Cisco IOS Voice Command Reference

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

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 voip

 icpif 0

id

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

Defaults

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 2333

The following example shows the the error response when an undefined neighbor ID is entered:

Router(config-annexg-neigh)#no id def 

% Entry not valid, id not configured. 

To deconfigure id under different neighbor you have to expilicitly go into that neighbor 
and deconfigure the id. 

Related 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

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 1/1

 idle-voltage high

The following example restores the default idle voltage (-24V) on voice port 1/1:

voice-port 1/1

 no idle-voltage

Related 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

Command Default

The default is mode-dependent:

•North American E&M:

–The receive B, C, and D bits are ignored

–The receive A bit is not ignored

•E&M MELCAS:

–The receive A bit is ignored

–The receive B, C, and D bits are not ignored

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 1/1

 ignore rx-a-bit

 ignore rx-b-bit

 ignore rx-c-bit

 no ignore rx-d-bit

To 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 1/0/0

 ignore rx-a-bit

 ignore rx-c-bit

 ignore rx-d-bit

 no ignore rx-b-bit

Related 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 0

Router(config-if)# ignore dcd

Related Commands

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

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:

•Modified Huffman (MH)—One-dimensional data compression scheme that compresses data in only one direction (horizontal). Modified Huffman compression does not allow the transmission of redundant data. This encoding method produces the largest image file size.

•Modified Read (MR)—Two-dimensional data compression scheme (used by fax devices) that handles the data compression of the vertical line and that concentrates on the space between lines and within given characters.

•Modified Modified Read (MMR)—Data compression scheme used by newer Group 3 fax devices. This encoding method produces the smallest possible image file size and is slightly more efficient than Modified Read.

•Passthrough—No encoding method is applied to the image—meaning that the image is encoded by whatever encoding method is used by the fax device.

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 mmoip

 image encoding mmr

Related Commands

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 mmoip

 image encoding mh

 image resolution fine

Related Commands

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

Note This table represents the full set of impedances. Not all modules support the full set of impedance values shown here. To determine which impedance values are available on your modules, enter impedance ? in the command-line interface to see a list of the values you can configure.

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 1/0/0

impedance 600r

shutdown/no shutdown

The following example configures an E&M voice port on a Cisco 2800 for an impedance of complex3:

voice-port 1/1

impedance complex3

shutdown/no shutdown

Related Commands

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-ua

Router(config-sip-ua)# no inband-alerting

Related 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

Defaults

120 seconds

Command Modes

Annex G neighbor service configuration (config-nxg-neigh-svc)

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 420

Related 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

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 3/0/0

Router(config-voiceport)# signal ground-start

Router(config-voiceport)# incoming alerting ring-only

Related Commands

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

+	(Optional) Character that indicates an E.164 standard number.
string	Series of digits that specify a pattern for the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A through D, and the following special characters: •The asterisk (*) and pound sign (#) that appear on standard touch-tone dial pads. •Comma (,), which inserts a pause between digits. •Period (.), which matches any entered digit (this character is used as a wildcard). •Percent sign (%), which indicates that the preceding digit occurred zero or more times; similar to the wildcard usage. •Plus sign (+), which indicates that the preceding digit occurred one or more times. Note The plus sign used as part of a digit string is different from the plus sign that can be used in front of a digit string to indicate that the string is an E.164 standard number. •Circumflex (^), which indicates a match to the beginning of the string. •Dollar sign ($), which matches the null string at the end of the input string. •Backslash symbol (\), which is followed by a single character, and matches that character. Can be used with a single character with no other significance (matching that character). •Question mark (?), which indicates that the preceding digit occurred zero or one time. •Brackets ( [ ] ), which indicate a range. A range is a sequence of characters enclosed in the brackets; only numeric characters from 0 to 9 are allowed in the range. •Parentheses ( ( ) ), which indicate a pattern and are the same as the regular expression rule.
T	(Optional) Control character that indicates that the destination-pattern value is a variable-length dial string. Using this control character enables the router to wait until all digits are received before routing the call.

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 voice

 incoming called-number 5550123

Related 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

+	(Optional) Character that indicates an E.164 standard number.
string	Series of digits that specify a pattern for the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A through D, and the following special characters: •The asterisk (*) and pound sign (#) that appear on standard touch-tone dial pads. •Comma (,), which inserts a pause between digits. •Period (.), which matches any entered digit (this character is used as a wildcard). •Percent sign (%), which indicates that the preceding digit occurred zero or more times; similar to the wildcard usage. •Plus sign (+), which indicates that the preceding digit occurred one or more times. Note The plus sign used as part of a digit string is different from the plus sign that can be used in front of a digit string to indicate that the string is an E.164 standard number. •Circumflex (^), which indicates a match to the beginning of the string. •Dollar sign ($), which matches the null string at the end of the input string. •Backslash symbol (\), which is followed by a single character, and matches that character. Can be used with a single character with no other significance (matching that character). •Question mark (?), which indicates that the preceding digit occurred zero or one time. •Brackets ( [ ] ), which indicate a range. A range is a sequence of characters enclosed in the brackets; only numeric characters from 0 to 9 are allowed in the range. •Parentheses ( ( ) ), which indicate a pattern and are the same as the regular expression rule.
T	(Optional) Control character that indicates that the destination-pattern value is a variable-length dial string. Using this control character enables the router to wait until all digits are received before routing the call.

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 signaling 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 voip

Router(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 pots

 incoming called-number 5550163

The following example sets the number (310) 555-0142 as the incoming called number for MMoIP dial peer 10:

dial-peer voice 10 mmoip

 incoming called-number 3105550142

incoming 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

+	(Optional) Character that indicates an E.164 standard number.
string	Series of digits that specify a pattern for the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A through D, and the following special characters: •The asterisk (*) and pound sign (#) that appear on standard touch-tone dial pads. •Comma (,), which inserts a pause between digits. •Period (.), which matches any entered digit (this character is used as a wildcard). •Percent sign (%), which indicates that the preceding digit occurred zero or more times; similar to the wildcard usage. •Plus sign (+), which indicates that the preceding digit occurred one or more times. Note The plus sign used as part of a digit string is different from the plus sign that can be used in front of a digit string to indicate that the string is an E.164 standard number. •Circumflex (^), which indicates a match to the beginning of the string. •Dollar sign ($), which matches the null string at the end of the input string. •Backslash symbol (\), which is followed by a single character, and matches that character. Can be used with a single character with no other significance (matching that character). •Question mark (?), which indicates that the preceding digit occurred zero or one time. •Brackets ( [ ] ), which indicate a range. A range is a sequence of characters enclosed in the brackets; only numeric characters from 0 to 9 are allowed in the range. •Parentheses ( ( ) ), which indicate a pattern and are the same as the regular expression rule.
T	(Optional) Control character that indicates that the destination-pattern value is a variable-length dial string. Using this control character enables the router to wait until all digits are received before routing the call.

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 voice

 incoming calling-number 5550125

Related 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 voice

 incoming dialpeer 12

Related 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 voice

 incoming media local ipv4 192.168.10.255

Related 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 voice

 incoming media remote ipv4 192.168.10.255

Related 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

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 voice

 incoming port 1/1/1

Related 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 voice

 incoming secondary-called-number 8288807

Related 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 voice

 incoming signaling local ipv4 192.168.10.255

Related 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 voice

 incoming signaling remote ipv4 192.168.10.255

Related 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 voice 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}

Session Initiation Protocol (SIP) Session Protocol

incoming uri {from | request | to | via} tag

no incoming uri {from | request | to | via}

Syntax Description

Command Default

No voice class is specified.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Usage Guidelines

•Before you use this command, configure the voice class by using the voice class uri command.

•The keywords depend on whether the dial peer is configured for SIP with the session protocol sipv2 command. The from, request, to, and via keywords are available only for SIP dial peers. The called and calling keywords are available only for dial peers using H.323.

•This command applies rules for dial peer matching. Table 29 and Table 30 show the rules and the order in which they are applied when the incoming uri command is used. The gateway compares the dial-peer command to the call parameter in its search to match an inbound call to a dial peer. All dial peers are searched based on the first match criterion. Only if no match is found does the gateway move on to the next criterion.

Note Calls using an E.164 number, rather than a URI, use the dial-peer matching rules that existed prior to Cisco IOS Release 15.1(2)T. For information, see theDial Peer Configuration on Voice Gateway Routers document, Cisco IOS Voice Configuration Library.

•You can use this command multiple times in the same dial peer with different keywords. For example, you can use incoming uri called and incoming uri calling in the same dial peer. The gateway then selects the dial peer based on the matching rules described in Table 29 and Table 30.

Examples

The following example matches on the destination telephone URI in incoming H.323 calls by using the ab100 voice class:

dial-peer voice 100 voip

 incoming uri called ab100

The following example matches on the incoming via URI for SIP calls by using the ab100 voice class:

dial-peer voice 100 voip 

 session protocol sipv2

 incoming uri via ab100

Related 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

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:

•The range of numbers must be in E.164 format.

•The beginning number and ending number must be the same length.

•The last digit of each number must be 0 to 9.

•Leading '+' (if used) must be defined from in the range of called numbers.

Examples

The following example shows the configuration for indexes in voice class called number pool 100:

voice class called number pool 100

 index 1 4085550100 - 4085550111 (Range of called numbers are 4085550100 up to 4085550111)

 index 2 +3227045000 

The following example shows configuration for indexes in voice class called number outbound 222:

voice class called number outbound 222

 index 1 4085550101

 index 2 4085550102

 index 2 4085550103

Related Commands

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 prepend the info-digits with "00" use the default info-digits form of this command. 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 xx

default info-digits

no info-digits

Syntax Description

Defaults

The dialed number is prepended with "00", indicating that the dialed number is a regular line.

Command Modes

Dial-peer configuration

Command History

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 tied to a voice-port that corresponds to Feature Group-D (FGD) Exchange Access North American (EANA) signaling that provides specific call services such as emergency 911 calls in the United States. Configuring the info-digit command for other voice-port types is not advised and may yield undesirable results.

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 pots

 info-digits 91

information-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

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 voip

 information-type fax

The following example shows the configuration for information type video for POTS dial peer 22:

dial-peer voice 22 pots

 information-type video

Related Commands

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

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 tone1

 inject guard-tone 2175 -30

Related Commands

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

Command Default

milliseconds: 0 milliseconds

Command Modes

Voice-class configuration

Command History

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 100

 inject tone 1 2000 0 200

 inject pause 2 100

Related Commands

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

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 100

 inject tone 1 1950 3 150

 inject tone 2 2000 0 60

 inject pause 3 60

 inject tone 4 2175 3 150

 inject tone 5 1000 0 50

Related Commands

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 dBm

Command 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:

•Output level: -9 dB

•Gain range: -12 dB to 20 dB

•Attack time (low to high): 30 milliseconds

•Attack time (high to low): 8 seconds

Examples

The following example inserts a 3-dB gain at the receiver side of the interface in the Cisco 3600 series router:

port 1/0/0

 input gain 3

Related Commands

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

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 Loopback1

 ip address 10.1.1.1 255.255.255.255

 interface Loopback2

 ip address 10.1.1.2 255.255.255.255

rlm group 1

 server r1-server

 link address 10.1.4.1 source Loopback1 weight 4

 link address 10.1.4.2 source Loopback2 weight 3

Related Commands

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

interface-number

Specifies the ISDN interface number. Note The interface-number argument depends on which controller the rlm-group subkeyword in the pri-group timeslots controller configuration command uses. For example, if the Redundant Link Manager (RLM) group is configured using the controller e1 2/3 command, the D-channel interface command will be interface Dchannel 2/3.

Command Default

No D-channel interface is specified.

Command Modes

Global configuration

Command History

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 1

 pri-group timeslots 1-3 nfas_d primary nfas_int 0 nfas_group 0 rlm-group 0 

 channel group 23 timeslot 24

 end

! D-channel interface is created for configuration of ISDN parameters:

interface Dchannel1

 isdn T309 4000

 end

Related 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

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•Enabling the gateway to write event logs to FTP could adversely impact gateway memory resources in some scenarios, for example, when:

•The gateway is consuming high processor resources and FTP does not have enough processor resources to flush the logged buffers to the FTP server.

•The designated FTP server is not powerful enough to perform FTP transfers quickly

•Bandwidth on the link between the gateway and the FTP server is not large enough

•The gateway is receiving a high volume of short-duration calls or calls that are failing

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:

application

monitor

interface event-log dump ftp ftp-server/elogs/int_elogs.log username myname password 0 
mypass

The 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:

application

monitor

interface event-log dump ftp 10.10.10.101/elogs/int_elogs.log username myname password 
0 mypass

Related 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

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:

application

monitor

interface event-log error-only

Related 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

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:

application

monitor

interface event-log max-buffer-size 8

Related 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

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:

application

monitor

interface max-server-records 50

Related 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

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:

application

monitor

interface stats

Related 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 voip 

 no allow-connections any to pots

 no allow-connections pots to any

 allow-connections h323 to h323

 h323

  ip circuit max-calls 1000

  ip circuit default only

The following example specifies a default carrier and incoming source carrier:

voice service voip 

 no allow-connections any to pots

 no allow-connections pots to any

 allow-connections h323 to h323

  h323

   ip circuit carrier-id AA reserved-calls 200

   ip circuit max-calls 1000

Related Commands

ip dhcp-client forcerenew

To enable forcerenew-message handling on the DHCP client when authentication is enabled, use the ip dhcp-client forcerenew command in global configuration mode. To disable the forced authentication, use the no form of this command.

ip dhcp-client forcerenew

no ip dhcp-client forcerenew

Syntax Description

This command has no arguments or keywords.

Command Default

Forcerenew messages are dropped.

Command Modes

Global configuration (config)

Command History

Usage Guidelines

DHCP forcerenew handling is not enabled until the CLI is configured.

Examples

The following example shows how to enable DHCP forcerenew-message handling on the DHCP client:

Router(config)# ip dhcp-client forcerenew 

Related 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

Command Default

The default value for this command is zero (0)

Command Modes

Dial peer configuration

Command History

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 voip

 ip precedence 5

ip qos defending-priority

To configure the Resource Reservation Protocol (RSVP) defending priority value for determining quality of service (QoS), use the ip qos defending-priority command in dial peer configuration mode. To disable RSVP defending priority as a QoS factor, use the no form of this command.

ip qos defending-priority defending-pri-value

no ip qos defending-priority

Syntax Description

Command Default

The RSVP defending priority value is disabled and is not a factor in determining QoS.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Usage Guidelines

To configure the RSVP defending priority value, use the ip qos defending-priority command in dial peer configuration mode. The defending priority value is passed to the QoS module during reservation initiation. In a situation where there is not enough bandwidth available to support all calls, this setting enables an existing call to avoid being preempted by a new call unless the preemption priority of the new call is higher than the defending priority of the existing call.

Examples

The following example shows how to specify the RSVP defending priority value:

dial-peer voice 100 voip

 ip qos defending-priority 1111

Related Commands

ip qos dscp

To configure the differentiated services code point (DSCP) value for quality of service (QoS), use the ip qos dscp command in dial peer configuration mode. To disable DSCP as a QoS factor, set the DSCP value to default (which sets the value to the 000000 bit pattern). To set DSCP values to their default settings, use the no form of this command.

ip qos dscp {dscp-value | set-af | set-cs | default | ef} {signaling | media [rsvp-pass | rsvp-fail] | video [rsvp-none | rsvp-pass | rsvp-fail]}

no ip qos dscp {dscp-value | set-af | set-cs | default | ef} {signaling | media [rsvp-pass | rsvp-fail] | video [rsvp-none | rsvp-pass | rsvp-fail]}

Syntax Description

Command Default

The DSCP default values are as follows:

•The default DSCP value for all signaling packets is af31.

•The default DSCP value for all media (voice and fax) packets is ef.

•The default DSCP value for all video packets is af41.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Usage Guidelines

To configure voice, signaling, and video traffic priorities, use the ip qos dscp command in dial peer configuration mode. The recommended value for media (voice and fax) packets is ef; for signaling packets, the recommended value is af31; and for video packets, it is af41 (all defaults).

Additionally, before you can specify RSVP QoS, you must first use the ip rsvp bandwidth command to enable RSVP on the IP interface.

Examples

The following example shows how to set the DSCP value to a class-selector code point value of 1 and apply that DSCP setting to media (voice and fax) payload packets with no RSVP configured:

dial-peer voice 1 voip

 ip qos dscp cs1 media

The following example shows how to set the DSCP value to the expedited forwarding bit pattern and apply that DSCP setting to media (voice and fax) payload packets with a successful RSVP connection:

dial-peer voice 1 voip

ip qos dscp ef media rsvp-pass

The following example shows how to set the DSCP value to an assured forwarding code point value of 22 and apply that DSCP setting to all signaling packets:

dial-peer voice 1 voip

ip qos dscp af22 signaling

The following example shows how to set the DSCP value to an assured forwarding code point value of 43 and apply that DSCP setting to video packets with a successful RSVP connection:

dial-peer voice 100 voip

 ip qos dscp af43 video rsvp-pass

Related Commands

ip qos policy-locator

To configure a quality of service (QoS) policy-locator (application ID) used to deploy Resource Reservation Protocol (RSVP) policies for specifying bandwidth reservations on Cisco IOS Session Initiation Protocol (SIP) devices, use the ip qos policy-locator command in dial peer configuration mode. To delete an application policy, use the no form of this command.