Table Of Contents
Cisco IOS Voice, Video, and Fax Commands:
Si Through Z
signal
signal keepalive
signal pattern
signal sequence oos
signal timing idle suppress-voice
signal timing oos
signal timing oos restart
signal timing oos slave-standby
signal timing oos suppress-all
signal timing oos suppress-voice
signal timing oos timeout
signal-type
sip-server
sip-ua
snmp enable peer-trap poor-qov
ss7 mtp2-variant bellcore
ss7 mtp2-variant itu
ss7 mtp2-variant ntt
ss7 mtp2-variant ttc
ss7 session
ss7 session cumack_t
ss7 session kp_t
ss7 session m_cumack
ss7 session m_outseq
ss7 session m_rcvnum
ss7 session m_retrans
ss7 session retrans_t
ss7 set failover-timer
station-id
subcell-mux
supervisory disconnect
supervisory disconnect anytone
supervisory disconnect dualtone voice-class
tdm-group
tech-prefix
test call fallback probe
test pots dial
test pots disconnect
test translation-rule
test voice port detector
test voice port inject-tone
test voice port loopback
test voice port relay
test voice port switch
test vrm busyout
test vrm reset
test vrm unbusyout
threshold noise
timeouts call-disconnect
timeouts initial
timeouts interdigit
timeouts ringing
timeouts wait-release
timers
timing clear-wait
timing delay-duration
timing delay-start
timing delay-with-integrity
timing dial-pulse min-delay
timing dialout-delay
timing digit
timing guard-out
timing hookflash-input
timing hookflash-out
timing interdigit
timing percentbreak
timing pulse
timing pulse-interdigit
timing wink-duration
timing wink-wait
token-root-name
tone ringback alert-no-PI
translate
translate-outgoing
translation-rule
transport
type (voice)
type (settlement)
unbundle vfc
url
use-proxy
vad (dial peer)
vad (voice-port)
vbr-rt
vofr
voice call convert-discpi-to-prog
voice call send-alert
voice-card
voice class busyout
voice class codec
voice-class codec (dial peer)
voice class dualtone
voice class h323
voice-class h323 (dial peer)
voice class permanent
voice-class permanent (dial-peer)
voice-class permanent (voice-port)
voice confirmation-tone
voice-encap
voice-group
voice hunt
voice local-bypass
voice-port
voice-port busyout
voice rtp send-recv
voice service
voice vad-time
voip-incoming translation-rule
zone access
zone bw
zone local
zone prefix
zone remote
zone subnet
Cisco IOS Voice, Video, and Fax Commands:
Si Through Z
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 Si. 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.
signal
To specify the type of signaling for a voice port, use the signal command in voice-port configuration mode. To restore the default value for this command, use the no form of this command.
FXO and FXS Voice Ports
signal {loop-start | ground-start}
no signal {loop-start | ground-start}
E&M Voice Ports
signal {wink-start | immediate | delay-dial}
no signal {wink-start | immediate | delay-dial}
Syntax Description
loop-start
|
Specifies loop start signaling. Used for Foreign Exchange Office (FXO) and Foreign Exchange Station (FXS) interfaces. With loop start signaling only one side of a connection can hang up. This is the default setting for FXO and FXS voice ports.
|
ground-start
|
Specifies ground start signaling. Used for FXO and FXS interfaces. Ground start signalling allows both sides of a connection to place a call and to hang up.
|
wink-start
|
Indicates that the calling side seizes the line by going off-hook on its E-lead then waits for a short off-hook "wink" indication on its M-lead from the called side before sending address information as dual tone multifrequency (DTMF) digits. Used for E&M tie trunk interfaces. This is the default setting for E&M voice ports.
|
immediate
|
Indicates that the calling side seizes the line by going off-hook on its E-lead and sends address information as DTMF digits. Used for E&M tie trunk interfaces.
|
delay-dial
|
Indicates that the calling side seizes the line by going off-hook on its E-lead. After a timing interval, the calling side looks at the supervision from the called side. If the supervision is on-hook, the calling side starts sending information as DTMF digits; otherwise, the calling side waits until the called side goes on-hook and then starts sending address information. Used for E&M tie trunk interfaces.
|
Defaults
Loop-start for FXO and FXS interfaces; wink-start for E&M interfaces.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series routers.
|
Usage Guidelines
This command applies to analog voice ports only.
Configuring the signal command for an FXO or FXS voice port changes the signal value for both voice ports on a voice port module (VPM) card.
Note
If you change the signal type for an FXO voice port on Cisco 3600 series routers, you need to move the appropriate jumper in the voice interface card of the voice network module. For more information about the physical characteristics of the voice network module, refer to the installation documentation, Voice Network Module and Voice Interface Card Configuration Note, that came with your voice network module.
Configuring this command for an E&M voice port changes only the signal value for the selected voice port. In either case, the voice port must be shut down and then activated before the configured values will take effect.
Some PBXs will miss initial digits if the E&M voice port is configured for Immediate signaling. If this occurs, use Delay-Dial signaling instead. Some non-Cisco devices have a limited number of DTMF receivers. This type of equipment must delay the calling side until a DTMF receiver is available.
Examples
The following example configures ground start signaling on the Cisco 3600 series as the signaling type for a voice port, which means that both sides of a connection can place a call and hang up:
signal keepalive
To configure the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks, use the signal keepalive command in voice-class configuration mode. To restore the default value, use the no form of this command.
signal keepalive number
no signal keepalive number
Syntax Description
number
|
Specifies the keepalive signaling packet interval, in seconds. The valid range is from 1 to 65,535.
|
Defaults
A keepalive packet is sent every 5 seconds.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
Before configuring the keepalive signaling interval, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. The voice class must then be assigned to a dial peer.
Examples
The following example, beginning in global configuration mode, sets the keepalive signaling interval to 3 seconds for voice class 10.
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies a dial-peer type.
|
signal pattern
|
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal timing oos
|
Configures the signal timing parameter for the OOS state of a call.
|
voice-class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice class permanent
|
Assigns a previously-configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.
|
signal pattern
To define the ABCD bit patterns that identify the idle and out-of-service (OOS) states for Cisco trunks and FRF.11 trunks, use the signal pattern command in voice-class configuration mode. To remove the signal pattern setting from the voice class, use the no form of this command.
signal pattern {idle receive | idle transmit | oos receive | oos transmit} bit-pattern
no signal pattern {idle receive | idle transmit | oos receive | oos transmit} bit-pattern
Syntax Description
idle receive
|
Defines the signaling pattern for identifying an idle message from the network. Also defines the idle signaling pattern to be sent to the PBX if the network trunk is out of service and the signal sequence oos idle-only or signal sequence oos both command is configured.
|
idle transmit
|
Defines the signaling pattern for identifying an idle message from the PBX.
|
oos receive
|
Defines the OOS signaling pattern to be sent to the PBX if the network trunk is out of service and the signal sequence oos oos-only or signal sequence oos both command is configured.
|
oos transmit
|
Defines the signaling pattern for identifying an OOS message from the PBX.
|
bit-pattern
|
Defines the ABCD bit pattern. Valid values are from 0000 to 1111.
|
Defaults
idle receive
|
For near-end E&M—0000 (for T1) or 0001 (for E1) For near-end FXO loop start—0101 For near-end FXO ground start—1111 For near-end FXS—0101 For near-end MELCAS—1101
|
idle transmit
|
For near-end E&M—0000 For near-end FXO—0101 For near-end FXS loop start—0101 For near-end FXS ground start—1111 For near-end MELCAS—1101
|
oos receive
|
For near-end E&M—1111 For near-end FXO loop start—1111 For near-end FXO ground start—0000 For near-end FXS loop start—1111 For near-end FXS ground start—0101 For near-end MELCAS—1111
|
oos transmit
|
No default signaling pattern is defined.
|
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.0(7)XK
|
Default signaling patterns were defined.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
Before configuring the signaling pattern, you must use the voice-class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you define the voice class, you assign it to a dial peer.
Idle Patterns
An idle state is generated if the router detects an idle signaling pattern coming from either direction. If an idle pattern is configured for only one direction (transmit or receive), an idle state can be detected only in the configured direction. Therefore, you should normally enter both the idle receive and the idle transmit keywords.
To suppress voice packets whenever the transmit or receive trunk is in the idle state, use the idle receive and idle transmit keywords in conjunction with the signal timing idle suppress-voice command.
OOS Patterns
An OOS state is generated differently in each direction under the following conditions:
•
If the router detects an oos transmit signaling pattern sent from the PBX, the router transmits the oos transmit signaling pattern to the network.
•
If the signal timing oos timeout timer expires and the router receives no signaling packets from the network (network is OOS), the router sends an oos receive signaling pattern to the PBX. (The oos receive pattern is not matched against the signaling packets received from the network; the receive packets indicate an OOS condition directly by setting the AIS alarm indication bit in the packet.)
To suppress voice packets whenever the transmit or receive trunk is in the OOS state, use the oos receive and oos transmit keywords in conjunction with the signal timing oos suppress-voice command.
To suppress voice and signaling packets whenever the transmit or receive trunk is in the OOS state, use the oos receive and oos transmit keywords in conjunction with the signal timing oos suppress-all command.
PBX Busyout
To "busy out" a PBX if the network connection fails, set the oos receive pattern to match the seized state (busy), and set the signal timing oos timeout value. When the timeout value expires and no signaling packets have been received, the router will send the oos receive pattern to the PBX.
Use the busy seized pattern only if the PBX does not have a specified pattern for indicating an OOS state. If the PBX has a specific OOS pattern, use that pattern instead.
Examples
The following example, beginning in global configuration mode, configures the signaling bit pattern for the idle receive and transmit states:
voice class permanent 10
signal keepalive 3
signal pattern idle receive 0101
signal pattern idle transmit 0101
exit
The following example, beginning in global configuration mode, configures the signaling bit pattern for the out-of-service receive and transmit states:
voice class permanent 10
signal keepalive 3
signal pattern oos receive 0001
signal pattern oos transmit 0001
The following example restores default signaling bit patterns for the receive and transmit idle states:
voice class permanent 10
signal keepalive 3
signal timing idle suppress-voice
no signal pattern idle receive
no signal pattern idle transmit
exit
The following example configures non-default signaling bit patterns for the receive and transmit out-of-service states:
voice class permanent 10
signal keepalive 3
signal pattern oos receive 0001
signal pattern oos transmit 0001
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies a dial-peer type.
|
signal timing idle suppress-voice
|
Specifies the length of time before voice traffic is stopped after a trunk goes into the idle state.
|
signal timing oos
|
Configures the signal timing parameter for the OOS call state.
|
signal timing oos slave-standby
|
Specifies that a slave port return to its initial standby state after the trunk has been OOS for a specified time.
|
signal timing oos suppress-all
|
Stops sending voice and signaling packets to the network if a transmit OOS signaling pattern id detected from the PBX for a specified time.
|
signal timing oos suppress-voice
|
Stops sending voice packets to the network if a transmit OOS signaling pattern is detected from the PBX for a specified time.
|
signal timing oos timeout
|
Changes the delay time between the loss of signaling packets from the network and the start time for the OOS state.
|
voice-class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice class permanent
|
Assigns a previously-configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.
|
signal sequence oos
To specify which signaling pattern is sent to the PBX when the far-end keepalive message is lost or an alarm indication signal (AIS) is received from the far end, use the signal sequence oos command in the voice-class configuration mode. To restore the default value, use the no form of this command.
signal sequence oos {no-action | idle-only | oos-only | both}
no signal sequence oos
Syntax Description
no-action
|
No signaling pattern is sent.
|
idle-only
|
Only the idle signaling pattern is sent.
|
oos-only
|
Only the out-of-service (OOS) signaling pattern is sent.
|
both
|
Both idle and OOS signaling patterns are sent. This is the default value.
|
Defaults
Both idle and OOS signaling patterns are sent.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 multiservice concentrators.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
Before configuring the idle or OOS signaling patterns to be sent, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you assign it to a dial peer.
Use the signal sequence oos command to specify which signaling pattern) to send. Use the signal pattern idle receive or the signal pattern oos receive command to define the bit patterns of the signaling patterns if other than the defaults.
Examples
The following example, beginning in global configuration mode, defines voice class 10, sets the signal sequence oos command to send only the idle signalin pattern to the PBX, and applies the voice class configuration to VoFR dial peer 100.
signal sequence oos idle-only
signal timing idle suppress-voice
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies a dial-peer type.
|
signal pattern
|
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Specifies the length of time before the router stops sending voice packets after a trunk goes into the idle state.
|
signal timing oos
|
Specifies that a permanent voice connection be torn down and restarted after the trunk has been OOS for a specified time.
|
signal timing oos slave-standby
|
Specifies that a slave port return to its initial standby state after the trunk has been OOS for a specified time.
|
signal timing oos suppress-all
|
Configures the router or concentrator to stop sending voice and signaling packets to the network if it detects an OOS signaling pattern from the PBX for a specified time.
|
signal timing oos suppress-voice
|
Configures the router or concentrator to stop sending voice packets to the network if it detects a transmit OOS signaling pattern from the PBX for a specified time.
|
signal timing oos timeout
|
Changes the delay time between the loss of signaling packets from the network and the start time for the OOS state.
|
voice-class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice class permanent
|
Assigns a previously-configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.
|
signal timing idle suppress-voice
To configure the signal timing parameter for the idle state of the call, use the signal timing idle suppress-voice command in voice-class configuration mode. To restore the default value, use the no form of this command.
signal timing idle suppress-voice seconds
no signal timing idle suppress-voice seconds
Syntax Description
seconds
|
Duration of the idle state, in seconds, before the voice traffic is stopped. The valid range is from 0 to 65,535.
|
Defaults
No signal timing idle suppress-voice timer is configured.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.0(7)XK
|
This command was modified to simplify the configuration process.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
Before configuring the signal timing idle suppress-voice timer, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. The voice class must then be assigned to a dial peer.
The signal timing idle suppress-voice command is used when the signal-type command is set to transparent in the dial peer for the Cisco trunk or FRF.11 trunk connection. The router stops sending voice packets when the timer expires. Signaling packets are still sent.
To detect an idle trunk state, the router or concentrator monitors both transmit and receive signaling for the idle transmit and idle receive signaling patterns. These can be configured by the signal pattern idle transmit or signal pattern idle receive command, or they can be the defaults. The default idle receive pattern is the idle pattern of the local voice port. The default idle transmit pattern is the idle pattern of the far-end voice port.
Examples
The following example, beginning in global configuration mode, sets the signal timing idle suppress-voice timer to 5 seconds for the idle state on voice class 10.
signal keepalive 3
signal pattern idle receive 0101
signal pattern idle transmit 0101
signal timing idle suppress-voice 5
The following example defines voice class 10, sets the idle detection time to 5 seconds, configures the trunk to use the default transmit and receive idle signal patterns, and applies the voice class configuration to VoFR dial peer 100.
signal keepalive 3
signal timing idle suppress-voice 5
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice encapsulation.
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Defines the ABCD bit patterns that identify the idle and oos states for Cisco trunks and FRF.11 trunks.
|
signal timing oos
|
Configures the signal timing parameter for the OOS state of a call.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
voice-class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice class permanent (dial peer)
|
Assigns a previously-configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.
|
signal timing oos
To configure the signal timing parameter for the out-of-service (OOS) state of the call, use the signal timing oos command in voice-class configuration mode. To restore the default value, use the no form of this command.
signal timing oos {restart | slave-standby | suppress-all | suppress-voice | timeout} seconds
no signal timing oos {restart | slave-standby | suppress-all | suppress-voice | timeout} seconds
Syntax Description
restart
|
If no signaling packets are received for this period, the permanent voice connection will be torn down and an attempt to achieve reconnection will be made.
|
slave-standby
|
If no signaling packets are received for this period, a slave port returns to its initial standby state. This option applies only to slave ports (ports configured using the connection trunk number answer-mode command).
|
suppress-all
|
If the transmit OOS pattern (from the PBX to the network) matches for this period of time, the router stops sending all packets to the network.
|
suppress-voice
|
If the transmit OOS pattern (from the PBX to the network) matches for this period of time, the router stops sending voice packets to the network. signaling packets continue to be sent with the alarm indication set (AIS).
|
timeout
|
If no signaling packets are received for this period of time, the router sends the configured receive OOS pattern to the PBX. Also, the router stops sending voice packets to the network. Use this option to perform busyout to the PBX.
|
seconds
|
Duration, in seconds, for the above settings. The valid range is from 0 to 65,535.
|
Defaults
No signal timing OOS pattern parameters are configured.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.0(4)T
|
This command was introduced.
|
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. The voice class must then be assigned to a dial peer.
You can enter several values for this command. However, the suppress-all and suppress-voice options are mutually exclusive.
Examples
The following example, beginning in global configuration mode, configures the signal timeout parameter for the OOS state on voice class 10. The signal timing oos timeout command is set to 60 seconds.
signal-keepalive 3
signal pattern oos receive 0001
signal pattern oos transmit 0001
signal timing oos timeout 60
Related Commands
Command
|
Description
|
connection
|
Specifies a connection mode for a voice port.
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice encapsulation.
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Defines the ABCD bit patterns that identify the idle and oos states for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of the call.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
voice class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice-class permanent (dial-peer)
|
Assigns a previously configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.
|
signal timing oos restart
To specify that a permanent voice connection be torn down and restarted after the trunk has been out-of-service (OOS) for a specified time, use the signal timing oos restart command in voice-class configuration mode. To restore the default value, use the no form of this command.
signal timing oos restart seconds
no signal timing oos restart
Syntax Description
seconds
|
Delay duration, in seconds, for the restart attempt. There is no default duration. The range is from 0 to 65,535.
|
Defaults
No restart attempt is made if the trunk becomes OOS.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 series.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. You then assign the voice class to a dial peer.
The signal timing oos restart command is valid only if the signal timing oos timeout command is enabled, which controls the start time for the OOS state. The timer for the signal timing oos restart command does not start until the trunk is OOS.
Examples
The following example, beginning in global configuration mode, creates voice class 10, sets the OOS timeout time to 60 seconds and sets the restart time to 30 seconds:
signal-keepalive 3
signal pattern oos receive 0001
signal pattern oos transmit 0001
signal timing oos timeout 60
signal timing oos restart 30
Related Commands
Command
|
Description
|
connection
|
Specifies a connection mode for a voice port.
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice encapsulation.
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Defines the ABCD bit patterns that identify the idle and oos states for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
voice class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice-class permanent (dial-peer)
|
Assigns a previously-configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.
|
signal timing oos slave-standby
To configure a slave port to return to its initial standby state after the trunk has been out-of-service (OOS) for a specified time, use the signal timing oos slave-standby command in voice-class configuration mode. To restore the default value, use the no form of this command.
signal timing oos slave-standby seconds
no signal timing oos slave-standby
Syntax Description
seconds
|
Delay duration, in seconds. If no signaling packets are received for this period, the slave port returns to its initial standby state. There is no default duration. The range is from 0 to 65,535.
|
Defaults
The slave port does not return to its standby state if the trunk becomes OOS.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 series.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you assign it to a dial peer.
If no signaling packets are received for the specified delay period, the slave port returns to its initial standby state. The signal timing oos slave-standby command is valid only if both of the following conditions are true:
•
The signal timing oos timeout command is enabled, which controls the start time for the OOS state. The timer for the signal timing oos slave-standby command does not start until the trunk is OOS.
•
The voice port is configured as a slave port with the connection trunk digits answer-mode command.
Examples
The following example, beginning in global configuration mode, creates a voice port as a slave voice port, creates voice class 10, sets the OOS timeout time to 60 seconds, and sets the return-to-slave-standby time to 120 seconds:
connection trunk 5559262 answer-mode
signal-keepalive 3
signal pattern oos receive 0001
signal pattern oos transmit 0001
signal timing oos timeout 60
signal timing oos slave-standby 120
Command History
Command
|
Description
|
connection
|
Specifies a connection mode for a voice port.
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice encapsulation.
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Defines the ABCD bit patterns that identify the idle and oos states for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
voice class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice-class permanent (dial-peer)
|
Assigns a previously configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.
|
signal timing oos suppress-all
To configure the router or concentrator to stop sending voice and signaling packets to the network if it detects a transmit out-of-service (OOS) signaling pattern from the PBX for a specified time, use the signal timing oos suppress-all command in voice-class configuration mode. To restore the default value, use the no form of this command.
signal timing oos suppress-all seconds
no signal timing oos suppress-all
Syntax Description
seconds
|
Delay duration, in seconds, before packet transmission is stopped. There is no default duration. The range is from 0 to 65,535.
|
Defaults
The router or concentrator does not stop sending packets to the network if it detects a transmit OOS signaling pattern from the PBX.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 series.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you assign it to a dial peer.
The signal timing oos suppress-all command is valid only if you configure an OOS transmit signaling pattern with the signal pattern oos transmit command. (There is no default oos transmit signaling pattern.)
The signal timing oos suppress-all command is valid whether or not the signal timing oos timeout command is enabled, which controls the start time for the OOS state. The timer for the signal timing oos suppress-all command starts immediately when the OOS transmit signaling pattern is matched.
Examples
The following example, beginning in global configuration mode, creates voice class 10, sets the OOS timeout time to 60 seconds, and sets the packet suppression time to 60 seconds:
signal-keepalive 3
signal pattern oos receive 0001
signal pattern oos transmit 0001
signal timing oos timeout 60
signal timing oos suppress-all 60
Related Commands
Command
|
Description
|
connection
|
Specifies a connection mode for a voice port.
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice encapsulation.
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Defines the ABCD bit patterns that identify the idle and oos states for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
voice class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice-class permanent (dial-peer)
|
Assigns a previously configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.
|
signal timing oos suppress-voice
To configure the router or concentrator to stop sending voice packets to the network if it detects a transmit out-of-service (OOS) signaling pattern from the PBX for a specified time, use the signal timing oos suppress-voice command in voice-class configuration mode. To restore the default value, use the no form of this command.
signal timing oos suppress-voice seconds
no signal timing oos suppress-voice
Syntax Description
seconds
|
Delay duration, in seconds, before voice-packet transmission is stopped. There is no default duration. The range is from 0 to 65,535.
|
Defaults
The router or concentrator does not stop sending voice packets to the network if it detects a transmit OOS signaling pattern from the PBX.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 series.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you assign it to a dial peer.
The signal timing oos suppress-voice command is valid only if you configure an OOS transmit signaling pattern with the signal pattern oos transmit command. (There is no default oos transmit signaling pattern.)
The signal timing oos suppress-voice s command is valid whether or not the signal timing oos timeout command is enabled, which controls the start time for the OOS state. The timer for the signal timing oos suppress-voice command starts immediately when the OOS transmit signaling pattern is matched.
Examples
The following example, beginning in global configuration mode, creates voice class 10, sets the OOS timeout time to 60 seconds, and sets the packet suppression time to 60 seconds:
signal-keepalive 3
signal pattern oos receive 0001
signal pattern oos transmit 0001
signal timing oos timeout 60
signal timing oos suppress-voice 60
Related Commands
Command
|
Description
|
connection
|
Specifies a connection mode for a voice port.
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice encapsulation.
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Defines the ABCD bit patterns that identify the idle and oos states for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
voice class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice-class permanent (dial-peer)
|
Assigns a previously configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.
|
signal timing oos timeout
To change the delay time between the loss of signaling packets from the network and the start time for the out-of-service (OOS) state, use the signal timing oos timeout command in voice-class configuration mode. To restore the default value, use the no form of this command.
signal timing oos timeout [seconds | disabled]
no signal timing oos timeout
Syntax Description
seconds
|
(Optional) Delay duration, in seconds, between the loss of signaling packets and the beginning of the OOS state. The default is 30 seconds. The range is from 1 to 65,535.
|
disabled
|
(Optional) Deactivates the detection of packet loss. If no signaling packets are received from the network, the router does not sent an OOS pattern to the PBX and it continues sending voice packets to the network. Use this option to disable busyout to the PBX.
|
Defaults
No signal timing OOS pattern parameters are configured.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 series.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you assign it to a dial peer.
You can use the signal timing oos timeout command to enable busyout to the PBX.
The signal timing oos timeout command controls the starting time for the signal timing oos restart and signal timing oos slave-standby commands. If this command is entered with the disabled keyword, the signal timing oos restart and signal timing oos slave-standby commands are ineffective.
Examples
The following example, beginning in global configuration mode, creates voice class 10 and sets the OOS timeout time to 60 seconds:
signal-keepalive 3
signal pattern oos receive 0001
signal pattern oos transmit 0001
signal timing oos timeout 60
Related Commands
Command
|
Description
|
connection
|
Specifies a connection mode for a voice port.
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice encapsulation.
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Defines the ABCD bit patterns that identify the idle and oos states for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
voice class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice-class permanent (dial-peer)
|
Assigns a previously configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.
|
signal-type
To set the signaling type to be used when connecting to a dial peer, use the signal-type command in dial-peer configuration mode. To return to the default signal type, use the no form of this command.
signal-type {cas | cept | ext-signal | transparent}
no signal-type
Syntax Description
cas
|
North American EIA-464 channel-associated signaling (robbed bit signaling).
If the Digital T1 Packet Voice Trunk Network Module is installed, this option might not be available.
|
cept
|
Provides a basic E1 ABCD signaling protocol. Used primarily for E&M interfaces. When used with FXS/FXO interfaces, this protocol is equivalent to MELCAS.
|
ext-signal
|
External signaling. The digital signal processor (DSP) does not generate any signaling frames. Use this option when there is an external signaling channel, for example, CCS, or when you need to have a permanent "dumb" voice pipe.
|
transparent
|
On the Cisco MC3810 multiservice concentrator, selecting this option produces different results depending on whether you are using a digital voice module (DVM) or an analog voice module (AVM).
For a DVM: The ABCD signaling bits are copied from or transported through the T1/E1 interface "transparently" without modification or interpretation. This enables the Cisco MC3810 multiservice concentrator to handle arbitrary or unknown signaling protocols.
For an AVM: It is not possible to provide "transparent" behavior because the Cisco MC3810 must interpret the signaling information in order to read and write the correct state to the analog hardware. This option is mapped to be equal to cas.
|
Defaults
cas
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
Support was added for the Cisco 7200 series router.
|
12.0(7)XK
|
In previous releases, the cept and transparent keywords were supported only on the Cisco MC3810 multiservice concentrator. Beginning in this release, these keywords are supported on the Cisco 2600, 3600 and 7200 series routers.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
This command applies to Voice over Frame Relay (VoFR) and Voice over ATM (VoATM) dial peers. It is used with permanent connections only (Cisco trunks and FRF.11 trunks), not with switched calls.
This command is used to inform the local telephony interface of the type of signaling it should expect to receive from the far-end dial peer. To turn signaling off at this dial peer, select the ext-signal option. If signaling is turned off and there are no external signaling channels, a "hot" line exists, enabling this dial peer to connect to anything at the far end.
When you connect an FXS to another FXS, or if you have anything other than an FXS/FXO or E&M/E&M pair, the appropriate signaling type on Cisco 2600 series and 3600 series routers is ext-signal (disabled).
If you have a digital E1 connection at the remote end that is running cept/MELCAS signaling and you then trunk that across to an analog port, you should make sure that you configure both ends for the cept signal type.
If you have a T1 or E1 connection at both ends and the T1/E1 is running a signaling protocol that is neither EIA-464, or cept/MELCAS, you might want to configure the signal type for the transparent option in order to pass through the signaling.
Examples
The following example shows how to disable signaling on a Cisco 2600 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
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a dial peer.
|
connection
|
Specifies the connection mode for a voice port.
|
destination-pattern
|
Specifies the telephone number associated with a dial peer.
|
dtmf-relay
|
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
|
preference
|
Enables the preferred dial peer to be selected when multiple dial peers within a hunt group are matched for a dial string.
|
sequence-numbers
|
Enables the generation of sequence numbers in each frame generated by the DSP.
|
session protocol
|
Establishes the VoFR protocol for calls between local and remote routers.
|
session target
|
Specifies a network-specific address for a dial peer.
|
sip-server
To configure a network address for the Session Initiation Protocol (SIP) server interface, use the sip-server command in SIP user-agent configuration mode.
sip-server {dns:[host-name] | ipv4:ipaddr[:port-num]}
Syntax Description
dns:
|
Sets the global SIP server interface to a Domain Name System (DNS) host name. If you do not specify a host name, the default DNS defined by the ip name-server command is used.
|
host-name
|
(Optional) A valid DNS host name takes the following format: name.gateway.xyz.
|
ipv4:ip_addr
|
Sets the global SIP server interface to an IP address. A valid IP address takes the following format: xxx.xxx.xxx.xxx.
|
:port-num
|
(Optional) Specifies the port number for the SIP server.
|
Defaults
The default for this command is a null value.
Command Modes
SIP user-agent configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco AS5300 universal access servers.
|
Usage Guidelines
If you use this command, you can then specify session target sip-server for each dial peer instead of repeatedly entering the SIP server interface address for each dial peer. To reset this command to a null value, use the default command.
Examples
The following example, beginning in global configuration mode, sets the global SIP server interface to the DNS host name of UA-1-f0.sip.com:
sip-server dns:UA-1-f0.sip.com
Related Commands
Command
|
Description
|
sip-ua
|
Enters SIP user-agent configuration mode, in which you configure the SIP user agent.
|
sip-ua
To enable the Session Initiation Protocol (SIP) user-agent configuration commands, with which you configure the user agent, use the sip-ua command in global configuration mode.
sip-ua
Syntax Description
This command has no arguments or keywords.
Defaults
No default behaviors or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco AS5300 universal access servers.
|
Usage Guidelines
Use the sip-ua command to enter the SIP user-agent configuration mode. Table 71 lists the SIP user-agent configuration mode commands:
Table 71 SIP User-Agent Configuration Mode Commands
Command
|
Description
|
exit
|
Exits SIP user-agent configuration mode.
|
inband-alerting
|
Specifies an inband-alerting SIP header.
|
retry
|
Configures the SIP signaling timers for retry attempts.
|
sip-server
|
Configures a SIP server interface.
|
timers
|
Configures the SIP signaling timers configuration.
|
transport
|
Enables or disables a SIP user agent transport for TCP or UDP, the protocol SIP user agents will be listening for on port 5060 (default).
|
Examples
The following example, beginning in global configuration mode, enters SIP user-agent configuration mode, configures the SIP user agent, then returns to global configuration mode:
sip-server ipv4:10.0.2.254
timers invite-wait-100 500
Related Commands
Command
|
Description
|
exit
|
Exits SIP user-agent configuration mode.
|
inband-alerting
|
Specifies an inband-alerting SIP header.
|
retry
|
Configures the retry attempts for SIP messages.
|
show sip-ua
|
Displays statistics for SIP retries, timers, and current listener status.
|
sip-server
|
Configures the SIP server interface.
|
timers
|
Configures the SIP signaling timers.
|
transport
|
Configures the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket.
|
snmp enable peer-trap poor-qov
To generate poor quality of voice notification for applicable calls associated with Voice over IP (VoIP) dial peers, use the snmp enable peer-trap poor-qov command in dial-peer configuration mode. To disable this notification, use the no form of this command.
snmp enable peer-trap poor-qov
no snmp enable peer-trap poor-qov
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
Usage Guidelines
Use the snmp enable peer-trap poor-qov command to generate poor quality of voice notifications for applicable calls associated with this dial peer. If you have a Simple Network Management Protocol (SNMP) manager that uses SNMP messages when voice quality drops, you might want to enable this command. Otherwise, you should disable this command to reduce unnecessary network traffic.
Examples
The following example enables poor quality of voice notifications for calls associated with VoIP dial peer 10:
snmp enable peer-trap poor-qov
Related Commands
Command
|
Description
|
snmp-server enable traps
|
Enables a router to send SNMP traps and information.
|
snmp trap link-status
|
Enables SNMP trap messages to be generated when a specific port is brought up or down.
|
ss7 mtp2-variant bellcore
To configure the router for Telcordia Technologies (formerly Bellcore) standards, use the ss7 mtp2-variant bellcore command in global configuration mode.
ss7 mtp-variant bellcore [channel] [parameters]
Syntax Description
channel
|
Specifies the channel, 0 through 3.
|
parameters
|
See table below for timer descriptions, defaults, and ranges.
|
Defaults
Bellcore is the default variant if no other is configured.
See Table 72 for default parameters.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
This MTP2 variant has timers and parameters that can be configured using the values listed in Table 72. To restore the designated default, use the no or the default form of the command (see example below).
Note
Timer durations are converted to 10 millisecond units. For example, a T1 value of 1005 is converted to 100 which results in a actual timeout duration of 1000 milliseconds.
This is true for all timers and all variants.
Table 72 Bellcore (Telcordia Technologies) Parameters and Values
Parameter
|
Description
|
Default
|
Range
|
T1
|
aligned/ready timer duration (milliseconds)
|
13000
|
1000 to 65535
|
T2
|
not aligned timer (milliseconds)
|
11500
|
1000 to 65535
|
T3
|
aligned timer (milliseconds)
|
11500
|
1000 to 65535
|
T4-Emergency-Proving
|
emergency proving timer (milliseconds)
|
600
|
1000 to 65535
|
T4-Normal-Proving
|
normal proving period (milliseconds)
|
2300
|
1000 to 65535
|
T5
|
sending SIB timer (milliseconds)
|
100
|
80 to 65535
|
T6
|
remote congestion timer (milliseconds)
|
6000
|
1000 to 65535
|
T7
|
excessive delay timer (milliseconds)
|
1000
|
500 to 65535
|
lssu-len
|
1 or 2 byte LSSU format
|
1
|
1 to 2
|
unacked-MSUs
|
Maximum number of MSUs waiting ACK
|
127
|
16 to 127
|
proving-attempts
|
Maximum number of attempts to prove alignment
|
5
|
3 to 8
|
SUERM-threshold
|
SUERM error rate threshold
|
64
|
32 to 128
|
SUERM-number-octets
|
SUERM octet counting mode
|
16
|
8 to 32
|
SUERM-number-signal- units
|
signal units (good or bad) needed to dec ERM
|
256
|
128 to 512
|
Tie-AERM-Emergency
|
AERM emergency error rate threshold
|
1
|
1 to 8
|
Tie-AERM-Normal
|
AERM normal error rate threshold
|
4
|
1 to 8
|
Examples
The following example sets the aligned/ready timer duration on channel 0 to 30,000 milliseconds:
ss7 mtp2-variant Bellcore 0
The following example restores the aligned/ready timer default value of 13,000 milliseconds:
ss7 mtp2-variant Bellcore 0
Related Commands
Command
|
Description
|
ss7 mtp2-variant itu
|
Specifies the mtp2-variant as ITU.
|
ss7 mtp2-variant ntt
|
Specifies the mtp2-variant as NTT.
|
ss7 mtp2-variant ttc
|
Specifies the mtp2-variant as TTC.
|
ss7 mtp2-variant itu
To configure the router for ITU (International Telecom United) standards, use the ss7 mtp2-variant itu command in global configuration mode.
ss7 mtp-variant itu [channel] [parameters]
Syntax Description
channel
|
Specifies the channel, 0 through 3.
|
parameters
|
See table below for timer descriptions, defaults, and ranges.
|
Defaults
Bellcore is the default variant if no other is configured.
See Table 73 for ITU default parameters.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The ITU MTP2 variant has timers and parameters that can be configured using the values listed in Table 73. To restore the designated default, use the no or the default form of the command (see the example below).
Table 73 ITU (White) Parameters and Values
Parameter
|
Description
|
Default
|
Range
|
T1
|
aligned/ready timer duration (milliseconds)
|
40000
|
1000 to 65535
|
T2
|
not aligned timer (milliseconds)
|
5000
|
1000 to 65535
|
T3
|
aligned timer (milliseconds)
|
1000
|
1000 to 65535
|
T4-Emergency-Proving
|
emergency proving timer (milliseconds)
|
500
|
1000 to 65535
|
T4-Normal-Proving
|
normal proving timer (milliseconds)
|
8200
|
1000 to 65535
|
T5
|
sending SIB timer (milliseconds)
|
100
|
80 to 65535
|
T6
|
remote congestion timer (milliseconds)
|
6000
|
1000 to 65535
|
T7
|
excessive delay timer (milliseconds)
|
1000
|
1000 to 65535
|
lssu-len
|
1 or 2 byte LSSU format
|
1
|
1 to 2
|
msu-len
|
|
|
|
unacked-MSUs
|
Maximum number of MSUs waiting ACK
|
127
|
16 to 127
|
proving-attempts
|
Maximum number of attempts to prove alignment
|
5
|
3 to 8
|
SUERM-threshold
|
SUERM error rate threshold
|
64
|
32 to 128
|
SUERM-number-octets
|
SUERM octet counting mode
|
16
|
8 to 32
|
SUERM-number-signal-units
|
signal units (good or bad) needed to dec ERM
|
256
|
128 to 512
|
Tie-AERM-Emergency
|
AERM emergency error rate threshold
|
1
|
1 to 8
|
Tin-AERM-Normal
|
AERM normal error rate threshold
|
4
|
1 to 8
|
Examples
The following example sets the emergency proving period on channel 1 to 10,000 milliseconds:
t4-Emergency-Proving 10000
The following example restores the emergency proving period default value of 5,000 milliseconds:
default t4-Emergency-Proving
Related Commands
Command
|
Description
|
ss7 mtp2-variant bellcore
|
Specifies the mtp2-variant as Bellcore.
|
ss7 mtp2-variant ntt
|
Specifies the mtp2-variant as NTT.
|
ss7 mtp2-variant ttc
|
Specifies the mtp2-variant as TTC.
|
ss7 mtp2-variant ntt
To configure the router for NTT (Japan) standards, use the ss7 mtp2-variant ntt command in global configuration mode.
ss7 mtp-variant ntt [channel] [parameters]
Syntax Description
channel
|
Specifies the channel, 0 through 3.
|
parameters
|
See table below for timer descriptions, defaults, and ranges.
|
Defaults
Bellcore is the default variant if no other is configured.
See Table 74 for NTT default parameters.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The NTT MTP2 variant has timers and parameters that can be configured using the values listed in Table 74. To restore the designated default, use the no or the default form of the command (see the example below).
Table 74 NTT Parameters and Values
Parameter
|
Description
|
Default
|
Range
|
T1
|
aligned/ready timer duration (milliseconds)
|
15000
|
1000 to 65535
|
T2
|
not aligned timer (milliseconds)
|
5000
|
1000 to 65535
|
T3
|
aligned timer (milliseconds)
|
3000
|
1000 to 65535
|
T4-Emergency- Proving
|
emergency proving timer (milliseconds)
|
3000
|
1000 to 65535
|
T5
|
sending SIB timer (milliseconds)
|
200
|
80 to 65535
|
T6
|
remote congestion timer (milliseconds)
|
2000
|
1000 to 65535
|
T7
|
excessive delay timer (milliseconds)
|
3000
|
1000 to 65535
|
TA
|
SIE interval timer (milliseconds)
|
20
|
10 to 500
|
TF
|
FISU interval timer (milliseconds)
|
20
|
10 to 500
|
TO
|
SIO interval timer (milliseconds)
|
20
|
10 to 500
|
TS
|
SIOS interval timer (milliseconds)
|
20
|
10 to 500
|
unacked-MSUs
|
Maximum number of MSUs waiting ACK
|
40
|
16 to 40
|
proving-attempts
|
Maximum number of attempts to prove alignment
|
5
|
3 to 8
|
SUERM-threshold
|
SUERM error rate threshold
|
64
|
32 to 128
|
SUERM-number-octets
|
SUERM octet counting mode
|
16
|
8 to 32
|
SUERM-number-signal-units
|
signal units (good or bad) needed to dec ERM
|
256
|
128 to 512
|
Tie-AERM-Emergency
|
AERM emergency error rate threshold
|
1
|
1 to 8
|
Examples
The following example sets the SUERM error rate threshold on channel 2 to 100:
The following example restores the SUERM error rate threshold default value of 64:
Related Commands
Command
|
Description
|
ss7 mtp2-variant bellcore
|
Specifies the mtp2-variant as Bellcore.
|
ss7 mtp2-variant itu
|
Specifies the mtp2-variant as ITU.
|
ss7 mtp2-variant ttc
|
Specifies the mtp2-variant as TTC.
|
ss7 mtp2-variant ttc
To configure the router for TTC (Japan Telecom) standards, use the ss7 mtp2-variant ttc command in global configuration mode.
ss7 mtp-variant ttc [channel] [parameters]
Syntax Description
channel
|
Specifies the channel, 0 through 3.
|
parameters
|
See table below for timer descriptions, defaults, and ranges.
|
Defaults
Bellcore is the default variant if no other is configured.
See Table 75 for TTC default parameters.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The TTC MTP2 variant has timers and parameters that can be configured using the values listed in Table 75. To restore the designated default, use the no or the default form of the command (see the example below).
Table 75 TTC Parameters and Values
Parameter
|
Description
|
Default
|
Range
|
T1
|
aligned/ready timer duration (milliseconds)
|
15000
|
1000 to 65535
|
T2
|
not aligned timer (milliseconds)
|
5000
|
1000 to 65535
|
T3
|
aligned timer (milliseconds)
|
3000
|
1000 to 65535
|
T4-Emergency-Proving
|
emergency proving timer (milliseconds)
|
3000
|
1000 to 65535
|
T5
|
sending SIB timer (milliseconds)
|
200
|
80 to 65535
|
T6
|
remote congestion timer (milliseconds)
|
2000
|
1000 to 65535
|
T7
|
excessive delay timer (milliseconds)
|
3000
|
1000 to 65535
|
TA
|
SIE interval timer (milliseconds)
|
20
|
10 to 500
|
TF
|
FISU interval timer (milliseconds)
|
20
|
10 to 500
|
TO
|
SIO interval timer (milliseconds)
|
20
|
10 to 500
|
TS
|
SIOS interval timer (milliseconds)
|
20
|
10 to 500
|
unacked-MSUs
|
Maximum number of MSUs waiting ACK
|
40
|
16 to 40
|
proving-attempts
|
Maximum number of attempts to prove alignment
|
5
|
3 to 8
|
SUERM-threshold
|
SUERM error rate threshold
|
64
|
32 to 128
|
SUERM-number-octets
|
SUERM octet counting mode
|
16
|
8 to 32
|
SUERM-number-signal-units
|
signal units (good or bad) needed to dec ERM
|
256
|
128 to 512
|
Tie-AERM-Emergency
|
AERM emergency error rate threshold
|
1
|
1 to 8
|
Examples
The following example sets the maximum number of proving attempts for channel 3 to 3:
The following example restores the maximum number of proving attempts to the default value:
Related Commands
Command
|
Description
|
ss7 mtp2-variant bellcore
|
Specifies the mtp2-variant as Bellcore.
|
ss7 mtp2-variant itu
|
Specifies the mtp2-variant as ITU.
|
ss7 mtp2-variant ntt
|
Specifies the mtp2-variant as NTT.
|
ss7 session
To create a Reliable User Datagram Protocol (RUDP) session, use the ss7 session command in global configuration mode. To delete the session, use the no form of this command.
ss7 session-session number {address remote-address remote-port local-address local-port}
no ss7session-session number address
Syntax Description
session-number
|
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
|
remote-address
|
The remote IP address of the Media Gateway Controller in four-part dotted-decimal format.
|
remote-port
|
The number of the remote UDP port on which the Media Gateway Controller is configured to listen. This UDP port cannot be used by another protocol as defined in RFC 1700 and cannot be otherwise used in the network.
|
local-address
|
The local IP address of the router in four-part dotted-decimal format.
The local IP address for both sessions, 0 and 1, must be the same.
|
local-port
|
The number of the local UDP port on which the router expects to receive messages from the Media Gateway Controller. Specify any UDP port that is not used by another protocol as defined in RFC 1700 and that is not otherwise used in your network.
The local UDP port must be different for session-0 and session-1.
|
Defaults
No session is configured.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
You can configure a maximum of two sessions, one for each signaling link. In a redundant Media Gateway Controller configuration, session-0 is configured to one MGC and session-1 is configured to the other.
The Media Gateway Controller must be configured to send messages to the local port, and it must be configured to listen on the remote port.
You must reload the router whenever you remove a session or change the parameters of a session.
Examples
The following example sets up two sessions on a Cisco 2611:
ss7 session-0 address 255.251.255.255 7000 255.255.255.254 7000
ss7 session-1 address 255.255.255.252 7002 255.255.255.254 7001
Note
The example above shows how the local IP addresses in session-0 and session-1 must be the same.
Related Commands
Command
|
Description
|
ss7 session retrans_t
|
Sets the retransmission timer.
|
ss7 session m_retrans
|
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
|
ss7 session m_rcvnum
|
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
|
ss7 session m_outseq
|
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
|
ss7 session m_cumack
|
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
|
ss7 session k_pt
|
Sets the null segment (keepalive) timer.
|
ss7 session cumack_t
|
Sets the cumulative acknowledgment timer.
|
ss7 session cumack_t
To set the Reliable User Datagram Protocol (RUDP) cumulative acknowledgment timer for a specific SS7 signaling link session, use the ss7 session cumack_t command in global configuration mode. To restore the default value, use the no form of this command.
ss7 session-session number cumack_t milliseconds
no ss7 session-session number cumack_t
Caution 
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
Syntax Description
session-number
|
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
|
milliseconds
|
Use this parameter to specify the amount of time (in milliseconds) that the RUDP waits before it sends an acknowledgment after receiving a segment.
Valid values are from 100 to 65535. This value should be less than the value configured for the retransmission timer by using the ss7 session-session number retrans_t command.
|
Defaults
The default value is 300 milliseconds.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The cumulative acknowledgment timer determines when the receiver sends an acknowledgment. If the timer is not already running, it is initialized when a valid data, null, or reset segment is received. When the cumulative acknowledgment timer expires, the last in-sequence segment is acknowledged. The RUDP typically tries to "piggyback" acknowledgments on data segments being sent. However, if no data segment is sent in this period of time, it sends a standalone acknowledgment.
Examples
The following example sets up two sessions and sets the cumulative acknowledgment timer to 320 milliseconds for each one:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7000
ss7 session-0 cumack_t 320
ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7001
ss7 session-1 cumack_t 320
Related Commands
Command
|
Description
|
ss7 session retrans_t
|
Sets the retransmission timer.
|
ss7 session m_retrans
|
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
|
ss7 session m_rcvnum
|
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
|
ss7 session m_outseq
|
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
|
ss7 session m_cumack
|
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
|
ss7 session k_pt
|
Sets the null segment (keepalive) timer.
|
show ss7
|
Displays the SS7 configuration.
|
ss7 session kp_t
To set the null segment (keepalive) timer for a specific SS7 signaling link session, use the ss7 session kp_t command in global configuration mode. To restore the default value, use the no form of this command.
ss7 session-session number kp_t milliseconds
no ss7 session-session number kp_t
Caution 
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
Syntax Description
session-number
|
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
|
milliseconds
|
Use this parameter to specify the amount of time (in milliseconds) that the Reliable User Datagram Protocol (RUDP) waits before sending a keepalive to verify that the connection is still active.
Valid values are 0 and from100 to 65535.
|
Defaults
The default value is 2000 milliseconds.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The null segment timer determines when a null segment (keepalive) is sent by the client Cisco 2600 series router. On the client, the timer starts when the connection is established and is reset each time a data segment is sent. If the null segment timer expires, the client sends a keepalive to the server to verify that the connection is still functional. On the server, the timer restarts each time a data or null segment is received from the client.
The value of the server's null segment timer is twice the value configured for the client. If no segments are received by the server in this period of time, the connection is no longer valid.
To disable keepalive, set this parameter to 0.
Examples
The following example sets up two sessions and sets a keepalive of 1,800 milliseconds for each one:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7000
ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7001
Related Commands
Command
|
Description
|
ss7 session retrans_t
|
Sets the retransmission timer.
|
ss7 session m_retrans
|
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
|
ss7 session m_rcvnum
|
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
|
ss7 session m_outseq
|
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
|
ss7 session m_cumack
|
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
|
ss7 session cumack_t
|
Sets the cumulative acknowledgment timer.
|
show ss7
|
Displays the SS7 configuration.
|
ss7 session m_cumack
To set the maximum number of segments that can be received before the Reliable User Datagram Protocol (RUDP) sends an acknowledgment in a specific SS7 signaling link session, use the ss7 session m_cumack command in global configuration mode. To restore the default value, use the no form of this command.
ss7 session-session number m_cumack segments
no ss7 session-session number m_cumack
Caution 
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
Syntax Description
session-number
|
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
|
segments
|
Use this parameter to specify maximum number of segments that can be received before the Reliable User Datagram Protocol (RUDP) sends an acknowledgment.
Valid values are from 0 to 255.
|
Defaults
The default value is 3 segments.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The cumulative acknowledgment counter records the number of unacknowledged, in-sequence data, null, or reset segments received without a data, null, or reset segment being sent to the transmitter. If this counter reaches the configured maximum, the receiver sends a standalone acknowledgment (a standalone acknowledgment is a segment that contains only acknowledgment information). The standalone acknowledgment contains the sequence number of the last data, null, or reset segment received.
If you set this parameter to 0, an acknowledgment is sent immediately after a data, null, or reset segment is received.
Examples
The following example sets up two sessions and in each session sets a maximum of two segments for receipt before acknowledgment:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001
ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000
Related Commands
Command
|
Description
|
ss7 session retrans_t
|
Sets the retransmission timer.
|
ss7 session m_retrans
|
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
|
ss7 session m_rcvnum
|
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
|
ss7 session m_outseq
|
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
|
ss7 session k_pt
|
Sets the null segment (keepalive) timer.
|
ss7 session cumack_t
|
Sets the cumulative acknowledgment timer.
|
show ss7
|
Displays the SS7 configuration.
|
ss7 session m_outseq
To set the maximum number of out-of-sequence segments that can be received before the Reliable User Datagram Protocol (RUDP) sends an extended acknowledgment in a specific SS7 signaling link session, use the ss7 session m_outseq command in global configuration mode. To restore the default value, use the no form of this command.
ss7 session-session number m_outseq segments
no ss7 session-session number m_outseq
Caution 
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
Syntax Description
session-number
|
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
|
segments
|
Use this parameter to specify the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment. If the specified number of segments are received out of sequence, an Extended Acknowledgment segment is sent to inform the sender which segments are missing.
Valid values are from 0 to 255.
|
Defaults
The default value is 3 segments.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The out-of-sequence acknowledgment counter records the number of data segments that have arrived out of sequence. If this counter reaches the configured maximum, the receiver sends an extended acknowledgment segment that contains the sequence numbers of the out-of-sequence data, null, and reset segments received. When the transmitter receives the extended acknowledgment segment, it retransmits the missing data segments.
If you set this parameter to 0, an acknowledgment is sent immediately after an out-of-sequence segment is received.
Examples
The following example sets up two sessions and sets a maximum number of four out-of-sequence segments for each session:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001
ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000
Related Commands
Command
|
Description
|
ss7 session retrans_t
|
Sets the retransmission timer.
|
ss7 session m_retrans
|
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
|
ss7 session m_rcvnum
|
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
|
ss7 session m_cumack
|
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
|
ss7 session k_pt
|
Sets the null segment (keepalive) timer.
|
ss7 session cumack_t
|
Sets the cumulative acknowledgment timer.
|
show ss7
|
Displays the SS7 configuration.
|
ss7 session m_rcvnum
To set the maximum number of segments that the remote end can send before receiving an acknowledgment in a specific SS7 signaling link session, use the ss7 session m_rcvnum command in global configuration mode. To restore the default value, use the no form of this command.
ss7 session-session number m_rcvnum segments
no ss7 session-session number m_rcvnum
Caution 
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
Syntax Description
session-number
|
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
|
segments
|
Use this parameter to specify the maximum number of segments that the remote (Cisco IOS software) end can send before receiving an acknowledgment.
Valid values are from 1 to 64.
|
Defaults
The default value is 32 segments.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The outstanding segments counter is the maximum number of segments that the Cisco IOS software end of the connection can send without getting an acknowledgment from the receiver. The receiver uses the counter for flow control.
Examples
The following example sets up two sessions and for each session sets a maximum of 36 segments for receipt before an acknowledgment:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001
ss7 session-0 m_rcvnum 36
ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000
ss7 session-1 m_rcvnum 36
Related Commands
Command
|
Description
|
ss7 session retrans_t
|
Sets the retransmission timer.
|
ss7 session m_retrans
|
Sets the maximum number of times that the Reliable User Datagram Protocol (RUDP) attempts to resend a segment before declaring the connection invalid.
|
ss7 session m_outseq
|
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
|
ss7 session m_cumack
|
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
|
ss7 session k_pt
|
Sets the null segment (keepalive) timer.
|
ss7 session cumack_t
|
Sets the cumulative acknowledgment timer.
|
show ss7
|
Displays the SS7 configuration.
|
ss7 session m_retrans
To set the maximum number of times that the Reliable User Datagram Protocol (RUDP) attempts to resend a segment before declaring the connection invalid in a specific SS7 signaling link session, use the ss7 session m_retrans command in global configuration mode. To restore the default value, use the no form of this command.
ss7 session-session number m_retrans number
no ss7 session-session number m_retrans
Caution 
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
Syntax Description
session-number
|
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
|
number
|
Use this parameter to specify the maximum number of times that the RRUDP attempts to resend a segment before declaring the connection broken.
Valid values are from 0 to 255.
|
Defaults
The default value is 2 times.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The retransmission counter is the number of times a segment has been retransmitted. If this counter reaches the configured maximum, the transmitter resets the connection and informs the upper-layer protocol.
If you set this parameter to 0, the RUDP attempts to resend the segment continuously.
Examples
The following example sets up two sessions and for each session sets a maximum number of three times to resend before a session becomes invalid:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001
ss7 session-0 m_retrans 3
ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000
ss7 session-1 m_retrans 3
Related Commands
Command
|
Description
|
ss7 session retrans_t
|
Sets the retransmission timer.
|
ss7 session m_rcvnum
|
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
|
ss7 session m_outseq
|
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
|
ss7 session m_cumack
|
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
|
ss7 session k_pt
|
Sets the null segment (keepalive) timer.
|
ss7 session cumack_t
|
Sets the cumulative acknowledgment timer.
|
show ss7
|
Displays the SS7 configuration.
|
ss7 session retrans_t
To set the amount of time that the Reliable User Datagram Protocol (RUDP) waits to receive an acknowledgment for a segment in a specific SS7 signaling link session, use the ss7 session retrans_t command in global configuration mode. If the RUDP does not receive the acknowledgment in this time period, the RUDP retransmits the segment. To restore the default value, use the no form of this command.
ss7 session-session number retrans_t milliseconds
no ss7 session-session number retrans_t
Caution 
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
Syntax Description
session-number
|
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
|
milliseconds
|
Use this parameter to specify the amount of time that the RUDP waits to receive an acknowledgment for a segment.
Valid values are from 100 to 65535.
|
Defaults
The default value is 600 milliseconds.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The retransmission timer is used to determine whether a packet must be retransmitted and is initialized each time a data, null, or reset segment is sent. If an acknowledgment for the segment is not received by the time the retransmission timer expires, all segments that have been transmitted—but not acknowledged—are retransmitted.
This value should be greater than the value configured for the cumulative acknowledgment timer by using the ss7 session cumack_t command.
Examples
The following example sets up two sessions and specifies 550 milliseconds as the time to wait for an acknowledgment for each session:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001
ss7 session-0 retrans_t 550
ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000
ss7 session-1 retrans_t 550
Related Commands
Command
|
Description
|
ss7 session m_retrans
|
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
|
ss7 session m_rcvnum
|
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
|
ss7 session m_outseq
|
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
|
ss7 session m_cumack
|
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
|
ss7 session k_pt
|
Sets the null segment (keepalive) timer.
|
ss7 session cumack_t
|
Sets the cumulative acknowledgment timer.
|
show ss7
|
Displays the SS7 configuration.
|
ss7 set failover-timer
To specify the amount of time that the SS7 Session Manager waits for the active session to recover or for the standby Media Gateway Controller to indicate that the SLT should switch traffic to the standby session, use the ss7 set failover-timer command in global configuration mode. To restore the default setting, use the no form of this command.
ss7 set failover-timer [seconds]
no ss7 set failover-timer
Syntax Description
seconds
|
Time in seconds that the Session Manager waits for a session to recover. Values from 1 through 10 are valid.
|
Defaults
The default is 3 seconds.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
This command specifies the number of seconds that the Session Manager waits for the the active session to recover or for the standby Media Gateway Controller to indicate that the SLT should switch traffic to the standby session and to make that session the active session. If the timer expires without a recovery of the original session or an active message from the standby Media Gateway Controller, the signaling links are taken out of service.
Examples
The following example sets the failover timer to four seconds:
Related Commands
Command
|
Description
|
show ss7 sm set
|
Displays the current failover timer setting.
|
ss7 session
|
Establishes a session.
|
station-id
To specify the name or number that will be sent as Caller-ID information and enable Caller-ID, use the station-id voice-port configuration command at the sending Foreign Exchange Station (FXS) voice port or at a Foreign Exchange Office (FXO) port through which routed Caller-ID calls pass. To remove the name or number, use the no form of this command.
station-id [name name | number number]
no station-id [name name | number number]
Syntax Description
name
|
A string of 1 to 15 characters to represent the station name.
|
number
|
A string of from 1 to 15 characters to represent the station number.
|
Defaults
The default is no station name or number.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.1(2)XH
|
This command was implemented for Cisco MC3810 and for Cisco 2600 and 3600 series routers.
|
12.1(3)T
|
This command was first supported on the T Train.
|
Usage Guidelines
This optional command is configured on FXS voice ports that are used to originate on-net calls. The information entered is displayed by the telephone attached to the FXS port at the far end of the on-net call. It can also be configured on the FXO port of a router on which Caller ID information is expected to be received from the CO, to suit situations where a call is placed from the CO, then goes through the FXO interface, and continues to a far-end FXS port through an on-net call. In this case, if no Caller ID information is received from the CO telephone line, the far-end call recipient receives the information configured on the FXO port.
Note
This feature applies only to Caller ID name display provided by an FXS port connection to a telephone device. The station name will not be passed through telephone trunk connections supporting Automatic Number Identification (ANI) calls. ANI supplies calling number identification only and does not support calling number names.
Do not use this command when the Caller-ID standard is dual-tone multifrequency (DTMF). DTMF Caller ID can carry only the calling number.
If the station-id or caller-id alerting command is configured on the voice port, these automatically enable Caller-ID, and the caller-id enable command is not necessary.
This command applies to the Cisco MC3810 and to Cisco 2600 and 3600 series routers.
Examples
The following example configures a Cisco 2600 or 3600 series router voice port from which Caller-ID information is sent:
station-id name A. Person
station-id number 4085551111
The following example configures a Cisco MC3810 voice port from which Caller-ID information is sent:
station-id name A. Person
station-id number 4085551111
caller-id alerting ring 1
Related Commands
Command
|
Description
|
caller-id enable
|
Enables Caller-ID operation.
|
subcell-mux
To enable subcell multiplexing on a Cisco MC3810 multiservice concentrator, use the subcell-mux command in voice-service configuration mode. To restore the default value, use the no form of the command.
subcell-mux
no subcell-mux
Syntax Description
This command has no arguments or keywords.
Defaults
Subcell multiplexing is not enabled.
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.1(1)XA
|
The command was introduced for the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
Use this command to enable ATM adaptation layer 2 (AAL2) common part sublayer (CPS) subcell multiplexing when the Cisco MC3810 interoperates with other equipment that uses subcell multiplexing.
Examples
The following example enables AAL2 CPS subcell multiplexing on a Cisco MC3810 multiservice concentrator:
voice service voatm
session protocol aal2
supervisory disconnect
To enable a supervisory disconnect signal on Foreign Exchange Office (FXO) ports, use the supervisory disconnect command in voice-port configuration mode. To disable the supervisory disconnect signal, use the no form of this command.
supervisory disconnect
no supervisory disconnect
Syntax Description
This command has no arguments or keywords.
Defaults
Enabled
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
This command indicates whether supervisory disconnect signaling is available on the FXO port. Supervisory disconnect signaling is a power denial from the switch lasting at least 350 milliseconds. When this condition is detected, the system interprets this as a disconnect indication from the switch and clears the call.
You should configure no supervisory disconnect on the voice port if there is no supervisory disconnect available from the switch.
Note
If there is no disconnect supervision on the voice port, the interface could be left active if the caller abandons the call before the far end answers. After the router collects the dialed digits but before the called party answers, the router starts a tone detector. Within this time window, the tone detector listens for signals (such as a fast busy signal) that occur if the originating caller hangs up. If this occurs, the router will interpret those tones as a disconnect indication and close the window.
Examples
The following example configures supervisory disconnect on a Cisco 3600 series voice port:
The following example configures supervisory disconnect on a Cisco MC3810 multiservice concentrator voice port:
supervisory disconnect anytone
To configure an Foreign Exchange Office (FXO) voice port to go on-hook if the router detects any tone from a PBX or public switched telephone network (PSTN) before the call is answered, use the supervisory disconnect anytone command in voice-port configuration mode. To restore the default, use the no form of this command.
supervisory disconnect anytone
no supervisory disconnect anytone
Syntax Description
This command has no arguments or keywords.
Defaults
The supervisory disconnect function is not enabled on voice ports.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600, 3600, and MC3810 series.
|
Usage Guidelines
The supervisory disconnect anytone voice-port configuration command can be used to provide the disconnect function if the PBX or PSTN does not provide a supervisory tone. This function is enabled only during call setup (before the call is answered); examples of tones that trigger a disconnect include busy tone, fast busy tone, and dial tone. You must enable echo cancellation; otherwise, the router's own ringback tone can trigger a disconnect.
This command replaces the no supervisory disconnect signal command. If you enter the no supervisory disconnect signal command, the supervisory disconnect any-tone feature will be enabled, and "supervisory disconnect" anytone will be displayed when show commands are entered.
Examples
The following example configures voice ports 1/4 and 1/5 to go on-hook if any tone from the PBX or PSTN is detected before the call is answered:
supervisory disconnect anytone
supervisory disconnect anytone
The following example disables the disconnect function on voice port 1/5:
no supervisory disconnect anytone
Related Commands
Command
|
Description
|
voice class dualtone
|
Creates a voice class for FXO tone detection parameters.
|
supervisory disconnect dualtone voice-class
To assign a previously configured voice class for Foreign Exchange Office (FXO) supervisory disconnect tone to a voice port, use the supervisory disconnect dualtone voice-class command in voice port configuration mode. To remove a voice class from a voice-port, use the no form of this command.
supervisory disconnect dualtone {mid-call | pre-connect} voice-class tag
no supervisory disconnect dualtone voice-class tag
Syntax Description
mid-call
|
Configures tone detection to operate throughout the duration of the call.
|
pre-connect
|
Configures tone detection to operate during call setup and to stop when the called telephone goes off-hook.
|
tag
|
A unique identification number assigned to one voice class. The tag number maps to the tag number assigned using the voice class dualtone global configuration command. The range is from 1 to 10,000.
|
Defaults
No voice class is assigned to a voice port.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600, 3600, and MC3810 series.
|
Usage Guidelines
You can apply an FXO supervisory disconnect tone voice class to multiple voice ports. You can assign only one FXO supervisory disconnect tone voice class to a voice port. If a second voice class is assigned to a voice port, the second voice class replaces the one previously assigned. You cannot assign separate FXO supervisory disconnect tone commands directly to the voice port.
This feature is applicable to analog FXO voice ports with loop-start signaling.
Examples
The following example assigns voice class 70 to FXO voice port 1/5 of a Cisco MC3810 series concentrator and specifies tone detection during the entire call duration:
supervisory disconnect dualtone mid-call voice-class 70
The following example assigns voice class 80 to FXO voice port 0/1/1 of a Cisco 3600 series router and specifies tone detection only during call setup:
supervisory disconnect dualtone pre-connect voice-class 80
Related Commands
Command
|
Description
|
channel-group
|
Defines the time slots of each T1 or E1 circuit.
|
mode
|
Sets the mode of the T1/E1 controller and enters specific configuration commands for each mode type in VoATM.
|
voice class dualtone
|
Creates a voice class for FXO tone detection parameters.
|
tdm-group
To configure a list of time slots for creating clear channel groups (pass-through) for time-division multiplexing (TDM) cross-connect, use the tdm-group command in controller configuration mode. To delete a clear channel group, use the no form of this command.
tdm-group tdm-group-no timeslot timeslot-list [type {e&m | fxs [loop-start | ground-start] | fxo
[loop-start | ground-start] | fxs-melcas | fxo-melcas | e&m-melcas}]
no tdm-group tdm-group-no timeslot timeslot-list [type {e&m | fxs [loop-start | ground-start] |
fxo [loop-start | ground-start] | fxs-melcas | fxo-melcas | e&m-melcas}]
Syntax Description
tdm-group-no
|
TDM group number.
|
timeslot
|
Time-slot number.
|
timeslot-list
|
Time-slot list. The valid range is from 1 to 24 for T1, and from 1 to 15 and 17 to 31 for E1.
|
type
|
(Optional) (Valid only when the mode cas command is enabled.) Specifies the voice signaling type of the voice port. If configuring a TDM group for data traffic only, do not specify the type keyword.
Choose from one of the following options:
• e&m—for E&M signaling
• fxs—for Foreign Exchange Station signaling (optionally, you can also specify loop-start or ground-start)
• fxo—for Foreign Exchange Office signaling (optionally, you can also specify loop-start or ground-start)
• fxs-melcas—for Foreign Exchange Station MEL CAS
• fxo-melcas—for Foreign Exchange Office MEL CAS
• e&m-melcas—for E&M Mercury Exchange Limited Channel-Associated signaling (MEL CAS)
The MELCAS options apply only to E1 lines and are used primarily in the United Kingdom.
|
Defaults
No TDM group is configured.
Command Modes
Controller configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on Cisco MC38310 multiservice concentrators.
|
12.1(1)T
|
This command was modified to include voice WAN interface cards (VWICs) for Cisco 2600 and Cisco 3600 series routers.
|
12.1(2)T
|
This command was modified for the OC-3/STM-1 ATM Circuit Emulation Service network module on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
The tdm-group command allows specific timeslots to switch from port 0 to port 1 and vice versa. This command is similar to the channel-group command, but it does not create a serial interface to terminate the specified channels.
Note
Channel groups, CAS voice groups, and TDM groups all use group numbers. All group numbers configured for channel groups, CAS voice groups, and TDM groups must be unique on the local router. For example, you cannot use the same group number for a channel group and for a TDM group.
Examples
The following example shows TDM group 1 being set up to include timeslots 13 through 20:
tdm-group 1 timeslots 13-20
The following example configures TDM group number 20 on controller T1 1 to support FXO ground-start:
tdm-group 20 timeslot 20 type fxs ground-start
Related Commands
Command
|
Description
|
connect
|
Starts passage of data between ports for cross-connect TDM.
|
tech-prefix
To specify that a particular technology prefix be prepended to the destination pattern of a specific dial peer, use the tech-prefix command in dial-peer configuration mode. To disable the defined technology prefix for this dial peer, use the no form of this command.
tech-prefix number
no tech-prefix number
Syntax Description
number
|
Defines the numbers used as the technology prefix. Each technology prefix can contain up to 11 characters. Although not strictly necessary, a pound (#) symbol is frequently used as the last character in a technology prefix. Valid characters are 0 though 9, the pound (#) symbol, and the asterisk (*).
|
Defaults
No technology prefix is defined.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(6)NA2
|
This command was introduced on Cisco 2500 and 3600 series routers.
|
Usage Guidelines
Technology prefixes are used to distinguish between gateways that have specific capabilities within a given zone. In the exchange between the gateway and the gatekeeper, the technology prefix is used to select a gateway after the zone has been selected. Use the tech-prefix command to define technology prefixes.
Technology prefixes can be used as a discriminator so that the gateway can tell the gatekeeper that a certain technology is associated with a particular call (for example, 15# could mean a fax transmission), or a technology prefix can be used like an area code for more generic routing. No standard defines what the numbers in a technology prefix mean; by convention, technology prefixes are designated by a pound (#) symbol as the last character.
In most cases, there is a dynamic protocol exchange between the gateway and the gatekeeper that enables the gateway to inform the gatekeeper about technology prefixes and where to forward calls. If, for some reason, that dynamic registry feature is not in effect, you can statically configure the gatekeeper to query the gateway for this information by configuring the gw-type-prefix command on the gatekeeper. Use the show gatekeeper gw-type-prefix command to display how the gatekeeper has mapped the technology prefixes to local gateways.
Note
Cisco gatekeepers use the asterisk (*) as a reserved character. If you are using Cisco gatekeepers, do not use the asterisk as part of the technology prefix.
Examples
The following example defines a technology prefix of 14# for the specified dial peer. In this example, the technology prefix means that the H.323 gateway will ask the RAS gatekeeper to direct calls using the technology prefix of 14#.
destination-pattern 14...
Related Commands
Command
|
Description
|
gw-type-prefix
|
Configures a technology prefix in the gatekeeper.
|
show gatekeeper gw-type-prefix
|
Displays the gateway technology prefix table.
|
test call fallback probe
To test a probe to a particular IP address and display the Calculated Planning Impairment Factor (ICPIF) response time reporter (RTR) values, use the test call fallback probe command in EXEC mode. This command has no impact on the cache.
test call fallback probe ip-address [codec 711/729]
Syntax Description
ip-address
|
Specifies the target IP address.
|
codec 711/729
|
(Optional) Specifies a specific codec type.
|
Defaults
This command is not configured by default.
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced.
|
Examples
The following example demonstrates a test probe to IP address 10.1.1.4, and shows that the ICPIF value to 10.1.1.4 is 0:
Router# test call fallback probe 10.1.1.4
Running a test RTR probe....
ICPIF value for the test probe is 0
Related Commands
Command
|
Description
|
call fallback active
|
Enables fallback to alternate dial peers in case of network congestion.
|
call fallback monitor
|
Enables the monitoring of destinations without fallback to alternate dial peers.
|
test pots dial
To dial a telephone number for the plain old telephone service (POTS) port on the router by using a dial application on your workstation, use the test pots dial command in EXEC mode.
test pots port dial number[#]
Syntax Description
port
|
Port number 1 or 2.
|
number
|
Telephone number to dial.
|
#
|
(Optional) Turns off dual tone multifrequency (DTMF) detection from the telephone while sending the enbloc signal. If you do not include the pound sign character (#) to terminate the number variable, you can use the telephone keypad to complete the call.
|
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(2)XF
|
The command test pots port dial was introduced on the Cisco 800 series routers.
|
Usage Guidelines
If the telephone is on the hook when you issue the dial command, the router rings the telephone, waits until the telephone is taken off the hook, and then dials the requested number. If the telephone is off the hook and providing a dial tone when you issue the command, the router dials the requested number.
Examples
The following POTS dial command dials the telephone number 4085551234:
Router# test pots 1 dial 4085551234#
For an example of the test pots port dial command with debug output, see the debug pots csm command in the Cisco IOS Debug Command Reference, Release 12.2.
Related Commands
Command
|
Description
|
show pots csm
|
Displays the current state of calls and the most recent event received by the CSM on the router.
|
test pots disconnect
|
Disconnects a telephone call for the POTS port on the router.
|
test pots disconnect
To disconnect a telephone call for the POTS port on the router, use the test pots disconnect command in EXEC mode.
test pots port disconnect
Syntax Description
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(2)XF
|
This command was introduced on the Cisco 800 series routers.
|
Examples
The following POTS disconnect command disconnects a telephone call from POTS port 1:
Router# test pots 1 disconnect
For an example of the test pots port disconnect command with debug output, see the debug pots csm command in the Cisco IOS Debug Command Reference, Release 12.2.
Related Commands
Command
|
Description
|
show pots csm
|
Displays the current state of calls and the most recent event received by the CSM on the router.
|
test pots dial
|
Dials a telephone number for the POTS port on the router by using a dial application on your workstation.
|
test translation-rule
To test the execution of the translation rules on a specific name tag, use the test translation-rule command in global configuration mode. To disable the test, use the no form of this command.
test translation-rule name-tag input-number [input-numbering-type]
no test translation-rule name-tag input-number [input-numbering-type]
Syntax Description
name-tag
|
The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. The range is 1 through 2,147,483,647.
|
input-number
|
The input string of digits for which a pattern matching is performed.
|
input-numbering-type
|
(Optional) The keyword choices for this field are international, national, subscriber, abbreviated, unknown, and any.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR1
|
This command was introduced for Voice over IP on the Cisco AS5300.
|
12.0(7)XK
|
This command was first supported for the following voice technologies on the following platforms:
• Voice over IP (Cisco 2600 series, Cisco 3600 series, Cisco MC3810 series)
• Voice over Frame Relay (Cisco 2600 series, Cisco 3600 series, Cisco MC3810 multiservice concentrator)
• Voice over ATM (Cisco 3600 series, Cisco MC3810 multiservice concentrator)
|
12.1(1)T
|
This command was first supported on the T train for the following voice technology on the following platforms:
• Voice over IP (Cisco 1750, Cisco 2600 series, Cisco 3600 series, Cisco AS5300, Cisco 7200 series, and Cisco 7500 series)
|
12.1(2)T
|
This command was first supported on the T train for the following voice technologies on the following platforms:
• Voice over IP (Cisco MC3810 multiservice concentrator)
• Voice over Frame Relay (Cisco 2600 series, Cisco 3600 series, Cisco MC3810 multiservice concentrator)
• Voice over ATM (Cisco 3600 series, Cisco MC3810 multiservice concentrator)
|
Examples
The following example shows output from the test translation-rule command:
Router# translation-rule 21
Rule 1 555.% 1408555 subscriber international
Rule 2 8.% 1408555 abbreviated international
Router# test translation-rule 21 45678 abbreviated
*Jan 19 16:39:14.578:The replace number 45614085558
Related Commands
Command
|
Description
|
numbering-type
|
Specifies number type for the VoIP or POTS dial peer.
|
rule
|
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
|
show translation-rule
|
Displays the contents of all the rules that have been configured for a specific translation name.
|
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.
|
test voice port detector
To test detector-related functions on a voice port, use the test voice port detector command in privileged EXEC mode.
Cisco 2600 and 3600 Series with Analog Voice Ports
test voice port slot/subunit/port detector {m-lead | battery-reversal | ring | tip-ground |
ring-ground | ring-trip} {on | off | disable}
Cisco 2600 and 3600 Series with Digital Voice Ports
test voice port slot/port:ds0-group detector {m-lead | battery-reversal | ring | tip-ground |
ring-ground | ring-trip} {on | off | disable}
Cisco MC3810 Multiservice Concentrator with Analog Voice Ports
test voice port slot/port detector {m-lead | battery-reversal | ring | tip-ground | ring-ground |
ring-trip} {on | off | disable}
Cisco MC3810 Multiservice Concentrator with Digital Voice Ports
test voice port slot:ds0-group detector {m-lead | battery-reversal | ring | tip-ground |
ring-ground | ring-trip} {on | off | disable}
Syntax DescriptionDescription
For the Cisco 2600 and 3600 Series Routers with Analog Voice Ports:
slot/subunit/port
|
Tests the voice port that you specify with the slot/subunit/port designation.
• slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• subunit specifies a voice interface card (VIC) in which the voice port is located. Valid entries are 0 and 1.
• port specifies an analog voice port number. Valid entries are 0 and 1.
|
For the Cisco 2600 and 3600 Series Routers with Digital Voice Ports:
slot/port:ds0-group
|
Tests the voice port that you specify with the slot/port:ds0-group designation.
• slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1.
• ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
For the Cisco MC3810 Multiservice Concentrator with Analog Voice Ports:
slot/port
|
Tests the voice port that you specify with the slot/port designation.
• slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810.
• port specifies an analog voice port number. Valid entries are 1 to 6.
|
For the Cisco MC3810 Multiservice Concentrator with Digital Voice Ports:
slot:ds0-group
|
Tests the voice port that you specify with the slot:ds0-group designation.
• slot specifies the module (and controller). Valid entries are 0 for the multiflex trunk module (MFT) (controller 0) and 1 for the DVM (controller 1).
• ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
For All Platforms:
m-lead
|
Forces the E&M m-lead detector to the specified state.
|
loop
|
Forces the FXO loop detector to the specified state.
|
battery-reversal
|
Forces the FXO battery-reversal detector to the specified state.
|
ring
|
Forces the FXO ringing detector to the specified state.
|
tip-ground
|
Forces the FXO tip-ground detector to the specified state.
|
ring-ground
|
Forces the FXS ring-ground detector to the specified state.
|
ring-trip
|
Forces the FXS ring-trip detector to the specified state.
|
on
|
Forces the selected item to the on state.
|
off
|
Forces the selected item to the off state.
|
disable
|
Ends the forced state for the selected item.
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
Use the test voice port detector privileged EXEC command to force a detector into specific states for testing. For each signaling type (E&M, FXO, FXS), only the applicable keywords are displayed. When you are finished testing, be sure to enter the command with the disable keyword to end the forced state. The disable keyword is available only if a test condition is already activated.
Examples
The following example forces the tip-ground detector to the off state on an FXO voice port (1/3) on a Cisco MC3810 and ends any call in progress:
Router# test voice port 1/3 detector tip-ground off
The following example ends the forced off state on an FXO voice port (1/3) on a Cisco MC3810:
Router# test voice port 1/3 detector tip-ground disable
The following example forces the ring-trip detector to the on state on an FXS port (0/0/1) on a Cisco 3600 series router and should start a call:
Router# test voice port 0/0/1 detector ring-trip on
The following example ends the forced on state on an FXS port (0/0/1) on a Cisco 3600 series router:
Router# test voice port 0/0/1 detector ring-trip disable
Related Commands
Command
|
Description
|
test voice port inject-tone
|
Injects a test tone into a voice port.
|
test voice port loopback
|
Performs loopback testing on a voice port.
|
test voice port relay
|
Tests relay-related functions on a voice port.
|
test voice port switch
|
Forces a voice port into fax or voice mode.
|
test voice port inject-tone
To inject a test tone into a voice port, use the test voice port inject-tone command in privileged EXEC mode.
Cisco 2600 and 3600 Series with Analog Voice Ports
test voice port slot/subunit/port inject-tone {local | network} {1000hz | 2000hz | 200hz | 3000hz
| 300hz | 3200hz | 3400hz | 500hz | quiet | disable}
Cisco 2600 and 3600 Series with Digital Voice Ports
test voice port slot/port:ds0-group inject-tone {local | network} {1000hz | 2000hz | 200hz |
3000hz | 300hz | 3200hz | 3400hz | 500hz | quiet | disable}
Cisco MC3810 Multiservice Concentrator with Analog Voice Ports
test voice port slot/port inject-tone {local | network} {1000hz | 2000hz | 200hz | 3000hz | 300hz
| 3200hz | 3400hz | 500hz | quiet | disable}
Cisco MC3810 Multiservice Concentrator with Digital Voice Ports
test voice port slot:ds0-group inject-tone {local | network} {1000hz | 2000hz | 200hz | 3000hz |
300hz | 3200hz | 3400hz | 500hz | quiet | disable}
Syntax Description
For the Cisco 2600 and 3600 Series with Analog Voice Ports:
slot/subunit/port
|
Tests the voice port that you specify with the slot/subunit/port designation.
• slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• subunit specifies a voice interface card (VIC) in which the voice port is located. Valid entries are 0 and 1.
• port specifies an analog voice port number. Valid entries are 0 and 1.
|
For the Cisco 2600 and 3600 Series with Digital Voice Ports:
slot/port:ds0-group
|
Tests the voice port that you specify with the slot/port:ds0-group designation.
• slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1.
• ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
For the Cisco MC3810 Multiservice Concentrator with Analog Voice Ports:
slot/port
|
Tests the voice port that you specify with the slot/port designation.
• slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810.
• port specifies an analog voice port number. Valid entries are 1 to 6.
|
For the Cisco MC3810 Multiservice Concentrator with Digital Voice Ports:
slot:ds0-group
|
Tests the voice port that you specify with the slot:ds0-group designation.
• slot specifies the module (and controller). Valid entries are 0 for the multiflex trunk module (MFT) (controller 0) and 1 for the DVM (controller 1).
• ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
For All platforms:
local
|
Directs the injected tone toward the local interface (near end).
|
network
|
Directs the injected tone toward the network (far end).
|
1000hz
|
Injects a 1-kilohertz test tone.
|
2000hz
|
Injects a 2-kilohertz test tone.
|
200hz
|
Injects a 200-hertz test tone.
|
3000hz
|
Injects a 3-kilohertz test tone.
|
300hz
|
Injects a 300-hertz test tone.
|
3200hz
|
Injects a 3.2-kilohertz test tone.
|
3400hz
|
Injects a 3.4-kilohertz test tone.
|
500hz
|
Injects a 500-hertz test tone.
|
quiet
|
Injects a quiet tone.
|
disable
|
Ends the test tone.
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
Use the test voice port inject-tone privileged EXEC command to inject a test tone or to end a test tone. A call must be established on the voice port under test. When you are finished testing, be sure to enter the disable keyword to end the test tone. The disable keyword is available only if a test condition is already activated.
When you enter the disable keyword, you must enter a direction (either network or local); however, you can enter either direction, regardless of which direction you entered to inject the test tone.
Examples
The following example injects a 1-kilohertz test tone into voice port 1/1, directed toward the network (far end), on a Cisco MC3810:
Router# test voice port 1/1 inject-tone network 1000hz
The following example removes the test tone from port 0/0/1 on a Cisco 3600 series router:
Router# test voice port 0/0/1 inject-tone network disable
or
Router# test voice port 0/0/1 inject-tone local disable
Related Commands
Command
|
Description
|
test voice port detector
|
Tests detector-related functions on a voice port.
|
test voice port loopback
|
Performs loopback testing on a voice port.
|
test voice port relay
|
Tests relay-related functions on a voice port.
|
test voice port switch
|
Forces a voice port into fax or voice mode.
|
test voice port loopback
To perform loopback testing on a voice port, use the test voice port loopback command in privileged EXEC mode.
Cisco 2600 and 3600 Series with Analog Voice Ports
test voice port slot/subunit/port loopback {local | network | disable}
Cisco 2600 and 3600 Series with Digital Voice Ports
test voice port slot/port:ds0-group loopback {local | network | disable}
Cisco MC3810 Multiservice Concentrator with Analog Voice Ports
test voice port slot/port loopback {local | network | disable}
Cisco MC3810 Multiservice Concentrator with Digital Voice Ports
test voice port slot:ds0-group loopback {local | network | disable}
Syntax Description
For the Cisco 2600 and 3600 Series with Analog Voice Ports:
slot/subunit/port
|
Tests the voice port that you specify with the slot/subunit/port designation.
• slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• subunit specifies a voice interface card (VIC) in which the voice port is located. Valid entries are 0 and 1.
• port specifies an analog voice port number. Valid entries are 0 and 1.
|
For the Cisco 2600 and 3600 Series with Digital Voice Ports:
slot/port:ds0-group
|
Tests the voice port that you specify with the slot/port:ds0-group designation.
• slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1.
• ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
For the Cisco MC3810 Multiservice Concentrator with Analog Voice Ports:
slot/port
|
Tests the voice port that you specify with the slot/port designation.
• slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810.
• port specifies an analog voice port number. Valid entries are 1 to 6.
|
For the Cisco MC3810 Multiservice Concentrator with Digital Voice Ports:
slot:ds0-group
|
Tests the voice port that you specify with the slot:ds0-group designation.
• slot specifies the module (and controller). Valid entries are 0 for the multiflex trunk module (MFT) (controller 0) and 1 for the DVM (controller 1).
• ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
All Platforms:
local
|
Forces a loopback at the voice port toward the customer premises equipment (CPE).
|
network
|
Forces a loopback at the voice port toward network.
|
disable
|
Ends the forced loopback.
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
Use the test voice port loopback privileged EXEC command to initiate or end a loopback at a voice port. A call must be established on the voice port under test. When you are finished testing, be sure to enter the disable keyword to end the forced loopback. The disable keyword is available only if a test condition is already activated.
Examples
The following example forces a loopback toward the CPE on voice port 1/1 on a Cisco MC3810:
Router# test voice port 1/1 loopback local
The following example ends a forced loopback on port 0/0/1 on a Cisco 3600 series router:
Router# test voice port 0/0/1 loopback disable
Related Commands
Command
|
Description
|
test voice port detector
|
Tests detector-related functions on a voice port.
|
test voice port inject-tone
|
Injects a test tone into a voice port.
|
test voice port relay
|
Tests relay-related functions on a voice port.
|
test voice port switch
|
Forces a voice port into fax or voice mode.
|
test voice port relay
To test relay-related functions on a voice port, use the test voice port relay command in privileged EXEC mode.
Cisco 2600 and 3600 Series with Analog Voice Ports
test voice port slot/subunit/port relay {e-lead | loop | ring-ground | battery-reversal |
power-denial | ring | tip-ground} {on | off | disable}
Cisco 2600 and 3600 Series with Digital Voice Ports
test voice port slot/port:ds0-group relay {e-lead | loop | ring-ground | battery-reversal |
power-denial | ring | tip-ground} {on | off | disable}
Cisco MC3810 Multiservice Concentrator with Analog Voice Ports
test voice port slot/port relay {e-lead | loop | ring-ground | battery-reversal | power-denial |
ring | tip-ground} {on | off | disable}
Cisco MC3810 Multiservice Concentrator with Digital Voice Ports
test voice port slot:ds0-group relay {e-lead | loop | ring-ground | battery-reversal | power-denial
| ring | tip-ground} {on | off | disable}
Syntax Description
For the Cisco 2600 and 3600 Series with Analog Voice Ports:
slot/subunit/port
|
Tests the voice port that you specify with the slot/subunit/port designation.
• slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• subunit specifies a voice interface card (VIC) in which the voice port is located. Valid entries are 0 and 1.
• port specifies an analog voice port number. Valid entries are 0 and 1.
|
For the Cisco 2600 and 3600 Series with Digital Voice Ports:
slot/port:ds0-group
|
Tests the voice port that you specify with the slot/port:ds0-group designation.
• slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1.
• ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
For the Cisco MC3810 Multiservice Concentrator with Analog Voice Ports:
slot/port
|
Tests the voice port that you specify with the slot/port designation.
• slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810.
• port specifies an analog voice port number. Valid entries are 1 to 6.
|
For the Cisco MC3810 Multiservice Concentrator with Digital Voice Ports:
slot:ds0-group
|
Tests the voice port that you specify with the slot:ds0-group designation.
• slot specifies the module (and controller). Valid entries are 0 for the multiflex trunk module (MFT) (controller 0) and 1 for the DVM (controller 1).
• ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
All Platforms:
e-lead
|
Forces the E&M e-lead relay to the specified state.
|
loop
|
Forces the FXO loop relay to the specified state.
|
ring-ground
|
Forces the FXO ring-ground relay to the specified state.
|
battery-reversal
|
Forces the FXO battery-reversal relay to the specified state.
|
power-denial
|
Forces the FXS power-denial relay to the specified state.
|
ring
|
Forces the FXS ringing relay to the specified state.
|
tip-ground
|
Forces the FXS tip-ground relay to the specified state.
|
on
|
Forces the selected item to the on state.
|
off
|
Forces the selected item to the off state.
|
disable
|
Ends the forced state for the selected item.
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
Use the test voice port relay privileged EXEC command to force a relay into specific states for testing. For each signaling type (E&M, FXO, FXS), only the applicable keywords are displayed. When you are finished testing, be sure to enter the disable keyword to end the forced state. The disable keyword is available only if a test condition is already activated.
Examples
The following example forces the E&M e-lead relay to the on state on port 0/0/1 on a Cisco 3600 series router:
Router# test voice port 0/0/1 relay e-lead on
The following example ends a forced actuation of the battery-reversal relay on an FXS port (0/0/1) on a Cisco 3600 series router:
Router# test voice port 0/0/1 relay battery-reversal disable
Related Commands
Command
|
Description
|
test voice port detector
|
Tests detector-related functions on a voice port.
|
test voice port inject-tone
|
Injects a test tone into a voice port.
|
test voice port switch
|
Forces a voice port into fax or voice mode.
|
test voice port switch
To force a voice port into fax mode, use the test voice port switch command in privileged EXEC mode.
Cisco 2600 and 3600 Series with Analog Voice Ports
test voice port slot/subunit/port switch {fax | disable}
Cisco 2600 and 3600 Series with Digital Voice Ports
test voice port slot/port:ds0-group switch {fax | disable}
Cisco MC3810 Multiservice Concentrator with Analog Voice Ports
test voice port slot/port switch {fax | disable}
Cisco MC3810 Multiservice Concentrator with Digital Voice Ports
test voice port slot:ds0-group switch {fax | disable}
Syntax Description
For the Cisco 2600 and 3600 Series with Analog Voice Ports:
slot/subunit/port
|
Tests the voice port that you specify with the slot/subunit/port designation.
• slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• subunit specifies a voice interface card (VIC) in which the voice port is located. Valid entries are 0 and 1.
• port specifies an analog voice port number. Valid entries are 0 and 1.
|
For the Cisco 2600 and 3600 Series with Digital Voice Ports:
slot/port:ds0-group
|
Tests the voice port that you specify with the slot/port:ds0-group designation.
• slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1.
• ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
For the Cisco MC3810 Multiservice Concentrator with Analog Voice Ports:
slot/port
|
Tests the voice port that you specify with the slot/port designation.
• slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810.
• port specifies an analog voice port number. Valid entries are 1 to 6.
|
For the Cisco MC3810 Multiservice Concentrator with Digital Voice Ports:
slot:ds0-group
|
Tests the voice port that you specify with the slot:ds0-group designation.
• slot specifies the module (and controller). Valid entries are 0 for the multiflex trunk module (MFT) (controller 0) and 1 for the DVM (controller 1).
• ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
For All Platforms:
fax
|
Forces a switch to fax mode.
|
disable
|
Ends fax mode; switches back to voice mode.
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
Use the test voice port switch privileged EXEC command to force a voice port into fax mode for testing. If no fax data is detected by the voice port, the voice port remains in fax mode for 30 seconds and then reverts automatically to voice mode. After you enter the test voice port switch fax command, you can use the show voice call or show voice call summary command to check whether the voice port is able to operate in fax mode.
The disable keyword ends the forced mode switch; however, the fax mode ends automatically after 30 seconds. The disable keyword is available only while the voice port is in fax mode.
Examples
The following example forces voice port 1/3 on a Cisco MC3810 into fax mode:
Router# test voice port 1/3 switch fax
The following example returns voice port 0/0/1 on a Cisco 3600 series router to voice mode:
Router# test voice port 0/0/1 switch disable
Related Commands
Command
|
Description
|
show voice call
|
Displays the call processing and protocol state-machine information for a voice port.
|
show voice call summary
|
Displays a summary of the call processing and protocol state-machine information for a voice port.
|
test vrm busyout
To busy out a specific digital signal processor (DSP) or channels on a specific DSP, use the test vrm busyout command in privileged EXEC mode.
test vrm busyout slot-number {first-dsp-number {last-dsp-number | channel number} | all}
Syntax Description
slot-number
|
Number that identifies the slot in which the voice feature card (VFC) is installed. Values for this argument are 0 to 11.
|
first-dsp-number
|
Specifies the first DSP in a range to be busied out. Each VFC holds 96 DSPs, so the value for this argument is 1 to 96.
|
last-dsp-number
|
Specifies the last DSP in a range to be busied out. Each VFC holds 96 DSPs, so the value for this argument is 1 to 96.
|
channel
|
Specifies that a certain channel on the specified DSPs will be busied out.
|
number
|
Indicates the channel to be busied out. Values are 1 or 2.
|
all
|
Indicates that all 96 DSPs on the VFC installed in the defined slot will be busied out.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on Cisco AS5800 universal access servers.
|
Usage Guidelines
Use the test vrm busyout command to busy out either one specific DSP or a range of DSPs on a specific VFC. In addition, you can use this command to busy out a particular channel on a specified DSP or range of DSPs. To restore the activity of the busied-out DSPs, use the test vrm unbusyout command.
Examples
The following example busies out all of the DSPs and associated channels for the VFC located in slot 4:
Router# test vrm busyout 4 all
The following example busies out all of the channels from DSP1 to DSP3 for the VFC located in slot 4:
Router# test vrm busyout 4 1 3
The following example busies out only channel 2 of DSP1 for the VFC located in slot 4:
Router# test vrm busyout 4 1 channel 2
Related Commands
Command
|
Description
|
test vrm unbusyout
|
Restores activity to a busied-out DSP or busied-out channels on a DSP.
|
test vrm reset
To reset a particular digital signal processor (DSP), use the test vrm reset command in privileged EXEC mode.
test vrm reset slot-number dsp-number
Syntax Description
slot-number
|
Number that identifies the slot in which the voice feature card (VFC) is installed.
|
dsp-number
|
Number that identifies the DSP to be reset.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco AS5300.
|
Usage Guidelines
Use the test vrm reset command to send a hard reset command to an identified DSP. When this command is used, any active calls on all channels associated with this DSP are dropped. Under most circumstances, you will never need to use this command.
Examples
The following example resets DSP 4 on the VFC installed in slot 2:
Router# test vrm reset 2 4
Resetting voice device may terminate active calls [confirm}
Reset command sent to voice card 4 for voice device 2.
test vrm unbusyout
To restore activity to a busied-out digital signal processor (DSP) or busied-out channels on a DSP, use the test vrm unbusyout command in privileged EXEC mode.
test vrm unbusyout slot-number {first-dsp-number {last-dsp-number | channel number} | all }
Syntax Description
slot-number
|
Number that identifies the slot in which the voice feature card (VFC) is installed. Values for this field are 0 to 11.
|
first-dsp-number
|
Specifies the first DSP in a range to be restored. Each VFC holds 96 DSPs, so the value for this argument is 1 to 96.
|
last-dsp-number
|
Specifies the last DSP in a range to be restored. Each VFC holds 96 DSPs, so the value for this argument is 1 to 96.
|
channel
|
Specifies that a certain channel on the specified DSPs will be restored.
|
number
|
Indicates the channel to be restored. Values are 1 or 2.
|
all
|
Indicates that all 96 DSPs on the VFC installed in the defined slot will be restored.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco AS5300.
|
Usage Guidelines
Use the test vrm unbusyout command to restore either one specific DSP or a range of DSPs on a specific VFC. In addition, you can use this command to restore a particular channel on a specified DSP or range of DSPs. To busy out a DSP (or range of DSPs) or to busy out a particular channel, use the test vrm busyout command.
Examples
The following example restores the activity of all DSPs and associated channels for the VFC located in slot 4:
Router# test vrm unbusyout 4 all
The following example restores the activity of all channels on the DSP from DSP1 to DSP3 for the VFC located in slot 4:
Router# test vrm unbusyout 4 1 3
The following example restores the activity of only channel 2 of DSP1 for the VFC located in slot 4:
Router# test vrm unbusyout 4 1 channel 2
Related Commands
Command
|
Description
|
test vrm busyout
|
Busy outs a specific DSP or channels on a specific DSP.
|
threshold noise
To configure a noise threshold for incoming calls, use the threshold noise command in voice-port configuration mode. To restore the default, use the no form of this command.
threshold noise {value}
no threshold noise {value}
Syntax Description
value
|
Number that establishes a noise threshold. Valid values are from -30 to -90 decibels (dBs). The default value is -62 dB.
|
Defaults
-62dB
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.2(16)
|
This command was introduced on the following platforms: Cisco 1700 Cisco 1751, Cisco 2600 (with and without the NM-HDA), Cisco 3600 (with and without the NM-HDA), Cisco 7200 (with and without the NM-HDA), Cisco AS5300, Cisco AS5800, and Cisco MC3810.
|
Usage Guidelines
Cisco voice activity detection (VAD) has two layers: application programming interface (API) layer and processing layer. There are 3 states that the processing layer classifies incoming signals: speech, unknown, and silence. The state of the incoming signals is determined by the noise threshold.
In earlier Cisco IOS Releases, the noise threshold is fixed between -62dB and -78 dB. If the voice level is below the noise threshold, then the signal is classified as silence. If the incoming signal cannot be classified, the variable thresholds that are computed with the statistics of speech and noise that VAD gathers is used to make a determination. If the signal still cannot be classified, then it is marked as unknown. The final decision is made by the API. For applications such as hoot-n-holler, you could have the noise create unwanted spurious packets (for example, a voice stream) taking up bandwidth.
With Cisco IOS Release 12.2(16), the noise threshold is configurable using the threshold noise command.
Examples
The following sample configuration shows a noise threshold level of -50 dB configured on a Cisco 3600:
timeouts call-disconnect
To configure the call disconnect timeout value for a specified voice port, use the timeouts call-disconnect command in voice-port configuration mode. To restore the default value, use the no form of this command.
timeouts call-disconnect seconds
no timeouts call-disconnect
Syntax Description
seconds
|
Sets the call-disconnect timeout duration, in seconds. Valid values are from 0 to 120.
|
Defaults
60 seconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(9)T
|
This command was introduced on Cisco 3600 series routers.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
Usage Guidelines
This command applies to Cisco 3600 series routers. To disable the timeouts call-disconnect timer, set the seconds value to 0. Use the timeouts call-disconnect command to specify the number of seconds for which the originating end system waits after receiving disconnect before notifying the user to hang up by playing a fast busy tone. During this duration, the user just hears silence. If the command is disabled by setting the value to 0, the user hears silence indefinitely.
Examples
The following example sets a call-disconnect timeout value of 10 seconds on a Cisco 3600 series router voice port:
timeouts call-disconnect 10
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timing delay-duration
|
Configures the delay dial signal duration for a specified voice port.
|
timeouts initial
To configure the initial digit timeout value for a specified voice port, use the timeouts initial command in voice-port configuration mode. To restore the default value, use the no form of this command.
timeouts initial seconds
no timeouts initial seconds
Syntax Description
seconds
|
Initial timeout duration, in seconds. Valid entries are any integer from 0 to 120.
|
Defaults
10 seconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
Usage Guidelines
Use the timeouts initial command to specify the number of seconds for which the system will wait for the caller to input the first digit of the dialed digits. The timeouts initial timer is activated when the call is accepted and is deactivated when the caller inputs the first digit. If the configured timeout value is exceeded, the caller is notified through the appropriate tone and the call is terminated.
To disable the timeouts initial timer, set the seconds value to 0.
Examples
The following example sets the initial digit timeout value on the Cisco 3600 series to 10 seconds:
The following example sets the initial digit timeout value on the Cisco MC3810 multiservice concentrator to 10 seconds:
Related Commands
Command
|
Description
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts interdigit
To configure the interdigit timeout value for a specified voice port, use the timeouts interdigit command in voice-port configuration mode. To restore the default value, use the no form of this command.
timeouts interdigit seconds
no timeouts interdigit seconds
Syntax Description
seconds
|
Interdigit timeout duration, in seconds. Valid entries are any integer from 0 to 120.
|
Defaults
10 seconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
Usage Guidelines
This command applies to both the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator.
Use the timeouts interdigit command to specify the number of seconds for which the system will wait (after the caller has input the initial digit) for the caller to input a subsequent digit of the dialed digits. The timeouts interdigit timer is activated when the caller inputs a digit and is restarted each time the caller inputs another digit until the destination address is identified. If the configured timeout value is exceeded before the destination address is identified, the caller is notified through the appropriate tone and the call is terminated.
To disable the timeouts interdigit timer, set the seconds value to 0.
Examples
The following example sets the interdigit timeout value on the Cisco 3600 series for 10 seconds:
The following example sets the interdigit timeout value on the Cisco MC3810 multiservice concentrator for 10 seconds:
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts ringing
To configure the timeout value for ringing, use the timeouts ringing command in voice-port configuration mode. To restore the default value, use the no form of this command.
timeouts ringing {seconds | infinity}
no timeouts ringing
Syntax Description
seconds
|
The duration, in seconds, for which a voice port allows ringing to continue if a call is not answered. The range is from 5 to 60,000. The default is 180.
|
infinity
|
Ringing continues until the caller goes on-hook.
|
Defaults
180 seconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators.
|
12.1(2)T
|
This command was integrated into Cisco Release 12.1(2)T.
|
Usage Guidelines
The timeouts ringing command provides the capability to limit the length of time for which a caller can continue ringing a telephone when there is no answer.
Examples
The following example configures voice port 1/1 on a Cisco MC3810 to allow ringing for 600 seconds:
The following example configures voice port 0/0/1 on a Cisco 3600 series router to allow ringing for 600 seconds:
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a voice port.
|
timeouts wait-release
To configure the delay timeout before the system starts the process for releasing voice ports, use the timeouts wait-release command in voice-port configuration mode. To restore the default value, use the no form of this command.
timeouts wait-release {seconds | infinity}
no timeouts wait-release
Syntax Description
seconds
|
The duration, in seconds, for which a voice port stays in the call-failure state while the Cisco router or concentrator sends a busy tone, reorder tone, or out-of-service tone to the port. The range is from 3 to 3600. The default is 30.
|
infinity
|
The voice port is never released as long as the call-failure state remains.
|
Defaults
30 seconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1) MA
|
This command was introduced on the Cisco MC3810 series.
|
12.0(7)XK
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
Use the timeouts wait-release command to limit the time a voice port can be held in a call failure state. After the timeout, the release sequence is enabled.
You can also use this command for voice ports with Foreign Exchange Station (FXS) loop-start signaling to specify the time allowed for a caller to hang up before the voice port goes into the parked state.
Examples
The following example configures voice port 1/1 on a Cisco MC3810 to stay in the call-failure state for 180 seconds while a busy tone, reorder tone, or out-of-service tone is sent to the voice port:
timeouts wait-release 180
The following example configures voice port 0/0/1 on a Cisco 3600 series router to stay in the call-failure state for 180 seconds while a busy tone, reorder tone, or out-of-service tone is sent to the voice port:
timeouts wait-release 180
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a voice port.
|
timers
To configure the Session Initiation Protocol (SIP) signaling timers, use the timers command in the Session Initiation Protocol (SIP) user agent configuration mode. To restore the default value, use the no form of this command.
timers {trying number | connect number | disconnect number | expires number}
no timers {trying number | connect number | disconnect number | expires number}
Syntax Description
trying number
|
Time (in milliseconds) to wait for a 100 response to an INVITE request. Possible values are 100 through 1000. The default is 500.
|
connect number
|
Time (in milliseconds) to wait for a 200 response to an ACK request. Possible values are 100 through 1000. The default is 500.
|
disconnect number
|
Time (in milliseconds) to wait for a 200 response to a BYE request. Possible values are 100 through 1000. The default is 500.
|
expires number
|
Time (in milliseconds) for which an INVITE request is valid. Possible values are 60,000 through 300,000. The default is 180,000.
|
Defaults
The default for trying, connect, and disconnect is 500. The default for expires is 180,000.
Command Modes
SIP user agent configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco AS5300 universal access servers.
|
12.1(3)T
|
This command was modified to change the names of the parameters. Two of the parameters (invite-wait-180 and invite-wait-200) were combined into one (trying).
|
Usage Guidelines
If you used the previous version of this command to configure timers, your previous timer settings will be maintained. The output of the show running configuration command will reflect both timers.
To reset this command to the default value, you can also use the default command.
Examples
The following example configures the SIP signaling timers to wait 500 milliseconds for a 100 response to an INVITE request:
timing clear-wait
To indicate the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port, use the timing clear-wait command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing clear-wait milliseconds
no timing clear-wait milliseconds
Syntax Description
milliseconds
|
Minimum amount of time, in milliseconds, between the inactive seizure signal and the call being cleared. Valid entries on the Cisco 3600 series are numbers from 200 to 2000. Valid entries on the Cisco MC3810 are numbers from 100 to 2000. Supported on E&M ports only.
|
Defaults
400 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 2600 and 3600 series routers.
|
Examples
The following example configures the clear-wait duration on a Cisco 3600 series voice port to 300 milliseconds:
The following example configures the clear-wait duration on a Cisco MC3810 multiservice concentrator voice port to 300 milliseconds:
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dial-pulse min-delay
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing delay-duration
To specify the delay signal duration for a specified voice port, use the timing delay-duration command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing delay-duration milliseconds
no timing delay-duration milliseconds
Syntax Description
milliseconds
|
Delay signal duration for delay dial signaling, in milliseconds. Valid entries are numbers from 100 to 5000. Supported on E&M ports only.
|
Defaults
2000 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
Usage Guidelines
The call direction for the timing delay-duration command is out.
Examples
The following example configures the delay signal duration on a Cisco 3600 series voice port for 3000 milliseconds:
timing delay-duration 3000
The following example configures the delay signal duration on a Cisco MC3810 multiservice concentrator voice port for 3000 milliseconds:
timing delay-duration 3000
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dial-pulse min-delay
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing delay-start
To specify the minimum delay time from outgoing seizure to out-dial address for a specified voice port, use the timing delay-start command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing delay-start milliseconds
no timing delay-start milliseconds
Syntax Description
milliseconds
|
Minimum delay time, in milliseconds, from outgoing seizure to outdial address. Valid entries are numbers from 20 to 2000. Supported on E&M ports only.
|
Defaults
300 milliseconds on the Cisco 3600 series.
150 milliseconds on the Cisco MC3810 multiservice concentrator.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
Usage Guidelines
The call direction for the timing delay-start command is out.
Examples
The following example configures the delay-start duration on a Cisco 3600 series voice port for 250 milliseconds:
The following example configures the delay-start duration on a Cisco MC3810 multiservice concentrator voice port for 250 milliseconds:
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dial-pulse min-delay
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing delay-with-integrity
To specify the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810, use the timing delay-with-integrity command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing delay-with-integrity milliseconds
no timing delay-with-integrity milliseconds
Syntax Description
milliseconds
|
Duration of the wink pulse for the delay dial, in milliseconds. Valid entries are numbers from 0 to 5000. Supported on E&M ports only.
|
Defaults
Zero (0)
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
This command applies only to the Cisco MC3810 multiservice concentrator.
Examples
The following example configures the duration of the wink pulse for the delay dial on a Cisco MC3810 voice port for 10 milliseconds:
timing delay-with-integrity 10
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dial-pulse min-delay
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing dial-pulse min-delay
To specify the time between wink-like pulses for a specified voice port, use the timing dial-pulse min-delay command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing dial-pulse min-delay milliseconds
no timing dial-pulse min-delay milliseconds
Syntax Description
milliseconds
|
Time, in milliseconds, between the generation of wink-like pulses. Valid entries are integers from 0 to 5000.
|
Defaults
300 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
Usage Guidelines
Use the timing dial-pulse min-delay command with PBXs that require a wink-like pulse, even though they have been configured for delay-dial signaling. If the value for this argument is set to 0, the router will not generate this wink-like pulse. The call signal direction for this command is in.
Examples
The following example configures the time between the generation of wink-like pulses on a Cisco 3600 series voice port for 350 milliseconds:
timing dial-pulse min-delay 350
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing dialout-delay
To specify the dial-out delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator, use the timing dialout-delay command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing dialout-delay milliseconds
no timing dialout-delay milliseconds
Syntax Description
milliseconds
|
Dial-out delay, in milliseconds, for the sending digit or cut-through on an FXO trunk or an E&M immediate trunk. Valid entries are from 100 to 5000 milliseconds.
|
Defaults
300 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
This command applies only to the Cisco MC3810 multiservice concentrator.
Examples
The following example configures the dial-out delay on a Cisco MC3810 multiservice concentrator voice port for 350 milliseconds:
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dial-pulse min-delay
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing digit
To specify the dual tone multifrequency (DTMF) digit signal duration for a specified voice port, use the timing digit command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing digit milliseconds
no timing digit milliseconds
Syntax Description
milliseconds
|
The DTMF digit signal duration, in milliseconds. Valid entries are integers from 50 to 100. Supported on FXO, FXS and E&M ports.
|
Defaults
100 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
Usage Guidelines
The call signal direction for the timing digit command is out.
Examples
The following example configures the DTMF digit signal duration on a Cisco 3600 series voice port for 50 milliseconds:
The following example configures the DTMF digit signal duration on a Cisco MC3810 multiservice concentrator voice port for 50 milliseconds:
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dial-pulse min-delay
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing guard-out
To specify the guard-out duration of an Foreign Exchange Office (FXO) voice port, use the timing guard-out command in voice-port configuration mode. To restore the default value, use the no form of this command.
timing guard-out milliseconds
no timing guard-out
Syntax Description
milliseconds
|
Duration, in milliseconds, of the guard-out period. The range is 300 to 3000. The default is 2000.
|
Defaults
2000 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on the Cisco MC3810 multiservice concentrator
|
12.0(7)XK
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
The timing guard-out command applies to the Cisco 2600, 3600, and MC3810 multiservice concentrator. This command is supported on FXO voice ports only.
Examples
The following example configures the timing guard-out duration on a Cisco MC3810 multiservice concentrator voice port for 1000 milliseconds:
The following example configures the timing guard-out duration on a Cisco 2600 or 3600 series voice port for 1000 milliseconds:
timing hookflash-input
To specify the maximum duration of a hookflash for an Foreign Exchange Station (FXS) interface, use the timing hookflash-input command in voice-port configuration mode. To restore the default duration for hookflash timing, use the no form of this command.
timing hookflash-input milliseconds
no timing hookflash-input
Syntax Description
milliseconds
|
Duration of the hookflash, in milliseconds. Possible values are 50 through 1550 milliseconds. Default is 600 milliseconds.
|
Defaults
600 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 3600 series.
|
Usage Guidelines
This command does not affect whether hookflash relay is enabled; hookflash relay is enabled only when the dtmf-relay h245-signal command is configured on the applicable VoIP dial peers. When the dtmf-relay h245-signal command is configured, the H.323 gateway relays hookflash by using an H.245 "signal" User Input Indication method.Hookflash is sent only when an h245 signal is available.
Use the timing hookflash-input command on FXS interfaces to specify the maximum duration (in milliseconds) of a hookflash indication. If the hookflash lasts longer than the specified limit, the FXS interface processes the indication as an on-hook.
Examples
The following example implements timing for the hookflash with a duration of 200 milliseconds:
timing hookflash-input 200
Related Commands
Command
|
Description
|
dtmf-relay (Voice over IP)
|
Specifies how an H.323 gateway relays DTMF tones between telephony interfaces and an IP network.
|
timing hookflash-out
To specify the duration of hookflash indications that the gateway generates on a Foreign Exchange Office (FXO) interface, use the timing hookflash-out command in voice-port configuration mode. To restore the default duration for hookflash timing, use the no form of this command.
timing hookflash-out milliseconds
no timing hookflash-out
Syntax Description
milliseconds
|
Duration of the hookflash, in milliseconds. Possible values are 50 through 1550 milliseconds.
|
Defaults
400 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on Cisco 2500, Cisco 2600, Cisco 3600, and Cisco 7200 series routers and Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
This command does not affect whether hookflash relay is enabled; hookflash relay is enabled only when the dtmf-relay h245-signal command is configured on the applicable Voice over IP (VoIP) dial peers. Hookflash is relayed by using an h245-signal indication and can be sent only when an h245 signal is available.
Use the timing hookflash-out command on FXO interfaces to specify the duration (in milliseconds) of a hookflash indication. To set hookflash timing parameters for analog voice interfaces, use the voice-port timing subcommand.
Examples
The following example implements timing for the hookflash with a duration of 200 milliseconds:
Related Commands
Command
|
Description
|
dtmf-relay (Voice over IP)
|
Specifies how an H.323 gateway relays DTMF tones between telephony interfaces and an IP network.
|
voice-port
|
Enters voice-port configuration mode.
|
timing interdigit
To specify the dual-tone multifrequency (DTMF) interdigit duration for a specified voice port, use the timing interdigit command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing interdigit milliseconds
no timing interdigit milliseconds
Syntax Description
milliseconds
|
DTMF interdigit duration, in milliseconds. Valid entries are numbers from 50 to 500 milliseconds. Supported on FXO, FXS and E&M ports.
|
Defaults
100 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
11.3(1)MA
|
This command was supported on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
The call signal direction for the timing interdigit command is out.
Examples
The following example configures the DTMF interdigit duration on a Cisco 3600 series voice port for 150 milliseconds:
The following example configures the DTMF interdigit duration on a Cisco MC3810 multiservice concentrator voice port for 150 milliseconds:
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dial-pulse min-delay
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing percentbreak
To specify the percentage of the break period for dialing pulses for a voice port, use the timing percentbreak command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing percentbreak percent
no timing percentbreak
Syntax Description
percent
|
Percentage of the break period for dialing pulses. Valid entries are from 20 to 80. The default is 50.
|
Defaults
50 percent
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA4
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
The timing percentbreak command is supported on Foreign Exchange Office (FXO) and ear and mouth (E&M) voice ports only.
Examples
The following example configures the break period percentage on a Cisco MC3810 multiservice concentrator voice port for 30 percent:
The following example configures the break period percentage on a Cisco 2600 or 3600 voice port for 30 percent:
Related Commands
Command
|
Description
|
timing pulse
|
Configures the pulse dialing rate for a voice port.
|
timing pulse-interdigit
|
Configures the pulse interdigit timing for a voice port.
|
timing pulse
To specify the pulse dialing rate for a specified voice port, use the timing pulse command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing pulse pulses-per-second
no timing pulse pulses-per-second
Syntax Description
pulses-per-second
|
Pulse dialing rate, in pulses per second. Valid entries are numbers from 10 to 20. Supported on Foreign Exchange Office (FXO) and ear and mouth (E&M) ports only.
|
Defaults
20 pulses per seconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
11.3(1)MA
|
This command was supported on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
The call signal direction for the timing pulse command is out.
Examples
The following example configures the pulse dialing rate on a Cisco 3600 series voice port for 15 pulses per second:
The following example configures the pulse dialing rate on a Cisco MC3810 multiservice concentrator voice port for 15 pulses per second:
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dial-pulse min-delay
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing pulse-interdigit
To specify the pulse interdigit timing for a specified voice port, use the timing pulse-interdigit command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing pulse-interdigit milliseconds
no timing pulse-interdigit milliseconds
Syntax Description
milliseconds
|
Pulse dialing interdigit timing, in milliseconds. Valid entries are integers from 100 to 1000. Supported on Foreign Exchange Office (FXO) and ear and mouth (E&M) ports only.
|
Defaults
500 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
11.3(1)MA
|
This command was supported on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
The call signal direction for the timing pulse-interdigit command is out.
Examples
The following example configures the pulse-dialing interdigit timing on a Cisco 3600 series voice port for 300 milliseconds:
timing pulse-interdigit 300
The following example configures the pulse-dialing interdigit timing on a Cisco MC3810 multiservice concentrator voice port for 300 milliseconds:
timing pulse-interdigit 300
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dial-pulse min-delay
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing wink-duration
To specify the maximum wink-signal duration for a specified voice port, use the timing wink-duration command in voice-port configuration mode. To restore the default value, use the no form of this command.
timing wink-duration milliseconds
no timing wink-duration milliseconds
Syntax Description
milliseconds
|
Maximum wink-signal duration, in milliseconds, for a wink-start signal. Valid entries are from 100 to 400 milliseconds. Supported on ear and mouth (E&M) ports only.
|
Defaults
200 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
11.3(1)MA
|
This command was supported on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
The call signal direction for the timing wink-duration command is out.
Examples
The following example configures the wink-signal duration on a Cisco 3600 series voice port for 300 milliseconds:
The following example configures the wink-signal duration on a Cisco MC3810 multiservice concentrator voice port for 300 milliseconds:
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing delay-with-integrity
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-wait
|
Specifies the maximum wink-wait duration for a specified voice port.
|
timing wink-wait
To specify the maximum wink-wait duration for a specified voice port, use the timing wink-wait command in voice-port configuration mode. To reset the default value, use the no form of this command.
timing wink-wait milliseconds
no timing wink-wait milliseconds
Syntax Description
milliseconds
|
Maximum wink-wait duration, in milliseconds, for a wink start signal. Valid entries are from 100 to 5000 milliseconds. Supported on ear and mouth (E&M) ports only.
|
Defaults
200 milliseconds
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
11.3(1)MA
|
This command was supported on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
The call signal direction for the timing wink-wait command is out.
Examples
The following example configures the wink-wait duration on a Cisco 3600 series voice port for 300 milliseconds:
The following example configures the wink-wait duration on a Cisco MC3810 multiservice concentrator voice port for 300 milliseconds:
Related Commands
Command
|
Description
|
timeouts initial
|
Configures the initial digit timeout value for a specified voice port.
|
timeouts interdigit
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts wait-release
|
Configures the timeout value for releasing voice ports on the Cisco MC3810 multiservice concentrator.
|
timing clear-wait
|
Indicates the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port.
|
timing delay-duration
|
Specifies the delay signal duration for a specified voice port.
|
timing delay-start
|
Specifies the minimum delay time from outgoing seizure to out-dial address for a specified voice port.
|
timing delay-with-integrity
|
Specifies the duration of the wink pulse for the delay dial for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dialout-delay
|
Specifies the dialout delay for the sending digit on a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing dial-pulse min-delay
|
Specifies the time between wink-like pulses for a specified voice port.
|
timing digit
|
Specifies the DTMF digit signal duration for a specified voice port.
|
timing interdigit
|
Specifies the DTMF interdigit duration for a specified voice port.
|
timing percentbreak
|
Specifies the percentage of a break period for a dialing pulse for a specified voice port on the Cisco MC3810 multiservice concentrator.
|
timing pulse
|
Specifies the pulse dialing rate for a specified voice port.
|
timing pulse-interdigit
|
Specifies the pulse interdigit timing for a specified voice port.
|
timing wink-duration
|
Specifies the maximum wink signal duration for a specified voice port.
|
token-root-name
To specify which root or Certificate Authority (CA) certificate the router should use to validate the settlement token in the incoming setup message, use the token-root-name command in settlement configuration mode. To restore the default value, use the no form of this command.
token-root-name name
no token-root-name name
Syntax Description
name
|
Specifies the name that is the certificate identification as configured through the crypto ca identity name command or the crypto ca trusted-root name command.
|
Defaults
The terminating gateway uses the CA certificate to validate the settlement token.
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco AS5300 and AS5800 universal access servers.
|
Examples
The following example defines the token-root-name as sample:
The following example shows new output for the show settlement command to display the value of the token-root-name command:
Address url = https://1.14.115.100:8444/
Encryption = all (default)
Max Concurrent Connections = 20 (default)
Connection Timeout = 3600 (s) (default)
Response Timeout = 1 (s) (default)
Retry Delay = 2 (s) (default)
Retry Limit = 1 (default)
Session Timeout = 86400 (s) (default)
Roaming = Disabled (default)
Number of Connections = 1
Number of Transactions = 0
Related Commands
Command
|
Description
|
crypto ca identity
|
Declares the Certificate Authority that your router should use.
|
crypto ca trusted-root
|
Configures the root certificate that the server uses to sign the settlement tokens.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
tone ringback alert-no-PI
To generate automatic ringback for the caller when no Progress Indicator (PI) alert has been received over the H.323 network, use the tone ringback alert-no-PI command in dial-peer configuration mode. To disable automatic ringback, use the no form of this command.
tone ringback alert-no-PI
no tone ringback alert-no-PI
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.2(1)
|
This command was introduced on Cisco 1700 series, Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, and Cisco 7200 series routers and on the Cisco AS5300 and Cisco AS5800 universal access servers.
|
Usage Guidelines
The tone ringback alert-no-PI command is used to generate ringback in an H.323 network when the attached device (for example, an ISDN device) cannot.
Examples
The following example activates ringback for a VoIP dial peer numbered 322:
router(config)# dial-peer voice 322 voip
router(config-dial-peer)# tone ringback alert-no-PI
Related Commands
Command
|
Description
|
progress_ind
|
Sets a specific PI in call Setup, Progress, or Connect messages from an H.323 VoIP gateway.
|
translate
To apply a translation rule to an inbound plain old telephone service (POTS) call leg, use the translate command in voice-port configuration mode. To remove the translation rule to an inbound POTS call leg, use the no form of this command.
translate {calling-number | called-number} name-tag
no translate {calling-number | called-number} name-tag
Syntax Description
calling-number
|
Applies the translation rule to the inbound calling party number.
|
called-number
|
Applies the translation rule to the inbound called party number.
|
name-tag
|
The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. The range is 1 through 2,147,483,647.
|
Defaults
No default behavior or values.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.0(7)XR1
|
This command was introduced for Voice over IP on the Cisco AS5300.
|
12.0(7)XK
|
This command was first supported for Voice over IP on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810 multiservice concentrator.
|
12.1(1)T
|
This command was first supported on the T train for Voice over IP on the Cisco 1750, Cisco 2600 series, Cisco 3600 series, Cisco AS5300, Cisco 7200 series, and Cisco 7500 series.
|
12.1(2)T
|
This command was first supported on the T train for Voice over IP on the Cisco MC3810 multiservice concentrator.
|
Examples
The following example applies translation rule 21 to the POTS inbound calling party number:
rule 1 555.% 1408555 subscriber international
rule 2 7.% 1408555 abbreviated international
translate calling-number 21
The following example applies translation rule 20 to the POTS inbound called party number:
rule 1 .%555.% 7 any abbreviated
translate called-number 20
Related Commands
Command
|
Description
|
numbering-type
|
Specifies number type for the VoIP or POTS dial peer.
|
rule
|
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
|
show translation-rule
|
Displays the contents of all the rules that have been configured for a specific translation name.
|
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.
|
translate-outgoing
To apply a translation rule to an outbound plain old telephone service (POTS) or Voice over IP (VoIP) call leg, use the translate-outgoing command in dial-peer configuration mode. To remove the translation rule to an outbound POTS or VoIP call leg, use the no form of this command.
translate-outgoing {calling-number | called-number} name-tag
no translate-outgoing {calling-number | called-number} name-tag
Syntax Description
calling-number
|
Applies the translation rule to the outbound calling party number.
|
called-number
|
Applies the translation rule to the outbound called party number.
|
name-tag
|
The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. The range is 1 through 2,147,483,647.
|
Defaults
No default behavior or values.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.0(7)XR1
|
This command was introduced for Voice over IP on the Cisco AS5300.
|
12.0(7)XK
|
This command was first supported for Voice over IP on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810 multiservice concentrator.
|
12.1(1)T
|
This command was first supported on the T train for Voice over IP on the Cisco 1750, Cisco 2600 series, Cisco 3600 series, Cisco AS5300, Cisco 7200 series, and Cisco 7500 series.
|
12.1(2)T
|
This command was first supported on the T train for Voice over IP on the Cisco MC3810 multiservice concentrator.
|
Examples
The following example applies translation rule 21 to the VoIP outbound calling number:
rule 1 555.% 1408555 subscriber international
rule 2 7.% 1408555 abbreviated international
translate-outgoing calling-number 21
The following example applies translation rule 20 to the VoIP called number:
rule 1 .%555.% 7 any abbreviated
translate-outgoing called-number 20
Related Commands
Command
|
Description
|
numbering-type
|
Specifies number type for the VoIP or POTS dial peer.
|
rule
|
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
|
show translation-rule
|
Displays the contents of all the rules that have been configured for a specific translation name.
|
translate
|
Applies a translation rule to a calling party number or a called party number for incoming 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.
|
translation-rule
To create a translation name and enter translation-rule configuration mode to apply rules to the translation name, use the translation-rule command in global configuration mode. To remove the translation rule, use the no form of this command.
translation-rule name-tag
no translation-rule name-tag
Syntax Description
name-tag
|
The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. The range is 1 through 2,147,483,647.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR1
|
This command was introduced for Voice over IP on the Cisco AS5300.
|
12.0(7)XK
|
This command was first supported for the following voice technologies on the following platforms:
• Voice over IP (Cisco 2600 series, Cisco 3600 series, Cisco MC3810 multiservice concentrator)
• Voice over Frame Relay (Cisco 2600 series, Cisco 3600 series, Cisco MC3810 multiservice concentrator)
• Voice over ATM (Cisco 3600 series, Cisco MC3810 multiservice concentrator)
|
12.1(1)T
|
This command was first supported on the T train for the following voice technology on the following platforms: Voice over IP (Cisco 1750, Cisco 2600 series, Cisco 3600 series, Cisco AS5300, Cisco 7200 series, and Cisco 7500 series)
|
12.1(2)T
|
This command was first supported on the T train for the following voice technologies on the following platforms:
• Voice over IP (Cisco MC3810 multiservice concentrator)
• Voice over Frame Relay (Cisco 2600 series, Cisco 3600 series, Cisco MC3810 multiservice concentrator)
• Voice over ATM (Cisco 3600 series, Cisco MC3810 multiservice concentrator)
|
Usage Guidelines
This command applies to all translation rules.
Examples
The following example creates translation rule 21 and applies a rule to it:
rule 1 555.% 1408555 subscriber international
Related Commands
Command
|
Description
|
numbering-type
|
Specifies number type for the VoIP or POTS dial peer.
|
rule
|
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
|
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.
|
voip-incoming translation-rule
|
Captures calls that originate from H.323-compatible clients.
|
transport
To configure the Session Inititation Protocol (SIP) user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket, use the transport command in SIP user-agent configuration mode. To block reception of SIP signaling messages on a particular socket, use the no form of this command.
transport {udp | tcp}
no transport {udp | tcp}
Syntax Description
udp
|
Configures the SIP user agent to receive SIP messages on UDP port 5060.
|
tcp
|
Configures the SIP user agent to receive SIP messages on TCP port 5060.
|
Defaults
Both UDP and TCP transport protocols are enabled.
Command Modes
SIP user-agent configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco AS5300 universal access servers.
|
12.1(3)T
|
Support for access platforms was added.
|
Usage Guidelines
This command controls whether messages reach the SIP service provider interface (SPI).
Examples
The following example configures the SIP user agent to block reception of SIP signaling messages on the TCP socket:
Related Commands
Command
|
Description
|
sip-ua
|
Enables the SIP user-agent configuration commands, with which you configure the user agent.
|
type (voice)
To specify the E&M interface type, use the type command in voice-port configuration mode. To reset the default value, use the no form of this command.
type {1 | 2 | 3 | 5}
no type {1 | 2 | 3 | 5}
Syntax Description
1
|
Indicates the following lead configuration:
• E—Output, relay to ground.
• M—Input, referenced to ground.
|
2
|
Indicates the following lead configuration:
• E—Output, relay to SG.
• M—Input, referenced to ground.
• SB—Feed for M, connected to -48V.
• SG—Return for E, galvanically isolated from ground.
|
3
|
Indicates the following lead configuration:
• E—Output, relay to ground.
• M—Input, referenced to ground.
• SB—Connected to -48V.
• SG—Connected to ground.
|
5
|
Indicates the following lead configuration:
• E—Output, relay to ground.
• M—Input, referenced to -48V.
|
Defaults
Type 1
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series routers.
|
11.3(1)MA
|
This command was first supported on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
Use the type command to specify the E&M interface for a particular voice port. With 1, the tie-line equipment generates the E-signal to the PBX type grounding the E-lead. The tie-line equipment detects the M-signal by detecting current flow to ground. If you select 1, a common ground must exist between the line equipment and the PBX.
With 2, the interface requires no common ground between the equipment, thereby avoiding ground loop noise problems. The E-signal is generated toward the PBX by connecting it to SG. The M-signal is indicated by the PBX connecting it to SB. While Type 2 interfaces do not require a common ground, they do have the tendency to inject noise into the audio paths because they are asymmetrical with respect to the current flow between devices.
Note
E&M Type 4 is not a supported option. However, Type 4 operates similarly to Type 2 except for the M-lead operation. On Type 4, the M-lead states are open/ground, compared to Type 2, which is open/battery. Type 4 can interface with Type 2. To use Type 4 you can set the E&M voice port to Type 2 and perform the necessary M-lead rewiring.
With 3, the interface operates the same as Type 1 interfaces with respect to the E-signal. The M-signal, however, is indicated by the PBX connecting it to SB on assertion and alternately connecting it to SG during inactivity. If you select 3, a common ground must be shared between equipment.
With 5, the Type 5 line equipment indicates E-signal to the PBX by grounding the E-lead. The PBX indicates M-signal by grounding the M-lead. A Type 5 interface is quasi-symmetrical in that while the line is up, current flow is more or less equal between the PBX and the line equipment, but noise injection is a problem.
Examples
The following example selects Type 3 as the interface type for the voice port on the Cisco 3600 series:
The following example selects Type 3 as the interface type for the voice port on the Cisco MC3810 multiservice concentrator:
type (settlement)
To point to the provider type and the specific settlement server, use the type command in settlement configuration mode. To disable this command, use the no form of this command.
type {osp | uni-osp}
no type
Syntax Description
osp
|
Enables the Open Settlement Protocol (OSP) server type.
|
uni-osp
|
Enables authentication of Voice over IP (VoIP) calls to the Public Switched Telephone Network (PSTN) using a single settlement server.
|
Defaults
osp
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco AS5300 universal access servers.
|
12.1(2)T
|
The uni-osp keyword was introduced.
|
Usage Guidelines
This command defines the settlement server that is doing the accounting and enables the server to do the accounting.
Examples
The following example enables authentication of VoIP calls to the PSTN using a single settlement server:
Related Commands
Command
|
Description
|
connection-timeout
|
Sets the connection timeout.
|
customer-id
|
Sets the customer identification.
|
device-id
|
Sets the device identification.
|
encryption
|
Specifies the encryption method.
|
max-connection
|
Sets the maximum simultaneous connections.
|
response-timeout
|
Sets the response timeout.
|
retry-delay
|
Sets the retry delay.
|
retry-limit
|
Sets the connection retry limit.
|
session-timeout
|
Sets the session timeout.
|
settlement
|
Enters settlement configuration mode.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown/no shutdown
|
Brings up the settlement provider and then shuts it down.
|
url
|
Specifies the Internet service provider (ISP) address.
|
unbundle vfc
To unbundle DSPWare from the VCWare and configure the default file and capability lists with default values, use the unbundle vfc command in privileged EXEC mode.
unbundle [high-complexity | medium-complexity] vfc slot-number
Syntax Description
high-complexity
|
(Optional) Unbundles the high-complexity firmware set.
|
medium-complexity
|
(Optional) Unbundles the medium-complexity firmware set.
|
slot-number
|
Indicates the voice feature card (VFC) slot number.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on Cisco AS5300 universal access servers.
|
12.0(2)XH
|
The high-complexity and medium-complexity keywords were added.
|
12.0(3)T
|
This command was integrated into Cisco IOS Release 12.0(3)T.
|
Usage Guidelines
VFCs come with a single bundled image, VCWare, stored in VFC Flash memory. Use the unbundle vfc command to unbundle this bundled image into separate files, which are then written to Flash memory. When VCWare is unbundled, it automatically adds DSPWare to Flash memory, creates both the capability and default file lists, and populates these lists with the default files for that version of VCWare. The default file list includes the files that will be used to boot up the system. The capability list defines the available voice codecs for H.323 capability negotiation. These files are used during initial card configuration and for subsequent firmware upgrades.
Before unbundling a VFC software image that you have just copied over to VFC Flash, use the clear vfc command. Unbundling a DSP firmware set rewrites the default-file and capabilities lists. After unbundling, you must reload the router for any changes to take effect.
Examples
The following example unbundles the high-complexity firmware set into slot 2:
Router# unbundle high-complexity vfc 2
Related Commands
Command
|
Description
|
copy flash vfc
|
Copies a new version of VCWare from the Cisco AS5300 motherboard to VFC Flash memory.
|
copy tftp vfc
|
Copies a new version of VCWare from a TFTP server to VFC Flash memory.
|
url
To configure the Internet service provider (ISP) address, use the url command in settlement configuration mode. To disable this command, use the no form of this command.
url url-address
no url url-address
Syntax Description
url-address
|
Specifies the URL address. A valid URL address is as follows: http://fully qualified domain name[:port]/[URL]
|
Defaults
No default behavior or values.
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco AS5300 universal access servers.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
You can configure the address type multiple times. If you configure multiple URLs for the settlement server, the gateway attempts to send the request to each URL in the order in which you configured these addresses.
Examples
The following example configures four URLs for the settlement server:
url https://1.2.3.4:4444/
url https://yourcompany.com:443/
Related Commands
Command
|
Description
|
connection-timeout
|
Sets the connection timeout.
|
customer-id
|
Sets the customer identification.
|
device-id
|
Sets the device identification.
|
encryption
|
Specifies the encryption method.
|
max-connection
|
Sets the maximum simultaneous connections.
|
response-timeout
|
Sets the response timeout.
|
retry-delay
|
Sets the retry delay.
|
retry-limit
|
Sets the connection retry limit.
|
session-timeout
|
Sets the session timeout.
|
settlement
|
Enters settlement configuration mode.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown/no shutdown
|
Brings up the settlement provider and then shuts it down.
|
type
|
Specifies the provider type.
|
use-proxy
To enable proxy communications for calls between local and remote zones, use the use-proxy command in gatekeeper configuration mode. To either remove a proxy configuration entry for a remote zone or disable proxy communications between local and remote zones, use the no form of this command.
use-proxy local-zone-name {default | remote-zone remote-zone-name}{inbound-to |
outbound-from}{gateway | terminal}
no use-proxy local-zone-name remote-zone remote-zone-name [{inbound-to |
outbound-from}{gateway | terminal}]
Syntax Description
local-zone-name
|
The name or zone name of the gatekeeper, which is usually the fully domain-qualified host name of the gatekeeper. For example, if the domain name is cisco.com, the gatekeeper name might be gk1.cisco.com. However, if the gatekeeper is controlling multiple zones, the name of the gatekeeper for each zone should be a unique string that has a mnemonic value.
|
default
|
Defines the default proxy policy for all calls that are not defined by a use-proxy command with the remote-zone keyword.
|
remote-zone remote-zone-name
|
Defines a proxy policy for calls to or from a specific remote gatekeeper or zone.
|
inbound-to
|
Applies the proxy policy to calls that are inbound to the local zone from a remote zone. Each use-proxy command defines the policy for only one direction.
|
outbound-from
|
Applies the proxy policy to calls that are outbound from the local zone to a remote zone. Each use-proxy command defines the policy for only one direction.
|
gateway
|
Defines the type of local device to which the policy applies. The gateway option applies the policy only to local gateways.
|
terminal
|
Defines the type of local device to which the policy applies. The terminal option applies the policy only to local terminals.
|
Defaults
The local zone uses proxy for both inbound and outbound calls to and from the local H.323 terminals only. Proxy is not used for both inbound and outbound calls to and from local gateways.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced on Cisco AS5300 universal access servers.
|
Usage Guidelines
This command replaces the zone access command used in the previous versions of the gatekeeper. When a previous version of gatekeeper is upgraded, any zone access commands are translated to use-proxy commands. You can use the show gatekeeper zone status command to see the gatekeeper proxy configuration.
Examples
In the following example, the local zone sj.xyz.com is configured to use a proxy for inbound calls from remote zones tokyo.xyz.com and milan.xyz.com to gateways in its local zone. The sj.xyz.com zone is also configured to use a proxy for outbound calls from gateways in its local zone to remote zones tokyo.xyz.com and milan.xyz.com.
use-proxy sj.xyz.com remote-zone tokyo.xyz.com inbound-to gateway
use-proxy sj.xyz.com remote-zone tokyo.xyz.com outbound-from gateway
use-proxy sj.xyz.com remote-zone milan.xyz.com inbound-to gateway
use-proxy sj.xyz.com remote-zone milan.xyz.com outbound-from gateway
Because the default mode disables proxy communications for all gateway calls, only the gateway call scenarios listed above can use the proxy.
In the following example, the local zone sj.xyz.com uses a proxy for only those calls that are outbound from H.323 terminals in its local zone to the specified remote zone germany.xyz.com:
no use-proxy sj.xyz.com default outbound-from terminal
use-proxy sj.xyz.com remote-zone germany.xyz.com outbound-from terminal
Note
Any calls inbound to H.323 terminals in the local zone sj.xyz.com from the remote zone germany.xyz.com use the proxy because the default applies.
The following example shows how to remove one or more proxy statements for the remote zone germany.xyz.com from the proxy configuration list:
no use-proxy sj.xyz.com remote-zone germany.xyz.com
The command above removes all special proxy configurations for the remote zone germany.xyz.com. After you enter a command like this, all calls between the local zone (sj.xyz.com) and germany.xyz.com are processed according to the defaults defined by any use-proxy commands that use the default option.
To prohibit proxy use for inbound calls to H.323 terminals in a local zone from a specified remote zone, enter a command similar to the following :
no use-proxy sj.xyz.com remote-zone germany.xyz.com inbound-to terminal
This command overrides the default and disables proxy use for inbound calls from remote zone germany.xyz.com to all H.323 terminals in the local zone sj.xyz.com.
Related Commands
Command
|
Description
|
show gatekeeper zone status
|
Displays the status of zones related to a gatekeeper.
|
vad (dial peer)
To enable voice activity detection (VAD) for the calls using a particular dial peer, use the vad command in dial-peer configuration mode. To disable VAD, use the no form of this command.
vad [aggressive]
no vad [aggressive]
Syntax Description
aggressive
|
Reduces noise threshold from -78 to -62 dBm. Available only when session protocol multicast is configured.
|
Defaults
VAD is enabled
Aggressive VAD is enabled in multicast dial peers
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series.
|
12.0(4)T
|
This command was implemented as a dial-peer command on Cisco MC3810 (in prior releases, the vad command was available only as a voice-port command).
|
12.2(10)
|
The aggressive keyword was added.
|
Usage Guidelines
Use this command to enable voice activity detection. With VAD, voice data packets fall into three categories: speech, silence, and unknown. Speech and unknown packets are sent over the network; silence packets are discarded. The sound quality is slightly degraded with VAD, but the connection monopolizes much less bandwidth. If you use the no form of this command, VAD is disabled and voice data is continuously sent to the IP backbone. When configuring voice gateways to handle fax calls, VAD should be disabled at both ends of the IP network because it can interfere with the successful reception of fax traffic.
When the aggressive keyword is used, the VAD noise threshold is reduced from -78 to -62 dBm. Noise that falls below the -62 dBm threshold is considered to be silence and is not sent over the network. Additionally, unknown packets are considered to be silence and are discarded.
On the Cisco MC3810, VAD can also be assigned to the voice port using the vad (voice-port) command. On the Cisco MC3810 multiservice concentrator, if you enable VAD on the dial peer for Voice over Frame Relay switched calls or permanent calls, the dial-peer setting overrides the VAD setting on the voice port.
Note
On the Cisco MC3810, the vad (dial-peer) command is enabled by default. The vad (voice-port) command is disabled by default.
Examples
The following example enables VAD for a Voice over IP (VoIP) dial peer, starting from global configuration mode:
Related Commands
Command
|
Description
|
comfort-noise
|
Generates background noise to fill silent gaps during calls if VAD is activated.
|
dial-peer voice
|
Enters dial-peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.
|
vad (voice-port)
|
Enables VAD for the calls using a particular voice port.
|
vad (voice-port)
To enable voice activity detection (VAD) for the calls using a particular voice port, use the vad command in voice-port configuration mode. To disable VAD, use the no form of this command.
vad
no vad
Syntax Description
This command has no arguments or keywords.
Defaults
VAD is not enabled.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced as a voice-port command on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
This command applies to Voice over Frame Relay and Voice over ATM on Cisco MC3810 multiservice concentrators.
Use the vad command to enable voice activity detection. With VAD, silence is not sent over the network; only audible speech is sent. If you enable VAD, the sound quality will be slightly degraded but the connection will monopolize much less bandwidth. If you use the no form of this command, VAD is disabled on the voice port.
Note
It is recommended that you use the vad command in dial-peer configuration mode.
Examples
The following example enables VAD:
Related Commands
Command
|
Description
|
comfort-noise
|
Generates background noise to fill silent gaps during calls if VAD is activated.
|
vad (dial peer)
|
Enables VAD for the calls using a particular dial peer.
|
vbr-rt
To configure the real-time variable bit rate (VBR) for Voice over ATM connections, use the vbr-rt command in ATM virtual circuit configuration mode. To restore the default, use the no form of this command.
vbr-rt peak-rate average-rate burst
no vbr-rt
Syntax Description
peak-rate
|
The peak information rate (PIR) of the voice connection, in kbps. The range is 56 to 10,000.
|
average-rate
|
The average information rate (AIR) of the voice connection, in kbps. The range is 1 to 56.
|
burst
|
Burst size, in number of cells. The range is 0 to 65,536.
|
Defaults
No vbr-rt settings are configured.
Command Modes
ATM virtual circuit configuration
Command History
Release
|
Modification
|
12.0
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
Support for this command was extended to the Cisco 3600 series.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
The vbr-rt command configures traffic shaping between voice and data permanent virtual circuits (PVCs). Traffic shaping is required so that the carrier does not discard calls. To configure voice and data traffic shaping, you must configure the peak, average, and burst options for voice traffic. Configure the burst value if the PVC will be carrying bursty traffic. The peak, average, and burst values are needed so that the PVC can effectively handle the bandwidth for the number of voice calls. To calculate the minimum peak, average, and burst values for the number of voice calls, use the following calculations:
•
Peak value: (2 x the maximum number of calls) x 16 kbps
•
Average value: (1 x the maximum number of calls) x 16 kbps
•
Burst value: (4 x the maximum number of calls)
Note
When you configure data PVCs that will be traffic shaped with voice PVCs, use the encapsulation aal5 snap command and calculate the overhead as 1.13 times the voice rate.
Examples
The following example configures the traffic shaping rate for ATM PVC 20 on a Cisco 3600. In the example, the peak, average, and burst rates are calculated based on a maximum of 20 calls on the PVC.
encapsulation aal5mux voice
Related Commands
Command
|
Description
|
encapsulation
|
Configures the ATM adaptation layer (AAL) and encapsulation type for an ATM PVC class.
|
vofr
To enable Voice over Frame Relay (VoFR) on a specific data-link connection identifier (DLCI) and to configure specific subchannels on that DLCI, use the vofr command in frame relay DLCI configuration mode. To disable VoFR on a specific DLCI, use the no form of this command.
Switched Calls
vofr [data cid] [call-control [cid]]
no vofr [data cid] [call-control [cid]]
Switched Calls to Cisco MC3810 Multiservice Concentrators Running Cisco IOS Releases Before 12.0(7)XK and 12.1(2)T
vofr [cisco]
no vofr [cisco]
Cisco-Trunk Permanent Calls
vofr data cid call-control cid
no vofr data cid call-control cid
Cisco-Trunk Permanent Calls to Cisco MC3810 Multiservice Concentrators Running Cisco IOS Releases Before 12.0(7)XK and 12.1(2)T
vofr cisco
no vofr cisco
FRF.11 Trunk Calls
vofr [data cid] [call-control cid]
no vofr [data cid] [call-control cid]
Syntax Description
data
|
(Required for Cisco-trunk permanent calls. Optional for switched calls.) Used to select a subchannel (CID) for data other than the default subchannel, which is 4.
|
cid
|
(Optional) Specifies the subchannel to be used for data. Valid values are from 4 through 255; the default is 4. If data is specified, enter a valid CID.
|
call-control
|
(Optional) Used to specify that a subchannel will be reserved for call-control signaling. This option is not supported on the Cisco MC3810 multiservice concentrator.
|
cisco
|
(Optional) Cisco proprietary voice encapsulation for VoFR with data is carried on CID 4 and call-control on CID 5. This option is required when configuring switched calls or Cisco trunks to Cisco MC3810 multiservice concentrators running Cisco IOS Releases before 12.0(7)XK and 12.1(2)T.
If you are configuring switched calls or Cisco trunks to Cisco MC3810 multiservice concentrators running Cisco IOS Release 12.0(7)XK and 12.1(2)T and later releases, do not use this option.
|
cid
|
(Optional) Specifies the subchannel to be used for call-control signaling. Valid values are from 4 to 255; the default is 5. If call-control is specified and a CID is not entered, the default CID will be used.
|
Defaults
Disabled
Command Modes
Frame relay DLCI configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on Cisco 2600, 3600, and 7200 series routers and Cisco MC3810 multiservice concentrators.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.0(7)XK
|
The use of the cisco option was modified. Beginning in this release, use the cisco option only when configuring connections to Cisco MC3810 concentrators running Cisco IOS Releases before 12.0(7)XK and 12.1(2)T.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
Table 76 lists the different options of the vofr command and which combination of options is used beginning in Cisco IOS Release 12.0(7)XK and 12.1(2)T.
Table 76 Combinations of the vofr Command
Type of Call
|
Command Combination to Use
|
Switched call (user dialed or auto-ringdown) to other routers supporting VoFR
|
vofr [data cid] [call-control [cid]]1
|
Switched call (user dialed or auto-ringdown) to a Cisco MC3810 running Cisco IOS Releases before 12.0(7)XK and 12.1(2)T
|
vofr cisco2
|
Cisco-trunk permanent call (private-line) to other routers supporting VoFR
|
vofr data cid call-control cid
|
Cisco-trunk permanent call (private-line) to a Cisco MC3810 running Cisco IOS Releases before 12.0(7)XK and 12.1(2)T
|
vofr cisco
|
FRF.11 trunk call (private-line) to other routers supporting VoFR
|
vofr [data cid] [call-control cid]3
|
Usage Restrictions for Cisco IOS Releases Before 12.0(7)XK and 12.1(2)T
This section describes restrictions for using the vofr command in releases before Cisco IOS Release 12.0(7)XK and 12.1(2)T. Beginning in Cisco IOS Release 12.0(7)XK and 12.1(2)T, these restrictions no longer apply.
When you use the vofr command without the cisco option, all subchannels on the DLCI are configured for FRF.11 encapsulation. If you enter the vofr command without any keywords or arguments, the data subchannel is CID 4 and there is no call-control subchannel.
Table 77 describes special conditions and restrictions for the use of the vofr command on the Cisco MC3810 running releases before 12.0(7)XK and 12.1(2)T.
Table 77 Using the vofr Command with the Cisco MC3810 Multiservice Concentrator
Type of Call
|
Conditions and Restrictions
|
FRF.11 trunks
|
1. Do not use the cisco option or the call-control option.
2. Use vofr or vofr data cid.
|
Cisco trunks
|
1. Must use vofr cisco.
|
switched-vofr
|
1. Must use vofr cisco.
|
If you select the "data" option, enter a numeric value to complete the command. If you select the call-control option, you do not enter a numeric value if you wish to accept the default call-control subchannel. See the following examples for clarification.
When you use the vofr command on a Cisco MC3810 multiservice concentrator without the "cisco" option, switched calls are not permitted. You can make only permanent FRF.11-trunk calls.
Note
It is not possible to configure the call-control option on a Cisco MC3810 multiservice concentrator. If you configure this option, the setting is ignored.
Examples
The following example, beginning in global configuration mode, shows how to enable VoFR on serial interface 1/1, DLCI 100 on a Cisco 2600 series, 3600 series, or 7200 series router or on a Cisco MC3810 multiservice concentrator. The example configures CID 4 for data; no call-control CID is defined.
frame-relay interface-dlci 100
To configure CID 4 for data and CID 5 for call-control (both defaults), enter the following command:
To configure CID 10 for data and CID 15 for call-control, enter the following command:
vofr data 10 call-control 15
To configure CID 4 for data and CID 15 for call-control, enter the following command:
To configure CID 10 for data and CID 5 for call-control, enter the following command:
vofr data 10 call-control
To configure CID 10 for data with no call-control, enter the following command:
To configure a Cisco router or MC3810 for a VoFR application with an older release of the Cisco MC3810 (before Release 12.0(3)XG), enter the following command:
Related Commands
Command
|
Description
|
class
|
Assigns a VC class to a PVC.
|
frame-relay interface-dlci
|
Assigns a DLCI to a specified Frame Relay subinterface.
|
voice call convert-discpi-to-prog
To convert a disconnect message with a progress indicator (PI) to a progress message, use the voice call convert-discpi-to-prog command in global configuration mode. To restore the default, use the no form of this command.
voice call convert-discpi-to-prog
no voice call convert-discpi-to-prog
Syntax Description
This command has no arguments or keywords.
Defaults
A disconnect message with a PI is not converted to a progress message.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(1)
|
This command was introduced.
|
Usage Guidelines
In Cisco IOS Release 12.2(1), a disconnect message with a PI is always converted to a progress message, even if the disconnect with PI is received after the connect message.
Examples
The following example changes a disconnect with PI to a progress message:
voice call convert-discpi-to-prog
Related Commands
Command
|
Description
|
disc_pi_off
|
Enables an H.323 gateway to disconnect a call when it receives a disconnect message with a PI.
|
voice call send-alert
To enable the terminating gateway to send an alert message instead of a progress message after it receives a call setup message, use the voice call send-alert command in global configuration mode. To restore the default behavior, use the no form of this command.
voice call send-alert
no voice call send-alert
Syntax Description
This command has no arguments or keywords.
Defaults
The terminating gateway sends a progress message after it receives a call Setup message.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)XI4
|
This command was introduced.
|
12.1(5)T
|
This command was not supported in this release.
|
12.1(5.3)T
|
This command was integrated into Cisco IOS Release 12.1(5.3)T.
|
12.2(1)
|
This command was integrated into Cisco IOS Release 12.2.
|
Usage Guidelines
In Cisco IOS Release 12.1(3)XI and later, the terminating gateway sends a Progress message with a progress indicator (PI) after it receives a Setup message. Previously, the gateway responded with an Alert message after receiving a call. In some cases, if the terminating switch does not forward the progress message to the originating gateway, the originating gateway does not cut-through the voice path until a Connect is received and the caller will not hear a ringback tone. In these cases, you can use the voice call send-alert command to make the gateway backward compatible with releases earlier than Cisco IOS Release 12.1(3)XI. If you configure the voice call send-alert command, the terminating gateway sends an Alert message after it receives a Setup message from the originating gateway.
To complete calls from a PRI to an FXS interface, configure the voice call send-alert command on the FXS device.
Examples
The following example configures the gateway to send an Alert message:
Related Commands
Command
|
Description
|
progress_ind
|
Sets a specific PI in call Setup, Progress, or Connect messages from an H.323 VoIP gateway.
|
voice-card
To configure a voice card, use the voice-card command in global configuration mode.
voice-card slot
Syntax Description
slot
|
On the Cisco 2600 and 3600 platforms, a value from 0 to 3 that identifies the physical slot in the chassis in which the voice card is located.
On Cisco MC3810 mutliservice concentrators with one or two HCMs installed, enter 0 only; this applies to the entire chassis.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)XK
|
The command was introduced for the Cisco 2600 and 3600 series.
|
12.0(7)T
|
This command was integrated into Cisco IOS Release 12.0(7)T.
|
12.0(7)XK
|
This command was first supported on the Cisco MC3810 series.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
You can configure codec complexity only in voice-card configuration mode. On the Cisco 2600 and 3600 series, the slot corresponds to the physical slot in the chassis. On the Cisco MC3810 series, the slot is always 0, and all changes made in voice-card configuration mode apply to the entire Cisco MC3810. On Cisco MC3810 series concentrators, this command is available only if the chassis is equipped with one or two HCMs.
Examples
The following example enters voice-card configuration mode for the voice card in slot 1 on a Cisco 2600 or 3600 router:
The following example enters voice-card configuration mode on a Cisco MC3810 concentrator:
Related Commands
Command
|
Description
|
codec complexity
|
Matches the DSP complexity packaging to the codecs to be supported. Codec complexity changes are made in the voice-card configuration mode.
|
voice class busyout
To create a voice class for local voice busyout functions, use the voice class busyout command in global configuration mode. To delete the voice class, use the no form of this command.
voice class busyout tag
no voice class busyout tag
Syntax Description
tag
|
A unique identification number assigned to one voice class. The range is 1 to 10,000.
|
Defaults
No voice class is configured for busyout functions.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
You can apply a busyout voice class to multiple voice ports. You can assign only one busyout voice class to a voice port. If a second busyout voice class is assigned to a voice port, the second voice class replaces the one previously assigned.
If you assign a busyout voice class to a voice port, you may not assign separate busyout commands directly to the voice port, such as busyout monitor serial, busyout monitor ethernet, or busyout monitor probe.
Examples
The following example configures busyout voice class 20, in which the connections to two remote interfaces are monitored by a response time reporter (RTR) probe with a G.711ulaw profile, and voice ports are busied out whenever both links have a packet loss exceeding 10 percent and a packet delay time exceeding 2 seconds:
busyout monitor probe 171.165.202.128 g711u loss 10 delay 2000
busyout monitor probe 171.165.202.129 g711u loss 10 delay 2000
The following example configures busyout voice class 30, in which voice ports are busied out when serial ports 0/0, 1/0, 2/0, and 3/0 go out of service.
busyout monitor serial 0/0
busyout monitor serial 1/0
busyout monitor serial 2/0
busyout monitor serial 3/0
Related Commands
Command
|
Description
|
busyout monitor ethernet
|
Configures a voice port to monitor a local Ethernet interface for events that would trigger a voice-port busyout.
|
busyout monitor probe
|
Configures a voice port to enter the busyout state if an RTR probe signal returned from a remote, IP-addressable interface crosses a specified delay or loss threshold.
|
busyout monitor serial
|
Configures a voice port to monitor a serial interface for events that would trigger a voice-port busyout.
|
show voice busyout
|
Displays information about the voice busyout state.
|
voice class codec
To enter voice-class configuration mode and assign an identification tag number for a codec voice class, use the voice class codec command in global configuration mode. To delete a codec voice class, use the no form of this command.
voice class codec tag
no voice class codec tag
Syntax Description
tag
|
The unique number you assign to the voice class. The valid range is 1 to 10,000. Each tag number must be unique on the router.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(2)XH
|
This command was introduced on the Cisco AS5300.
|
12.0(7)T
|
This command was first supported on the Cisco 2600 and Cisco 3600 series routers.
|
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.
|
Usage Guidelines
This command only creates the voice class for codec selection preference and assigns an identification tag. Use the codec preference command to specify the parameters of the voice class, and use the voice-class codec dial-peer command to apply the voice class to a Voice over IP (VoIP) dial peer.
Note
The voice class codec command in global configuration mode is entered without the hyphen. The voice-class codec command in dial-peer configuration mode is entered with the hyphen.
Examples
The following example shows how to enter voice-class configuration mode and assign a voice class tag number starting from global configuration mode:
After you enter voice-class configuration mode for codecs, use the codec preference command to specify the parameters of the voice class.
The following example creates preference list 99, which can be applied to any dial peer:
codec preference 1 g711alaw
codec preference 2 g711ulaw bytes 80
codec preference 3 g723ar53
codec preference 4 g723ar63 bytes 144
codec preference 5 g723r53
codec preference 6 g723r63 bytes 120
codec preference 7 g726r16
codec preference 8 g726r24
codec preference 9 g726r32 bytes 80
codec preference 11 g729br8
codec preference 12 g729r8 bytes 50
Related Commands
Command
|
Description
|
codec preference
|
Specifies a list of preferred codecs to use on a dial peer.
|
test voice port detector
|
Defines the order of preference in which network dial peers select codecs.
|
voice-class codec (dial peer)
|
Assigns a previously configured codec selection preference list to a dial peer.
|
voice-class codec (dial peer)
To assign a previously configured codec selection preference list (codec voice class) to a Voice over IP (VoIP) dial peer, enter the voice-class codec command in dial-peer configuration mode. To remove the codec preference assignment from the dial peer, use the no form of this command.
voice-class codec tag
no voice-class codec tag
Syntax Description
tag
|
The unique number assigned to the voice class. The valid range for this tag is 1 to 10,000. The tag number maps to the tag number created using the voice class codec global configuration command.
|
Defaults
Dial peers have no codec voice class assigned.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.0(2)XH
|
This command was introduced on the Cisco AS5300.
|
12.0(7)T
|
This command was supported on the Cisco 2600 and 3600 series routers.
|
12.0(7)XK
|
This command was supported on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
You can assign one voice class to each VoIP dial peer. If you assign another voice class to a dial peer, the last voice class assigned replaces the previous voice class.
Note
The voice-class codec command in dial-peer configuration mode is entered with the hyphen. The voice class codec command in global configuration mode is entered without the hyphen.
Examples
The following example shows how to assign a previously configured codec voice class to a dial peer:
Related Commands
Command
|
Description
|
show dial-peer voice
|
Displays the configuration for all dial peers configured on the router.
|
test voice port detector
|
Defines the order of preference in which network dial peers select codecs.
|
voice class codec
|
Enters voice-class configuration mode and assigns an identification tag number for a codec voice class.
|
voice class dualtone
To create a voice class for Foreign Exchange Office (FXO) supervisory disconnect tone detection parameters, use the voice class dualtone command in global configuration mode. To delete the voice class, use the no form of this command.
voice class dualtone tag
no voice class dualtone tag
Syntax Description
tag
|
A unique identification number assigned to one voice class. The range is from 1 to 10,000.
|
Defaults
No voice class is configured for tone detection parameters.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
Use this command first to create the voice class. Then use the supervisory disconnect dualtone voice-class command to assign the voice class to a voice port.
A voice class can define any number of tones to be detected. You need to define a matching tone for each supervisory disconnect tone expected from a PBX or from the public switched telephone network (PSTN).
Examples
The following example configures voice class dualtone 70, which defines one tone with two frequency components, and does not configure a cadence list:
The following example configures voice class dualtone 100, which defines one tone with two frequency components, and configures a cadence list:
cadence-list 1 100 100 300 300
The following example configures voice class dualtone 90, which defines three tones, each with two frequency components, and configures two cadence lists:
cadence-list 1 100 100 300 300 100 200
cadence-list 2 100 200 100 400
Related Commands
Command
|
Description
|
supervisory disconnect dualtone voice-class
|
Assigns a previously configured voice class for FXO supervisory disconnect tone to a voice port.
|
voice class h323
To create an H.323 voice class that is independent of a dial peer and can be used on multiple dial peers, use the voice class h323 command in global configuration mode. To remove the voice class, use the no form of this command.
voice class h323 tag
no voice class h323
Syntax Description
tag
|
Specifies a number to identify the voice class. The valid range for this tag is 1 to 10,000. The tag number must be unique on the router.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(2)T
|
This command was introduced on Cisco 1700 routers, Cisco 2600, 3600 and 7200 series routers, Cisco AS5300 universal access servers, and Cisco uBR910 and uBR924 cable access routers.
|
Usage Guidelines
The voice class h323 command in global configuration mode does not include a hyphen. The voice-class h323 command in dial-peer configuration mode uses a hyphen.
Examples
The following example creates an H.323 voice class labeled 1:
Related Commands
Command
|
Description
|
h225 timeout tcp establish
|
Sets the H.225 TCP timeout value.
|
voice-class h323 (dial peer)
To assign an H.323 voice class to a VoIP dial peer, use the voice-class h323 command in dial-peer configuration mode. To remove the voice class from the dial peer, use the no form of this command.
voice-class h323 tag
no voice-class h323 tag
Syntax Description
tag
|
Specifies a number to identify the voice class. The valid range for this tag is 1 to 10,000. The tag number must be unique on the router.
|
Defaults
The dial peer does not use an H.323 voice class.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1(2)T
|
This command was introduced.
|
Usage Guidelines
The voice class that you assign to the dial peer must be configured using the voice class h323 in global configuration mode. The voice-class h323 command in dial-peer configuration mode uses a hyphen and in global configuration mode does not include a hyphen.
You can assign one voice class to each VoIP dial peer. If you assign another voice class to a dial peer, the last voice class assigned replaces the previous voice class.
Examples
The following example shows how to create an H.323 voice class and then assign it to a dial peer:
Related Commands
Command
|
Description
|
show dial-peer voice
|
Displays the configuration for all dial peers configured on the router.
|
voice class h323
|
Enters voice-class configuration mode and assigns an identification tag number for an H.323 voice class.
|
voice class permanent
To create a voice class for a Cisco trunk or FRF.11 trunk, use the voice class permanent command in global configuration mode. To delete the voice class, use the no form of this command.
voice class permanent tag
no voice class permanent tag
Syntax Description
tag
|
The unique number that you assign to the voice class. The tag number must be unique on the router. The valid range for this tag is 1 to 10,000.
|
Defaults
No voice class is configured.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 series.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
The voice class permanent command can be used for Voice over Frame Relay (VoFR), Voice over ATM (VoATM), and Voice over IP (VoIP) trunks.
Note
The voice class permanent command in global configuration mode is entered without the hyphen. The voice-class permanent command in dial-peer and voice-port configuration modes is entered with the hyphen.
Examples
The following example shows how to create a permanent voice class starting from global configuration mode:
Related Commands
Command
|
Description
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal timing oos
|
Configures the signal timing parameter for the OOS state of a call.
|
signal-type
|
Sets the signaling type for a network dial peer.
|
voice-class permanent
|
Assigns a previously configured voice class for a Cisco trunk or FRF.11 trunk to a network dial peer.
|
voice-class permanent (dial-peer)
To assign a previously configured voice class for a Cisco trunk or FRF.11 trunk to a network dial peer, use the voice-class permanent command in dial-peer configuration mode. To remove the voice-class assignment from the network dial peer, use the no form of this command.
voice-class permanent tag
no voice-class permanent tag
Syntax Description
tag
|
The unique number assigned to the voice class. The tag number maps to the tag number created using the voice class permanent global configuration command. The valid range is from 1 to 10,000.
|
Defaults
Network dial peers have no voice class assigned.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
You can assign one voice class to any given network dial peer. If you assign another voice class to a dial peer, the last voice class assigned replaces the previous voice class.
Note
You cannot assign a voice class to a plain old telephone service (POTS) dial peer.
Note
The voice-class permanent command in dial-peer configuration mode is entered with the hyphen. The voice class permanent command in global configuration mode is entered without the hyphen.
Examples
The following example assigns a previously configured voice class to a Voice over Frame Relay (VoFR) network dial peer:
Related Commands
Command
|
Description
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal timing oos
|
Configures the signal timing parameter for the OOS state of a call.
|
signal-type
|
Sets the signaling type for a network dial peer.
|
voice class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice-class permanent (voice-port)
To assign a previously configured voice class for a Cisco trunk or FRF.11 trunk to a voice port, use the voice-class permanent command in voice-port configuration mode. To remove the voice-class assignment from the voice port, use the no form of this command.
voice-class permanent tag
no voice-class permanent tag
Syntax Description
tag
|
The unique number assigned to the voice class. The tag number maps to the tag number created using the voice class permanent global configuration command. The valid range is from 1 to 10,000.
|
Defaults
Voice ports have no voice class assigned.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(3)T
|
This command was first supported as a voice port configuration command and on Cisco 2600 and 3600 series routers.
|
Usage Guidelines
You can assign one voice class to any given voice port. If you assign another voice class to a voice port, the last voice class assigned replaces the previous voice class.
Note
The voice-class permanent command in voice-port configuration mode is entered with a hyphen. The voice class permanent command in global configuration mode is entered without the hyphen.
Examples
The following example assigns a previously configured voice class to voice port 1/1 in a Cisco MC3810 series concentrator:
The following example assigns a previously configured voice class to voice port 1/1/0 in a Cisco 3600 series router:
Related Commands
Command
|
Description
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal timing oos
|
Configures the signal timing parameter for the OOS state of a call.
|
signal-type
|
Sets the signaling type for a network dial peer.
|
voice class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice confirmation-tone
To disable the two-beep confirmation tone for private line, automatic ringdown (PLAR) or PLAR off premises extension (OPX) connections, use the voice confirmation-tone command in voice-port configuration mode. To enable the two-beep confirmation tone, use the no form of this command.
voice confirmation-tone
no voice confirmation-tone
Syntax Description
This command has no arguments or keywords.
Defaults
The two-beep confirmation tone is heard on the PLAR or PLAR OPX connection.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
This command applies only to the Cisco MC3810 multiservice concentrator.
Use this command to disable the two-beep confirmation tone that a caller hears when picking up the handset for PLAR and PLAR OPX connections. This command is valid only if the voice-port connection command is set to PLAR or PLAR OPX.
Examples
The following example disables the two-beep confirmation tone on voice port 1/1 on the Cisco MC3810 multiservice concentrator:
Related Commands
Command
|
Description
|
connection
|
Specifies a connection mode for a voice port.
|
voice-encap
This command was added in Cisco IOS Release 11.3(1)MA on the Cisco MC3810 multiservice concentrator. This command is not supported in Cisco IOS Release 12.2.
voice-group
This command was added in Cisco IOS Release 11.3(1)MA for the Cisco MC3810 multiservice concentrator. This command is not supported in Cisco IOS Release 12.2.
voice hunt
To configure an originating or tandem router so that it continues dial-peer hunting if it receives a user-busy disconnect code from a destination router, use the voice hunt command in global configuration mode. To configure the router so that it stops dial-peer hunting if it receives a user-busy disconnect code (the default option), use the no form of this command.
voice hunt {user-busy | invalid-number | unassigned-number}
no voice {user-busy | invalid-number | unassigned-number}
Syntax Description
user-busy
|
Router continues to dial-peer hunting if it receives a user-busy disconnect cause code from a destination router.
|
invalid-number
|
Router stops dial-peer hunting if it receives a an invalid-number disconnect cause code from a destination router.
|
unassigned-number
|
Router stops dial-peer hunting if it receives an unassigned-number disconnect cause code from a destination router.
|
Defaults
The default depends on the disconnect cause code. By default, the router stops dial-peer hunting if it receives the user-busy disconnect cause code. By default, the router continues dial-peer hunting if it receives an invalid-number, or an unassigned-number disconnect cause code.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced on the Cisco 2600 series and 3600 series routers, and the Cisco MC3810 multiservice concentrator for Voice over Frame Relay (VoFR). It was also supported for Voice over IP (VoIP) on the 2600 series and 3600 series routers.
|
12.0(7)T
|
This command was first supported on the Cisco AS5300 and Cisco AS5800 for VoIP.
|
12.0(7)XK
|
This command was first used for VoIP on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
The support for VoIP on the Cisco MC3810 multiservice concentrator was implemented in Cisco IOS Release 12.1(2)T.
|
12.1(3)XI
|
The invalid-number and unassigned-number keywords were added, and the command name was changed to voice hunt.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Usage Guidelines
This command applies to routers that act as originating or tandem nodes in a Voice over IP, Voice over Frame Relay, or Voice over ATM environment.
This command is used for a configuration in which an originating or tandem router is configured with multiple dial peer entries that route a call to the same destination number, but on different destination routers. In this configuration, after all routes to the first router entry in the dial-peer list are active, a new call will not "roll over" to the next router in the dial-peer list.
This failure to route to the second destination router happens when the bandwidth on the voice interface is greater than the maximum capacity of the first destination router. This condition allows the originating or tandem router to attempt to place a new call to the first destination router because it has indications from the first destination router that there is more capacity based on the bandwidth setting. When the first destination router receives the call, if all of the ports are in use, the destination router returns a "user-busy" disconnect reason code to the originating or tandem router.
The originating or tandem router interprets the disconnect reason code as "unavailable destination" for the call and returns a busy tone to the initiating caller.
The originating or tandem router fails to try other routers in the dial-peer list after receiving a "user disconnect" reason code, and so it terminates the call attempt. By using this command, you can perform dial-peer hunting on multiple destination routers even if the originating or tandem router receives a "user-busy" disconnect reason code from one of the destination routers.
Examples
The following example displays configuring the originating or tandem router to continue dial-peer hunting if it receives a "user-busy" disconnect code from a destination router:
The following example displays configuring the originating or tandem router to continue dial-peer hunting if it receives an "invalid-number" disconnect code from a destination router:
voice hunt invalid-number
Related Commands
Command
|
Description
|
huntstop
|
Disables all further dial-peer hunting if a call fails when using hunt groups.
|
preference
|
Indicates the preferred order of a dial peer within a rotary hunt group.
|
voice local-bypass
To configure local calls to bypass the digital signal processor (DSP), use the voice local-bypass command in global configuration mode. To direct local calls through the DSP, use the no form of this command.
voice local-bypass
no voice local-bypass
Syntax Description
This command has no arguments or keywords.
Defaults
Local calls bypass the DSP.
Command Modes
Global configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced.
|
12.0(7)XK
|
This command was implemented on the Cisco 2600 and 3600 series routers and the MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
Local calls (calls between voice ports on a router or concentrator) normally bypass the DSP to minimize use of system resources. Use the no form of the voice local-bypass command if you need to direct local calls through the DSP. Input gain and output attenuation can be configured only if calls are directed through the DSP.
Examples
The following example configures a Cisco MC3810 multiservice concentrator, 2600 or 3600 series routers to pass local calls through the DSP:
Related Commands
Command
|
Description
|
input gain
|
Configures a specific input gain value.
|
output attenuation
|
Configures a specific output attenuation value.
|
voice-port
To enter voice-port configuration mode, use the voice-port command in global configuration mode.
Cisco 1750 Router
voice-port slot-number/port
Cisco 2600 and Cisco 3600 Series Router
voice-port {slot-number/subunit-number/port} | {slot/port:ds0-group-no}
Cisco MC3810 Multiservice Concentrator
voice-port slot/port
Cisco AS5300 Universal Access Server
voice-port controller number:D
Cisco AS5800 Universal Access Server
voice-port {shelf/slot/port:D} | {shelf/slot/parent:port:D}
Cisco 7200 Series Router
voice-port {slot/port:ds0-group-no} | {slot-number/subunit-number/port}
Syntax Description
For the Cisco 1750 Router:
slot-number
|
Slot number in the router in which the voice interface card (VIC) is installed. Valid entries are from 0 to 2, depending on the slot in which it has been installed.
|
port
|
Indicates the voice port. Valid entries are 0 or 1.
|
For the Cisco 2600 and Cisco 3600 Series Routers:
slot-number
|
Slot number in the Cisco router in which the VIC is installed. Valid entries are from 0 to 3, depending on the slot where it has been installed.
|
subunit-number
|
Subunit on the VIC in which the voice port is located. Valid entries are 0 or 1.
|
port
|
Voice port number. Valid entries are 0 or 1.
|
slot
|
The router location in which the voice port adapter is installed. Valid entries are from 0 to 3.
|
port
|
Indicates the voice interface card location. Valid entries are 0 or 3.
|
dso-group-no
|
Indicates the defined DS0 group number. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card.
|
For the Cisco MC3810 Multiservice Concentrator:
slot/port
|
The slot argument specifies the slot number in the Cisco router in which the VIC is installed. The only valid entry is 1.
The port variable specifies the voice port number. Valid ranges are as follows:
• Analog voice ports: from 1 to 6.
• Digital voice port:
• Digital T1: from 1 to 24.
• Digital E1: from 1 to 15, and from 17 to 31.
|
For the Cisco AS5300 Universal Access Server:
controller-number
|
Specifies the T1 or E1 controller.
|
:D
|
Indicates the D channel associated with ISDN PRI.
|
For the Cisco AS5800 Universal Access Server:
shelf/slot/port
|
Specifies the T1 or E1 controller on the T1 card. Valid entries for the shelf argument are 0 to 9999. Valid entries for the slot value is 0 to 11. Valid entries for the port variable is 0 to 11.
|
shelf/slot/parent:port
|
Specifies the T1 controller on the T3 card. Valid entries for the shelf argument are 0 to 9999. Valid entries for the slot argument are 0 to 11. Valid entries for the port argument are 1 to 28. The value for the parent argument is always 0.
|
:D
|
Indicates the D channel associated with ISDN PRI.
|
For the Cisco 7200 Series Router:
slot
|
The router location in which the voice port adapter is installed. Valid entries are from 0 to 3.
|
port
|
Indicates the VIC location. Valid entries are 0 or 1.
|
dso-group-no
|
Indicates the defined DS0 group number. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card.
|
slot-number
|
Indicates the slot number in the Cisco router in which the VIC is installed. Valid entries are from 0 to 3, depending on the slot in which it has been installed.
|
subunit-number
|
Indicates the subunit on the VIC in which the voice port is located. Valid entries are 0 or 1.
|
port
|
Indicates the voice port number. Valid entries are 0 or 1.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
11.3(3)T
|
Support for Cisco 2600 series routers was added.
|
12.0(3)T
|
Support for the Cisco AS5300 access server was added.
|
12.0(7)T
|
Support for the Cisco AS5800 universal access server, the Cisco 7200 series router, and the Cisco 1750 router was added. Arguments for the Cisco 2600 and Cisco 3600 series router were added.
|
Usage Guidelines
Use the voice-port global configuration command to switch to voice-port configuration mode from global configuration mode. Use the exit command to exit voice-port configuration mode and return to global configuration mode.
Examples
The following example accesses voice-port configuration mode for port 0, located on subunit 0 on a voice interface card installed in slot 1 for the Cisco 3600 series:
The following example accesses the voice-port configuration mode for digital voice port 24 on a Cisco MC3810 with a digital voice module (DVM) installed:
The following example accesses the voice-port configuration mode for the Cisco AS5300:
The following example accesses the voice-port configuration mode for the Cisco AS5800 (T1 card):
The following example accesses the voice-port configuration mode for the Cisco AS5800 (T3 card):
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice encapsulation.
|
voice-port busyout
To place all voice ports associated with a serial or ATM interface into a busyout state, use the voice-port busyout command in interface configuration mode. To remove the busyout state on the voice ports associated with this interface, use the no form of this command.
voice-port busyout
no voice-port busyout
Syntax Description
This command has no arguments or keywords.
Defaults
The voice ports on the interface are not in busyout state.
Command Modes
Interface configuration
Command History
Release
|
Modification
|
12.0(3)T
|
This command was introduced on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
This command busies out all voice ports associated with the interface, except any voice ports configured to busy out under specific conditions using the busyout monitor and busyout seize commands.
Examples
The following example places the voice ports associated with serial interface 1 into busyout state:
interface serial 1
voice-port busyout
The following example places the voice ports associated with ATM interface 0 into busyout state:
Related Commands
Command
|
Description
|
busyout forced
|
Forces a voice port on the Cisco MC3810 multiservice concentrator into the busyout state.
|
busyout monitor
|
Places a voice port on the Cisco MC3810 multiservice concentrator into the busyout monitor state.
|
busyout seize
|
Changes the busyout action for an FXO or FXS voice port.
|
show voice busyout
|
Displays information about the voice busyout state on the Cisco MC3810 multiservice concentrator.
|
voice rtp send-recv
To establish a two-way voice path when the Real-Time Transport Protocol (RTP) channel is opened, use the voice rtp send-recv command in global configuration mode. To restore the default condition, use the no form of this command.
voice rtp send-recv
no voice rtp send-recv
Syntax Description
This command has no arguments or keywords.
Defaults
The voice path is cut-through in only the backward direction when the RTP channel is opened.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(5)T
|
This command was introduced on Cisco 2600, Cisco 3600, Cisco 7200, and Cisco 7500 series routers, Cisco AS5300 and Cisco AS5800 universal access servers, and Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
The voice rtp send-recv command should be enabled only when the voice path must be cut-through (established) in both the backward and forward directions before a Connect message is received from the destination switch. The voice rtp send-recv command affects all Voice over IP (VoIP) calls when it is enabled.
Examples
The following example enables the voice path to cut-through in both directions when the RTP channel is opened:
voice service
To specify the voice encapsulation type, use the voice service command in global configuration mode.
voice service {voip | voatm}
Syntax Description
voip
|
Specifies Voice over IP (VoIP) parameters.
|
voatm
|
Specifies Voice over ATM (VoATM) parameters.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(1)XA
|
This command was introduced for VoATM for the Cisco MC3810 multiservice concentrators.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T for the Cisco MC3810 multiservice concentrators.
|
12.1(3)T
|
This command was integrated into Cisco IOS Release 12.1(3)T for VoIP on the Cisco 2600 series routers, Cisco 3600 series routers, and Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
Use the voice service command to switch to voice-service configuration mode from global configuration mode and to specify a voice encapsulation type. Use the exit command to exit the voice-service configuration mode and return to the global configuration mode.
Examples
The following example shows how to access voice-service configuration mode and specify VoIP parameters, beginning in global configuration mode:
Related Commands
Command
|
Description
|
fax protocol
|
Specifies the global default fax protocol for all the VoIP dial peers.
|
modem passthrough
|
Configures modem pass-through over VoIP.
|
voice vad-time
To change the minimum silence detection time for voice activity detection (VAD), use the voice vad-time command in global configuration mode. To restore the default value, use the no form of this command.
voice vad-time milliseconds
no voice vad-time
Syntax Description
milliseconds
|
The waiting period, in milliseconds, before silence detection and suppression of voice-packet transmission. The range is 250 to 65,536. The default is 250.
|
Defaults
250 milliseconds
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco 2600, 3600, and MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
The voice vad-time command affects all voice ports on a router or concentrator, but it does not affect calls already in progress.
You can use this command in transparent common channel signaling (CCS) applications in which you want VAD to activate when the voice channel is idle, but not during active calls. With a longer silence detection delay, VAD reacts to the silence of an idle voice channel, but not to pauses in conversation.
The voice vad-time command does not affect voice codecs that have ITU-standardized built-in VAD features—for example, G.729B, G.729AB, G.723.1A. The VAD behavior and parameters of these codecs are defined exclusively by the applicable ITU standard.
Examples
The following example configures a 20-second delay before VAD silence detection is enabled:
Related Commands
Command
|
Description
|
vad (dial peer)
|
Enables voice activity detection on a network dial peer.
|
voip-incoming translation-rule
To set the incoming translation rule for calls that originate from H.323-compatible clients, use the voip-incoming translation-rule command in global configuration mode. To disable the incoming translation rule, use the no form of this command.
voip-incoming translation-rule name-tag {calling-number | called-number}
no voip-incoming translation-rule name-tag {calling-number | called-number}
Syntax Description
name-tag
|
The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. The range is 1 through 2,147,483,647.
|
calling-number
|
The automatic number identification (ANI) number or the number of the calling party.
|
called-number
|
The Dial Number Information Service (DNIS) number or the number of the called party.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR1
|
This command was introduced for Voice over IP on the Cisco AS5300.
|
12.0(7)XK
|
This command was first supported for Voice over IP on the Cisco 2600 series and Cisco 3600 series routers, and on the Cisco MC3810 multiservice concentrator.
|
12.1(1)T
|
This command was first supported on the T train for Voice over IP on the Cisco 1750, Cisco 2600 series, and Cisco 3600 series routers, the Cisco AS5300 universal access server, and the Cisco 7200 series and Cisco 7500 series routers.
|
12.1(2)T
|
This command was first supported on the T train for Voice over IP on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
With this command, all IP-based calls are captured and handled, depending on either the calling number or the called number to the specified tag name.
Examples
The following example identifies the rule set for calls that originate from H.323-compatible clients:
voip-incoming translation-rule 5 called-number
Related Commands
Command
|
Description
|
numbering-type
|
Matches one number type for a dial-peer call leg.
|
rule
|
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
|
show translation-rule
|
Displays the contents of all the rules that have been configured for a specific translation name.
|
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.
|
zone access
To configure the accessibility of your local-zone zone, use the zone access command in gatekeeper configuration mode. To remove any accessibility configurations, use the no form of this command.
zone access local-zone-name {default | remote-zone remote-zone-name}{direct | proxied}
no zone access local-zone-name remote-zone remote-zone-name
Syntax Description
local-zone-name
|
Name of local zone (synonymous with local gatekeeper).
|
default
|
Use with the direct or proxied keyword to define the mode of behavior for all remote zones that have not been specially named using the remote-zone remote-zone-name keyword and argument combination.
|
remote-zone remote-zone-name
|
Name of remote zone (synonymous with remote gatekeeper) for which a special mode of behavior is defined.
|
direct
|
Configures direct calls (without use of proxies) between endpoints. The local zone (or gatekeeper) offers the local endpoint IP address instead of the IP address of a local proxy.
|
proxied
|
Configures calls using proxies between endpoints. The local zone (or gatekeeper) offers the IP address of a local proxy instead of the local endpoint address.
|
Defaults
The local zone allows proxied access for all remote zones.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on Cisco 2500 and 3600 series routers.
|
Usage Guidelines
By default, a gatekeeper will offer a local proxy IP address when queried by a remote gatekeeper about a target local endpoint. This is considered proxied access. By using the zone access command, you can configure the local gatekeeper to offer the local endpoint address instead of the local proxy address. This is considered direct access.
Note
The zone access command, configured on your local gatekeeper, affects onlythe use of proxies for incoming calls (that is, it does not affect the use of local proxies for outbound calls). When originating a call, a gatekeeper will use a proxy only if the remote gatekeeper offers a proxy at the remote end. A call between two endpoints in the same zone will always be a direct (nonproxied) call.
You can define the accessibility behavior of a local zone relative to certain remote zones using the remote-zone remote-zone-name keyword and argument combination with the direct or proxied keyword. You can define the default behavior of a local zone relative to all other remote zones using the default keyword with the direct or proxied keywords. To remove an explicitly named remote zone so that it is governed by the default-behavior rule, use the no zone access command.
Examples
The following example allows direct access to the local zone eng.xyz.com from remote zones within xyz corporation. All other remote locations will have proxied access to eng.xzy.com.
zone local eng.xyz.com xyz.com
zone access eng.xyz.com remote-zone mfg.xyz.com direct
zone access eng.xyz.com remote-zone mktg.xyz.com direct
zone access eng.xyz.com remote-zone sales.xyz.com direct
zone access eng.xyz.com default proxied
The following example supposes that only local gatekeepers within xyz.com have direct access to each other because your corporation has firewalls or you do not advertise your gatekeepers externally. You have excellent Quality of Service (QoS) within your corporate network, except for a couple of foreign offices. In this case, use proxies with the foreign offices (in Milan and Tokyo) and nowhere else.
zone local sanjose.xyz.com xyz.com
zone access sanjose.xyz.com default direct
zone access sanjose.xyz.com remote-zone milan.xyz.com proxied
zone access sanjose.xyz.com remote-zone tokyo.xyz.com proxied
Related Commands
Command
|
Description
|
show proxy h323 calls
|
Displays a list of each active call on the proxy.
|
zone local
|
Specifies a zone controlled by a gatekeeper.
|
zone bw
To set the maximum bandwidth allowed in a gatekeeper zone at any one time, use the zone bw command in gatekeeper configuration mode. To remove the maximum bandwidth setting and make the bandwidth unlimited, use the no form of this command.
zone bw gatekeeper-name max-bandwidth
no zone bw gatekeeper-name max-bandwidth
Syntax Description
gatekeeper-name
|
Name of the gatekeeper that controlsthe zone.
|
max-bandwidth
|
Maximum bidirectional bandwidth, in kbps, allowed in the zone at any one time.
|
Defaults
Bandwidth is unlimited.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on Cisco 2500 and 3600 series routers.
|
Examples
The following example sets the maximum bandwidth to 1000 kbps for zone gk1:
Related Commands
Command
|
Description
|
show proxy h323 calls
|
Displays a list of each active call on the proxy.
|
zone local
To specify a zone controlled by a gatekeeper, use the zone local command in gatekeeper configuration mode. To remove a zone controlled by a gatekeeper, use the no form of this command.
zone local gatekeeper-name domain-name [ras-IP-address]
no zone local gatekeeper-name domain-name
Syntax Description
gatekeeper-name
|
The gatekeepers name or zone name. This is usually the fully domain-qualified host name of the gatekeeper. For example, if the domain-name is cisco.com, the gatekeeper-name might be gk1.cisco.com. However, if the gatekeeper is controlling multiple zones, the gatekeeper-name for each zone should be some unique string that has a mnemonic value.
|
domain-name
|
The domain name served by this gatekeeper.
|
ras-IP-address
|
(Optional) The IP address of one of the interfaces on the gatekeeper. When the gatekeeper responds to gatekeeper discovery messages, it signals the endpoint or gateway to use this address in future communications.
Note Setting this address for one local zone makes it the address used for all local zones.
|
Defaults
No local zone is defined.
Note
The gatekeeper cannot operate without at least one local zone definition. Without local zones, the gatekeeper goes to an inactive state when the no shutdown command is issued.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on Cisco 2500 and 3600 series routers.
|
Usage Guidelines
Multiple local zones can be defined. The gatekeeper manages all configured local zones. Intrazone and interzone behavior remains the same (zones are controlled by the same or different gatekeepers).
Only one ras-IP-address argument can be defined for all local zones. You cannot configure each zone to use a different RAS IP address. If you define this in the first zone definition, you can omit it for all subsequent zones, which automatically pick up this address. If you set it in a subsequent zone local command, it changes the RAS address of all previously configured local zones as well. Once defined, you can change it by reissuing any zone local command with a different ras-IP-address argument.
If the ras-IP-address argument is a Hot Standby Router Protocol (HSRP) virtual address, it automatically puts the gatekeeper into HSRP mode. In this mode, the gatekeeper assumes STANDBY or ACTIVE status according to whether the HSRP interface is on STANDBY or ACTIVE status.
You cannot remove a local zone if there are endpoints or gateways registered in it. To remove the local zone, shut down the gatekeeper first, which forces unregistration.
Multiple zones are controlled by multiple logical gatekeepers on the same Cisco IOS platform.
The maximum number of local zones defined in a gatekeeper should not exceed 100.
This command can also be used to change the IP address used by the gatekeeper.
Examples
The following example creates a zone controlled by a gatekeeper in the domain called cisco.com:
zone local gk1.cisco.com cisco.com
Related Commands
Command
|
Description
|
show proxy h323 calls
|
Displays a list of each active call on the proxy.
|
zone subnet
|
Specifies a zone controlled by a gatekeeper.
|
zone prefix
To add a prefix to the gatekeeper zone list, use the zone prefix command in gatekeeper configuration mode. To remove knowledge of a zone prefix, use the no form of this command with the gatekeeper name and prefix. To remove the priority assignment for a specific gateway, use the no form of this command with the gw-priority option.
zone prefix gatekeeper-name e164-prefix [blast | seq] [gw-priority priority gw-alias
[gw-alias, ...]]
no zone prefix gatekeeper-name e164-prefix [blast | seq] [gw-priority priority gw-alias
[gw-alias, ...]]
Syntax Description
gatekeeper-name
|
The name of a local or remote gatekeeper, which must have been defined by using the zone local or zone remote command.
|
e164-prefix
|
An E.164 prefix in standard form followed by dots (.). Each dot represents a number in the E.164 address. For example, 212....... is matched by 212 and any seven numbers.
Note Although a dot representing each digit in an E.164 address is the preferred configuration method, you can also enter an asterisk (*) to match any number of digits.
|
blast
|
(Optional) If you list multiple hopoffs, this indicates that the LRQs should be sent simultaneously to the gatekeepers based on the order in which they were listed. The default is seq.
|
seq
|
(Optional) If you list multiple hopoffs, this indicates that the LRQs should be sent sequentially to the gatekeepers based on the order in which they were listed. The default is seq.
|
gw-priority pri-0-to-10 gw-alias
|
(Optional) Use the gw-priority option to define how the gatekeeper selects gateways in its local zone for calls to numbers beginning with prefix e164-prefix. Do not use this option to set priority levels for a prefix assigned to a remote gatekeeper.
Use values from 0 to 10. A 0 value prevents the gatekeeper from using the gateway gw-alias for that prefix. Value 10 places the highest priority on gateway gw-alias. If you do not specify a priority value for a gateway, the value 5 is assigned.
To assign the same priority value for one prefix to multiple gateways, list all the gateway names after the pri-0-to-10 value.
The gw-alias name is the H.323 ID of a gateway that is registered or will register with the gatekeeper. This name is set on the gateway with the h323-gateway voip h.323-id command.
|
Defaults
No knowledge of its own prefix or the prefix of any other zone is defined.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(6)Q
|
This command was introduced.
|
11.3(7)NA
|
This command was modified for H.323 Version 1.
|
12.0(5)T
|
The display format was modified for H.323 Version 2.
|
Usage Guidelines
A gatekeeper can handle more than one zone prefix, but a zone prefix cannot be shared by more than one gatekeeper. If you have defined a zone prefix as being handled by a gatekeeper and now define it as being handled by a second gatekeeper, the second assignment cancels the first.
If you need a gatekeeper to handle more than one prefix, but for cost reasons you want to be able to group its gateways by prefix usage, there are two ways to do it.
The first method is simpler, has less overhead, and is recommended if your gateways can be divided into distinct groups, in which each group is to be used for a different set of prefixes. For instance, if a group of gateways is used for calling area codes 408 and 650, and another group is used for calling area code 415, you can use this method. In this case, you define a local zone for each set of prefixes, and have the group of gateways to be used for that set of prefixes register with that specific local zone. Do not define any gateway priorities. All gateways in each local zone are treated equally in the selection process.
However, if your gateways cannot be cleanly divided into nonintersecting groups (for instance if one gateway is used for calls to 408 and 415 and another gateway is used for calls to 415 and 650, and so on), you can put all these gateways in the same local zone and use the gw-priority option to define which gateways will be used for which prefixes.
When choosing a gateway, the gatekeeper first looks for the longest zone prefix match; then it uses the priority and the gateway status to select from the gateways. If all gateways are available, the gatekeeper chooses the highest priority gateway. If all the highest priority gateways are busy (see the gateway resource threshold command), a lower priority gateway is selected.
Note
The zone prefix command matches a prefix to a gateway. It does not register the gateway. The gateway must register with the gatekeeper before calls can be completed through that gateway.
Examples
The following example shows how you can define multiple local zones for separating your gateways:
zone local gk408or650 xyz.com
zone prefix gk408or650 408.......
zone prefix gk408or650 650.......
zone prefix gk415 415.......
Now you need to configure all the gateways to be used for area codes 408 or 650 to register with gk408or650 and all gateways to be used for area code 415 to register with gk415. On Cisco voice gateways, you configure the gateways to register with the appropriate gatekeepers by using the h323 voip id command.
The following example shows how you can put all your gateways in the same zone but use the gw-priority keyword to determine which gateways will be used for calling different area codes:
zone local localgk xyz.com
zone prefix localgk 408.......
zone prefix localgk 415....... gw-priority 10 gw1 gw2
zone prefix localgk 650....... gw-priority 0 gw1
The commands shown accomplish the following tasks:
•
Domain xyz.com is assigned to gatekeeper localgk.
•
Prefix 408...... is assigned to gatekeeper localgk, and no gateway priorities are defined for it; therefore, all gateways registering to localgk can be used equally for calls to the 408 area code. No special gateway lists are built for the 408....... prefix; selection is made from the master list for the zone.
•
The prefix 415....... is added to gatekeeper localgk, and priority 10 is assigned to gateways gw1 and gw2.
•
Prefix 650....... is added to gatekeeper localgk, and priority 0 is assigned to gateway gw1.
A priority 0 is assigned to gateway gw1 to exclude it from the gateway pool for prefix 650........ When gateway gw2 registers with gatekeeper localgk, it is added to the gateway pool for each prefix as follows:
•
For gateway pool for 415......., gateway gw2 is set to priority 10.
•
For gateway pool for 650......., gateway gw2 is set to priority 5.
The following example changes gateway gw2 from priority 10 for zone 415....... to the default priority 5:
no zone prefix localgk 415....... gw-pri 10 gw2
The following example changes both gateways gw1 and gw2 from priority 10 for zone 415....... to the default priority 5:
no zone prefix localgk 415....... gw-pri 10 gw1 gw2
In the preceding example, the prefix 415....... remains assigned to gatekeeper localgk. All gateways that do not specify a priority level for this prefix are assigned a default priority of 5. The following example removes the prefix and all associated gateways and priorities from this gatekeeper:
no zone prefix localgk 415.......
Related Commands
Command
|
Description
|
register
|
Configures a gateway to register or deregister a fully qualified dial-peer E.164 address with a gatekeeper.
|
resource threshold
|
Configures a gateway to report H.323 resource availability to the gatekeeper of the 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.
|
zone local
|
Specifies a zone controlled by a gatekeeper.
|
zone remote
|
Statically specifies a remote zone if DNS is unavailable or undesirable.
|
zone remote
To statically specify a remote zone if domain name service (DNS) is unavailable or undesirable, use the zone remote command in gatekeeper configuration mode. To remove the remote zone, use the no form of this command.
zone remote other-gatekeeper-name other-domain-name other-gatekeeper-ip-address
[port-number]
no zone remote other-gatekeeper-name other-domain-name other-gatekeeper-ip-address
[port-number]
Syntax Description
other-gatekeeper-name
|
Name of the remote gatekeeper.
|
other-domain-name
|
Domain name of the remote gatekeeper.
|
other-gatekeeper-ip-address
|
IP address of the remote gatekeeper.
|
port-number
|
(Optional) RAS signaling port number for the remote zone. Value ranges from 1 to 65,535. If this is not set, the default is the well-known RAS port number 1719.
|
Defaults
No remote zone is defined. DNS will locate the remote zone.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on Cisco 2500 and 3600 series routers.
|
12.0(3)T
|
This command was integrated into Cisco IOS Release 12.0(3)T.
|
Usage Guidelines
All gatekeepers do not have to be in DNS. For those that are not, use the zone remote command so that the local gatekeeper knows how to access them. In addition, you may wish to improve call response time slightly for frequently accessed zones. If the zone remote command is configured for a particular zone, you do not need to make a DNS lookup transaction.
The maximum number of zones defined on a gatekeeper varies depending on the mode or the call model or both. For example, a directory gatekeeper may be in the mode of being responsible for forwarding location request (LRQ) messages and not handling any local registrations and calls; the call model might be E.164 addressed calls instead of H.323-ID addressed calls.
For a directory gatekeeper that does not handle local registrations and calls, the maximum remote zones defined should not exceed 10,000; an additional 4 MB of memory is required to store this maximum number of remote zones.
For a gatekeeper that handles local registrations and only E.164 addressed calls, the number of remote zones defined should not exceed 2000.
For a gatekeeper that handles H.323-ID calls, the number of remote zones defined should not exceed 200.
Examples
The following example configures the local gatekeeper to reach targets of the form xxx.cisco.com by sending queries to the gatekeeper named sj3.cisco.com at IP address 10.1.1.12:
zone remote sj3.cisco.com cisco.com 10.1.1.12
Related Commands
Command
|
Description
|
show proxy h323 calls
|
Displays a list of each active call on the proxy.
|
zone local
|
Specifies a zone controlled by a gatekeeper.
|
zone subnet
To configure a gatekeeper to accept discovery and registration messages sent by endpoints in designated subnets, use the zone subnet command in gatekeeper configuration mode. To disable the gatekeeper from acknowledging discovery and registration messages from subnets or to remove subnets entirely, use the no form of this command.
zone subnet local-gatekeeper-name {default | subnet-address {/bits-in-mask | mask-address}}
enable
no zone subnet local-gatekeeper-name {default | subnet-address {/bits-in-mask | mask-address}}
enable
Syntax Description
local-gatekeeper-name
|
Name of the local gatekeeper.
|
default
|
Applies to all other subnets that are not specifically defined by the zone subnet command.
|
subnet-address
|
Address of the subnet being defined.
|
/bits-in-mask
|
Number of bits of the mask to be applied to the subnet address.
|
mask-address
|
Mask (in dotted string format) to be applied to the subnet address.
|
enable
|
Gatekeeper accepts discovery and registration from the specified subnets.
|
Defaults
The local gatekeeper accepts discovery and registration requests from all subnets. If the request specifies a gatekeeper name, it must match the local gatekeeper name or the request will not be accepted.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on Cisco 2500 and 3600 series routers.
|
12.0(3)T
|
This command was integrated into Cisco IOS Release 12.0(3)T.
|
Usage Guidelines
You can use the zone subnet command more than once to create a list of subnets controlled by a gatekeeper. The subnet masks do not have to match actual subnets in use at your site. For example, to specify a particular endpoint, you can supply its address with a 32-bit netmask.
Examples
The following example starts by disabling the gatekeeper, gk1.cisco.com, from accepting discovery and registration messages from all subnets. Next, gk1.cisco.com is configured to accept discovery and registration messages from all H.323 nodes on the subnet 172.21.127.0.
In addition, gk1.cisco.com is configured to accept discovery and registration messages from a particular endpoint with the IP address 172.21.128.56.
no zone subnet gk1.cisco.com default enable
zone subnet gk1.cisco.com 172.21.127.0/24 enable
zone subnet gk1.cisco.com 172.21.128.56/32 enable
Related Commands
Command
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Description
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show gatekeeper zone status
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Displays the status of zones related to a gatekeeper.
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zone local
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Specifies a zone controlled by a gatekeeper.
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