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Cisco IOS Voice Command Reference
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About Cisco IOS Software Documentation
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Using Cisco IOS Software
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Commands: A
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Commands: B
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Commands:cac master through call application stats
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Commands: call application voice through call-denial
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Commands: call fallback through called-number (dial-peer)
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Commands: caller-id (dial peer) through ccm-manager switchover-to-backup
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Commands: ccs connect (controller) through clear vsp statistics
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Commands: clid through cptone
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Commands: default (MGCP profile) through direct-inward-dial
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Commands: disable-early-media 180 through dualtone
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Commands: E
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Commands: F
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Commands: G
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Commands: H
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Commands: icpif through irq global-request
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Commands: isdn bind-l3 through ixi transport http
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Commands: K
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Commands: L
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Commands: map q850-cause through mgcp package-capability
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Commands: mgcp persistent through mmoip aaa send-id secondary
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Commands: mode (ATM/T1/E1 controller) through mwi-server
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Commands: N
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Commands: O
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Commands: package through pattern
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Commands: permit hostname through proxy h323
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Commands: Q
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Commands: R
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Commands: sccp through service-type call-check
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Commands: session through sgcp tse payload
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Commands: show aal2 profile through show call filter match-list
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Commands: show call history fax through show debug condition
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Commands: show dial-peer video through show gatekeeper zone status
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Commands: show gateway through show modem relay statistics
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Commands: show mrcp client session active through show sgcp statistics
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Commands: show sip service through show trunk hdlc
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Commands: show vdev through show voice statistics memory-usage
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Commands: show voice translation-profile through shutdown (voice-port)
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Commands: signal through srv version
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Commands: ss7 mtp2-variant through switchover method
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Commands: target carrier-id through threshold noise
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Commands: timeout (auto-config application) through timing clear-wait
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Commands:timing delay-duration through type (voice)
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Commands: U
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Commands: vad (dial peer) through voice class dualtone-detect-params
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Commands: voice class h323 through vxml tree memory
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Commands: W
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Commands: Z
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Feedback
Table Of Contents
incoming called-number (call filter match list)
incoming called-number (dial peer)
incoming calling-number (call filter match list)
incoming secondary-called-number
incoming signaling remote ipv4
interface event-log error only
interface event-log max-buffer-size
Cisco IOS Voice Commands:
I
This chapter contains commands to configure and maintain Cisco IOS voice applications. The commands are presented in alphabetical order. Some commands required for configuring voice may be found in other Cisco IOS command references. Use the master index of commands or search online to find these commands.
For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice Configuration Library.
icpif
To specify the Calculated Planning Impairment Factor (ICPIF) for calls sent by a dial peer, use the icpif command in dial peer configuration mode. To reset to the default, use the no form of this command.
icpif number
no icpif
Syntax Description
number
Integer, expressed in equipment impairment factor units, that specifies the ICPIF value. Range is 0 to 55. The default is 20.
Command Default
20
Command Modes
Dial peer configuration
Command History
Usage Guidelines
This command is applicable only to VoIP dial peers.
Use this command to specify the maximum acceptable impairment factor for the voice calls sent by the selected dial peer.
Examples
The following example disables the icpif command:
dial-peer voice 10 voipicpif 0id
To configure the local identification (ID) for a neighboring border element (BE), use the id command in Annex G neighbor border element (BE) configuration mode. To remove the local ID, use the no form of this command.
id neighbor-id
no id neighbor-id
Syntax Description
Command Default
No default behavior or values
Command Modes
Annex G neighbor BE configuration
Command History
Examples
The following example configures the local ID for a neighboring BE. The identifier is 2333.
Router(config-annexg-neigh)# id 2333Related Commands
idle-voltage
To specify the idle voltage on an Foreign Exchange Station (FXS) voice port, use the idle-voltage command in voice-port configuration mode. To reset to the default, use the no form of this command.
idle-voltage {high | low}
no idle-voltage
Syntax Description
high
The talk-battery (tip-to-ring) voltage is high (-48V) when the FXS port is idle.
low
The talk-battery (tip-to-ring) voltage is low (-24V) when the FXS port is idle.
Command Default
The idle voltage is -24V
Command Modes
Voice-port configuration
Command History
Usage Guidelines
Some fax equipment and answering machines require a -48V idle voltage to be able to detect an off-hook condition in a parallel phone.
If the idle voltage setting is high, the talk battery reverts to -24V whenever the voice port is active (off hook).
Examples
The following example sets the idle voltage to -48V on voice port 1/1:
voice-port 1/1idle-voltage highThe following example restores the default idle voltage (-24V) on voice port 1/1:
voice-port 1/1no idle-voltageRelated Commands
ignore
To configure the North American E&M or E&M MELCAS voice port to ignore specific receive bits, use the ignore command in voice-port configuration mode. To reset to the default, use the no form of this command.
ignore {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit}
no ignore {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit}
Syntax Description
rx-a-bit
Ignores the receive A bit.
rx-b-bit
Ignores the receive B bit.
rx-c-bit
Ignores the receive C bit.
rx-d-bit
Ignores the receive D bit.
Command Default
The default is mode-dependent:
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North American E&M:
–
–
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–
–
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The ignore command applies to E&M digital voice ports associated with T1/E1 controllers. Repeat the command for each receive bit to be configured. Use this command with the define command.
Examples
To configure voice port 1/1 to ignore receive bits A, B, and C and to monitor receive bit D, enter the following commands:
voice-port 1/1ignore rx-a-bitignore rx-b-bitignore rx-c-bitno ignore rx-d-bitTo configure voice port 1/0/0 to ignore receive bits A, C, and D and to monitor receive bit B, enter the following commands:
voice-port 1/0/0ignore rx-a-bitignore rx-c-bitignore rx-d-bitno ignore rx-b-bitRelated Commands
ignore (interface)
To configure the serial interface to ignore the specified serial signals as the line up/down indicator, use the ignore command in interface configuration mode. To restore the default, use the no form of this command.
DCE Asynchronous Mode
ignore [dtr | rts]
no ignore [dtr | rts]
DCE Synchronous Mode
ignore [dtr | local-loopback | rts]
no ignore [dtr | local-loopback | rts]
DTE Asynchronous Mode
ignore [cts | dsr]
no ignore [cts | dsr]
DTE Synchronous Mode
ignore [cts | dcd | dsr]
no ignore [cts | dcd | dsr]
Syntax Description
Command Default
The no form of this command is the default. The serial interface monitors the serial signal as the line up/down indicator.
Command Modes
Interface configuration
Command History
Usage Guidelines
Serial Interfaces in DTE Mode
When the serial interface is operating in DTE mode, it monitors the DCD signal as the line up/down indicator. By default, the attached DCE device sends the DCD signal. When the DTE interface detects the DCD signal, it changes the state of the interface to up.
SDLC Multidrop Environments
In some configurations, such as a Synchronous Data Link Control (SDLC) multidrop environment, the DCE device sends the DSR signal instead of the DCD signal, which prevents the interface from coming up. Use this command to tell the interface to monitor the DSR signal instead of the DCD signal as the line up/down indicator.
Examples
The following example shows how to configure serial interface 0 to ignore the DCD signal as the line up/down indicator:
Router(config)# interface serial 0Router(config-if)# ignore dcdRelated Commands
debug serial lead-transition
Activates the leads status transition debug capability for all capable ports.
show interfaces serial
Displays information about a serial interface.
image encoding
To specify an encoding method for fax images associated with a Multimedia Mail over IP (MMoIP) dial peer, use the image encoding command in dial peer configuration mode. To reset to the default, use the no form of this command.
image encoding {mh | mr | mmr | passthrough}
no image encoding {mh | mr | mmr | passthrough}
Syntax Description
mh
Modified Huffman image encoding. This is the IETF standard.
mr
Modified Read image encoding.
mmr
Modified Modified Read image encoding.
passthrough
The image is not modified by an encoding method.
Command Default
Passthrough encoding
Command Modes
Dial peer configuration
Command History
Usage Guidelines
Use this command to specify an encoding method for e-mail fax TIFF images for a specific MMoIP dial peer. This command applies primarily to the on-ramp MMoIP dial peer. Although you can optionally create an off-ramp dial peer and configure a particular image encoding value for that off-ramp call leg, store-and-forward fax ignores the off-ramp MMoIP setting and sends the file using Modified Huffman encoding.
There are four available encoding methods:
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The IETF standard for sending fax TIFF images is Modified Huffman encoding with fine or standard resolution. RFC 2301 requires that compliant receivers support TIFF images with MH encoding and fine or standard resolution. If a receiver supports features beyond this minimal requirement, you might want to configure the Cisco AS5300 universal access server to send enhanced-quality documents to that receiver.
The primary reason to use a different encoding scheme from MH is to save network bandwidth. MH ensures interoperability with all Internet fax devices, but it is the least efficient of the encoding schemes for sending fax TIFF images. For most images, MR is more efficient than MH, and MMR is more efficient than MR. If you know that the recipient is capable of receiving more efficient encodings than just MH, store-and-forward fax allows you to send the most efficient encoding that the recipient can process. For end-to-end closed networks, you can choose any encoding scheme because the off-ramp gateway can process MH, MR, and MMR.
Another factor to consider is the viewing software. Many viewing applications (for example, those that come with Windows 95 or Windows NT) are able to display MH, MR, and MMR. Therefore you should decide, on the basis of the viewing application and the available bandwidth, which encoding scheme is right for your network.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following example selects Modified Modified Read as the encoding method for fax TIFF images sent by MMoIP dial peer 10:
dial-peer voice 10 mmoipimage encoding mmrRelated Commands
image resolution
Specifies a particular fax image resolution for a specific MMoIP dial peer.
image resolution
To specify a particular fax image resolution for a specific multimedia mail over IP (MMoIP) dial peer, use the image resolution command in dial peer configuration mode. To reset to the default, use the no form of this command.
image resolution {fine | standard | superfine | passthrough}
no image resolution {fine | standard | superfine | passthrough}
Syntax Description
Command Default
passthrough
Command Modes
Dial peer configuration
Command History
Usage Guidelines
Use this command to specify a resolution (in pixels per inch) for e-mail fax TIFF images sent by the specified MMoIP dial peer. This command applies primarily to the on-ramp MMoIP dial peer. Although you can optionally create an off-ramp dial peer and configure a particular image resolution value for that off-ramp call leg, store-and-forward fax ignores the off-ramp MMoIP setting and sends the file using fine resolution.
This command enables you to increase or decrease the resolution of a fax TIFF image, thereby changing not only the resolution but also the size of the fax TIFF file. The IETF standard for sending fax TIFF images is Modified Huffman encoding with fine or standard resolution. The primary reason to configure a different resolution is to save network bandwidth.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following example selects fine resolution (204-by-196 pixels per inch) for e-mail fax TIFF images associated with MMoIP dial peer 10:
dial-peer voice 10 mmoipimage encoding mhimage resolution fineRelated Commands
image encoding
Specifies an encoding method for fax images associated with an MMoIP dial peer.
impedance
To specify the terminating impedance of a voice-port interface, use the impedance command in voice-port configuration mode. To reset to the default, use the no form of this command.
impedance {600c | 600r | 900c | 900r | complex1 | complex2 | complex3 | complex4 | complex5 | complex6}
no impedance {600c | 600r | 900c | 900r | complex1 | complex2 | complex3 | complex4 | complex5 | complex6}
Syntax Description
600c
600 ohms + 2.15uF1.
600r
Resistive 600-ohm termination.
900c
900 ohms + 2.15uF1.
900r
Resistive 900-ohm termination.
complex1
220 ohms + (820 ohms || 115 nF)1 .
complex2
270 ohms + (750 ohms || 150 nF)1.
complex3
370 ohms + (620 ohms || 310 nF)1.
complex4
600r, line = 270 ohms + (750 ohms || 150 nF)1.
complex5
320 + (1050 ohms || 230 nF), line = 12 Kft1.
complex6
600r, line = 350 + (1000 ohms || 210 nF)1.
1 The plus symbol (+) indicates serial. The double pipe ( || ) indicates parallel.
Note
Command Default
600r
Command Modes
Voice-port configuration
Command History
Usage Guidelines
Use this command to specify the terminating impedance of analog telephony interfaces. The impedance value must match the specifications from the telephony system to which it is connected. Different countries often have different standards for impedance. CO switches in the United States are predominantly 600r. PBXs in the United States are 600r or 900c.
If the impedance is set incorrectly (if there is an impedance mismatch), a significant amount of echo is generated (which could be masked if the echo-cancel command has been enabled). In addition, gains might not work correctly if there is an impedance mismatch.
Configuring the impedance on a voice port changes the impedance on both voice ports of a VPM card. This voice port must be shut down and then opened for the new value to take effect.
Examples
The following example configures an FXO voice port on the Cisco 3600 series router for an impedance of 600 ohms (real):
voice-port 1/0/0impedance 600rshutdown/no shutdownThe following example configures an E&M voice port on a Cisco 2800 for an impedance of complex3:
voice-port 1/1impedance complex3shutdown/no shutdownRelated Commands
voice-port
Enters voice-port configuration mode.
echo-cancel enable
Enables the cancellation of voice that is sent out the interface and received back on the same interface.
inband-alerting
To enable inband alerting, use the inband-alerting command in the SIP user agent configuration mode. To disable inband alerting, use the no form of this command.
inband-alerting
no inband-alerting
Syntax Description
This command has no arguments or keywords.
Command Default
Enabled
Command Modes
SIP user agent configuration
Command History
Usage Guidelines
If inband alerting is enabled, the originating gateway can open an early media path (upon receiving a 180 or 183 message with a SDP body). Inband alerting allows the terminating gateway or switch to feed tones or announcements before a call is connected. If inband alerting is disabled, local alerting is generated on the originating gateway.
To reset this command to the default value, use the default command.
Examples
The following example disables inband alerting:
Router(config)# sip-uaRouter(config-sip-ua)# no inband-alertingRelated Commands
inbound ttl
To set the inbound time-to-live value, use the inbound ttl command in Annex G neighbor service configuration mode. To reset to the default, use the no form of this command.
inbound ttl ttl-value
no inbound ttl
Syntax Description
ttl-value
Inbound time-to-live (TTL) value, in seconds. Range is 0 to 4294967295. When set to 0, the service relationship does not expire. The default is 120.
Command Default
120 seconds
Command Modes
Annex G neighbor service configuration
Command History
Usage Guidelines
Service relationships are defined to be unidirectional. Establishing a service relationship between border element A and border element B entitles A to send requests to B and expect responses. For B to send requests to A and expect responses, a second service relationship must be established. From A's perspective, the service relationship that B establishes with A is designated the "inbound" service relationship. Use this command to indicate the duration of the relationship between border elements that participate in a service relationship.
Examples
The following example sets the inbound time-to-live value to 420 seconds (7 minutes):
Router(config-nxg-neigh-svc)# inbound ttl 420Related Commands
incoming alerting
To instruct an FXO ground-start voice port to modify its means of detecting an incoming call, use the incoming alerting command in voice-port configuration mode. To return to the default call detection method, use the no form of this command.
incoming alerting {ring-only}
no incoming alerting
Syntax Description
ring-only
Count incoming rings to detect incoming calls to the voice port that should be answered by the router.
Command Default
The FXO ground-start voice port detects an incoming call either by detecting the ring voltage applied to the line by the PSTN central office (CO) or by detecting that tip-ground is present for greater than about 7 seconds.
Command Modes
Voice-port configuration
Command History
Usage Guidelines
This command is valid only on FXO ports that have been configured with the signal ground-start command.
This command is necessary when two Cisco Unified CallManager Express (Cisco Unified CME) routers are used to provide redundant failover for incoming PSTN FXO ground-start lines. The voice ports for these trunk lines are wired in parallel between the two routers. The primary router is set to answer incoming calls after the first ring by default. The secondary router is set to answer incoming calls after 2 or 3 rings using the ring number command in voice-port configuration mode. As long as the primary router is operating, then the secondary router will not see enough rings to trigger it to answer the call. When the primary router is not operating, the secondary router has to be able to detect incoming ring signals so that it can answer calls. The default method of incoming call detection is not appropriate for voice ports on a secondary Cisco Unified CME router. The incoming alerting ring-only command must be used to modify the incoming call detection logic so that the voice port counts the number of incoming call rings instead of using the default call detection method.
Examples
The following example sets ring-only as the detection method for incoming calls on voice port 3/0/0, which is an FXO ground-start voice port.
Router(config)# voice-port 3/0/0Router(config-voiceport)# signal ground-startRouter(config-voiceport)# incoming alerting ring-onlyRelated Commands
ring number
Specifies the maximum number of rings to be detected before an incoming call is answered by the router.
signal
Specifies the type of signaling for a voice port.
incoming called-number (call filter match list)
To configure debug filtering for incoming called numbers, use the incoming called-number command in call filter match list configuration mode. To disable, use the no form of this command.
incoming called-number [+]string[T]
no incoming called-number [+]string[T]
Syntax Description
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Examples
The following example shows the voice call debug filter set to match incoming called number 5550123:
call filter match-list 1 voiceincoming called-number 5550123Related Commands
incoming called-number (dial peer)
To specify a digit string that can be matched by an incoming call to associate the call with a dial peer, use the incoming called-number command in dial peer configuration mode. To reset to the default, use the no form of this command.
incoming called-number [+]string[T]
no incoming called-number [+]string[T]
Syntax Description
Command Default
No incoming called number is defined
Command Modes
Dial peer configuration
Command History
Usage Guidelines
When a Cisco device is handling both modem and voice calls, it needs to be able to identify the service type of the call—meaning whether the incoming call to the server is a modem or a voice call. When the access server handles only modem calls, the service type identification is handled through modem pools. Modem pools associate calls with modem resources based on the dialed number identification service (DNIS). In a mixed environment, in which the server receives both modem and voice calls, you need to identify the service type of a call by using this command.
If you do not use this command, the server attempts to resolve whether an incoming call is a modem or voice call on the basis of the interface over which the call arrives. If the call comes in over an interface associated with a modem pool, the call is assumed to be a modem call; if a call comes in over a voice port associated with a dial peer, the call is assumed to be a voice call.
By default, there is no called number associated with the dial peer, which means that incoming calls are associated with dial peers by matching calling number with answer address, call number with destination pattern, or calling interface with configured interface.
Use this command to define the destination telephone number for a particular dial peer. For the on-ramp POTS dial peer, this telephone number is the DNIS number of the incoming fax call. For the off-ramp MMoIP dial peer, this telephone number is the telephone number of the destination fax machine.
This command applies to both VoIP and POTS dial peers and to on-ramp and off-ramp store-and-forward fax functions.
This command is also used to provide a matching VoIP dial peer on the basis of called number when fax or modem pass-through with named service events (NSEs) is defined globally on a terminating gateway.
You can ensure that all calls will match at least one dial peer by using the following commands:
Router(config)# dial-peer voice tag voipRouter(config-dial-peer)# incoming called-number.Examples
The following example configures calls that come into the router with a called number of 555-0163 as being voice calls:
dial peer voice 10 potsincoming called-number 5550163The following example sets the number (310) 555-0142 as the incoming called number for MMoIP dial peer 10:
dial-peer voice 10 mmoipincoming called-number 3105550142incoming calling-number (call filter match list)
To configure debug filtering for incoming calling numbers, use the incoming calling-number command in call filter match list configuration mode. To disable, use the no form of this command.
incoming calling-number [+]string[T]
no incoming calling-number [+]string[T]
Syntax Description
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Examples
The following example shows the voice call debug filter set to match incoming calling number 5550125:
call filter match-list 1 voiceincoming calling-number 5550125Related Commands
incoming dialpeer
To configure debug filtering for the incoming dial peer, use the incoming dialpeer command in call filter match list configuration mode. To disable, use the no form of this command.
incoming dialpeer tag
no incoming dialpeer tag
Syntax Description
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Examples
The following example shows the voice call debug filter set to match incoming dial peer 12:
call filter match-list 1 voiceincoming dialpeer 12Related Commands
incoming media local ipv4
To configure debug filtering for the incoming media local IPv4 addresses for the voice gateway receiving the media stream, use the incoming media local ipv4 command in call filter match list configuration mode. To disable, use the no form of this command.
incoming media local ipv4 ip_address
no incoming media local ipv4 ip_address
Syntax Description
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Examples
The following example shows the voice call debug filter set to match incoming media on the local voice gateway, which has IP address 192.168.10.255:
call filter match-list 1 voiceincoming media local ipv4 192.168.10.255Related Commands
incoming media remote ipv4
To configure debug filtering for the incoming media remote IPv4 addresses for the voice gateway receiving the media stream, use the incoming media remote ipv4 command in call filter match list configuration mode. To disable, use the no form of this command.
incoming media remote ipv4 ip_address
no incoming media remote ipv4 ip_address
Syntax Description
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Examples
The following example shows the voice call debug filter set to match incoming media on the remote IP device, which has IP address 192.168.10.255:
call filter match-list 1 voiceincoming media remote ipv4 192.168.10.255Related Commands
incoming port
To configure debug filtering for the incoming port, use the incoming port command in call filter match list configuration mode. To disable, use the no form of this command.
Cisco 2600, Cisco 3600, and Cisco 3700 Series
incoming port {slot-number/subunit-number/port | slot/port:ds0-group-no}
no incoming port {slot-number/subunit-number/port | slot/port:ds0-group-no}
Cisco 2600 and Cisco 3600 Series with a High-Density Analog Network Module (NM-HDA)
incoming port {slot-number/subunit-number/port}
no incoming port {slot-number/subunit-number/port}
Cisco AS5300
incoming port controller-number:D
no incoming port controller-number:D
Cisco AS5400
incoming port card/port:D
no incoming port card/port:D
Cisco AS5800
incoming port {shelf/slot/port:D | shelf/slot/parent:port:D}
no incoming port {shelf/slot/port:D | shelf/slot/parent:port:D}
Cisco MC3810
incoming port slot/port
no incoming port slot/port
Syntax Description
Cisco 2600, Cisco 3600 Series and Cisco 3700 Series
Cisco AS5300
Cisco AS5400
card
Specifies the T1 or E1 card. Valid entries for the card argument are 1 to 7.
port
Specifies the voice port number. Valid entries are 0 to 7.
:D
Indicates the D channel associated with ISDN PRI.
Cisco AS5800
Cisco MC3810
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Examples
The following example shows the voice call debug filter set to match incoming port 1/1/1 on a Cisco 3660 voice gateway:
call filter match-list 1 voiceincoming port 1/1/1Related Commands
incoming secondary-called-number
To configure debug filtering for incoming called numbers from the second stage of a two-stage scenario, use the incoming secondary-called-number command in call filter match list configuration mode. To disable, use the no form of this command.
incoming secondary-called-number string
no incoming secondary-called-number string
Syntax Description
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Usage Guidelines
Two-stage dialing occurs when the voice gateway presents a dial-tone before accepting digits. When a voice call comes into the Cisco IOS voice gateway, the voice port on the router is seized inbound by a PBX or CO switch. The voice gateway then presents a dial tone to the caller and collects digits until it can identify an outbound dial-peer. Dial-peer matching is done digit-by-digit whether the digits are dialed with irregular intervals by humans or in a regular fashion by telephony equipment sending the precollected digits. The voice gateway attempts to match a dial-peer after each digit is received.
Examples
The following example shows the voice call debug filter set to match incoming secondary called number 8288807:
call filter match-list 1 voiceincoming secondary-called-number 8288807Related Commands
incoming signaling local ipv4
To configure debug filtering for the incoming signaling local IPv4 addresses for the gatekeeper managing the signaling, use the incoming signaling local ipv4 command in call filter match list configuration mode. To disable, use the no form of this command.
incoming signaling local ipv4 ip_address
no incoming signaling local ipv4 ip_address
Syntax Description
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Examples
The following example shows the voice call debug filter set to match incoming signaling on the local voice gateway, which has IP address 192.168.10.255:
call filter match-list 1 voiceincoming signaling local ipv4 192.168.10.255Related Commands
incoming signaling remote ipv4
To configure debug filtering for the incoming signaling remote IPv4 addresses for the gatekeeper managing the signaling, use the incoming signaling remote ipv4 command in call filter match list configuration mode. To disable, use the no form of this command.
incoming signaling remote ipv4 ip_address
no incoming signaling remote ipv4 ip_address
Syntax Description
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Examples
The following example shows the voice call debug filter set to match incoming signaling on the remote IP device, which has IP address 192.168.10.255:
call filter match-list 1 voiceincoming signaling remote ipv4 192.168.10.255Related Commands
incoming uri
To specify the voice class used to match a VoIP dial peer to the uniform resource identifier (URI) of an incoming call, use the incoming uri command in dial peer configuration mode. To remove the URI voice class from the dial peer, use the no form of this command.
H.323 Session Protocol
incoming uri {called | calling} tag
no incoming uri {called | calling}
SIP Session Protocol
incoming uri {from | request | to} tag
no incoming uri {from | request | to}
Syntax Description
Command Default
No default behavior or values
Command Modes
Dial peer configuration
Command History
Usage Guidelines
•
•
•
Note
•
Examples
The following example matches on the destination TEL URI in incoming H.323 calls by using the ab100 voice class:
dial-peer voice 100 voipincoming uri called ab100Related Commands
index (voice class)
To define one or more numbers for a voice class called number, or a range of numbers for a voice class called number pool, use the index command in voice class configuration mode. To remove the number or range of numbers, use the no form of this command.
index number called-number
no index number called-number
Syntax Description
number
Digits that identify this index. Range is 1 to 2147483647.
called-number
Specifies a called number, or a range of called numbers, in E.164 format.
Command Default
No index is configured.
Command Modes
Voice class configuration
Command History
Usage Guidelines
Use this command to define one or more numbers for a voice class called number, or a range of numbers for a voice class called number pool. You can define multiple indexes for any inbound or outbound voice class called number or voice class called number pool.
When defining a range of numbers for a called number pool:
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Examples
The following example shows the configuration for indexes in voice class called number pool 100:
voice class called number pool 100index 1 4085550100 - 4085550111 (Range of called numbers are 4085550100 up to 4085550111)index 2 +3227045000The following example shows configuration for indexes in voice class called number outbound 222:
voice class called number outbound 222index 1 4085550101index 2 4085550102index 2 4085550103Related Commands
voice class called number
One or more called numbers configured for a voice class.
info-digits
To automatically prepend two information digits to the beginning of a dialed number associated with the given POTS dial peer, use the info-digits command in dial peer configuration mode. To keep the router from automatically prepending the two-digit information numbers to the beginning of the POTS dial peer, use the no form of this command.
info-digits string
no info-digits
Syntax Description
Command Default
No default behavior or values.
Command Modes
Dial peer configuration
Command History
12.2(1)T
This command was introduced on Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series routers, and Cisco AS5300 series universal access servers.
Usage Guidelines
This command is designed to prepend a pair of information digits to the beginning of the dialed number string for the POTS dial peer that will enable you to dynamically redirect the outgoing call. The info-digits command is only available for POTS dial peers.
Examples
The following example prepends the information number string 91 to the beginning of the dialed number for POTS dial peer 10:
dial-peer voice 10 potsinfo-digits 91information-type
To select a specific information type for a Voice over IP (VoIP) or plain old telephone service (POTS) dial peer, use the information-type command in dial peer configuration mode. To remove the current information type setting, use the no form of this command. To return to the default configuration, use the no form of this command.
information-type {fax | voice | video}
no information-type
Syntax Description
fax
The information type is set to store-and-forward fax.
voice
The information type is set to voice. This is the default.
video
The information type is set to video.
Command Default
Voice
Command Modes
Dial peer configuration
Command History
Usage Guidelines
The fax keyword applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following example shows the configuration for information type fax for VoIP dial peer 10:
dial-peer voice 10 voipinformation-type faxThe following example shows the configuration for information type video for POTS dial peer 22:
dial-peer voice 22 potsinformation-type videoRelated Commands
isdn integrate calltype all
Enables integrated mode (for data, voice, and video) on ISDN BRI or PRI interfaces.
inject guard-tone
To play out a guard tone with the voice packet, use the inject guard-tone command in voice-class configuration mode. To remove the guard tone, use the no form of this command.
inject guard-tone frequency amplitude [idle]
no inject guard-tone frequency amplitude [idle]
Syntax Description
Command Default
No guard tone is injected.
Command Modes
Voice-class configuration
Command History
12.3(4)XD
This command was introduced.
12.3(7)T
This command was integrated into Cisco IOS Release 12.3(7)T.
Usage Guidelines
The inject guard-tone command has an effect on an ear and mouth (E&M) analog or digital voice port only if the signal type for that port is Land Mobile Radio (LMR). The guard tone is played out with the voice packet to keep the radio channel up. Guard tones of 1950 Hz and 2175 Hz can be filtered out before the voice packet is sent from the digital signal processor (DSP) to the network using the digital-filter command.
Examples
The following example configures a guard tone of 1950 Hz and -10 dBm to be played out with voice packets:
voice class tone-signal tone1inject guard-tone 2175 -30Related Commands
digital-filter
Specifies the digital filter to be used before the voice packet is sent from the DSP to the network.
inject pause
To specify a pause between injected tones, use the inject pause command in voice-class configuration mode. To remove the pause, use the no form of this command.
inject pause index milliseconds
no inject pause index milliseconds
Syntax Description
index
Order of pauses and tones. Range is integers from 1 to 10.
milliseconds
Duration, in milliseconds, of the pause between injected tones. Range is integers from 10 to 500.
Command Default
milliseconds: 0 milliseconds
Command Modes
Voice-class configuration
Command History
12.3(4)XD
This command was introduced.
12.3(7)T
This command was integrated into Cisco IOS Release 12.3(7)T.
Usage Guidelines
The inject pause command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). Use this command to specify the pause between injected tones specified with the inject tone command. Use the index argument of this command in conjunction with the index argument of the inject tone command to specify the order of the pauses and tones.
Examples
The following example configures a pause of 100 milliseconds after the injected tone:
voice class tone-signal 100inject tone 1 2000 0 200inject pause 2 100Related Commands
inject tone
Specifies a wakeup or frequency selection tone to be played out before the voice packet.
inject tone
To specify a wakeup or frequency selection tone to be played out before the voice packet, use the inject tone command in voice-class configuration mode. To remove the tone, use the no form of this command.
inject tone index frequency amplitude duration
no inject tone index frequency amplitude duration
Syntax Description
Command Default
No tone is injected.
Command Modes
Voice-class configuration
Command History
12.3(4)XD
This command was introduced.
12.3(7)T
This command was integrated into Cisco IOS Release 12.3(7)T.
Usage Guidelines
The inject tone command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). Use this command with the inject pause command to configure wakeup and frequency selection tones. Use the index argument of this command in conjunction with the index argument of the inject pause command to specify the order of the pauses and tones.
If you configure injected tones with this command, be sure to use the timing delay-voice tdm command to configure a delay before the voice packet is played out. The delay must be equal to the sum of the durations of the injected tones and pauses in the tone-signal voice class.
Examples
The following example configures a frequency selection tone to be played out before the voice packet:
voice class tone-signal 100inject tone 1 1950 3 150inject tone 2 2000 0 60inject pause 3 60inject tone 4 2175 3 150inject tone 5 1000 0 50Related Commands
inject pause
Specifies a pause between injected tones.
timing delay-voice tdm
Specifies the delay before a voice packet is played out.
input gain
To configure a specific input gain value or enable automatic gain control, use the input gain command in voice-port configuration mode. To disable the selected amount of inserted gain, use the no form of this command.
input gain {decibels | auto-control [auto-dbm]}
no input gain {decibels | auto-control [auto-dbm]}
Syntax Description
Command Default
decibels: 0 decibels
auto-dbm: -9 dBmCommand Modes
Voice-port configuration
Command History
Usage Guidelines
A system-wide loss plan must be implemented using both the input gain and output attenuation commands. You must consider other equipment (including PBXs) in the system when creating a loss plan. The default value for this command assumes that a standard transmission loss plan is in effect, meaning that there is typically a minimum attenuation of -6 dB between phones, especially if echo cancellers are present. Connections are implemented to provide 0 dB of attenuation when the input gain and output attenuation commands are configured with the default value of 0 dB.
You cannot increase the gain of a signal to the public switched telephone network (PSTN), but you can decrease it. If the voice level is too high, you can decrease the volume by either decreasing the input gain or increasing the output attenuation.
You can increase the gain of a signal coming into the router. If the voice level is too low, you can increase the input gain by using the input gain command.
Typical Land Mobile Radio (LMR) signaling systems send 0 dB out and expect -10 dB in. Setting output attenuation to 10 dB is typical. Output attenuation should be adjusted to provide the voice level required by the radio to produce correct transmitter modulation.
The auto-control keyword and auto-dbm argument are available on an ear and mouth (E&M) voice port only if the signal type for that port is LMR. The auto-control keyword enables automatic gain control, which is performed by the digital signal processor (DSP). Automatic gain control adjusts speech to a comfortable volume when it becomes too loud or too soft. Because of radio network loss and other environmental factors, the speech level arriving at a router from an LMR system could be very low. You can use automatic gain control to ensure that the speech is played back at a more comfortable level. Because the gain is inserted digitally, the background noise can also be amplified. Automatic gain control is implemented as follows:
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Examples
The following example inserts a 3-dB gain at the receiver side of the interface in the Cisco 3600 series router:
port 1/0/0input gain 3Related Commands
output attenuation
Configures a specific output attenuation value or enables automatic gain control for a voice port.
interface (RLM server)
To define the IP addresses of the Redundant Link Manager (RLM) server, use the interface command in interface configuration mode. To disable this function, use the no form of this command.
interface name-tag
no interface name-tag
Syntax Description
name-tag
Name to identify the server configuration so that multiple entries of server configuration can be entered.
Command Default
Disabled
Command Modes
Interface configuration
Command History
Usage Guidelines
Each server can have multiple entries of IP addresses or aliases.
Examples
The following example configures the access-server interfaces for RLM servers "Loopback1" and "Loopback2":
interface Loopback1ip address 10.1.1.1 255.255.255.255interface Loopback2ip address 10.1.1.2 255.255.255.255rlm group 1server r1-serverlink address 10.1.4.1 source Loopback1 weight 4link address 10.1.4.2 source Loopback2 weight 3Related Commands
interface Dchannel
To specify an ISDN D-channel interface and enter interface configuration mode, use the interface Dchannel command in global configuration mode.
interface Dchannel interface-number
Syntax Description
Command Default
No D-channel interface is specified.
Command Modes
Global configuration
Command History
12.2(8)B
This command was introduced.
12.2(15)T
This command was integrated into Cisco IOS Release 12.2(15)T.
Usage Guidelines
This command is used specifically in Voice over IP (VoIP) applications that require release of the ISDN PRI signaling time slot for RLM configurations.
Examples
The following example configures a D-channel interface for a Signaling System 7 (SS7)-enabled shared T1 link:
controller T1 1pri-group timeslots 1-3 nfas_d primary nfas_int 0 nfas_group 0 rlm-group 0channel group 23 timeslot 24end! D-channel interface is created for configuration of ISDN parameters:interface Dchannel1isdn T309 4000endRelated Commands
interface event-log dump ftp
To enable the gateway to write the contents of the interface event log buffer to an external file, use the interface event-log dump ftp command in application configuration monitor mode. To reset to the default, use the no form of this command.
interface event-log dump ftp server[:port]/file username username password [encryption-type] password
no interface event-log dump ftp server[:port]/file username username password [encryption-type] password
Syntax Description
Command Default
Interface event log buffer is not written to an external file.
Command Modes
Application configuration monitor
Command History
12.3(14)T
This command was introduced to replace the call application interface event-log dump ftp command.
Usage Guidelines
This command enables the gateway to automatically write the interface event log buffer to the named file when the buffer becomes full. The default buffer size is 4 KB. To modify the size of the buffer, use the interface event-log max-buffer-size command. To manually flush the event log buffer, use the interface dump event-log command in privileged EXEC mode.
Note
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You should enable FTP dumping only when necessary and not enable it in situations where it might adversely impact system performance.
Examples
The following example specifies that interface event log are written to an external file named int_elogs.log on a server named ftp-server:
applicationmonitorinterface event-log dump ftp ftp-server/elogs/int_elogs.log username myname password 0 mypassThe following example specifies that application event logs are written to an external file named int_elogs.log on a server with the IP address of 10.10.10.101:
applicationmonitorinterface event-log dump ftp 10.10.10.101/elogs/int_elogs.log username myname password 0 mypassRelated Commands
interface event-log error only
To restrict event logging to error events only for application interfaces, use the interface event-log error-only command in application configuration monitor mode. To reset to the default, use the no form of this command.
interface event-log error-only
no interface event-log error-only
Syntax Description
This command has no arguments or keywords.
Command Default
All events are logged.
Command Modes
Application configuration monitor
Command History
12.3(14)T
This command was introduced to replace the call application interface event-log error only command.
Usage Guidelines
This command limits the severity level of the events that are logged; it does not enable logging. You must use this command with the interface event-log command, which enables event logging for all application interfaces.
Examples
The following example enables event logging for error events only:
applicationmonitorinterface event-log error-onlyRelated Commands
interface event-log max-buffer-size
To set the maximum size of the event log buffer for each application interface, use the interface event-log max-buffer-size command in application configuration monitor mode. To reset to the default, use the no form of this command.
interface event-log max-buffer-size kbytes
no interface event-log max-buffer-size
Syntax Description
Command Default
4 KB
Command Modes
Application configuration monitor
Command History
12.3(14)T
This command was introduced to replace the call application interface event-log max-buffer-size command.
Usage Guidelines
If the event log buffer reaches the limit set by this command, the gateway allocates a second buffer of equal size. The contents of both buffers is displayed when you use the show call application interface command. When the first event log buffer becomes full, the gateway automatically appends its contents to an external FTP location if the interface event-log dump ftp command is used.
A maximum of two buffers are allocated for an event log. If both buffers are filled, the first buffer is deleted and another buffer is allocated for new events (buffer wraps around). If the interface event-log dump ftp command is configured and the second buffer becomes full before the first buffer is dumped, event messages are dropped and are not recorded in the buffer.
Examples
The following example sets the maximum buffer size to 8 KB:
applicationmonitorinterface event-log max-buffer-size 8Related Commands
interface max-server-records
To set the maximum number of application interface records that are saved, use the interface max-server-records command in application configuration monitor mode. To reset to the default, use the no form of this command.
interface max-server-records number
no interface max-server-records
Syntax Description
Command Default
10
Command Modes
Application configuration monitor
Command History
12.3(14)T
This command was introduced to replace the call application interface max-server-records command.
Usage Guidelines
Only the specified number of records from the most recently accessed servers are kept.
Examples
The following example sets the maximum saved records to 50:
applicationmonitorinterface max-server-records 50Related Commands
interface stats
To enable statistics collection for application interfaces, use the interface stats command in application configuration monitor mode. To reset to the default, use the no form of this command.
interface stats
no interface stats
Syntax Description
This command has no arguments or keywords.
Command Default
Statistics collection is disabled.
Command Modes
Application configuration monitor
Command History
12.3(14)T
This command was introduced to replace the call application interface stats command.
Usage Guidelines
To display the interface statistics enabled by this command, use the show call application interface command. To reset the interface counters to zero, use the clear call application interface command.
Examples
The following example enables statistics collection for application interfaces:
applicationmonitorinterface statsRelated Commands
ip circuit
To create carrier IDs on an IP virtual trunk group, and create a maximum capacity for the IP group, use the ip circuit command. To remove a trunk group or maximum capacity, use the no form of the command.
ip circuit {carrier-id carrier-name [reserved-calls reserved] | max-calls maximum-calls | default {only | name carrier-name}}
no ip circuit {carrier-id carrier-name | default {only | name carrier-name}}
Syntax Description
Command Default
If this command is not specified, no IP carriers and no maximum call leg values are defined.
Command Modes
H.323 configuration.
Command History
Usage Guidelines
You can use the ip circuit command only when no calls are active. You can define multiple carrier IDs, and the ordering does not matter. IP circuit default only is mutually exclusive with defining carriers with circuit carrier id.
If ip circuit default only is specified, the maximum calls value is set to 1000.
Examples
The following example specifies a default circuit and maximum number of calls:
voice service voipno allow-connections any to potsno allow-connections pots to anyallow-connections h323 to h323h323ip circuit max-calls 1000ip circuit default onlyThe following example specifies a default carrier and incoming source carrier:
voice service voipno allow-connections any to potsno allow-connections pots to anyallow-connections h323 to h323h323ip circuit carrier-id AA reserved-calls 200ip circuit max-calls 1000Related Commands
ip precedence (dial peer)
To set IP precedence (priority) for packets sent by the dial peer, use the ip precedence command in dial peer configuration mode. To reset to the default, use the no form of this command.
ip precedence number
no ip precedence number
Syntax Description
number
Integer specifying the IP precedence value. Range is 0 to 7. A value of 0 means that no precedence (priority) has been set. The default is 0.
Command Default
The default value for this command is zero (0)
Command Modes
Dial peer configuration
Command History
11.3(1)NA
This command was introduced on the following platforms: Cisco 2500 series, Cisco 3600 series, and Cisco AS5300.
Usage Guidelines
Use this command to configure the value set in the IP precedence field when voice data packets are sent over the IP network. This command should be used if the IP link utilization is high and the quality of service for voice packets needs to have a higher priority than other IP packets. This command should also be used if RSVP is not enabled and the user would like to give voice packets a higher priority than other IP data traffic.
This command applies to VoIP peers.
Examples
The following example sets the IP precedence to 5:
dial-peer voice 10 voipip precedence 5ip qos dscp
To set the DSCP for the quality of service, use the ip qos dscp command in dial peer configuration mode. To disable DSCP, use the no form of this command.
ip qos dscp [number | set-af | set-cs | default | ef] [media | signaling | video {rsvp-none | rsvp-pass | rsvp-fail}]
no ip qos dscp [number | set-af | set-cs | default | ef] [media | signaling | video {rsvp-none | rsvp-pass | rsvp-fail}]
Syntax Description
Command Default
DSCP is set to bit pattern af41.
Command Modes
Dial peer configuration
Command History
12.2(2)T
This command was introduced.It replaced the ip precedence (dial peer) command
12.3(4)T
Keywords were added to support DSCP configuration for video streams.
Usage Guidelines
To configure voice and signaling traffic priorities, use the ip qos dscp command.
Recommended values are ip qos dscp ef media and ip qos dscp af31 signaling.
You must use the ip rsvp bandwidth command to enable RSVP on an IP interface before you can specify RSVP QoS.Examples
The following example specifies DSCP is set to precedence 1 and is applied to media payload packets.
dial-peer voice 1 voipip qos dscp cs1 mediaThe following example specifies the DSCP for video streams:
dial-peer voice 100 voipip qos dscp cs5 video rsvp-passip qos dscp cs6 video rsvp-failRelated Commands
ip rtcp report interval
To configure the average reporting interval between subsequent Real-Time Control Protocol (RTCP) report transmissions, use the ip rtcp report interval command in global configuration mode. To reset to the default, use the no form of this command.
ip rtcp report interval value
no ip rtcp report interval
Syntax Description
Command Default
5000 ms
Command Modes
Global configuration
Command History
Usage Guidelines
This command configures the average interval between successive RTCP report transmissions for a given voice session. For example, if the value argument is set to 25,000 milliseconds, an RTCP report is sent every 25 seconds, on average.
For more information about RTCP, see RFC 1889, RTP: A Transport Protocol for Real-Time Applications.
Examples
The following example sets the reporting interval to 5000 ms:
Router(config)# ip rtcp report interval 5000Related Commands
ip udp checksum
To calculate the UDP checksum for voice packets sent by the dial peer, use the ip udp checksum command in dial peer configuration mode. To disable this feature, use the no form of this command.
ip udp checksum
no ip udp checksum
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
Dial peer configuration
Command History
Usage Guidelines
Use this command to enable UDP checksum calculation for each of the outbound voice packets. This command is disabled by default to speed up the transmission of the voice packets. If you suspect that the connection has a high error rate, you should enable this command to prevent corrupted voice packets forwarded to the digital signal processor (DSP).
This command applies to VoIP peers.
Note
Examples
The following example calculates the UDP checksum for voice packets sent by dial peer 10:
dial-peer voice 10 voipip udp checksumRelated Commands
loop-detect
Enables loop detection for T1 for Voice over ATM, Voice over Frame Relay, and Voice over HDLC.
irq global-request
To configure the gatekeeper to send information-request (IRQ) messages with the call-reference value (CRV) set to zero, use the irq global-request command in gatekeeper configuration mode. To disable the gatekeeper from sending IRQ messages, use the no form of this command.
irq global-request
no irq global-request
Syntax Description
This command has no arguments or keywords.
Command Default
The gatekeeper sends IRQ messages with the CRV set to zero.
Command Modes
Gatekeeper configuration
Command History
Usage Guidelines
Use this command to disable the gatekeeper from sending an IRQ message with the CRV set to zero when the gatekeeper requests the status of all calls after its initialization. Disabling IRQ messages can eliminate unnecessary information request response (IRR) messages if the reconstruction of call structures can be postponed until the next IRR or if the call information is no longer required because calls are terminated before the periodic IRR message is sent. Disabling IRQ messages is advantageous if direct bandwidth control is not used in the gatekeeper.
Examples
The following example shows that IRQ messages are not sent from the gatekeeper:
...lrq reject-resource-lowno irq global-requesttimer lrq seq delay 10timer lrq window 6timer irr period 6no shutdown...Related Commands
