default is no signal timing OOS pattern parameters are configured.

Note An OOS transmit signaling pattern must be configured with the signal pattern oos transmit command (see Step 4).
Step 9
Router(config-voice-class)# signal timing idle 
suppress-voice seconds
(Optional) Configures the router or concentrator to stop sending voice packets after the trunk has been idle for a specified time. The default is no signal timing OOS pattern parameters are configured.
Assigning Trunk-Conditioning Attributes to Network Dial Peers

After the voice class has been created, it must be applied to the dial-peer configuration. The trunk-conditioning attributes can be assigned to VoIP, VoFR, or VoATM dial peers, but not to POTS dial peers.

Note This feature applies only to Cisco trunk (private-line) or FRF.11 trunk calls and does not apply to digital T1/E1 trunks using CCS.

To apply trunk-conditioning signaling attributes to a network dial peer, specify the dial peer type and then use the following command in dial-peer voice configuration mode:

Command

Purpose

Router(config-dial-peer)# voice-class permanent tag

Assigns the voice class to the dial peer. The tag argument specifies the unique number. The valid range is from 1 to 10000.

Note The voice-class command in dial-peer configuration mode is entered with a hyphen. The voice class command in global configuration mode is entered without the hyphen.

Assigning Voice Classes to Voice Ports

To assign a voice class to a voice port, specify the voice port, and then use the following command in voice-port configuration mode:
Command

Purpose
Router(config-voice-port)# voice-class permanent tag
Assigns the voice class to a voice port. The tag argument is a unique number assigned to the voice class. Valid range is from 1 to 10000.

Note The voice-class command for assigning a voice class to a voice port has a hyphen. The voice class command in global configuration mode is entered without the hyphen.
Verifying the Signaling Attributes and Trunk Conditioning

To verify the signaling attributes (timing parameters) using voice-port 1/5 on a Cisco MC3810 multiservice concentrator, enter the show voice trunk-conditioning signaling command. The following is a sample output from this command:
Router# show voice trunk-conditioning signaling 1/5
1/5 : 
TX INFO :slow-mode seq#= 25, sig pkt cnt= 42, last-ABCD=0000
hardware-state ACTIVE signal type is NorthamericanCAS
signal path is OPEN
 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
RX INFO :slow-mode, sig pkt cnt= 37
missing = 0, out of seq = 0, very late = 0 
playout depth = 0 (ms), refill count = 1
prev-seq#= 25, last-ABCD=0000
trunk_down_timer = 4212 (ms), idle timer = 0 (sec),
tx_oos_timer = 0 (sec), rx_ais_duration = 0 (ms)
forced playout signal pattern = NONE
signaling playout history
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
To verify the status of trunk supervision and configuration parameters on a Cisco MC3810 multiservice concentrator, enter the show voice trunk-conditioning supervisory command. The following is a sample output from this command.
Router# show voice trunk-conditioning supervisory 1/5
1/5 : state : TRUNK_SC_CONNECT, voice : on, signal : on, slave 
status: trunk connected
sequence oos : idle and oos
pattern :rx_idle = 0x0 rx_oos = 0xF tx_oos =  0xF
timing : idle = 0, restart = 0, standby = 0, timeout = 40
supp_all = 50, supp_voice = 0, keep_alive = 5
timer: oos_ais_timer = 0, timer = 0
To verify signaling and timing parameters for the configuration for voice-ports 0:0, 0:1, and 0:2 on a Cisco MC3810 multiservice concentrator, enter the show running-config command. The trunks do not have to be connected and active. The following is a sample output from this command.
Router# show running-config
Building configuration...
Current configuration:
.
.
.
voice class permanent 100
signal timing idle suppress-voice 2000
signal timing oos restart 1000
.
.
.
voice-port 0:0
 voice-class permanent 100
 compand-type a-law
!
voice-port 0:1
 voice-class permanent 100
 compand-type a-law
!
voice-port 0:2
 voice-class permanent 100
 compand-type a-law
.
.
.
To display the status of trunk-conditioning signaling and timing parameters for a voice port on a Cisco MC3810 multiservice concentrator, enter one of the following commands:

•show voice trunk-conditioning signaling. The following output sample is for voice port 1/5 on a Cisco MC3810 multiservice concentrator:
Router# show voice trunk-conditioning signaling 1/5
1/5 : 
TX INFO :slow-mode seq#= 25, sig pkt cnt= 42, last-ABCD=0000
hardware-state ACTIVE signal type is NorthamericanCAS
signal path is OPEN
 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
RX INFO :slow-mode, sig pkt cnt= 37
missing = 0, out of seq = 0, very late = 0 
playout depth = 0 (ms), refill count = 1
prev-seq#= 25, last-ABCD=0000
trunk_down_timer = 4212 (ms), idle timer = 0 (sec),
tx_oos_timer = 0 (sec), rx_ais_duration = 0 (ms)
forced playout signal pattern = NONE
signaling playout history
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
•show voice trunk-conditioning signaling summary. The following output sample is for voice ports on a Cisco MC3810 multiservice concentrator:
Router# show voice trunk-conditioning signaling summary
1/1 is shutdown
1/4 is shutdown
1/5 : 
TX INFO :slow-mode seq#= 25, sig pkt cnt= 40, last-ABCD=0000
hardware-state ACTIVE signal type is NorthamericanCAS signal path is OPEN
RX INFO :slow-mode, sig pkt cnt= 36, prev-seq#= 25, last-ABCD=0000
•show voice call summary. The following output sample is for voice port 1/5 on a Cisco MC3810 multiservice concentrator:
Router# show voice call summary
PORT      CODEC    VAD VTSP STATE            VPM STATE
========= ======== === ===================== ========================
1/1                                          *shutdown*
1/2       -         -  -                     FXSLS_ONHOOK 
1/3       -         -  -                     FXSLS_ONHOOK 
1/4                                          *shutdown* 
1/5       g729r8    n  S_CONNECT             S_TRUNKED 
1/6       -         -  -                     EM_ONHOOK 
Configuring Trunk Connections

This section covers the following three types of trunk connections:

•PLARs (switched) connections enable the user to make a call without dialing any digits. The router uses the digits that follow the command internally to send the call to a dial peer.

•Trunk and tie-line connections are virtual connections to PBXs and are dedicated until disabled.

•OPXs are off-premise extension connections that are used with the Cisco MC3810 concentrators only.

Configuring PLAR (Switched) Connections

To configure a PLAR connection, enter voice-port configuration mode for the required voice port.

Note The syntax of the voice-port command is hardware specific. Refer to the Cisco IOS Voice, Video, and Fax Command Reference for more information.

To configure a PLAR connection, use the following command in voice-port configuration mode:

Command

Purpose

Router(config-voice-port)# connection plar string

Specifies a PLAR connection and associates a peer directly with an interface. The string argument is a destination telephone number. Valid entries are any series of digits that specify the E.164 standard.

Configuring Trunk/Tie-Line Connections

The following restrictions apply to the trunk/tie-line configuration:

•Trunk/tie-line connections are applicable only to Cisco 2600 and 3600 series routers.

•Use the following voice port combinations:

–E&M to E&M (same type)

–FXS to FXO

–FXS to FXS (without signaling)

•Do not perform number expansion on the destination pattern telephone numbers configured for trunk connection.

•Configure both end routers to establish the trunk connection.

•Use the shutdown/no shutdown command sequence on the voice port to activate the configuration.

To configure a trunk or tie-line connection, use the following commands in dial-peer configuration mode for the required POTS dial peer:

Command

Purpose

Step 1

Router(config-dial-peer)# destination-pattern [+]string [T]

Defines the telephone number associated with the POTS dial peer. The keywords and argument are as follows:

•Plus sign (+)—(Optional) Character indicating an E.164 standard number. The plus sign (+) is not supported on the Cisco MC3810.

•string—Series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A through D, and the following special characters:

–Asterisk (*) and pound sign (#) that appear on standard touch-tone dial pad.

–Comma (,) inserts a pause between digits.

–Period (.) matches any entered digit (this character is used as a wildcard).

•T—Indicates that the control character that the destination-pattern value is a variable length dial-string.

Step 2

Router(config-dial-peer)# port {slot-number/subunit-number/port} | {slot/port:ds0-group-no}

Associates the POTS dial peer with a specific logical dial interface. The arguments are as follows:

•slot-number—Location of the voice interface card. Valid entries are from 0 to 3, depending on the slot where the card is installed.

•subunit-number—Subunit on the voice interface card where the voice port is located. Valid entries are 0 and 1.

•port—Voice-port number. Valid entries are 0 and 1.

•slot—Router location of the installed voice port adapter. Valid entries are from 0 to 3.

•port—Voice interface card location. Valid entries are from 0 to 3.

•ds0-group-no—Defined DS0 group number. Each group number is represented on a separate voice port. This enables definition of individual DS0s on the digital T1/E1 card.

Step 3

Router(config-dial-peer)# prefix string

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

Step 4

Router(config-dial-peer)# exit

Exits dial-peer configuration mode.

Step 5

Router(config)# dial-peer voice number voip

Configures a VoIP peer. The number argument uniquely identifies the VoIP dial peer.

Step 6

Router(config-dial-peer)# destination-pattern [+]string [T]

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

Step 7

Router(config-dial-peer)# session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.]host-name | loopback:rtp | loopback:compressed | loopback:uncompressed | ras}

Identifies the IP address of the appropriate port on the destination end router. The keywords and arguments are as follows:

•ipv4:destination-address—Specifies the IP address of the dial peer.

•dns:host-name—Specifies the domain name server that is the name of the IP address. Valid entries are characters representing the name of the host device.

–$s$.—Source destination pattern is part of the domain name.

–$d$.—Destination number is part of the domain name.

–$e$.—Called number digits are reversed, periods are added in-between each digit of the called number. The string is part of the domain name.

–$u$.—Unmatched portion of the destination pattern (such as a defined extension number) is part of the domain name.

•loopback:rtp—Specifies that all voice data is looped back to the originating source. Applicable for VoIP peers.

•loopback:compressed—Specifies that all voice data is looped back in compressed mode to the originating source. Applicable for POTS peers.

•loopback:uncompressed—Specifies that all voice data is looped back in an uncompressed mode to the originating source. Applicable for POTS peers.

•ras—Indicates that the RAS signaling function protocol is used. A gatekeeper will translate the E.164 address into an IP address.

Step 8

Router(config-dial-peer)# exit

Exits dial-peer configuration mode and returns to global configuration mode.

Step 9

Router(config)# voice-port {slot-number/subunit-number/port} | {slot/port:ds0-group-no}

Enters voice-port configuration mode. The arguments are as follows:

•slot-number—Defines the location of the voice interface card. Valid entries are from 0 to 3, depending on the slot where the card is installed.

•subunit-number—Specifies the subunit on the voice interface card where the voice port is located. Valid entries are 0 and 1.

•port—Specifies the voice-port number. Valid entries are 0 and 1.

•slot—Defines the router location of the installed voice port adapter. Valid entries are from 0 to 3.

•port—Indicates the voice interface card location. Valid entries are from 0 to 3.

•ds0-group-no—Defines the DS0 group number. Each group number is represented on a separate voice port. This enables definition of individual DS0s on the digital T1/E1 card.

Step 10

Router(config-voice-port)# connection {tie-line | trunk [answer-mode]} string

Specifies a tie-line connection to a PBX. The keywords and arguments are as follows:

•tie-line—Used only on the Cisco MC3810 multiservice concentrator when additional prefixed digits are required. The combined set of digits route the call into the network using the dial peers. The tie-line digits are automatically stripped by a terminating port.

•trunk—Specifies a straight tie-line connection to a PBX.

•answer-mode—(Optional) Specifies that the router should not attempt to initiate a trunk connection, but should wait for an incoming call before establishing the trunk. If one of the devices is for receiving calls only, use this option.

•string—Specifies the destination telephone number configured for the destination VoIP dial peer. The value configured for the connection trunk command must match the value configured for the VoIP dial peer exactly.

Configuring PLAR-OPX Connections

The plar-opx command is specific to the Cisco MC3810 concentrator and configures an OPX connection. The local voice port provides a local response before the remote voice port receives an answer. On FXO interfaces, the voice port does not answer until the remote side answers.

To configure a PLAR-OPX connection, use the following command in voice-port configuration mode for the required voice port:

Command

Purpose

Router(config-voice-port)# connection plar-opx string

Specifies a PLAR-OPX connection, associating a peer directly with an interface.

Configuring T1/E1 Alarm Generation Parameters

A network can be configured to monitor any combination of DS0 groups on a T1 or E1 trunk. An alarm is triggered only if all of the monitored DS0 groups on a T1 or E1 trunk are OOS. If one monitored DS0 group is in service, no alarm is triggered. The DS0 groups can be either of the following types:

•DS0 groups configured as voice ports for permanent point-to-point voice connections created using the connection command (for private lines and tie-lines). These DS0 groups can go OOS due to a trunk-conditioning event or busyout event (except forced busyout).

•DS0 groups configured as voice ports for switched voice traffic using CAS. These DS0 groups can go OOS, because of a busyout event (except forced busyout).

Note Alarm conditioning is not supported on CCS trunks.

To specify the DS0 group to be monitored and the alarm type, use the following commands beginning in global configuration mode:
Command

Purpose

Step 1

Router(config)# controller {t1 | e1} {0 | 1}

Enters controller configuration mode.

Step 2

Router(config-controller)# mode {cas | atm}

Configures the controller for CAS.

Step 3

Router(config-controller)# ds0-group ds0-group-no timeslots timeslot-list type {e&m-immediate | e&m-delay | e&m-wink | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxp-loop-start}

Configures DS0 groups on the controller. The keywords and arguments are as follows:

•ds0-group-no—Identifies the DS0 group and must be a value from 0 to 23 for T1 and 0 to 30 for E1.

•timeslots timeslot-list—Specifies a single time slot number, a single range of numbers, or multiple ranges of numbers separated by commas. For T1/E1, allowable values are from 1 to 24. Examples are:

–2

–1-15, 17-24

–1-23

–2, 4, 6-12
•type—Specifies the signaling method that depends upon the connection. The E&M interface enables connection for PBX lines and telephone equipment. The FXS interface connects basic telephone equipment and the PBX. The FXO interface connects the CO to a standard PBX interface where permitted by local regulations. It is often used for OP