Table Of Contents
Cisco IOS Voice Commands:
I
icpif
id
idle-voltage
ignore
ignore (interface)
image encoding
image resolution
impedance
inband-alerting
inbound ttl
incoming alerting
incoming called-number (call filter match list)
incoming called-number (dial peer)
incoming calling-number (call filter match list)
incoming dialpeer
incoming media local ipv4
incoming media remote ipv4
incoming port
incoming secondary-called-number
incoming signaling local ipv4
incoming signaling remote ipv4
incoming uri
index (voice class)
info-digits
information-type
inject guard-tone
inject pause
inject tone
input gain
interface (RLM server)
interface Dchannel
interface event-log dump ftp
interface event-log error only
interface event-log max-buffer-size
interface max-server-records
interface stats
ip circuit
ip precedence (dial peer)
ip qos dscp
ip rtcp report interval
ip udp checksum
irq global-request
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
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.2(8)T
|
The number default value for this command was changed from 30 to 20.
|
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:
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
neighbor-id
|
ID for a neighboring BE. The identification ID must be an International Alphabet 5 (IA5) string and cannot include spaces. This identifier is local and is not related to the border element ID.
|
Command Default
No default behavior or values
Command Modes
Annex G neighbor BE configuration
Command History
Release
|
Modification
|
12.2(2)XA
|
This command was introduced.
|
12.2(4)T
|
This command was integrated into Cisco IOS Release 12.2(4)T. This command is not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850.
|
12.2(11)T
|
This command was integrated into Cisco IOS Release 12.2(11)T.
|
Examples
The following example configures the local ID for a neighboring BE. The identifier is 2333.
Router(config-annexg-neigh)# id 2333
Related Commands
Command
|
Description
|
advertise (annex G)
|
Controls the type of descriptors that the BE advertises to its neighbors.
|
port
|
Configures the port number of the neighbor that is used for exchanging Annex G messages.
|
query-interval
|
Configures the interval at which the local BE queries the neighboring BE.
|
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
Release
|
Modification
|
12.0(4)T
|
This command was introduced on the Cisco MC3810.
|
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:
The following example restores the default idle voltage (-24V) on voice port 1/1:
Related Commands
Command
|
Description
|
show voice port
|
Displays voice port configuration information.
|
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:
•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
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on the Cisco MC3810.
|
12.0(7)XK
|
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
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:
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:
Related Commands
Command
|
Description
|
condition
|
Manipulates the signaling bit pattern for all voice signaling types.
|
define
|
Defines the transmit and receive bits for North American E&M and E&M MELCAS voice signaling.
|
show voice port
|
Displays configuration information for voice ports.
|
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
dtr
|
Specifies that the DCE ignores the Data Terminal Ready (DTR) signal.
|
rts
|
Specifies that the DCE ignores the Request To Send (RTS) signal.
|
local-loopback
|
Specifies that the DCE ignores the local loopback signal.
|
cts
|
Specifies that the DTE ignores the Clear To Send (CTS) signal.
|
dsr
|
Specifies that the DTE ignores the Data Set Ready (DSR) signal.
|
dcd
|
Specifies that the DTE ignores the Data Carrier Detect (DCD) signal.
|
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
Release
|
Modification
|
12.2(15)ZJ
|
This command was introduced on the following platforms: Cisco 2610XM, Cisco 2611XM, Cisco 2620XM, Cisco 2621XM, Cisco 2650XM, Cisco 2651XM, Cisco 2691, Cisco 3631, Cisco 3660, Cisco 3725, and Cisco 3745 routers.
|
12.3(2)T
|
This command was integrated into Cisco IOS Release 12.3(2)T.
|
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
Command
|
Description
|
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
Release
|
Modification
|
12.0(4)XJ
|
This command was introduced.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
12.2(4)T
|
This command was implemented on the Cisco 1750.
|
12.2(8)T
|
This command was implemented on the following platforms: Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
|
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:
Related Commands
Command
|
Description
|
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
fine
|
Configures the fax TIFF image resolution to be 204-by-196 pixels per inch.
|
standard
|
Configures the fax TIFF image resolution to be 204-by-98 pixels per inch.
|
superfine
|
Configures the fax TIFF image resolution to be 204-by-391 pixels per inch.
|
passthrough
|
Indicates that the resolution of the fax TIFF image is not altered.
|
Command Default
passthrough
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
12.0(4)XJ
|
This command was introduced.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
12.2(4)T
|
This command was implemented on the Cisco 1750 access router.
|
12.2(8)T
|
This command was implemented on the following platforms: Cisco 1751, Cisco 2600, Cisco 3600, Cisco 3725, and Cisco 3745.
|
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:
Related Commands
Command
|
Description
|
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.
|
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
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series.
|
12.3(7)T
|
This command was integrated into Cisco IOS Release 12.3(7)T and support was added for the complex3, complex4, complex5, and complex6 keywords on the Cisco 2600XM series, Cisco 2691, Cisco 2800 series, Cisco 3662 (telco models), Cisco 3700 series, and Cisco 3800 series.
|
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):
The following example configures an E&M voice port on a Cisco 2800 for an impedance of complex3:
Related Commands
Command
|
Description
|
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
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
12.1(3)T
|
This command was limited to enabling and disabling inband alerting.
|
12.2(2)XA
|
This command was implemented on the Cisco AS5350 and Cisco AS5400.
|
12.2(2)XB1
|
This command was introduced on the Cisco AS5850.
|
12.2(11)T
|
This command was integrated into Cisco IOS Release 12.2(11)T.
|
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)# no inband-alerting
Related Commands
Command
|
Description
|
default
|
Sets a command to its default.
|
exit
|
Exits the SIP user agent configuration mode.
|
max-forwards
|
Specifies the maximum number of hops for a request.
|
no
|
Negates a command or set its defaults.
|
retry
|
Configures the SIP signaling timers for retry attempts.
|
timers
|
Configures the SIP signaling timers.
|
transport
|
Enables SIP UA transport for TCP/UDP.
|
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
Release
|
Modification
|
12.2(11)T
|
This command was introduced.
|
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
Command
|
Description
|
access-policy
|
Requires that a neighbor be explicitly configured.
|
outbound retry-interval
|
Defines the retry period for attempting to establish the outbound relationship between border elements.
|
retry interval
|
Defines the time between delivery attempts.
|
retry window
|
Defines the total time that a border element attempts delivery.
|
service-relationship
|
Establishes a service relationship between two border elements.
|
shutdown
|
Enables or disables the border element.
|
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
Cisco IOS Release
|
Modification
|
12.4(4)XC
|
This command was introduced.
|
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
Command
|
Description
|
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
+
|
(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
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
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
Command
|
Description
|
call filter match-list voice
|
Create a call filter match list for debugging voice calls.
|
debug condition match-list
|
Run a filtered debug on a voice call.
|
incoming calling-number
|
Configure debug filtering for incoming calling numbers.
|
incoming dialpeer
|
Configure debug filtering for the incoming dial peer.
|
incoming secondary-called-number
|
Configure debug filtering for incoming called numbers from the second stage of a two-stage scenario.
|
outgoing called-number
|
Configure debug filtering for outgoing called numbers.
|
outgoing calling-number
|
Configure debug filtering for outgoing calling numbers.
|
outgoing dialpeer
|
Configure debug filtering for the outgoing dial peer.
|
show call filter match-list
|
Display call filter match lists.
|
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
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series.
|
11.3NA
|
This command was implemented on the Cisco AS5800.
|
12.0(4)XJ
|
This command was modified for store-and-forward fax.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
12.2(4)T
|
This command was implemented on the Cisco 1750.
|
12.2(8)T
|
This command was implemented on the following platforms: Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
|
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 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:
incoming called-number 5550163
The following example sets the number (310) 555-0142 as the incoming called number for MMoIP dial peer 10:
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
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
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
Command
|
Description
|
call filter match-list voice
|
Create a call filter match list for debugging voice calls.
|
debug condition match-list
|
Run a filtered debug on a voice call.
|
incoming called-number (call filter match list)
|
Configure debug filtering for incoming called numbers.
|
incoming dialpeer
|
Configure debug filtering for the incoming dial peer.
|
incoming secondary-called-number
|
Configure debug filtering for incoming called numbers from the second stage of a two-stage scenario.
|
outgoing called-number
|
Configure debug filtering for outgoing called numbers.
|
outgoing calling-number
|
Configure debug filtering for outgoing calling numbers.
|
outgoing dialpeer
|
Configure debug filtering for the outgoing dial peer.
|
show call filter match-list
|
Display call filter match lists.
|
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
tag
|
Digits that define a specific dial peer. Valid entries are 1 to 2,147,483,647.
|
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
Examples
The following example shows the voice call debug filter set to match incoming dial peer 12:
call filter match-list 1 voice
Related Commands
Command
|
Description
|
call filter match-list voice
|
Create a call filter match list for debugging voice calls.
|
debug condition match-list
|
Run a filtered debug on a voice call.
|
incoming called-number (call filter match list)
|
Configure debug filtering for incoming called numbers.
|
incoming calling-number
|
Configure debug filtering for incoming calling numbers.
|
incoming port
|
Configure debug filtering for the incoming port.
|
incoming secondary-called-number
|
Configure debug filtering for incoming called numbers from the second stage of a two-stage scenario.
|
outgoing called-number
|
Configure debug filtering for outgoing called numbers.
|
outgoing calling-number
|
Configure debug filtering for outgoing calling numbers.
|
outgoing dialpeer
|
Configure debug filtering for the outgoing dial peer.
|
outgoing port
|
Configure debug filtering for the outgoing port.
|
show call filter match-list
|
Display call filter match lists.
|
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
ip_address
|
IP address of the local voice gateway
|
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
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
Command
|
Description
|
call filter match-list voice
|
Create a call filter match list for debugging voice calls.
|
debug condition match-list
|
Run a filtered debug on a voice call.
|
incoming media remote ipv4
|
Configure debug filtering for the incoming media IPv4 addresses for calls to the IP side from the remote IP device.
|
incoming port
|
Configure debug filtering for the incoming port.
|
outgoing media local ipv4
|
Configure debug filtering for the outgoing media IPv4 addresses for calls to the IP side from the local voice gateway.
|
outgoing media remote ipv4
|
Configure debug filtering for the outgoing media IPv4 addresses for calls to the IP side from the remote IP device.
|
outgoing port
|
Configure debug filtering for the outgoing port.
|
show call filter match-list
|
Display call filter match lists.
|
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
ip_address
|
IP address of the remote IP device
|
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
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
Command
|
Description
|
call filter match-list voice
|
Create a call filter match list for debugging voice calls.
|
debug condition match-list
|
Run a filtered debug on a voice call.
|
incoming media local ipv4
|
Configure debug filtering for the incoming media IPv4 addresses for calls to the IP side from the local voice gateway.
|
incoming port
|
Configure debug filtering for the incoming port.
|
outgoing media local ipv4
|
Configure debug filtering for the outgoing media IPv4 addresses for calls to the IP side from the local voice gateway
|
outgoing media remote ipv4
|
Configure debug filtering for the outgoing media IPv4 addresses for calls to the IP side from the remote IP device.
|
outgoing port
|
Configure debug filtering for the outgoing port.
|
show call filter match-list
|
Display call filter match lists.
|
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
slot-number
|
Number of the slot in the router in which the VIC is installed. Valid entries are 0 to 3, depending on the slot in which it has been installed.
|
subunit-number
|
Subunit on the VIC in which the voice port is located. Valid entries are 0 or 1.
|
port
|
Voice port number. Valid entries are 0 and 1.
|
slot
|
The router location in which the voice port adapter is installed. Valid entries are 0 to 3.
|
port:
|
Indicates the voice interface card location. Valid entries are 0 and 3.
|
ds0-group-no
|
Indicates the defined DS0 group number. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card.
|
Cisco AS5300
controller-number
|
T1 or E1 controller.
|
:D
|
D channel associated with ISDN PRI.
|
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
shelf
|
Specifies the T1 or E1 controller on the T1 card, or the T1 controller on the T3 card. Valid entries for the shelf argument are 0 to 9999.
|
slot
|
Specifies the T1 or E1 controller on the T1 card, or the T1 controller on the T3 card. Valid entries for the slot argument are 0 to 11.
|
port
|
Specifies the voice port number.
•T1 or E1 controller on the T1 card —Valid entries are 0 to 11.
•T1 controller on the T3 card—Valid entries are 1 to 28.
|
:port
|
Specifies the value for the parent argument. The valid entry is 0.
|
:D
|
Indicates the D channel associated with ISDN PRI.
|
Cisco MC3810
slot
|
The slot argument specifies the number slot in the router in which the VIC is installed. The only valid entry is 1.
|
port
|
The port variable specifies the voice port number. Valid interface ranges are as follows:
•T1—ANSI T1.403 (1989), Telcordia TR-54016.
•E1— ITU G.703.
•Analog Voice—Up to six ports (FXS, FXO, E & M).
•Digital Voice— Single T1/E1 with cross-connect drop and insert, CAS and CCS signaling, PRI QSIG.
•Ethernet—Single 10BASE-T.
•Serial—Two five-in-one synchronous serial (ANSI EIA/TA-530, EIA/TA-232, EIA/TA-449; ITU V.35, X.21, Bisync, Polled async).
|
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
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
Related Commands
Command
|
Description
|
call filter match-list voice
|
Create a call filter match list for debugging voice calls.
|
debug condition match-list
|
Run a filtered debug on a voice call.
|
outgoing port
|
Configure debug filtering for the outgoing port.
|
show call filter match-list
|
Display call filter match lists.
|
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
string
|
Series of digits that specify a pattern for the E.164 or private dialing plan telephone number. Valid entries are the digits 0 to 9, the letters A to D, and the following special characters:
•The asterisk (*) and pound sign (#) that appear on standard touchtone dial pads. On the Cisco 3600 series routers only, these characters cannot be used as leading characters in a string (for example, *650).
•Comma (,), which inserts a pause between digits.
•Period (.), which matches any entered digit (this character is used as a wildcard). On the Cisco 3600 series routers, the period cannot be used as a leading character in a string (for example, .650).
•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; 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 0 to 9 are allowed in the range.
•Parentheses ( ), which indicate a pattern and are the same as the regular expression rule.
|
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
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
Command
|
Description
|
call filter match-list voice
|
Create a call filter match list for debugging voice calls.
|
debug condition match-list
|
Run a filtered debug on a voice call.
|
incoming called-number (call filter match list)
|
Configure debug filtering for incoming called numbers.
|
incoming calling-number
|
Configure debug filtering for incoming calling numbers.
|
incoming dialpeer
|
Configure debug filtering for the incoming dial peer.
|
outgoing called-number
|
Configure debug filtering for outgoing called numbers.
|
outgoing calling-number
|
Configure debug filtering for outgoing calling numbers.
|
outgoing dialpeer
|
Configure debug filtering for the outgoing dial peer.
|
show call filter match-list
|
Display call filter match lists.
|
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
ip_address
|
IP address of the local voice gateway
|
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
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
Command
|
Description
|
call filter match-list voice
|
Create a call filter match list for debugging voice calls.
|
debug condition match-list
|
Run a filtered debug on a voice call.
|
incoming port
|
Configure debug filtering for the incoming port.
|
incoming signaling remote ipv4
|
Configure debug filtering for the incoming signaling IPv4 addresses for calls to the IP side from the remote IP device.
|
outgoing port
|
Configure debug filtering for the outgoing port.
|
outgoing signaling local ipv4
|
Configure debug filtering for the outgoing signaling IPv4 addresses for calls to the IP side from the local voice gateway.
|
outgoing signaling remote ipv4
|
Configure debug filtering for the outgoing signaling IPv4 addresses for calls to the IP side from the remote IP device.
|
show call filter match-list
|
Display call filter match lists.
|
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
ip_address
|
IP address of the remote IP device
|
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
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
Command
|
Description
|
call filter match-list voice
|
Create a call filter match list for debugging voice calls.
|
debug condition match-list
|
Run a filtered debug on a voice call.
|
incoming port
|
Configure debug filtering for the incoming port.
|
incoming signaling local ipv4
|
Configure debug filtering for the incoming signaling IPv4 addresses for calls to the IP side from the local voice gateway.
|
outgoing port
|
Configure debug filtering for the outgoing port.
|
outgoing signaling local ipv4
|
Configure debug filtering for the outgoing signaling IPv4 addresses for calls to the IP side from the local voice gateway.
|
outgoing signaling remote ipv4
|
Configure debug filtering for the outgoing signaling IPv4 addresses for calls to the IP side from the remote IP device.
|
show call filter match-list
|
Display call filter match lists.
|
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
called
|
Destination URI in the H.225 message of an H.323 call.
|
calling
|
Source URI in the H.225 message of an H.323 call.
|
from
|
From header in an incoming SIP Invite message.
|
request
|
Request-URI in an incoming SIP Invite message.
|
to
|
To header in an incoming SIP Invite message.
|
tag
|
Alphanumeric label that uniquely identifies the voice class. This tag must be configured with the voice class uri command.
|
Command Default
No default behavior or values
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
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, and to 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 new rules for dial peer matching. Table 27 and Table 28 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 criteria. Only if no match is found does the gateway move on to the next criteria.
Table 27 Dial-Peer Matching Rules for Inbound URI in SIP Calls
Match Order
|
Cisco IOS Command
|
Incoming Call Parameter
|
1
|
incoming uri request
|
Request-URI
|
2
|
incoming uri to
|
To URI
|
3
|
incoming uri from
|
From URI
|
4
|
incoming called-number
|
Called number
|
5
|
answer-address
|
Calling number
|
6
|
destination-pattern
|
Calling number
|
7
|
carrier-id source
|
Carrier-is associated with the call
|
Table 28 Dial-Peer Matching Rules for Inbound URI in H.323 Calls
Match Order
|
Cisco IOS Command
|
Incoming Call Parameter
|
1
|
incoming uri called
|
Destination URI in H.225 message
|
2
|
incoming uri calling
|
Source URI in H.225 message
|
3
|
incoming called-number
|
Called number
|
4
|
answer-address
|
Calling number
|
5
|
destination-pattern
|
Calling number
|
6
|
carrier-id source
|
Source carrier-id associated with the call
|
Note Calls using an E.164 number, rather than a URI, use the previously existing dial-peer matching rules. For information, refer to the Dial Peer Configuration on Voice Gateway Routers document, Cisco IOS Voice Configuration Library, Release 12.3.
•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 27 and Table 28.
Examples
The following example matches on the destination TEL URI in incoming H.323 calls by using the ab100 voice class:
incoming uri called ab100
Related Commands
Command
|
Description
|
answer-address
|
Specifies calling number to match for a dial peer.
|
debug voice uri
|
Displays debugging messages related to URI voice classes.
|
destination-pattern
|
Specifies telephone number to match for a dial peer.
|
dial-peer voice
|
Enters dial peer configuration mode to create or modify a dial peer.
|
incoming called-number
|
Incoming called number matched to a dial peer.
|
session protocol
|
Specifies the session protocol in the dial peer for calls between the local and remote router.
|
show dialplan incall uri
|
Displays which dial peer is matched for a specific URI in an incoming voice call.
|
voice class uri
|
Creates or modifies a voice class for matching dial peers to calls containing a SIP or TEL URI.
|
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
Release
|
Modification
|
12.4(11)T
|
This command was introduced.
|
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)
The following example shows configuration for indexes in voice class called number outbound 222:
voice class called number outbound 222
Related Commands
Command
|
Description
|
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
string
|
Specifies the two-digit prefix that the router will automatically prepend to the dialed number for the given POTS dial peer.
Note This string cannot contain any more or any less than two digits.
|
Command Default
No default behavior or values.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
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:
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
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
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series.
|
12.0(4)XJ
|
This command was modified for store-and-forward fax.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
12.2(4)T
|
This command was implemented on the Cisco 1750.
|
12.2(8)T
|
This command was implemented on the following platforms: Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
|
12.4(11)T
|
The video keyword was added.
|
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:
The following example shows the configuration for information type video for POTS dial peer 22:
Related Commands
Command
|
Description
|
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
frequency
|
Frequency, in Hz, of the tone to be injected. Range is integers from 1 to 4000.
|
amplitude
|
Amplitude, in dBm, of the tone to be injected. Range is integers from -50 to -3.
|
idle
|
(Optional) Play out the inverse of the guard tone when there are no voice packets. Idle tone and guard tone are mutually exclusive.
|
Command Default
No guard tone is injected.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
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 tone1
inject guard-tone 2175 -30
Related Commands
Command
|
Description
|
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
Release
|
Modification
|
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 100
Related Commands
Command
|
Description
|
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
index
|
Order of pauses and tones. Range is integers from 1 to 10.
|
frequency
|
Frequency, in Hz, of the tone to be injected. Range is integers from 1 to 4000.
|
amplitude
|
Amplitude, in dBm, of the tone to be injected. Range is integers from -30 to 3.
|
duration
|
Duration, in milliseconds, of the tone to be injected. Range is integers from 10 to 500.
|
Command Default
No tone is injected.
Command Modes
Voice-class configuration
Command History
Release
|
Modification
|
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 100
Related Commands
Command
|
Description
|
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
decibels
|
Gain, in decibels (dB), to be inserted at the receiver side of the interface. Range is integers from -27 to 16. The default is 0.
|
auto-control
|
Enable automatic gain control.
|
auto-dbm
|
(Optional) Target speech level, in decibels per milliwatt (dBm), to be achieved at the receiver side of the interface. Range is integers from -30 to 3. The default is -9.
|
Command Default
decibels: 0 decibels
auto-dbm: -9 dBm
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
11.3(1)MA
|
This command was implemented on the Cisco MC3810.
|
12.3(4)XD
|
The range of values for the decibels argument was increased.
|
12.3(7)T
|
This command was integrated into Cisco IOS Release 12.3(7)T.
|
12.3(14)T
|
This command was implemented on the Cisco 2800 series and Cisco 3800 series.
|
12.4(2)T
|
The auto-control keyword and auto-dbm argument were added.
|
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:
Related Commands
Command
|
Description
|
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
Release
|
Modification
|
11.3(7)
|
This command was introduced.
|
Usage Guidelines
Each server can have multiple entries of IP addresses or aliases.
Examples
The following example configures the access-server interfaces for RLM servers "Loopback1" and "Loopback2":
ip address 10.1.1.1 255.255.255.255
ip address 10.1.1.2 255.255.255.255
link address 10.1.4.1 source Loopback1 weight 4
link address 10.1.4.2 source Loopback2 weight 3
Related Commands
Command
|
Description
|
clear interface
|
Resets the hardware logic on an interface.
|
clear rlm group
|
Clears all RLM group time stamps to zero.
|
link (RLM)
|
Specifies the link preference.
|
protocol rlm port
|
Reconfigures the port number for the basic RLM connection for the whole rlm-group.
|
retry keepalive
|
Allows consecutive keepalive failures a certain amount of time before the link is declared down.
|
server (RLM)
|
Defines the IP addresses of the server.
|
show rlm group statistics
|
Displays the network latency of the RLM group.
|
show rlm group status
|
Displays the status of the RLM group.
|
show rlm group timer
|
Displays the current RLM group timer values.
|
shutdown (RLM)
|
Shuts down all of the links under the RLM group.
|
timer
|
Overwrites the default setting of timeout values.
|
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
Release
|
Modification
|
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:
pri-group timeslots 1-3 nfas_d primary nfas_int 0 nfas_group 0 rlm-group 0
channel group 23 timeslot 24
! D-channel interface is created for configuration of ISDN parameters:
Related Commands
Command
|
Description
|
pri-group timeslots
|
Specifies an ISDN PRI group on a channelized T1 or E1 controller, and releases the ISDN PRI signaling time slot for environments that require that SS7-enabled VoIP applications share all slots in a PRI group.
|
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
server
|
Name or IP address of FTP server where the file is located.
|
port
|
(Optional) Specific port number on server.
|
file
|
Name and path of file.
|
username
|
Username required to access file.
|
encryption-type
|
(Optional) The Cisco proprietary algorithm used to encrypt the password. Values are 0 or 7. To disable encryption enter 0; to enable encryption enter 7. If you specify 7, you must enter an encrypted password (a password already encrypted by a Cisco router).
|
password
|
Password required to access file.
|
Command Default
Interface event log buffer is not written to an external file.
Command Modes
Application configuration monitor
Command History
Release
|
Modification
|
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•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:
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:
interface event-log dump ftp 10.10.10.101/elogs/int_elogs.log username myname password
0 mypass
Related Commands
Command
|
Description
|
call application interface event-log dump ftp
|
Enable the gateway to write the contents of the interface event log buffer to an external file.
|
interface dump event-log
|
Flushes the event log buffer for application interfaces to an external file.
|
interface event-log
|
Enables event logging for external interfaces used by voice applications.
|
interface event-log max-buffer-size
|
Sets the maximum size of the event log buffer for each application interface.
|
interface max-server-records
|
Sets the maximum number of application interface records that are saved.
|
show call application interface
|
Displays event logs and statistics for application interfaces.
|
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
Release
|
Modification
|
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:
interface event-log error-only
Related Commands
Command
|
Description
|
call application interface event-log error-only
|
Restricts event logging to error events only for application interfaces.
|
interface event-log
|
Enables event logging for external interfaces used by voice applications.
|
interface event-log max-buffer-size
|
Sets the maximum size of the event log buffer for each application interface.
|
interface max-server-records
|
Sets the maximum number of application interface records that are saved.
|
show call application interface
|
Displays event logs and statistics for application interfaces.
|
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
kbytes
|
Maximum buffer size, in kilobytes. Range is 1 to 10. Default is 4.
|
Command Default
4 KB
Command Modes
Application configuration monitor
Command History
Release
|
Modification
|
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:
interface event-log max-buffer-size 8
Related Commands
Command
|
Description
|
call application interface event-log max-buffer-size
|
Sets the maximum size of the event log buffer for each application interface.
|
interface dump event-log
|
Flushes the event log buffer for application interfaces to an external file.
|
interface event-log dump ftp
|
Enables the gateway to write the contents of the interface event log buffer to an external file.
|
interface max-server-records
|
Sets the maximum number of application interface records that are saved.
|
show call application interface
|
Displays event logs and statistics for application interfaces.
|
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
number
|
Maximum number of records to save. Range is 1 to 100. Default is 10.
|
Command Default
10
Command Modes
Application configuration monitor
Command History
Release
|
Modification
|
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:
interface max-server-records 50
Related Commands
Command
|
Description
|
call application interface max-server-records
|
Sets the maximum number of application interface records that are saved.
|
interface event-log
|
Enables event logging for external interfaces used by voice applications.
|
interface event-log max-buffer-size
|
Sets the maximum size of the event log buffer for each application interface.
|
show call application interface
|
Displays event logs and statistics for application interfaces.
|
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
Release
|
Modification
|
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:
Related Commands
Command
|
Description
|
call application interface stats
|
Enables statistics collection for application interfaces.
|
clear call application interface
|
Clears application interface statistics or event logs.
|
interface event-log
|
Enables event logging for external interfaces used by voice applications.
|
show call application interface
|
Displays event logs and statistics for application interfaces.
|
stats
|
Enables statistics collection for voice applications.
|
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
carrier-id
|
Sets the IP circuit associated with a specific carrier.
|
carrier-name
|
Defines an IP circuit using the specified name as the circuit ID.
|
reserved-calls reserved
|
(Optional) Specifies the maximum number of calls for the circuit ID. Default value is 200.
|
max-calls maximum-calls
|
Sets the number of maximum aggregate H.323 IP circuit carrier call legs. Default value is 1000.
|
default only
|
Creates a single carrier using the default carrier name.
|
default name
|
Changes the default circuit name.
|
carrier-name
|
Default carrier name.
|
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
Release
|
Modification
|
12.2(13)T3
|
This command was introduced.
|
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:
no allow-connections any to pots
no allow-connections pots to any
allow-connections h323 to h323
ip circuit max-calls 1000
The following example specifies a default carrier and incoming source carrier:
no allow-connections any to pots
no allow-connections pots to any
allow-connections h323 to h323
ip circuit carrier-id AA reserved-calls 200
ip circuit max-calls 1000
Related Commands
Command
|
Description
|
show crm
|
Displays some of the values set by this command.
|
voice-source group
|
Assigns a name to a set of source IP group characteristics, which are used to identify and translate an incoming VoIP call.
|
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
Release
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Modification
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11.3(1)NA
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This command was introduced on the following platforms: Cisco 2500 series, Cisco 3600 series, and Cisco AS5300.
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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:
ip 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
number
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DSCP value. Valid entries are from 0 to 63.
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set-af
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Sets DSCP to assured forwarding bit pattern.
•af11—bit pattern 001010
•af12—bit pattern 001100
•af13—bit pattern 001110
•af21—bit pattern 010010
•af22—bit pattern 010100
•af23—bit pattern 010110
•af31—bit pattern 011010
•af32—bit pattern 011100
•af33—bit pattern 011110
•af41—bit pattern 100010
•af42—bit pattern 100100
•af43—bit pattern 100110
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set-cs
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Sets DSCP to class-selector codepoint.
•cs1—codepoint 1 (precedence 1)
•cs2—codepoint 2 (precedence 2)
•cs3—codepoint 3 (precedence 3)
•cs4—codepoint 4 (precedence 4)
•cs5—codepoint 5 (precedence 5)
•cs6—codepoint 6 (precedence 6)
•cs7—codepoint 7 (precedence 7)
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default
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Sets DSCP to default bit pattern of 000000.
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ef
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Sets DSCP to expedited forwarding bit pattern 101110.
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media
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Applies DSCP to media payload packets.
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signaling
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Applies DSCP to signaling packets.
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video rsvp-none
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Applies DSCP to video stream with no RSVP reservations.
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video rsvp-pass
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Applies DSCP to video stream with successful RSVP reservations.
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video rsvp-fail
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Applies DSCP to video stream with failed RSVP reservations.
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Command Default
DSCP is set to bit pattern af41.
Command Modes
Dial peer configuration
Command History
Release
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Modification
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12.2(2)T
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This command was introduced.It replaced the ip precedence (dial peer) command
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12.3(4)T
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Keywords were added to support DSCP configuration for video streams.
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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.
The following example specifies the DSCP for video streams:
ip qos dscp cs5 video rsvp-pass
ip qos dscp cs6 video rsvp-fail
Related Commands
Command
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Description
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call rsvp-sync
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Enables synchronization between Resource Reservation Protocol (RSVP) signaling and the voice signaling protocol.
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ip rsvp signalling dscp
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Specifies the DSCP to be used on all RSVP messages transmitted on an interface.
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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
value
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Average interval for RTCP report transmissions, in ms. Range is 1 to 65535. Default is 5000.
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Command Default
5000 ms
Command Modes
Global configuration
Command History
Release
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Modification
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12.2(2)XB
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This command was introduced.
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12.2(8)T
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This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
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12.2(11)T
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This command was implemented on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5800.
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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 5000
Related Commands
Command
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Description
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debug ccsip events
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Displays all SIP SPI event tracing and traces the events posted to SIP SPI from all interfaces.
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timer receive-rtcp
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Enables the RTCP timer and configures a multiplication factor for the RTCP timer interval.
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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
Release
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Modification
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11.3(1)T
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This command was introduced on the Cisco 3600 series.
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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 To maintain performance and scalability of the Cisco 5850 when using images before 12.3(4)T, enable no more than 10% of active calls with UDP checksum.
Examples
The following example calculates the UDP checksum for voice packets sent by dial peer 10:
Related Commands
Command
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Description
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loop-detect
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Enables loop detection for T1 for Voice over ATM, Voice over Frame Relay, and Voice over HDLC.
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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
Release
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Modification
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12.2(11)T
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This command was introduced on the Cisco 3600 series.
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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:
Related Commands
Command
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Description
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timer irr period
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Configures the IRR timer.
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