Table Of Contents
Cisco IOS Voice, Video, and Fax Commands:
R Through Sh
register e164
registered-caller ring
req-qos
reset
resource threshold
response-timeout
retry-delay
retry-limit
retry (SIP user-agent)
ring
ring cadence
ring frequency
ring number
roaming (dial-peer)
roaming (settlement)
rtsp client session history duration
rtsp client session history records
rule
security
sequence-numbers
server (RLM)
server registration-port
server trigger
session
session protocol
session protocol (Voice over Frame Relay)
session protocol aal2
session protocol multicast
session target (VoATM)
session target (VoFR)
session target (VoIP)
session transport
set
settle-call
settlement
settlement roam-pattern
sgcp
sgcp call-agent
sgcp graceful-shutdown
sgcp max-waiting-delay
sgcp modem passthru
sgcp quarantine-buffer disable
sgcp request retries
sgcp request timeout
sgcp restart
sgcp retransmit timer
sgcp timer
sgcp tse payload
show aal2 profile
show atm video-voice address
show backhaul-session-manager group
show backhaul-session-manager session
show backhaul-session-manager set
show call active
show call application voice
show call fallback cache
show call fallback config
show call fallback stats
show call history
show call history video record
show call history voice record
show call resource voice stats
show call resource voice threshold
show call rsvp-sync conf
show call rsvp-sync stats
show cdapi
show ces clock-select
show connect
show controllers rs366
show controllers timeslots
show controllers voice
show csm
show dial-peer video
show dial-peer voice
show dialplan incall number
show dialplan number
show frame-relay vofr
show gatekeeper calls
show gatekeeper endpoints
show gatekeeper gw-type-prefix
show gatekeeper servers
show gatekeeper status
show gatekeeper zone prefix
show gatekeeper zone status
show gateway
show interface dspfarm
show mgcp
show mgcp connection
show mgcp endpoint
show mgcp statistics
show num-exp
show pots csm
show pots status
show proxy h323 calls
show proxy h323 detail-call
show proxy h323 status
show rawmsg
show rlm group statistics
show rlm group status
show rlm group timer
show rtsp client session
show rudpv0 failures
show rudpv0 statistics
show rudpv1
show settlement
show sgcp connection
show sgcp endpoint
show sgcp statistics
show sip-ua
show ss7 mtp2 ccb
show ss7 mtp2 state
show ss7 mtp2 stats
show ss7 mtp2 timer
show ss7 mtp2 variant
show ss7 sm session
show ss7 sm set
show ss7 sm stats
show translation-rule
show vfc
show vfc cap-list
show vfc default-file
show vfc directory
show vfc version
show video call summary
show voice busyout
show voice call
show voice dsp
show voice permanent-call
show voice port
show voice trunk-conditioning signaling
show voice trunk-conditioning supervisory
show vrm active_calls
show vrm vdevices
shut
shutdown (dial-peer)
shutdown (DS1)
shutdown (gatekeeper)
shutdown (RLM)
shutdown (settlement)
shutdown (voice-port)
Cisco IOS Voice, Video, and Fax Commands:
R Through Sh
This chapter presents the commands to configure and maintain Cisco IOS voice, video, and fax applications. The commands are presented in alphabetical order beginning with R. 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.
register e164
To configure a gateway to register or deregister (remove the registration for) a fully qualified plain old telephone service (POTS) dial-peer E.164 address with a gatekeeper, use the register e164 command in dial-peer configuration mode. To deregister an E.164 address, use the no form of this command.
register e164
no register e164
Syntax Description
This command has no keywords or arguments.
Defaults
No E.164 addresses are registered until you enter this command.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use this command to register the E.164 address of an analog telephone line attached to a Foreign Exchange Station (FXS) port on a router. The gateway automatically registers fully qualified E164 addresses. Use the no register e164 command to deregister an address. Use the register e164 command to register a deregistered address.
Before you automatically or manually register an E.164 address with a gatekeeper, you must create a dial peer (using the dial-peer command), assign an FXS port to the peer (using the port command), and assign an E.164 address (using the destination-pattern command). The E.164 address must be a fully qualified address. For example, +5551212, 5551212, and 4085551212 are fully qualified addresses; 408555.... is not a fully qualified address. E.164 addresses are registered only for active interfaces—those that are not shut down. If an FXS port or its interface is shut down, the corresponding E.164 address is deregistered.
Tips
You can use the show gateway command to find out if the gateway is connected to a gatekeeper and if a fully qualified E.164 address is assigned to the gateway. Use the zone-prefix command at the gatekeeper to define prefix patterns, such as 408555...., that apply to one or more gateways.
Examples
The following command sequence places the gateway in dial-peer configuration mode, assigns an E.164 address to the interface, and registers that address with the gatekeeper:
destination-pattern 5551212
The following commands deregister an address with the gatekeeper:
The following example shows that you must have a connection to a gatekeeper and define a unique E.164 address before you can register an address:
ERROR-register-e164:Dial-peer destination-pattern is not a full E.164 number
ERROR-register-e164:No gatekeeper
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
|
Enters dial-peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.
|
port
|
Enables an interface on a PA-4R-DTR to operate as a concentrator port.
|
show gateway
|
Displays the current gateway status.
|
zone prefix
|
Configures the gatekeeper with knowledge of its own prefix and the prefix of any remote zone.
|
registered-caller ring
To configure the Nariwake service registered caller ring cadence, use the registered-caller ring command in dial-peer configuration mode.
registered-caller ring cadence
Syntax Description
cadence
|
A value of 0, 1, or 2. The default ring cadence for registered callers is 1 and for unregistered callers is 0. The on and off periods of ring 0 (normal ringing signals) and ring 1 (ringing signals for the Nariwake service) are defined in the NTT user manual.
|
Defaults
The default Nariwake service registered caller ring cadence is ring 1.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1.(2)XF
|
The command registered-caller ring was introduced on the Cisco 800 series routers.
|
Usage Guidelines
If your ISDN line is provisioned for the I Number or dial-in services, you must also configure a dial peer by using the destination-pattern not-provided command. Either port 1 or port 2 can be configured under this dial peer. The router then forwards the incoming call to voice port 1. (See the "Examples" section below.
If more than one dial peer is configured with the destination-pattern not-provided command, the router uses the first configured dial peer for the incoming calls. To display the Nariwake ring cadence setting, use the show run command.
Examples
The following example sets the ring cadence for registered callers to 2.
req-qos
To specify the desired quality of service to be used in reaching a specified dial peer, use the req-qos command in dial-peer configuration mode. To restore the default value for this command, use the no form of this command.
req-qos {best-effort | controlled-load | guaranteed-delay}
no req-qos
Syntax Description
best-effort
|
Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation.
|
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 assure 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.
|
Usage Guidelines
This command is applicable only to VoIP dial peers.
Use the req-qos command to request a specific quality of service to be used in reaching a dial peer. Like acc-qos, when you issue this command, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation Protocol (RSVP) to request quality of service guarantees from the network.
Examples
The following example configures guaranteed-delay as the desired (requested) quality of service to a dial peer:
Related Commands
Command
|
Description
|
acc-qos
|
Defines the acceptable QoS for any inbound and outbound call on a VoIP dial peer.
|
reset
To reset a set of digital signal processors (DSPs), use the reset command in global configuration mode.
reset number
Syntax Description
number
|
Specifies the number of DSPs to be reset. The number of DSPs ranges from 0 to 30.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
12.0(5)XE
|
This command was introduced on the Cisco 7200 series routers.
|
12.0(7)T
|
This command was integrated into the Cisco IOS Release 12.0(7)T.
|
Examples
The following example displays the reset command configuration for DSP 1:
01:24:54:%DSPRM-5-UPDOWN: DSP 1 in slot 1, changed state to up
resource threshold
To configure a gateway to report H.323 resource availability to the its gatekeeper, use the resource threshold command in gateway configuration mode. To disable gateway resource-level reporting, use the no form of this command.
resource threshold [all] [high percentage-value] [low percentage-value]
no resource threshold
Syntax Description
all
|
(Optional) Applies the high- and low- parameter settings to all monitored H.323 resources. This is the default condition.
|
high percentage-value
|
(Optional) A resource utilization level that triggers a Resource Availability Indicator (RAI) message indicating that H.323 resource use is high. Enter a number between 1 and 100 that represents the high-resource utilization percentage. A value of 100 specifies high-resource usage when any H.323 resource is unavailable. The default is 90 percent.
|
low percentage-value
|
(Optional) Resource utilization level that triggers an RAI message indicating that H.323 resource usage has dropped below the high-usage level. Enter a number between 1 and 100 that represents the acceptable resource utilization percentage. After the gateway sends a high-utilization message, it waits to send the resource recovery message until the resource use drops below the value defined by the low parameter. The default is 90 percent.
|
Defaults
Reports low resources when 90 percent of resources are in use, and reports resource availability when resource use drops below 90 percent.
Command Modes
Gateway configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
The resource threshold command defines the resource load levels that trigger Resource Availability Indicator (RAI) messages. To view the monitored resources, enter the show gateway command.
The monitored H.323 resources include digital signal processor (DSP) channels and DS0s. Use the show call resource voice stats command to see the total amount of resources available for H.323 calls.
Note
The DS0 resources that are monitored for H.323 calls are limited to the ones that are associated with a voice POTS dial peer.
See the dial-peer configuration commands for details on how to associate a dial peer with a PRI or CAS group.
When any monitored H.323 resources exceed the threshold level defined by the high parameter, the gateway sends an RAI message to the gatekeeper with the AlmostOutOfResources field flagged. This message reports high resource usage.
When all gateway H.323 resources drop below the level defined by the low parameter, the gateway sends the RAI message to the gatekeeper with the AlmostOutOfResources field cleared.
When a gatekeeper can choose between multiple gateways for call completion, the gatekeeper uses internal priority settings and gateway resource statistics to determine which gateway to use. When all other factors are equal, a gateway that has available resources will be chosen over a gateway that has reported limited resources.
Examples
The following command defines the H.323 resource limits for a gateway:
resource threshold high 70 low 60
Related Commands
Command
|
Description
|
show call resource voice stats
|
Displays resource statistics for an H.323 gateway.
|
show call resource voice threshold
|
Displays the threshold configuration settings and status for an H.323 gateway.
|
show gateway
|
Displays the current gateway status.
|
response-timeout
To configure the maximum time to wait for a response from a server, use the response-timeout command in settlement configuration mode. To restore the default value of this command, use the no form of this command.
response-timeout number
no response-timeout number
Syntax Description
number
|
Response waiting time in seconds.
|
Defaults
The default response timeout is one (1) second.
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and the Cisco AS5300 universal access server.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
If no response is received within the response-timeout time limit, the current connection ends, and the router attempts to contact the next service point.
Examples
The following example illustrates a response-timeout set to 1 second.
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the time for which a connection is maintained after completion of a communication exchange.
|
customer-id
|
Identifies a carrier or ISP with a settlement provider.
|
device-id
|
Specifies a gateway associated with a settlement provider.
|
encryption
|
Sets the encryption method to be negotiated with the provider.
|
max-connection
|
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
|
retry-delay
|
Sets the time between attempts to connect with the settlement provider.
|
retry-limit
|
Sets the maximum number of attempts to connect to the provider.
|
session-timeout
|
Sets the interval for closing the connection when there is no input or output traffic.
|
settlement
|
Enters settlement mode and specifies the attributes specific to a settlement provider.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown/no shutdown
|
Deactivates the settlement provider/activates the settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
url
|
Specifies the Internet service provider address.
|
retry-delay
To set the time between attempts to connect with the settlement provider, use the retry-delay command in settlement configuration mode. To restore the default value, use the no form of this command.
retry-delay number
no retry-delay
Syntax Description
number
|
Length of time (in seconds) between attempts to connect with the settlement provider. The valid range for retry delay is from 1 to 600 seconds.
|
Defaults
The default retry delay is two seconds.
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on the Cisco 2600 and 3600 series routers and the Cisco AS5300 universal access server.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
After exhausting all service points for the provider, the router is delayed for the specified length of time before resuming connection attempts.
Examples
The following example sets a retry value of 15 seconds:
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the time for which a connection is maintained after completion of a communication exchange.
|
customer-id
|
Identifies a carrier or ISP with a settlement provider.
|
device-id
|
Specifies a gateway associated with a settlement provider.
|
encryption
|
Sets the encryption method to be negotiated with the provider.
|
max-connection
|
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
|
response-timeout
|
Configures the maximum time to wait for a response from a server.
|
retry-limit
|
Sets the maximum number of attempts to connect to the provider.
|
session-timeout
|
Sets the interval for closing the connection when there is no input or output traffic.
|
settlement
|
Enters settlement configuration mode and specifies the attributes specific to a settlement provider.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown/no shutdown
|
Deactivates the settlement provider/activates the settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
retry-limit
To set the maximum number of attempts to connect to the provider, use the retry-limit command in settlement configuration mode. To restore the default value, use the no form of this command.
retry-limit number
no retry-limit number
Syntax Description
number
|
Maximum number of connection attempts in addition to the first attempt.
|
Defaults
The default retry limit is one (1) retry.
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
If no connection is established after the configured retries, the router ceases connection attempts. The retry limit number does not count the initial connection attempt. A retry limit of one (default) results in a total of two connection attempts to every service point.
Examples
The following example sets the number of retries to 1:
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the time for which a connection is maintained after a communication exchange is complete.
|
customer-id
|
Identifies a carrier or ISP with a settlement provider.
|
device-id
|
Specifies a gateway associated with a settlement provider.
|
encryption
|
Sets the encryption method to be negotiated with the provider.
|
max-connection
|
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
|
response-timeout
|
Configures the maximum time to wait for a response from a server.
|
retry-delay
|
Sets the time between attempts to connect with the settlement provider.
|
session-timeout
|
Sets the length of interval for closing the connection when there is no input or output traffic.
|
settlement
|
Enters settlement mode and specifies the attributes specific to a settlement provider.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown
|
Brings up the settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
retry (SIP user-agent)
To configure the number of retry attempts for Session Initiation Protocol (SIP) messages, use the retry command in SIP user-agent configuration mode. To reset this command to the default value, use the no form of this command.
retry {invite number | response number | bye number | cancel number}
no retry {invite number | response number | bye number | cancel number}
Syntax Description
invite number
|
Number of INVITE retries: 1 through 10 are valid inputs; default = 6.
|
response number
|
Number of RESPONSE retries: 1 through 10 are valid inputs; default = 6.
|
bye number
|
Number of BYE retries: 1 through 10 are valid inputs; default = 10.
|
cancel number
|
Number of CANCEL retries: 1 through 10 are valid inputs; default = 10.
|
Defaults
invite: 6
response: 6
bye: 10
cancel: 10
Command Modes
SIP user-agent configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
Usage Guidelines
To reset this command to the default value, you can also use the default command.
Examples
In the following example, the number of invite retries has been set to 5.
Related Commands
Command
|
Description
|
sip-ua
|
Enables the sip-ua configuration commands, with which you configure the user agent.
|
ring
To set up a distinctive ring for your connected telephones, fax machines, or modems, use the ring command in interface configuration mode. To disable the specified distinctive ring, use the no form of this command.
ring cadence-number
no ring cadence-number
Syntax Description
cadence-number
|
Number from 0 through 2:
• Type 0 is a primary ringing cadence—default ringing cadence for the country your router is in.
• Type 1 is a distinctive ring—0.8 seconds on, 0.4 seconds off, 0.8 seconds on, 0.4 seconds off.
• Type 2 is a distinctive ring—0.4 seconds on, 0.2 seconds off, 0.4 seconds on, 0.2 seconds off, 0.8 seconds on, 4 seconds off.
|
Defaults
The default is 0.
Command Modes
Interface configuration
Command History
Release
|
Modification
|
12.0(3)T
|
This command was introduced on the Cisco 800 series router.
|
Usage Guidelines
This command applies to Cisco 800 series routers.
You can 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 specifies the type 1 distinctive ring:
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.
|
no call-waiting
|
Disables call waiting.
|
port (dial-peer)
|
Enables an interface on a PA-4R-DTR port adapter to operate as a concentrator port.
|
pots distinctive-ring-guard-time
|
Specifies a delay in which a telephone port can be rung after a previous call is disconnected (for Cisco 800 series routers).
|
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.
|
ring cadence
To specify the ring cadence for a Foreign Exchange Station (FXS) voice port, use the ring cadence command in voice-port configuration mode. To restore the default value, use the no form of this command.
ring cadence {pattern-number | define pulse interval}
no ring cadence
Syntax Description
pattern-number
|
Predefined ring cadence patterns. Each pattern specifies a ring-pulse time and a ring-interval time.
• pattern01—2 seconds on, 4 seconds off
• pattern02—1 second on, 4 seconds off
• pattern03—1.5 seconds on, 3.5 seconds off
• pattern04—1 second on, 2 seconds off
• pattern05—1 second on, 5 seconds off
• pattern06—1 second on, 3 seconds off
• pattern07—0.8 second on, 3.2 seconds off
• pattern08—1.5 seconds on, 3 seconds off
• pattern09—1.2 seconds on, 3.7 seconds off
• pattern09—1.2 seconds on, 4.7 seconds off
• pattern11—0.4 second on, 0.2 second off, 0.4 second on, 2 seconds off
• pattern12—0.4 second on, 0.2 second off, 0.4 second on, 2.6 seconds off
|
define
|
User-definable ring cadence pattern. Each number pair specifies one ring-pulse time and one ring-interval time. You must enter numbers in pairs, and you can enter from 1 to 6 pairs. The second number in the last pair that you enter specifies the interval between rings.
|
pulse
|
A number (1 or 2 digits) specifying ring pulse (on) time in hundreds of milliseconds.
The range is from 1 to 50, for pulses of 100 ms to 5000 ms. For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.
|
interval
|
A number (1 or 2 digits) specifying ring interval (off) time in hundreds of milliseconds.
The range is from 1 to 50, for pulses of 100 to 5000 ms. For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.
|
Defaults
Ring cadence defaults to the pattern you specify with the cptone command.
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, and the patternXX keyword was introduced.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
Usage Guidelines
The patternXX keyword provides preset ring cadence patterns for use on any platform. The define keyword allows you to create a custom ring cadence. On the Cisco 2600 and 3600 series routers, only one or two pairs of digits can be entered under the define keyword.
Examples
The following example configures the ring cadence for 1 second on and 4 seconds off on voice port 1/1 on a Cisco MC3810 multiservice concentrator:
The following example configures the ring cadence for 1 second on, 1 second off, 1 second on, and
5 seconds off on voice port 1/2 on a Cisco MC3810 multiservice concentrator:
ring cadence define 10 10 10 50
The following example configures the ring cadence for 1 second on and 2 seconds off on voice port 1/0/0 on a Cisco 2600 or 3600 series router:
Related Commands
Command
|
Description
|
ring frequency
|
Specifies the ring frequency for a specified FXS voice port.
|
cptone
|
Specifies the default tone, ring, and cadence settings according to country.
|
ring frequency
To specify the ring frequency for a specified Foreign Exchange Station (FXS) voice port, use the ring frequency command in voice-port configuration mode. To restore the default value, use the no form of this command.
ring frequency number
no ring frequency number
Syntax Description
number
|
Ring frequency (hertz) used in the FXS interface. Valid entries on the Cisco 3600 series are 25 and 50. Valid entries on the Cisco MC3810 multiservice concentrator are 20 and 30.
|
Defaults
25 Hz on the Cisco 3600 series routers and 20 Hz on the Cisco MC3810 multiservice concentrators.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
Use the ring frequency command to select a specific ring frequency for an FXS voice port. Use the no form of this command to reset the default value. The ring frequency you select must match the connected equipment. If set incorrectly, the attached phone might not ring or might buzz. In addition, the ring frequency is usually country-dependent. You should take into account the appropriate ring frequency for your area before configuring this command.
This command does not affect ringback, which is the ringing a user hears when placing a remote call.
Examples
The following example configures the ring frequency on the Cisco 3600 series for 25 Hz:
The following example configures the ring frequency on the Cisco MC3810 multiservice concentrator for 20 Hz:
Related Commands
Command
|
Description
|
ring cadence
|
Specifies the ring cadence for an FXS voice port on the Cisco MC3810 multiservice concentrator.
|
ring number
|
Specifies the number of rings for a specified FXO voice port.
|
ring number
To specify the number of rings for a specified Foreign Exchange Office (FXO) voice port, use the ring number command in voice-port configuration mode. To restore the default value, use the no form of this command.
ring number number
no ring number number
Syntax Description
number
|
Number of rings detected before answering the call. Valid entries are numbers from 1 to 10. The default is 1.
|
Defaults
One ring
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series router.
|
Usage Guidelines
Use the ring number command to set the maximum number of rings to be detected before answering a call over an FXO voice port. Use the no form of this command to reset the default value, which is one ring.
Normally, this command should be set to the default so that incoming calls are answered quickly. If you have other equipment available on the line to answer incoming calls, you might want to set the value higher to give the equipment sufficient time to respond. In that case, the FXO interface would answer if the equipment online did not answer the incoming call in the configured number of rings.
This command is not applicable to Foreign Exchange Station (FXS) or E&M interfaces because they do not receive ringing on incoming calls.
Examples
The following example on the Cisco 3600 series sets five rings as the maximum number of rings to be detected before closing a connection over this voice port:
The following example on the Cisco MC3810 multiservice concentrator sets five rings as the maximum number of rings to be detected before closing a connection over this voice port:
Related Commands
Command
|
Description
|
ring frequency
|
Specifies the ring frequency for a specified FXS voice port.
|
roaming (dial-peer)
To enable the roaming capability for the dial peer, use the roaming command in dial-peer configuration mode. To disable the roaming capability, use the no form of this command.
roaming
no roaming
Syntax Description
This command has no arguments or keywords.
Defaults
No roaming
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server`.
|
Usage Guidelines
Enable the roaming capability of a dial peer if that dial peer can terminate roaming calls. If a dial peer is dedicated to local calls only, disable the roaming capability.
The roaming dial peer must work with a roaming service provider. If the dial peer allows a roaming user to go through, and the service provider is not roaming-enabled, the call fails.
Examples
The following example enables the roaming capability for the dial peer:
Related Commands
Command
|
Description
|
roaming (settlement)
|
Enables the roaming capability for a settlement provider.
|
settle-call
|
Limits the dial peer to using only the specific clearinghouse identified by the specified provider-number.
|
settlement roam-pattern
|
Configures a pattern to match against when determining roaming.
|
roaming (settlement)
To enable the roaming capability for a settlement provider, use the roaming command in settlement configuration mode. To disable the roaming capability, use the no form of this command.
roaming
no roaming
Syntax Description
This command has no arguments or keywords.
Defaults
No roaming
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
Usage Guidelines
Enable roaming capability of a settlement provider if that provider can authenticate a roaming user and route roaming calls.
A roaming call is successful only if both the settlement provider and the outbound dial peer for that call are roaming-enabled.
Examples
The following example enables the roaming capability for the settlement provider:
Related Commands
Command
|
Description
|
roaming (dial-peer mode)
|
Enables the roaming capability for the dial peer.
|
settle-call
|
Limits the dial peer to using only the specific clearinghouse identified by the specified provider-number.
|
settlement roam-pattern
|
Configures a pattern to match against when determining roaming.
|
rtsp client session history duration
To specify how long to keep Real Time Streaming Protocol (RTSP) session history records in memory, use the rtsp client session history duration command in global configuration mode. To set the value to the default, use the no form of this command.
rtsp client session history duration number
no rtsp client session history duration number
Syntax Description
number
|
Specifies how long, in minutes, to keep the record.
|
Defaults
10 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 sets the RTSP session history to 500 minutes:
rtsp client session history duration 500
Related Commands
Command
|
Description
|
call application voice load
|
Allows reload of an aplication that was loaded via the MGCP scripting package.
|
rtsp client session history records
|
Specifies the number of RTSP client session history records kept during the session.
|
show call application voice
|
Displays all TCL or MGCP scripts that are loaded.
|
show rtsp client session
|
Displays cumulative information about the RTSP session records.
|
rtsp client session history records
To configure the number of records to keep in the RTSP client session history, use the rtsp client session history records command in global configuration mode. To set the value to the default, use the no form of this command.
rtsp client session history records number
no rtsp client session history records number
Syntax Description
number
|
Specifies the number of records to retain in a session history. Values range from 1 to 100000.
|
Defaults
50 records
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 sets the RTSP client history to 500 records:
rtsp client session history records 500
Related Commands
Command
|
Description
|
call application voice load
|
Allows reload of an aplication that was loaded via the MGCP scripting package.
|
rtsp client session history duration
|
Specifies the how long the RTSP is kept during the session.
|
show call application voice
|
Displays all TCL or MGCP scripts that are loaded.
|
rule
To apply a translation rule to a calling party number or a called party number for both incoming and outgoing calls, use the rule command in translation-rule configuration mode. To remove the translation rule, use the no form of this command.
rule name-tag input-matched-pattern substituted-pattern [match-type substituted-type]
no rule name-tag input-matched-pattern substituted-pattern [match-type substituted-type]
Syntax Description
name-tag
|
The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. Range is from 1 through 2147483647.
|
input-matched-pattern
|
The input string of digits for which pattern matching is performed.
|
substituted-pattern
|
The replacement digit string that results after pattern matching is performed. Regular expressions are used to carry out this process.
|
match-type
|
(Optional) The choices for this field are international, national, subscriber, abbreviated, unknown, and any, as defined by the International Telecommunication Union Telecommunication Standardization Sector (ITU-T) Q.931 specification. If you enter the match-type value, then you must also enter the substituted-type value.
|
substituted-type
|
(Optional) The choices for this field are international, national, subscriber, abbreviated and unknown, as defined by the ITU Q.931 specification.
|

Note
In the syntax description above, the square brackets indicate optional values. When using this command, do not include these square brackets as part of the syntax. They are not valid parameters in the rule command. The square brackets can only be used in actual syntax for such commands as the destination-pattern and incoming called-number commands, where the syntax specifically allows this delimiter.
Defaults
No default behavior or values.
Command Modes
Translation-rule configuration
Command History
Release
|
Modification
|
12.0(7)XR1
|
This command was introduced for Voice over IP on the Cisco AS5300 universal access server.
|
12.0(7)XKs
|
This command was first supported for Voice over IP on the following platforms: Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators.
|
12.1(1)T
|
This command was first supported on the T train for Voice over IP on the following platforms: Cisco 1750 routers, Cisco 2600 and 3600 series routers, Cisco AS5300 universal access servers, Cisco 7200 series routers, and Cisco 7500 series routers.
|
12.1(2)T
|
This command was first supported for Voice over IP on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
When configuring your dial peers, you are provided with an option called the translation rule. This option applies a translation rule to a calling party number (Automatic Number Identification [ANI]) or a called party number (Dial Number Information Service [DNIS]) for both incoming and outgoing calls within Cisco H.323 voice-enabled gateways. Also, the rule allows translation of the type of number.
Examples
The following example applies a translation rule. If a called number starts with 5552205 or 52205, then translation rule 21 will use the rule command to forward the number to 14085552205 instead.
rule 1 555.% 1408555 subscriber international
rule 2 7.% 1408555 abbreviated international
In the next example, if a called number is either 14085552205 or 014085552205, then after the execution of the translation rule 345, the forwarding digits will be 52205. If the match type is configured and the type is not "unknown," then the dial peer matching will be required to match input string numbering type.
rule 1 .%555.% 7 any abbreviated
Related Commands
Command
|
Description
|
numbering-type
|
Specifies number type for the VoIP or POTS dial peer.
|
test translation-rule
|
Tests the execution of the translation rules on a specific name tag.
|
translate
|
Applies a translation rule to a calling party number or a called party number for incoming calls
|
translate-outgoing
|
Applies a translation rule to a calling party number or a called party number for outgoing calls
|
translation-rule
|
Creates a translation name and enters translation-rule configuration mode.
|
voip-incoming translation-rule
|
Captures calls that originate from H.323-compatible clients.
|
security
To enable authentication and authorization on a gatekeeper, use the security command in gatekeeper configuration mode. To disable security, use the no form of this command.
security {any | h323-id | e164} {password default password | password separator character}
no security {any | h323-id | e164} {password default password | password separator character}
Syntax Description
any
|
Uses the first alias of an incoming registration, admission, and status (RAS) protocol registration, regardless of its type, as the means of identifying the user to RADIUS/TACACS+.
|
h323-id
|
Uses the first H.323 ID type alias as the means of identifying the user to RADIUS/TACACS+.
|
e164
|
Uses the first E.164 address type alias as the means of identifying the user to RADIUS/TACACS+.
|
password default password
|
Specifies the default password that the gatekeeper associates with endpoints when authenticating them with an authentication server. The password must be identical to the password on the authentication server.
|
password separator character
|
Specifies the character that endpoints use to separate the H.323-ID from the piggybacked password in the registration. Specifying this character allows each endpoint to supply a user-specific password. The separator character and password will be stripped from the string before it is treated as an H.323-ID alias to be registered.
Note that passwords may only be piggybacked in the H.323-ID, not the E.164 address, because the E.164 address allows a limited set of mostly numeric characters. If the endpoint does not wish to register an H.323-ID, it can still supply an H.323-ID consisting of just the separator character and password. This H.323-ID consisting of just the separator character and password will be understood to be a password mechanism and no H.323-ID will be registered.
|
Defaults
No default
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers.
|
Usage Guidelines
Use the security command to enable identification of registered aliases by RADIUS/TACACS+. If the alias does not exist in RADIUS/TACACS+, the endpoint will not be allowed to register.
A RADIUS/TACACS+ server and encryption key must have been configured in Cisco IOS software for security to work.
Only the first alias of the proper type will be identified. If no alias of the proper type is found, the registration will be rejected.
This command does not allow you to define the password mechanism unless the security type (h323-id or e164 or any) has been defined. Although the no security password command undefines the password mechanism, it leaves the security type unchanged, so security is still enabled. However, the no security command disables security entirely, including removing any existing password definitions.
Examples
The following example enables identification of registrations using the first H.323 ID found in any registration:
The following example enables security, authenticating all users by using their H.323-IDs and a password of qwerty2x:
security password qwerty2x
The next example enables security, authenticating all users by using their H.323-IDs and the password entered by the user in the H.323-ID alias he or she registers:
security password separator !
Now if a user registers with an H.323-ID of joe!024aqx, the gatekeeper authenticates user joe with password 024aqx, and if that is successful, registers the user with the H.323-ID of joe. If the exclamation point is not found, the user is authenticated with the default password, or a null password if no default has been configured.
The following example enables security, authenticating all users by using their E.164 IDs and the password entered by the user in the H.323-ID alias he or she registers:
security password separator !
Now if a user registers with an E.164 address of 5551212 and an H.323-ID of !hs8473q6, the gatekeeper authenticates user 5551212 and password hs8473q6. Because the H.323-ID string supplied by the user begins with the separator character, no H.323-ID is registered, and the user is known only by the E.164 address.
Related Commands
Command
|
Description
|
accounting (gatekeeper)
|
Enables the accounting security feature on the gatekeeper.
|
radius-server host
|
Specifies a RADIUS server host.
|
radius-server key
|
Sets the authentication and encryption key for all RADIUS communications between the router and the RADIUS daemon.
|
sequence-numbers
To enable the generation of sequence numbers in each frame generated by the digital signal processor (DSP) for Voice over Frame Relay applications, use the sequence-numbers command in dial-peer configuration mode. To disable the generation of sequence numbers, use the no form of this command.
sequence-numbers
no sequence-numbers
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco 2600 and 3600 series routers and the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
This command was integrated into the Cisco IOS Release 12.0(4)T.
|
Usage Guidelines
Sequence numbers on voice packets allow the digital signal processor (DSP) at the playout side to detect lost packets, duplicate packets, or out-of-sequence packets. This helps the DSP to mask out occasional drop-outs in voice transmission at the cost of one extra byte per packet. The benefit of using sequence numbers versus the cost in bandwidth of adding an extra byte to each voice packet on the Frame Relay network must be weighed to determine whether to disable this function for your application.
Another factor to consider is that this command does not affect codecs that require a sequence number, such as G.726. If you are using a codec that requires a sequence number, the DSP will generate one regardless of the configuration of this command.
Examples
The following example shows how to disable the generation of sequence numbers for VoFR frames on a Cisco 2600 series or 3600 series router or on a Cisco MC3810 multiservice concentrator for VoFR dial peer 200, starting from global configuration mode:
Related Commands
Command
|
Description
|
called-number (dial-peer)
|
Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11 trunk connection.
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a Voice over Frame Relay dial peer.
|
cptone
|
Specifies a regional analog voice interface-related tone, ring, and cadence setting.
|
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 (Voice over Frame Relay)
|
Enables the generation of FRF.11 Annex A frames for a dial peer.
|
session protocol (Voice over Frame Relay)
|
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.
|
server (RLM)
To identify an RLM server, use the server RLM configuration command. To remove the identification, use the no form of this command
server name-tag
no server name-tag
Syntax Description
name-tag
|
Name to identify the server configuration so that multiple entries of server configuration can be entered.
|
Defaults
Disabled
Command Modes
RLM configuration
Command History
Release
|
Modification
|
11.3(7)
|
This command was introduced.
|
Usage Guidelines
Each server can have multiple entries of IP addresses or aliases.
Examples
The following example identifies the RLM server and defines the associated IP addresses:
rlm group 1
server r1-server
link address 10.1.4.1 source Loopback1 weight 4
link address 10.1.4.2 source Loopback2 weight 3
Related Commands
Command
|
Description
|
clear interface
|
Resets the hardware logic on an interface.
|
clear rlm group
|
Clears all RLM group time stamps to zero.
|
interface
|
Defines the IP addresses of the server, configures an interface type, and enters interface configuration mode.
|
link (RLM)
|
Specifies the link preference.
|
protocol rlm port
|
Reconfigures the port number for the basic RLM connection for the whole rlm-group.
|
retry keepalive
|
Allows consecutive keepalive failures a certain amount of time before the link is declared down.
|
show rlm group statistics
|
Displays the network latency of the RLM group.
|
show rlm group status
|
Displays the status of the RLM group.
|
show rlm group timer
|
Displays the current RLM group timer values.
|
shutdown (RLM)
|
Shuts down all of the links under the RLM group.
|
timer
|
Overwrites the default setting of timeout values.
|
server registration-port
To configure the listener port for the server to establish a connection with the gatekeeper, use the server registration-port command in gatekeeper configuration mode. To force the gatekeeper to close the listening socket so that no more new registration takes place, use the no form of this command.
server registration-port port number
no server registration-port port number
Syntax Description
port number
|
Specifies a single range of values from 1 through 65535 for the port number on which the gatekeeper listens for external server connections.
|
Defaults
The registration port of the gatekeeper is not configured, so no external server can register with this gatekeeper.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2500 series, Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
Use this command to configure a server registration port to poll for servers that want to establish connections with the gatekeeper on this router.
Note
The no form of this command forces the gatekeeper on this router to close the listen socket, so it cannot accept more registrations. However, existing connections between the gatekeeper and servers are left open.
Examples
The following example shows how a listener port for a server is established for connection with a gatekeeper:
server registration-port 20000
Related Commands
Command
|
Description
|
server trigger
|
Configure static server triggers for specific RAS messages to be forwarded to a specified server.
|
server trigger
To configure a static server trigger for external applications, use the server trigger command in gatekeeper configuration mode. To remove a single statically configured trigger entry, use the no form of this command. To remove every static trigger you configured if you want to delete them all, use the all keyword.
server trigger {arq | lcf | lrj | lrq | rrq | urq} gkid priority server-id server-ipaddress server-port
no server trigger {arq | lcf | lrj | lrq | rrq | urq} gkid priority
no server trigger all
Syntax Description
all
|
Specified to delete all command-line interface configured triggers.
|
arq, lcf, lrj, lrq, rrq, urq
|
Registration, admission, and status (RAS) protocol message types. Use these message types to specify a submode in the gatekeeper configuration mode in which you configure a trigger for the gatekeeper to act upon. Specify only one message type per server trigger command. There is a different trigger submode for each message type. Each trigger submode has its own set of applicable commands.
|
gkid
|
The local gatekeeper identifier.
|
priority
|
The priority for each trigger. The range is from 1 through 20, with 1 being the highest priority.
|
server-id
|
The ID number of the external application.
|
server-ipaddress
|
The IP address of the server.
|
server-port
|
The port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
|
Defaults
No server triggers are set.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2500 series, Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
Use this command to configure a static server trigger. There are six different server triggers—one for each of the RAS messages. To configure a trigger, go to its submode where a set of subcommands are used to trigger a condition. See the following examples.
In ARQ submode, enter the following syntax:
server trigger arq gkid priority server-id server-ipaddress server-port
In LCF submode, enter the following syntax:
server trigger lcf gkid priority server-id server-ipaddress server-port
In LRJ submode, enter the following syntax:
server trigger lrj gkid priority server-id server-ipaddress server-port
In LRQ submode, enter the following syntax:
server trigger lrq gkid priority server-id server-ipaddress server-port
In RRQ submode, enter the following syntax:
server trigger rrq gkid priority server-id server-ipaddress server-port
In URQ submode, enter the following syntax:
server trigger urq gkid priority server-id server-ipaddress server-port
The following options are available in all submodes:
info-only
|
Information only—no need to wait for acknowledgment.
|
shutdown
|
Enter this subcommand to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward.
|
The destination-info argument is under the ARQ, LRQ, LCF, and LRJ submode and has the following options:
destination-info
|
Configure destination-info to trigger one of the following conditions:
|
e164
email-id
h323-id
word
|
Configure an E.164 pattern.
|
Configure an email ID.
|
Configure an H.323 ID.
|
When configuring the e164 address option, the email-id option, or the h323-id option above, the E.164 address can end in a trailing `., `s, or `*'.
|
The redirect-reason argument is under the ARQ and LRQ submodes and has the following options:
redirect-reason
|
Configure a redirect-reason to trigger on (range of 0 through 65535) with the following reserved values:
|
0
1
2
4
9
10
15
|
Unknown reason.
|
Call forwarding busy or called DTE busy.
|
Call forwarded no reply.
|
Call deflection.
|
Called DTE out of order.
|
Call forwarding by the call DTE.
|
Call forwarding unconditionally.
|
The remote-ext-address argument is under the LCF trigger submode and has the following options:
remote-ext-address
|
Configure remote extension addresses, with the following options:
|
e164
word
|
Configure an E.164 pattern.
|
When configuring the e164 address option, the email-id option, or the h323-id option above, the E.164 address can end in a trailing `., `s, or `*'.
|
The endpoint-type argument is under the RRQ and URQ trigger submodes and has the following options:
endpoint-type
|
Configure the type of endpoint to trigger, with the following options:
|
gatekeeper
h320-gateway
mcu
other-gateway
|
The endpoint is an H.323 gatekeeper.
|
The endpoint is an H.320 gateway.
|
The endpoint is a multipoint control unit (MCU).
|
The endpoint is another type of gateway not specified on this list.
|
proxy
terminal
voice-gateway
|
The endpoint is a H.323 proxy.
|
The endpoint is an H.323 proxy.
|
The endpoint is a voice gateway.
|
The supported-prefix keyword is under the RRQ and URQ submodes and has the following options:
supported-prefix
|
Configure the gateway technology prefix to trigger on.
|
word
|
Enter a word within the set of "0123456789#*" when configuring the E.164 pattern for a gateway technology prefix.
|
Entering the no form of the server trigger command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Examples
The following example configures a server trigger on gatekeeper sj.xyz.com to notify external server "Server-123" of any call to an E.164 number that starts with 1800 followed by any 7 digits (1800551212, for example):
server trigger arq sj.xyz.com 1 Server-123 1.14.93.130 1751
destination-info e164 1800.......
Related Commands
Command
|
Description
|
server registration port
|
Configure a gatekeeper listening port to listen for external server connections.
|
show gatekeeper servers
|
Show a list of currently registered and statically configured triggers on this gatekeeper router.
|
session
To associate a transport session with a specified session-group, use the session group command in backhaul session manager configuration mode. It is assumed that the server is located on a remote machine. To delete the session, use the no form of this command.
session group group-name remote_ip remote_port local_ip local_port priority
no session group group-name remote_ip remote_port local_ip local_port priority
Syntax Description
group
|
Specifies the session-group name.
|
group-name
|
Session-group name.
|
remote_ip
|
Remote IP address.
|
remote_port
|
Remote port number. Range is 1024 through 9999.
|
local_ip
|
Local IP address.
|
local_port
|
Local port number. Range is 1024 through 9999.
|
priority
|
Priority of the session-group. Range is 0 through 9999 and 0 is the highest priority.
|
Defaults
No default behavior or values.
Command Modes
Backhaul session manager configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850 platform.
|
Examples
To associate a transport session with the session-group Group5 and specify the parameters described above, see the following example:
Router(config-bsm)# session group group5 161.44.2.72 5555 172.18.72.198 5555 1
session protocol
To specify a session protocol for calls between the local and remote routers using the packet network, use the session protocol command in dial-peer configuration mode. To reset the default value for this command, use the no form of this command.
session protocol {cisco | sipv2 | aal2-trunk | smtp}
no session protocol
Syntax Description
cisco
|
Configure the dial peer to use proprietary Cisco VoIP session protocol.
|
sipv2
|
SIP users should use this option. This option configures the VoIP dial peer to use IETF SIP.
|
aal2-trunk
|
AAL2 nonswitched trunk session protocol.
|
smtp
|
Specifies Simple Mail Transfer Protocol (SMTP) session protocol.
|
Defaults
No default behaviors or values.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series router.
|
12.0(4)XJ
|
This command was modified for store-and-forward fax on the Cisco AS5300 universal access server.
|
12.1(1)T
|
The sipv2 option was added.
|
12.1(1)XA
|
Support was added for VoATM dial peers on the Cisco MC3810 multiservice concentrator with the aal2-trunk keyword.
|
12.1(2)T
|
Modifications to this command in Cisco IOS Release 12.1(1)XA were integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
The keyword cisco is applicable only to VoIP on the Cisco 3600 series routers. The keyword aal2-trunk is applicable only to VoATM on the Cisco MC3810 multiservice concentrator.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following is an example of configuring a VoIP dial peer for H.323 or SIP as the session protocol for VoIP call signaling:
The following example selects AAL2 trunking as the session protocol on a Cisco MC3810 multiservice concentrator:
session protocol aal2-trunk
The following example selects Cisco Session Protocol as the session protocol on a Cisco 3600 series router:
The following example selects SMTP as the session protocol:
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.
|
session target (VoIP)
|
Configures a network-specific address for a dial peer.
|
session protocol (Voice over Frame Relay)
To establish a Voice over Frame Relay protocol for calls between the local and remote routers via the packet network, use the session protocol command in dial-peer configuration mode. To reset the default value, use the no form of this command.
session protocol {cisco-switched | frf11-trunk}
no session protocol
Syntax Description
cisco-switched
|
Specifies proprietary Cisco VoFR session protocol. (This is the only valid session protocol for the Cisco 7200 series.)
|
frf11-trunk
|
Specifies FRF.11 session protocol.
|
Defaults
cisco-switched
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced for VoIP.
|
12.0(3)XG
|
This command was modified to support VoFR on the Cisco 2600, 3600, and 7200 series routers and the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
The cisco-switched and frf11-trunk keywords were added for VoFR dial peers.
|
Usage Guidelines
For Cisco-to-Cisco dial peer connections, Cisco recommends that you use the default session protocol because of the advantages it offers over a pure FRF.11 implementation. When connecting to FRF.11-compliant equipment from other vendors, use the FRF.11session protocol.
Note
When using the FRF.11 session protocol on Cisco 2600 series and 3600 series routers, you must also use the called-number command.
Examples
The following example shows how to configure the FRF.11 session protocol on a Cisco 2600 series or 3600 series router for VoFR dial peer 200:
session protocol frf11-trunk
The following example shows how to configure the FRF.11 session protocol on a Cisco MC3810 multiservice concentrator for VoFR dial peer 200:
session protocol frf11-trunk
Related Commands
Command
|
Description
|
called-number (dial-peer)
|
Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11 trunk connection.
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a Voice over Frame Relay dial peer.
|
cptone
|
Specifies a regional analog voice interface-related tone, ring, and cadence setting.
|
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 (Voice over Frame Relay)
|
Enables the generation of FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred order of a dial peer within a rotary hunt group.
|
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.
|
session protocol aal2
To enter the voice-service-session configuration mode and specify AAL2 trunking on a Cisco MC3810 multiservice concentrator, use the session protocol aal2 command in voice-service configuration mode.
session protocol aal2
Syntax Description
This command has no keywords or arguments.
Defaults
There is no default setting for this command.
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.1(1)XA
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
Usage Guidelines
This command applies to VoATM on the MC3810 multiservice concentrator.
In the voice-service-session configuration mode for AAL2, you can configure only AAL2 features, such as call admission control and subcell multiplexing.
Examples
The following example shows how to access the voice-service-session configuration mode, beginning in global configuration mode:
session protocol multicast
To set the session protocol as multicast, use the session protocol multicast command dial-peer configuration mode. To negate this command and return to the Cisco default session protocol, use the no version of this command.
session protocol multicast
no session protocol multicast
Syntax Description
There are no keywords or arguments.
Defaults
When this command is not implemented, the default session protocol is cisco.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1(2)XH
|
This command was introduced on Cisco 2600 and Cisco 3600 series routers for the Cisco Hoot and Holler over IP application.
|
12.1(3)T
|
This command was integrated into the Cisco IOS Release 12.1(3)T.
|
Usage Guidelines
Use the session protocol multicast dial-peer configuration command for voice conferencing in a
Hoot and Holler networking implementation. This command allows more than two ports to join the same session simultaneously. It is supported on Cisco 2600 and Cisco 3600 series routers.
Examples
The following example shows the use of the session protocol multicast dial-peer configuration command in context with its accompanying commands:
session protocol multicast
session target ipv4:237.111.0.111:22222
Related Commands
Command
|
Description
|
session target ipv4
|
Assigns the session target for voice-multicasting dial peers.
|
session target (VoATM)
To specify a network-specific address for a specified VoATM dial peer, use the session target command in dial-peer configuration mode. To restore default values for this parameter, use the no form of this command.
Cisco 3600 Series Routers Voice over ATM Dial Peers
session target interface pvc {name | vpi/vci | vci}
no session target
Cisco MC3810 Multiservice Concentrator Voice over ATM Dial Peers
session target {serial | atm} interface pvc {word | vpi/vci | vci} cid
no session target
Syntax Description
serial
|
Specifies the serial interface for the dial-peer address.
|
atm
|
Specifies the ATM interface. The only valid number is 0.
|
interface
|
Interface type and interface number on the router.
|
pvc
|
The specific ATM permanent virtual circuit (PVC) for this dial peer.
|
word
|
(Optional) A name that identifies the PVC. The argument can identify the PVC if a word identifier was assigned when the PVC was created.
|
name
|
The PVC name.
|
vpi/vci
|
ATM network virtual path identifier (VPI) and virtual channel identifier (VCI) of this PVC.
On the Cisco 3600, if you have the Multiport T1/E1 ATM network module with IMA installed, the valid range for vpi is from 0 to 5, and the valid range for vci is from 1 to 255.
If you have the OC3 ATM Network Module installed, the valid range for vpi is from 0 to 15, and the valid range for vci is from 1 to 1023.
|
vci
|
ATM network virtual channel identifier (VCI) of this PVC.
|
cid
|
ATM network channel identifier (CID) of this PVC. The valid range is from 8 to 255.
|
Defaults
The default for this command is enabled with no IP address or domain name defined.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
11.3(1)MA
|
Support was added for VoATM, VoHDLC, and POTS dial peers on the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
Support was added for VoATM dial peers on the Cisco 3600 series routers. Support for VoHDLC on the Cisco MC3810 multiservice concentrator was removed.
|
12.1(2)T
|
Support was added for VoATM on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origin and the loopback type selected.
This command applies to on-ramp store-and-forward fax functions.
You must enter the session protocol aal2-trunk dial-peer configuration command before you can specify a cid for a dial peer for VoATM on the Cisco MC3810 multiservice concentrator.
Note
This command does not apply to plain old telephone service (POTS) dial peers.
Examples
The following example configures a session target for Voice over ATM on a Cisco MC3810 multiservice concentrator. The session target is sent to ATM interface 0, and for a PVC with a VCI of 20.
destination-pattern 13102221111
session target atm0 pvc 20
The following example delivers fax-mail to multiple recipients:
session target marketing-information@mailer.example.com
Assuming that mailer.example.com is running sendmail, you can put the following information into its /etc/aliases file:
fax=+14085551212@sj-offramp.example.com
The following example displays configuring a session target for Voice over ATM on the Cisco 3600 series. The session target is sent to ATM interface 0, and is for a PVC with a VPI/VCI of 1/100.
destination-pattern 13102221111
session target atm1/0 pvc 1/100
Related Commands
Command
|
Description
|
called-number
|
Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a dial peer.
|
cptone
|
Specifies a regional tone, ring, and cadence setting for an analog voice port.
|
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.
|
dtmf-relay
|
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred selection order of a dial peer within a hunt group.
|
session protocol
|
Establishes a VoFR protocol for calls between the local and the remote routers via the packet network.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
session target (VoFR)
To specify a network-specific address for a specified VoFR dial peer, use the session target command in dial-peer configuration mode. To restore default values for this parameter, use the no form of this command.
Cisco 2600 and 3600 Series Routers Voice over Frame Relay Dial Peers
session target interface dlci [cid]
no session target
Cisco MC3810 Multiservice Concentrator Voice over Frame Relay Dial Peers
session target interface dlci [cid]
no session target
Cisco 7200 Series Routers Voice over Frame Relay Dial Peers
session target interface dlci
no session target
Syntax Description
interface
|
Specifies the serial interface and interface number (slot number and port number) associated with this dial peer. For the range of valid interface numbers for the selected interface type, enter a ? character after the interface type.
|
dlci
|
Specifies the data link connection identifier for this dial peer. The valid range is from 16 to 1007.
|
cid
|
(Optional) Specifies the DLCI subchannel to be used for data on FRF.11 calls. A CID must be specified only when the session protocol is frf11-trunk. When the session protocol is cisco-switched, the CID is dynamically allocated. The valid range is from 4 to 255.
Note By default, CID 4 is used for data; CID 5 is used for call-control. We recommend that you select CID values between 6 and 63 for voice traffic. If the CID is greater than 63, the FRF.11 header will contain an extra byte of data.
|
Defaults
The default for this command is enabled with no IP address or domain name defined.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
11.3(1)MA
|
Support was added for VoFR, VoHDLC, and POTS dial peers on the Cisco MC3810 multiservice concentrator.
|
12.0(3)XG
|
Support was added for VoFR dial peers on the Cisco 2600 series and 3600 series routers. The cid option was added.
|
12.0(4)T
|
Support was added for VoFR and POTS dial peers on the Cisco 7200 series routers and the support added in Cisco IOS Release 12.0(3)XG was integrated into Cisco IOS Release 12.0(4)T.
|
Usage Guidelines
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origin and the loopback type selected.
For VoFR dial peers, the cid option is not allowed when the cisco-switched option for the session protocol command is used.
Examples
The following example configures a session target for Voice over Frame Relay on a Cisco MC3810 multiservice concentrator with a session target on serial port1 and a DLCI of 200:
destination-pattern 13102221111
session target serial1 200
The following example shows how to configure serial interface 1/0, DLCI 100 as the session target for VoFR dial peer 200 (an FRF.11 dial peer) on a Cisco 2600 series or 3600 series router, starting from global configuration mode and using the FRF.11 session protocol:
destination-pattern 13102221111
session protocol frf11-trunk
session target serial 1/0 100 20
The following example delivers fax-mail to multiple recipients:
session target marketing-information@mailer.example.com
Assuming that mailer.example.com is running sendmail, you can put the following information into its /etc/aliases file:
fax=+14085551212@sj-offramp.example.com
Related Commands
Command
|
Description
|
called-number
|
Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a dial peer.
|
cptone
|
Specifies a regional tone, ring, and cadence setting for an analog voice port.
|
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.
|
dtmf-relay
|
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred selection order of a dial peer within a hunt group.
|
session protocol
|
Establishes a VoFR protocol for calls between the local and the remote routers via the packet network.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
session target (VoIP)
To specify a network-specific address for a specified VoIP dial peer, use the session target command in dial-peer configuration mode. To restore default values for this parameter, use the no form of this command.
Cisco 2600 and Cisco 3600 Series Routers and Cisco MC8310 Multiservice Concentrator Voice over IP Dial Peers
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp
| loopback:compressed | loopback:uncompressed | ras | settlement}
no session target
Cisco AS5300 Universal Access Server Voice over IP Dial Peers
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp
| loopback:compressed | loopback:uncompressed | mailto: | {name | $d$}@domain-name |
ipv4:destination-address | dns:[$s$. | $d$. | $u$. | $e$.] host-name}
no session target
Cisco AS5800 Universal Access Server Voice over IP Dial Peers
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp
| loopback:compressed | loopback:uncompressed}
no session target
Syntax Description
ipv4:destination-address
|
IP address of the dial peer.
|
dns:[$s$...] host-name
|
Indicates that the domain name server will be used to resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device.
(Optional) Use one of the following three wildcards with this keyword when defining the session target for Voice over IP (VoIP) peers:
$s$.—Indicates that the source destination pattern will be used as part of the domain name.
$d$.—Indicates that the destination number will be used as part of the domain name.
$e$.—Indicates that the digits in the called number will be reversed, periods will be added between the digits of the called number, and this string will be used as part of the domain name.
$u$.—Indicates that the unmatched portion of the destination pattern (such as a defined extension number) will be used as part of the domain name.
|
loopback:rtp
|
Indicates that all voice data will be looped back to the source. This is applicable for VoIP peers.
|
loopback:compressed
|
Indicates that all voice data will be looped back in compressed mode to the source. This is applicable for POTS peers.
|
loopback:uncompressed
|
Indicates that all voice data will be looped-back in uncompressed mode to the source. This is applicable for POTS peers.
|
ras
|
Indicates that the registration, admission, and status (RAS) signaling function protocol is being used, meaning that a gatekeeper will be consulted to translate the E.164 address into an IP address.
|
settlement provider-number
|
Indicates that the settlement server is the target to resolve the terminating gateway address. Enter the provider IP address for provider number.
|
Defaults
The default state for this command is enabled, with no IP address or domain name defined.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers.
|
12.0(3)T
|
Support was added for VoIP and POTS dial peers on the Cisco AS5300 universal access server. The parameter was added for RAS.
|
12.0(4)XJ
|
Support was added for store-and-forward fax on the Cisco AS5300 universal access server platform.
|
12.1(1)T
|
Support was added for session target type of settlement.
|
Usage Guidelines
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origin and the loopback type selected.
The session target dns command can be used with or without the specified wildcards. Using the optional wildcards can reduce the number of VoIP dial peer session targets you must configure if you have groups of numbers associated with a particular router.
Use the session target ras command to specify that the RAS protocol is being used to determine the IP address of the session target.
In Cisco IOS Release 12.1(1)T the session target command configuration cannot combine the target of RAS with the settle-call command. When configuring the VoIP dial peers for a settlement server, if session target type is settlement, the provider-number parameter in the session target and settle-call commands should be identical.
When the VoIP dial peers are configured for a settlement server, if the session target type is settlement, the provider-number parameter in the session target and settle-call commands should be identical.
Examples
The following example configures a session target using DNS for a host, "voice_router," in the domain cisco.com:
session target dns:voice_router.cisco.com
The following example configures a session target using DNS, with the optional $u$. wildcard. In this example, the destination pattern has been configured to allow for any four-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. indicates that the router will use the unmatched portion of the dialed number—in this case, the four-digit extension—to identify the dial peer. As in the preceding example, the domain is "cisco.com."
destination-pattern 1310222....
session target dns:$u$.cisco.com
The following example configures a session target using DNS, with the optional $d$. wildcard. In this example, the destination pattern has been configured for 13102221111. The optional wildcard $d$. indicates that the router will use the destination pattern to identify the dial peer in the "cisco.com" domain.
destination-pattern 13102221111
session target dns:$d$.cisco.com
The following example configures a session target using DNS, with the optional $e$. wildcard. In this example, the destination pattern has been configured for 12345. The optional wildcard $e$. indicates that the router will reverse the digits in the destination pattern, add periods between the digits, and then use this reverse-exploded destination pattern to identify the dial peer in the "cisco.com" domain.
destination-pattern 12345
session target dns:$e$.cisco.com
The following example configures a session target using RAS:
destination-pattern 13102221111
The following example configures a session target using settlement:
session target settlement:0
Related Commands
Command
|
Description
|
called-number
|
Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a dial peer.
|
cptone
|
Specifies a regional tone, ring, and cadence setting for an analog voice port.
|
dtmf-relay
|
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred selection order of a dial peer within a hunt group.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
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.
|
session protocol
|
Establishes a session protocol for calls between the local and remote routers through the packet network in Voice over IP.
|
settle-call
|
Specifies that settlement is to be used for this dial peer, regardless of session target type.
|
session transport
To configure the VoIP dial peer to use TCP or User Datagram Protocol (UDP) as the underlying transport layer protocol for Session Initiation Protocol (SIP) messages, use the session transport command in dial-peer configuration mode. To reset the value to the default, use the no form of this command.
session transport {udp | tcp }
Syntax Description
udp
|
Configure the SIP dial peer to use the UDP transport layer protocol. This is the default.
|
tcp
|
Configure the SIP dial peer to use the TCP transport layer protocol.
|
Defaults
The SIP dial peer uses UDP.
Note
The transport protocol specified with the transport command and the one specified with the session transport command must be the same.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use show sip-ua status to ensure that the transport protocol that you set using the session transport command matches the protocol set using the transport command.
Examples
The following example shows a VoIP dial peer configured to use UDP as the underlying transport
layer protocol for SIP messages:
set
To create a fault-tolerant or non-fault-tolerant session-set with the client or server option, use the set command in backhaul session manager configuration mode. To delete the set, use the no form of this command.
set set-name { client | server } { ft | nft }
no set set-name { client | server } { ft | nft }
Syntax Description
set-name
|
Session-set name.
|
client
|
Client option. The session-set should only be configured as client for backhaul.
|
server
|
Server option.
|
ft
|
Fault-tolerant. Fault-tolerance is the level of ability within a system to operate properly even if a group in the set fails.
|
nft
|
Non-fault-tolerant. Only one group is allowed in a non-fault-tolerant set.
|
Defaults
No default behavior or values.
Command Modes
Backhaul session manager configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
Usage Guidelines
There can be multiple groups associated with a session-set.
The session-set should only be configured for the client for backhaul (not the server).
A set cannot be deleted unless the groups associated with the set are deleted first.
Examples
To specify the client set named Set1 to fault-tolerant, see the following example:
Router(config-bsm)# set set1 client ft
settle-call
To force a call to be authorized with a settlement server that uses the address resolution method specified in the session target type command, use the settle-call command in dial-peer configuration mode. To make sure that no authorization will be performed by a settlement server, use the no form of this command.
settle-call provider-number
no settle-call provider-number
Syntax Description
provider-number
|
Digit defining the ID of a particular settlement server. The only valid entry is 0.
Note If session target type is settlement, the provider-number argument in the session target and settle-call commands should be identical.
|
Defaults
No default behavior or values.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
Usage Guidelines
Using the session target command, a dial peer can determine the address of the terminating gateway through the ipv4, dns, ras, and settlement keywords.
If the session target is not settlement, and the settle-call provider-number argument is set, the gateway resolves the terminating gateway's address using the specified method and then requests the settlement server to authorize that address and create a settlement token for that particular address. If the server cannot authorize the terminating gateway address suggested by the gateway, the call fails.
Do not combine the session target types ras and settle-call. Combination of session target types is not supported in Cisco IOS Release 12.1(1)T.
Examples
The following example sets a call to be authorized with a settlement server that uses the address resolution method specified in the session target:
destination-pattern 1408.......
session target ipv4:172.22.95.14
Related Commands
Command
|
Description
|
session target
|
Specifies a network-specific address for a specified dial peer.
|
settlement
To enter settlement configuration mode and specify the attributes specific to a settlement provider, use the settlement command in global configuration mode. To disable the settlement provider, use the no form of this command.
settlement provider-number
no settlement provider-number
Syntax Description
provider-number
|
Specifies a digit that defines a particular settlement server. The only valid entry is 0.
|
Defaults
0
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The variable provider-number defines a particular settlement provider. For Cisco IOS Release 12.1, only one clearinghouse per system is allowed, and the only valid value for provider-number is 0.
Examples
This example shows how to enter settlement configuration mode:
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the length of time for which a connection is maintained after a communication exchange is completed.
|
customer-id
|
Identifies a carrier or ISP with a settlement provider.
|
device-id
|
Specifies a gateway associated with a settlement provider.
|
encryption
|
Sets the encryption method to be negotiated with the provider.
|
max-connection
|
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
|
response-timeout
|
Configures the maximum time to wait for a response from a server.
|
retry-delay
|
Sets the time between attempts to connect with the settlement provider.
|
retry-limit
|
Sets the connection retry limit.
|
session-timeout
|
Sets the interval for closing the connection when there is no input or output traffic.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown
|
Brings up the settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
settlement roam-pattern
To configure a pattern that must be matched to determine if a user is roaming, use the settlement roam-pattern command in global configuration mode. To delete a particular pattern, use the no form of this command.
settlement provider-number roam-pattern pattern {roaming | no roaming}
no settlement provider-number roam-pattern pattern {roaming | no roaming}
Syntax Description
provider-number
|
Digit defining the ID of particular settlement server. The only valid entry is 0.
|
pattern
|
Specifies a user account pattern.
|
roaming | no roaming
|
Determines whether a user is roaming.
|
Defaults
No default pattern
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
Usage Guidelines
Multiple "roam patterns" could be entered on one gateway.
Examples
The following example will configure a pattern that determines if a user is roaming:
settlement 0 roam-pattern 1222 roam
settlement 0 roam-pattern 1333 noroam
settlement roam-pattern 1444 roam
settlement roam-pattern 1555 noroam
Related Commands
Command
|
Description
|
roaming (settlement)
|
Enables the roaming capability for a settlement provider.
|
settlement
|
Enters settlement configuration mode.
|
sgcp
To start and allocate resources for the Simple Gateway Control Protocol (SGCP) daemon, use the sgcp command in global configuration mode. To terminate all calls, release all allocated resources, and kill the SGCP daemon, use the no form of this command.
sgcp
no sgcp
Syntax Description
This command has no arguments or keywords.
Defaults
The SGCP daemon is not enabled.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command 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 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
When the SGCP daemon is not active, all SGCP messages are ignored.
When you enter the no sgcp command, the SGCP process is removed.
Note
After you enter the no sgcp command, you must save the configuration and reboot the router for the disabling of SGCP to take effect.
Examples
The following example shows the SGCP daemon being enabled:
The following example shows the SGCP daemon being disabled:
Related Commands
Command
|
Description
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp call-agent
To define the IP address of the default Simple Gateway Control Protocol (SGCP) call agent in the router configuration file, use the sgcp call-agent command in global configuration mode. To remove the IP address of the default SGCP call agent from the router configuration, use the no form of this command.
sgcp call-agent ipaddress [:udp port]
no sgcp call-agent ipaddress
Syntax Description
ipaddress
|
Specifies the IP address or hostname of the call agent.
|
:udp port
|
(Optional) Specifies the UDP port of the call agent.
|
Defaults
No IP address is configured.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command 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 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
Setting this command defines the IP address of the default SGCP call agent to which the router sends an initial RSIP (Restart In Progress) packet when the router boots up. This is used for initial boot-up only before the SGCP call agent contacts the router acting as the gateway.
When you enter the no sgcp call-agent command, only the IP address of the default SGCP call agent is removed.
Examples
The following example shows SGCP being enabled and the IP address of the call agent being specified:
sgcp call-agent 209.165.200.225
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp graceful-shutdown
To block all new calls and gracefully terminate all existing calls (wait for the caller to end the call), use the sgcp graceful-shutdown command in global configuration mode. To unblock all calls and allow new calls to go through, use the no form of this command.
sgcp graceful-shutdown
no sgcp graceful-shutdown
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command 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 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
Once you issue this command, all requests for new connections (CreateConnection requests) are denied. All existing calls are maintained until users terminate them, or until you enter the no sgcp command. When the last active call is terminated, the SGCP daemon is terminated, and all resources allocated to it are released.
Examples
The following example shows all new calls being blocked and existing calls being terminated:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp max-waiting-delay
To set the Simple Gateway Control Protocol (SGCP) maximum waiting delay to prevent restart avalanches, use the sgcp max-waiting-delay command in global configuration mode. To restore the default value, use the no form of this command.
sgcp max-waiting-delay delay
no sgcp max-waiting-delay delay
Syntax Description
delay
|
Sets the maximum waiting delay (MWD) value in milliseconds. The valid range is from 0 to 600,000. The default is 3000.
|
Defaults
3,000 milliseconds
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only, and was not generally available.
|
12.0(7)XK
|
Support for this command 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 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Examples
The following example shows the maximum wait delay value set to 40 milliseconds:
sgcp max-waiting-delay 40
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp modem passthru
To enable Simple Gateway Control Protocol (SGCP) modem or fax pass-through, use the sgcp modem passthru command in global configuration mode. To disable SGCP modem or fax pass-through, use the no form of this command.
sgcp modem passthru {ca | cisco | nse}
no sgcp modem passthru {ca | cisco | nse}
Syntax Description
ca
|
Uses the call agent controlled modem upspeed method violation message.
|
cisco
|
Uses a Cisco-proprietary upspeed method based on the protocol.
|
nse
|
Uses the NSE-based modem upspeed method.
|
Defaults
SGCP modem or fax pass-through is disabled by default.
Command Modes
Global configuration.
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 multiservice concentrator and Cisco 3600 series routers (except the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
You can use this command for fax pass-through because the answer tone can come from either modem or fax transmissions. The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions.
If you use the nse option, you must also configure the sgcp tse payload command.
Examples
The following example shows SGCP modem pass-through configured using the call agent upspeed method:
The following example shows SGCP modem pass-through configured using the proprietary Cisco upspeed method:
sgcp modem passthru cisco
The following example shows SGCP modem pass-through configured using the NSE-based modem upspeed:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp quarantine-buffer disable
To disable the Simple Gateway Control Protocol (SGCP) quarantine buffer, use the sgcp quarantine-buffer disable command in global configuration mode. To reenable the SGCP quarantine buffer, use the no form of this command.
sgcp quarantine-buffer disable
no sgcp quarantine-buffer disable
Syntax Description
This command has no arguments or keywords.
Defaults
The SGCP quarantine buffer is enabled.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on 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 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
The SGCP quarantine buffer is the mechanism for buffering the SGCP events between two RQNT messages.
Examples
The following example shows the SGCP quarantine buffer being disabled:
sgcp quarantine-buffer disable
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp request retries
To specify the number of times to retry sending "notify" and "delete" messages to the Simple Gateway Control Protocol (SGCP) call agent, use the sgcp request retries command in global configuration mode. To restore the default value, use the no form of this command.
sgcp request retries count
no sgcp request retries
Syntax Description
count
|
Specifies the number of times a "notify" and "delete" message is retransmitted to the SGCP call agent before it is dropped. The valid range is from 1 to 100. The default is 3.
|
Defaults
The default for the number of times a "notify" and "delete" message is retransmitted to the SGCP call agent before it is dropped is 3
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command 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 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
The actual retry count may be different from the value you enter for this command. The retry count is also limited by the call agent. If there is no response from the call agent after 30 seconds, the gateway will not retry anymore, even though the number set using the sgcp request retries command has not been reached.
The router will stop sending retries after 30 seconds, regardless of the setting for this command.
Examples
The following example shows the system configured to send the sgcp command 10 times before dropping the request:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp request timeout
To specify how long the system should wait for a response to a request, use the sgcp request timeout command in global configuration mode. To restore the default value, use the no form of this command.
sgcp request timeout timeout
no sgcp request timeout
Syntax Description
timeout
|
Specifies the number of milliseconds to wait for a response to a request. Valid range is from 1 to 10,000.
|
Defaults
500 milliseconds
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command 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 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
This command is used for "notify" and "delete" messages, which are sent to the SGCP call agent.
Examples
The following example shows the system configured to wait 40 milliseconds for a reply to a request:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp restart
To trigger the router to send a Restart in Progress (RSIP) message to the Simple Gateway Control Protocol (SGCP) call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller, use the sgcp restart command in global configuration mode. To restore the default value, use the no form of this command.
sgcp restart {delay delay | notify}
no sgcp restart {delay delay | notify}
Syntax Description
delay delay
|
Specifies the restart delay timer value in milliseconds. The valid range is from 0 to 600, and the default value is 0.
|
notify
|
Enables the restart notification upon the SGCP/digital interface state transition.
|
Defaults
Zero (0)
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 multiservice concentrator and Cisco 3600 series routers (except the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
This command is used to send RSIP messages from the router to the SGCP call agent. The RSIP messages are used to synchronize the router and the call agent. RSIP messages are also sent when the sgcp command is entered to enable the SGCP daemon.
You must enter the notify option to enable RSIP messages to be sent.
Examples
The following example shows the system configured to wait 40 milliseconds before restarting SGCP:
The following example shows the system configured to send an RSIP notification to the SGCP call agent when the T1 controller state changes:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp retransmit timer
To configure the Simple Gateway Control Protocol (SGCP) retransmission timer to use a random algorithm, use the sgcp retransmit timer command in global configuration mode. To restore the default value, use the no form of this command.
sgcp retransmit timer {random}
no sgcp retransmit timer {random}
Syntax Description
random
|
Enables the SGCP retransmission timer to use a random algorithm.
|
Defaults
The SGCP retransmission timer does not use the random algorithm.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco 3600 and Cisco MC3810 multiservice concentrator in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
Use this command to enable the random algorithm component of the retransmission timer. For example, if the retransmission timer is set to 200 milliseconds, the first retransmission timer is 200 milliseconds, but the second retransmission timer picks up a timer value randomly between either 200 or 400. The third retransmission timer picks up a timer value randomly of 200, 400, or 800 as shown below:
•
First retransmission timer: 200
•
Second retransmission timer: 200 or 400
•
Third retransmission timer: 200, 400, or 800
•
Fourth retransmission timer: 200, 400, 800, or 1600
•
Fifth retransmission timer: 200, 400, 800, 1600, or 3200 and so on.
After 30 seconds, the retransmission timer no longer retries.
Examples
The following example shows the retransmission timer set to use the random algorithm:
sgcp retransmit timer random
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp timer
To configure how the gateway detects the Real-Time Transport Protocol (RTP) stream lost, use the sgcp timer command in global configuration mode. To restore the default value, use the no form of this command.
sgcp timer {receive-rtcp timer | rtp-nse timer}
no sgcp timer {receive-rtcp timer | rtp-nse timer}
Syntax Description
receive-rtcp timer
|
Sets the multiples of the RTP Control Protocol (RTCP) transmission interval in milliseconds. The valid range is from 1 to 100, and the default is 5.
|
rtp-nse timer
|
Sets the multiples of the RTP named signaling event (NSE) timeout in milliseconds. The valid range is from 100 to 3000, and the default is 200.
|
Defaults
Default for receive-rtcp timer is 5.
Default for rtp-nse timer is 200.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command 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 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
The RTP NSE timer is used for proxy ringing (the ringback tone is provided at the originating gateway).
Examples
The following example shows the receive-rtcp timer set to 100 milliseconds:
sgcp timer receive-rtcp 100
The following example shows the rtp-nse timer set to 1000 milliseconds:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp tse payload
To enable Inband Telephony Signaling Events (TSE) for fax and modem operation, use the sgcp tse payload command in global configuration mode. To restore the default value, use the no form of this command.
sgcp tse payload type
no sgcp tse payload type
Syntax Description
type
|
Sets the TSE payload type. The valid range is from 96 to 119. The default is 0, meaning that the command is disabled.
|
Defaults
Zero (0)
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 multiservice concentrator and Cisco 3600 series routers (except the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
Because this command is disabled by default, you must specify a TSE payload type.
If you configure the sgcp modem passthru command to the nse value, then you must configure this command.
Examples
The following example shows the Simple Gateway Control Protocol (SGCP) modem pass-through set using the NSE-based modem upspeed and the Inband Telephony Signaling Events payload value set to 110:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.up or down so that the call agent can synchronize
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
show aal2 profile
To display the ATM adaptation layer 2 (AAL2) profiles configured on the system, use the show aal2 profile command in privileged EXEC mode.
show aal2 profile all | {itut profile-number | custom profile-number | atmf profile-number}
Syntax Description
all
|
Displays International Telecommunication Union Telecommunication Standardization Sector (ITU-T), ATM Forum, and custom AAL2 profiles configured on the system.
|
itut
|
Displays ITU-T profiles configured on the system.
|
profile-number
|
Specifies the profile number of the AAL2 profile to display. The available choices are as follows:
For ITU-T:
• 1 = G.711 u-law
• 2 = G.711 u-law with silence insertion descriptor (SID)
• 7 = G.711 u-law and G.729ar8
For ATMF: None. ATMF is not supported.
For custom:
• 100 = G.711 u-law and G.726r32
• 110 = G.711 u-law, G.726r32, and G.729ar8
|
custom
|
Displays custom profiles configured on the system.
|
atmf
|
Displays ATM Forum profiles configured on the system.
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(1)XA
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
Usage Guidelines
This command applies to AAL2 Voice over ATM (VoATM) applications on the Cisco MC3810 multiservice concentrator.
Use the show aal2 profile EXEC command to display the AAL2 profiles configured in the system.
Examples
The following is sample output from the show aal2 profile command for displaying all the profiles configured in the system:
Router# show aal2 profile all
Printing all the Profiles in the system
Profile Type: ITUT Profile Number: 1 SID Support: 0
Red enable: 1 Num entries: 1
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15
Profile Type: ITUT Profile Number: 2 SID Support: 1
Red enable: 1 Num entries: 1
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15
Profile Type: custom Profile Number: 100 SID Support: 1
Red enable: 1 Num entries: 2
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 7
Coding type: g726r32 Packet length: 40 UUI min: 8 UUI max: 15
Profile Type: ITUT Profile Number: 7 SID Support: 1
Red enable: 1 Num entries: 2
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15
Coding type: g729ar8 Packet length: 10 UUI min: 0 UUI max: 15
Profile Type: custom Profile Number: 110 SID Support: 1
Red enable: 1 Num entries: 3
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 7
Coding type: g726r32 Packet length: 40 UUI min: 8 UUI max: 15
Coding type: g729ar8 Packet length: 30 UUI min: 8 UUI max: 15
Table 26 provides an alphabetical listing of the fields in this output and a description of each field.
Table 26 show aal2 profile Field Descriptions
Field
|
Description
|
Profile Type
|
Category of codec types configured on DSP. Possible types are ITU-T, ATMF, and custom.
|
ITUT Profile Number
|
Predefined combination of one or more codec types configured for a digital signal processor (DSP).
|
SID Support
|
Silence insertion descriptor.
|
Red enable
|
Redundancy enable for type3 packets.
|
Num entries
|
Number of profile elements.
|
Coding type
|
Voice compression algorithm.
|
Packet length
|
Sample size.
|
UUI min
|
Minimum sequence number on the voice packets.
|
UUI max
|
Maximum sequence number on the voice packets.
|
Related Commands
Command
|
Description
|
codec aal2-profile
|
Sets the codec profile for a DSP on a per-call basis.
|
show atm video-voice address
To display the network service access point (NSAP) address for the ATM interface, enter the show atm video-voice address command in privileged EXEC mode.
show atm video-voice address
Syntax Description
This command has no keywords or arguments.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(5)XK
|
This command was introduced for the Cisco MC3810 multiservice concentrator.
|
12.0(7)T
|
Cisco IOS Release 12.0(5)XK was integrated into Cisco IOS Release 12.0(7)T.
|
Usage Guidelines
Enter this command to review ATM interface NSAP addresses that have been assigned with the atm video aesa command and to ensure that ATM management is confirmed for those addresses.
Examples
On a Cisco MC3810 multiservice concentrator, the following example displays ATM interface NSAP addresses:
Router# show atm video-voice address
nsap address type ilmi status
47.0091810000000002F26D4901.00107B4832E1.FE VOICE_AAL5 Confirmed
47.0091810000000002F26D4901.00107B4832E1.C8 VIDEO_AAL1 Confirmed
Related Commands
Command
|
Description
|
codec aal2-profile
|
Sets the codec profile for a DSP on a per-call basis.
|
show backhaul-session-manager group
To display status, statistics, or configuration information for all available session-groups, use the show backhaul-session-manager group command in privileged EXEC mode.
show backhaul-session-manager group { status | stats | cfg } { all | name group-name }
Syntax Description
status
|
Displays status information for session-groups.
|
stats
|
Displays statistics for session-groups.
|
cfg
|
Displays configuration information for session-groups.
|
all
|
Displays information for all available session-groups.
|
name group-name
|
Displays information for a specific session-group. The group-name argument specifies the name of the session-group.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
Examples
The following displays statistics for all session-groups:
Router# show backhaul-session-manager group stats all
Session-Group grp1 statistics
Un-Successful Fail-Over attempts:0
Active Pkts receive count :0
Standby Pkts receive count :0
Total PDUs dispatch err :0
The following displays the current configuration for all session-groups:
Router# show backhaul-session-manager group cfg all
Dest:10.5.0.3 8304 Local:10.1.2.15 8304 Priority:0
Dest:10.5.0.3 8300 Local:10.1.2.15 8300 Priority:2
Dest:10.5.0.3 8303 Local:10.1.2.15 8303 Priority:2
timer cumulative ack :100
timer transfer state :2000
The following displays the current status of all session-groups. This group named grp1 belongs to the set named set1.
The Status will be either Group-OutOfService (no session in the group has been established) or Group-Inservice (at least one session in the group has been established).
The Status(use) will be either Group-Standby (the VSC connected to the other end of this group will go into standby mode), Group-Active (the VSC connected to the other end of this group will be the active VSC), or Group-None (the VSC has not declared its intent yet).
Router# show backhaul-session-manager group status all
Status :Group-OutOfService
Related Commands
Command
|
Description
|
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 backhaul-session-manager session
To display various information for about a session or sessions, use the show backhaul-session-manager session command in privileged EXEC mode.
show backhaul-session-manager session { all | ip ip_address }
Syntax Description
all
|
All available sessions.
|
ip ip_address
|
The IP address of the local or remote session.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
Examples
To display information for all available sessions, see the following example.
The State will be OPEN (the connection is established), OPEN_WAIT (the connection is awaiting establishment), OPEN_XFER (session failover is in progress for this session, which is a transient state), or CLOSE (this session is down, also a transient state). The session will move to OPEN_WAIT after waiting a fixed amount of time.
The Use-status field indicates whether PRI signaling traffic is currently being transported over this session . The field will be either OOS (this session is not being used to transport signaling traffic) or IS (this session is being used currently to transport all PRI signaling traffic). OOS does not indicate if the connection is established and IS indicates that the connection is established.
Router# show backhaul-session-manager session all
Group:grp1 /*this session belongs to the group named 'grp1' */
Local:10.1.2.15 , port:8303
Remote:10.5.0.3 , port:8303
RUDP Option:Client, Conn Id:0x2
Status:OPEN_WAIT, Use-status:OOS, /*see explanation below */
# of unexpected RUDP transitions (total) 0
# of unexpected RUDP transitions (since last reset) 0
Receive pkts - Total:0 , Since Last Reset:0
Recieve failures - Total:0 ,Since Last Reset:0
Transmit pkts - Total:0, Since Last Reset:0
Transmit Failures (PDU Only)
Due to Blocking (Not an Error) - Total:0, Since Last Reset:0
Due to causes other than Blocking - Total:0, Since Last
Transmit Failures (NON-PDU Only)
Due to Blocking(Not an Error) - Total:0, Since Last Reset:0
Due to causes other than Blocking - Total:0, Since Last
Send window full failures:0
Resource unavailble failures:0
Related Commands
Command
|
Description
|
show backhaul-session-manager group
|
Displays status, statistics, or configuration of a specified or all session-groups.
|
show backhaul-session-manager set
|
Displays session-groups associated with a specified or all session-sets.
|
show backhaul-session-manager set
To display session-groups associated with a specified session-set or all session-sets, use the show backhaul-session-manager set command in privileged EXEC mode.
show backhaul-session-manager set { all | name session-set-name }
Syntax Description
all
|
All available session-sets.
|
name session-set-name
|
A specified session-set.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
Examples
To show session groups associated with all session-sets, see the following example:
Router# show backhaul-session-manager set all
Related Commands
Command
|
Description
|
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 a session or all sessions.
|
show call active
To display active call information for voice calls or fax transmissions in progress, use the show call active command in user EXEC or privileged EXEC mode.
show call active {voice | fax}[brief]
Syntax Description
voice
|
Specifies that information be displayed for all active voice calls.
|
fax
|
Specifies that information be displayed for all active fax calls.
|
brief
|
(Optional) Displays a truncated version of the active call information.
|
Defaults
No default behavior or values.
Command Modes
User EXEC or
Privileged EXEC
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 2600 series and 3600 series.
|
12.0(3)XG
|
Support for VoFR was added.
|
12.0(4)XJ
|
This command was modified for store-and-forward fax on the Cisco AS5300 universal access server.
|
12.0(4)T
|
This command was first supported on the Cisco 7200 series.
|
12.0(7)XK
|
This command was first supported on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.1(3)T
|
This command was modified for Modem Passthrough over VoIP on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use the show call active command to display the contents of the active call table. This command displays information about call times, dial peers, connections, quality of service, and other status and statistical information. If you use the voice keyword, information is displayed about all voice calls currently connected through the router or access server. If you use the fax keyword, information is displayed about all fax calls currently connected.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
See Table 19 for a listing of the information types associated with this command.
Examples
The following is sample output from the show call active voice command:
Router# show call active voice
ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
VoiceTxDuration=155310 ms
ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
RemoteIPAddress=1.14.82.14
tx_DtmfRelay=inband-voice
SessionTarget=ipv4:1.14.82.14
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=67 ms
LoWaterPlayoutDelay=67 ms
Modem passthrough signaling method is nse:
Consecutive-packets-lost Events = 0
Corrected packet-loss Events = 0
Last Buffer Drain/Fill Event = 157sec
Time between Buffer Drain/Fills = Min 0sec Max 0sec
The following is sample output from the show call active voice brief command:
Router# show call active voice brief
<ID>: <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
Tele <int>: tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
3 : 104443hs.1 +521 pid:100 Answer 50110 active
dur 00:03:28 tx:20151/3036404 rx:20102/3517936
Tele 0:D:1: tx:199630/199630/0ms g711ulaw noise:-75 acom:11 i/0:-22/-13 dBm
3 : 104648hs.1 +316 pid:2 Originate 55240 active
dur 00:03:28 tx:20102/3276712 rx:20151/3277628
IP 1.14.82.14:18202 rtt:3ms pl:40/0ms lost:0/0/0 delay:67/67/67ms g729r8
MODEMPASS nse buf:0/0 loss 0% 0/0 last 195s dur:0/0s
The following is sample output from the show call active fax command:
Router# show call active fax
ConnectionId[0x37EC7F41 0xB0110001 0x0 0x35C34]
ConnectionId=[0x37EC7F41 0xB0110001 0x0 0x35C34]
The following is sample output from the show call active fax brief command:
Router# show call active fax brief
<ID>: <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state> \
tx:<packets>/<bytes> rx:<packets>/<bytes> <state>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
Tele <int>: tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
1 : 22021hs.1 +2263 pid:0 Answer wook song active
IP 0.0.0.0 AcceptedMime:2 DiscardedMime:1
1 : 23193hs.1 +1091 pid:3469 Originate 527.... active
Tele : tx:31200/10910/20290ms noise:-1 acom:-1 i/0:0/0 dBm
Table 27 provides an alphabetical listing of the fields displayed in the output from the show call active command and a description of each field.
Table 27 show call active Field Descriptions
Field
|
Description
|
ACOM Level
|
Current ACOM level for this call. ACOM is the combined loss achieved by the echo canceler, which is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.
|
Buffer Drain Events
|
Total number of jitter buffer drain events.
|
Buffer Fill Events
|
Total number of jitter buffer fill events.
|
CallDuration
|
Length of the call in hours, minutes, and seconds, hh:mm:ss.
|
CallOrigin
|
Call origin: answer or originate.
|
CallState
|
Current state of the call.
|
ChargedUnits
|
Total number of charging units that apply to this peer since system startup. The unit of measure for this field is hundredths of second.
|
CodecBytes
|
Payload size in bytes for the codec used.
|
CoderTypeRate
|
Negotiated coder rate. This value specifies the send rate of voice or fax compression to its associated call leg for this call.
|
ConnectionId
|
Global call identifier for this gateway call.
|
ConnectTime
|
Time at which the call was connected.
|
Consecutive-packets-lost Events
|
Total number of consecutive (two or more) packet-loss events.
|
Corrected packet-loss Events
|
Total number of packet loss events that were corrected using the RFC 2198 method.
|
Dial-Peer
|
Tag of the dial peer sending this call.
|
ERLLevel
|
Current Echo Return Loss (ERL) level for this call.
|
FaxTxDuration
|
Duration of fax transmission from this peer to the voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.
|
GapFillWithInterpolation
|
Duration of a voice signal played out with a signal synthesized from parameters, or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call.
|
GapFillWithRedundancy
|
Duration of a voice signal played out with a signal synthesized from available redundancy parameters because voice data was lost or not received in time from the voice gateway for this call.
|
GapFillWithPrediction
|
Duration of the voice signal played out with signal synthesized from parameters, or samples of data preceding in time, because voice data was lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser or frame-concealment strategies in G.729 and G.723.1 compression algorithms.
|
GapFillWithSilence
|
Duration of a voice signal replaced with silence because voice data was lost or not received in time for this call.
|
HiWaterPlayoutDelay
|
High-water mark Voice Playout FIFO Delay during this call.
|
Index
|
Dial peer identification number.
|
InfoActivity
|
Active information transfer activity state for this call.
|
InfoType
|
Information type for this call, for example, voice or fax.
|
InSignalLevel
|
Active input signal level from the telephony interface used by this call.
|
Last Buffer Drain/Fill Event
|
Time since the last jitter buffer drain or fill event, in seconds.
|
LogicalIfIndex
|
Index number of the logical interface for this call.
|
LoWaterPlayoutDelay
|
Low water mark Voice Playout FIFO Delay during this call.
|
Modem passthrough signaling method in use
|
Indicates that this is a modem pass-through call and that named signaling events (NSEs)—also called telephone-events in RFC 2833—are used for signaling codec upspeed. The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions. This means that you might move to a faster codec when you have both voice and data calls and then slow down when there is only voice traffic.
|
NoiseLevel
|
Active noise level for this call.
|
OnTimeRvPlayout
|
Duration of voice playout from data received on time for this call. Derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.
|
OutSignalLevel
|
Active output signal level to the telephony interface used by this call.
|
PeerAddress
|
Destination pattern or number associated with this peer.
|
PeerId
|
ID value of the peer table entry to which this call was made.
|
PeerIfIndex
|
Voice port index number for this peer. For ISDN media, this would be the index number of the B channel used for this call.
|
PeerSubAddress
|
Subaddress when this call is connected.
|
Percent Packet Loss
|
Total percent packet loss.
|
ReceiveBytes
|
Number of bytes received by the peer during this call.
|
ReceiveDelay
|
Average Playout FIFO Delay plus the Decoder Delay during this voice call.
|
ReceivePackets
|
Number of packets received by this peer during this call.
|
RemoteIPAddress
|
Remote system IP address for the VoIP call.
|
RemoteUDPPort
|
Remote system UDP listener port to which voice packets are sent.
|
RoundTripDelay
|
Voice packet round trip delay between the local and remote systems on the IP backbone for this call.
|
SelectedQoS
|
Selected RSVP quality of service (QoS) for this call.
|
SessionProtocol
|
Session protocol used for an Internet call between the local and remote routers through the IP backbone.
|
SessionTarget
|
Session target of the peer used for this call.
|
SetupTime
|
Value of the system UpTime when the call associated with this entry was started.
|
SignalingType
|
Signaling type for this call; for example, channel-associated signaling (CAS) or common-channel signaling (CCS).
|
Time between Buffer Drain/Fills
|
Minimum and maximum durations between jitter buffer drain or fill events, in seconds.
|
TransmitBytes
|
Number of bytes sent by this peer during this call.
|
TransmitPackets
|
Number of packets sent by this peer during this call.
|
TxDuration
|
Duration of transmit path open from this peer to the voice gateway for this call.
|
VAD
|
Whether voice activation detection (VAD) was enabled for this call.
|
VoiceTxDuration
|
Duration of voice transmission from this peer to the voice gateway for this call. Derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.
|
Related Commands
Command
|
Description
|
show call history
|
Displays the call history table.
|
show dial-peer voice
|
Displays configuration information for dial peers.
|
show num-exp
|
Displays how the number expansions are configured in Voice over IP.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show call application voice
To define the names of the audio files that the interactive voice response (IVR) script will play, the operation of the abort keys, the prompts that are used, and caller interaction, use the show call application voice command in EXEC mode.
show call application voice [name | summary]
Syntax Description
name
|
(Optional) The name of the desired IVR application.
|
summary
|
(Optional) Displays a one-line summary. If the command is entered without the summary keyword, a complete detailed description is displayed of the application.
|
Defaults
No default behavior or values.
Command Modes
EXEC
Command History
Release
|
Modification
|
11.3(6)NA2
|
This command was introduced on the Cisco 2500 series and Cisco 3600 series routers and the Cisco AS5300 universal access server.
|
Usage Guidelines
If the name of a specific application is entered, it will give information about that application.
If the summary keyword is entered, a one-line summary will be displayed about each application.
If the command is entered without the summary, a detailed description of the entered IVR application is displayed.
Examples
This example shows the output for the clid_authen_collect IVR script:
Router# show call application voice clid_authen_collect
Application clid_authen_collect has 10 states with 0 calls active
State start has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=ani, pinName=dnis
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_SETUP_IND do action IVR_ACT_CALL_SETUP_ACK
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_AAA_FAIL goto state get_account
State end has 1 actions and 3 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_DISCONNECT_DONE do action IVR_ACT_CALL_DESTROY
State get_account has 4 actions and 7 events
URL: flash:enter_account.au
allowInt=1, pContent=0x60E4C564
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern account is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state get_pin
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_TIMEOUT goto state get_account count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_account
State get_pin has 4 actions and 7 events
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern pin is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state authenticate
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_TIMEOUT goto state get_pin count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_pin
State authenticate has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=account, pinName=pin
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_TIMEOUT do nothing count=0
If Event IVR_EV_AAA_FAIL goto state authenticate_fail
State collect_dest has 4 actions and 8 events
URL: flash:enter_destination.au
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_DIALPLAN.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
If Event IVR_EV_DIAL_COL_SUCCESS goto state place_call
If Event IVR_EV_DIAL_COL_FAIL goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
State place_call has 1 actions and 4 events
Do Action IVR_ACT_PLACE_CALL.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_UP goto state active
If Event IVR_EV_CALL_FAIL goto state place_fail
State active has 0 actions and 2 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State authenticate_fail has 1 actions and 2 events
URL: flash:auth_failed.au
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State place_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY_FAILURE_TONE.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
Router# show call application voice clid_authen_collect
Application clid_authen_collect has 10 states with 0 calls active
State start has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=ani, pinName=dnis
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_SETUP_IND do action IVR_ACT_CALL_SETUP_ACK
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_AAA_FAIL goto state get_account
State end has 1 actions and 3 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_DISCONNECT_DONE do action IVR_ACT_CALL_DESTROY
State get_account has 4 actions and 7 events
URL: flash:enter_account.au
allowInt=1, pContent=0x60E4C564
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern account is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state get_pin
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_TIMEOUT goto state get_account count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_account
State get_pin has 4 actions and 7 events
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern pin is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state authenticate
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_TIMEOUT goto state get_pin count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_pin
State authenticate has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=account, pinName=pin
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_TIMEOUT do nothing count=0
If Event IVR_EV_AAA_FAIL goto state authenticate_fail
State collect_dest has 4 actions and 8 events
URL: flash:enter_destination.au
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_DIALPLAN.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
If Event IVR_EV_DIAL_COL_SUCCESS goto state place_call
If Event IVR_EV_DIAL_COL_FAIL goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
State place_call has 1 actions and 4 events
Do Action IVR_ACT_PLACE_CALL.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_UP goto state active
If Event IVR_EV_CALL_FAIL goto state place_fail
State active has 0 actions and 2 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State authenticate_fail has 1 actions and 2 events
URL: flash:auth_failed.au
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State place_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY_FAILURE_TONE.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
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
|
Reloads the designated TCL script.
|
show call fallback cache
To see the current Calculated Planning Impairment Factor (ICPIF) estimates for all IP addresses in cache, use the show call fallback cache command in EXEC mode.
show call fallback cache [ip-address]
Syntax Description
ip-address
|
(Optional) Specifies a specific IP address.
|
Defaults
This command is not configured by default.
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
To clear all entries in the cache, use the clear call fallback cache command.
Examples
The following example displays output from the show call fallback cache command:
Router# show call fallback cache
Probe IP Address Codec Delay Loss ICPIF Reject Accept
----- ---------- ----- ----- ---- ----- ------ ------
1 1.1.1.4 g729r8 40 0 0 0 9
2 122.24.56.25 g729r8 148 10 5 1 4
IP Address IP Address to which the probe is sent
Codec Codec Type of the probe
Delay Delay in milliseconds that the probe incurred
Loss Loss in % that the probe incurred
ICPIF Computed ICPIF value for the probe
Reject Number of times that calls of Codec Type <Codec>
were rejected to the IP Address
Accept Number of times that calls of Codec Type <Codec>
were accepted to the IP Address
active probes Number of destinations being probed
Router# show call fallback cache 10.14.115.53
Probe IP Address Codec ICPIF Reject Accept
----- ---------- ----- ----- ------ ------
1 10.14.115.53 g729r8 0 0 2
Related Commands
Command
|
Description
|
show call fallback stats
|
Displays the call fallback statistics.
|
show call fallback config
To display the call fallback configuration, use the show call fallback config command in EXEC mode.
show call fallback config
Syntax Description
This command has no arguments or keywords.
Defaults
This command is not configured by default.
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
Examples
The following example displays output from the show call fallback config command:
Router# show call fallback config
ICPIF value timeout:20 seconds
Number of packets in a probe:20
IP precedence of probe packets:2
Fallback cache size:2 entries
Fallback cache timeout:240 seconds
Instantaneous value weight:65
Related Commands
Command
|
Description
|
call fallback monitor
|
Enables the monitoring of destinations without fallback to alternate dial peers.
|
show voice trunk-conditioning signaling
|
Enables fallback to alternate dial peers in case of network congestion.
|
show call fallback stats
To display the call fallback statistics, use the show call fallback stats command in EXEC mode.
show call fallback stats
Syntax Description
This command has no arguments or keywords.
Defaults
This command is not configured by default.
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
To remove all values, use the clear call fallback stats command.
Examples
The following example displays output from the show call fallback stats command:
Router# show call fallback stats
Total accepted calls Number of times that calls were successful over IP.
Total rejected calls Number of times that calls were rejected over IP.
Total cache overflows Number of times that the fallback cache overflowed and requied
pruning.
Related Commands
Command
|
Description
|
clear call fallback stats
|
Clears the call fallback statistics.
|
show call fallback cache
|
Displays the current ICPIF estimates for all IP addresses in the cache.
|
show call history
To display the call history table for voice calls or fax transmissions, use the show call history command in user EXEC or privileged EXEC mode.
show call history {voice | fax}[last number | brief]
Syntax Description
voice
|
Specifies that call history information be displayed for voice calls.
|
fax
|
Specifies that call history information be displayed for fax calls.
|
last number
|
(Optional) Displays the last calls connected, where the number of calls that appear is defined by the number argument. Valid values are from 1 to 100.
|
brief
|
(Optional) Displays a truncated version of the call history table.
|
Defaults
No default behavior or values.
Command Modes
User EXEC
Privileged EXEC
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series.
|
12.0(3)XG
|
Support for Voice over Frame Relay (VoFR) was added on the Cisco 2600 and Cisco 3600 series.
|
12.0(4)XJ
|
This command was modified for store-and-forward fax.
|
12.0(4)T
|
The brief keyword was added and the command was first supported on the Cisco 7200 series.
|
12.0(7)XK
|
Support for the brief keyword was added on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS 12.1(2)T.
|
Usage Guidelines
The show call history command displays a call history table containing a list of voice or fax calls connected through the router in descending time order. The maximum number of calls contained in the table can be set to a number between 0 and 500 using the dial-control-mib command in global configuration mode. The default maximum number of table entries is 50. Each call record is aged out of the table after a configurable number of minutes has elapsed, also specified by the dial-control-mib command. The default timer value is 15 minutes.
You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the keyword last, and define the number of calls to be displayed with the number argument.
To display a truncated version of the call history table, use the brief keyword.
When using the fax keyword, this command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following is sample output from the show call history voice command:
Router# show call history voice
DisconnectText=normal call clearing.
ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
RemoteIPAddress=1.14.82.14
tx_DtmfRelay=inband-voice
SessionTarget=ipv4:1.14.82.14
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=67 ms
LoWaterPlayoutDelay=67 ms
Modem passthrough signaling method is nse
Consecutive-packets-lost Events = 0
Corrected packet-loss Events = 0
Last Buffer Drain/Fill Event = 373sec
Time between Buffer Drain/Fills = Min 0sec Max 0sec
DisconnectText=normal call clearing.
ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
VoiceTxDuration=375300 ms
The following is sample output from the show call history voice brief command:
Router# show call history voice brief
<ID>: <start>hs.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
Telephony <int>: tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm
The following is sample output from the show call history fax command:
Router# show call history fax
DisconnectText=normal call clearing.: Normal connection
ConnectionId=[0x37EC7F41 0xB0110001 0x0 0x35C34]
DisconnectText=normal call clearing.