Downloads |
Table Of Contents
Information About Trunk Management
Congestion Monitoring and Management Features
T1/E1 Alarm Conditioning Feature
Call Admission Control Features
Analog Centralized Automatic Message Accounting E911 Trunk Feature
How to Configure Trunk Conditioning and Connections
Prerequisites for Configuring Trunk Conditioning and Connections
Configuring Trunk Conditioning
Configuring Trunk-Conditioning Signaling Attributes
Assigning Trunk-Conditioning Attributes to Network Dial Peers
Assigning Voice Classes to Voice Ports
Verifying Signaling Attributes and Trunk Conditioning
Configuring T1/E1 Alarm Conditioning
Assigning Alarm-Generation Parameters
Verifying Alarm-Generation Parameters
Configuring PLAR (Switched) Connections
Configuring Trunk and Tie-Line Connections
Configuring PLAR-OPX Connections
Configuration Examples for Trunk Conditioning and Connections
PLAR (Switched Calls) Configuration: Example
Permanent Trunks Configuration: Example
How to Configure Trunk Monitoring and Management
Configuring Analog Centralized Automatic Message Accounting E911 Trunk
Configuring CAMA Card for CAMA Signaling
Monitoring and Maintaining Analog CAMA-E911
Configuring Voice Ports to Support DID
Verifying DID Voice-Port Configuration
Configuring Call Admission Control
Configuring Call Admission Control for H.323 VoIP Gateways
Configuring MGCP VoIP Call Admission Control
Configuring Local and Advanced Voice Busyout
Configuring the Busyout Trigger Event
Configuring a Voice Port to Busy Out
Configuring a Voice Port to Monitor the Link to a Remote Interface
Configuring a Busyout-Monitoring Voice Class
Configuring a Graceful Busyout
Configuring Busyout Monitor Gatekeeper
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 Wait-Timeout
Configuring VoIP Alternate Path Fallback SNMP Trap
Configuring Call-Fallback Map Parameters
Verifying PSTN Fallback Configuration
Monitoring and Maintaining PSTN Fallback
Configuration Examples for Trunk Monitoring and Management
Analog Centralized Automatic Message Accounting E911 Trunk: Examples
Local Voice Busyout Configuration: Examples
Alarm Trigger for Busyout of Voice Ports Configuration: Example
Call Admission Control: Examples
Call Admission Control for H.323 VoIP Gateways: Examples
MGCP VoIP Call Admission Control: Examples
Trunk-Management Features
This document describes how to condition and connect trunks and how to configure the following trunk-management features:
•
Analog Centralized Automatic Message Accounting E911 Trunk
•
•
–
–
•
–
–
•
•
•
Feature History for Analog Centralized Automatic Message Accounting E911 Trunk
12.2(11)T
This feature was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series.
Feature History for Analog DID (Direct Inward Dial)
12.1(5)XM
This feature was introduced on the Cisco 2600 series and Cisco 3600 series.
12.2(2)T
This feature was integrated into this release.
Feature History for Busyout Monitor
Feature History for CAC Features (Call Admission Control for H.323 VoIP Gateways and MGCP VoIP Call Admission Control
12.1(5)XM
This feature was introduced on the Cisco AS5300, Cisco AS5400, and Cisco AS5800.
12.2(2)T
This feature was integrated into this release.
Feature History for Local and Advanced Voice Busyout
Feature History for PSTN Fallback
Feature History for T1/E1 Alarm Conditioning
12.1(3)T
This feature was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
Feature History for Trunk Conditioning
12.0(3)XG
This feature was introduced on the Cisco MC3810.
12.0(4)T
This feature was integrated into this release.
Finding Support Information for Platforms and Cisco IOS Software Images
Use Cisco Feature Navigator to find information about platform support and Cisco IOS and CatOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn.
Note
Contents
•
•
•
•
•
Information About Trunk Management
A trunk is a communication line between two switching systems—in this case, the switching equipment in a central office (CO) and a 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.
VoIP simulates trunk connections between PBXs that are connected to Cisco routers or access servers on each side of the network.
In Figure 1, two PBXs connect to a router using a simulated trunk and a recEive and transMit (E&M) voice port. In this case, a permanent, nonswitched connection transparently connects the two PBXs.
Figure 1 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. The user hears a second dial tone from the remote PBX. You can configure two types of simulated connection—switched and permanent—for both analog and digital systems. The connection command creates these connections.
The connection trunk command creates a permanent call that is connected as soon as the routers on each end are booted (see Figure 2). Permanent calls pass limited telephony signaling and operate without collecting digits or requiring changes to the overall dial plan.
Figure 2 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 either E&M to E&M or 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 3), the user can make a call without dialing any digits. Telephony signaling, such as hookflash, is not passed. If the remote telephone does not answer and digits from an attached telephony device are not collected, the call does not roll over to voice mail.
Figure 3 Connection Private-Line Auto Ringback (PLAR) Configuration
The switched-call configuration works with any type of voice port (E&M, FXO, or FXS) and 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, rather than the router, provides dial tone to the extensions.
The connection tie-line command creates a switched call between two stations or PBXs, and this call 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
The Trunk Conditioning feature enables you to create a voice class, configure specific signaling attributes to the voice class, and then map the attributes in the voice class to either a Voice over Frame Relay, Voice over ATM, or a Voice over HDLC dial peer. Using the voice class, you can define the keepalive-signaling packet interval and the signal pattern (ABCD) bit pattern for Cisco-trunk (private-line) calls.
Trunk-conditioning signaling attributes apply to permanent point-to-point voice connections (private lines and tie lines) that you create using the connection trunk command.
Trunk conditioning enables 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 can 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) ABCD signaling. It does not apply to common-channel signaling (CCS).
When T1/E1 CAS is carried in transparent pass-through mode for arbitrary, unknown, or unsupported CAS protocols, you must define on-hook/idle patterns so that the digital signal processor (DSP) 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
Congestion Monitoring and Management Features
Congestion monitoring of permanent and switched calls is performed with the following features:
•
•
•
These features provides the following capabilities:
•
•
•
•
•
An OOS condition can be signaled 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 Feature
The T1/E1 Alarm Conditioning feature provides status monitoring on T1/E1 PBX voice interfaces for simulated lines and trunks that you create using the connection command. It supports operation with CAS but not with 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:
•
•
When all 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 that connects to the PBX or PSTN.
Note
PSTN Fallback Feature
The PSTN Fallback feature monitors congestion in the IP network and redirects calls to the PSTN or rejects calls based on network congestion. You define congestion thresholds based on your configured network. This feature can reroute calls to an alternate IP destination or, if the IP network is found unsuitable for voice traffic during periods of network congestion, to the PSTN. This enables a service provider to give a reasonable guarantee about conversation quality to its VoIP users at the time of call admission.
Note
Note
PSTN fallback includes the following capabilities:
•
–
–
•
•
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, they remain until the cache ages out or overflows. Until a value ages out, probes are sent periodically for that destination. The time interval is user configurable.
The following is an example of the PSTN fallback sequence. In the example, call fallback active is enabled and an ICPIF threshold is defined. Call control would be similar if loss and delay thresholds were defined.
1.
Note
2.
Note
3.
If the timeout threshold is set and the router has not received calls for a particular destination after the threshold expires, the router removes that destination's traffic information from the cache.
Caution
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 threshold is based on ICPIF, which is derived as part of ITU G.113 including the following: G.729, G.711, G.723, G.726, GSM, and G.728.
The ICPIF handling has been introduced for compatibility with 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.
Service Assurance Agent
SAA is a network-congestion analysis mechanism. SAA provides delay, jitter, and packet-loss information for 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 the UDP port (port 1976) for sending the SAA control message to the terminating gateway. SAA probe packets go out on randomly selected ports from the top end of the audio UDP port range (16384 to 32767).
The port pair (RTP and 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 Features
This section describes the following busyout features:
•
Local Voice Busyout Feature
The Local Voice Busyout feature busies out trunks that are assigned to PVCs so that the PBX does not seize the circuit. It enables the PBX to route a call based on the actual availability of trunks. Local voice busyout enables the following:
•
•
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 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.
This feature differs 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
Advanced Voice Busyout Feature
The Advanced Voice Busyout feature adds the following functionality to the local voice busyout feature:
•
•
This feature enables you to do the following:
•
•
SAA-probe monitoring of remote interfaces is intended for use with VoIP networks, although it can also be used with Voice over Frame Relay (VoFR) and Voice over ATM (VoATM) networks.
Busyout Monitor Feature
The Busyout Monitor feature is one aspect of call admission control that uses a data network and the PSTN to provide the best possible quality and cost savings for VoIP calls. It also provides the following:
•
•
•
Call Admission Control Features
This section describes the following CAC features:
•
•
Call Admission Control for H.323 VoIP Gateways Feature
This feature provides the ability to support resource-based CAC processes. These resources include system resources such as CPU, memory, and call volume, and interface resources such as call volume.
If system resources are not available to admit the call, the result is either a system denial (which busies out all T1 or E1) or per-call denial (which disconnects, hairpins, or plays, a message or tone). If the interface-based resource is not available to admit the call, the call is dropped from the session protocol (such as H.323).
PSTN Fallback offers CAC based on congestion thresholds in H.323 networks.
Note
User-Selected CAC Commands
The feature allows you to configure thresholds for local resources and memory and CPU resources.
Note
With the call threshold command, you can configure two thresholds, high and low, for each resource. Call treatment is triggered when the current value of a resource exceeds the configured high. The call treatment remains in effect until the current resource value falls below the configured low. Having high and low thresholds prevents call admission flapping and provides hysteresis in call admission decision making.
With the call spike command, you can configure the limit for incoming calls during a specified time period. A call spike is the term for when a large number of incoming calls arrive from the PSTN in a very short period of time (for example, 100 incoming calls in 10 ms).
With the call treatment command, you can select how the call should be treated when local resources are not available to handle the call. For example, when the current resource value for any one of the configured triggers for call threshold has exceeded the configured threshold, the call treatment choices are as follows:
•
•
•
Resource-Unavailable Signaling
The Resource Unavailable Signaling feature supports the autobusyout feature where channels are busied out when local resources are not available to handle the call. Autobusyout is supported on both CAS and PRI channels:
•
•
MGCP VoIP Call Admission Control Feature
The MGCP VoIP Call Admission Control feature enables certain Cisco CAC capabilities on VoIP networks that are managed by Media Gateway Control Protocol (MGCP) call agents. These capabilities permit the gateway to identify and gracefully refuse calls that are susceptible to poor voice quality.
Poor voice quality on an MGCP voice network can result from transmission artifacts such as echo, from the use of low quality codecs, from network congestion and delay, or from overloaded gateways. The first two causes can be overcome by using echo cancellation and better codec selection. The last two causes are addressed by MGCP VoIP CAC.
Before the release of MGCP VoIP CAC, MGCP voice calls were often established regardless of the availability of resources for those calls in the gateway and the network. MGCP VoIP CAC ensures resource availability by disallowing calls when gateway and network resources are below configured thresholds and by reserving guaranteed bandwidth throughout the network for each completed call.
MGCP VoIP CAC has three components for improving voice quality and reliability (see Table 48).
Table 48 MGCP VoIP CAC Components
System Resource Check (SRC) CAC
Evaluates memory and call resources local to the gateway
•
•
Resource Reservation Protocol (RSVP) CAC
Surveys bandwidth availability on the network
•
•
Cisco Service Assurance Agent (SA Agent)1 CAC
Appraises network congestion conditions on the network
•
1 SA Agent was called Response Time Reporter (RTR) in earlier Cisco IOS software releases.
If all three CAC types are configured on a gateway, the gateway checks resources in this order: SRC CAC, then RSVP CAC, then SA Agent CAC. If any resource check fails, the call fails and no further checks are performed. When the call fails, the gateway refuses to accept it.
MGCP VoIP CAC supports several types of calls, depending on platform (Table 49).
Fax/modem pass-through and fax/modem relay are not supported.
Note
•
SRC CAC
When a call agent attempts to set up or modify a call, MGCP SRC CAC measures available local resources on the gateway and compares them to the configured thresholds for those resources. If one or more resources are beyond their thresholds, SRC CAC notifies the call agent of the results and refuses the call. If resources are within bounds and a call is subsequently established, local resources are guaranteed for the duration of the call.
SRC CAC checks these gateway thresholds, as configured by the user:
•
•
•
If several types of thresholds are configured on the gateway, the gateway checks them in sequence to determine if sufficient resources are available to continue setting up the call.
Note
When the gateway sends an unavailable condition to the call agent, the call agent takes responsibility for the type of treatment to attach to the call attempt. The call agent may choose to handle such situations by rerouting the call, playing an announcement that the call cannot be completed, playing special tones, or sending the call back to take a different path. Once resources become available again, the gateway resumes the acceptance of new calls.
RSVP CAC
MGCP RSVP CAC determines if sufficient bandwidth exists across the IP network to accept a call and refuses the call if end-to-end bandwidth is not available.
To accept a call, MGCP RSVP CAC checks for and reserves the network bandwidth between the originating gateway and the terminating gateway before attempting to complete the call. If sufficient bandwidth is not available or cannot be reserved, the gateway alerts the call agent to this condition and the call agent applies a previously configured treatment to the refused call (plays an announcement or special tones, or sends the call back to take a different path).
RSVP is an out-of-band, end-to-end signaling protocol that requests a certain amount of bandwidth and latency with each network hop that supports RSVP. If a network node (router) does not support RSVP, RSVP moves on to the next hop. A network node has the option to approve or deny the reservation on the basis of the load of the interface to which the service is requested.
A voice call triggers two RSVP reservations because the reservation and admission control mechanisms provided by RSVP are unidirectional. Each voice gateway is responsible for initiating and maintaining one reservation toward the other voice gateway. RSVP CAC for a VoIP call fails if at least one of the reservations fails.
Cisco VoIP CAC applications use RSVP to limit the accepted voice load on the IP network and guarantee the QoS levels of calls. RSVP CAC synchronizes RSVP signaling with the call setup signaling protocol (MGCP, in this case) to ensure that the bandwidth reservation is established in both directions before a call moves to the alerting phase (ringing). This synchronization ensures that the called-party phone rings only after the resources for the call have been reserved. Using RSVP-based admission control, VoIP applications can reserve network bandwidth and react appropriately if bandwidth reservation fails.
SA Agent CAC
Cisco SAA is a Cisco IOS feature that allows users to monitor network performance and congestion between a Cisco router and a remote device, which can be another Cisco router, an IP host, or a multiple virtual storage (MVS) host. Performance can be measured for real-world scenarios through the configuration of SA Agent operations that are executed periodically. Performance metrics include round-trip response time, connect time, packet loss, application performance, and interpacket delay variance (jitter). The SA Agent feature allows users to receive notifications and perform troubleshooting and problem analysis on the basis of the statistics collected by the SA Agent.
SA Agent probes traverse the network to a given IP destination and measure the loss and delay characteristics of the network along the path traveled. These values are returned to the outgoing gateway to use in making a decision on the condition of the network and its ability to carry a voice call. SA Agent probes do not provide any bandwidth information, either configured or available. However, if bandwidth across a link anywhere in the path that the voice call will follow is oversubscribed, it is reasonable to assume that the packet delay and loss values returned by the probe will indeed reflect this condition, even if indirectly. The SA Agent protocol is a client/server protocol defined in User Datagram Protocol (UDP). The client builds and sends the probe, and the server (previously the RTR Responder) returns the probe to the sender.
SA Agent probe delay and loss information is used in calculating a single value that can be used as a gauge of network impairment and as a threshold for CAC decisions.
Analog DID Feature
The Analog Direct Inward Dialing (DID) feature is a service offered by telephone companies that enables callers to dial directly to an extension on a PBX without the assistance of an operator or automated call attendant. It makes use of DID trunks, which forward only the last three to five digits of a phone number to the PBX. If, for example, a company has a PBX with extensions 555-1000 to 555-1999, and a caller dials 555-1234, the local CO would forward 234 to the PBX. The PBX would then ring extension 234. This entire process is transparent to the caller.
When this feature is configured, a voice-enabled router equipped with an analog DID interface can receive calls from a DID trunk and connect them to the appropriate extensions. The DID state machine is identical to the E&M state machine and uses one of the following signaling types:
•
•
•
Figure 4 shows a hypothetical topology where a user connected to the PSTN (Caller A) dials various numbers and is connected to the appropriate extensions on a PBX.
Figure 4 DID Support
555-1234
234
User C
555-1345
345
User D
555-1456
456
User B
555-1678
678
No dial-peer match found; fast busy tone is played.
Note
Analog Centralized Automatic Message Accounting E911 Trunk Feature
Because E911 calls require special routing, the North American emergency E911 phone system is built largely outside of the normal PSTN on which common voice traffic rides. Calls to emergency services are routed based on the calling number, not the called number. The calling number is checked against a database of emergency service providers that cross-references the providers for the caller's particular location. The call is then routed to the proper public-service answering point (PSAP), which, in turn, dispatches those services for the caller's location.
During setup of a call to E911, before the audio channel is connected, the calling number is sent to each switching point (known as a Selective Router [SR]) via an old telephony protocol known as CAMA. CAMA was originally designed as a protocol for long-distance billing, because it provides for carrying both calling and called number using in-band signaling. CAMA allows the telephone system to send a station identification number to the PSAP via multifrequency (MF) signaling through the telephone company's E911 equipment. CAMA trunks are used in 80 percent of E911 networks.
The calling number is needed at the PSAP for two reasons:
•
•
Figure 5 illustrates the topography in existing E911 networks. Figure 6 illustrates an E911 network using the VIC-2CAMA card with Cisco 2600 series or Cisco 3600 series routers.
Figure 5 Existing E911 Networks
Figure 6 Analog CAMA E911 Networks
How to Configure Trunk Conditioning and Connections
This section contains the following procedures:
•
•
•
•
Prerequisites for Configuring Trunk Conditioning and Connections
Note
•
–
–
–
–
Configuring Trunk Conditioning
This section contains the following procedures:
•
•
•
•
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, you configure these attributes by creating a voice class for each set of attributes required. You then configure trunk-conditioning attributes for each voice class and assign the voice class to one or more dial peers. You must configure a voice class and assign it to at least one dial peer before trunk-conditioning signaling attributes take effect.
Note
To configure trunk-conditioning signaling attributes, use the following commands.
SUMMARY STEPS
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
DETAILED STEPS
Assigning Trunk-Conditioning Attributes to Network Dial Peers
After you create a voice class, you must apply it to the dial-peer configuration. You can assign trunk-conditioning attributes to VoIP, VoFR, or VoATM dial peers, but not to POTS dial peers.
Note
To apply trunk-conditioning signaling attributes to a network dial peer, use the following commands.
SUMMARY STEPS
1.
2.
3.
DETAILED STEPS
Assigning Voice Classes to Voice Ports
To assign a voice class to a voice port, use the following commands.
SUMMARY STEPS
1.
2.
3.
DETAILED STEPS
Verifying Signaling Attributes and Trunk Conditioning
Step 1
Use this command to verify signaling attributes (timing parameters).
The following is sample output for voice-port 1/5 on a Cisco MC3810.
Router# show voice trunk-conditioning signaling 1/51/5 :TX INFO :slow-mode seq#= 25, sig pkt cnt= 42, last-ABCD=0000hardware-state ACTIVE signal type is NorthamericanCASsignal path is OPEN0000 0000 0000 0000 0000 0000 0000 0000 0000 00000000 0000 0000 0000 0000 0000 0000 0000 0000 00000000 0000 0000 0000 0000 0000 0000 0000 0000 0000RX INFO :slow-mode, sig pkt cnt= 37missing = 0, out of seq = 0, very late = 0playout depth = 0 (ms), refill count = 1prev-seq#= 25, last-ABCD=0000trunk_down_timer = 4212 (ms), idle timer = 0 (sec),tx_oos_timer = 0 (sec), rx_ais_duration = 0 (ms)forced playout signal pattern = NONEsignaling playout history0000 0000 0000 0000 0000 0000 0000 0000 0000 00000000 0000 0000 0000 0000 0000 0000 0000 0000 00000000 0000 0000 0000 0000 0000 0000 0000 0000 0000The following is sample summary output for voice ports on a Cisco MC3810:
Router# show voice trunk-conditioning signaling summary1/1 is shutdown1/4 is shutdown1/5 :TX INFO :slow-mode seq#= 25, sig pkt cnt= 40, last-ABCD=0000hardware-state ACTIVE signal type is NorthamericanCAS signal path is OPENRX INFO :slow-mode, sig pkt cnt= 36, prev-seq#= 25, last-ABCD=0000Step 2
Use this command to determine the status of trunk supervision and configuration parameters.
The following is sample output for a Cisco MC3810 multiservice concentrator.
Router# show voice trunk-conditioning supervisory 1/51/5 : state : TRUNK_SC_CONNECT, voice : on, signal : on, slavestatus: trunk connectedsequence oos : idle and oospattern :rx_idle = 0x0 rx_oos = 0xF tx_oos = 0xFtiming : idle = 0, restart = 0, standby = 0, timeout = 40supp_all = 50, supp_voice = 0, keep_alive = 5timer: oos_ais_timer = 0, timer = 0Step 3
Use this command to show summary call data.
The following is sample output sample is for voice port 1/5 on a Cisco MC3810:
Router# show voice call summaryPORT CODEC VAD VTSP STATE VPM STATE========= ======== === ===================== ========================1/1 *shutdown*1/2 - - - FXSLS_ONHOOK1/3 - - - FXSLS_ONHOOK1/4 *shutdown*1/5 g729r8 n S_CONNECT S_TRUNKED1/6 - - - EM_ONHOOKStep 4
Use this command to verify signaling and timing parameters. The trunks do not have to be connected and active.
The following is sample output for voice-ports 0:0, 0:1, and 0:2 on a Cisco MC3810.
Router# show running-configurationBuilding configuration...Current configuration:...voice class permanent 100signal timing idle suppress-voice 2000signal timing oos restart 1000...voice-port 0:0voice-class permanent 100compand-type a-law!voice-port 0:1voice-class permanent 100compand-type a-law!voice-port 0:2voice-class permanent 100compand-type a-law...
Configuring T1/E1 Alarm Conditioning
This section contains the following procedures:
•
•
You can configure a network 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. DS0 groups can be either of the following types:
•
•
Note
Prerequisites
Note
•
•
•
Assigning Alarm-Generation Parameters
To assign alarm-generation parameters, use the following commands.
SUMMARY STEPS
1.
2.
3.
4.
5.
DETAILED STEPS
Verifying Alarm-Generation Parameters
Step 1
Use this command to verify that the T1/E1 controller is correctly configured for generating alarms.
The following is sample output for a Cisco MC3810 with controller E1 0 configured so that a blue alarm is generated if DS0 groups 0, 1, and 2 (voice ports 0:0, 0:1, and 0:2) are all busied out:
Router# show running-configurationBuilding configuration....controller E1 0mode casds0-group 0 timeslots 1-10 type e&m-immediate-startds0-group 1 timeslots 11-15,17-20 type e&m-immediate-startds0-group 2 timeslots 21-30 type e&m-immediate-startalarm-trigger blue 0-2Step 2
•
a.
b.
Note
•
Step 3
Use this command to display alarm status.
The following sample output displays the alarm status of the T1 or E1 trunk on a Cisco MC3810. A yellow alarm is received and detected, and a blue alarm is generated and transmitted:
Router# show controller t1 0T1 0 is up.Applique type is Channelized T1Cablelength is long gain36 0dbYellow alarm detected.alarm-trigger is set to BlueAlarm is triggeredSlot 3 CSU Serial #00000056 Model TEB HWVersion 3.70 RX level = 0DBFraming is ESF, Line Code is B8ZS, Clock Source is Line.Data in current interval (827 seconds elapsed):
Configuring Trunk Connections
This section contains the following procedures:
•
•
•
Configuring PLAR (Switched) 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.
To configure a PLAR connection, use the following command in voice-port configuration mode.
Note
To configure PLAR connections, use the following commands.
SUMMARY STEP
1.
2.
3.
DETAILED STEP
Configuring Trunk and Tie-Line Connections
Trunk and tie-line connections are virtual connections to PBXs. They are dedicated until disabled.
Restrictions
•
•
–
–
–
•
•
•
SUMMARY STEPS
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
11.
12.
DETAILED STEPS
Configuring PLAR-OPX Connections
OPXs are off-premise extension connections that are used with the Cisco MC3810 only.
To configure PLAR-OPX connections, use the following commands.
SUMMARY STEP
1.
2.
3.
DETAILED STEP
Configuration Examples for Trunk Conditioning and Connections
This section provides the following configuration examples:
•
•
Trunk-Conditioning: Example
The following example configures a voice class and then applies it to a VoFR and VoATM dial peer on a Cisco MC3810 series.
Router(config)# voice class permanent 10Router(config-class)# signal keepalive 10Router(config-class)# signal pattern idle receive 0101Router(config-class)# signal pattern idle transmit 0101Router(config-class)# signal timing idle suppress-voice 5Router(config-class)# signal pattern oos receive 0001Router(config-class)# signal pattern oos transmit 0001Router(config-class)# signal timing oos timeout 60Router(config-class)# signal timing oos restart 120Router(config-class)# signal timing oos suppress-voice 30Router(config)# dial peer voice vofr 10Router(config-dial-peer)# voice-class permanent 10Router(config)# dial peer voice voatm 20Router(config-dial-peer)# voice-class permanent 10The following example configures a voice class using default idle and OOS signaling patterns and configures busyout to the PBX after a 60-second loss of signaling packets, with restart after 120 seconds.
Router(config)# voice class permanent 10Router(config-class)# signal keepalive 10Router(config-class)# signal timing oos timeout 60Router(config-class)# signal timing idle suppress-voice 5Router(config-class)# signal timing oos restart 120Router(config-class)# exitRouter(config)# dial peer voice vofr 10Router(config-dial-peer)# voice-class permanent 10Router(config-dial-peer)# exitRouter(config)# dial peer voice voatm 20Router(config-dial-peer)# voice-class permanent 10Router(config-dial-peer)# exitThe following configuration example shows a voice class with specified signaling bit patterns for the idle receive and transmit; OOS receive and transmit states; and busyout to the PBX after a 90-second loss of signaling packets with restart after 240 seconds:
Router(config)# voice class permanent 30Router(config-class)# signal keepalive 10Router(config-class)# signal pattern idle receive 0101Router(config-class)# signal pattern idle transmit 0101Router(config-class)# signal pattern oos receive 0001Router(config-class)# signal pattern oos transmit 0001Router(config-class)# signal timing oos timeout 90Router(config-class)# signal timing idle suppress-voice 5Router(config-class)# signal timing oos restart 240Router(config-class)# exitRouter(config)# voice-port 0/1:5Router(config-voiceport)# voice-class permanent 30The following configuration example shows a voice class using default idle and OOS signaling patterns and configures busyout after 60 seconds to the PBX, with restart after 120 seconds. It applies the voice class to both VoFR and VoATM dial peers:
Router(config)# voice class permanent 10Router(config-class)# signal keepalive 10Router(config-class)# signal timing oos timeout 60Router(config-class)# signal timing idle suppress-voice 5Router(config-class)# signal timing oos restart 120Router(config-class)# exitRouter(config)# dial peer voice vofr 10Router(config-dial-peer)# voice-class permanent 10Router(config-dial-peer)# exitRouter(config)# dial peer voice voatm 20Router(config-dial-peer)# voice-class permanent 10Router(config-dial-peer)# exitThe following example configures a voice class with specified signaling bit patterns for the idle receive, idle transmit, OOS receive, and OOS transmit states, and configures busyout after 90 seconds to the PBX, with restart after 240 seconds. It applies the voice class to digital voice port 0:5 on a Cisco MC3810.
Router(config)# voice class permanent 30Router(config-class)# signal keepalive 10Router(config-class)# signal pattern idle receive 0101Router(config-class)# signal pattern idle transmit 0101Router(config-class)# signal pattern oos receive 0001Router(config-class)# signal pattern oos transmit 0001Router(config-class)# signal timing oos timeout 90Router(config-class)# signal timing idle suppress-voice 5Router(config-class)# signal timing oos restart 240 Router(config-class)# exitRouter(config)# voice-port 0:5Router(config-voiceport)# voice-class permanent 30PLAR (Switched Calls) Configuration: Example
The following example configures the DTMF relay and PLAR for router alpha.
hostname router-alpha!voice-card 1!controller T1 1/0framing esflinecode b8zsds0-group 1 timeslot 1 type fxo-loopds0-group 2 timeslot 2 type fxo-loop!dial-peer voice 1 voipdtmf-relay h245-alphacodec g729adestination-pattern 2..session target ipv4:192.168.100.2!dial-peer voice 2 potsdestination-pattern 101port 1/0:1!dial-peer voice 3 potsdestination-pattern 102port 1/0:2!voice-port 1/0:1connection plar 201!voice-port 1/0:2connection plar 202!interface s0/0ip address 192.168.100.1 255.255.255.0The following example configures the DTMF relay for router beta.
hostname router-beta!dial-peer voice 1 voipdestination-pattern 1..dtmf-relay h245-alphacodec g729asession target ipv4:192.168.100.1!dial-peer voice 2 potsdestination-pattern 201port 1/1!dial-peer voice 3 potsdestination-pattern 202port 1/2!voice-port 1/1!voice-port 1 / 2!interface serial 0/0ip address 192.168.100.2 255.255.255.0Permanent Trunks Configuration: Example
A trunk connection can be used only between E&M ports or with FXO-to-FXS connections. The following example configures the alpha router:
hostname router-alpha!voice-card 1!controller T1 1/0framing esflinecode b8zsds0-group 1 timeslot 1 type e&m-winkds0-group 2 timeslot 2 type e&m-winkclock source line!voice-port 1/0:1connection trunk 1111!voice-port 1/0:2connection trunk 1112!dial-peer voice 1 voipdtmf-relay h245-alphacodec g729adestination-pattern 111.session target ipv4:192.168.100.2!dial-peer voice 2 potsdestination-pattern 2221port 1/0:1!dial-peer voice 3 potsdestination-pattern 2222port 1/0:2!interface serial 0/0ip address 192.168.100.1 255.255.255.0The following example configures the beta router:
hostname router-beta!voice-card 1!controller T1 1/0framing esflinecode b8zsds0-group 1 timeslot 1 type e&m-winkds0-group 2 timeslot 2 type e&m-winkclock source line!voice-port 1/0:1connection trunk 2221!voice-port 1/0:2connection trunk 2222!dial-peer voice 1 voipdtmf-relay h245-alphacodec g729adestination-pattern 222.session target ipv4:192.168.100.1!dial-peer voice 2 potsdestination-pattern 1111port 1/0:1!dial-peer voice 3 potsdestination-pattern 1112port 1/0:2!interface serial 0/0ip address 192.168.100.2 255.255.255.0In this configuration, a permanent and transparent path is set up between individual DS0s on each router. It passes dial tone from the remote PBX and passes DTMF digits out of band.
The connection command, using the keyword trunk, establishes the permanent trunk connection between the routers. The digits after the command are passed internally within the router to match a dial peer so that the call can be set up.
How to Configure Trunk Monitoring and Management
This section contains the following procedures:
•
•
•
Configuring Analog Centralized Automatic Message Accounting E911 Trunk
This section contains the following procedures (each task identified as either optional or required):
•
•
•
•
•
Restrictions
•
•
–
–
–
–
Prerequisites
Note
•
•
•
•
•
Configuring CAMA Card for CAMA Signaling
To configure a CAMA card for CAMA signaling, use the following commands.
Note
SUMMARY STEPS
1.
2.
3.
4.
5.
DETAILED STEPS
Note
Configuring ANI Mapping
To configure a CAMA card for 8-digit ANI transmission, use the following commands.
SUMMARY STEPS
1.
2.
3.
4.
5.
DETAILED STEPS
Verifying CAMA Signaling
•
•
Troubleshooting Tips
To troubleshoot the analog CAMA E911 feature, perform the following steps:
•
•
To troubleshoot specific areas of the analog CAMA E911 feature, use the following commands in EXEC mode:
Monitoring and Maintaining Analog CAMA-E911
To display configuration information about a specific voice port, use the following commands in EXEC mode:
Configuring Analog DID
This section contains the following procedures (each identified as either optional or required):
•
•
Restrictions
•
•
Prerequisites
Note
•
•
•
•
•
•
Configuring Voice Ports to Support DID
To configure voice ports for DID, use the following commands. Not all commands required to configure voice ports appear here. Use the reference information in the "Analog DID Feature" section to find out more about voice-port configuration.
SUMMARY STEPS
1.
2.
3.
4.
5.
6.
7.
8.
DETAILED STEPS
Verifying DID Voice-Port Configuration
To verify voice-port configuration, use the show voice port command. You can specify a voice port or display the status of all configured voice ports. In the following example, the specified Cisco 2600 FXS port is configured for DID:
Router# show voice port 1/0/0Foreign Exchange Station with Direct Inward Dialing (FXS-DID) 1/0/0 Slot is 1, Sub-unit is 0, Port is 0Type of VoicePort is DID-INOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is not setNoise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to -38 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 8 msPlayout-delay Mode is set to defaultPlayout-delay Nominal is set to 60 msPlayout-delay Maximum is set to 200 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sRinging Time Out is set to 180 sCompanding Type is u-lawRegion Tone is set for USAnalog Info Follows:Currently processing noneMaintenance Mode Set to None (not in mtc mode)Number of signaling protocol errors are 0Impedance is set to 600r OhmWait Release Time Out is 30 sStation name None, Station number NoneVoice card specific Info Follows:Signal Type is wink-startDial Type is dtmfIn Seizure is inactiveOut Seizure is inactiveDigit Duration Timing is set to 100 msInterDigit Duration Timing is set to 100 msPulse Rate Timing is set to 10 pulses/secondInterDigit Pulse Duration Timing is set to 750 msClear Wait Duration Timing is set to 400 msWink Wait Duration Timing is set to 200 msWait Wink Duration Timing is set to 550 msWink Duration Timing is set to 200 msDelay Start Timing is set to 300 msDelay Duration Timing is set to 2000 msDial Pulse Min. Delay is set to 140 msPercent Break of Pulse is 60 percentAuto Cut-through is disabledDialout Delay for immediate start is 300 msConfiguring Call Admission Control
This section contains the following procedures:
•
•
Configuring Call Admission Control for H.323 VoIP Gateways
This section contains the following procedures (each identified as either required or optional):
•
•
•
•
•
Restrictions
The following are applicable to the CAC Control for H.323 VoIP Gateways feature in conjunction with the PSTN Fallback feature only:
•
•
•
•
Prerequisites
Note
•
•
Configuring Call Spike
To configure a call spike, use the following command.
SUMMARY STEP
call spike call-number [steps number-of-steps size ms]
DETAILED STEP
Router(config)# call spike call-number [steps number-of-steps size ms]
Configures the limit for the number of incoming calls in a short period of time.
Configuring Call Threshold
To configure a call threshold, use the following command.
SUMMARY STEP
call threshold {global trigger-name | interface interface-name interface-number int-calls} low value high value [busyout | treatment]
DETAILED STEP
Configuring Call-Threshold Poll Interval
To configure a call-threshold poll interval, use the following command.
SUMMARY STEP
call threshold poll-interval {cpu-average | memory} seconds
DETAILED STEP
Router(config)# call threshold poll-interval {cpu-average | memory} seconds
Enables a polling interval threshold for CPU or memory.
Configuring Call Treatment
To configure call treatment, use the following command.
SUMMARY STEP
call treatment {on | action action [value] | cause-code cause-code | isdn-reject value}
DETAILED STEP
Verifying Call Admission Control
Step 1
Use this command to display the configured call spike threshold and statistics for incoming calls.
Step 2
Use this command to display enabled triggers, current values for configured triggers, and number of application programming interface (API) calls that were made to global and interface resources.
Step 3
Use this command to display the call treatment configuration and the statistics for handling the calls based upon resource availability.
Step 4
Use this command to display the full running configuration.
Configuring MGCP VoIP Call Admission Control
This section contains the following procedures (each identified as either required or optional; you can perform any combination of optional procedures):
•
•
•
•
•
•
Restrictions
•
•
•
•
•
•
•
•
Note
Prerequisites
Note
•
•
•
•
•
Configuring MGCP for Call Admission Control
Only the mgcp command and the mgcp call-agent command are required to configure MGCP on a gateway. Other commands may be used to fine-tune the MGCP application. They are described in the documents listed in the "MGCP VoIP Call Admission Control Feature" section.
To configure MGCP for CAC, use the following commands.
SUMMARY STEPS
1.
2.
3.
4.
DETAILED STEPS
Configuring MGCP SRC CAC
To determine if the local gateway has sufficient resources to handle voice calls, MGCP SRC CAC checks those resources against the thresholds that you specify in this configuration task. The commands listed here are the minimum required to configure MGCP SRC CAC. Other commands to fine-tune SRC CAC are described in the SRC CAC document listed in the "MGCP VoIP Call Admission Control Feature" section.
Note
To configure MGCP SRC CAC, use the following commands.
SUMMARY STEPS
1.
2.
3.
DETAILED STEPS
Configuring MGCP RSVP CAC
MGCP RSVP CAC configuration requires the synchronization of the call setup signaling and the RSVP signaling. This synchronization guarantees that the called-party phone rings only after the resources for the call have been reserved. This synchronization also gives voice gateways the control of what action to take before the call setup moves to the alerting stage if the reservation fails or cannot be completed within a predefined period of time.
A timer can be set by using the call rsvp-sync resv-timer command to limit the number of seconds for which the terminating gateway waits for bandwidth reservation setup before proceeding with the call setup or releasing the call, depending on the QoS level configured in the dial peers. The timer defaults to 10 seconds.
Enable RSVP on the appropriate interfaces on your gateway by using the ip rsvp bandwidth interface configuration command. You must also enable fair queueing on these interfaces by using the fair-queue interface configuration command.
The commands listed here are the minimum required to configure MGCP RSVP CAC.
Note
•
To configure MGCP RSVP CAC, use the following commands.
SUMMARY STEPS
1.
2.
3.
4.
5.
6.
DETAILED STEPS
Configuring MGCP-SA-Agent CAC
The Cisco SA Agent is an application-aware synthetic operation agent that monitors network performance by measuring response time, network resource availability, application performance, jitter (interpacket delay variance), connect time, throughput, and packet loss. Performance can be measured between any Cisco device that supports this feature and any remote IP host (server), Cisco routing device, or mainframe host. Performance measurement statistics provided by this feature can be used for troubleshooting, for problem analysis, and for designing network topologies.
The SA Agent Responder that is enabled using the rtr responder command is a component embedded in the target Cisco routing device that allows the system to anticipate and respond to SA Agent request packets. The responder can listen on any user-defined port for UDP and TCP protocol messages. In a client/server terminology, the SA Agent Responder is a concurrent multiservice server.
Note
To configure MGCP-SA-Agent CAC, use the following commands. The commands listed here are the minimum required to configure MGCP-SA-Agent CAC. Other fine-tuning commands are described in the SA-Agent CAC documents listed in the "MGCP VoIP Call Admission Control Feature" section.
SUMMARY STEPS
1.
2.
3.
DETAILED STEPS
Verifying the MGCP VoIP CAC Configuration
Step 1
Use this command to display the current configuration settings.
Step 2
Use this command to display MGCP configuration information.
Bold lines in the following command output indicate that the MGCP VoIP SA Agent CAC and SRC CAC are disabled.
Router# show mgcpMGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONEMGCP call-agent: 172.18.195.147 2300 Initial protocol service is SGCP 1.5MGCP block-newcalls DISABLEDMGCP send RSIP for SGCP is ENABLEDMGCP quarantine mode discard/stepMGCP quarantine of persistent events is ENABLEDMGCP dtmf-relay for VoIP disabled for all codec typesMGCP dtmf-relay voaal2 codec allMGCP voip modem passthrough mode: NSE, codec: g711ulaw, redundancy: DISABLED,MGCP voaal2 modem passthrough mode: NSE, codec: g711ulawMGCP TSE payload: 100MGCP T.38 Named Signalling Event (NSE) response timer: 200MGCP Network (IP/AAL2) Continuity Test timer: 3000MGCP 'RTP stream loss' timer: 2MGCP request timeout 500MGCP maximum exponential request timeout 4000MGCP gateway port: 2427, MGCP maximum waiting delay 3000MGCP restart delay 0, MGCP vad DISABLEDMGCP rtrcac DISABLEDMGCP system resource check DISABLEDMGCP xpc-codec: DISABLED, MGCP persistent hookflash: DISABLEDMGCP persistent offhook: ENABLED, MGCP persistent onhook: DISABLEDMGCP piggyback msg DISABLED, MGCP endpoint offset DISABLEDMGCP simple-sdp DISABLEDMGCP undotted-notation DISABLEDMGCP codec type g711ulaw, MGCP packetization period 20MGCP JB threshold lwm 30, MGCP JB threshold hwm 150MGCP LAT threshold lmw 150, MGCP LAT threshold hwm 300MGCP PL threshold lwm 1000, MGCP PL threshold hwm 10000MGCP CL threshold lwm 1000, MGCP CL threshold hwm 10000MGCP playout mode is adaptive 60, 4, 200 in msecMGCP IP ToS low delay disabled, MGCP IP ToS high throughput disabledMGCP IP ToS high reliability disabled, MGCP IP ToS low cost disabledMGCP IP RTP precedence 5, MGCP signaling precedence: 3MGCP default package: line-packageMGCP supported packages: gm-package dtmf-package trunk-package line-package hs-package atm-package ms-package dt-package res-package mt-packageStep 3
Use this command to display the SRC CAC configuration.
Router# show call threshold configurationSome resource polling interval:CPU_AVG interval: 60Memory interval: 5IF Type Value Low High Enable----- ---- ----- ---- ---- ------N/A cpu-5sec 43 0 80 treatmentN/A cpu-avg 27 60 80 treatmentN/A io-mem 15 60 80 treatmentN/A proc-mem 24 60 80 treatmentN/A total-mem 22 60 80 treatmentN/A total-calls 0 5 12 treatment
Troubleshooting the MGCP VoIP CAC Configuration
MGCP VoIP CAC has several commands available to analyze call statistics and operation of applications on the gateway. They are classified into these groups for clarity:
•
•
Troubleshooting MGCP
To provide information about the operation of the MGCP application, use the following commands in privileged EXEC mode:
Troubleshooting MGCP SRC CAC
To help identify SRC CAC problems, use the following commands in privileged EXEC mode:
Troubleshooting MGCP RSVP CAC
To identify and trace RSVP CAC problems, use the following commands in privileged EXEC mode:
Troubleshooting MGCP SA Agent CAC
To help identify SA Agent CAC problems, use the following commands in privileged EXEC mode:
Configuring Local and Advanced Voice Busyout
A busyout trigger event can be configured at both the serial interface level and the voice-port level. If there is a conflict between the interface-level trigger event and the voice-port-level trigger event (trigger events for each are different), the voice-port-level trigger event overrides the interface-level trigger event.
If more than one interface is configured for a busyout trigger event, voice ports are not busied out until all of the interfaces are down.
Note
This section contains the following procedures:
•
•
•
•
•
•
Restrictions
•
•
•
Configuring the Busyout Trigger Event
To configure the voice-port busyout trigger event for a serial or ATM network interface, use the following commands.
Note
SUMMARY STEPS
1.
2.
3.
4.
DETAILED STEPS
Step 1
Router(config)# interface type [number]
Enters interface configuration mode.
Step 2
Router(config-if)# voice-port busyout
Busies out all voice ports associated with this serial interface.
Note
Step 3
Router(config-if)# Ctrl-Z
Exits the current mode and enters EXEC mode.
Step 4
Router# show voice busyout
Displays the voice busyout status.
Configuring a Voice Port to Busy Out
You can configure a voice port to busy out under specified conditions or you can manually force it to busy out using the following procedures:
•
•
The default is to busy out when the monitored interface is OOS.
Configuring Busyout Under Specified Conditions
To configure busyout under specified conditions, use the following commands.
SUMMARY STEP
1.
2.
3.
DETAILED STEP
Step 1
Router(config)# voice-port slot/subunit/port
Enters voice-port configuration mode for a specified voice port.
Step 2
Specifies an interface to be monitored. When multiple interfaces are configured for OOS, busy out occurs only if all of the interfaces are OOS. When multiple interfaces are configured for in-service, busy out occurs only when any one interface returns to service.
Keywords and arguments are as follows:
•
•
•
•
•
Note
Reenter the command for each additional interface to be monitored.
Step 3
Configuring Busyout-Seize Conditions
To configure busyout-seize conditions, use the following commands.
SUMMARY STEPS
1.
2.
3.
4.
DETAILED STEPS
Note
The busyout seize action depends on the voice-port signaling type. Table 50 lists the busyout actions that take place. For E&M voice ports, the busyout action is always seize.
Forcing Busyout
When you configure busyout, the specified voice port is forced into a busyout state when the interface is down. If you enter the busyout forced command, the voice port is forced unconditionally into a busyout state. If you enter the voice-port busyout command on more than one interface, all interfaces must be down for the busyout to take effect. To configure a forced busyout condition, use the following commands.
Note
SUMMARY STEPS
1.
2.
3.
4.
5.
DETAILED STEPS
Configuring a Voice Port to Monitor the Link to a Remote Interface
Restrictions
•
•
•
•
•
•
•
You can configure individual voice ports for busyout, or you can apply a voice class that includes all of the busyout parameters (see the "Assigning Voice Classes to Voice Ports" section).
Note
SUMMARY STEP
1.
2.
3.
DETAILED STEP
Verifying the Voice-Port Busyout Configuration
Step 1
Step 2
Use this command to display information about the busyout state.
The following is sample output for voice ports on a Cisco MC3810:
Router# show voice busyoutVoice port busyout will be triggered by thefollowing network interfaces states 1/1 probe 192.168.202.128 codec g711u icpif 25 1/2 probe 192.168.202.128 codec g711u icpif 25 1/3 probe 192.168.202.128 codec g711u icpif 25The following voice ports are in busyout state1/1 is in busyout state caused by probe 192.168.202.128 codec g711u icpif 21/2 is in busyout state caused by probe 192.168.202.128 codec g711u icpif 21/3 is in busyout state caused by probe 192.168.202.128 codec g711u icpif 2
Configuring a Busyout-Monitoring Voice Class
A busyout voice class monitors local ports (serial and Ethernet) and links to remote IP addresses. Busyout occurs when all of the monitored local ports are OOS or when all of the monitored links go below the configured threshold value. If a voice port is configured to monitor multiple links, busyout occurs only when all of the monitored links go below the threshold.
To configure a busyout-monitoring voice class, use the following commands.
SUMMARY STEPS
1.
2.
3.
4.
5.
6.
7.
8.
9.
DETAILED STEPS
Verifying the Voice and Voice-Class Busyout Configuration
Step 1
–
–
The voice port monitors a remote IP address for OOS only.
Note
Step 2
Use this command to display information about the busyout state.
The following is sample output for voice ports on a Cisco MC3810:
Router# show voice busyoutVoice port busyout will be triggered by the following network interfaces states 1/2 busyout monitor ATM0 1/3 busyout monitor ATM0 1/4 busyout monitor Serial0 1/5 busyout monitor Serial0 1/6 probe 192.168.202.128 codec g711u icpif 25The following voice ports are in busyout state1/1 is forced into busyout state 1/2 is in busyout state caused by ATM0 1/3 is in busyout state caused by ATM0 1/4 is in busyout state caused by Serial0 1/5 is in busyout state caused by Serial01/6 is in busyout state caused by probe 192.168.202.128 codec g711u icpif 2
Configuring a Graceful Busyout
To configure a graceful busyout, use the following commands.
SUMMARY STEPS
1.
2.
3.
DETAILED STEPS
Configuring Busyout Monitor
Prerequisites
Note
•
•
•
•
SUMMARY STEPS
1.
2.
3.
4.
DETAILED STEPS
Configuring Busyout Monitor Gatekeeper
To configure a voice port to busy out a voice port if the gateway loses connection to the primary gatekeeper, use the following commands.
SUMMARY STEPS
1.
2.
3.
DETAILED STEPS
Verifying Busyout Status
To verify that busyout is configured correctly, use the show running-configuration command to display the command settings for the router, as shown in the "Configuring Local and Advanced Voice Busyout" section.
Configuring PSTN Fallback
This section contains the following procedures (each identified as either optional or required):
•
•
•
•
•
•
•
•
•
•
•
Restrictions
•
•
•
•
Prerequisites
Note
•
Configuring Fallback to Alternate Dial Peers
To configure fallback to alternate dial peers, use the following commands.
SUMMARY STEPS
1.
2.
DETAILED STEPS
Configuring Destination Monitoring without Fallback to Alternate Dial Peers
To configure destination monitoring without fallback to alternate dial peers, use the following commands.
SUMMARY STEP
call fallback monitor
DETAILED STEP
Router(config)# call fallback monitor
Enables the monitoring of destinations without fallback to alternate dial peers.
Configuring Call-Fallback Cache Parameters
To configure the call-fallback cache parameters, use the following commands.
SUMMARY STEPS
1.
2.
3.
DETAILED STEPS
Configuring Call-Fallback Jitter-Probe Parameters
To configure call-fallback jitter-probe parameters, use the following commands.
SUMMARY STEPS
1.
2.
or
call fallback jitter-probe dscp dscp-number3.
DETAILED STEPS
Configuring Call-Fallback Probe-Timeout and Weight Parameters
To configure call-fallback jitter-probe parameters, use the following commands.
SUMMARY STEPS
1.
2.
DETAILED STEPS
Configuring Call-Fallback Threshold Parameters
To configure call-fallback threshold parameters, use the following commands.
SUMMARY STEPS
call fallback threshold delay delay-value loss loss-value
or
call fallback threshold icpif threshold-value
DETAILED STEPS
Configuring Call-Fallback Wait-Timeout
To configure the call-fallback wait-timeout parameters, use the following commands:
Summary Steps
call fallback wait-timeout milliseconds
DETAILED STEPS
Configuring VoIP Alternate Path Fallback SNMP Trap
The VoIP Alternate Path Fallback SNMP Trap feature adds a Simple Network Management Protocol (SNMP) trap generation capability. This feature is built on top of the fallback subsystem to provide an SNMP notification trap when the fallback subsystem redirects or rejects a call because a network condition has failed to meet the configured threshold. The SNMP trap provides VoIP management status MIB information without flooding management systems with unnecessary messages about call status by triggering only when a call has been redirected to the public switched telephone network (PSTN) or the alternative IP port. A call can be rejected because of a network problem such as loss of WAN connection, delay, packet loss, or jitter. This feature supports only VoIP signaling protocol with H.323 in this release.
This feature has to be configured on the originating gateway and the terminating gateway. To configure the SNMP trap parameters, use the following commands:
SUMMARY STEPS
1.
2.
DETAILED STEPS
What to Do Next
Configure the rtr responder command on the terminating voice gateway. If the rtr responder is enabled on the terminating gateway, the terminating gateway responds to the probe request when the originating gateway sends an Response Time Report (RTR) probe to the terminating gateway to check the network conditions. The details on how to configure RTR can be found in the Network Monitoring Using Cisco Service Assurance Agent section of the Cisco IOS Configuration Fundamentals Configuration Guide, Release 12.2.
Configuring Call-Fallback Map Parameters
To configure call-fallback map parameters, use the following commands.
SUMMARY STEPS
call fallback map map target ip-address address-list ip-address1 ip-address2 ... ip-address7
or
call fallback map map target ip-address subnet ip-network netmask
DETAILED STEPS
Verifying PSTN Fallback Configuration
To verify PSTN Fallback configuration, use the following commands.
Step 1
Use this command to display the contents of the currently running configuration file to see if the new feature is configured.
Step 2
Use this command to display the call history table for voice calls and verify call fallback, call delay, and call loss parameters.
Step 3
Use this command to display the current Calculated Planning Impairment Factor (ICPIF) estimates for all IP addresses in the call fallback cache.
Step 4
Use this command to display the current configuration.
Step 5
Use this command to display the call fallback statistics.
Monitoring and Maintaining PSTN Fallback
SUMMARY STEPS
1.
2.
3.
4.
5.
6.
DETAILED STEPS
Configuration Examples for Trunk Monitoring and Management
This section provides the following configuration examples:
•
–
–
•
–
–
Analog Centralized Automatic Message Accounting E911 Trunk: Examples
VIC-2CAMA for CAMA Signaling
The following example shows that the VIC-2CAMA is configured for 8-digit transmission:
!voice-port 1/0/0timing digit 75timing inter-digit 65ani mapping 0 408ani mapping 1 510ani mapping 2 610ani mapping 3 710signal cama KP-NPD-NXX-XXXX-ST!voice-port 1/0/1timing digit 75timing inter-digit 65ani mapping 0 408ani mapping 1 510ani mapping 2 610ani mapping 3 710signal cama KP-NPD-NXX-XXXX-ST!ANI Mapping
The following example shows port 0 and port 1 with the Numbering Plan Area (NPA), or area code, preprogrammed into a single MF digit. The Numbering Plan Digit (NPD) table is preprogrammed in the sending and receiving equipment on each end of the MF trunk. Configuration of only one port is necessary, because both ports are configured simultaneously.
The following example shows configuration for the following NPAs (area codes): 0=408, 1=510, 2=610, 3=710.
!voice-port 1/0/0timing digit 75timing inter-digit 65ani mapping 0 408ani mapping 1 510ani mapping 2 610ani mapping 3 710signal cama KP-NPD-NXX-XXXX-ST!voice-port 1/0/1timing digit 75timing inter-digit 65ani mapping 0 408ani mapping 1 510ani mapping 2 610ani mapping 3 710signal cama KP-NPD-NXX-XXXX-ST!Busyout: Examples
This section provides the following configuration examples:
•
•
Local Voice Busyout Configuration: Examples
The following example configures digital voice port 0:0.4 on a Cisco MC3810 series to go into the busyout state if serial interface 0:0 goes out of service:
Router(config)# voice-port 0:0.4Type of VoicePort is FXSrouter(config-voiceport)# busyout monitor interface serial 0:01/2 is in busyout stateRouter(config-voiceport)# endRouter# show voice busyout!If following network interfaces are down, voice port will be put into busyout stateThe following voice ports are in busyout state1/1 is forced into busyout state1/2 is in busyout state caused by Serial0The following example configures digital voice port 2/1:7 on a Cisco 3600 series to go into the busyout state if serial interface 0:0 goes out of service:
Router(config)# voice-port 2/1:7Type of VoicePort is FXSRouter(config-voiceport)# busyout monitor interface serial 0:01/2 is in busyout stateRouter(config-voiceport)# endRouter# show voice busyout!If following network interfaces are down, voice port will be put into busyout stateThe following voice ports are in busyout state2/1:7 is forced into busyout state2/1:8 is in busyout state caused by Serial0The following example configures the busyout seize action for analog voice port 0/2/1 on a Cisco 3600 series to repeat:
Router(config)# voice-port 0/2/1Type of VoicePort is FXORouter(config-voiceport)# busyout seize repeatRouter(config-voiceport)# endRouter# show voice busyout!If following network interfaces are down, voice port will be put into busyout stateThe following voice ports are in busyout state0/2/1 is forced into busyout state0/2/2 is in busyout state caused by Serial0The following example forces DS0 timeslots 1 through 12 on controller T1 0 on a Cisco MC3810 into the busyout state:
Router(config)# controller t1 0Router(config-controller)# ds0 busyout 1-12Router(config-controller)# endThe following example configures busyout voice class 35, which initiates voice-port busyout whenever either serial port 0 or 1 is in service, and it applies voice class 35 to voice port 1/3:
Router(config)# voice class busyout 35Router(config-class)# busyout monitor serial 0 in-serviceRouter(config-class)# busyout monitor serial 1 in-serviceRouter(config-class)# exitRouter(config)# voice-port 1/3Router(config-voiceport)# voice class 35The following example configures busyout voice class 40, which initiates voice-port busyout whenever an SAA probe sent to both of the two specified remote interfaces results in a link with an ICPIF delay/loss average of more than 15, and it applies voice class 40 to voice port 1/4:
Router(config)# voice class busyout 40Router(config-class)# busyout monitor probe 192.168.202.128 icpif 15Router(config-class)# busyout monitor probe 192.168.202.129 icpif 15Router(config-class)# exitRouter(config)# voice-port 1/4Router(config-voiceport)# voice class 40The following example configures analog voice port 1/1 on a Cisco MC3810 to use an SAA probe with a G.711 alaw profile to probe the link to the remote interface with IP address 192.168.202.128, and to busyout the voice port if the link has a packet loss of more than 50 percent and a packet delay of more than 25 ms:
Router(config)# voice-port 1/1Router(config-voiceport)# busyout monitor probe 192.168.202.128 codec g711a loss 50 delay 25The following example configures voice port 1/0/1 on a Cisco 3600 series to use an SAA probe with the default (G.711 ulaw) profile to probe the link to the remote interface with IP address 192.168.202.128, and to busyout the voice port if the link has packet loss and delay that exceed the threshold values configured by the call fallback active command:
Router(config)# voice-port 1/0/1Router(config-voiceport)# busyout monitor probe 192.168.202.128The following example configures busyout voice class 60, which configures multiple parameters for voice-port busyout, and it applies voice class 60 to voice ports 1/0/0 and 1/0/1 on a Cisco 3600 series. The voice ports busy out under any one the following conditions:
•
•
•
Router(config)# voice class busyout 60Router(config-class)# busyout monitor serial 0/0Router(config-class)# busyout monitor serial 0/1Router(config-class)# busyout monitor serial 1/0 in-serviceRouter(config-class)# busyout monitor serial 1/1 in-serviceRouter(config-class)# busyout monitor probe 192.168.202.128 loss 50 delay 1000Router(config-class)# busyout monitor probe 192.168.202.129 loss 50 delay 1000Router(config-class)# exitRouter(config)# voice-port 1/0/0Router(config-voiceport)# voice class 60Router(config-voiceport)# exitRouter(config)# voice-port 1/0/1Router(config-voiceport)# voice class 60Router(config-voiceport)# exitThe following example configures voice port 1/1 into forced busyout state:
Router(config)# voice-port 1/1Type of VoicePort is FXSRouter(config-voiceport)# busyout forced 00:09:46: port 0 is forced into busyout stateRouter(config-voiceport)# endRouter# show voice busyout!If following network interfaces are down, voice port will be put into busyout state.The following voice ports are in busyout state1/1 is forced into busyout stateThe following example configures voice port 1/2 to busyout monitor mode, monitoring serial 0:
Router(config)# voice-port 1/2Type of VoicePort is FXSRouter(config-voiceport)# busyout-monitor serial 01/2 is in busyout stateRouter(config-voiceport)# endRouter# show voice busyout!If following network interfaces are down, voice port will be put into busyout state.The following voice ports are in busyout state1/1 is forced into busyout state1/2 is in busyout state caused by Serial0The following example configures voice port 1/3 to the busyout seize repeat state:
Router(config)# voice-port 1/3Type of VoicePort is FXOrouter(config-voiceport)# busyout-seize repeatRouter(config-voiceport)# endRouter# show voice busyout!If following network interfaces are down, voice port will be put into busyout state.The following voice ports are in busyout state1/1 is forced into busyout state1/2 is in busyout state caused by Serial0The following is a sample configuration of the busyout monitor action graceful and busyout monitor gatekeeper commands:
Router# show running-configurationBuilding configuration...Current configuration :2143 bytes!version 12.2service timestamps debug uptimeservice timestamps log uptimeno service password-encryption!hostname 2600a!enable secret 5 $1$QHAX$W3J2KNkDTkB99UmLZ7rq9.enable password xxx!username user password 0 passwdip subnet-zerono ip routing!!no ip domain-lookup!!fax interface-type fax-mailmta receive maximum-recipients 0!controller T1 0/2framing sflinecode ami!!interface Ethernet0/0ip address 10.4.170.95 255.255.255.0no ip route-cacheno ip mroute-cachehalf-duplexh323-gateway voip interfaceh323-gateway voip id test ipaddr 10.4.170.77 1719h323-gateway voip h323-id morpheus!interface Serial0/0no ip addressno ip route-cacheno ip mroute-cacheshutdownno fair-queue!interface Serial0/1no ip addressno ip route-cacheno ip mroute-cacheshutdown!ip local pool setup_pool 1.2.71.1 1.2.71.255ip default-gateway 1.2.0.1ip classlessno ip http serverip pim bidir-enable!!dialer-list 1 protocol ip permitdialer-list 1 protocol ipx permit!!snmp-server community public RO!voice-port 1/0/0type 5!voice-port 1/0/1type 5!voice-port 1/1/0!voice-port 1/1/1busyout monitor action gracefulbusyout monitor gatekeeperbusyout monitor Ethernet0/0!mgcpmgcp default-package dtmf-package!mgcp profile default!dial-peer cor custom!!dial-peer voice 1 voipincoming called-number 308destination-pattern ...session protocol sipv2session target ipv4:10.4.170.77codec g711ulaw!dial-peer voice 2 potsdestination-pattern 308port 1/1/0prefix 308!dial-peer voice 3 potsdestination-pattern 309port 1/1/1!dial-peer voice 4 potsapplication mgcpapp!dial-peer voice 7 potsapplication sdfjsadf!dial-peer voice 88 pots!dial-peer voice 33 voipfax rate 12000!dial-peer voice 34 voatm!dial-peer voice 35 vofr!dial-peer voice 37 pots!dial-peer voice 90 voatm!dial-peer voice 91 voipfax rate 4800 bytes 41!dial-peer voice 92 vofr!dial-peer voice 999 voipdestination-pattern 1234session target ras!gateway!!line con 0exec-timeout 0 0line aux 0line vty 0 4password labloginline vty 5 15login!!endAlarm Trigger for Busyout of Voice Ports Configuration: Example
This example creates three permanent trunks on controller T1 0 and configures T1 0 to send a blue (AIS) alarm if all three permanent trunks are OOS. These steps create the voice ports and configure the alarm trigger:
Router(config)# controller t1 0Router(config-controller)# mode casRouter(config-controller)# ds0-group 0 timeslots 1-10 type fxs-ground-startRouter(config-controller)# ds0-group 1 timeslots 11 type fxs-ground-startRouter(config-controller)# ds0-group 2 timeslots 12-23 type fxs-ground-startRouter(config-controller)# alarm-trigger blue 0-2Router(config-controller)# exitRouter(config)#These steps create a voice class to define the trunk conditioning parameters for permanent trunks (in which the default values are not used):
Router(config)# voice class permanent 8Router(config-class)# signal keepalive 10Router(config-class)# signal timing oos timeout 60Router(config-class)# signal timing idle suppress-voice 5Router(config-class)# signal timing oos restart 120Router(config-class)# exitRouter(config)#These steps create a VoIP dial peer to define the network connectivity and trunk conditioning parameters for permanent trunks:
Router(config)# dial-peer voice 100 voipRouter(config-dial-peer)# session target ipv4:172.20.10.10Router(config-dial-peer)# destination-pattern 10..Router(config-dial-peer)# voice-class permanent 8Router(config-dial-peer)# exitRouter(config)#These steps assign each voice port to a permanent trunk and associate each trunk with a network dial peer:
Router(config)# voice-port 0:0Router(config-voiceport)# connection trunk 1001Router(config-voiceport)# exitRouter(config)# voice-port 0:1Router(config-voiceport)# connection trunk 1002Router(config-voiceport)# exitRouter(config)# voice-port 0:2Router(config-voiceport)# connection trunk 1003Router(config-voiceport)# exitRouter(config)#This example configures voice port 0:0 for busyout if serial port 0.1, 0.2, and Ethernet port 0 all go out of service, or serial port 1 comes into service:
Router(config)# voice-port 0:0Router(config-voiceport)# busyout monitor serial 0.1Router(config-voiceport)# busyout monitor serial 0.2Router(config-voiceport)# busyout monitor ethernet 0Router(config-voiceport)# busyout monitor serial 1 in-serviceRouter(config-voiceport)# exitThis example configures voice port 0:1 for busyout if the connections to both of two remote IP addresses are OOS:
Router(config)# voice-port 0:1Router(config-voiceport)# busyout monitor probe 192.168.202.128 codec g711a icpif 15Router(config-voiceport)# busyout monitor probe 192.168.202.129 codec g711a icpif 15Router(config-voiceport)# exitThis example configures voice port 0:2 for busyout under any one of the following conditions:
•
•
•
Router(config)# voice-port 0:2Router(config-voiceport)# busyout monitor serial 0.1Router(config-voiceport)# busyout monitor serial 0.2Router(config-voiceport)# busyout monitor serial 1 in-serviceRouter(config-voiceport)# busyout monitor probe 192.168.202.128 codec g711a icpif 15Router(config-voiceport)# busyout monitor probe 192.168.202.129 codec g711a icpif 15Router(config-voiceport)# exitRouter(config)# exitCall Admission Control: Examples
This section provides the following configuration examples:
•
•
Call Admission Control for H.323 VoIP Gateways: Examples
Call Spike Configuration
The following configuration of the call spike command has a call number of 30, 10 steps, and a step size of 2000 ms:
call threshold global cpu-avg low 70 high 80call spike 30 steps 10 size 2000cns event-service serverCall Threshold Configuration
The following example busies out the total-calls resource of 5 (low) or 5000 (high):
call threshold global total-calls low 5 high 5000 busyoutThe following example enables thresholds of 5 (low) and 2500 (high) on Ethernet interface 0:
call threshold interface Ethernet 0 int-calls low 5 high 2500The following example busies out the average CPU utilization if 5 percent (low) or 65 percent (high) is reached:
call threshold global cpu-avg low 5 high 65 busyoutCall Threshold Poll Interval Configuration
The following example enables a polling interval threshold for memory of 10 seconds:
call threshold poll-interval memory 10The following example enables a polling interval threshold of 50 seconds:
call threshold poll-interval cpu-average 50Call Treatment Configuration
The following example enables the Call Treatment feature with a "hairpin" action:
call treatment oncall treatment action hairpinThe following example displays proper formatting of the action playmsg keywords:
call treatment oncall treatment action playmsg tftp://keyer/prompts/congestion.au
Note
The following example configures a call treatment cause code to display no-qos when local resources are unavailable to process a call:
call treatment oncall treatment cause-code no-qosMGCP VoIP Call Admission Control: Examples
MGCP RSVP and SA Agent CAC
The following example shows a configuration of MGCP RSVP and SA Agent CAC on a Cisco 3660:
version 12.2no service single-slot-reload-enableservice timestamps debug uptimeservice timestamps log uptimeno service password-encryption!hostname host1!no logging bufferedno logging bufferedlogging rate-limit console 10 except errors!!ip subnet-zero!!no ip fingerno ip domain-lookupip host lab 192.168.254.254!call fallback activecall rsvp-sync!!interface FastEthernet0/0ip address 172.16.125.4 255.255.0.0duplex autospeed autoip rsvp bandwidth 512 512!interface FastEthernet0/1no ip addressshutdownduplex autospeed auto!ip kerberos source-interface anyip classlessip route 172.16.173.1 255.255.255.255 172.16.0.1ip route 192.168.254.254 255.255.255.255 FastEthernet0/0no ip http server!!voice-port 1/1/0!voice-port 1/1/1!mgcpmgcp call-agent 172.16.173.1 service-type mgcp version 1.0mgcp modem passthrough voip mode nsemgcp modem passthrough voaal2 modemgcp rtrcacno mgcp timer receive-rtcp!mgcp profile default!dial-peer cor custom!!dial-peer voice 1 potsapplication mgcpappport 1/1/0!dial-peer voice 2 potsapplication mgcpappport 1/1/1rtr responder!line con 0transport input noneline aux 0line vty 0 4login!endMGCP VoIP CAC on a Trunking Gateway
This configuration enables all three types of MGCP VoIP CAC: SRC, RSVP, and SA Agent. Comment lines are provided above the CAC commands to help you identify the commands needed for a particular CAC type.
version 12.2service timestamps debug uptimeservice timestamps log uptimeno service password-encryption!hostname host1!!voice-card 2!voice-card 3!ip subnet-zeroip dhcp smart-relay!! The following command is used in MGCP SA Agent CAC.call fallback active! The following command is used in MGCP RSVP CAC.call rsvp-sync! The following six commands are used in MGCP SRC CAC.call threshold global cpu-5sec low 55 high 70 treatmentcall threshold global cpu-avg low 70 high 80 treatmentcall threshold global total-mem low 70 high 80 treatmentcall threshold global io-mem low 70 high 80 treatmentcall threshold global proc-mem low 70 high 80 treatmentcall threshold global total-calls low 10 high 12 treatment!!controller T1 2/0!controller T1 2/1!controller T1 3/0framing esfclock source internalds0-group 1 timeslots 1-5 type none service mgcpds0-group 2 timeslots 6-24 type none service mgcp!controller T1 3/1framing esfds0-group 1 timeslots 1-10 type none service mgcpds0-group 2 timeslots 11-24 type none service mgcp!!interface FastEthernet0/0ip address 192.168.1.61 255.255.255.0duplex autospeed auto! The following command is used in MGCP RSVP CAC to configure the bandwidth allocated! for VoIP calls through the interface.ip rsvp bandwidth 512 512!interface FastEthernet0/1ip address 172.20.1.1 255.255.0.0duplex autospeed auto!ip kerberos source-interface anyip classlessip route 10.0.0.0 10.0.0.0 192.168.1.10no ip http server!snmp-server engineID local 0000000902000002B95D89F0no snmp-server ifindex persistsnmp-server manager!voice-port 3/0:1!voice-port 3/0:2!voice-port 3/1:1!voice-port 3/1:2!mgcpmgcp call-agent 10.13.57.88 service-type mgcp version 1.0mgcp modem passthrough voip mode nsemgcp modem passthrough voaal2 modemgcp package-capability trunk-package! The following command is used for MGCP SA Agent CAC.mgcp rtrcac! The following command is used in MGCP SRC CAC.mgcp src-cacno mgcp timer receive-rtcp!mgcp profile default!dial-peer cor custom!dial-peer voice 1 pots
