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icpif through irq global-request
- 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
- intensity
- 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 address trusted
- ip circuit
- ip dhcp-client forcerenew
- ip precedence (dial-peer)
- ip qos defending-priority
- ip qos dscp
- ip qos policy-locator
- ip qos preemption-priority
- ip rtcp report interval
- ip rtcp sub-rtcp
- ip udp checksum
- irq global-request
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.
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.
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 2333The following example shows the the error response when an undefined neighbor ID is entered:
Router(config-annexg-neigh)#no id def % Entry not valid, id not configured. To deconfigure id under different neighbor you have to expilicitly go into that neighbor and deconfigure the id.idle-voltage
To specify the idle voltage on a 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.
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).
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 }
Usage Guidelines
The ignore command applies to E&M digital voice ports associated with T1/E1 controllers. Repeat the command for each receive bit to be configured. Use this command with the define command.
Examples
To configure voice port 1/1 to ignore receive bits A, B, and C and to monitor receive bit D, enter the following commands:
voice-port 1/1 ignore rx-a-bit ignore rx-b-bit ignore rx-c-bit no ignore rx-d-bitTo configure voice port 1/0/0 to ignore receive bits A, C, and D and to monitor receive bit B, enter the following commands:
voice-port 1/0/0 ignore rx-a-bit ignore rx-c-bit ignore rx-d-bit no ignore rx-b-bitignore (interface)
To configure the serial interface to ignore the specified serial signals as the line up/down indicator, use the ignorecommand 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 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.
image encoding
To specify an encoding method for fax images associated with a Multimedia Mail over IP (MMoIP) dial peer, use the image encodingcommand in dial-peer configuration mode. To reset to the default, use the no form of this command.
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.
image resolution
To specify a particular fax image resolution for a specific multimedia mail over IP (MMoIP) dial peer, use the image resolutioncommand 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 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 series, Cisco 3600, Cisco 3725, and Cisco 3745.
Usage Guidelines
Use thiscommand 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.
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 + 2.15uF2.
900r
Resistive 900-ohm termination.
complex1
220 ohms + (820 ohms || 115 nF)3.
complex2
270 ohms + (750 ohms || 150 nF)4.
complex3
370 ohms + (620 ohms || 310 nF)5.
complex4
600r, line = 270 ohms + (750 ohms || 150 nF)6.
complex5
320 + (1050 ohms || 230 nF), line = 12 Kft7.
complex6
600r, line = 350 + (1000 ohms || 210 nF)8.
1 The plus symbol (+) indicates serial. The double pipe ( || ) indicates parallel.2 The plus symbol (+) indicates serial. The double pipe ( || ) indicates parallel.3 The plus symbol (+) indicates serial. The double pipe ( || ) indicates parallel.4 The plus symbol (+) indicates serial. The double pipe ( || ) indicates parallel.5 The plus symbol (+) indicates serial. The double pipe ( || ) indicates parallel.6 The plus symbol (+) indicates serial. The double pipe ( || ) indicates parallel.7 The plus symbol (+) indicates serial. The double pipe ( || ) indicates parallel.8 The plus symbol (+) indicates serial. The double pipe ( || ) indicates parallel.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 thiscommand 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.
NoteThe values in the syntax description 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.
If the impedance is set incorrectly (if there is an impedance mismatch), a significant amount of echo is generated (which could be masked if the echo-cancel command has been enabled). In addition, gains might not work correctly if there is an impedance mismatch.
Configuring the impedance on a voice port changes the impedance on both voice ports of a VPM card. This voice port must be shut down and then opened for the new value to take effect.
Examples
The following example configures an FXO voice port on the Cisco 3600 series router for an impedance of 600 ohms (real):
voice-port 1/0/0 impedance 600r shutdown/no shutdownThe following example configures an E&M voice port on a Cisco 2800 for an impedance of complex3:
voice-port 1/1 impedance complex3 shutdown/no shutdowninband-alerting
To enable inband alerting, use the inband-alertingcommand in the SIP user agent configuration mode. To disable inband alerting, use the no form of this command.
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 Router(config-sip-ua)# no inband-alertingRelated 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 ttlcommand in Annex G neighbor service configuration mode. To reset to the default, use the noform of this command.
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 thiscommand to indicate the duration of the relationship between border elements that participate in a service relationship.
Examples
The following example sets the inbound time-to-live value to 420 seconds (7 minutes):
Router(config-nxg-neigh-svc)# inbound ttl 420Related Commands
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.
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.
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 numbercommand 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.
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.
Syntax Description
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 5550123Related 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-numbercommand in dial-peer configuration mode. To reset to the default, use the no form of this command.
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 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.
Cisco IOS XE Release 3.3S
This command was integrated into Cisco IOS XE Release 3.3S.
Usage Guidelines
When a Cisco device is handling both modem and voice calls, it needs to be able to identify the service type of the call--meaning whether the incoming call to the server is a modem or a voice call. When the access server handles only modem calls, the service type identification is handled through modem pools. Modem pools associate calls with modem resources based on the dialed number identification service (DNIS). In a mixed environment, in which the server receives both modem and voice calls, you need to identify the service type of a call by using this command.
If you do not use this command, the server attempts to resolve whether an incoming call is a modem or voice call on the basis of the interface over which the call arrives. If the call comes in over an interface associated with a modem pool, the call is assumed to be a modem call; if a call comes in over a voice port associated with a dial peer, the call is assumed to be a voice call.
By default, there is no called number associated with the dial peer, which means that incoming calls are associated with dial peers by matching calling number with answer address, call number with destination pattern, or calling interface with configured interface.
Use this command to define the destination telephone number for a particular dial peer. For the on-ramp POTS dial peer, this telephone number is the DNIS number of the incoming fax call. For the off-ramp MMoIP dial peer, this telephone number is the telephone number of the destination fax machine.
This command applies to both VoIP and POTS dial peers and to on-ramp and off-ramp store-and-forward fax functions.
This command is also used to provide a matching VoIP dial peer on the basis of called number when fax or modem pass-through with named signaling events (NSEs) is defined globally on a terminating gateway.
You can ensure that all calls will match at least one dial peer by using the following commands:
Router(config)# dial-peer voice tag voip Router(config-dial-peer)# incoming called-number.Examples
The following example configures calls that come into the router with a called number of 555-0163 as being voice calls:
dial peer voice 10 pots incoming called-number 5550163The following example sets the number (310) 555-0142 as the incoming called number for MMoIP dial peer 10:
dial-peer voice 10 mmoip incoming called-number 3105550142incoming calling-number (call filter match list)
To configure debug filtering for incoming calling numbers, use the incoming calling-number command in call filter match list configuration mode. To disable, use the no form of this command.
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.
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 5550125Related 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.
Examples
The following example shows the voice call debug filter set to match incoming dial peer 12:
call filter match-list 1 voice incoming dialpeer 12Related 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.
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 192.168.10.255Related 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.
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 192.168.10.255Related 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}
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
Cisco 2600, Cisco 3600 Series and Cisco 3700 Series
Syntax Description
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.
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.
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.
:port
Specifies the value for the parent argument. The valid entry is 0.
:D
Indicates the D channel associated with ISDN PRI.
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-T V.35, X.21, Bisync, Polled async).
Examples
The following example shows the voice call debug filter set to match incoming port 1/1/1 on a Cisco 3660 voice gateway:
call filter match-list 1 voice incoming port 1/1/1incoming 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.
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.
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 5550156:
call filter match-list 1 voice incoming secondary-called-number 5550156Related 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.
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 192.168.10.255Related 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.
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 192.168.10.255Related 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 voice configuration mode. To remove the URI voice class from the dial peer, use the no form of this command.
H.323 Session Protocol
incoming uri { called | calling } tag
no incoming uri { called | calling }
Session Initiation Protocol (SIP) Session Protocol
incoming uri { from | request | to | via } tag
no incoming uri { from | request | to | via }
Syntax Description
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.
tag
Alphanumeric label that uniquely identifies the voice class. This tag argument must be configured with the voice class uri command.
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.
via
Via header in an incoming SIP Invite message.
Usage Guidelines
- Before you use this command, configure the voice class by using the voice class uri command.
- The keywords depend on whether the dial peer is configured for SIP with the session protocol sipv2 command. The from, request, to, and via keywords are available only for SIP dial peers. The called and calling keywords are available only for dial peers using H.323.
- This command applies rules for dial peer matching. The tables below show the rules and the order in which they are applied when the incoming uri command is used. The gateway compares the dial-peer command to the call parameter in its search to match an inbound call to a dial peer. All dial peers are searched based on the first match criterion. Only if no match is found does the gateway move on to the next criterion.
Table 1 Dial-Peer Matching Rules for Inbound URI in SIP Calls Match Order
Cisco IOS Command
Incoming Call Parameter
1
incoming uri via
Via URI
2
incoming uri request
Request-URI
3
incoming uri to
To URI
4
incoming uri from
From URI
5
incoming called-numbe r
Called number
6
answer-address
Calling number
7
destination-pattern
Calling number
8
carrier-id source
Carrier-ID associated with the call
Table 2 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
NoteCalls using an E.164 number, rather than a URI, use the dial-peer matching rules that existed prior to Cisco IOS Release 15.1(2)T. For information, see theDial Peer Configuration on Voice Gateway Routers document, Cisco IOS Voice Configuration Library.
Examples
The following example matches on the destination telephone URI in incoming H.323 calls by using the ab100 voice class:
dial-peer voice 100 voip incoming uri called ab100The following example matches on the incoming via URI for SIP calls by using the ab100 voice class:
dial-peer voice 100 voip session protocol sipv2 incoming uri via ab100Related Commands
Command
Description
answer-address
Specifies the calling number to match for a dial peer.
debug voice uri
Displays debugging messages related to URI voice classes.
destination-pattern
Specifies the telephone number to match for a dial peer.
dial-peer voice
Enters dial peer voice configuration mode to create or modify a dial peer.
incoming called-number
Specifies the 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.
Usage Guidelines
Use this command to define one or more numbers for a voice class called number, or a range of numbers for a voice class called number pool. You can define multiple indexes for any inbound or outbound voice class called number or voice class called number pool.
When defining a range of numbers for a called number pool:
- The range of numbers must be in E.164 format.
- The beginning number and ending number must be the same length.
- The last digit of each number must be 0 to 9.
- Leading '+' (if used) must be defined from in the range of called numbers.
Examples
The following example shows the configuration for indexes in voice class called number pool 100:
voice class called number pool 100 index 1 4085550100 - 4085550111 (Range of called numbers are 4085550100 up to 4085550111) index 2 +3227045000The following example shows configuration for indexes in voice class called number outbound 222:
voice class called number outbound 222 index 1 4085550101 index 2 4085550102 index 2 4085550103info-digits
To automatically add the two-digit prefix to the beginning of a dialed number string associated with the given POTS dial peer, use the info-digits command in dial-peer configuration mode. To specify that the two-digit prefix is "00" use the default info-digits form of this command. To prevent the router from automatically adding the two-digit prefix to the beginning of the POTS dial peer, use the no form of this command.
Syntax Description
prefix-number
Specifies the two-digit prefix that the router will automatically add to the dialed number string for the given POTS dial peer to identify the type of phone originating the call. This value cannot contain any more or less than two digits. Valid values include:
- 00--Regular line
- 01--4- and 8-party
- 06--Hotel or Motel
- 07--Coinless
- 10--Test call
- 27--Coin
- 95--Test call
Note Values 12 through 19 cannot be assigned because of conflicts with international 20 Automatic Identification of Outward listed directory number sent.
Command Default
The dialed number string is added with 00, indicating that the dialed number string originates from a regular line.
Usage Guidelines
This command adds a two-digit prefix to the dialed number string for the POTS dial peer that will enable you to dynamically redirect the outgoing call. The info-digits command is only available for POTS dial peers tied to a voice-port that corresponds to Feature Group-D (FGD) Exchange Access North American (EANA) signaling that provides specific call services such as emergency 911 calls in the United States. Configuring the info-digit command for other voice port types is not advised and may yield undesirable results.
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-typecommand 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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
input gain
To configure a specific input gain value or to enable automatic gain control, use the input gain command in voice-port configuration mode. To disable the selected value of the 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
The gain, in decibels (dB), to be inserted at the receiver side of the interface. The range is integers from –6 to 14. The default is 0 decibels.
auto-control
Enables automatic gain control.
auto-dBm
(Optional) The target speech level, in decibels per milliwatt (dBm), to be achieved at the receiver side of the interface. The range is integers from –30 to 3. The default is –9 dBm.
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
This command was modified. 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
This command was modified. The auto-control keyword and auto-dBm argument were added.
Usage Guidelines
A system-wide loss plan must be implemented by using both the input gain and output attenuation commands. You must consider other equipment (including PBXs) in the system when you create a loss plan. The default value for the input gain command assumes that a standard transmission loss plan is in effect; that is, 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 by increasing the output attenuation.
You can increase the gain of a signal coming into the device. If the voice level is too low, use the input gain command to increase the input gain.
Typical Land Mobile Radio (LMR) signaling systems send 0 dB out and expect –10 dB in. Setting the 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. Radio network loss and other environmental factors could cause the speech level arriving at a device from an LMR system to 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:
intensity
To configure the intensity or depth of the noise reduction process, use the intensity command in media profile configuration mode. To disable the configuration, use the no form of this command.
Usage Guidelines
Use the intensity command to configure the intensity or depth of the noise reduction process. You must create a media profile for noise reduction and then configure the intensity level.
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.
Examples
The following example configures the access-server interfaces for RLM servers "Loopback1" and "Loopback2":
interface Loopback1 ip address 10.1.1.1 255.255.255.255 interface Loopback2 ip address 10.1.1.2 255.255.255.255 rlm group 1 server r1-server link address 10.1.4.1 source Loopback1 weight 4 link address 10.1.4.2 source Loopback2 weight 3Related 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.
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 timeslotscontroller 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.
Usage Guidelines
This command is used specifically in Voice over IP (VoIP) applications that require release of the ISDN PRI signaling time slot for RLM configurations.
Examples
The following example configures a D-channel interface for a Signaling System 7 (SS7)-enabled shared T1 link:
controller T1 1 pri-group timeslots 1-3 nfas_d primary nfas_int 0 nfas_group 0 rlm-group 0 channel group 23 timeslot 24 end ! D-channel interface is created for configuration of ISDN parameters: interface Dchannel1 isdn T309 4000 endinterface 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 ftpcommand 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.
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-sizecommand. To manually flush the event log buffer, use the interface dump event-log command in privileged EXEC mode.
NoteEnabling the gateway to write event logs to FTP could adversely impact gateway memory resources in some scenarios, for example, when:
- The gateway is consuming high processor resources and FTP does not have enough processor resources to flush the logged buffers to the FTP server.
- The designated FTP server is not powerful enough to perform FTP transfers quickly
- Bandwidth on the link between the gateway and the FTP server is not large enough
- The gateway is receiving a high volume of short-duration calls or calls that are failing
You should enable FTP dumping only when necessary and not enable it in situations where it might adversely impact system performance.
Examples
The following example specifies that interface event log are written to an external file named int_elogs.log on a server named ftp-server:
application monitor interface event-log dump ftp ftp-server/elogs/int_elogs.log username myname password 0 mypassThe following example specifies that application event logs are written to an external file named int_elogs.log on a server with the IP address of 10.10.10.101:
application monitor interface event-log dump ftp 10.10.10.101/elogs/int_elogs.log username myname password 0 mypassRelated 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.
Usage Guidelines
This command limits the severity level of the events that are logged; it does not enable logging. You must use this command with the interface event-log command, which enables event logging for all application interfaces.
Examples
The following example enables event logging for error events only:
application monitor interface event-log error-onlyRelated 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-sizecommand in application configuration monitor mode. To reset to the default, use the no form of this 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:
application monitor interface event-log max-buffer-size 8Related 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.
Usage Guidelines
Only the specified number of records from the most recently accessed servers are kept.
Examples
The following example sets the maximum saved records to 50:
application monitor interface max-server-records 50Related 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.
Usage Guidelines
To display the interface statistics enabled by this command, use the show call application interface command. To reset the interface counters to zero, use the clear call application interface command.
Examples
The following example enables statistics collection for application interfaces:
application monitor interface statsRelated 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 address trusted
To set up toll-fraud prevention support on a device, use the ip address trusted command in voice-service configuration mode. To disable the setup, use the no form of this command.
ip address trusted { authenticate | call-block cause code | list}
no ip address trusted { authenticate | call-block cause | list}
Syntax Description
authenticate Enables IP address authentication on incoming H.323 or Session Initiation Protocol (SIP) trunk calls.
call-block cause code Enables issuing a cause code when an incoming call is rejected on the basis of failed IP address authentication. By default, the device issues a call-reject (21) cause code.
list Enables manual addition of IPv4 and IPv6 addresses to the trusted IP address list.
Usage Guidelines
Use the ip address trusted command to modify the default behavior of a device, which is to not trust a call setup from a VoIP source. With the introduction of this command, the device checks the source IP address of the call setup before routing the call.
A device rejects a call if the source IP address does not match an entry in the trusted IP address list that is a trusted VoIP source. To create a trusted IP address list, use the ip address trusted list command in voice service configuration mode, or use the IP addresses that have been configured using the session target command in dial peer configuration mode. You can issue a cause code when an incoming call is rejected on the basis of failed IP address authentication.
Examples
The following example displays how to enable IP address authentication on incoming H.323 or SIP trunk calls for toll-fraud prevention support.:
Device(config)# voice service voip Device(conf-voi-serv)# ip address trusted authenticateThe following example displays the number of rejected calls:
Device# show call history voice last 1 | inc Disc DisconnectCause=15 DisconnectText=call rejected (21) DisconnectTime=343939840 msThe following example displays the error message code and the error description:
Device# show call history voice last 1 | inc Error InternalErrorCode=1.1.228.3.31.0The following example displays the error description:
Device# show voice iec description 1.1.228.3.31.0 IEC Version: 1 Entity: 1 (Gateway) Category: 228 (User is denied access to this service) Subsystem: 3 (Application Framework Core) Error: 31 (Toll fraud call rejected) Diagnostic Code: 0The following example shows how to issue a cause code when an incoming call is rejected on the basis of failed IP address authentication:
Device(config)# voice service voip Device(conf-voi-serv)# ip address trusted call-block cause call-rejectThe following example displays how to enable the addition of IP addresses to a trusted IP address list:
Device(config)# voice service voip Device(conf-voi-serv)# ip address trusted listRelated Commands
Command
Description
debug voip ccapi inout Traces the execution path through the call control API.
show call history voice Displays the call history table for voice calls.
show ip address trusted list Displays a list of valid IP addresses for incoming H.323 or SIP trunk calls.
voice iec syslog Enables viewing of internal error codes as they are encountered in real time.
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.
Usage Guidelines
You can use the ip circuit command only when no calls are active. You can define multiple carrier IDs, and the ordering does not matter. IP circuit default only is mutually exclusive with defining carriers with circuit carrier id.
If ip circuit default only is specified, the maximum calls value is set to 1000.
Examples
The following example specifies a default circuit and maximum number of calls:
voice service voip no allow-connections any to pots no allow-connections pots to any allow-connections h323 to h323 h323 ip circuit max-calls 1000 ip circuit default onlyThe following example specifies a default carrier and incoming source carrier:
voice service voip no allow-connections any to pots no allow-connections pots to any allow-connections h323 to h323 h323 ip circuit carrier-id AA reserved-calls 200 ip circuit max-calls 1000ip dhcp-client forcerenew
To enable forcerenew-message handling on the DHCP client when authentication is enabled, use the ip dhcp-client forcerenew command in global configuration mode. To disable the forced authentication, use the no form of this command.
Examples
The following example shows how to enable DHCP forcerenew-message handling on the DHCP client:
Router(config)# ip dhcp-client forcerenewRelated Commands
Command
Description
ip dhcp client authentication key-chain
Specifies the key chain to be used in DHCP authentication requests.
ip dhcp client authentication mode
Specifies the type of authentication to be used in DHCP messages on the interface.
key chain
Identifies a group of authentication keys for routing protocols.
ip precedence (dial-peer)
To set IP precedence (priority) for packets sent by the dial peer, use the ip precedencecommand in dial-peer configuration mode. To reset to the default, use the no form of this command.
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.
ip qos defending-priority
To configure the Resource Reservation Protocol (RSVP) defending priority value for determining quality of service (QoS), use the ip qos defending-prioritycommand in dial peer configuration mode. To disable RSVP defending priority as a QoS factor, use the no form of this command.
Command Default
The RSVP defending priority value is disabled and is not a factor in determining QoS.
Usage Guidelines
To configure the RSVP defending priority value, use the ip qos defending-priority command in dial peer configuration mode. The defending priority value is passed to the QoS module during reservation initiation. In a situation where there is not enough bandwidth available to support all calls, this setting enables an existing call to avoid being preempted by a new call unless the preemption priority of the new call is higher than the defending priority of the existing call.
Examples
The following example shows how to specify the RSVP defending priority value:
dial-peer voice 100 voip ip qos defending-priority 1111Related Commands
Command
Description
acc-qos
Defines the acceptable QoS for inbound and outbound calls on a VoIP dial peer.
ip qos dscp
Configures the DSCP value for QoS.
ip qos policy-locator
Configures the application ID of RSVP.
ip qos preemption-priority
Configures the RSVP preemption priority.
ip rsvp policy preempt
Enables RSVP to take bandwidth from lower-priority reservations and give it to new, higher-priority reservations.
req-qos
Requests a particular QoS using RSVP to be used in reaching a specified dial peer in VoIP.
show-sip-ua calls
Displays the active UAC and UAS information for SIP calls on a Cisco IOS device.
voice-class sip rsvp-fail-policy
Configures RSVP failure policies.
ip qos dscp
To configure the differentiated services code point (DSCP) value for quality of service (QoS), use the ip qos dscp command in dial peer configuration mode. To disable DSCP as a QoS factor, set the DSCP value to default (which sets the value to the 000000 bit pattern). To set DSCP values to their default settings, use the noform of this command.
ip qos dscp { dscp-value | set-af | set-cs | default | ef } { signaling | media [ rsvp-pass | rsvp-fail ] | video [ rsvp-none | rsvp-pass | rsvp-fail ] }
no ip qos dscp { dscp-value | set-af | set-cs | default | ef } { signaling | media [ rsvp-pass | rsvp-fail ] | video [ rsvp-none | rsvp-pass | rsvp-fail ] }
Syntax Description
dscp-value
DSCP value. Valid entries are from 0 to 63.
set-af
An assured forwarding bit pattern as the DSCP value:
set-cs
C lass-selector code point as the DSCP value:
default
Specifies the default bit pattern 000000 as the DSCP value.
ef
Specifies the expedited forwarding bit pattern 101110 as the DSCP value.
signaling
Specifies that the DSCP value applies to signaling packets.
media
Specifies that the DSCP value applies to media packets (voice and fax).
rsvp-pass
(Optional) Specifies that the DSCP value applies to packets with successful Resource Reservation Protocol (RSVP) reservations.
rsvp-fail
(Optional) Specifies that the DSCP value applies to packets (media or video) with failed RSVP reservations.
video
Specifies that the DSCP value applies to video packets. This option is valid only for Cisco Unified Communications Manager Express (Cisco Unified CME) on a Cisco Unified Border Element.
rsvp-none
(Optional) Specifies that the DSCP value applies to video packets with no RSVP reservations (valid only for video packets.)
Command Default
The DSCP default values are as follows:
- The default DSCP value for all signaling packets is af31.
- The default DSCP value for all media (voice and fax) packets is ef.
- The default DSCP value for all video packets is af41.
Command History
Release
Modification
12.2(2)T
This command was introduced. It replaced the ip precedence (dial peer) command
12.3(4)T
This command was modified. Keywords were added to support DSCP configuration for video streams.
12.4(22)T
This command was modified. Keywords were added to apply a DSCP value to media (voice and fax) packets with a specified (successful or failed) RSVP connection.
Cisco IOS XE Release 3.3S
This command was integrated into Cisco IOS XE Release 3.3S.
Usage Guidelines
To configure voice, signaling, and video traffic priorities, use the ip qos dscp command in dial peer configuration mode. The recommended value for media (voice and fax) packets is ef;for signaling packets, the recommended value is af31; and for video packets, it is af41 (all defaults).
Additionally, before you can specify RSVP QoS, you must first use the ip rsvp bandwidth command to enable RSVP on the IP interface.
Examples
The following example shows how to set the DSCP value to a class-selector code point value of 1 and apply that DSCP setting to media (voice and fax) payload packets with no RSVP configured:
dial-peer voice 1 voip ip qos dscp cs1 mediaThe following example shows how to set the DSCP value to the expedited forwarding bit pattern and apply that DSCP setting to media (voice and fax) payload packets with a successful RSVP connection:
dial-peer voice 1 voip ip qos dscp ef media rsvp-passThe following example shows how to set the DSCP value to an assured forwarding code point value of 22 and apply that DSCP setting to all signaling packets:
dial-peer voice 1 voip ip qos dscp af22 signalingThe following example shows how to set the DSCP value to an assured forwarding code point value of 43 and apply that DSCP setting to video packets with a successful RSVP connection:
dial-peer voice 100 voip ip qos dscp af43 video rsvp-passRelated Commands
Command
Description
call rsvp-sync
Enables synchronization between RSVP signaling and the voice signaling protocol.
ip qos defending-priority
Configures the RSVP defending priority value.
ip qos policy-locator
Configures the application ID of RSVP.
ip qos preemption-priority
Configures the RSVP preemption priority value.
ip rsvp bandwidth
Enables RSVP for IP on an interface.
ip rsvp signalling dscp
Configures the DSCP settings to be used on RSVP messages on an interface.
ip qos policy-locator
To configure a quality of service (QoS) policy-locator (application ID) used to deploy Resource Reservation Protocol (RSVP) policies for specifying bandwidth reservations on Cisco IOS Session Initiation Protocol (SIP) devices, use the ip qos policy-locatorcommand in dial peer configuration mode. To delete an application policy, use the no form of this command.
ip qos policy-locator { video | voice } [ app app-string ] [ guid guid-string ] [ sapp subapp-string ] [ ver version-string ]
no ip qos policy-locator { video | voice } [ app app-string ] [ guid guid-string ] [ sapp subapp-string ] [ ver version-string ]
Syntax Description
video
Specifies that the application ID applies to RSVP for video streams.
voice
Specifies that the application ID applies to RSVP for voice streams.
app
(Optional) Specifies an application.
app-string
Application ID. Consists of 1 to 31 alphanumeric characters.
guid
(Optional) Specifies a globally unique identifier (GUID).
guid-string
GUID. Consists of 1 to 31 alphanumeric characters.
sapp
(Optional) Specifies a subapplication.
sapp-string
Subapplication ID. Consists of 1 to 31 alphanumeric characters.
ver
(Optional) Specifies a version.
ver-string
Version ID. Consists of 1 to 15 alphanumeric characters.
Usage Guidelines
In Cisco IOS software, the RSVP can process and accept requests by referring to multiple bandwidth pools. To enhance the granularity of local policy match criteria on Cisco IOS SIP devices, bandwidth pools can include policies based on application IDs. You can use these application-specific IDs to reserve bandwidth for each until specified bandwidth limits are reached.
To prevent one application type from consuming all bandwidth, RFC 2872 , Application and Sub Application Identity Policy Element for Use with RSVP , allows for the creation of separate bandwidth reservation pools. For example, an RSVP reservation pool can be created for voice traffic and another for video traffic so that reservations tagged with these application IDs can then be matched to the interface bandwidth pools using RSVP local policies. To limit bandwidth per application, though, you must configure a bandwidth limit for each application and configure each with a reservation flag that associates the application with the appropriate bandwidth limit.
Before you can configure bandwidth limits for any application-specific policy, however, you must create application IDs. To create application IDs (application-specific reservation profiles), use the ip qos policy-locatorcommand in dial peer configuration mode. After creating the necessary application IDs, you can then use the appropriate commands listed in the "Related Commands" section to configure bandwidth reservation. However, this feature is available only on supported devices that are running Cisco IOS Release 12.4(22)T or a later release.
For more information about configuring SIP RSVP features, see the "Configuring SIP RSVP Features" chapter in the Cisco IOS SIP Configuration Guide. For more general information about the application-specific policy feature, see the "Configuring RSVP" chapter in the RSVP section of the "Signaling" part in the Cisco IOS Quality of Service Solutions Configuration Guide.
Examples
The following example shows how to configure a policy for the application ID:
dial-peer voice 100 voip ip qos policy-locator voice app MyApp1 sapp MySubApp4Related Commands
Command
Description
acc-qos
Defines the acceptable QoS for inbound and outbound calls on a VoIP dial peer.
handle-replaces
Configures fallback to legacy handling of SIP INVITE.
ip qos defending-priority
Configures the RSVP defending priority value.
ip qos dscp
Sets the DSCP value for QoS.
ip qos preemption-priority
Configures the RSVP preemption priority value.
ip rsvp bandwidth
Enables RSVP for IP on an interface.
ip rsvp policy default-reject
Configures blocking or passing of all messages that do not match any existing RSVP policies.
ip rsvp policy identity
Defines RSVP application IDs used to deploy RSVP policies.
ip rsvp policy preempt
Enables RSVP to take bandwidth from lower-priority reservations and give it to new, higher-priority reservations.
maximum (local policy)
Configures a local policy that limits RSVP resources.
preempt-priority
Configures RSVP QoS priorities to be inserted into PATH and RESV messages when they are not signaled from an upstream or downstream neighbor or local client application.
req-qos
Requests a particular QoS using RSVP to be used in reaching a specified dial peer in VoIP.
show sip-ua calls
Displays the active UAC and UAS information on SIP calls.
voice-class sip rsvp-fail-policy
Specifies the action that takes place when RSVP negotiation fails.
ip qos preemption-priority
To configure the Resource Reservation Protocol (RSVP) preemption priority value for determining quality of service (QoS), use the ip qos preemption-prioritycommand in dial peer configuration mode. To disable RSVP preemption priority as a QoS factor, use the no form of this command.
Command Default
The RSVP preemption priority value is disabled and is not a factor in determining QoS.
Usage Guidelines
To configure an RSVP preemption priority value, use the ip qos preemption-prioritycommand in dial peer configuration mode. The preemption priority value is passed to the QoS module during reservation initiation. In a situation where there is not enough bandwidth available to support all calls, this setting enables a new call to preempt an existing call unless the defending priority of the existing call is higher than the preemption priority of the new call.
Examples
The following example shows how to specify the RSVP preemption priority value:
dial-peer voice 100 voip ip qos preemption-priority 1111Related Commands
Command
Description
acc-qos
Defines the acceptable QoS for inbound and outbound calls on a VoIP dial peer.
ip qos dscp
Configures the DSCP value for QoS.
ip qos policy-locator
Configures the application ID of RSVP.
ip qos defending-priority
Configures the defending priority value of RSVP.
ip rsvp policy preempt
Enables RSVP to take bandwidth from lower-priority reservations and give it to new, higher-priority reservations.
req-qos
Requests a particular QoS using RSVP to be used in reaching a specified dial peer in VoIP.
show-sip-ua calls
Displays the active UAC and UAS information for SIP calls on a Cisco IOS device.
voice-class sip rsvp-fail-policy
Configures RSVP failure policies.
ip rtcp report interval
To configure the average reporting interval between subsequent Real-Time Control Protocol (RTCP) report transmissions, use the ip rtcp report intervalcommand in global configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(8)T
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.
12.2(11)T
This command was implemented on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5800.
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.
ip rtcp sub-rtcp
To specify sub-Real-Time Control Protocol (RTCP) message types, use the ip rtcp sub-rtcpcommand in global configuration mode. To disable the configuration, use the no form of this command.
ip udp checksum
To calculate the UDP checksum for voice packets sent by the dial peer, use the ip udp checksumcommand in dial-peer configuration mode. To disable this feature, use the no form of this command.
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.
NoteTo maintain performance and scalability of the Cisco AS5850 when using images before Cisco IOS Release 12.3(4)T, enable no more than 10% of active calls with UDP checksum.
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.
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.
