Table Of Contents
Cisco IOS Voice, Video, and Fax Commands:
A Through C
aaa nas port voip
acc-qos
alarm-trigger
alias static
alt-dial
answer-address
application
arq reject-unknown-prefix
as
asp
atm scramble-enable
atm video aesa
audio-prompt load
auto-cut-through
backhaul-session-manager
bandwidth
bandwidth remote
battery-reversal
block-caller
busyout forced
busyout monitor
busyout monitor probe
busyout seize
cac master
cadence-list
cadence-max-off-time
cadence-min-on-time
cadence-variation
call application cache reload time
call application voice
call application voice access-method
call application voice accounting enable
call application voice accounting-list
call application voice authen-list
call application voice authen-method
call application voice authentication enable
call application voice global-password
call application voice language
call application voice load
call application voice pin-len
call application voice redirect-number
call application voice retry-count
call application voice set-location
call application voice uid-length
call application voice warning-time
call fallback active
call fallback cache-size
call fallback cache-timeout
call fallback instantaneous-value-weight
call fallback jitter-probe num-packets
call fallback jitter-probe precedence
call fallback jitter-probe priority-queue
call fallback key-chain
call fallback map target address-list
call fallback map target subnet
call fallback monitor
call fallback probe-timeout
call fallback threshold delay loss
call fallback threshold icpif
call rsvp-sync
call rsvp-sync resv-timer
call start
call-waiting
called-number (dial-peer)
caller-id
caller-id alerting dsp-pre-alloc
caller-id alerting line-reversal
caller-id alerting pre-ring
caller-id alerting ring
caller-id attenuation
caller-id block
caller-id enable
calling-number outbound
cap-list vfc
card type
ccm-manager application redundant-link port
ccm-manager mgcp
ccm-manager redundant-host
ccm-manager switchback
ccs connect (interface)
ccs connect (controller)
ccs encap frf11
ces cell-loss-integration-period
ces-clock
ces clockmode synchronous
ces connect
ces initial-delay
ces max-buf-size
ces service
clear backhaul-session-manager group
clear call fallback cache
clear call fallback stats
clear controllers call-counters
clear csm-statistics modem
clear csm-statistics voice
clear h323 gatekeeper call
clear ip sctp statistics
clear mgcp statistics
clear rlm group
clear rudpv0 statistics
clear rudpv1 statistics
clear sgcp statistics
clear ss7 sm stats
clear voice port
clock-select
codec (dial-peer)
codec (dsp)
codec (voice-port)
codec aal2-profile
codec complexity
codec preference
comfort-noise
compand-type
condition
connect (atm)
connect (drop-and-insert)
connect (global)
connect voice
connection
connection-timeout
copy flash vfc
copy tftp vfc
cptone
cross-connect
customer-id
Cisco IOS Voice, Video, and Fax Commands:
A Through C
This chapter presents the commands to configure and maintain Cisco IOS voice, video, and fax applications. The commands are presented in alphabetical order. Some commands required for configuring voice, video, and fax may be found in other Cisco IOS command references. Use the command reference master index or search online to find these commands.
For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice, Video, and Fax Configuration Guide.
aaa nas port voip
To send out the standard NAS-Port attribute (RADIUS IETF Attribute 5) on voice interfaces, use the aaa nas port voip command in global configuration mode. To disable the command, use the no form of the command.
aaa nas port voip
no aaa nas port voip
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(1)T
|
This command was introduced on the Cisco AS5300.
|
Usage Guidelines
This command brings back the original behavior of the AAA NAS-Port on Voice over IP (VoIP) interfaces. By default this feature should not be enabled.
Note
Some customers using the Cisco AS5300 voice gateway have had the Debit Card application stop working after upgrading from 12.1(5)T to 12.1(5.3)T.
Examples
The following example shows how to return to the original behavior of the AAA NAS-Port:
Related Commands
Command
|
Description
|
aaa nas port extended
|
Replaces the NAS-port attribute with RADIUS IETF attribute 26 and displays extended field information.
|
acc-qos
To define the acceptable quality of service (QoS) for any inbound and outbound call on a Voice over IP (VoIP) dial peer, use the acc-qos command in dial peer configuration mode. To restore the default QoS setting, use the no form of this command.
acc-qos {best-effort | controlled-load | guaranteed-delay}
no acc-qos
Syntax Description
best-effort
|
Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation. This is the default.
|
controlled-load
|
Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to ensure that preferential service is received even when the bandwidth is overloaded.
|
guaranteed-delay
|
Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded.
|
Defaults
best-effort
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series routers.
|
12.1(5)T
|
The description of the command was modified.
|
Usage Guidelines
This command is applicable only to VoIP dial peers.
When VoIP dial peers are used, the Cisco IOS software uses RSVP to reserve a certain amount of bandwidth so that the selected QoS can be provided by the network. Call setup is aborted if the RSVP resource reservation does not satisfy the acceptable QoS for both peers.
To select the most appropriate value for this command, you need to be familiar with the amount of traffic this connection supports and what kind of impact you are willing to have on it. The Cisco IOS software generates a trap message when the bandwidth required to provide the selected quality of service is not available.
Examples
The following example selects guaranteed-delay as the acceptable QoS for inbound and outbound calls on VoIP dial peer 10:
Related Commands
Command
|
Description
|
req-qos
|
Requests a particular QoS using RSVP to be used in reaching a specified dial peer in VoIP.
|
alarm-trigger
To configure a T1 or E1 controller to send an alarm to the public switched telephone network (PSTN) or switch if specified T1 or E1 DS0 groups are out of service, use the alarm-trigger command in controller configuration mode. To configure a T1 or E1 controller not to send an alarm, use the no form of this command.
alarm-trigger blue ds0-group-list
no alarm-trigger
Syntax Description
blue
|
Specifies the alarm type to be sent is "blue," also known as an Alarm Indication Signal (AIS).
|
ds0-group-list
|
Specifies the DS0 group or groups to be monitored for permanent trunk connection status or busyout status.
|
Defaults
No alarm is sent.
Command Modes
Controller configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600, 3600, and MC3810 multiservice concentrator.
|
Usage Guidelines
Any monitored time slot can be used for either permanent trunk connections or switched connections. Permanent virtual circuits (PVCs) and switched virtual circuits (SVCs) can be combined on a T1 or E1 controller and monitored for alarm conditioning.
An alarm is sent only if all of the time slots configured for alarm conditioning on a T1 or E1 controller are out of service. If one monitored time slot remains in service or returns to service, no alarm is sent.
Examples
The following example configures T1 0 to send a blue (AIS) alarm if DS0 groups 0 and 1 are out of service:
Related Commands
Command
|
Description
|
busyout monitor
|
Configures a voice port to monitor an interface for events that would trigger a voice-port busyout.
|
connection trunk
|
Creates a permanent trunk connection (private line or tie-line) between a voice port and a PBX.
|
voice class permanent
|
Creates a voice class for a Cisco or FRF-11 permanent trunk.
|
alias static
To create a static entry in the local alias table, use the alias static command in gatekeeper configuration mode. To remove a static entry, use the no form of this command.
alias static ip-signaling-addr [port] gkid gatekeeper-name [ras ip-ras-addr port] [terminal | mcu |
gateway {h320 | h323-proxy | voip}] [e164 e164-address] [h323id h323-id]
no alias static ip-signaling-addr [port] gkid gatekeeper-name [ras ip-ras-addr port] [terminal |
mcu | gateway {h320 | h323-proxy | voip}] [e164 e164-address] [h323id h323-id]
Syntax Description
ip-signaling-addr
|
IP address of the H.323 node, used as the address to signal when establishing a call.
|
port
|
(Optional) Port number other than the endpoint Call Signaling well-known port number (1720).
|
gkid gatekeeper-name
|
Name of the local gatekeeper of whose zone this node is a member.
|
ras ip-ras-addr
|
(Optional) Node remote access server (RAS) signaling address. If omitted, the ip-signaling-addr parameter is used in conjunction with the RAS well-known port.
|
port
|
(Optional) Port number other than the RAS well-known port number (1719).
|
terminal
|
(Optional) Indicates that the alias refers to a terminal.
|
mcu
|
(Optional) Indicates that the alias refers to a multiple control unit (MCU).
|
gateway
|
(Optional) Indicates that the alias refers to a gateway.
|
h320
|
(Optional) Indicates that the alias refers to an H.320 node.
|
h323-proxy
|
(Optional) Indicates that the alias refers to an H.323 proxy.
|
voip
|
(Optional) Indicates that the alias refers to VoIP.
|
e164 e164-address
|
(Optional) Specifies the node E.164 address. This keyword and argument can be used more than once to specify as many E.164 addresses as needed. Note that there is a maximum number of 128 characters that can be entered for this address. To avoid exceeding this limit, you can enter multiple alias static commands with the same call signaling address and different aliases.
|
h323id h323-id
|
(Optional) Specifies the node H.323 alias. This keyword and argument can be used more than once to specify as many H.323 identification (ID) aliases as needed. Note that there is a maximum number of 256 characters that can be entered for this address. To avoid exceeding this limit, you can enter multiple alias static commands with the same call signaling address and different aliases.
|
Defaults
No static aliases exist.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on the Cisco 2500 and 3600 series.
|
12.0(3)T
|
This command was integrated into Cisco IOS Release 12.0(3)T.
|
Usage Guidelines
The local alias table can be used to load static entries by performing as many of the commands as necessary. Aliases for the same IP address can be added in different commands, if required.
Typically, static aliases are needed to access endpoints that do not belong to a zone (that is, they are not registered with any gatekeeper) or whose gatekeeper is inaccessible for some reason.
Examples
The following example creates a static terminal alias in the local zone:
zone local gk.zone1.com zone1.com
alias static 191.7.8.5 gkid gk.zone1.com terminal e164 14085551212 h323id bobs_terminal
alt-dial
To configure an alternate dial-out string for dial peers on the Cisco MC3810 multiservice concentrator, use the alt-dial command in dial peer configuration mode. To delete the alternate dial-out string, use the no form of this command.
alt-dial string
no alt-dial string
Syntax Description
string
|
The alternate dial-out string.
|
Defaults
No alternate dial-out string is configured.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
This command applies to Cisco MC3810 multiservice concentrator plain old telephone service (POTS), Voice over Frame Relay (VoFR), and Voice over ATM (VoATM) dial peers.
The alt-dial command is used for the on-net-to-off-net alternative dialing function. The string replaces the destination-pattern string for dialing out.
Examples
The following example configures an alternate dial-out string of 9,5559871:
answer-address
To specify the full E.164 telephone number to be used to identify the dial peer of an incoming call, use the answer-address command in dial peer configuration mode. To disable the configured telephone number, use the no form of this command.
answer-address [+]string[T]
no answer-address
Syntax Description
+
|
(Optional) Character indicating an E.164 standard number.
|
string
|
Series of digits that specify the E.164 or private dial plan telephone number. Valid entries are as follows:
• Digits 0 through 9, letters A through D, pound sign (#), and asterisk (*), which represent specific digits that can be entered.
• Comma (,), which inserts a pause between digits.
• Period (.), which matches any entered digit.
|
T
|
(Optional) Control character indicating that the answer-address value is a variable-length dial string.
|
Defaults
The default value is enabled with a null string.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
Usage Guidelines
Use the answer-address command to identify the origin (or dial peer) of incoming calls from the IP network. Cisco IOS software identifies the dial peers of a call in one of two ways: either by identifying the interface through which the call is received or through the telephone number configured with the answer-address command. In the absence of a configured telephone number, the peer associated with the interface will be associated with the incoming call.
For calls coming in from a POTS interface, the answer-address command is not used to select an incoming dial peer. The incoming POTS dial peer is selected on the basis of the port configured for that dial peer.
There are certain areas in the world (for example, in certain European countries) where valid telephone numbers can vary in length. Use the optional control character T to indicate that a particular answer-address value is a variable-length dial string. In this case, the system will not match the dialed numbers until the interdigit timeout value has expired.
Note
The Cisco IOS software does not check the validity of the E.164 telephone number;
it accepts any series of digits as a valid number.
Examples
The following example configures the E.164 telephone number 555-9626 as the dial peer of an incoming call:
Related Commands
Command
|
Description
|
destination-pattern
|
Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.
|
port (dial peer)
|
Enables an interface on a PA-4R-DTR port adapter to operate as a concentrator port.
|
prefix
|
Specifies the prefix of the dialed digits for this dial peer.
|
application
To enable a specific interactive voice response (IVR) application on a dial peer, use the application command in dial-peer configuration mode. To remove the application from the dial peer, use the no form of this command.
application application-name [out-bound]
no application application-name [out-bound]
Syntax Description
application-name
|
Indicates the name of the predefined application you wish to enable on the dial peer. For H.323 networks, the application is defined by a Tool Command Language/interactive voice response (TCL/IVR) filename and location. Incoming calls using plain old telephone service (POTS) dial peers and outgoing calls using Multimedia Mail over IP (MMoIP) dial peers are handed off to this application. For Media Gateway Control Protocol (MGCP) or Simple Gateway Control Protocol (SGCP) networks, see the usage guidelines below for valid application names.
|
out-bound
|
The named application will handle the MMoIP dial peer in the outgoing mode.
|
Defaults
No default behavior or values.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(6)NA2
|
This command was introduced on the Cisco 2500 series, 3600 series, and AS5300.
|
12.0(5)T
|
The SGCPAPP application was supported initially on the Cisco AS5300 universal access server in a private release that was not generally available.
|
12.0(7)XK
|
Support for the SGCPAPP application was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.1(3)T
|
The MGCPAPP application was supported initially on the Cisco AS5300 universal access server.
|
12.1(3)XI
|
The out-bound keyword was added for the store-and-forward fax feature on the Cisco AS5300 universal access server.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
Use this command to associate a predefined session application with an incoming POTS dial peer or an outgoing MMoIP dial peer. Calls using this incoming POTS dial peer or this outgoing MMoIP dial peer will be handed to the predefined specified session application.
SGCP Networks
For SGCP networks, enter SGCPAPP in uppercase characters. This application can be applied only to POTS dial peers. Note that SGCP dial peers do not use dial peer hunting.
Note
In Cisco IOS Release 12.2, you cannot mix SGCP and non-SGCP endpoints in the same T1 controller. You also cannot mix SGCP and non-SGCP endpoints in the same DS0 group.
MGCP Networks
For MGCP networks, enter MGCPAPP in upper-case characters. This application can be applied only to POTS dial peers. Note that MGCP dial peers do not use dial peer hunting.
Examples
The following example shows how to define an application and how to apply it to an outbound MMoIP dial peer for the fax onramp operation:
call application voice fax_on_vfc_onramp http://santa/username/clid_4digits_npw_3.tcl
application fax_on_vfc_onramp out-bound
destination-pattern 57108..
session target mailto:$d$@mail-server.cisco.com
The following example shows how to apply the MGCP application to a dial peer:
Related Commands
Command
|
Description
|
call application voice
|
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
|
mgcp
|
Starts the MGCP daemon.
|
sgcp
|
Starts and allocates resources for the SCGP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
arq reject-unknown-prefix
To enable the gatekeeper to reject admission requests (ARQs) for zone prefixes that are not configured, use the arq reject-unknown-prefix command in gatekeeper configuration mode. To reenable the gatekeeper to accept and process all incoming ARQs, use the no form of this command.
arq reject-unknown-prefix
no arq reject-unknown-prefix
Syntax Description
This command has no arguments or keywords.
Defaults
The gatekeeper accepts and processes all incoming ARQs.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(6)Q, 11.3(7)NA
|
This command was introduced.
|
12.0(3)T
|
This command was integrated into Cisco IOS Release 12.0(3)T.
|
Usage Guidelines
Use the arq reject-unknown-prefix command to configure the gatekeeper to reject any incoming ARQs for a destination E.164 address that does not match any of the configured zone prefixes.
When an endpoint or gateway initiates an H.323 call, it sends an ARQ to its gatekeeper. The gatekeeper uses the configured list of zone prefixes to determine where to direct the call. If the called address does not match any of the known zone prefixes, the gatekeeper attempts to hairpin the call out through a local gateway. If you do not want your gateway to do this, then use the arq reject-unknown-prefix command. (The term hairpin is used in telephony. It means to send a call back in the direction from which it came. For example, if a call cannot be routed over IP to a gateway that is closer to the target phone, the call is typically sent back out through the local zone, back the way it came.)
This command is typically used to either restrict local gateway calls to a known set of prefixes or deliberately fail such calls so that an alternate choice on a gateway's rotary dial peer is selected.
Examples
Consider a gatekeeper configured as follows:
zone local gk408 cisco.com
zone remote gk415 cisco.com 172.21.139.91
zone prefix gk408 1408.......
zone prefix gk415 1415.......
In this example configuration, the gatekeeper manages a zone containing gateways to the 408 area code, and it knows about a peer gatekeeper that has gateways to the 415 area code. Using the zone prefix command, the gatekeeper is then configured with the appropriate prefixes so that calls to those area codes hop off in the optimal zone.
If the arq request-unknown-prefix command is not configured, the gatekeeper handles calls in the following way:
•
A call to the 408 area code is routed out through a local gateway.
•
A call to the 415 area code is routed to the gk415 zone, where it hops off on a local gateway.
•
A call to the 212 area code is routed to a local gateway in the gk408 zone.
If the arq reject-unknown-prefix command is configured, the gatekeeper handles calls in the following way:
•
A call to the 408 area code is routed out through a local gateway.
•
A call to the 415 area code is routed to the gk415 zone, where it hops off on a local gateway.
•
A call to the 212 area code is rejected because the destination address does not match any configured prefix.
as
To define an application server for backhaul, use the as command in IUA configuration mode. To disable, use the no form of this command.
Note
All of the ASPs in an application server must be removed before an application server can be unconfigured.
as as-name {localip1 [localip2]} [local-sctp-port] | [fail-over-timer] [sctp-startup-rtx]
[sctp-streams] [sctp-t1init]
no as as-name
Syntax Description
as-name
|
Defines the protocol name (only ISDN is supported).
|
localip1
|
Defines the local IP address(es) for all the ASPs in a particular AS.
|
localip2
|
Defines the local IP address(es) for all the ASPs in a particular AS.
|
local-sctp-port
|
Defines a specific local SCTP port rather than an IUA well-known port.
|
fail-over-timer
|
(Optional) Configures the failover timer for a particular AS.
|
sctp-startup-rtx
|
(Optional) Configures the SCTP maximum startup retransmission timer.
|
sctp-streams
|
(Optional) Configures the number of SCTP streams for a particular AS.
|
sctp-t1init
|
(Optional) Configures the SCTP t1 init timer.
|
Defaults
No default behavior or values.
Command Modes
IUA configuration
Command History
Release
|
Modification
|
12.2(4)T
|
This command was introduced.
|
12.2(11)T
|
This command was integrated into Cisco IOS Release 12.2(11)T and support was added for the Cisco AS5300 platform.
|
12.2(11)T
|
This command was integrated into Cisco IOS Release 12.2(11)T on Cisco 2420, Cisco 2600, Cisco 3600, Cisco 3700, Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 network access server (NAS) platforms.
|
Usage Guidelines
A maximum of two local IP addresses can be specified (note that SCTP has built-in support for multi-homed machines).
The default value of the SCTP streams is determined by the hardware that you have installed. The value of failover timer is found in the show iua as all command output.
The number of streams to assign to a given association is implementation dependent. During the initialization of the IUA association, you need to specify the total number of streams that can be used. Each D channel is associated with a specific stream within the association. With multiple trunk group support, every interface can potentially be a separate D channel.
At start-up the IUA code checks for all the possible T1, E1, or T3 interfaces and sets the total number of inbound and outbound streams supported accordingly. In most cases, there is only a need for one association between the GW and the MGC. For the rare case that you are configuring multiple AS associations to various MGCs, the overhead from the unused streams would have minimal impact. The NFAS D channels are configured for one or more interfaces, where each interface is assigned a unique stream ID.
The total number of streams for the association needs to include an additional stream for the SCTP management messages. So during start-up the IUA code adds one to the total number of interfaces (streams) found.
You have the option to manually configure the number of streams per association. In the backhaul scenario, if the number of D channel links is limited to one, allowing the number of streams to be configurable avoids the unnecessary allocation of streams in an association that will never be used. For multiple associations between a GW and multiple MGCs, the configuration utility is useful in providing only the necessary number of streams per association. The overhead from the streams allocated but not used in the association is negligible.
If the number of streams is manually configured through the CLI, the IUA code cannot distinguish between a start-up event, which automatically sets the streams to the number of interfaces, or if the value is set manually during runtime. If you are configuring the number of SCTP streams manually, you must add one plus the number of interfaces using the sctp-streams keyword. Otherwise, IUA needs to always add one for the management stream, and the total number of streams increments by one after every reload.
When you set the SCTP stream with CLI, you cannot change the inbound and outbound stream support once the association is established with SCTP. The value takes effect when you first remove the IUA AS
configuration and then configure it back as the same AS or a new one. The other option is to reload the router.
Examples
An application server (AS) and the application server process (ASP) should be configured first to allow a National ISDN-2 with Cisco extensions (NI2+) to be bound to this transport layer protocol. The AS is a logical representation of SCTP local end point. The local end point can have more than one IP address but must use the same port number.
The following is an example of an AS configuration on a gateway:
Router(config-iua)# as as5400-3 10.1.2.34, 10.1.2.35 2577
In the configuration above, an AS named as5400-3 is configured to use two local IP addresses and a port number of 2577.
The following output shows options available when you use this command:
Router(config-iua)# as as5400-3 fail-over ?
<1000-10000> set Fail-Over time (in milliseconds) between 1 and 10 seconds
Router(config-iua)# as as5400-3 sctp-stre ?
<2-57> Specify number of SCTP streams for association
Router(config-iua)# as as5400-3 sctp-startup ?
<2-20> Set SCTP Maximum Startup Retransmission Interval
Router(config-iua)# as as5400-3 sctp-t1init ?
<1000-60000> Set SCTP T1 init timer (in milliseconds)
Related Commands
Command
|
Description
|
asp
|
Defines an ASP for backhaul.
|
asp
To define an ASP for backhaul, use the asp command in IUA configuration mode. To disable, use the no form of this command.
Note
All of the ASPs in an application server must be removed before an application server can be unconfigured.
asp asp-name as as-name {remoteip1 [remoteip2]} [remote-sctp-port] | [ip-precedence
[sctp-keepalives] [sctp-max-associations] [sctp-path-retransmissions] [sctp-t3-timeout]
no asp asp-name
Syntax Description
asp-name
|
Names the current ASP.
|
as
|
The application server to which the ASP belongs.
|
as-name
|
Name of the application server to which the ASP belongs.
|
remoteip1
|
Designates the remote IP address for this SCTP association.
|
remoteip2
|
Designates the remote IP address for this SCTP association.
|
remote-sctp-port
|
Connects to a remote SCTP port rather than the IUA well-known port.
|
ip-precedence
|
(Optional) Sets IP Precedence bits for protocol data units (PDUs). IP precedence is expressed in the type of service (ToS) field of the show ip sctp association parameters output. Default ToS is 0.
|
sctp-keepalives
|
(Optional) Modifies the keepalive behavior of an IP address in a particular ASP. The default is 500 ms (see the show ip sctp association parameters output under heartbeats).
|
sctp-max-associations
|
(Optional) Sets the SCTP max association retransmissions for a particular ASP. The default value is 3.
|
sctp-path- retransmissions
|
(Optional) Sets the SCTP path retransmissions for a particular ASP. The default value is 5.
|
sctp-t3-timeout
|
(Optional) Sets the SCTP T3 retransmission timeout for a particular ASP. The default value is 900 ms.
|
Defaults
No default behavior or values.
Command Modes
IUA configuration
Command History
Release
|
Modification
|
12.2(4)T
|
This command was introduced.
|
12.2(11)T
|
This command was integrated into Cisco IOS Release Cisco IOS Release 12.2(11)T on Cisco 2420, Cisco 2600, Cisco 3600, Cisco 3700, Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 network access server (NAS) platforms.
|
Usage Guidelines
You can configure the precedence value in IUA in the range of 0 through 7 for a given IP address. Within IUA, the upper three bits representing the IP precedence in the ToS byte (used in the IP header) is set based on the user input before passing down the value to SCTP. In turn, SCTP passes the ToS byte value to IP. The default value is 0 for "normal" IP precedence handling.
The asp-name argument specifies the name of this ASP. The ip-precedence keyword sets the precedence and ToS field. The remote-ip_address argument specifies the IP address of the remote end-point (the address of MGC, for example). The number argument can be any IP precedence bits in the range 1 through 255.
The no form of the command results in precedence bits not being explicitly set by SCTP. The default is to set all bits in the ToS field to zero by SCTP.
In the case of a hot-standby PGW pair, from the GW perspective there is usually be one ASP active and another in the INACTIVE state. The ASP_UP message is used to bring the ASP state on the GW to the INACTIVE state, followed by the ASPTM message, ASP_ACTIVE to ready the IUA link for data exchange (eventually the QPTM Establish Request message actually initiates the start of the D channel for the given interface). In the event that the GW detects a failure on the active ASP, it can send a NTFY message to the standby ASP to request that it become active.
Examples
An ASP can be viewed as a local representation of an SCTP association since it specifies a remote end point that will be in communication with an AS local end point. An ASP is defined for a given AS. For example, the following configuration defines a remote signaling controller asp-name at two IP addresses for AS as-name. The remote SCTP port number is 2577:
AS as-name 10.4.8.69, 10.4.9.69 2477
ASP asp-name AS as-name 10.4.8.68 10.4.9.68 2577
Multiple ASPs can be defined for a single AS for the purpose of redundancy, but only one ASP can be active. The ASPs are inactive and only become active after fail-over.
In the Cisco MGC solution, a signaling controller is always the client that initiates the association with
a gateway. During the initiation phase, you can request outbound and inbound stream numbers, but the
gateway only allows a number that is at least one digit higher than the number of interfaces (T1/E1)
allowed for the platform.
The following shows options for this command:
Router(config-iua)# asp asp-name ip-precedence 10.1.2.345 ?
<0-7> specify precedence level (0 - 7)
default use default value of IP precedence for this address
Router(config-iua)# asp asp-name sctp-keep ?
A.B.C.D specify the IP address to enable/disable keep alives
Router(config-iua)# asp asp-name sctp-keep 10.1.2.234 ?
<1000-60000> specify keep alive interval (in milliseconds)
Router(config-iua)# asp asp-name sctp-max-ass ?
<2-20> specify maximum associations
default use default value of max associations for this address
Router(config-iua)# asp asp-name sctp-path-retran ?
<2-10> specify maximum path retransmissions
default use default value of max path retrans for this address
Router(config-iua)# asp asp-name sctp-t3-time ?
<300-60000> specify T3 retransmission timeout (in milliseconds)
default use default value of T3 for this address
Related Commands
Command
|
Description
|
as
|
Defines an application server (AS) for backhaul.
|
atm scramble-enable
To enable scrambling on E1 links, use the atm scramble-enable command in interface configuration mode. To disable scrambling, use the no form of this command.
atm scramble-enable
no atm scramble-enable
Syntax Description
This command has no arguments or keywords.
Defaults
By default, payload scrambling is set off.
Command Modes
Interface configuration
Command History
Release
|
Modification
|
12.0(5)XK
|
This command was introduced for ATM interface configuration on the Cisco MC3810 multiservice concentrator.
|
12.0(7)T
|
This command was integrated into Cisco IOS Release 12.0(7)T.
|
Usage Guidelines
Enable scrambling on E1 links only. On T1 links, the default B8ZS line encoding normally ensures sufficient reliability. Scrambling improves data reliability on E1 links by randomizing the ATM cell payload frames to avoid continuous nonvariable bit patterns and to improve the efficiency of the ATM cell delineation algorithms.
The scrambling setting must match that of the far end.
Examples
On a Cisco MC3810, the following example shows how to set the ATM0 E1 link to scramble payload:
atm video aesa
To set the unique ATM end-station address (AESA) for an ATM video interface that is using switched virtual circuit (SVC) mode, use the atm video aesa command in ATM interface configuration mode. To remove any configured address for the interface, use the no form of this command.
atm video aesa [default | esi-address]
no atm video aesa
Syntax Description
default
|
(Optional) Automatically creates a network service access point (NSAP) address for the interface, based on a prefix from the ATM switch (26 hexadecimal characters), the MAC address (12 hexadecimal characters) as the end station identifier (ESI), and a selector byte (two hexadecimal characters).
|
esi-address
|
(Optional) Defines the 12 hexadecimal characters used as the ESI. The ATM switch provides the prefix (26 hexadecimal characters), and the video selector byte provides the remaining two hexadecimal characters.
|
Defaults
default
Command Modes
ATM Interface configuration
Command History
Release
|
Modification
|
12.0(5)XK
|
This command was introduced for ATM interface configuration on the Cisco MC3810 multiservice concentrator.
|
12.0(7)T
|
This command was integrated into Cisco IOS Release 12.0(7)T.
|
Usage Guidelines
You cannot specify the ATM interface NSAP address in its entirety. The system creates either all of the address or part of it, depending on how you use this command.
Examples
On a Cisco MC3810 multiservice concentrator, the following example shows the ATM interface NSAP address set automatically:
On a Cisco MC3810 multiservice concentrator, the following example shows the ATM interface NSAP address set to a specific ESI value:
atm video aesa 444444444444
Related Commands
Command
|
Description
|
show atm video-voice address
|
Displays the NSAP address for the ATM interface.
|
audio-prompt load
To initiate loading the selected audio file (.au), the file that contains the announcement prompt for the caller from Flash memory into RAM, use the audio-prompt load command in privileged EXEC mode.
audio-prompt load name
Syntax Description
name
|
Indicates the location of the audio file that you want to have loaded from memory, Flash memory, or an FTP server.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(6)NA2
|
This command was introduced.
|
Usage Guidelines
The first time the interactive voice response (IVR) application plays a prompt, it reads it from the URL (or the specified location for the .au file, such as Flash or TFP) into RAM. Then it plays the script from RAM. An example of the sequence of events is as follows:
•
When the first caller is asked to enter the account and personal identification numbers (PINs), the enter_account.au and enter_pin.au files are loaded into RAM from Flash memory.
•
When the next call comes in, these prompts are played from the RAM copy.
•
If all callers enter valid account numbers and PINs, then the auth_failed.au file is not loaded from Flash memory into RAM memory.
The router will load the audio file only when the script initially plays that prompt after the router restarts. If the audio file is changed, you must run this EXEC command to reread the file. This will generate an error message if the file is not accessible or if there is a format error.
Note
With Cisco IOS Release 11.3(6)NA2, the URL pointer refers to the directory where Flash memory is stored.
Examples
The following example shows how to load the enter_pin.au audio file from Flash memory into RAM:
audio-prompt load flash:enter_pin.au
auto-cut-through
To enable call completion when a PBX does not provide an M-lead response, use the auto-cut-through command in voice-port configuration mode. To disable the auto-cut-through operation, use the no form of this command.
auto-cut-through
no auto-cut-through
Syntax Description
This command has no arguments or keywords.
Defaults
Auto-cut-through is enabled.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
The auto-cut-through command applies to ear and mouth (E&M) voice ports only.
Examples
The following example shows enabling of call completion on a Cisco MC3810 multiservice concentrator when a PBX does not provide an M-lead response:
The following example shows enabling of call completion on a Cisco 2600 or 3600 router when a PBX does not provide an M-lead response:
Related Commands
Command
|
Description
|
show voice port
|
Displays voice port configuration information.
|
backhaul-session-manager
To enter backhaul session manager configuration mode, use the backhaul-session-manager command in global configuration mode.
backhaul-session-manager
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
Usage Guidelines
Use the backhaul-session-manager command to enter the backhaul session manager configuration mode. Table 3 lists the backhaul session manager configuration mode commands:
Table 3 Backhaul Session Manager Configuration Mode Commands
Command
|
Description
|
group
|
Creates a session-group and associates it with a specified session-set.
|
group auto-reset
|
Configures the maximum auto-reset value.
|
group cumulative-ack
|
Configures maximum cumulative acknowledgments.
|
group out-of-sequence
|
Configures maximum out-of-sequence segments that are received before an EACK is sent.
|
group receive
|
Configures maximum receive segments.
|
group retransmit
|
Configures maximum retransmits.
|
group timer cumulative-ack
|
Configures cumulative acknowledgment timeout.
|
group timer keepalive
|
Configures keepalive (or null segment) timeout.
|
group timer retransmit
|
Configures retransmission timeout.
|
group timer transfer
|
Configures state transfer timeout.
|
session group
|
Associates a transport session with a specified session-group.
|
set
|
Creates a fault-tolerant or non-fault-tolerant session-set with the client or server option.
|
Examples
The following example shows how to enter backhaul-session-manager configuration mode:
Router(config)# backhaul-session-manager
Related Commands
Command
|
Description
|
clear backhaul-session-manager group
|
Resets the stastistics or traffic counters for a specified session-group.
|
clear rudpv1 statistics
|
Clears the RUDP statistics and failure counters.
|
group
|
Creates a session-group and associates it with a specified session-set.
|
group auto-reset
|
Configures the maximum auto-reset value.
|
group cumulative-ack
|
Configures maximum cumulative acknowledgments.
|
group out-of-sequence
|
Configures maximum out-of-sequence segments that are received before an EACK is sent.
|
group receive
|
Configures maximum receive segments.
|
group retransmit
|
Configures maximum retransmits.
|
group timer cumulative-ack
|
Configures cumulative acknowledgment timeout.
|
group timer keepalive
|
Configures keepalive (or null segment) timeout.
|
group timer retransmit
|
Configures retransmission timeout.
|
group timer transfer
|
Configures state transfer timeout.
|
isdn bind-l3
|
Configures the ISDN serial interface for backhaul.
|
session group
|
Associates a transport session with a specified session-group.
|
set
|
Creates a fault-tolerant or non-fault-tolerant session-set with the client or server option.
|
show backhaul-session-manager group
|
Displays status, statistics, or configuration of a specified or all session-groups.
|
show backhaul-session-manager session
|
Displays status, statistics, or configuration of sessions.
|
show backhaul-session-manager set
|
Displays session-groups associated with a specific or all session-sets.
|
show rudpv1
|
Displays RUDP statistics.
|
bandwidth
To specify the maximum aggregate bandwidth for H.323 traffic, use the bandwidth command in gatekeeper configuration mode. To remove the maximum aggregate bandwidth value, use the no form of this command.
bandwidth {interzone | total | session} {default | zone zone-name} bandwidth-size
no bandwidth {interzone | total | session} {default | zone zone-name} bandwidth-size
Syntax Description
interzone
|
Specifies the maximum bandwidth for H.323 traffic between one zone and another zone.
|
total
|
Specifies the maximum bandwidth for H.323 traffic within a zone and between zones (intrazone and interzone).
|
session
|
Specifies the maximum bandwidth allowed for a single session in a specific zone or in all zones.
|
default
|
Specifies the maximum bandwidth for all applicable zones, depending on the keyword with which it is used.
|
zone
zone-name
|
Specifies a particular zone.
Names the particular zone.
|
bandwidth-size
|
Maximum bandwidth. For interzone and total, the range is from 1 to 10,000,000 kbps. For session, the range is from 1 to 5000 kbps.
|
Defaults
None
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on the Cisco 2500 series and 3600 series routers and on the AS5300 universal access server.
|
12.1(5)T
|
The bandwidth command replaced the zone bw command.
|
Usage Guidelines
The functionality of this command in previous Cisco IOS software releases was enabled by using the zone bw command.
To specifiy maximum bandwidth for traffic between one zone and any other zone, use the default keyword with the interzone keyword.
To specify maximum bandwidth for traffic within one zone or for traffic between that zone and another zone (interzone and intrazone), use the default keyword with the total keyword.
To specify maximum bandwidth for a single session within a specific zone, use the zone keyword with the session keyword.
To specify maximum bandwidth for a single session within any zone, use the default keyword with the session keyword.
Examples
The following example configures the default maximum bandwidth for traffic between one zone and another zone to 5000 kbps:
bandwidth interzone default 5000
The following example configures the default maximum bandwidth for all zones to 5000 kbps:
bandwidth total default 5000
The following example configures the default maximum bandwidth for a single session within any zone to 2000 kbps:
bandwidth session default 2000
The following example configures the default maximum bandwidth for a single session with a specific zone to 1000 kbps:
bandwidth session zone denver 1000
Related Commands
Command
|
Description
|
bandwidth remote
|
Specifies the total bandwidth for H.323 traffic between this gatekeeper and any other gatekeeper.
|
h323 interface
|
Defines on which port the proxy will listen.
|
h323 t120
|
Enables the T.120 capabilities on your router and specifies bypass or proxy mode.
|
bandwidth remote
To specify the total bandwidth for H.323 traffic between this gatekeeper and any other gatekeeper, use the bandwidth remote command in gatekeeper configuration mode. To disable the total bandwidth specified, use the no form of this command.
bandwidth remote bandwidth-size
no bandwidth remote bandwidth-size
Syntax Description
bandwidth-size
|
Maximum bandwidth. The range is from 1 to 10,000,000 kbps.
|
Defaults
None
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
12.1(5)T
|
This command was introduced on the Cisco 2600, 3600, and 7200 series routers and on the MC3810 multiservice concentrator.
|
Usage Guidelines
The functionality of this command in previous Cisco IOS software releases was enabled by using the zone gatekeeper command.
Examples
The following example configures the remote maximum bandwidth to 100,000 kbps:
Related Commands
Command
|
Description
|
bandwidth
|
Specifies the maximum aggregate bandwidth for H.323 traffic from a zone to another zone, within a zone, or for a session in a zone.
|
h323 interface
|
Defines which port the proxy will listen on.
|
h323 t120
|
Enables the T.120 capabilities on your router and specifies bypass or proxy mode.
|
battery-reversal
To specify battery polarity reversal on a Foreign Exchange Office (FXO) or Foreign Exchange Station (FXS) port, use the battery-reversal command in voice-port configuration mode. To disable battery reversal, use the no form of this command.
battery-reversal
no battery-reversal
Syntax Description
This command has no arguments or keywords.
Defaults
Battery reversal is enabled.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco 2600 and 3600 series routers and on the MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
The battery-reversal command applies to FXO and FXS voice ports. On Cisco 2600 and 3600 series routers, only analog voice ports in VIC-2FXO-M1 and VIC-2FXO-M2 voice interface cards are able to detect battery reversal; analog voice ports in VIC-2FXO and VIC-2FXO-EU voice interface cards do not detect battery reversal. On digital voice ports, battery reversal is supported only on E1 Mercury Exchange Limited Channel Associated Signaling (MELCAS); it is not supported in T1 channel associated signaling (CAS) or E1 CAS.
FXS ports normally reverse battery upon call connection. If an FXS port is connected to an FXO port that does not support battery reversal detection, you can use the no battery-reversal command on the FXS port to prevent unexpected behavior.
FXO ports in loopstart mode normally disconnect calls when they detect a second battery reversal (back to normal). You can use the no battery-reversal command on FXO ports to disable this action.
The battery-reversal command restores voice ports to their default battery-reversal operation.
Examples
The following example disables battery reversal on voice port 1/1 on a Cisco MC3810:
The following example disables battery reversal on voice port 1/0/0 on a Cisco 2600 or 3600 series router:
Related Commands
Command
|
Description
|
show voice port
|
Displays voice port configuration information.
|
block-caller
To configure call blocking on caller ID, use the block-caller command in dial peer voice configuration mode. To disable call blocking on caller ID, use the no form of this command.
block-caller number
no block-caller number
Syntax Description
number
|
Specifies the telephone number to block. You can use a period (.) as a digit wildcard. For example, the command block-caller 5.51234 blocks all numbers beginning with the digit 5, followed by any digit, and then sequentially followed by the digits 5, 1, 2, 3, and 4.
|
Defaults
Call blocking is disabled; the router does not block any calls for any listed directory numbers (LDNs) based on caller ID numbers.
Command Modes
Dial peer voice configuration
Command History
Release
|
Modification
|
12.1.(2)XF
|
This command was introduced on the Cisco 800 series routers.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
This command is available on Cisco 800 series routers that have plain old telephone service (POTS) ports. For each dial peer, you can enter up to ten caller ID numbers to block. The routers do not accept additional caller ID numbers if ten numbers are already present. In that case, a number must be removed before another caller ID number can be added for blocking.
If you do not specify the block-caller command for a local directory, all voice calls to that local directory are accepted. If you specify the block-caller command for a local directory, the router verifies that the incoming calling-party number does not match any caller ID numbers in that local directory before processing or accepting the voice call. Each specified caller ID number and incoming calling-party number is compared from right to left, up to the number of digits in the specified caller ID number or incoming calling-party number, whichever has fewer digits.
This command is effective only if you subscribe to caller ID service. If you enable call blocking on caller ID without subscribing to the caller ID service, the routers do not perform the verification process on calling-party numbers and do not block any calls.
Examples
The following example configures a router to block calls from a caller whose caller ID number is 408-555-1234.
Related Commands
Command
|
Description
|
caller-id
|
Identifies incoming calls with caller ID.
|
debug pots csm csm
|
Activates events from which an application can determine and display the status and progress of calls to and from POTS ports.
|
isdn i-number
|
Configures several terminal devices to use one subscriber line.
|
pots call-waiting
|
Enables local call waiting on a router.
|
registered-caller ring
|
Configures the Nariwake service registered caller ring cadence.
|
busyout forced
To force a voice port into the busyout state, use the busyout forced command in voice-port configuration mode. To remove the voice port from the busyout state, use the no form of this command.
busyout forced
no busyout forced
Syntax Description
This command has no arguments or keywords.
Defaults
The voice-port is not in the busyout state.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.0(3)T
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
This command was first supported on the Cisco 2600 and 3600 series routers. On the Cisco MC3810, the voice-port busyout command was eliminated in favor of this command.
|
12.1(2)T
|
The modifications in Cisco IOS Release 12.0(7)XK were integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
If a voice port is in the forced busyout state, only the no busyout forced command can restore the voice port to service.
To avoid conflicting command-line interface (CLI) commands, do not use the busyout forced command and the ds0 busyout command on the same controller.
Examples
The following example forces analog voice port 1/1 on a Cisco MC3810 multiservice concentrator into the busyout state:
The following example forces digital voice port 0:8 on a Cisco MC3810 multiservice concentrator into the busyout state:
The following example forces analog voice port 3/1/1 on a Cisco 3600 router into the busyout state:
The following example forces digital voice port 0/0:12 on a Cisco 3600 router into the busyout state:
Related Commands
Command
|
Description
|
busyout-monitor interface
|
Configures a voice port to monitor a serial interface for events that would trigger a voice-port busyout.
|
busyout seize
|
Changes the busyout seize procedure for a voice port.
|
show voice busyout
|
Displays information about the voice busyout state.
|
busyout monitor
To place a voice port into the busyout monitor state, enter the busyout monitor command in voice-port configuration mode. To remove the busyout monitor state from the voice port, use the no form of this command.
busyout monitor {serial interface-number | ethernet interface-number} [in-service]
no busyout monitor {serial interface-number | ethernet interface-number}
Syntax Description
serial
|
Specifies monitoring of a serial interface. More than one interface can be entered for a voice port.
|
ethernet
|
Specifies monitoring of an Ethernet interface. More than one interface can be entered for a voice port.
|
interface-number
|
Identifies an interface to be monitored for the voice port busyout function.
Interface choices include serial port, serial port subinterface, Ethernet port, and ATM interface.
|
in-service
|
(Optional) Configures the voice port to be busied out when any monitored interface comes into service (its state changes to up). If the keyword is not entered, the voice port is busied out when all monitored interfaces go out of service (their state changes to down).
|
Defaults
The voice port does not monitor any interfaces.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.0(3)T
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(5)XE
|
This command was implemented on the Cisco 7200 series routers.
|
12.0(5)XK
|
This command was implemented on the Cisco 2600 and 3600 series routers.
|
12.0(7)T
|
The Cisco 2600 and 3600 series router implementation was integrated into Cisco IOS Release 12.0(7)T.
|
12.0(7)XK
|
The ability to monitor an Ethernet port was introduced and the in-service keyword was added. The serial keyword was first supported on the Cisco 2600 and 3600 series routers.
|
12.1(1)T
|
The implementation of this command on the Cisco 7200 series routers was integrated into Cisco IOS Release 12.1(1)T.
|
12.1(2)T
|
The serial and ethernet keywords were added, the in-service keyword was integrated into Cisco IOS Release 12.1(2)T, and the interface number argument was changed to go with the serial and Ethernet keywords.
|
12.1(3)T
|
The interface keyword was removed.
|
Usage Guidelines
When you place a voice port in the busyout monitor state, the voice port monitors the specified interface and enters the busyout state when the interface is down. This down state forces the rerouting of calls.
The command monitors only the up or down status of an interface—not end-to-end TCP/IP connectivity.
When an interface is operational, a busied-out voice port returns to its normal state.
This feature can monitor LAN, WAN, and virtual interfaces as well as subinterfaces.
The Cisco 2600 and 3600 series routers and the MC3810 multiservice concentrator support ATM interfaces. To monitor an ATM interface, enter ATM and the interface number.
A voice port can monitor multiple interfaces at the same time. To configure a voice port to monitor multiple interfaces, reenter the busyout monitor command for each additional interface to be monitored.
If you specify more than one monitored interface for a voice port, all the monitored interfaces must be down to trigger busyout on the voice port.
You can combine in-service and out-of-service monitoring on a voice port. The following rule describes the actions if monitored interfaces change state.
A voice port is busied out if either of the following occurs:
•
Any interface monitored for coming into service comes up.
•
All interfaces monitored for going out of service go down.
Examples
The following example shows configuration of analog voice port 1/1 on a Cisco MC3810 multiservice concentrator to busyout if serial ports 1 and 0:0 both go out of service:
busyout monitor serial 0:0
The following example shows configuration of analog voice port 1/2 on a Cisco MC3810 multiservice concentrator to busy out if serial port 0 or 1 comes into service:
busyout monitor serial 0 in-service
busyout monitor serial 1 in-service
The following example shows configuration of digital voice port 1/2/2 on a Cisco 3600 series router to busy out if serial port 0 goes out of service:
voice-port 1/2/2
busyout monitor serial 0
The following example shows configuration of digital voice port 0:6 on a Cisco MC3810 multiservice concentrator to busy out if both Ethernet port 0 and serial port 0 go out of service:
busyout monitor ethernet 0
The following example shows configuration of the voice port to monitor two serial interfaces and an Ethernet interface. When all these interfaces are down, the voice port is busied out. When at least one interface is operating, the voice port is put back into a normal state.
busyout monitor ethernet 0/0
busyout monitor serial 1/0
busyout monitor serial 2/0
Related Commands
Command
|
Description
|
busyout forced
|
Forces a voice port into the busyout state.
|
busyout monitor probe
|
Configures a voice port to enter the busyout state if a Service Assurance Agent (SAA) probe signal returned from a remote, IP-addressable interface crosses a specified delay or loss threshold.
|
busyout seize
|
Changes the busyout seize procedure for a voice port.
|
show voice busyout
|
Displays information about the voice busyout state.
|
voice-port busyout
|
Places all voice ports associated with a serial or ATM interface into a busyout state.
|
busyout monitor probe
To configure a voice port to enter the busyout state if a Service Assurance Agent (SAA) probe signal returned from a remote, IP-addressable interface crosses a specified delay or loss threshold, use the busyout monitor probe command in voice-port configuration mode. To configure a voice port not to monitor SAA probe signals, use the no form of this command.
busyout monitor probe ip-address [codec codec-type] [icpif number | loss percent delay
milliseconds]
no busyout monitor probe ip-address
Syntax Description
ip-address
|
The IP address of a target interface for the SAA probe signal.
|
codec
|
(Optional) Configures the profile of the SAA probe signal to mimic the packet size and interval of a specific codec type.
|
codec-type
|
(Optional) The codec type for the SAA probe signal.
Available options are as follows:
• g711a—G.711 A-law
• g711u—G.711 U-law (the default)
• g729—G.729
• g729a—G.729
|
icpif
|
(Optional) Configures the busyout monitor probe to use an Impairment/Calculated Planning Impairment Factor (ICPIF) loss/delay busyout threshold, in accordance with ITU-T G.113. The ICPIF numbers represent predefined combinations of loss and delay.
|
number
|
(Optional) The ICPIF threshold for initiating a busyout. The range is from 0 to 30. Lower numbers are equivalent to lower loss and delay thresholds.
|
loss
|
(Optional) Configures the percentage-of-packets-lost threshold for initiating a busyout.
|
percent
|
(Optional) The loss value (expressed as a percentage) for initiating a busyout. The range is from 1 to 100.
|
delay
|
(Optional) Configures the average packet delay threshold for initiating a busyout.
|
milliseconds
|
(Optional) The delay threshold, in milliseconds, for initiating a busyout. The range is from 1 to 2147483647.
|
Defaults
If the busyout monitor probe command is not entered, the voice port does not monitor SAA probe signals.
If the busyout monitor probe command is entered with no optional keywords or arguments, the default codec type is G.711 alaw, and the default loss and delay thresholds are the threshold values configured with the pstn fallback command.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 and 3600 series and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
A voice port can monitor multiple interfaces at the same time. To configure a voice port to monitor multiple interfaces, enter the busyout monitor probe command for each additional interface to be monitored.
The busyout monitor probe command is effective only if the call fallback function is enabled on this router and the SAA responder is enabled on the target router.
The SAA probe is transmitted periodically with a period determined by the call fallback function.
Refer to the PSTN Fallback feature module for Cisco IOS Release 12.1(3)T for details of the call fallback function and ICPIF values.
Lower thresholds of ICPIF, loss, and delay result in earlier busyout when the link deteriorates, thereby raising the voice minimum quality level. Higher thresholds prevent busyout until loss and delay are greater, allowing transmission of lower-quality voice.
Caution 
If thresholds are set too low, the link can alternate between in-service and out-of-service states, causing repeated interruptions of traffic.
Examples
The following example configures analog voice port 1/1 on a Cisco MC3810 multiservice concentrator to use an SAA probe with a G.711alaw profile to probe the link to two remote interfaces that have IP addresses and to busy out the voice port. Both links have a loss exceeding 25 percent or a packet delay of more than 1.5 seconds.
busyout monitor probe 209.165.202.128 codec g711a loss 25 delay 1500
busyout monitor probe 209.165.202.129 codec g711a loss 25 delay 1500
Related Commands
Command
|
Description
|
busyout monitor
|
Places a voice port into the busyout monitor state.
|
pstn fallback
|
Forces a voice port into the busyout state.
|
show voice busyout
|
Displays information about the voice busyout state.
|
voice class busyout
|
Creates a voice class for local voice busyout functions.
|
busyout seize
To change the busyout action for a Foreign Exchange Office (FXO) or Foreign Exchange Station (FXS) voice port, use the busyout seize command in voice-port configuration mode. To restore the default busyout action, use the no form of this command.
busyout seize {ignore | repeat}
no busyout seize
Syntax Description
ignore
|
Specifies the type of ignore procedure, depending on the type of voice port signaling. See Table 4 for more information.
|
repeat
|
Specifies the type of repeat procedure, depending on the type of voice port signaling. See Table 4 for more information.
|
Defaults
See Table 4 for the default actions for different voice ports and signaling types.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.0(3)T
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
This command was first supported on Cisco 2600 and 3600 series routers.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
The busyout seize command is valid for both analog and digital voice ports. On digital voice ports, the busyout actions are valid whether the busyout results from a voice-port busyout event or from the ds0-busyout command.
The voice port returns to an idle state when the event that triggered the busyout disappears.
Table 4 describes the busyout actions for the busyout seize settings on each voice port type.
The busyout action for E and M voice ports is to seize the far end by setting lead busy.
Table 4 Busyout Seize Actions for Voice Ports
Voice Port Signaling Type
|
Procedure Setting
(busyout-option command)
|
Busyout Actions
|
FXS loop start
|
Default
|
Removes the power from the loop. For analog voice ports, this is equivalent to removing the ground from the tip lead. For digital voice ports, the port will generate the bit pattern equivalent to removing the ground from the tip lead, or it will busy out if the bit pattern exists.
|
FXS loop start
|
Ignore
|
Ignores the ground on the ring lead.
|
FXS ground start
|
Default
|
Grounds the tip lead and stays at this state.
|
FXS ground start
|
Ignore
|
1. Leaves the tip lead open.
2. Ignores the ground on the ring lead.
|
FXS ground start
|
Repeat
|
1. Grounds the tip lead.
2. Waits for the far end to close the loop.
3. The far end closes the loop.
4. If the far end then opens the loop, FXS removes the ground from the tip lead.
5. FXS waits for several seconds before returning to Step 1.
|
FXO loop start
|
Default
|
Closes the loop and stays at this state.
|
FXO loop start
|
Ignore
|
1. Leaves the loop open.
2. Ignores the ringing current on the ring level.
|
FXO loop start
|
Repeat
|
1. Closes the loop.
2. After the detected far end starts the power denial procedure, FXO opens the loop.
3. After the detected far end has completed the power denial procedure, FXO waits for several seconds before returning to Step 1.
|
FXO ground start
|
Default
|
Grounds the tip lead.
|
FXO ground start
|
Ignore
|
1. Leaves the loop open.
2. Ignores the running current on the ring lead, or the ground current on the tip lead.
|
FXO ground start
|
Repeat
|
1. Grounds the ring lead.
2. Removes the ground from the ring lead and closes the loop after the detected far end grounds the tip lead.
3. When the detected far end removes the ground from tip lead, FXO opens the loop.
4. FXO waits for several seconds before returning to Step 1.
|
Examples
The following example shows configuration of analog voice port 1/1 to perform the ignore actions when busied out:
voice-port 1/1
busyout seize ignore
The following example shows configuration of digital voice port 0:2 to perform the repeat actions when busied out:
voice-port 0:2
busyout seize repeat
Related Commands
Command
|
Description
|
busyout forced
|
Forces a voice port into the busyout state.
|
busyout-monitor interface
|
Configures a voice port to monitor an interface for events that would trigger a voice port busyout.
|
ds0 busyout
|
Forces a DS0 time slot on a controller into the busyout state.
|
show voice busyout
|
Displays information about the voice busyout state.
|
voice-port busyout
|
Places all voice ports associated with a serial or ATM interface into a busyout state.
|
cac master
To configure the call admission control (CAC) operation as master, enter the cac master command in voice-service configuration mode. To restore the default value, use the no form of this command.
cac master
no cac master
Syntax Description
No arguments or keywords
Defaults
The Cisco MC3810 multiservice concentrator is enabled as a CAC slave.
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.1(1)XA
|
The command was introduced for the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
You should configure the Cisco MC3810 multiservice concentrators at opposite ends of an AAL2 trunk for the opposite CAC operation—master at one end and slave at the other end.
A Cisco MC3810 multiservice concentrator configured as a master always performs CAC during fax/modem upspeed. A Cisco MC3810 multiservice concentrator configured as a slave sends a request for CAC to the CAC master.
Examples
The following example shows configuration of the CAC operation of a Cisco MC3810 multiservice concentrator as master:
voice service voatm
session protocol aal2
The following example shows the CAC operation of a Cisco MC3810 multiservice concentrator being returned to slave:
voice service voatm
session protocol aal2
cadence-list
To specify a tone cadence pattern to be detected, use the cadence-list command in voice-class configuration mode. To delete a cadence pattern, use the no form of this command.
cadence-list cadence-id cycle-1-on-time cycle-1-off-time [cycle-2-on-time cycle-2-off-time]
[cycle-3-on-time cycle-3-off-time] [cycle-4-on-time cycle-4-off-time]
no cadence-list cadence-id
Syntax Description
cadence-id
|
A tag to identify this cadence list. The range is from 1 to 10.
|
cycle-1-on-time
|
The tone duration for the first cycle of the cadence pattern. The range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
|
cycle-1-off-time
|
The silence duration for the first cycle of the cadence pattern. The range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
|
cycle-2-on-time
|
(Optional) The tone duration for the second cycle of the cadence pattern. The range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
|
cycle-2-off-time
|
(Optional) The silence duration for the second cycle of the cadence pattern. The range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
|
cycle-3-on-time
|
(Optional) The tone duration for the third cycle of the cadence pattern. The range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
|
cycle-3-off-time
|
(Optional) The silence duration for the third cycle of the cadence pattern. The range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
|
cycle-4-on-time
|
(Optional) The tone duration for the fourth cycle of the cadence pattern. The range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
|
cycle-4-off-time
|
(Optional) The silence duration for the fourth cycle of the cadence pattern. The range is from 0 to 1000 (0 milliseconds to 100 seconds). The default is 0.
|
Defaults
No cadence pattern is configured.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 and Cisco 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
A cadence list enables the router to match a complex tone pattern from a PBX or public switched telephone network (PSTN). A tone is detected if it matches any configured cadence list. You can create up to ten cadence lists, enabling the router to detect up to ten different tone patterns. If the tone to be detected consists of only one on-off cycle, you can configure this in either of two ways:
•
Create a cadence list using only the cycle-1-on-time and cycle-1-off-time variables.
•
Use the cadence-max-off-time and cadence-min-on-time commands.
You must also configure the times of the cadence-max-off-time and cadence-min-on-time commands to be compatible with the on and off times specified by the cadence-list command. The time of the cadence-max-off-time must be equal to or greater than the longest off-time in the cadence list; the cadence-min-on-time must be equal to or less than the shortest on-time in the cadence list.
Examples
The following example shows configuration of cadence list 1 with three on/off cycles and cadence list 2 with two on/off cycles for voice class 100:
cadence-list 1 100 100 300 300 100 200
cadence-list 2 100 200 100 400
Related Commands
Command
|
Description
|
cadence-max-off-time
|
Specifies the maximum off duration for detection of a tone.
|
cadence-min-on-time
|
Specifies the minimum on duration for detection of a tone.
|
voice class dualtone
|
Creates a voice class for FXO tone detection parameters.
|
cadence-max-off-time
To specify the maximum off duration for detection of a tone, use the cadence-max-off-time command in voice-class configuration mode. To restore the default, use the no form of this command.
cadence-max-off-time time
no cadence-max-off-time
Syntax Description
time
|
The maximum off time of a tone that can be detected, in 10-millisecond increments. The range is from 0 to 5000 (0 milliseconds to 50 seconds). The default is 0.
|
Defaults
No cadence maximum off time is configured.
Command Modes
Voice-class configuration
Related Commands
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 and 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
You must specify a time value greater than the off time of the tone to be detected. You must specify a time value greater than 0 to enable detection of a tone. With the default (0), the router will detect only a continuous tone.
Examples
The following example shows configuration of a maximum off duration of 20 seconds for voice class 100:
cadence-max-off-time 2000
Related Commands
Command
|
Description
|
cadence-min-on-time
|
Specifies the minimum on duration for detection of a tone.
|
cadence-variation
|
Specifies the cadence variation time allowed for detection of a tone.
|
voice class dualtone
|
Creates a voice class for FXO tone detection parameters.
|
cadence-min-on-time
To specify the minimum on duration for detection of a tone, use the cadence-min-on-time command in voice-class configuration mode. To restore the default, use the no form of this command.
cadence-min-on-time time
no cadence-min-on-time
Syntax Description
time
|
The minimum on time of a tone that can be detected, in 10-millisecond increments. The range is from from 0 to 100 (0 milliseconds to 1 seconds). The default is 0.
|
Defaults
No cadence minimum on time is configured.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 and 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
You must specify a time value shorter than the on time of the tone to be detected. With the default (0), a tone of any length will be detected.
Examples
The following example shows configuration of a minimum on duration of 30 milliseconds for voice class 100:
Related Commands
Command
|
Description
|
cadence-max-off-time
|
Specifies the maximum off duration for detection of a tone.
|
cadence-variation
|
Specifies the cadence variation time allowed for detection of a tone.
|
voice class dualtone
|
Creates a voice class for Foreign Exchange Office (FXO) tone detection parameters.
|
cadence-variation
To specify the cadence variation time allowed for detection of a tone, use the cadence-variation command in voice-class configuration mode. To restore the default cadence variation time, use the no form of this command.
cadence-variation time
no cadence-variation
Syntax Description
time
|
The maximum time by which the tone onset can vary from the specified onset time and still be detected, in 10-millisecond increments. The range is from 0 to 200 (0 milliseconds to 2 seconds). The default is 0.
|
Defaults
Zero for zero millisecond cadence variation allowed
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 and 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
You should specify a time value greater than the cadence variation of the tone to be detected. With the default of 0, only those tones that match the configured cadence will be detected.
Examples
The following example shows configuration of a cadence variation time of 30 milliseconds for voice class 100:
Related Commands
Command
|
Description
|
cadence-max-off-time
|
Specifies the maximum off duration for detection of a tone.
|
cadence-min-on-time
|
Specifies the minimum on duration for detection of a tone.
|
voice class dualtone
|
Creates a voice class for FXO tone detection parameters.
|
call application cache reload time
To configure the router to reload the Media Gateway Control Protocol (MGCP) scripts from cache on a regular interval, use the call application cache reload time command in global configuration mode. To set the value to the default, use the no form of this command.
call application cache reload time bg-minutes
no call application cache reload time
Syntax Description
bg-minutes
|
Specifies the number of minutes after which the background process is awakened. This background process checks the time elapsed since the script was last used and whether the script is current:
• If the script has not been used in the last "unload time," it will unload the script and quit. The unload time is not configurable.
• If the script has been used, the background process will load the script from the URL. It compares the scripts, and if they do not match, it begins using the new script for new calls.
|
Defaults
30 minutes
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Examples
The following example displays the call application cache reload time command configured to specify 30 minutes before a background process is awakened:
call application cache reload time 30
Related Commands
Command
|
Description
|
call application voice load
|
Allows reload of an application that was loaded via the MGCP scripting package.
|
show call application voice
|
Displays all TCL or MGCP scripts that are loaded.
|
call application voice
To create an application and to indicate the location of the corresponding Tool Command Language (TCL) files that implement this application, use the call application voice command in global configuration mode. To remove the defined application and all configured parameters associated with it, use the no form of this command.
call application voice application-name location {word}
no call application voice application-name location {word}
Syntax Description
application-name
|
Character string that defines the name of the application.
|
location
|
Location of the TCL file in URL format. Valid storage locations are TFTP, FTP, and Flash.
|
word
|
Text string that defines an attribute-value pair specified by the TCL script and understood by the RADIUS server.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
12.1(3)T
|
The word argument was added for attribute-value (AV) pairs.
|
Usage Guidelines
Use this command when configuring interactive voice response (IVR) or one of the IVR-related features (such as Debit Card) to define the name of an application and to identify the location of the TCL script associated with this application.
Note
The command no call application voice application-name removes the entire application and all parameters, if configured.
Examples
This example shows how to define the application "prepaid" and the TFTP server location of the associated TCL script:
call application voice prepaid tftp://keyer/debitcard.tcl
The following is an example of AV pair configuration:
set avsend(h323-ivr-out,)) "payphone:true"
set avsend(323-ivr-out,1) "creditTime:3400"
The AV pair (after the array is defined, as in the prior example) must be sent to the server, using the authentication, authorization, and accounting (AAA) authenticate or AAA authorize verbs as follows:
aaa authenticate $account $password $avsend
The script would use this AV pair whenever it is needed to convey information to the RADIUS server that cannot be represented by the standard vendor-specific attributes (VSAs).
Related Commands
Command
|
Description
|
call application voice language
|
Defines the language of the audio file for the designated application and passes that information to the application.
|
call application voice load
|
Reloads the designated TCL script.
|
call application voice pin-len
|
Defines the number of characters in the PIN for the application and passes that information to the application.
|
call application voice redirect-number
|
Defines the telephone number to which a call will be redirected—for example, the operator telephone number of the service provider—for the designated application.
|
call application voice retry-count
|
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
|
call application voice set-location
|
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
|
call application voice uid-len
|
Defines the number of characters in the UID for the designated application and passes that information to the application.
|
call application voice warning-time
|
Defines the number of seconds of warning that a user receives before the allowed calling time runs out for the designated application.
|
call application voice access-method
To specify the access method for two-stage dialing for the designated application, use the call application voice access-method command in global configuration mode. To restore default values for this command, use the no form of this command.
call application voice application-name access-method {prompt-user | redialer}
no call application voice application-name access-method
Syntax Description
application-name
|
The name of the application.
|
prompt-user
|
Specifies that no direct inward dialing (DID) is set in the incoming plain old telephone service (POTS) dial peer and that a Tool Command Language (TCL) script in the incoming POTS dial peer will be used for two-stage dialing.
|
redialer
|
Specifies that no DID is set in the incoming POTS dial peer and that the redialer device will be used for two-stage dialing.
|
Defaults
Prompt-user when DID is not set in the dial peer
Command Modes
Global configuration mode
Command History
Release
|
Modification
|
12.1(3)XI
|
This command was introduced on the Cisco AS5300 universal access server.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
Use the call application voice access-method command to specify the access method for two-stage dialing when DID is disabled in the POTS dial peer.
Examples
The following example specifies prompt-user as the access method for two-stage dialing for the app_libretto_onramp9 IVR application:
call application voice app_libretto_onramp9 access-method prompt-user
Related Commands
Command
|
Description
|
call application voice
|
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
|
call application voice language
|
Defines the language of the audio file for the designated application and passes that information to the application.
|
call application voice load
|
Reloads the designated TCL script.
|
call application voice pin-len
|
Defines the number of characters in the PIN for the application and passes that information to the application.
|
call application voice redirect-number
|
Defines the telephone number to which a call will be redirected—for example, the operator telephone number of the service provider—for the designated application.
|
call application voice retry-count
|
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
|
call application voice set-location
|
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
|
call application voice uid-len
|
Defines the number of characters in the UID for the designated application and passes that information to the application.
|
call application voice warning-time
|
Defines the number of warning seconds a user receives before the allowed calling time runs out for the designated application.
|
call application voice accounting enable
To enable authentication, authorization, and accounting (AAA) accounting for a Tool Command Language (TCL) application, use the call application voice accounting enable command in global configuration mode. To disable accounting for a TCL application, use the no form of this command.
call application voice application-name accounting enable
no call application voice application-name accounting enable
Syntax Description
application-name
|
The name of the application.
|
Defaults
Disabled
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)XI
|
This command was introduced on the Cisco AS5300 universal access server.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
This command enables AAA accounting services if an AAA accounting method list has been defined using both the aaa accounting command and the mmoip aaa method fax accounting command.
This command applies to off-ramp store-and-forward fax functions on Cisco AS5300 universal access server voice feature cards (VFCs). It is not used on modem cards.
Examples
The following example enables AAA accounting to be used with outbound store-and-forward fax:
call application voice app_libretto_onramp9 accounting enable
Related Commands
Command
|
Description
|
mmoip aaa method fax accounting
|
Defines the name of the method list to be used for AAA accounting with store-and-forward fax.
|
call application voice accounting-list
To define the accounting list name of the voice feature card (VFC), use the call application voice accounting-list command in global configuration mode. To restore the default value, use the no form of this command.
call application voice application-name accounting-list method-list-name
no call application voice application-name accounting-list method-list-name
Syntax Description
application-name
|
The name of the application.
|
method-list-name
|
Character string used to name a list of accounting methods to be used with store-and-forward fax.
|
Defaults
No AAA accounting method list is defined.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)XI
|
This command was introduced on the Cisco AS5300 universal access server.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
This command defines the name of the accounting feature of the authentication, authorization, and accounting (AAA) method list to be used with store-and-forward fax. The method list itself, which defines the type of accounting services provided for store-and-forward fax, is defined using the aaa accounting global configuration command. Unlike standard AAA (where each defined method list can be applied to specific interfaces and lines), the AAA accounting method lists used in store-and-forward fax are applied globally on the Cisco AS5300 universal access server.
After the accounting method lists have been defined, they are enabled by using the mmoip aaa receive-accounting enable command.
This command applies to both on-ramp and off-ramp store-and-forward fax functions on Cisco AS5300 universal access server voice feature cards. It is not used on modem cards.
Examples
The following example defines a AAA accounting method list (called "sherman") to be used with store-and-forward fax:
call application voice app_libretto_onramp9 accounting-list sherman
Related Commands
Command
|
Description
|
call application voice accounting enable
|
Enables on-ramp AAA accounting services.
|
call application voice authen-list
To specify the name of an authentication method list for a Tool Command Language (TCL) application, use the call application voice authen-list command in global configuration mode. To disable the authentication method list for a TCL application, use the no form of this command.
call application voice application-name authen-list method-list-name
no call application voice application-name authen-list method-list-name
Syntax Description
application-name
|
The name of the application.
|
method-list-name
|
Character string used to name a list of authentication methods to be used with T.38 fax relay and T.37 store-and-forward fax.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)XI
|
This command was introduced on the Cisco AS5300 universal access server.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
This command defines the name of the authentication, authorization, and accounting (AAA) authentication method list to be used with fax applications on voice feature cards. The method list itself, which defines the type of authentication services provided for store-and-forward fax, is defined using the aaa authentication global configuration command. Unlike standard AAA (where each defined method list can be applied to specific interfaces and lines), AAA authentication method lists used with fax applications are applied globally on the Cisco AS5300 universal access server.
After the authentication method lists have been defined, they are enabled by using the call application voice authentication enable command.
Examples
The following example defines an AAA authentication method list (called "fax") to be used with T.38 fax relay and T.37 store-and-forward fax:
call application voice app_libretto_onramp9 authen-list fax
Related Commands
Command
|
Description
|
call application voice authentication enable
|
Enables AAA authentication services for a TCL application.
|
call application voice authen-method
|
Specifies the authentication method for a TCL application.
|
call application voice authen-method
To specify an authentication, authorization, and accounting (AAA) authentication method for a Tool Command Language (TCL) application, use the call application voice authen-method command in global configuration mode. To disable the authentication method for a TCL application, use the no form of this command.
call application voice application-name authen-method {prompt-user | ani | dnis | gateway |
redialer-id | redialer-dnis}
no call application voice application-name authen-method {prompt-user | ani | dnis | gateway |
redialer-id | redialer-dnis}
Syntax Description
application-name
|
The name of the application.
|
prompt-user
|
Indicates that the user is prompted for the TCL application account identifier.
|
ani
|
Indicates that the calling-party telephone number (automatic number identification [ANI]) is used as the TCL application account identifier.
|
dnis
|
Indicates that the called party telephone number (dialed number identification service [DNIS]) is used as the TCL application account identifier.
|
gateway
|
Indicates that the router-specific name derived from the host name and domain name is used as the TCL application account identifier. It is displayed in the following format: router-name.domain-name.
|
redialer-id
|
Indicates that the account string returned by the external redialer device is used as the TCL application account identifier. In this case, the redialer ID is either the redialer serial number or the redialer account number.
|
redialer-dnis
|
Indicates that the called party telephone number (dialed number identification service or DNIS) is used as the TCL application account identifier captured by the redialer if a redialer device is present.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)XI
|
This command was introduced on the Cisco AS5300 universal access server.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
Normally, when AAA is being used for simple user authentication, AAA uses the username information defined in the user profile for authentication. With T.37 store-and-forward fax and T.38 real-time fax, you can specify that the ANI, DNIS, gateway identification (ID), redialer ID, or redialer DNIS be used to identify the user for authentication or that the user be prompted for the TCL application.
Examples
The following example shows how to configure the router-specific name derived from the host name and domain name as the TCL application account identifier for the app_libretto_onramp9 TCL application:
call application voice app_libretto_onramp9 authen-method gateway
Related Commands
Command
|
Description
|
call application voice authentication enable
|
Enables AAA authentication services for a TCL application.
|
call application voice authen-list
|
Specifies the name of an authentication method list for a TCL application.
|
call application voice authentication enable
To enable AAA authentication services for a tool command line (TCL) application, use the call application voice authentication enable command in global configuration mode. To disable authentication for a TCL application, use the no form of this command.
call application voice application-name authentication enable
no call application voice application-name authentication enable
Syntax Description
application-name
|
The name of the application.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)XI
|
This command was introduced on the Cisco AS5300 universal access server.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
This command enables AAA authentication services for a TCL application if a AAA authentication method list has been defined using both the aaa authentication command and the call application voice authen-list command.
Examples
The following example enables a AAA authentication method list (called peabody) to be used with outbound store and forward fax.
call application voice app_onramp6 authen-list peabody
call application voice app_onramp6 authentication enable
Related Commands
Command
|
Description
|
call application voice authen-list
|
Specifies the name of an authentication method list for a TCL application.
|
call application voice authen-method
|
Specifies the authentication method for a TCL application.
|
call application voice global-password
To define a password to be used with CiscoSecure for Windows NT when using store-and-forward fax on a voice feature card, use the call application voice global-password command in global configuration mode. To restore the default value, use the no form of this command.
call application voice application-name global-password password
no call application voice application-name global-password password
Syntax Description
application-name
|
The name of the application.
|
password
|
Character string used to define the CiscoSecure for Windows NT password to be used with store-and-forward fax. The maximum length is 64 alphanumeric characters.
|
Defaults
No password is defined.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)XI
|
This command was introduced on the Cisco AS5300 universal access server.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
CiscoSecure for Windows NT might require a separate password to complete authentication, no matter what security protocol you use. This command defines the password to be used with CiscoSecure for Windows NT. All records on the Windows NT server use this defined password.
This command applies to on-ramp store-and-forward fax functions on Cisco AS5300 universal access server voice feature cards. It is not used on modem cards.
Examples
The following example shows a password (abercrombie) being used by AAA for the app_libretto_onramp9 TCL application:
call application voice app_libretto_onramp9 global-password abercrombie
call application voice language
To define the language of the audio file for the specified application and to pass that information to the specified application, use the call application voice language command in global configuration mode. To remove the associated language of the audio file from the application, use the no form of this command.
call application voice application-name language number language
no call application voice application-name language number language
Syntax Description
application-name
|
The name of the application to which the language parameters are being passed.
|
number
|
Tag that uniquely identifies an audio file. Valid entries are from 0 to 9.
|
language
|
Defines the language of the associated audio file. Valid entries are as follows:
• en—English
• sp—Spanish
• ch—Mandarin
• aa—all
|
Defaults
No default behavior or values.
Command Modes
Global configuration mode
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced.
|
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool Command Language (TCL) script being used—or one of the IVR-related features (such as Debit Card) to define the language of the audio file for the specified application and to pass that information to the specified application.
Table 5 lists TCL script names and the corresponding parameters that are required for each TCL script.
Table 5 TCL Scripts and Parameters
TCL Script Name
|
Description
|
Parameters to Configure
|
clid_4digits_npw_3_cli.tcl
|
This script authenticates the account number and personal identification number (PIN), respectively, using automatic number identification (ANI) and NULL. The numbers of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, it allows the caller to retry. The retry number is also configured through the CLI.
|
• call application voice uid-len minimum = 1, maximum = 20, default = 10
• call application voice pin-len minimum = 0, maximum = 10, default = 4
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_col_npw_cli.tcl
|
This script authenticates the account number and PIN, respectively, using ANI and NULL. If the authentication fails, it allows the caller to retry. The retry number is configured through CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_collect_cli.tcl
|
This script authenticates the account number and PIN using ANI and dialed number identification service (DNIS). If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_3_cli.tcl
|
This script authenticates using ANI and NULL for account numbers and PINs, respectively. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_npw_cli.tcl
|
This script authenticates using ANI and NULL for account and PIN, respectively. If authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected together.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
Examples
The following example shows how to define English and Spanish as the languages of the audio files associated with the application (named "prepaid"):
call application voice prepaid language 1 en
call application voice prepaid language 2 sp
Related Commands
Command
|
Description
|
call application voice
|
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
|
call application voice load
|
Reload the designated TCL script.
|
call application voice pin-len
|
Defines the number of characters in the PIN for the application and passes that information to the application.
|
call application voice redirect-number
|
Defines the telephone number to which a call will be redirected—for example, the operator telephone number of the service provider—for the designated application.
|
call application voice retry-count
|
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
|
call application voice set-location
|
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
|
call application voice uid-len
|
Defines the number of characters in the UID for the designated application and passes that information to the application. the application.
|
call application voice warning-time
|
Defines the number of second's warning a user receives before the allowed calling time runs out for the designated application.
|
call application voice load
To reload the selected Tool Command Language (TCL) script from the URL, use the call application voice load command in privileged EXEC mode.
call application voice load name
Syntax Description
name
|
Defines the TCL script to use for the call. Enter the name of the TCL or Media Gateway Control Protocol (MGCP) script you want this dial peer to use.
|
Defaults
TCL or scripts are not loaded.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco 2600 series routers, Cisco 3600 series (except for the 3660), and on the Cisco AS5300.
|
12.1(3)T
|
Support for dynamic script loading of MGCP scripts was added.
|
Usage Guidelines
The software checks the signature lock to ensure that it is a Cisco-supported TCL script.
If the TCL script does not have a valid Cisco-supported signature, the software fails to load the script and generates the following error message:
00:02:54: %IVR-3-BAD_IVR_SIG: Script signature is invalid
Examples
The following example shows the loading of a MGCP script package:
Router# call application voice load mgcp-script-pkg
Related Commands
Command
|
Description
|
call application cache reload time
|
Configures the interval for reloading MGCP scripts.
|
call application voice
|
Creates and calls the application that will interact with the IVR feature.
|
show call application voice
|
Displays a list of the voice applications that are configured.
|
call application voice pin-len
To define the number of characters in the personal identification number (PIN) for the designated application, use the call application voice pin-len command in global configuration mode. To restore default values for this command, use the no form of this command.
call application voice application-name pin-len number
no call application voice application-name pin-len number
Syntax Description
application-name
|
The name of the application to which the PIN length parameter is being passed.
|
number
|
Defines the number of allowable characters in PINs associated with the specified application. Valid entries are from 0 to 10.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced in the Cisco 2600 series routers, Cisco 3600 series routers, and Cisco AS5300 universal access routers.
|
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the TCL script being used—or one of the IVR-related features (such as Debit Card) to define the number of allowable characters in a PIN for the specified application and to pass that information to the specified application.
Table 6 lists TCL script names and the corresponding parameters that are required for each TCL script.
Table 6 TCL Scripts and Parameters
TCL Script Name
|
Description
|
Parameters to Configure
|
clid_4digits_npw_3_cli.tcl
|
This script authenticates the account number and PIN, respectively, using automatic number identification (ANI) and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, it allows the caller to retry. The retry number is also configured through the CLI.
|
• call application voice uid-len minimum = 1, maximum = 20, default = 10
• call application voice pin-len minimum = 0, maximum = 10, default = 4
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_col_npw_cli.tcl
|
This script authenticates the account number and PIN, respectively, using ANI and NULL. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_collect_cli.tcl
|
This script authenticates the account number and PIN using ANI and dialed number identification service (DNIS). If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_3_cli.tcl
|
This script authenticates using ANI and NULL for account numbers and PINs, respectively. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_npw_cli.tcl
|
This script authenticates using ANI and NULL for account and PIN, respectively. If authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected together.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
Examples
The following example shows how to define a PIN length of four characters for the application (named "prepaid"):
call application voice prepaid pin-len 4
Related Commands
Command
|
Description
|
call application voice
|
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
|
call application voice language
|
Defines the language of the audio file for the designated application and passes that information to the application.
|
call application voice load
|
Reload this designated TCL script.
|
call application voice redirect-number
|
Defines the telephone number to which a call will be redirected—for example, the operator telephone number of the service provider—for the designated application.
|
call application voice retry-count
|
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
|
call application voice set-location
|
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
|
call application voice uid-len
|
Defines the number of characters in the UID for the designated application and passes that information to the application.
|
call application voice warning-time
|
Defines the number of seconds of warning that a user receives before the allowed calling time runs out for the designated application.
|
call application voice redirect-number
To define the telephone number to which a call will be redirected—for example, the operator telephone number of the service provider—for the designated application, use the call application voice redirect-number command in global configuration mode. To cancel this particular parameter, use the no form of this command.
call application voice application-name redirect-number number
no call application voice application-name redirect-number number
Syntax Description
application-name
|
The name of the application to which the redirect telephone number parameter is being passed.
|
number
|
Defines the designated operator telephone number of the service provider (or any other number designated by the customer). This is the number that calls are terminated to when, for example, debit time allowed has run out or the debit amount is exceeded.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series routers , and the AS5300 universal access server.
|
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool Command Language (TCL) script being used—or one of the IVR-related features (such as Debit Card) to define the telephone number to which a call will be redirected.
Table 7 lists TCL script names and the corresponding parameters that are required for each TCL script.
Table 7 TCL Scripts and Parameters
TCL Script Name
|
Description
|
Parameters to Configure
|
clid_4digits_npw_3_cli.tcl
|
This script authenticates the account number and personal identification number (PIN), respectively, using automatic number identification (ANI) and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, it allows the caller to retry. The retry number is also configured through the CLI.
|
• call application voice uid-len minimum = 1, maximum = 20, default =10
• call application voice pin-len minimum = 0, maximum = 10, default = 4
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_col_npw_cli.tcl
|
This script authenticates the account number and PIN, respectively, using ANI and NULL. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_collect_cli.tcl
|
This script authenticates the account number and PIN using ANI and dialed number identification service (DNIS). If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_3_cli.tcl
|
This script authenticates using ANI and NULL for account and PIN, respectively. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_npw_cli.tcl
|
This script authenticates using ANI and NULL for account numbers and PINs, respectively. If authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected together.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
Examples
The following example shows how to define a redirect number for the application (named "prepaid"):
call application voice prepaid redirect-number 5551111
Related Commands
Command
|
Description
|
call application voice
|
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
|
call application voice language
|
Defines the language of the audio file for the designated application and passes that information to the application.
|
call application voice load
|
Reloads the designated TCL script.
|
call application voice pin-len
|
Defines the number of characters in the PIN for the application and passes that information to the application.
|
call application voice retry-count
|
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
|
call application voice set-location
|
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
|
call application voice uid-len
|
Defines the number of characters in the UID for the designated application and passes that information to the application.
|
call application voice warning-time
|
Defines the number of seconds of warning that a user receives before the allowed calling time runs out for the designated application.
|
call application voice retry-count
To define the number of times a caller is permitted to reenter the personal identification number (PIN) for the designated application, use the call application voice retry-count command in global configuration mode. To cancel this particular parameter, use the no form of this command.
call application voice application-name retry-count number
no call application voice application-name retry-count number
Syntax Description
application-name
|
The name of the application to which the number of possible retries is being passed.
|
number
|
Defines the number of times the caller is permitted to reenter personal identification number (PIN) digits.Valid entries for this parameter are from 1 to 5.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the AS5300.
|
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool Command Language (TCL) script being used—or one of the IVR-related features (such as Debit Card) to define how many times a user can reenter a PIN.
Table 8 lists TCL script names and the corresponding parameters that are required for each TCL script.
Table 8 TCL Scripts and Parameters
TCL Script Name
|
Description Summary
|
Parameters to Configure
|
clid_4digits_npw_3_cli.tcl
|
This script authenticates the account number and PIN, respectively, using automatic number identification (ANI) and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, it allows the caller to retry. The retry number is also configured through the CLI.
|
• call application voice uid-len minimum = 1, maximum = 20, default =10
• call application voice pin-len minimum = 0, maximum = 10, default = 4
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_col_npw_cli.tcl
|
This script authenticates the account number and PIN, respectively, using ANI and NULL. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_collect_cli.tcl
|
This script authenticates the account number and PIN using ANI and dialed number identification service (DNIS). If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_3_cli.tcl
|
This script authenticates using ANI and NULL for account numbers and PINs, respectively. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_npw_cli.tcl
|
This script authenticates using ANI and NULL for account and PIN, respectively. If authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected together.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
Examples
The following example shows how to define that a user can re-enter a PIN three times before being disconnected for the application (named "prepaid"):
call application voice prepaid retry-count 3
Related Commands
Command
|
Description
|
call application voice
|
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
|
call application voice language
|
Defines the language of the audio file for the designated application and passes that information to the application.
|
call application voice load
|
Reloads the designated TCL script.
|
call application voice pin-len
|
Defines the number of characters in the PIN for the application and passes that information to the application.
|
call application voice redirect-number
|
Defines the telephone number to which a call will be redirected—for example, the operator telephone number of the service provider—for the designated application.
|
call application voice set-location
|
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
|
call application voice uid-len
|
Defines the number of characters in the UID for the designated application and passes that information to the application. the application.
|
call application voice warning-time
|
Defines the number of second's warning a user receives before the allowed calling time runs out for the designated application.
|
call application voice set-location
To define the location, language, and category of the audio files for the specified application, use the call application voice set-location command in global configuration mode. To cancel this particular parameter, use the no form of this command.
call application voice application-name set-location language category location
no call application voice application-name set-location language category location
Syntax Description
application-name
|
The name of the application to which the set-location parameters are being passed.
|
language
|
Defines the language associated with the audio files. Possible values for this parameter are as follows:
• en = English
• ch = Mandarin
• sp = Spanish
|
category
|
Defines a particular category group. Audio files can be divided into category groups (from 0 to 4). For example, audio files representing the days and months can be category 1, audio files representing units of currency can be category 2, audio files representing units of time—seconds, minutes, and hours—can be category 3. The minimum is 0; the maximum is 4 (0 means all).
|
location
|
Defines the location (audio file URL or directory in the TFTP server) where the audio files are stored.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the AS5300.
|
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the TCL script being used—or one of the IVR-related features (such as Debit Card) to define the location, language, and category of the audio files for the designated application and pass that information to the application.
Table 9 lists TCL script names and the corresponding parameters that are required for each TCL script.
Table 9 TCL Scripts and Parameters
TCL Script Name
|
Description
|
Parameters to Configure
|
clid_4digits_npw_3_cli.tcl
|
This script authenticates the account number and PIN, respectively, using automatic number identification (ANI) and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, it allows the caller to retry. The retry number is also configured through the CLI.
|
• call application voice uid-len minimum = 1, maximum = 20, default =10
• call application voice pin-len minimum = 0, maximum = 10, default = 4
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_col_npw_cli.tcl
|
This script authenticates the account number and PIN, respectively, using ANI and NULL. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_collect_cli.tcl
|
This script authenticates the account number and PIN using ANI and dialed number identification service (DNIS). If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_3_cli.tcl
|
This script authenticates using ANI and NULL for account and PIN respectively. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_npw_cli.tcl
|
This script authenticates using ANI and NULL for account numbers and PINs, respectively. If authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected together.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
Examples
The following example shows how to configure the call application voice set-location command for the application (named "prepaid"). In this example, the language defined is English, the category into which the audio files are group is Category 0 (meaning all), and the location is the keyer directory on the TFTP server.
call application voice prepaid set-location en 0 tftp://keyer/
Related Commands
Command
|
Description
|
call application voice
|
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
|
call application voice language
|
Defines the language of the audio file for the designated application and passes that information to the application.
|
call application voice load
|
Reloads the designated TCL script.
|
call application voice pin-len
|
Defines the number of characters in the PIN for the application and passes that information to the application.
|
call application voice redirect-number
|
Defines the telephone number to which a call will be redirected—for example, the operator telephone number of the service provider—for the designated application.
|
call application voice retry-count
|
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
|
call application voice uid-len
|
Defines the number of characters in the UID for the designated application and passes that information to the application. the application.
|
call application voice warning-time
|
Defines the number of seconds of warning that a user is warned before their allowed calling time runs out for the designated application.
|
call application voice uid-length
To define the number of characters in the user identification number (UID) for the designated application, use the call application voice uid-length command in global configuration mode. To delete the specification of the number of characters in the user identification number, use the no form of this command.
call application voice application-name uid-length number
no call application voice application-name uid-length number
Syntax Description
application-name
|
The name of the application to which the UID length parameter is being passed.
|
number
|
Defines the number of allowable characters in UIDs associated with the specified application. Valid entries are from 1 to 20.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco 2600 series routers, Cisco 3600 series, and on the Cisco AS5300.
|
Usage Guidelines
Use this command when configuring interactive voice response (IVR), depending on the Tool Command Language (TCL) script being used or one of the IVR-related features (such as Debit Card) to define the number of allowable characters in a UID for the specified application and to pass that information to the specified application.
Table 7 lists TCL script names and the corresponding parameters that are required for each TCL script.
Table 10 TCL Script Names and Parameters
TCL Script Name
|
Description
|
Parameters to Configure
|
clid_4digits_npw_3_cli.tcl
|
This script authenticates the account number and PIN, respectively, using automatic number identification (ANI) and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, it allows the caller to retry. The retry number is also configured through the CLI.
|
• call application voice uid-len minimum = 1, maximum = 20, default =10
• call application voice pin-len minimum = 0, maximum = 10, default = 4
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_col_npw_cli.tcl
|
This script authenticates the account number and personal identification number (PIN), respectively, using ANI and NULL. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_collect_cli.tcl
|
This script authenticates the account number and PIN using ANI and dialed number identification service (DNIS). If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_3_cli.tcl
|
This script authenticates using ANI and NULL for account numbers and PINs, respectively. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_npw_cli.tcl
|
This script authenticates using ANI and NULL for account numbers and PINs, respectively. If authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected together.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
Examples
The following example shows how to configure four allowable characters in the UID for the application (named "prepaid"):
call application voice prepaid uid-len 4
Related Commands
Command
|
Description
|
call application voice
|
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
|
call application voice language
|
Defines the language of the audio file for the designated application and passes that information to the application.
|
call application voice load
|
Reloads the designated TCL script.
|
call application voice pin-len
|
Defines the number of characters in the PIN for the application and passes that information to the application.
|
call application voice redirect-number
|
Defines the telephone number to which a call will be redirected—for example, the operator telephone number of the service provider—for the designated application.
|
call application voice retry-count
|
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
|
call application voice set-location
|
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
|
call application voice warning-time
|
Defines the number of seconds of warning that a user receives before the allowed calling time runs out for the designated application.
|
call application voice warning-time
To define the number of seconds of warning that a user receives before the allowed calling time runs out, use the call application voice warning-time command in global configuration mode. To restore default values for this command, use the no form of this command.
call application voice application-name warning-time number
no call application voice application-name warning-time number
Syntax Description
application-name
|
The name of the application to which the warning time parameter is being passed.
|
number
|
Defines the length of the warning period, in seconds, before the allowed calling time runs out. Valid entries are from 10 to 600.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series routers, and AS5300 universal access server.
|
Usage Guidelines
Use this command when configuring interactive voice response (IVR)—depending on the Tool Command Language (TCL) script being used—or one of the IVR-related features (such as Debit Card) to define how many seconds in the warning period before the allowed calling time runs out for the specified application and to pass that information to the specified application.
Table 11 lists TCL script names and the corresponding parameters that are required for each TCL script.
.
Table 11 TCL Scripts and Parameters
TCL Script Name
|
Description
|
Parameters to Configure
|
clid_4digits_npw_3_cli.tcl
|
This script authenticates the account number and PIN, respectively, using automatic number identification (ANI) and NULL. The number of digits allowed for the account number and password, respectively, are configurable through the command-line interface (CLI). If the authentication fails, it allows the caller to retry. The retry number is also configured through the CLI.
|
• call application voice uid-len minimum = 1, maximum = 20, default =10
• call application voice pin-len minimum = 0, maximum = 10, default = 4
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_col_npw_cli.tcl
|
This script authenticates the account number and personal identification number (PIN), respectively, using ANI and NULL. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_authen_collect_cli.tcl
|
This script authenticates the account number and PIN using ANI and dialed number identification service (DNIS). If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected separately.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_3_cli.tcl
|
This script authenticates using ANI and NULL for account and PIN, respectively. If the authentication fails, it allows the caller to retry. The retry number is configured through the CLI.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
clid_col_npw_npw_cli.tcl
|
This script authenticates using ANI and NULL for account numbers and PINs, respectively. If authentication fails, it allows the caller to retry. The retry number is configured through the CLI. The account number and PIN are collected together.
|
• call application voice retry-count minimum = 1, maximum = 5, default = 3
|
Examples
The following example shows how to configure a 30-second warning time for the application (named "prepaid"):
call application voice prepaid warning-time 30
Related Commands
Command
|
Description
|
call application voice language
|
Defines the language of the audio file for the designated application and passes that information to the application.
|
call application voice load
|
Reloads the designated TCL script.
|
call application voice location
|
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
|
call application voice pin-len
|
Defines the number of characters in the PIN for the application and passes that information to the application.
|
call application voice redirect-number
|
Defines the telephone number to which a call will be redirected—for example, the operator telephone number of the service provider—for the designated application.
|
call application voice retry-count
|
Defines the number of times a caller is permitted to reenter the PIN for a designated application and passes that information to the application.
|
call application voice set-location
|
Defines the location, language, and category of the audio files for the designated application and passes that information to the application.
|
call application voice uid-len
|
Defines the number of characters in the UID for the designated application and passes that information to the application. the application.
|
call fallback active
To enable a call request to fall back to alternate dial peers in case of network congestion, use the call fallback active command in global configuration mode. To disable public switched telephone network (PSTN) fall back, use the no form of this command.
call fallback active
no call fallback active
Syntax Description
This command has no arguments or keywords.
Defaults
This command is disabled by default.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
Enabling the call fallback active command determines whether calls should be accepted or rejected based on probing of network conditions. The command call fallback active checks each H.323 call request and rejects the call if the network congestion parameters are greater than the value of the configured threshold parameters of the destination. If this is the case, alternative dial peers are tried from the session application layer.
Use the call fallback threshold delay loss or call fallback threshold icpif command to set the threshold parameters.
Connected calls are not affected by this feature.
Examples
The following example enables the call fallback active command:
Related Commands
Command
|
Description
|
call fallback cache-size
|
Specifies the call fallback cache size for network traffic probe entries.
|
call fallback cache-timeout
|
Specifies the time after which the cache entries of network conditions are purged.
|
call fallback instantaneous-value-weight
|
Configures the call fallback subsystem to take an average from the last two cache entries for call requests.
|
call fallback jitter-probe num-packets
|
Specifies the number of packets in a jitter probe used to determine network conditions.
|
call fallback jitter-probe precedence
|
Specifies the priority of the jitter-probe transmission.
|
call fallback jitter-probe priority-queue
|
Assigns a priority-queue for jitter-probe transmissions.
|
call fallback key-chain
|
Specifies use of MD5 authentication for sending and receiving SAA probes.
|
call fallback map address-list
|
Configures the call fallback router to keep a cache table by IP addresses of distances for several destination peers sitting behind the router.
|
call fallback map subnet
|
Configures the call fallback router to keep a cache table by subnet addresses of distances for several destination peers sitting behind the router.
|
call fallback probe-timeout
|
Sets the timeout for a SAA probe for call fallback purposes.
|
call fallback threshold delay loss
|
Configures the call fallback threshold to use solely packet delay and loss values.
|
call fallback threshold icpif
|
Configures call fallback to use the Impairment/Calculated Planning Impairment Factor (ICPIF) threshold.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback cache-size
To specify the call fallback cache size for network traffic probe entries, use the call fallback cache-size command in global configuration mode. To restore the default value, use the no form of this command.
call fallback cache-size number
no call fallback cache-size number
Syntax Description
number
|
Specifies the cache size in number of entries. The valid range is from 1 to 256.
|
Defaults
128 entries
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
The cache size can be changed only when the call fallback active command is not enabled.
The pverflow process deletes up to one-fourth of the cache entries to allow for additional calls beyond the specified cache size. The cache entries chosen for deletion are the oldest entries in the cache.
Examples
The following example specifies 120 cache entries:
call fallback cache-size 120
When call fallback is already configured, the output is as follows:
Cache size left unchanged (can be changed only when Fallback is OFF (use no call fallback)
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
call fallback cache-timeout
|
Specifies the time after which the cache entry is purged.
|
show call fallback cache
|
Displays the current IPCIF estimates for all IP addresses in the cache.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback cache-timeout
To specify the time after which the cache entries of network conditions are purged, use the call fallback cache-timeout command in global configuration mode. To disable, use the no form of this command.
call fallback cache-timeout seconds
no call fallback cache-timeout seconds
Syntax Description
seconds
|
Specifies the cache timeout value in seconds. The valid range is from 1 to 2,147,483.
|
Defaults
600 seconds
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
Enabling the call fallback cache-timeout command sends a Service Assurance Agent (SAA) probe out to the network to determine the amount of congestion in terms of configured thresholds.The network condition terms are based on delay and loss, or Impairment/Calculated Planning Impairment Factor (ICPIF) thresholds. Use the call fallback threshold delay loss or call fallback threshold icpif command to set the threshold parameters.
The cache keeps entries for every network congestion-checking probe sent and received between timeouts. The cache updates after each probe returns the current condition of network traffic. To set the probe frequency, use the call fallback probe-timeout command.
A call comes into the router. The router matches a dial peer and obtains the destination information. The router calls the fall back subsystem to look up the specified destination in its network traffic cache. If the delay and loss or ICPIF threshold 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 in the cache), then the call is connected.
After each call request, the timer is reset. Purging of the cache occurs only when the cache has received no call requests during the timeout (seconds) period. When the cache timeout expires, the entire cache is deleted, and a probe is sent to start a new cache entry. A call cannot be completed until this probe returns with network traffic information.
The network congestion probes continue in the background as long as the entry for the last call request remains in the cache.
Examples
The following example specifies 1200 seconds before the cache times out:
call fallback cache-timeout 1200
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
call fallback cache-size
|
Specifies the call fallback cache size.
|
call fallback probe-timeout
|
Specifies the call fallback probe timeout.
|
call fallback threshold delay loss
|
Configures the call fallback threshold to use only packet delay and loss values.
|
call fallback threshold icpif
|
Configures the call fallback to use the ICPIF threshold.
|
show call fallback cache
|
Displays the current ICPIF estimates for all IP addresses in the cache.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback instantaneous-value-weight
To configure the call fallback subsystem to take an average from the last two probes registered in the cache for call requests, use the call fallback instantaneous-value-weight command in global configuration mode. To return to the default values, use the no form of this command.
call fallback instantaneous-value-weight weight
no call fallback instantaneous-value-weight weight
Syntax Description
weight
|
Specifies the instantaneous value weight. The valid range is from 0 to 100 percent.
|
Defaults
66 percent
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
Probes returning with network congestion information are logged into the cache to determine whether the next call request will be granted. The network can be busy regularly, and the cache entries reflect these heavy traffic conditions. However, one probe may return with low traffic conditions, which are in contrast to normal conditions. All call requests received between the time of this probe and the next use this entry to determine call acceptance. These calls are allowed through the network, but before the next probe is sent and received, the normal heavy traffic conditions may have returned. The calls sent through congest the network and result in worse traffic conditions.
Use the call fallback instantaneous-value-weight command to recover gradually from heavy traffic network conditions. While the system waits for a call, probes are received updating the cache. When a new probe is received, the weight calculates how much to rely upon the new probe and how much to rely upon the previous cache entry. If the weight is set to 50(%), the system enters a cache entry based on an average from the new probe and the most recent entry in the cache. Call requests use this blended entry to determine acceptance. This system allows the call fallback subsystem to keep conservative measures of network congestion.
The configured weight applies to the new probe first. If the call fallback instantaneous-value-weight command is configured with the default weight of 66(%), the new probe is given a higher value than the earlier one in calculating the average for the new cache entry.
Examples
The following example specifies a fall back value weight of 50 percent:
call fallback instantaneous-value-weight 50
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback jitter-probe num-packets
To specify the number of packets in a jitter probe used to determine network conditions, use the call fallback jitter-probe num-packets command in global configuration mode. To restore the default value, use the no form of this command.
call fallback jitter-probe num-packets number-of-packets
no call fallback jitter-probe num-packets number-of-packets
Syntax Description
number-of-packets
|
Specifies the number of packets value. The valid range is from 2 to 50.
|
Defaults
15 packets
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
A jitter probe, consisting of 2 to 50 packets, details the conditions of the network. More than one packet is used by the probe to calculate an average of delay and loss or Impairment/Calculated Planning Impairment Factor (ICPIF). After the packets return to the probe, the probe delivers the traffic information to the cache, where it is logged for call acceptance or denial. Use the call fallback threshold delay loss or call fallback threshold icpif command to set the threshold parameters.
To get a more realistic estimate of the network congestion, increase the number of packets. More probing packets give better estimates of network conditions, but also negatively affect the bandwidth for other network operations. Use fewer packets when you need to focus on bandwidth.
Examples
The following example specifies 20 packets for jitter:
call fallback jitter-probe num-packets 20
If the call fallback command has been enabled before configuring the number of jitter-probe packets, the output is as follows:
call fallback jitter-probe num-packets 20
The new num-packets will take effect only for new probes
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
call fallback jitter-probe precedence
|
Specifies the jitter-probe precedence.
|
call fallback jitter-probe priority-queue
|
Assigns a priority queue for jitter-probe transmissions.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback jitter-probe precedence
To specify the treatment of the jitter-probe transmission, use the call fallback jitter-probe precedence command in global configuration mode. To restore the default value, use the no form of this command.
call fallback jitter-probe precedence precedence-value
no call fallback jitter-probe precedence precedence-value
Syntax Description
precedence-value
|
Specifies the jitter-probe precedence. The valid range is from 0 to 6.
|
Defaults
Precedence of 2
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
In every IP packet, there is a precedence header. Precedence is used by various queueing mechanisms in different companies' routers to determine the priority of allowing traffic through the system.
Use the call fallback jitter-probe precedence command if there are different queueing mechanisms in your network. Enabling the call fallback jitter-probe precedence command sets the precedence for jitter probes to pass through your network.
If you require your probes to be sent and returned quickly, set the precedence to a low number (zero or one): the lower the precedence, the higher the priority given.
Examples
The following example specifies a jitter-probe precedence of 5, or low priority:
call fallback jitter-probe precedence 5
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
call fallback jitter-probe num-packets
|
Specifies the number of packets in a jitter probe used to determine network conditions.
|
call fallback jitter-probe priority-queue
|
Assigns a priority queue for jitter-probe transmissions.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback jitter-probe priority-queue
To assign a priority queue, use the call fallback jitter-probe priority-queue command in global configuration mode. To return to default values, use the no form of this command.
call fallback jitter-probe priority-queue
no call fallback jitter-probe priority-queue
Syntax Description
This command has no arguments or keywords.
Defaults
This command is disabled by default.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
This command is applicable only if the queueing method used is IP RTP Priority. This command is unnecessary when low latency queueing (LLQ) is used because these packets follow the priority queue path (or not) based on the LLQ classification criteria and not this command.
The command works by choosing between sending the probe on an odd or even RTP port number. The Service Assurance Agent (SAA) probe packets go out on randomly selected ports chosen from within the top end of the audio User Datagram Protocol (UDP) defined port range (16384-32767). The port pair (Real-Time Transport Protocol [RTP] & Real-Time Transport Control Protocol [RTCP] port) is selected, and, by default, SAA probes for call fallback use the RTCP port (odd) to avoid going into the priority queue, if enabled. If call fallback is configured to use the priority queue, the RTP port (even) is selected.
Examples
The following example specifies the setting of the call fallback jitter-probe priority queue command:
call fallback jitter-probe priority-queue
Warning:In order for this command to have any affect on the probes, IP priority queueing
must be set for UDP voice ports 16384-32767.
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
call fallback jitter-probe num-packets
|
Specifies the number of packets in a jitter probe used to determine network conditions.
|
call fallback jitter-probe precedence
|
Specifies the jitter-probe precedence.
|
ip rtp priority
|
Provides a strict priority queueing scheme for delay-sensitive data.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback key-chain
To specify use of Message Digest 5 (MD5) authentication for sending and receiving Service Assurance Agent (SAA) probes, use the call fallback key-chain command in global configuration mode. To disable MD5 use, use the no form of this command.
call fallback key-chain name-of-chain
no call fallback key-chain name-of-chain
Syntax Description
name-of-chain
|
Specifies the name of the chain. This line is to be alphanumeric and case-sensitive text.
|
Defaults
No call fallback key chain is defined.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
This command is used to enable Service Assurance Agent (SAA) probe authentication using MD5. If authentication is used, the keys on the sender and receiver routers must match.
Examples
The following example specifies "secret" as the fall back key chain:
call fallback key-chain secret
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
key chain
|
Enables authentication for routing protocols by identifying a group of authentication keys.
|
key-string
|
Specifies the authentication string for a key.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback map target address-list
To configure the call fallback router to keep a cache table by IP addresses of distances for several destination peers sitting behind the router, use the call fallback map target address-list command in global configuration mode. To restore the default values, use the no form of this command.
call fallback map map target ip-address address-list ip-address1 ip-address2 ... ip-address7
no call fallback map map target ip-address address-list ip-address1 ip-address2 ... ip-address7
Syntax Description
map
|
Specifies the fall back map. The valid range is from 1 to 16.
|
target ip-address
|
Specifies the target IP address.
|
ip-address1 ... ip-address7
|
Lists the IP addresses that will be kept in the cache table. The maximum number of IP addresses is seven.
|
Defaults
No call fallback maps are defined.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
Use this command when several destination peers are connected to a single access point.
Call fallback map setup allows the decongestion of traffic caused by a high volume of call probes sent across a network to query a large number of dial peers. One router/common node can keep the distances in a cache table to numerous IP addresses or destination peers in a network. When the fall back is queried for network congestion to a particular IP address (that is, the common node), the map addresses are searched to find the target IP address. If a match is determined, the probes are sent to the target address rather than to the particular IP address.
In Figure 2, the three routers (1, 2, and 3) keep the cache tables of distances for the destination peers behind them. When a call probe comes from somewhere in the IP cloud, the cache routers check their distance tables for the IP address or destination peer where the call probe is destined. This distance checking limits congestion on the networks behind these routers by directing the probe to the particular IP address and not to the entire network.
Figure 2 Call Fallback Map with IP Addresses
Examples
The following example specifies call fallback map target address-list configurations for 172.31.10.1 and 172.26.10.1:
call fallback map 1 target 172.31.10.1
address-list 172.31.10.2 172.31.10.3 172.31.10.4 172.31.10.5
172.31.10.6 172.31.10.7 172.31.10.8
call fallback map 2 target 172.26.10.1
address-list 172.26.10.2 172.26.10.3 172.26.10.4 172.26.10.5
172.26.10.6 172.26.10.7 172.26.10.8
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
call fallback map target subnet
|
Specifies the call fallback router to keep a cache table (by subnet addresses) of distances for several destination peers sitting behind the router.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback map target subnet
To configure the call fallback router to keep a cache table by subnet addresses of distances for several destination peers sitting behind the router, use the call fallback map target subnet command in global configuration mode. To restore the default values, use the no form of this command.
call fallback map map target ip-address subnet ip-network netmask
no call fallback map map target ip-address subnet ip-network netmask
Syntax Description
map
|
Specifies the fall back map. The valid range is from 1 to 16.
|
target ip-address
|
Specifies the target IP address.
|
subnet ip-network
|
Specifies the subnet IP address.
|
netmask
|
Specifies the network mask number.
|
Defaults
No call fallback maps are defined.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
Use this command when several destination peers are sitting behind one common node.
call fallback map setup allows the decongestion of traffic caused by a high volume of call probes sent across a network to query a large number of dial peers. One router/common node can keep the distances in a cache table to numerous IP addresses within a subnet (destination peers) in a network. When the fall back is queried for network congestion to a particular IP address (that is, the common node), the map addresses are searched to find the target IP address. If a match is determined, the probes are sent to the target address rather than to the particular IP address.
In Figure 3, the three routers (1, 2, and 3) keep the cache tables of distances for the destination peers behind them. When a call probe comes from somewhere in the IP cloud, the cache routers check their distance tables for the subnet address/destination peer where the call probe is destined. This distance checking limits congestion on the networks behind these routers by directing the probe to the particular subnet address and not to the entire network.
Figure 3 Call Fallback Map with Subnet Addresses
Examples
The following example specifies the call fallback map target subnet command configuration for 209.165.201.225:
call fall back map 1 209.165.201.225 subnet
209.165.201.224 255.255.255.224
call fall back map 2 209.165.202.225 subnet
209.165.202.224 255.255.255.224
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
call fallback map target address-list
|
Specifies the call fallback router to keep a cache table (by IP addresses) of distances for several destination peers sitting behind the router.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback monitor
To enable the monitoring of destinations with no provision for call fallback to alternate dial peers, use the call fallback monitor command in global configuration mode. To disable monitoring without fall back, use the no form of this command.
call fallback monitor
no call fallback monitor
Syntax Description
This command has no arguments or keywords.
Defaults
This command is disabled by default.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
The call fallback monitor command is used as a statistics collector of network conditions based on probes (detailing network traffic) and connected calls. There is no H.323 call checking and rejecting as with the call fallback active command. All call requests are granted, regardless of network traffic conditions.
Configure the call fallback threshold delay loss or call fallback threshold icpif command to set threshold parameters. The thresholds are ignored, but for statistics collecting, configuring one of the thresholds allows you to monitor cache entries for either delay and loss or Impairment/Calculated Planning Impairment Factor (ICPIF) values.
Examples
The following example shows that the call fallback monitor command has been enabled:
Related Commands
Command
|
Description
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback probe-timeout
To set the timeout for a Service Assurance Agent (SAA) probe for call fallback purposes, use the call fallback probe-timeout command in global configuration command. To restore the default value, use the no form of this command.
call fallback probe-timeout seconds
no call fallback probe-timeout seconds
Syntax Description
seconds
|
Specifies the interval in seconds. The valid range is from 1 to 2,147,483.
|
Defaults
30 seconds
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810.
|
Usage Guidelines
SAA probes collect network traffic information based on configured delay and loss or Impairment/Calculated Planning Impairment Factor (ICPIF) values and report this information to the cache for call request determinations. Use the call fallback threshold delay loss or call fallback threshold icpif command to set the threshold parameters.
When the probe timeout expires, a new probe is sent to collect network statistics. To reduce the bandwidth taken up by the probes, increase the probe-timeout interval (seconds). Probes do not have a great affect upon bandwidth unless several thousand destinations are involved. If this is the case in your network, use a longer timeout. If you need more network traffic information and bandwidth is not an issue, use a lower timeout. The default interval, 30 seconds, is a low timeout.
When the call fallback cache-timeout command is configured or expires, new probes are initiated for data collection.
Examples
The following example configures a 120-second interval:
call fallback probe-timeout 120
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback threshold delay loss
To configure the call fallback threshold to use only packet delay and loss values, use the call fallback threshold delay loss command in global configuration mode. To restore the default value, use the no form of this command.
call fallback threshold delay delay-value loss loss-value
no call fallback threshold delay delay-value loss loss-value
Syntax Description
delay-value
|
Sets the delay value. The valid range is from 1 to 2,147,483,647 milliseconds.
|
loss-value
|
Sets the loss value. The valid range is from 0 to 100 percent.
|
Defaults
There are no values set for delay and loss by default.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
For voice traffic, delays and loss of voice packets create unhappy customers. During times of heavy voice traffic, two parties in a conversation may notice a significant delay in transmission or hear only part of a conversation due to loss of voice packets.
Use the call fallback threshold delay loss command to configure parameters for voice quality. Lower values of delay and loss allow higher quality of voice. Call requests match the network information in the cache with the configured thresholds of delay and loss. If you enable call fallback active, the call fallback subsystem uses the last cache entry compared with the configured delay and loss threshold to determine whether the call is connected or denied. If you enable call fallback monitor, all calls are connected, regardless of the configured threshold or voice quality. In this case, configuring the call fallback threshold delay loss command allows you to collect network statistics for further tracking.

Note
The call fallback threshold delay loss command differs from the call fallback threshold icpif command because the call fallback threshold delay loss command uses only packet delay and loss parameters. The call fallback threshold icpif command uses packet delay and loss plus other ITU G.113 factors to gather impairment information.
Setting this command does not affect bandwidth. Available bandwidth for call requests is determined by the call fallback subsystem using probes. The number of probes on the network affects bandwidth.
Examples
The following example configures a threshold delay of 20 milliseconds and a threshold loss of 50 percent:
call fallback threshold delay 20 loss 50
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
call fallback threshold icpif
|
Specifies the ICPIF threshold.
|
show call fallback config
|
Displays the call fallback configuration.
|
call fallback threshold icpif
To configure call fallback to use the Impairment/Calculated Planning Impairment Factor (ICPIF) threshold, use the call fallback threshold icpif command in global configuration mode. To restore the default value, use the no form of this command.
call fallback threshold icpif threshold-value
no call fallback threshold icpif threshold-value
Syntax Description
threshold-value
|
Sets the threshold value. The valid range is from 0 to 34.
|
Defaults
ICPIF threshold of 5
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and on the MC3810 multiservice concentrator.
|
Usage Guidelines
During times of heavy voice traffic, two parties in a conversation may notice a significant delay in transmission or hear only part of a conversation because of loss of voice packets.
Use the call fallback threshold icpif command to configure parameters for voice quality. A low ICPIF value allows for higher quality of voice. Call requests match the network information in the cache with the configured ICPIF threshold. If you enable the call fallback active command, the call fallback subsystem uses the last cache entry compared with the configured ICPIF threshold to determine whether the call is connected or denied. If you enable the call fallback monitor command, all calls are connected regardless of the configured threshold or voice quality. In this case, configuring the call fallback threshold icpif command allows you to collect network statistics for further tracking.
A lower value of ICPIF tolerates less delay and loss (according to ICPIF calculations) of voice packets. Use lower values for higher quality of voice. Configuring a value of 34 equates to 100 percent packet loss.
The ICPIF is calculated and used according to the International Telecommunication Union (ITU) G.113 specifications.
Note
The call fallback threshold delay loss command differs from the call fallback threshold icpif command because the call fallback threshold delay loss command uses only packet delay and loss parameters. The call fallback threshold icpif command uses packet delay and loss plus other ITU G.113 factors to gather impairment information.
Setting this command does not affect bandwidth. Available bandwidth for call requests is determined by the call fallback subsystem using probes. The number of probes on the network affect bandwidth.
Examples
The following example sets the ICPIF threshold to 20:
call fallback threshold icpif 20
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
call fallback threshold delay loss
|
Specifies the call fallback threshold delay and loss values.
|
show call fallback config
|
Displays the call fallback configuration.
|
call rsvp-sync
To enable synchronization between Resource Reservation Protocol (RSVP) signaling and the voice signaling protocol, use the call rsvp-sync command in global configuration mode. To disable synchronization, use the no form of this command.
call rsvp-sync
no call rsvp-sync
Syntax Description
This command has no keywords or arguments.
Defaults
Synchronization is enabled between RSVP and the voice signaling protocol (for example, H.323).
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)XI
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and 7200 series and on the AS5300, AS5800, and MC3810.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
The call rsvp-sync command is enabled by default.
Examples
The following example enables synchronization between RSVP and the voice signaling protocol:
Related Commands
Command
|
Description
|
call rsvp-sync resv-timer
|
Sets the timer for reservation requests.
|
call start
|
Forces the H.323 Version 2 gateway to use fast connect or slow connect procedures for a dial peer.
|
debug call rsvp-sync events
|
Displays the events that occur during RSVP synchronization.
|
h323 call start
|
Forces an H.323 Version 2 gateway to use fast connect or slow connect procedures for all VoIP services.
|
ip rsvp bandwidth
|
Enables the use of RSVP on an interface.
|
show call rsvp-sync conf
|
Displays the RSVP synchronization configuration.
|
show call rsvp-sync stats
|
Displays statistics for calls that have attempted RSVP reservation.
|
call rsvp-sync resv-timer
To set the timer on the terminating VoIP gateway for completing RSVP reservation setups, use the call rsvp-sync resv-timer command in global configuration mode. To restore the default value, use the no form of this command.
call rsvp-sync resv-timer seconds
no call rsvp-sync resv-timer
Syntax Description
seconds
|
Number of seconds in which the reservation setup must be completed, in both directions. The value range is from 1 to 60 seconds.
|
Defaults
The timer default is 10 seconds.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)XI
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and 7200 series and on the AS5300, AS5800, and MC3810.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
The reservation timer is started on the terminating gateway when the session protocol receives an indication of the incoming call. This timer is not set on the originating gateway because the resource reservation is confirmed at the terminating gateway. If the reservation timer expires before the RSVP setup is complete, the outcome of the call depends on the acceptable quality of service (QoS) level configured in the dial peer; either the call proceeds without any bandwidth reservation or it is released. The timer must be set long enough to allow calls to complete but short enough to free up resources. The optimum number of seconds depends on the number of hops between the participating gateways and the delay characteristics of the network.
Examples
The following example sets the reservation timer to 30 seconds:
call rsvp-sync resv-timer 30
Related Commands
Command
|
Description
|
call rsvp-sync
|
Enables synchronization of RSVP and the H.323 voice signaling protocol.
|
debug call rsvp-sync events
|
Displays the events that occur during RSVP synchronization.
|
show call rsvp-sync conf
|
Displays the RSVP synchronization configuration.
|
show call rsvp-sync stats
|
Displays statistics for calls that have attempted RSVP reservation.
|
call start
To force the H.323 Version 2 gateway to use fast connect or slow connect procedures for a dial peer, use the call start command in voice-class configuration mode. To restore the default condition, use the no form of this command.
call start {fast | slow | system}
no call start
Syntax Description
fast
|
Gateway uses H.323 Version 2 (fast connect) procedures.
|
slow
|
Gateway uses H.323 Version 1 (slow connect) procedures.
|
system
|
Gateway defaults to the voice service configuration that is defined using the h323 call start command in voice-service configuration mode.
|
Defaults
The default is system.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.1(3)XI
|
This command was introduced on the Cisco 2600 series routers, 3600 series, and 7200 series and on the AS5300, AS5800, and MC3810.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
In Cisco IOS Release 12.1(3)XI and later, H.323 Voice over IP (VoIP) gateways by default use H.323 Version 2 (fast connect) for all calls, including those initiating RSVP. Previously, gateways used only slow connect procedures for RSVP calls. To enable Cisco IOS Release 12.1(3)XI gateways to be backward compatible with earlier releases of Cisco IOS Release 12.1 T, the call start command allows the originating gateway to initiate calls using slow connect.
The call start command is configured as part of the voice class assigned to an individual VoIP dial peer. It takes precedence over the h323 call start voice-service configuration command, which applies globally to all VoIP calls, unless the system keyword is selected. If the system keyword is used for the call start voice-class command, the gateway defaults to the voice-service configuration.
Examples
The following example selects slow connect for voice class 1000:
call-waiting
To enable call waiting, use the call-waiting command in interface configuration mode. To disable call waiting, use the no form of this command.
call-waiting
no call-waiting
Syntax Description
This command has no arguments or keywords.
Defaults
Call waiting is enabled.
Command Modes
Interface configuration
Command History
Release
|
Modification
|
12.0(3)T
|
This command was introduced on the Cisco 800 series.
|
Usage Guidelines
This command is applicable to Cisco 800 series routers.
You must specify this command when creating a dial peer. This command will not work if it is not specified within the context of a dial peer. For information on creating a dial peer, refer to the Cisco 800 Series Routers Software Configuration Guide.
Examples
The following example disables call waiting:
Related Commands
Command
|
Description
|
destination-pattern
|
Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
|
dial peer voice
|
Enters dial peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.
|
port (dial peer)
|
Enables an interface on a PA-4R-DTR port adapter to operate as a concentrator port.
|
ring
|
Sets up a distinctive ring for telephones, fax machines, or modems connected to a Cisco 800 series router.
|
show dial peer voice
|
Displays configuration information and call statistics for dial peers.
|
called-number (dial-peer)
To enable an incoming Voice over Frame Relay (VoFR) call leg to get bridged to the correct plain old telephone service (POTS) call leg when a static FRF.11 trunk connection is used, use the called-number command in dial peer configuration mode. To disable a static trunk connection, use the no form of this command.
called-number string
no called-number
Syntax Description
string
|
A string of digits, including wildcards, that specifies the telephone number of the voice port dial peer.
|
Defaults
This command is disabled.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
12.0(4)T
|
This command was introduced on the Cisco 2600 and 3600 series.
|
Usage Guidelines
This command applies to the Cisco 2600 and 3600 series routers only. It is ignored on the Cisco MC3810 multiservice concentrator and on the Cisco 7200 series routers.
The called-number command is used only when the dial peer type is VoFR and you are using the frf11-trunk (FRF.11) session protocol. It is ignored at all times on the Cisco MC3810 multiservice concentrator and on all other platforms when using the Cisco-switched session protocol.
Because FRF.11 does not provide any end-to-end messaging to manage a trunk, the called-number command is necessary to allow the router to establish an incoming trunk connection. The E.164 number is used to find a matching dial peer during call setup.
Examples
The following example shows how to configure a Cisco 2600 series routers or 3600 series router for a static FRF.11 trunk connection to a specific telephone number (555-2150), beginning in global configuration mode:
connection trunk 55Router0
destination pattern 5552150
session protocol frf11-trunk
destination pattern 55Router0
Related Commands
Command
|
Description
|
codec (dial peer)
|
Specifies the voice coder rate of speech for a VoFR dial peer.
|
connection
|
Specifies a connection mode for a voice port.
|
destination-pattern
|
Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
|
dtmf-relay (VoFR)
|
Enables the generation of FRF.11 Annex A frames for a dial peer.
|
fax-rate
|
Establishes the rate at which a fax will be sent to the specified dial peer.
|
preference
|
Indicates the preferred order of a dial peer within a rotary hunt group.
|
session protocol
|
Establishes a session protocol for calls between the local and remote routers via the packet network.
|
session target
|
Specifies a network-specific address for a specified dial peer or destination gatekeeper.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
vad (dial peer)
|
Enables voice-activated dialing (VAD) for the calls using a particular dial peer.
|
caller-id
To enable caller ID, use the caller-id command in dial peer configuration mode. To disable caller ID, use the no form of the command.
caller-id
no caller-id
Syntax Description
This command contains no arguments or keywords.
Defaults
Caller ID is disabled.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
12.1.(2)XF
|
This command was introduced on the Cisco 800 series routers.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
This command is available on Cisco 800 series routers that have plain old telephone service (POTS) ports. The command is effective only if you subscribe to caller ID service. If you enable caller ID on a router without subscribing to the caller ID service, caller ID information does not appear on the telephone display.
The configuration of caller ID must match the device connected to the POTS port. That is, if a telephone supports the caller ID feature, use the command caller-id to enable the feature. If the telephone does not support the caller ID feature, use the command default or disable the caller ID feature. Odd ringing behavior might occur if the caller ID feature is disabled when it is a supported telephone feature or enabled when it is not a supported telephone feature.
Note
Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.
Examples
The following example enables a router to use the caller ID feature:
Related Commands
Command
|
Description
|
block-caller
|
Configures call blocking on caller ID.
|
debug pots csm csm
|
Activates events from which an application can determine and display the status and progress of calls to and from POTS ports.
|
isdn i-number
|
Configures several terminal devices to use one subscriber line.
|
pots call-waiting
|
Enables local call waiting on a router.
|
registered-caller ring
|
Configures the Nariwake service-registered caller ring cadence.
|
caller-id alerting dsp-pre-alloc
To statically allocate a digital signal processor (DSP) resource for receiving caller ID information for on-hook (Type 1) Caller ID at a receiving Foreign Exchange Office (FXO) voice port, use the caller-id alerting dsp-pre-alloc command in voice-port configuration mode. To disable the command's effect, use the no form of this command.
caller-id alerting dsp-pre-alloc
no caller-id alerting dsp-pre-alloc
Syntax Description
This command contains no keywords or arguments.
Defaults
No pre-allocation of DSP resources
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.1(2)XH
|
This command was implemented for the Cisco MC3810 multiservice concentrator and for Cisco 2600 and 3600 series routers.
|
12.1(3)T
|
This command was integrated into Cisco IOS Release 12.1(3)T.
|
Usage Guidelines
The caller-id alerting dsp-pre-alloc command may be required on an FXO port if the central office uses line polarity reversal to signal the start of Caller-ID information transmission. Pre-allocating a DSP allows the DSP to listen for Caller-ID information continuously without requiring an alerting signal from the CO.
This command is the FXO counterpart to the caller-id alerting line-reversal command, which is applied to the Foreign Exchange Station (sending) end of the Caller-ID call.
This command applies to the Cisco MC3810 multiservice concentrator and to Cisco 2600 and 3600 series routers.
Note
Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.
Examples
The following example configures a voice port on a Cisco 2600 or 3600 router where Caller-ID information is received:
caller-id alerting line-reversal
caller-id alerting dsp-pre-alloc
The following example configures a voice port on a Cisco MC3810 multiservice concentrator where Caller-ID information is received:
caller-id alerting line-reversal
caller-id alerting dsp-pre-alloc
Related Commands
caller-id alerting line-reversal
To set the line-reversal alerting method for Caller-ID information for on-hook (Type 1) Caller ID at a sending Foreign Exchange Station (FXS) voice port, use the caller-id alerting line-reversal command in voice-port configuration mode. To disable the command's effect, use the no form of this command.
caller-id alerting line-reversal
no caller-id alerting line-reversal
Syntax Description
This command has no keywords or arguments.
Defaults
No line-reversal alert
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.1(2)XH
|
This command was implemented for the Cisco MC3810 multiservice concentrator and for Cisco 2600 and 3600 series routers.
|
12.1(3)T
|
This command was integrated into Cisco IOS Release 12.1(3)T.
|
Usage Guidelines
This command is only required when the telephone device attached to an FXS port requires the line-reversal method to signal the start of a Caller-ID transmission. Use it on FXS voice ports that send Caller-ID information.
This command is the FXS counterpart to the caller-id alerting dsp-pre-alloc command, which is applied to the FXO (receiving) end of the Caller-ID call with the line-reversal alerting method.
This command applies to the Cisco MC3810 multiservice concentrator and to Cisco 2600 and 3600 series routers.
Note
Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.
Examples
The following example configures a voice port on a Cisco 2600 or 3600 series router from which Caller-ID information is sent:
station number 4085551111
caller-id alerting line-reversal
caller-id alerting dsp-pre-alloc
The following example configures a voice port on a Cisco MC3810 multiservice concentrator from which Caller-ID information is sent:
station number 4085551111
caller-id alerting line-reversal
caller-id alerting dsp-pre-alloc
Related Commands
Command
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Description
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caller-id alerting dsp-pre-alloc
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At the receiving end of a line-reversal alerting Caller-ID call, pre-allocates DSPs for caller ID calls.
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caller-id alerting pre-ring
To set a 250-millisecond pre-ring alerting method for caller ID information for on-hook (Type 1) Caller ID at a sending Foreign Exchange Station (FXS) voice port, use the caller-id alerting pre-ring command in voice-port configuration mode. To disable the command, use the no form of this command.
caller-id alerting pre-ring
no caller-id alerting pre-ring
Syntax Description
This command has no keywords or arguments.
Defaults
No pre-ring alert
Command Modes
Voice-port configuration
Command History
Release
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Modification
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12.1(2)XH
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This command was implemented for the Cisco MC3810 multiservice concentrator and for Cisco 2600 and 3600 series routers.
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12.1(3)T
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This command was integrated into Cisco IOS Release 12.1(3)T.
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Usage Guidelines
This command is required only when the telephone device attached to an FXS port requires the pre-ring (immediate ring) method to signal the start of caller ID transmission. Use it on FXS voice ports that send caller ID information. This command allows the FXS port to send a short pre-ring preceding the normal ring cadence. On an FXO port, an incoming pre-ring (immediate ring) is simply counted as a normal ring using the caller-id alerting ring command.
This command applies to the Cisco MC3810 multiservice concentrator and to Cisco 2600 and 3600 series routers.
Note
Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.
Examples
The following example configures a voice port on a Cisco 2600 or 3600 series router from which caller ID information is sent: