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This chapter documents new commands. All other commands used with this feature are documented in the Cisco IOS Release 11.3 command reference documents.
The following new commands are used to configure and monitor Voice over IP:
Table 3-1 describes the syntax used with the commands in this chapter.
| Convention | Description |
|---|---|
| boldface font | Commands and keywords. |
| italic font | Command input that is supplied by you. |
| [ ] | Keywords or arguments that appear within square brackets are optional. |
| { x | x | x } | A choice of keywords (represented by x) appears in braces separated by vertical bars. You must select one. |
| ^ or Ctrl | Represent the key labeled Control. For example, when you read ^D or Ctrl-D, you should hold down the Control key while you press the D key. |
screen font
| Examples of information displayed on the screen. |
boldface screen font
| Examples of information that you must enter. |
| < > | Nonprinting characters, such as passwords, appear in angled brackets. |
| [ ] | Default responses to system prompts appear in square brackets. |
| Note | Means reader take note. Notes contain helpful suggestions or references to additional information and material. |
| best-effort | Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation. |
| controlled-load | Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to assure that preferential service is received even when the bandwidth is overloaded. |
| guaranteed-delay | Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queuing if the bandwidth reserved is not exceeded. |
The default value is best-effort. Using the no form of this command is the same as the default.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the acc-qos dial peer command to generate an SNMP event if the quality of service for specified dial peer drops below the specified level. When a dial peer is used, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation Protocol (RSVP) to request quality of service guarantees from the network.
To select the most appropriate value for this command, you need to be familiar with the amount of traffic this connection supports and what kind of impact you are willing to have on it. The Cisco IOS software generates a trap message when the bandwidth required to provide the selected quality of service is not available.
This command is only applicable to VoIP peers.
The following example selects guaranteed-delay as the specified level below which an SNMP trap message will be generated:
dial-peer voice 10 voip acc-qos guaranteed-delay
You can use the master index or search online to find documentation of related commands.
req-qos
| string | Series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are:
· Digits 0 through 9, letters A through D, pound sign (#), and asterisk (*), which represent specific digits that can be entered. · Plus sign (+), which is optionally used as the first digit to indicate an E.164 standard number. · Comma (,), which inserts a pause between digits. · Period (.), which matches any entered digit. |
The default value is enabled with a null string.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the answer-address command to identify the origin (or dial peer) of incoming calls from the IP network. Cisco IOS software identifies the dial peers of a call in one of two ways: either by identifying the interface through which the call is received or through the telephone number configured with the answer-address command. In the absence of a configured telephone number, the peer associated with the interface will be associated with the incoming call.
For calls coming in from a POTS interface, the answer-address command is not used to select an incoming dial peer. The incoming POTS dial peer is selected on the basis of the port configured for that dial peer.
This command is applicable to both VoIP and POTS dial peers.
The following example configures the E.164 telephone number, "379-9626" as the dial peer of an incoming call:
dial-peer voice 10 pots answer-address +3799626
You can use the master index or search online to find documentation of related commands.
destination-pattern
port
prefix
| g711alaw | G.711 A-Law 64,000 bits per second (bps) |
| g711ulaw | G.711 u-Law 64,000 bps |
| g729r8 | G.729 8000 bps. |
The default value for this command is g729r8.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the codec command to define a specific voice coder rate of speech for a dial peer.
For toll quality, use g711alaw or g711ulaw. These values provide high-quality voice transmission but use a significant amount of bandwidth. For almost toll quality (and a significant savings in bandwidth), use the g729r8 value.
If codec values for the VoIP peers of a connection do not match, the call will fail.
This command is only applicable to VoIP peers.
The following example configures a voice coder rate that provides toll quality and uses a relatively high amount of bandwidth:
dial-peer voice 10 voip codec g711alaw
This command has no arguments or keywords.
The default value for this command is enabled.
Voice-port configuration.
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the comfort-noise command to generate background noise to fill silent gaps during calls if VAD is activated. If comfort noise is not enabled, and VAD is enabled at the remote end of the connection, the user will hear dead silence when the remote party is not speaking.
The configuration of comfort noise only affects the silence generated at the local interface; it does not affect the use of VAD on either end of the connection, or the silence generated at the remote end of the connection.
The following example enables background noise:
voice-port 1/0/0 comfort-noise
You can use the master index or search online to find documentation of related commands.
vad
| plar | Specifies a private line auto ringdown (PLAR) connection. PLAR is handled by associating a peer directly with an interface; when an interface goes off-hook, the peer is used to set up the second call leg and conference them together without the caller having to dial any digits. |
| string | Specifies the destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number. |
The default value for this command is no connection.
Voice-port configuration.
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the connection command to specify a connection mode for a specific interface. Use the connection plar command to specify a PLAR interface. The string you configure for this command is used as the called number for all calls coming in over this voice port. The destination dial peer is determined on the basis of this called number.
If the connection command is not configured, the standard session application outputs a dial tone when the interface goes off-hook until enough digits are collected to match a dial-peer and complete the call.
The following example selects PLAR as the connection mode, with a destination telephone number of 379-9262:
voice-port 1/0/0 connection plar 3799262
You can use the master index or search online to find documentation of related commands.
session-protocol
| australia | Specifies an analog voice interface-related default tone, ring, and cadence setting for Australia. |
| brazil | Specifies an analog voice interface-related default tone, ring, and cadence setting for Brazil. |
| china | Specifies an analog voice interface-related default tone, ring, and cadence setting for China. |
| finland | Specifies an analog voice interface-related default tone, ring, and cadence setting for Finland. |
| france | Specifies an analog voice interface-related default tone, ring, and cadence setting for France. |
| germany | Specifies an analog voice interface-related default tone, ring, and cadence setting for Germany. |
| japan | Specifies an analog voice interface-related default tone, ring, and cadence setting for Japan. |
| northamerica | Specifies an analog voice interface-related default tone, ring, and cadence setting for North America. |
| unitedkingdom | Specifies an analog voice interface-related default tone, ring, and cadence setting for the United Kingdom. |
The default value for this command is northamerica.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the cptone command to specify a regional analog voice interface-related tone, ring, and cadence setting for a specified voice port. This command only affects the tones generated at the local interface. It does not affect any information passed to the remote end of a connection, or any tones generated at the remote end of a connection.
The following example configures North America as the call progress tone locale:
voice-port 1/0/0 cptone northamerica
| string | Character string from 1 to 255 characters. |
The default for this command is enabled with a null string.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the description command to include descriptive text about this voice-port connection. This information is displayed when you issue a show command and does not affect the operation of the interface in any way.
The following example identifies this voice port as being connected to the Purchasing department:
voice-port 1/0/0 description purchasing_dept
| string | Series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are:
· Digits 0 through 9, letters A through D, pound sign (#), and asterisk (*), which represent specific digits that can be entered. · Plus sign (+), which is optionally used as the first digit to indicate an E.164 standard number. · Comma (,), which inserts a pause between digits. · Period (.), which matches any entered digit. |
The default value for this command is enabled with a null string.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the destination-pattern command to define the E.164 telephone number for this dial peer. For the Cisco IOS software to recognize a series of digits as the destination pattern, it must be preceded by a "+."
This pattern is used to match dialed digits to a dial peer. The dial peer is then used to complete the call.
This command is applicable to both VoIP and POTS dial peers.
The following example configures the E.164 telephone number, "479-7922," for a dial peer:
dial-peer voice 10 pots destination-pattern +4797922
You can use the master index or search online to find documentation of related commands.
answer-address
prefix
| max-size number | Specifies the maximum size of the call history table. Valid entries are from 0 to 500 table entries. A value of 0 will prevent any history from being retained. |
| retain-timer number | Specifies the length of time, in minutes, for entries in the call history table. Valid entries are from 0 to 2147483647 minutes. A value of 0 will prevent any history from being retained. |
The default call history table length is 50 table entries. The default retain timer is 15 minutes.
Global configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
The following example configures the call history table to hold 400 entries, with each entry remaining in the table for 10 minutes:
configure terminal dial-control-mib max-size 400 dial-control-mib retain-timer 10
| number | Digit(s) defining a particular dial peer. Valid entries are from 1 to 2147483647. |
| voip | Indicates that this is a VoIP peer using voice encapsulation on the POTS network. |
| pots | Indicates that this is a POTS peer using Voice over IP encapsulation on the IP backbone. |
Global configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the dial-peer voice global configuration command to switch to the dial peer configuration mode from the global configuration mode. Use the exit command to exit the dial peer configuration mode and return to the global configuration mode.
The following example accesses the dial peer configuration mode and configures a POTS peer identified as dial peer 10:
configure terminal dial-peer voice 10 pots
You can use the master index or search online to find documentation of related commands.
voice-port
| dtmf | Specifies a touch-tone dialer. |
| pulse | Specifies a pulse dialer. |
The default value for this command is dtmf.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the dial-type command to specify an out-dialing type for an FXO or E&M voice port interface; this command is not applicable to FXS voice ports because they do not generate out-dialing. Voice ports can always detect dtmf and pulse signals. This command does not affect voice port dialing detection.
The dial-type command affects out-dialing as configured for the dial peer.
The following example configures a voice port to support a touch-tone dialer:
voice-port 1/0/0 dial-type dtmf
| value | Number of milliseconds the echo-canceller will cover on a given signal. Valid values are 16, 24, and 32. |
The default value for this command is 16.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the echo-cancel coverage command to adjust the coverage size of the echo canceller. This command enables cancellation of voice that is sent out the interface and received back on the same interface within the configured amount of time. If the local loop (the distance from the analog interface to the connected equipment producing the echo) is longer, the configured value of this command should be extended.
If you configure a longer value for this command, it will take the echo canceller longer to converge; in this case, the user might hear slight echo when the connection is initially set up. If the configured value for this command is too short, the user might hear some echo for the duration of the call because the echo canceller is not cancelling the longer delay echoes.
There is no echo or echo cancellation on the IP side of the connection.
The following example adjusts the size of the echo canceller to 16 milliseconds:
voice-port 1/0/0 echo-cancel enable echo-cancel coverage 16
You can use the master index or search online to find documentation of related commands.
echo-cancel enable
This command has no arguments or keywords.
The default value for this command is enabled for all interface types.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
The echo-cancel command enables cancellation of voice that is sent out the interface and is received back on the same interface. Disabling echo cancellation might cause the remote side of a connection to hear an echo. Because echo cancellation is an invasive process that can minimally degrade voice quality, this command should be disabled if it is not needed.
The echo-cancel command does not affect the echo heard by the user on the analog side of the connection.
There is no echo path for a 4-wire E&M interface. The echo canceller should be disabled for that interface type.
The following example enables the echo cancel feature for 16-millisecond echo coverage:
voice-port 1/0/0 echo-cancel enable echo-cancel coverage 16
You can use the master index or search online to find documentation of related commands.
echo-cancel coverage
non-linear
| value | Integers that represent the ITU specification for quality of voice as described in G.113. Valid entries are from 0 to 20, with 0 representing toll quality. |
The default value for this command is 10.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Voice over IP monitors the quality of voice received over the network. Use the expect-factor command to specify when the router will generate an SNMP trap to the network manager.
This command is only applicable to VoIP peers.
The following example configures toll quality of voice when connecting to a dial peer:
dial-peer voice 10 voip expect-factor 0
| 2400 | Specifies a fax transmission speed of 2400 bits per second (bps). |
| 4800 | Specifies a fax transmission speed of 4800 bps. |
| 7200 | Specifies a fax transmission speed of 7200 bps. |
| 9600 | Specifies a fax transmission speed of 9600 bps. |
| 14400 | Specifies a fax transmission speed of 14,400 bps. |
| disable | Disables fax relay transmission capability. |
| voice | Specifies the highest possible transmission speed allowed by voice rate. |
The default value for this command is voice.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the fax-rate command to specify the fax transmission rate to the specified dial peer.
The values for this command apply only to the fax transmission speed and do not affect the quality of the fax itself. The higher values provide a faster transmission speed but monopolize a significantly larger portion of the available bandwidth. Slower transmission speeds use less bandwidth.
If the fax-rate command is set above the codec rate in the same dial peer, the data sent over the network for fax transmission will be above the bandwidth reserved for RVSP. Because more network bandwidth will be monopolized by the fax transmission, we do not recommend setting the fax-rate value higher than the codec value. If the fax-rate value is set lower than the codec value, faxes will take longer to transmit but will use less bandwidth.
This command is only applicable to VoIP peers.
The following example configures a facsimile rate of 9600 bps for faxes sent to a dial peer:
dial-peer voice 10 voip fax-rate 9600
You can use the master index or search online to find documentation of related commands.
codec
| number | Integer, expressed in equipment impairment factor units, specifying the ICPIF value. Valid entries are 0 to 55. |
The default value for this command is 30.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the icpif command to specify the maximum acceptable impairment factor for the voice calls sent by the selected dial peer.
This command is applicable only to VoIP peers.
The following example disables the icpif command:
dial-peer voice 10 voip icpif 0
| 600c | Specifies 600 Ohms complex. |
| 600r | Specifies 600 Ohms real. |
| 900c | Specifies 900 Ohms complex. |
| complex1 | Specifies Complex 1. |
| complex2 | Specifies Complex 2. |
The default value for this command is 600.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the impedance command to specify the terminating impedance of an FXO voice-port interface. The impedance value selected needs to match the specifications from the specific 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 normally either 600r or 900c.
If the impedance is set incorrectly (if there is an impedance mismatch), there will be a significant amount of echo 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 will change 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.
This command is applicable to FXS, FXO, and E&M voice ports.
The following example configures an FXO voice port for a terminating impedance of 600 Ohms:
impedance 600r
| value | Specifies, in decibels, the amount of gain to be inserted at the receiver side of the interface. Acceptable value is any integer from -6 to 14. |
The default value for FXO, FXS, and E&M ports is 0.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
A system-wide loss plan must be implemented using both input gain and output attenuation commands. Other equipment (including PBXs) in the system must be taken into account when creating a loss plan. This default value for this command assumes that a standard transmission loss plan is in effect, meaning that normally, there must be -6dB attenuation between phones. Connections are implemented to provide -6dB of attenuation when the input gain and output attenuation commands are configured with the default value of 0.
Please note that you can't increase the gain of a signal going out into the PSTN, but you can decrease it. Therefore, if the voice level is too high, you can decrease the volume by either decreasing the input gain value or by increasing the output attenuation.
You can increase the gain of a signal coming in to the router. If the voice level is too low, you can increase the input gain.
The following example configures a 3-decibel gain to be inserted at the receiver side of the interface:
input gain 3
You can use the master index or search online to find documentation of related commands.
output attenuation
| number | Integer specifying the IP precedence value. Valid entries are 0 to 7. A value of 0 means that no precedence (priority) has been set. |
The default value for this command is 0.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the ip precedence 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 QoS for voice packets need to have a higher priority than other IP packets. The ip precedence command should also be used if RSVP is not enabled, and the user would like to give voice packets a higher priority over other IP data traffic.
This command is applicable to VoIP peers.
The following example sets the ip precedence at 5:
dial-peer voice 10 voip ip precedence 5
This command has no arguments or keywords.
The default value for this command is disabled.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the ip udp checksum 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 ip udp checksum to prevent bad voice packets forwarded to the DSP.
This command is applicable to VoIP peers.
The following example calculates the UDP checksum for voice packets transmitted by this dial peer:
dial-peer voice 10 voip ip udp checksum
| number | Specifies the on-hold music threshold in decibels. Valid entries are any integer from -70 to -30. |
The default value for this command is -38.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the music-threshold command to specify the decibel level of music played when calls are put on hold. This command tells the firmware to pass steady data above the specified level. It only affects the operation of VAD when receiving voice.
If the value for this command is set too high, VAD will interpret music-on-hold as silence, and the remote end will not hear the music. If the value for this command is set too low, VAD will compress and pass silence when the background is noisy, creating unnecessary voice traffic.
The following sets the decibel threshold for the music played when calls are put on hold to -35:
voice port 1/0/0 music-threshold -35
This command has no arguments or keywords.
The default for this command is enabled for all voice-port types.
Voice-port configuration.
This command first appeared in Cisco IOS Release 11.3(1)T.
This command is associated with the echo canceller operation. The echo-cancel enable command must be enabled for the non-linear command to take effect. Use the non-linear command to shut off any signal if no near-end speech is detected.
Enabling the non-linear command normally improves performance, although some users might perceive truncation of consonants at the end of sentences when this command is enabled.
This feature is also generally known as residual echo suppression.
The following example enables non-linear call processing:
voice-port 1/0/0 non-linear
You can use the master index or search online to find documentation of related commands.
echo-cancel enable
| extension-number | Digit(s) defining an extension number for a particular dial peer. |
| expanded-number | Digit(s) defining the expanded telephone number or destination pattern for the extension number listed. |
Global configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the num-exp global configuration command to define how to expand a particular set of numbers (for example, an extension number) into a particular destination pattern. With this command, you can map specific extensions and expanded numbers together by explicitly defining each number, or you can define extensions and expanded numbers using variables. You can also use this command to convert seven-digit numbers to numbers containing less than seven digits.
Use a period (.) as a variable or wild card, representing a single number. Use a separate period for each number you want to represent with a wildcard--meaning that if you want to replace four numbers in an extension with wildcards, type in four periods.
The following example expands the extension number 65541 to be expanded to 14085665541:
num-exp 65541 14085665541
The following example shows how to expand all five-digit extensions beginning with 6 to append the following numbers at the beginning of the extension number 1408566:
num-exp 6.... 1408566....
| 2-wire | Specifies a two-wire E&M cabling scheme. |
| 4-wire | Specifies a four-wire E&M cabling scheme. |
2-wire operation
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
The operation command will only affect voice traffic. Signaling is independent of two-wire versus four-wire settings. If the wrong cable scheme is specified, the user might get voice traffic in only one direction.
Configuring the operation command on a voice port changes the operation of both voice ports on a VPM card. The voice port must be shut down and then opened again for the new value to take effect.
This command is not applicable to FXS or FXO interfaces because those are, by definition, two-wire interfaces.
The following example specifies that an E&M port uses a four-wire cabling scheme:
voice-port 1/0/0 operation 4-wire
| value | Specifies, in decibels, the amount of attenuation at the transmit side of the interface. Acceptable value is any integer from 0 to 14. The default value for FXO, FXS, and E&M ports is 0. |
The default value for FXO, FXS, and E&M ports is 0.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
A system-wide loss plan must be implemented using both input gain and output attenuation commands. Other equipment (including PBXs) in the system must be taken into account when creating a loss plan. This default value for this command assumes that a standard transmission loss plan is in effect, meaning that normally, there must be -6 dB attenuation between phones. Connections are implemented to provide -6 dB of attenuation when the input gain and output attenuation commands are configured with the default value of 0.
Please note that you can't increase the gain of a signal going out into the PSTN, but you can decrease it. Therefore, if the voice level is too high, you can decrease the volume by either decreasing the input gain value or by increasing the output attenuation.
The following example configures a three-decibel gain to be inserted at the transmit side of the interface:
voice-port 1/0/0 output attenuation 3
You can use the master index or search online to find documentation of related commands.
input gain
| slot-number/ | Specifies the slot number in the Cisco router where the voice interface card is installed. Valid entries are from 0 to 3, depending on the slot where it has been installed. |
| subunit-number/ | Specifies the subunit on the voice interface card where the voice port is located. Valid entries are 0 or 1. |
| port | Specifies the voice port. Valid entries are 0 or 1. |
No port is configured.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the port configuration command to associate the designated voice port with the selected dial peer.
This command is used for calls incoming from a telephony interface to select an incoming dial peer and for calls coming from the VoIP network to match a port with the selected outgoing dial peer.
This command is applicable only to POTS peers.
The following example associates a dial peer with voice port 1, which is located on subunit 0, and accessed through port 0:
dial-peer voice 10 pots port 1/0/0
| string | Integers representing the prefix of the telephone number associated with the specified dial peer. Valid numbers are 0 through 9, and a comma (,). Use a comma to include a pause in the prefix. |
The default for this command is a null string.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the prefix command to specify a prefix for a specific dial peer. When an outgoing call is initiated to this dial peer, the prefix string value is sent to the telephony interface first, before the telephone number associated with the dial peer.
If you want to configure different prefixes for dialed numbers on the same interface, you need to configure different dial peers.
This command is applicable only to POTS peers.
The following example specifies a prefix of "9" and then a pause:
dial-peer voice 10 pots prefix 9,
You can use the master index or search online to find documentation of related commands.
answer-address
destination-pattern
| best-effort | Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation. |
| controlled-load | Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to assure that preferential service is received even when the bandwidth is overloaded. |
| guaranteed-delay | Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queuing if the bandwidth reserved is not exceeded. |
The default value for this command is best-effort. The no form of this command restores the default value.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the req-qos command to request a specific quality of service to be used in reaching a dial peer. Like acc-qos, when you issue this command, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation Protocol (RSVP) to request quality of service guarantees from the network.
This command is applicable only to VoIP peers.
The following example configures guaranteed-delay as the desired (requested) quality of service to a dial peer:
dial-peer voice 10 voip req-qos guaranteed-delay
You can use the master index or search online to find documentation of related commands.
acc-qos
| number | Specifies the ring frequency (Hertz) used in the FXS interface. Valid entries are 25 and 50. |
The default value for this command is 25 Hertz.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the ring frequency command to select a specific ring frequency for an FXS voice port. Use the no form of this command to reset the default value for this command, which is 25 Hertz. The ring frequency you select must match the connected equipment. If set incorrectly, the attached phone might not ring or might buzz. In addition, the ring frequency is usually country-dependent and you should take into account the appropriate ring frequency for your area before configuring this command.
This command does not affect ringback, which is the ringing a user hears when placing a remote call.
The following example configures the ring frequency for 50 Hertz:
voice-port 1/0/0 ring frequency 50
You can use the master index or search online to find documentation of related commands.
ring number
| number | Specifies the number of rings detected before answering the call. Valid entries are numbers from 1 to 10. |
The default value is one ring.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the ring number command to set the maximum number of rings to be detected before answering a call over an FXO voice port. Use the no form of this command to reset the default value, which is 1 ring.
Normally, this command should be set to the default so that incoming calls are answered quickly. If you have other equipment available on the line to answer incoming calls, you might want to set the value higher to give the equipment sufficient time to respond. In that case, the FXO interface would answer if the equipment on line did not answer the incoming call in the configured number of rings.
This command is not applicable to FXS or E&M interfaces because they do not receive ringing to receive a call.
The following example sets five rings as the maximum number of rings to be detected before closing a connection over this voice port:
voice port 1/0/0 ring number 5
You can use the master index or search online to find documentation of related commands.
ring frequency
| cisco | Specifies Cisco Session Protocol. |
The default value for this command is cisco.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
For this release, Cisco Session Protocol (cisco) is the only applicable session protocol. This command is applicable only to VoIP peers.
The following example selects Cisco Session Protocol as the session protocol:
dial-peer voice 10 voip session protocol cisco
You can use the master index or search online to find documentation of related commands.
session target
| ipv4:destination-address | IP address of the dial peer. |
| dns:host-name | Indicates that the domain name server will be used to resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device.
(Optional) You can use one of the following three wildcards with this keyword when defining the session target for VoIP peers: · $s$.--Indicates that the source destination pattern will be used as part of the domain name. · $d$.--Indicates that the destination number will be used as part of the domain name. · $u$.--Indicates that the unmatched portion of the destination pattern (such as a defined extension number) will be used as part of the domain name. |
| loopback:rtp | Indicates that all voice data will be looped-back to the originating source. This is applicable for VoIP peers. |
| loopback:compressed | Indicates that all voice data will be looped-back in compressed mode to the originating source. This is applicable for POTS peers. |
| loopback:uncompressed | Indicates that all voice data will be looped-back in uncompressed mode to the originating source. This is applicable for POTS peers. |
The default for this command is enabled with no IP address or domain name defined.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origination and the loopback type selected.
The session target dns command can be used with or without the specified wildcards. Using the optional wildcards can reduce the number of VoIP dial peer session targets you need to configure if you have groups of numbers associated with a particular router.
The following example configures a session target using dns for a host, "voice_router," in the domain "cisco.com":
dial-peer voice 10 voip session target dns:voice_router.cisco.com
The following example configures a session target using dns, with the optional $u$. wildcard. In this example, the destination pattern has been configured to allow for any four-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. indicates that the router will use the unmatched portion of the dialed number--in this case, the four-digit extension, to identify the dial peer. As in the previous example, the domain is "cisco.com."
dial-peer voice 10 voip destination-pattern 1310222.... session target dns:$u$.cisco.com
The following example configures a session target using dns, with the optional $d$. wildcard. In this example, the destination pattern has been configured for 13102221111. The optional wildcard $d$. indicates that the router will use the destination pattern to identify the dial peer in the "cisco.com" domain.
dial-peer voice 10 voip destination-pattern 13102221111 session target dns:$d$.cisco.com
You can use the master index or search online to find documentation of related commands.
destination-pattern
session protocol
To show the active call table, use the show call active voice privileged EXEC command.
show call active voiceThis command contains no arguments or keywords.
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show call active voice privileged EXEC command to display the contents of the active call table, which shows all of the calls currently connected through the router.
For each call, there are two call legs, usually a POTS call leg and a VoIP call leg. A call leg is a discrete segment of a call connection that lies between two points in the connection. Each dial peer creates a call leg, as shown in Figure 3-1.
These two call legs are associated by the connection ID. The connection ID is global across the voice network, so that you can associate two call legs on one router with two call legs on another router, thereby providing an end-to-end view of a call.
The following is sample output from the show call active voice command:
sloth#show call active voiceGENERIC: SetupTime=21072 Index=0 PeerAddress= PeerSubAddress= PeerId=0 PeerIfIndex=0 LogicalIfIndex=0 ConnectTime=0 CallState=3 CallOrigin=2 ChargedUnits=0 InfoType=0 TransmitPackets=375413 TransmitBytes=7508260 ReceivePackets=377734 ReceiveBytes=7554680VOIP: ConnectionId[0x19BDF910 0xAF500007 0x0 0x58ED0] RemoteIPAddress=17635075 RemoteUDPPort=16394 RoundTripDelay=0 SelectedQoS=0 SessionProtocol=1 SessionTarget= OnTimeRvPlayout=0 GapFillWithSilence=0 GapFillWithPrediction=600 GapFillWithInterpolation=0 GapFillWithRedundancy=0 HiWaterPlayoutDelay=110 LoWaterPlayoutDelay=64 ReceiveDelay=94 VADEnable=0 CoderTypeRate=0GENERIC: SetupTime=21072 Index=1 PeerAddress=+14085271001 PeerSubAddress= PeerId=0 PeerIfIndex=0 LogicalIfIndex=5 ConnectTime=21115 CallState=4 CallOrigin=1 ChargedUnits=0 InfoType=1 TransmitPackets=377915 TransmitBytes=7558300 ReceivePackets=375594 ReceiveBytes=7511880TELE: ConnectionId=[0x19BDF910 0xAF500007 0x0 0x58ED0] TxDuration=16640 VoiceTxDuration=16640 FaxTxDuration=0 CoderTypeRate=0 NoiseLevel=0 ACOMLevel=4 OutSignalLevel=-440 InSignalLevel=-440 InfoActivity=2 ERLLevel=227 SessionTarget=
Table 3-2 provides an alphabetical listing of the fields in this output and a description of each field.
| Field | Description |
|---|---|
| ACOM Level | Current ACOM level for the call. This value is sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call. |
| CallOrigin | Call origin; answer versus originate. |
| CallState | Current state of the call. |
| CoderTypeRate | Negotiated coder transmit rate of voice/fax compression during the call. |
| ConnectionId | Global call identifier of a gateway call. |
| ConnectTime | Time at which the call was connected. |
| Dial-Peer | Tag of the dial peer transmitting this call. |
| ERLLevel | Current Echo Return Loss (ERL) level for this call. |
| FaxTxDuration | Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value. |
| GapFillWith Silence | Duration of voice signal replaced with silence because voice data was lost or not received on time for this call. |
| GapFillWithPrediction | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call. An example of such pullout is frame-eraser or frame-concealment strategies in G.729 and G.723.1 compression algorithms. |
| GapFillWithInterpolation | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because voice data was lost or not received on time from voice gateway for this call. |
| GapFillWith Redundancy | Duration of voice signal played out with signal synthesized from redundancy parameters available because voice data was lost or not received on time from voice gateway for this call. |
| HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during this call. |
| Index | Dial peer identification number. |
| InfoActivity | Active information transfer activity state for this call. |
| InfoType | Information type for this call. |
| InSignalLevel | Active input signal level from the telephony interface used by this call. |
| LogicalIfIndex | Index number of the logical interface for this call. |
| LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during the call. |
| NoiseLevel | Active noise level for the call. |
| OnTimeRvPlayout | Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
| OutSignalLevel | Active output signal level to telephony interface used by this call. |
| PeerAddress | Destination pattern associated with this peer. |
| PeerId | ID value of the peer table entry to which this call was made. |
| PeerIfIndex | Voice-port index number for this peer. |
| PeerSubaddress | Subaddress to which this call is connected. |
| ReceiveBytes | Number of bytes received by the peer during this call. |
| ReceiveDelay | Average Playout FIFO Delay plus the decoder delay during the voice call. |
| ReceivePackets | Number of packets received by this peer during this call. |
| RemoteIPAddress | Remote system IP address for the VoIP call. |
| RemoteUDPPort | Remote system UDP listener port to which voice packets are transmitted. |
| RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone during the call. |
| SelectedQoS | Selected RSVP quality of service (QoS) for the call. |
| SessionProtocol | Session protocol used for an Internet call between the local and remote router via the IP backbone. |
| SessionTarget | Session target of the peer used for the call. |
| SetupTime | Value of the System UpTime when the call associated with this entry was started. |
| TransmitBytes | Number of bytes transmitted from this peer during the call. |
| TransmitPackets | Number of packets transmitted from this peer during the call. |
| TxDuration | Duration of transmit path open from this peer to the voice gateway for the call. |
| VADEnable | Whether or not voice activation detection (VAD) was enabled for this call. |
| VoiceTxDuration | Duration of voice transmission from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value. |
You can use the master index or search online to find documentation of related commands.
show call history voice
show dial-peer voice
show num-exp
show voice port
To display the call history table, use the show call history voice privileged EXEC command.
show call history voice last number| last number | Displays the last calls connected, where the number of calls displayed is defined by the argument number. Valid entries for the argument number is any number from 1 to 2147483647. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show call history voice privileged EXEC command to display the call history table. The call history table contains a listing of all calls connected through this router in descending time order since Voice over IP was enabled. You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the keyword last, and define the number of calls to be displayed with the argument number.
The following is sample output from the show call history voice command:
sloth#show call history voiceGENERIC: SetupTime=20405 Index=0 PeerAddress= PeerSubAddress= PeerId=0 PeerIfIndex=0 LogicalIfIndex=0 DisconnectCause=NORMAL DisconnectText= ConnectTime=0 DisconectTime=20595 CallOrigin=2 ChargedUnits=0 InfoType=0 TransmitPackets=0 TransmitBytes=0 ReceivePackets=0 ReceiveBytes=0 VOIP: ConnectionId[0x19BDF910 0xAF500006 0x0 0x56590] RemoteIPAddress=17635075 RemoteUDPPort=16392 RoundTripDelay=0 SelectedQoS=0 SessionProtocol=1 SessionTarget= OnTimeRvPlayout=0 GapFillWithSilence=0 GapFillWithPrediction=0 GapFillWithInterpolation=0 GapFillWithRedundancy=0 HiWaterPlayoutDelay=0 LoWaterPlayoutDelay=0 ReceiveDelay=0 VADEnable=0 CoderTypeRate=0 TELE: ConnectionId=[0x19BDF910 0xAF500006 0x0 0x56590] TxDuration=3030 VoiceTxDuration=2700 FaxTxDuration=0 CoderTypeRate=0 NoiseLevel=0 ACOMLevel=0 SessionTarget=
Table 3-3 provides an alphabetical listing of the fields in this output and a description of each field.
| Field | Description |
|---|---|
| ACOMLevel | Average ACOM level for this call. This value is sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call. |
| CallOrigin | Call origin; answer versus originate. |
| CoderTypeRate | Negotiated coder rate. This value specifies the transmit rate of voice/fax compression to its associated call leg for the call. |
| ConnectionID | Global call identifier for the gateway call. |
| ConnectTime | Time the call was connected. |
| DisconnectCause | Description explaining why the call was disconnected. |
| DisconnectText | Descriptive text explaining the disconnect reason. |
| DisconnectTime | Time the call was disconnected. |
| FaxDuration | Duration of fax transmitted from this peer to the voice gateway for this call. You can derive the Fax Utilization Rate by dividing this value by the TxDuration value. |
| GapFillWithSilence | Duration of voice signal replaced with silence because the voice data was lost or not received on time for this call. |
| GapFillWithPrediction | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call. |
| GapFillWithInterpolation | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call. |
| GapFillWithRedundancy | Duration of voice signal played out with signal synthesized from redundancy parameters available because the voice data was lost or not received on time from the voice gateway for this call. |
| HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during the voice call. |
| Index | Index number identifying the voice-peer for this call. |
| InfoType | Information type for this call. |
| LogicalIfIndex | Index of the logical voice port for this call. |
| LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during the voice call. |
| NoiseLevel | Average noise level for this call. |
| OnTimeRvPlayout | Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
| PeerAddress | Destination pattern or number to which this call is connected. |
| PeerId | ID value of the peer entry table to which this call was made. |
| PeerIfIndex | Index number of the logical interface through which this call was made. For ISDN media, this would be the index number of the B channel used for the call. |
| PeerSubAddress | Subaddress to which this call is connected. |
| ReceiveBytes | Number of bytes received by the peer during this call. |
| ReceiveDelay | Average Playout FIFO Delay plus the decoder delay during the voice call. |
| ReceivePackets | Number of packets received by this peer during the call. |
| RemoteIPAddress | Remote system IP address for the call. |
| RemoteUDPPort | Remote system UDP listener port to which voice packets for this call are transmitted. |
| RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone for this call. |
| SelectedQoS | Selected RSVP quality of service for the call. |
| Session Protocol | Session protocol to be used for an Internet call between the local and remote router via the IP backbone. |
| Session Target | Session target of the peer used for the call. |
| SetUpTime | Value of the System UpTime when the call associated with this entry was started. |
| TransmitBytes | Number of bytes transmitted by this peer during the call. |
| TransmitPackets | Number of packets transmitted by this peer during the call. |
| TxDuration | Duration of the transmit path open from this peer to the voice gateway for the call. |
| VADEnable | Whether or not voice activation detection (VAD) was enabled for this call. |
| VoiceTxDuration | Duration of voice transmitted from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration by the TxDuration value. |
You can use the master index or search online to find documentation of related commands.
show call active voice
show dial-peer voice
show num-exp
show voice port
| number | Displays configuration for the dial peer identified by the argument number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show dial-peer voice privileged EXEC command to display the configuration for all VoIP and POTS dial peers configured for the router. To show configuration information for only one specific dial peer, use the argument number to identify the dial peer.
The following is sample output from the show dial-peer voice command for a POTS dial peer:
sloth# show dial-peer voice 1
VoiceEncapPeer1
tag = 1, dest-pat = `+14085291000',
answer-address = `',
group = 0, Admin state is up, Operation state is down
Permission is Both,
type = pots, prefix = `',
session-target = `', voice-port =
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
Last Setup Time = 0
The following is sample output from the show dial-peer voice command for a VoIP dial peer:
sloth# show dial-peer voice 10
VoiceOverIpPeer10
tag = 10, dest-pat = `',
incall-number = `+14087',
group = 0, Admin state is up, Operation state is down
Permission is Answer,
type = voip, session-target = `',
sess-proto = cisco, req-qos = bestEffort,
acc-qos = bestEffort,
fax-rate = voice, codec = g729r8,
Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled,
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
Last Setup Time = 0
Table 3-4 explains the fields contained in both of these examples.
| Field | Description |
|---|---|
| AcceptedCalls | Number of calls from this peer accepted since system startup. |
| acc-qos | Lowest acceptable quality of service configured for calls for this peer. |
| Admin state | Administrative state of this peer. |
| Charged Units | Total number of charging units applying to this peer since system startup. |
| codec | Default voice coder rate of speech for this peer. |
| Connect Time | Accumulated connect time to the peer since system startup for both incoming and outgoing calls. |
| dest-pat | Destination pattern (telephone number) for this peer. |
| Expect factor | User-requested Expectation Factor of voice quality for calls via this peer. |
| fax-rate | Fax transmission rate configured for this peer. |
| Failed Calls | Number of failed call attempts to this peer since system startup. |
| group | Group number associated with this peer. |
| ICPIF | Configured Calculated Planning Impairment Factor (ICPIF) value for calls sent by a dial peer. |
| incall-number | Full E.164 telephone number to be used to identify the dial peer. |
| Last Disconnect Cause | Encoded network cause associated with the last call. This value will be updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface. |
| Last Disconnect Text | ASCII text describing the reason for the last call termination. |
| Last Setup Time | Value of the System Up Time when the last call to this peer was started. |
| Operation state | Operational state of this peer. |
| Permission | Configured permission level for this peer. |
| Poor QOV Trap | Whether Poor Quality of Voice trap messages have been enabled or disabled. |
| Refused Calls | Number of calls from this peer refused since system startup. |
| req-qos | Configured requested quality of service for calls for this dial peer. |
| session-target | Session target of this peer. |
| sess-proto | Session protocol to be used for Internet calls between local and remote router via the IP backbone. |
| Successful Calls | Number of completed calls to this peer. |
| tag | Unique dial peer ID number. |
| VAD | Whether or not voice activation detection (VAD) is enabled for this dial peer. |
You can use the master index or search online to find documentation of related commands.
show call active voice
show call-history voice
show num-exp
show voice port
| slot-number/ | Specifies the slot number in the Cisco router where the voice network module is installed. Valid entries are from 0 to 3, depending on the voice interface card you have installed. |
| subunit-number/ | Specifies the subunit on the voice network module where the voice port is located. Valid entries are 0 or 1. |
| port | Specifies the voice port. Valid entries are 0 or 1. |
| dial string | Specifies a particular destination pattern (telephone number). |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Occasionally, an incoming call cannot be matched to a dial peer in the dial peer database. One reason this might occur is that the specified destination cannot be reached via the voice interface through which the incoming call came. Use the show dialplan incall number command as a troubleshooting method to resolve the call destination by pairing voice ports and telephone numbers together until there is a match.
The following example tests whether the telephone extension 57681 can be reached through voice port 1/0/1:
show dialplan incall 1/0/1 number 57681
You can use the master index or search online to find documentation of related commands.
show dialplan number
| dial string | Specifies a particular destination pattern (telephone number). |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
The following example displays the dial peer associated with the destination pattern of 54567:
show dialplan number 54567
You can use the master index or search online to find documentation of related commands.
show dialplan incall number
To show the number expansions configured, use the show num-exp privileged EXEC command.
show num-exp [dialed- number]| dialed-number | Displays number expansion for the specified dialed number. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show num-exp privileged EXEC command to display all of the number expansions configured for this router. To display number expansion for only one number, specify that number by using the dialed-number argument.
The following is sample output from the show num-exp command:
sloth# show num-exp
Dest Digit Pattern = '0...' Translation = '+14085270...'
Dest Digit Pattern = '1...' Translation = '+14085271...'
Dest Digit Pattern = '3..' Translation = '+140852703..'
Dest Digit Pattern = '4..' Translation = '+140852804..'
Dest Digit Pattern = '5..' Translation = '+140852805..'
Dest Digit Pattern = '6....' Translation = '+1408526....'
Dest Digit Pattern = '7....' Translation = '+1408527....'
Dest Digit Pattern = '8...' Translation = '+14085288...'
Table 3-5 explains the fields in the sample output.
| Field | Description |
|---|---|
| Dest Digit Pattern | Index number identifying the destination telephone number digit pattern. |
| Translation | Expanded destination telephone number digit pattern. |
You can use the master index or search online to find documentation of related commands.
show call active voice
show call history voice
show dial-peer voice
show voice port
| slot-number/ | Specifies the slot number in the Cisco router where the voice interface card is installed. Valid entries are from 0 to 3, depending on the slot where it has been installed. |
| subunit-number/ | Specifies the subunit on the voice interface card where the voice port is located. Valid entries are 0 or 1. |
| port | Specifies the voice port. Valid entries are 0 or 1. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show voice port privileged EXEC command to display configuration and voice interface card-specific information about a specific port.
The following is sample output from the show voice port command for an E&M voice port:
sloth#show voice port 1/0/0E&M Slot is 1, Sub-unit is 0, Port is 0Type of VoicePort is E&MOperation State is unknownAdministrative State is unknownThe Interface Down Failure Cause is 0Alias is NULLNoise Regeneration is disabledNon Linear Processing is disabledMusic On Hold Threshold is Set to 0 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is disabledEcho Cancel Coverage is set to 16msConnection Mode is NormalConnection Number isInitial Time Out is set to 0 sInterdigit Time Out is set to 0 sAnalog Info Follows:Region Tone is set for northamericaCurrently processing noneMaintenance Mode Set to None (not in mtc mode)Number of signaling protocol errors are 0Voice card specific Info Follows:Signal Type is wink-startOperation Type is 2-wireImpedance is set to 600r OhmE&M Type is unknownDial Type is dtmfIn Seizure is inactiveOut Seizure is inactiveDigit Duration Timing is set to 0 msInterDigit Duration Timing is set to 0 msPulse Rate Timing is set to 0 pulses/secondInterDigit Pulse Duration Timing is set to 0 msClear Wait Duration Timing is set to 0 msWink Wait Duration Timing is set to 0 msWink Duration Timing is set to 0 msDelay Start Timing is set to 0 msDelay Duration Timing is set to 0 ms
The following is sample output from the show voice port command for an FXS voice port:
sloth# show voice port 1/0/0 Foreign Exchange Station 1/0/0 Slot is 1, Sub-unit is 0, Port is 0 Type of VoicePort is FXS Operation State is DORMANT Administrative State is UP The Interface Down Failure Cause is 0 Alias is NULL Noise Regeneration is enabled Non Linear Processing is enabled Music On Hold Threshold is Set to 0 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 16ms Connection Mode is Normal Connection Number is Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Analog Info Follows: Region Tone is set for northamerica Currently processing none Maintenance Mode Set to None (not in mtc mode) Number of signaling protocol errors are 0 Voice card specific Info Follows: Signal Type is loopStart Ring Frequency is 25 Hz Hook Status is On Hook Ring Active Status is inactive Ring Ground Status is inactive Tip Ground Status is inactive Digit Duration Timing is set to 100 ms InterDigit Duration Timing is set to 100 ms Hook Flash Duration Timing is set to 600 ms
Table 3-6 explains the fields in the sample output.
| Field | Description |
|---|---|
| Administrative State | Administrative state of the voice port. |
| Alias | User-supplied alias for this voice port. |
| Clear Wait Duration Timing | Time of inactive seizure signal to declare call cleared. |
| Connection Mode | Connection mode of the interface. |
| Connection Number | Full E.164 telephone number used to establish a connection with the trunk or PLAR mode. |
| Currently Processing | Type of call currently being processed: none, voice, or fax. |
| Delay Duration Timing | Maximum delay signal duration for delay dial signaling. |
| Delay Start Timing | Timing of generation of delayed start signal from detection of incoming seizure. |
| Dial Type | Out-dialing type of the voice port. |
| Digit Duration Timing | DTMF Digit duration in milliseconds. |
| E&M Type | Type of E&M interface. |
| Echo Cancel Coverage | Echo Cancel Coverage for this port. |
| Echo Cancellation | Whether or not echo cancellation is enabled for this port. |
| Hook Flash Duration Timing | Maximum length of hook flash signal. |
| Hook Status | Hook status of the FXO/FXS interface. |
| Impedance | Configured terminating impedance for the E&M interface. |
| In Gain | Amount of gain inserted at the receiver side of the interface. |
| In Seizure | Incoming seizure state of the E&M interface. |
| Initial Time Out | Amount of time the system waits for an initial input digit from the caller. |
| InterDigit Duration Timing | DTMF interdigit duration in milliseconds. |
| InterDigit Pulse Duration Timing | Pulse dialing interdigit timing in milliseconds. |
| Interdigit Time Out | Amount of time the system waits for a subsequent input digit from the caller. |
| Maintenance Mode | Maintenance mode of the voice-port. |
| Music On Hold Threshold | Configured Music-On-Hold Threshold value for this interface. |
| Noise Regeneration | Whether or not background noise should be played to fill silent gaps if VAD is activated. |
| Number of signaling protocol errors | Number of signaling protocol errors. |
| Non-Linear Processing | Whether or not Non-Linear Processing is enabled for this port. |
| Operations State | Operation state of the port. |
| Operation Type | Operation of the E&M signal: two-wire or four-wire. |
| Out Attenuation | Amount of attenuation inserted at the transmit side of the interface. |
| Out Seizure | Outgoing seizure state of the E&M interface. |
| Port | Port number for this interface associated with the voice interface card. |
| Pulse Rate Timing | Pulse dialing rate in pulses per second (pps). |
| Regional Tone | Configured regional tone for this interface. |
| Ring Active Status | Ring active indication. |
| Ring Frequency | Configured ring frequency for this interface. |
| Ring Ground Status | Ring ground indication. |
| Signal Type | Type of signaling for a voice port: loop-start, ground-start, wink-start, immediate, and delay-dial. |
| Slot | Slot used in the voice interface card for this port. |
| Sub-unit | Subunit used in the voice interface card for this port. |
| Tip Ground Status | Tip ground indication. |
| Type of VoicePort | Type of voice port: FXO, FXS, and E&M. |
| The Interface Down Failure Cause | Text string describing why the interface is down, |
| Wink Duration Timing | Maximum wink duration for wink start signaling. |
| Wink Wait Duration Timing | Maximum wink wait duration for wink start signaling. |
You can use the master index or search online to find documentation of related commands.
show call active voice
show call history voice
show dial-peer voice
show num-exp
This command has no arguments or keywords.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
When a dial peer is shut down, you cannot initiate calls to that peer. This command is applicable to both VoIP and POTS peers.
The following example changes the administrative state of voice telephony dial peer 10 to down:
configure terminal dial-peer voice 10 pots shutdown
This command has no arguments or keywords.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
When you enter the shutdown command, all ports on the voice interface card are disabled. When you enter the no shutdown command, all ports on the voice interface card are enabled. A telephone connected to an interface will hear dead silence when a port is shut down.
The following example takes voice port 1/1/0 offline:
configure terminal voice port 1/1/0 shutdown
| loop-start | Specifies Loop Start signaling. Used for FXO and FXS interfaces. With Loop Start signalling only one side of a connection can hang up. This is the default setting for FXO and FXS voice ports. |
| ground-start | Specifies Ground Start signaling. Used for FXO and FXS interfaces. Ground Start allows both sides of a connection to place a call and to hang up. |
| wink-start | Indicates that the calling side seizes the line by going off-hook on its E lead then waits for a short off-hook "wink" indication on its M lead from the called side before sending address information as DTMF digits. Used for E&M tie trunk interfaces. This is the default setting for E&M voice ports. |
| immediate | Indicates that the calling side seizes the line by going off-hook on its E lead and sends address information as DTMF digits. Used for E&M tie trunk interfaces. |
| delay-dial | Indicates that the calling side seizes the line by going off-hook on its E lead. After a timing interval, the calling side looks at the supervision from the called side. If the supervision is on-hook, the calling side starts sending information as DTMF digits; otherwise, the calling side waits until the called side goes on-hook and then starts sending address information. Used for E&M tie trunk interfaces. |
The default value is loop-start for FXO and FXS interfaces. The default value is wink-start for E&M interfaces.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Configuring the signal command for an FXS or FXO voice port will change the signal value for both voice ports on a VPM card.
Configuring this command for an E&M voice port will change only the signal value for the selected voice port. In either case, the voice port must be shut down and then activated before the configured values will take effect.
Some PBXs will miss initial digits if the E&M voice port is configured for Immediate signaling. If this occurs, use Delay-Dial signaling instead. Some non-Cisco devices have a limited number of DTMF receivers. This type of equipment must delay the calling side until a DTMF receiver is available.
The following example configures Ground Start signaling, which means that both sides of a connection can place a call and hang up, as the signaling type for a voice port:
configure terminal voice-port 1/1/1 signal ground-start
This command has no arguments or keywords.
Disabled
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the snmp enable peer-trap poor qov command to generate poor quality of voice notifications for applicable calls associated with this dial peer. If you have an SNMP manager that will use SNMP messages when voice quality drops, you might want to enable this command. Otherwise, you should disable this command to reduce unnecessary network traffic.
This command is applicable only to VoIP peers.
The following example enables poor quality of voice notifications for calls associated with VoIP dial peer 10:
dial-peer voice 10 voip snmp enable peer-trap poor-qov
You can use the master index or search online to find documentation of related commands.
snmp-server enable trap voice poor-qov
snmp trap link-status
To enable the router to send SNMP traps, use the snmp-server enable traps global configuration command. Use the no form of this command to disable SNMP traps.
snmp-server enable traps [trap-type] [trap-option]| trap-type | (Optional) Type of trap to enable. If no type is specified, all traps are sent (including the envmon and repeater traps). The trap type can be one of the following keywords:
· bgp--Sends Border Gateway Protocol (BGP) state change traps. · config--Sends configuration traps. · entity--Sends Entity MIB modification traps. · envmon--Sends Cisco enterprise-specific environmental monitor traps when an environmental threshold is exceeded. When the envmon keyword is used, you can specify a trap-option value. · frame-relay--Sends Frame Relay traps. · isdn--Sends Integrated Services Digital Network (ISDN) traps. When the isdn keyword is used on Cisco 1600 series routers, you can specify a trap-option value. · repeater--Sends Ethernet hub repeater traps. When the repeater keyword is selected, you can specify a trap-option value. · rtr--Sends response time reporter (RTR) traps. · snmp--Sends Simple Network Management Protocol (SNMP) traps. When the snmp keyword is used, you can specify a trap-option value. · syslog--Sends error message traps (Cisco Syslog MIB). Specify the level of messages to be sent with the logging history level command. · voice--Sends SNMP poor quality of voice traps, when used with the qov trap-option. |
| trap-option | (Optional) When the envmon keyword is used, you can enable a specific environmental trap type, or accept all trap types from the environmental monitor system. If no option is specified, all environmental types are enabled. The option can be one or more of the following keywords: voltage, shutdown, supply, fan, and temperature.
When the isdn keyword is used on Cisco 1600 series routers, you can specify the call-information keyword to enable an SNMP ISDN call information trap for the ISDN MIB subsystem, or you can specify the isdnu-interface keyword to enable an SNMP ISDN U interface trap for the ISDN U interface MIB subsystem. When the repeater keyword is used, you can specify the repeater option. If no option is specified, all repeater types are enabled. The option can be one or more of the following keywords: · health--Enables IETF Repeater Hub MIB (RFC 1516) health trap. · reset--Enables IETF Repeater Hub MIB (RFC 1516) reset trap. When the snmp keyword is used, you can specify the authentication option to enable SNMP Authentication Failure traps. (The snmp-server enable traps snmp authentication command replaces the snmp-server trap-authentication command.) If no option is specified, all SNMP traps are enabled. When the voice keyword is used, you can enable SNMP poor quality of voice traps by using the qov option. |
This command is disabled by default. No traps are enabled.
Some trap types cannot be controlled with this command. These traps are either always enabled or enabled by some other means. For example, the linkUpDown messages are disabled by the no snmp trap link-status command.
If you enter this command with no keywords, the default is to enable all trap types.
Global configuration
This command first appeared in Cisco IOS Release 11.1.
This command is useful for disabling traps that are generating a large amount of uninteresting or useless noise.
If you do not enter an snmp-server enable traps command, no traps controlled by this command are sent. In order to configure the router to send these SNMP traps, you must enter at least one snmp-server enable traps command. If you enter the command with no keywords, all trap types are enabled. If you enter the command with a keyword, only the trap type related to that keyword is enabled. In order to enable multiple types of traps, you must issue a separate snmp-server enable traps command for each trap type and option.
The snmp-server enable traps command is used in conjunction with the snmp-server host command. Use the snmp-server host command to specify which host or hosts receive SNMP traps. In order to send traps, you must configure at least one snmp-server host command.
For a host to receive a trap controlled by this command, both the snmp-server enable traps command and the snmp-server host command for that host must be enabled. If the trap type is not controlled by this command, just the appropriate snmp-server host command must be enabled.
The trap types used in this command all have an associated MIB object that allows them to be globally enabled or disabled. Not all of the trap types available in the snmp-server host command have notificationEnable MIB objects, so some of these cannot be controlled using the snmp-server enable traps command.
The following example enables the router to send SNMP poor quality of voice traps:
configure terminal snmp-server enable trap voice poor-qov
The following example enables the router to send all traps to the host myhost.cisco.com using the community string public:
snmp-server enable traps snmp-server host myhost.cisco.com public
The following example enables the router to send Frame Relay and environmental monitor traps to the host myhost.cisco.com using the community string public:
snmp-server enable traps frame-relay snmp-server enable traps envmon temperature snmp-server host myhost.cisco.com public
The following example will not send traps to any host. The BGP traps are enabled for all hosts, but the only traps enabled to be sent to a host are ISDN traps.
snmp-server enable traps bgp snmp-server host bob public isdn
You can use the master indexes or search online to find documentation of related commands.
snmp enable peer-trap peer-qov
snmp-server host
snmp-server trap-source
snmp trap illegal-address
snmp trap link-status
This command contains no arguments or keywords.
The default for this command is enabled.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the snmp trap link-status command to enable SNMP trap messages (linkup and linkdown) to be generated whenever this voice port is brought online or offline.
If you are managing the equipment with an SNMP manager (such as Maestro) this command should be enabled. Enabling link-status messages will allow the SNMP manager to learn of a status change without polling the equipment. If you are not using an SNMP manager, this command should be disabled to avoid unnecessary network traffic.
The following example enables SNMP trap messages for voice-port 2/1/0:
voice-port 2/1/0 snmp trap link-stat
You can use the master index or search online to find documentation of related commands.
smnp enable peer-trap poor-qov
snmp-server enable trap poor-qov
| initial seconds | Specifies the initial timeout duration in seconds. Valid entries are any integer from 0 to 120. |
The default value is 10 seconds.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the timeouts initial command to specify the number of seconds the system will wait for the caller to input the first digit of the dialed digits. The timeouts initial timer is activated when the call is accepted and is deactivated when the caller inputs the first digit. If the configured timeout value is exceeded, the caller is notified through the appropriate tone and the call is terminated.
To disable the timeouts initial timer, set the seconds value to 0.
The following example sets the initial digit timeout value to 15 seconds:
voice-port 1/0/0 timeouts initial 15
You can use the master index or search online to find documentation of related commands.
timeouts interdigit
timing
| seconds | Specifies the interdigit timeout duration in seconds. Valid entries are any integer from 0 to 120. |
The default value is 10 seconds.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the timeouts interdigit command to specify the number of seconds the system will wait (after the caller has input the initial digit) for the caller to input a subsequent digit of the dialed digits. The timeouts interdigit timer is activated when the caller inputs a digit and restarted each time the caller inputs another digit until the destination address is identified. If the configured timeout value is exceeded before the destination address is identified, the caller is notified through the appropriate tone and the call is terminated.
To disable the timeouts interdigit timer, set the seconds value to 0.
The following example sets the interdigit timeout value for 15 seconds:
voice-port 1/0/0 timeouts interdigit 15
You can use the master index or search online to find documentation of related commands.
timeouts initial
timing
| timing-value | One of the keyword/argument pairs listed in Table 3-7. |
| clear-wait milliseconds | Indicates the minimum amount of time, in milliseconds, between the inactive seizure signal and the call being cleared. Valid entries are numbers from 200 to 2000. |
| delay-duration milliseconds | Indicates the delay signal duration for delay dial signaling, in milliseconds. Valid entries are numbers from 100 to 5000. |
| delay-start milliseconds | Indicates the minimum delay time, in milliseconds, from outgoing seizure to outdial address. Valid entries are numbers from 20 to 2000. |
| dial-pulse min-delay milliseconds | Indicates the time, in milliseconds, between the generation of wink-like pulses. Valid entries are numbers from 0 to 5000. |
| digit milliseconds | Indicates the DTMF digit signal duration, in milliseconds. Valid entries are numbers from 50 to 100. |
| inter-digit milliseconds | Indicates the DTMF inter-digit duration, in milliseconds. Valid entries are numbers from 50 to 500. |
| pulse pulses per second | Indicates the pulse dialing rate, in pulses per second. Valid entries are numbers from 10 to 20. |
| pulse-inter-digit milliseconds | Indicates the pulse dialing inter-digit timing, in milliseconds. Valid entries are numbers from 100 to 1000. |
| wink-duration milliseconds | Indicates the maximum wink signal duration, in milliseconds, for a wink start signal. Valid entries are numbers from 100 to 400. |
| wink-wait milliseconds | Indicates the maximum wink-wait duration, in milliseconds, for a wink start signal. Valid entries are numbers from 100 to 5000. |
The default values are listed in Table 3-8.
| clear-wait milliseconds | The default value is 400. |
| delay-duration milliseconds | The default value is 2000. |
| delay-start milliseconds | The default value is 300. |
| dial-pulse min-delay milliseconds | The default value is 140. |
| digit milliseconds | The default value is 100. |
| inter-digit milliseconds | The default value is 100. |
| pulse pulses per second | The default is 20. |
| pulse-inter-digit milliseconds | The default value is 500. |
| wink-duration milliseconds | The default value is 200. |
| wink-wait milliseconds | The default value is 200. |
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the timing command to specify timing parameters other than those defined by the timeouts commands.
Use the timing command with the dial-pulse min-delay keyword with PBXs requiring a wink-like pulse, even though they have been configured for delay-dial signaling. If the value for this keyword is set to 0, the router will not generate this wink-like pulse.
Table 3-9 lists the call signal directions for the Timing/Keyword argument pairs.
| Timing Keyword/Argument Pairs | Call Signal Direction |
|---|---|
| clear-wait milliseconds | Not applicable |
| delay-duration milliseconds | Out |
| delay-start milliseconds | Out |
| dial-pulse min-delay milliseconds | In |
| digit milliseconds | Out |
| inter-digit milliseconds | Out |
| pulse pulses per second | Out |
| pulse-inter-digit milliseconds | Out |
| wink-duration milliseconds | Out |
| wink-wait milliseconds | Out |
The following example configures the clear-wait duration to 300 milliseconds:
voice-port 1/0/0 timing clear-wait 300
You can use the master index or search online to find documentation of related commands.
timeouts initial
timeouts interdigit
| 1 | Indicates the following lead configuration: E--output, relay to ground. M--input, referenced to ground. |
| 2 | Indicates the following lead configuration: E--output, relay to SG. M--input, referenced to ground. SB--feed for M, connected to -48V. SG--return for E, galvanically isolated from ground. |
| 3 | Indicates the following lead configuration: E--output, relay to ground. M--input, referenced to ground. SB--connected to -48V. SG--connected to ground. |
| 5 | Indicates the following lead configuration: E--output, relay to ground. M--input, referenced to -48V. |
The default value is 1.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the type command to specify the E&M interface for a particular voice port. With 1, the tie-line equipment generates the E-signal to the PBX type grounding the E-lead. The tie line equipment detects the M-signal by detecting current flow to ground. If you select 1, a common ground must exist between the line equipment and the PBX.
With 2, the interface requires no common ground between the equipment, thereby avoiding ground loop noise problems. The E-signal is generated toward the PBX by connecting it to SG. M-signal is indicated by the PBX connecting it to SB. While Type 2 interfaces do not require a common ground, they do have the tendency to inject noise into the audio paths because they are asymmetrical with respect to the current flow between devices.
With 3, the interface operates the same as Type 1 interfaces with respect to the E-signal. The M-signal, however, is indicated by the PBX connecting it to SB on assertion and alternately connecting it to SG during inactivity. If you select 3, a common ground must be shared between equipment.
With 5, the Type 5 line equipment indicates E-signal to the PBX by grounding the E-lead. The PBX indicates M-signal by grounding the M-lead. A Type 5 interface is quasi-symmetrical in that while the line is up, current flow is more or less equal between the PBX and the line equipment but noise injection is a problem.
The following example selects Type 3 as the interface type for your voice port:
voice-port 1/0/0 type 3
This command has no arguments or keywords.
Enabled
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the vad command to enable voice activity detection. With VAD, silence is not transmitted over the network, only audible speech. If you enable VAD, the sound quality will be slightly degraded but the connection will monopolize much less bandwidth. If you use the no form of this command, VAD is disabled and voice data is continuously transmitted to the IP backbone.
This command is applicable only to VoIP peers.
The following example enables VAD:
dial-peer voice 10 voip vad
You can use the master index or search online to find documentation of related commands.
comfort-noise
To enter the voice-port configuration mode, use the voice-port global configuration command.
voice-port slot-number/subunit-number/port| slot-number/ | Specifies the slot number in the Cisco router where the voice network module is installed. Valid entries are from 0 to 3, depending on the voice interface card you have installed. |
| subunit-number/ | Specifies the subunit on the voice network module where the voice port is located. Valid entries are 0 or 1. |
| port | Specifies the voice port. Valid entries are 0 or 1. |
Global configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the voice-port global configuration command to switch to the voice-port configuration mode from the global configuration mode. Use the exit command to exit the voice-port configuration mode and return to the global configuration mode.
For more information about the physical characteristics of the voice network module, or how to install it, refer to installation documentation that came with your voice network module.
The following example accesses the voice-port configuration mode for port 0, located on subunit 0 on a voice interface card installed in slot 1:
configure terminal voice-port 1/0/0
You can use the master index or search online to find documentation of related commands.
dial-peer
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