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Cisco IOS Voice Command Reference
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About Cisco IOS Software Documentation
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Using Cisco IOS Software
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Commands: A
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Commands: B
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Commands:cac master through call application stats
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Commands: call application voice through call-denial
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Commands: call fallback through called-number (dial-peer)
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Commands: caller-id (dial peer) through ccm-manager switchover-to-backup
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Commands: ccs connect (controller) through clear vsp statistics
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Commands: clid through cptone
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Commands: default (MGCP profile) through direct-inward-dial
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Commands: disable-early-media 180 through dualtone
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Commands: E
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Commands: F
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Commands: G
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Commands: H
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Commands: icpif through irq global-request
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Commands: isdn bind-l3 through ixi transport http
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Commands: K
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Commands: L
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Commands: map q850-cause through mgcp package-capability
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Commands: mgcp persistent through mmoip aaa send-id secondary
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Commands: mode (ATM/T1/E1 controller) through mwi-server
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Commands: N
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Commands: O
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Commands: package through pattern
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Commands: permit hostname through proxy h323
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Commands: Q
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Commands: R
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Commands: sccp through service-type call-check
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Commands: session through sgcp tse payload
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Commands: show aal2 profile through show call filter match-list
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Commands: show call history fax through show debug condition
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Commands: show dial-peer video through show gatekeeper zone status
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Commands: show gateway through show modem relay statistics
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Commands: show mrcp client session active through show sgcp statistics
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Commands: show sip service through show trunk hdlc
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Commands: show vdev through show voice statistics memory-usage
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Commands: show voice translation-profile through shutdown (voice-port)
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Commands: signal through srv version
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Commands: ss7 mtp2-variant through switchover method
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Commands: target carrier-id through threshold noise
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Commands: timeout (auto-config application) through timing clear-wait
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Commands:timing delay-duration through type (voice)
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Commands: U
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Commands: vad (dial peer) through voice class dualtone-detect-params
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Commands: voice class h323 through vxml tree memory
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Commands: W
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Commands: Z
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Feedback
Table Of Contents
playout-delay mode (dial peer)
playout-delay mode (voice-port)
pots distinctive-ring-guard-time
permit hostname (SIP)
To store hostnames used during validatation of initial incoming INVITE messages, use the permit hostname command in SIP-ua configuration mode. To remove a stored hostname, use the no form of this command.
permit hostname dns: domain name
no permit hostname
Syntax Description
dns: domain name
Domain name in DNS format. Domain names can be up to 30 characters in length; domain names exceeding 30 characters will be truncated.
Command Modes
SIP-ua configuration
Command History
Usage Guidelines
The permit hostname command allows you to specify hostnames in FQDN (fully qualified domain name) format used during validation of incoming initial INVITE messages. The length of the hostname can be up to 30 characters; hostnames exceeding 30 characters will be truncated. You can store up to 10 hostnames by repeating the permit hostname command.
Once configured, initial INVITEs with a hostname in the requested Universal Resource Identifier (URI) are compared to the configured list of hostnames. If there is a match, the INVITE is processed; if there is a mismatch, a "400 Bad Request - Invalid Host" is sent, and the call is rejected.
Note
Before Software Release 12.4(9)T, hostnames in incoming INVITE-request messages were only validated when they were in IPv4 format; now you can specify hostnames in fully qualified domain name (FQDN) format.
Examples
The following example show you how to set the hostname to sip.example.com:
Router(config)# sip-uaRouter(conf-sip-ua)# permit hostname dns:sip.example.comphone context
To filter out uniform resource identifiers (URIs) that do not contain a phone-context field that matches the configured pattern, use the phone context command in voice URI class configuration mode. To remove the pattern, use the no form of this command.
phone context phone-context-pattern
no phone context
Syntax Description
phone-context-pattern
Cisco IOS regular expression pattern to match against the phone context field in a SIP or TEL URI. Can be up to 32 characters.
Command Default
No default behavior or values
Command Modes
Voice URI class configuration
Command History
Usage Guidelines
•
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Examples
The following example sets a match on the phone context in the URI voice class:
voice class uri 10 telphone number ^408phone context 555Related Commands
phone number
To match a call based on the phone-number field in a telephone (TEL) uniform resource identifier (URI), use the phone number command in voice URI class configuration mode. To remove the pattern, use the no form of this command.
phone number phone-number-pattern
no phone number
Syntax Description
phone-number-pattern
Cisco IOS regular expression pattern to match against the phone-number field in a TEL URI. Can be up to 32 characters.
Command Default
No default behavior or values
Command Modes
Voice URI class configuration
Command History
Usage Guidelines
•
•
Examples
The following example defines a voice class that matches on the phone number field in a TEL URI:
voice class uri r101 telphone number ^408Related Commands
pickup direct
To designate an SCCP telephony control (STC) feature access code for directed call pickup, use the pickup direct command in STC application feature access-code configuration mode. To return the code to its default, use the no form of this command.
pickup direct keypad-character
no pickup direct
Syntax Description
Command Default
The default code for directed call pickup is 6.
Command Modes
STC application feature access-code configuration
Command History
Usage Guidelines
This command is used with the STC application, which enables certain features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control.
Use this command to set the feature access code (FAC) for the directed pickup feature. Use this command only if you want to change the code from its default (6).
Note
To use the directed pickup feature on a phone, dial the FAC prefix and the FAC that has been configured with this command (or the default if this command has not been used). Then dial the number that is ringing. For example, if the prefix is ** (the default), the feature access code is 6 (the default), and the ringing extension is 310, dial **6310 to pick up the call.
The directed pickup feature can also be used to retrieve the last call that was parked at a call-park slot by this extension. To retrieve the last parked call, dial the following sequence: FAC prefix, directed pickup FAC, and asterisk (*). The default is **6*.
If you set the directed pickup FAC code to a value that is already in use for another code, you receive a warning message. If you configure a duplicate code, the system implements the first matching feature in the order of precedence shown in the output of the show stcapp feature codes command.
The show running-config command displays nondefault FACs only. The show stcapp feature codes command displays all feature access codes.
Examples
The following example sets a feature access code prefix of two pound signs (##) and a directed pickup code of 3. After these values are configured, a phone user presses ##3 and the ringing extension number to pick up the incoming call.
Router(config)# stcapp feature access-codeRouter(stcapp-fac)# prefix ##Router(stcapp-fac)# call forward all 2Router(stcapp-fac)# call forward cancel 4Router(stcapp-fac)# pickup local 6Router(stcapp-fac)# pickup group 5Router(stcapp-fac)# pickup direct 3Router(stcapp-fac)# exitRelated Commands
pickup group
To designate an SCCP telephony control (STC) feature access code for group call pickup, use the pickup group command in STC application feature access-code configuration mode. To return the code to its default, use the no form of this command.
pickup group keypad-character
no pickup group
Syntax Description
Command Default
The default code for group call pickup is 4.
Command Modes
STC application feature access-code configuration
Command History
Usage Guidelines
This command is used with the STC application, which enables certain features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control.
Use this command to set the feature access code (FAC) for the group call pickup feature. Use this command only if you want to change the FAC from its default (4).
To use the group call pickup feature on a phone, dial the FAC prefix and the FAC that has been configured with this command (or the default if this command has not been used). Then dial the pickup group number of the phone that is ringing. For example, if the prefix is ** (the default), the FAC is 4 (the default), and the pickup group number of the ringing extension is 88, dial **488 to pick up the call.
If you set this FAC to a value that is already in use for another FAC, you receive a warning message. If you configure a duplicate code, the system implements the first matching feature in the order of precedence shown in the output of the show stcapp feature codes command.
The show running-config command displays nondefault FACs only. The show stcapp feature codes command displays all FACs.
Examples
The following example sets a FAC prefix of two pound signs (##) and a group pickup FAC of 3. After these values are configured, a phone user presses ##3 and the group number of the ringing extension number to pick up the incoming call.
Router(config)# stcapp feature access-codeRouter(stcapp-fac)# prefix ##Router(stcapp-fac)# call forward all 2Router(stcapp-fac)# call forward cancel 4Router(stcapp-fac)# pickup local 6Router(stcapp-fac)# pickup group 3Router(stcapp-fac)# pickup direct 5Router(stcapp-fac)# exitRelated Commands
pickup local
To designate an SCCP telephony control (STC) feature access code for group call pickup within a local group, use the pickup local command in STC application feature access-code configuration mode. To return the code to its default, use the no form of this command.
pickup local keypad-character
no pickup local
Syntax Description
Command Default
The default code for local group pickup is 3.
Command Modes
STC application feature access-code configuration
Command History
Usage Guidelines
This command is used with the STC application, which enables certain features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control.
Use this command to set the feature access code (FAC) for the local group pickup feature. Use this command only if you want to change the FAC from its default (3).
To use the local group pickup feature on a phone, dial the FAC prefix and the FAC that has been configured with this command (or the default if this command has not been used). For example, if the prefix is ** (the default), the FAC is 3 (the default), and the pickup group of the ringing extension is the same as yours, dial **3 to pick up the call.
If you set this FAC to a value that is already in use for another FAC, you receive a warning message. If you configure a duplicate code, the system implements the first matching feature in the order of precedence shown in the output of the show stcapp feature codes command.
The show running-config command displays nondefault FACs only. The show stcapp feature codes command displays all FACs.
Examples
The following example sets a FAC prefix of two pound signs (##) and a local group pickup FAC of 6. After these values are configured, a phone user presses ##6 to pick up an incoming call in the same group.
Router(config)# stcapp feature access-codeRouter(stcapp-fac)# prefix ##Router(stcapp-fac)# call forward all 2Router(stcapp-fac)# call forward cancel 4Router(stcapp-fac)# pickup local 6Router(stcapp-fac)# pickup group 5Router(stcapp-fac)# pickup direct 3Router(stcapp-fac)# exitRelated Commands
playout-delay (dial peer)
To tune the playout buffer on digital signal processors (DSPs) to accommodate packet jitter caused by switches in the WAN, use the playout-delay command in dial peer configuration mode. To reset the playout buffer to the default, use the no form of this command.
playout-delay {fax milliseconds | maximum milliseconds | minimum {default | low | high} | nominal milliseconds}
no playout-delay {fax | maximum | minimum | nominal}
Syntax Description
Defaults
fax—300 milliseconds
maximum—200 milliseconds
minimum—default (40 milliseconds)
nominal—200 millisecondsCommand Modes
Dial peer configuration (config-dial-peer)
Command History
Usage Guidelines
Before Cisco IOS Release 12.1(5)T, this command was used in voice-port configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay should be configured in dial-peer configuration mode on the Voice over IP (VoIP) dial peer that is on the receiving end of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the gateway, playout delay should be configured in dial-peer configuration mode. When there are numerous dial peers to configure, it might be simpler to configure playout delay on a voice port. If conflicting playout-delay values have been configured on a voice port and on a dial peer, the dial-peer configuration takes precedence.
Playout delay is the amount of time that elapses between the time at which a voice packet is received at the jitter buffer on the DSP and the time at which it is played out to the codec. In most networks with normal jitter conditions, the defaults are adequate and you will not need to configure this command.
In situations in which you want to improve voice quality by reducing jitter or you want to reduce network delay, you can configure playout-delay parameters. The parameters are slightly different for each of the two playout-delay modes, adaptive and fixed (see the playout-delay mode command).
In adaptive mode, the average delay for voice packets varies depending on the amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum and minimum limits that have been configured. The maximum limit establishes the highest value to which the adaptive delay is set. The minimum limit is the low-end threshold for the delay of incoming packets by the adaptive jitter buffer. Algorithms in the DSPs that control the growth and shrinkage of the jitter buffer are weighted toward the improvement of voice quality at the expense of network delay: jitter buffer size increases rapidly in response to spikes in network transmissions and decreases slowly in response to reduced congestion.
In fixed mode, the nominal value is the amount of playout delay applied at the beginning of a call by the jitter buffer in the gateway and is also the maximum size of the jitter buffer throughout the call.
As a general rule, if there is excessive breakup of voice due to jitter with the default playout-delay settings, increase playout delay times. If your network is small and jitter is minimal, decrease playout-delay times for a smaller overall delay.
When there is bursty jitter in the network, voice quality can be degraded even though the jitter buffer is actually adjusting the playout delay correctly. The constant readjustment of playout delay to erratic network conditions causes voice quality problems that are usually alleviated by increasing the minimum playout delay-value in adaptive mode or by increasing the nominal delay for fixed mode.
Use the show call active voice command to display the current delay, as well as high- and low-water marks for delay during a call. Other fields that can help determine the size of a jitter problem are ReceiveDelay, GapFillWith..., LostPackets, EarlyPackets, and LatePackets. The following is sample output from the show call active voice command:
VOIP:ConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4]IncomingConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4]RemoteIPAddress=192.168.100.101RemoteUDPPort=18834RoundTripDelay=26 msSelectedQoS=best-efforttx_DtmfRelay=inband-voiceFastConnect=TRUESeparate H245 Connection=FALSEH245 Tunneling=FALSESessionProtocol=ciscoSessionTarget=OnTimeRvPlayout=417000GapFillWithSilence=850 msGapFillWithPrediction=2590 msGapFillWithInterpolation=0 msGapFillWithRedundancy=0 msHiWaterPlayoutDelay=70 msLoWaterPlayoutDelay=29 msReceiveDelay=39 msLostPackets=0EarlyPackets=0LatePackets=86Examples
The following example uses default adaptive mode with a minimum playout delay of 10 ms and a maximum playout delay of 60 ms on VoIP dial peer 80. The size of the jitter buffer is adjusted up and down on the basis of the amount of jitter that the DSP finds, but is never smaller than 10 ms and never larger than 60 ms.
dial-peer 80 voipplayout-delay minimum lowplayout-delay maximum 60Related Commands
playout-delay (voice-port)
To tune the playout buffer to accommodate packet jitter caused by switches in the WAN, use the playout-delay command in voice-port configuration mode. To reset the playout buffer to the default, use the no form of this command.
playout-delay {fax | maximum | nominal} milliseconds
no playout-delay {fax | maximum | nominal}
Syntax Description
Defaults
fax—300 milliseconds
maximum—160 milliseconds
nominal—80 millisecondsCommand Modes
Voice-port configuration
Command History
Usage Guidelines
If there is excessive breakup of voice due to jitter with the default playout delay settings, increase the delay times. If your network is small and jitter is minimal, decrease the delay times to reduce delay.
Before Cisco IOS Release 12.1(5)T, the playout-delay command was configured in voice-port configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay should be configured in dial-peer configuration mode on the Voice over IP (VoIP) dial peer that is on the receiving end of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the gateway, playout delay should be configured in dial-peer configuration mode. When there are numerous dial peers to configure, it might be simpler to configure playout delay on a voice port. If conflicting playout-delay values have been configured on a voice port and on a dial peer, the dial-peer configuration takes precedence.
Playout delay is the amount of time that elapses between the time at which a voice packet is received at the jitter buffer on the DSP and the time at which it is played out to the codec. In most networks with normal jitter conditions, the defaults are adequate and you will not need to configure the playout-delay command.
In situations in which you want to improve voice quality by reducing jitter or you want to reduce network delay, you can configure playout-delay parameters. The parameters are slightly different for each of the two playout-delay modes, adaptive and fixed (see the playout-delay mode command).
In adaptive mode, the average delay for voice packets varies depending on the amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum and minimum limits that have been configured. The maximum limit establishes the highest value to which the adaptive delay will be set. The minimum limit is the low-end threshold for incoming packet delay that is created by the adaptive jitter buffer. Algorithms in the DSPs that control the growth and shrinkage of the jitter buffer are weighted toward the improvement of voice quality at the expense of network delay: jitter buffer size increases rapidly in response to spikes in network transmissions and decreases slowly in response to reduced congestion.
In fixed mode, the nominal value is the amount of playout delay applied at the beginning of a call by the jitter buffer in the gateway and is also the maximum size of the jitter buffer throughout the call.
As a general rule, if there is excessive breakup of voice due to jitter with the default playout-delay settings, increase playout-delay times. If your network is small and jitter is minimal, decrease playout-delay times for a smaller overall delay.
When there is bursty jitter in the network, voice quality can be degraded even though the jitter buffer is actually adjusting the playout delay correctly. The constant readjustment of playout delay to erratic network conditions causes voice quality problems that are usually alleviated by increasing the minimum playout-delay value in adaptive mode or by increasing the nominal delay for fixed mode.
Note
Use the show call active voice command to display the current delay, as well as high- and low-water marks for delay during a call. Other fields that can help determine the size of a jitter problem are GapFillWith..., ReceiveDelay, LostPackets, EarlyPackets, and LatePackets. The following is sample output from the show call active voice command:
VOIP:ConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4]IncomingConnectionId[0xECDE2E7B 0xF46A003F 0x0 0x47070A4]RemoteIPAddress=192.168.100.101RemoteUDPPort=18834RoundTripDelay=26 msSelectedQoS=best-efforttx_DtmfRelay=inband-voiceFastConnect=TRUESeparate H245 Connection=FALSEH245 Tunneling=FALSESessionProtocol=ciscoSessionTarget=OnTimeRvPlayout=417000GapFillWithSilence=850 msGapFillWithPrediction=2590 msGapFillWithInterpolation=0 msGapFillWithRedundancy=0 msHiWaterPlayoutDelay=70 msLoWaterPlayoutDelay=29 msReceiveDelay=39 msLostPackets=0EarlyPackets=0LatePackets=86Examples
The following example sets nominal playout delay to 80 ms and maximum playout delay to 160 ms on voice port 1/0/0:
voice-port 1/0/0playout-delay nominal 80playout-delay maximum 160Related Commands
playout-delay mode (dial peer)
To select fixed or adaptive mode for playout delay from the jitter buffer on digital signal processors (DSPs), use the playout-delay mode command in dial peer configuration mode. To reset to the default, use the no form of this command.
playout-delay mode {adaptive | fixed}
no playout-delay mode
Syntax Description
Command Default
Adaptive jitter buffer size
Command Modes
Dial peer configuration
Command History
12.1(5)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco MC3810, and Cisco ICS 7750. The no-timestamps keyword was removed.
Usage Guidelines
Before Cisco IOS Release 12.1(5)T, this command was used only in voice-port configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay should be configured in dial peer configuration mode on the VoIP dial peer that is on the receiving end of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the gateway, playout delay should be configured in dial peer configuration mode.
Tip
In most networks with normal jitter conditions, the default is adequate and you do not need to configure this command.
The default is adaptive mode, in which the average delay for voice packets varies depending on the amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum and minimum limits that have been configured.
Select fixed mode only when you understand your network conditions well, and when you have a network with very poor quality of service (QoS) or when you are interworking with a media server or similar transmission source that tends to create a lot of jitter at the transmission source. In most situations it is better to configure adaptive mode and let the DSP size the jitter buffer according to current conditions.
Examples
The following example sets adaptive playout-delay mode with a high (80 ms) minimum delay on a VoIP dial peer 80:
dial-peer 80 voipplayout-delay mode adaptiveplayout-delay minimum highRelated Commands
playout-delay
Tunes the jitter buffer on DSPs for playout delay of voice packets.
show call active voice
Displays active call information for voice calls.
playout-delay mode (voice-port)
To select fixed or adaptive mode for playout delay from the jitter buffer on digital signal processors (DSPs), use the playout-delay mode command in voice port configuration mode. To reset to the default, use the no form of this command.
playout-delay mode {adaptive | fixed}
no playout-delay mode
Syntax Description
Command Default
Adaptive jitter buffer size
Command Modes
Voice-port configuration
Command History
Usage Guidelines
Before Cisco IOS Release 12.1(5)T, this command was used only in voice-port configuration mode. For Cisco IOS Release 12.1(5)T and later releases, in most cases playout delay should be used in dial peer configuration mode on the VoIP dial peer that is on the receiving end of the voice traffic that is to be buffered. This dial peer senses network conditions and relays them to the DSPs, which adjust the jitter buffer as necessary. When multiple applications are configured on the gateway, playout delay should be configured in dial peer configuration mode.
Tip
In most networks with normal jitter conditions, the default is adequate and you do not need to configure the playout-delay mode command.
The default is adaptive mode, in which the average delay for voice packets varies depending on the amount of interarrival variation that packets have as the call progresses. The jitter buffer grows and shrinks to compensate for jitter and to keep voice packets playing out smoothly, within the maximum and minimum limits that have been configured.
Select fixed mode only when you understand your network conditions well, and when you have a network with very poor quality of service (QoS) or when you are interworking with a media server or similar transmission source that tends to create a lot of jitter at the transmission source. In most situations it is better to configure adaptive mode and let the DSP size the jitter buffer according to current conditions.
Examples
The following example sets fixed mode on a Cisco 3640 voice port with a nominal delay of 80 ms.
voice-port 1/1/0playout-delay mode fixedplayout-delay nominal 80Related Commands
playout-delay
Tunes the jitter buffer on DSPs for playout delay of voice packets.
show call active voice
Displays active call information for voice calls.
port (Annex G neighbor BE)
To configure the port number of the neighbor that is used for exchanging Annex G messages, use the port command in Annex G Neighbor BE configuration mode. To remove the port number, use the no form of this command.
port neighbor-port
no port
Syntax Description
neighbor-port
Port number of the neighbor. This number is used for exchanging Annex G messages. The default port number is 2099.
Command Default
2099
Command Modes
Annex G Neighbor BE Configuration
Command History
Examples
The following example sets a neighbor BE to port number 2010.
Router(config-annexg-neigh)# port 2010Related Commands
port (dial peer)
To associate a dial peer with a specific voice port, use the port command in dial peer configuration mode. To cancel this association, use the no form of this command.
Cisco 1750 and Cisco 3700 Series
port slot-number/port
no port slot-number/port
Cisco 2600 series, Cisco 3600 Series, and Cisco 7200 Series
port {slot-number/subunit-number/port | slot/port:ds0-group-no}
no port {slot-number/subunit-number/port | slot/port:ds0-group-no}
Cisco AS5300
port controller-number:D
no port controller-number:D
Cisco uBR92x Series
port slot/subunit/port
no port slot/subunit/port
Syntax Description
Cisco 1750 and Cisco 3700 Series
Cisco 2600 series, Cisco 3600 Series, and Cisco 7200 Series
Cisco AS5300
controller-number
Specifies the T1 or E1 controller.
:D
Indicates the D channel associated with ISDN PRI.
Cisco uBR92x series
Command Default
No port is configured.
Command Modes
Dial peer configuration
Command History
Usage Guidelines
This command is used for calls that come from a telephony interface to select an incoming dial peer and for calls that come from the VoIP network to match a port with the selected outgoing dial peer.
This command applies only to POTS peers.
Note
Examples
The following example associates POTS dial peer 10 with voice port 1, which is located on subunit 0 and accessed through port 0:
dial-peer voice 10 potsport 1/0/0The following example associates POTS dial peer 10 with voice port 0:D:
dial-peer voice 10 potsport 0:DThe following example associates POTS dial peer 10 with voice port 1/0/0:D (T1 card):
dial-peer voice 10 potsport 1/0/0:DRelated Commands
port (MGCP profile)
To associate a voice port with the Media Gateway Control Protocol (MGCP) profile that is being configured, use the port command in MGCP profile configuration mode. To disassociate the voice port from the profile, use the no form of this command.
port port-number
no port port-number
Syntax Description
port-number
Voice port or DS0-group number to be used as an MGCP endpoint associated with an MGCP profile.
Command Default
No default behavior or values
Command Modes
MGCP profile configuration
Command History
Usage Guidelines
This command is used when values for an MGCP profile are configured.
This command associates a voice port with the MGCP profile that is being defined. To associate multiple voice ports with a profile, repeat this command with different voice port arguments.
This command is not used when the default MGCP profile is configured because the values in the default profile configuration apply to all parameters that have not been otherwise configured for a user-defined MGCP profile.
Examples
The following example associates an analog voice port with an MGCP profile on a Cisco uBR925 platform:
Router(config)# mgcp profile ny110caRouter(config-mgcp-profile)# port 0Related Commands
port media
To specify the serial interface to which the local video codec is connected for a local video dial peer, use the port media command in video dial peer configuration mode. To remove any configured locations from the dial peer, use the no form of this command.
port media interface
no port media
Syntax Description
Command Default
No interface is specified
Command Modes
Video dial peer configuration
Command History
12.0(5)XK
This command was introduced for ATM video dial peer configuration on the Cisco MC3810.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Examples
The following example specifies serial interface 0 as the specified interface for the codec local video dial peer 10:
dial-peer video 10 videocodecport media Serial0Related Commands
port signal
Specifies the slot location of the VDM and the port location of the EIA/TIA-366 interface for signaling.
show dial-peer video
Displays dial peer configuration.
port signal
To specify the slot location of the video dialing module (VDM) and the port location of the EIA/TIA-366 interface for signaling for a local video dial peer, use the port signal command in video dial peer configuration mode. To remove any configured locations from the dial peer, use the no form of this command.
port signal slot/port
no port signal
Syntax Description
slot
Slot location of the VDM. Valid values are 1 and 2.
port
Port location of the EIA/TIA-366 interface.
Command Default
No locations are specified
Command Modes
Video dial peer configuration
Command History
12.0(5)XK
This command was introduced for ATM video dial peer configuration on the Cisco MC3810.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
Examples
The following example sets up the VDM and EIA/TIA-366 interface locations for the local video dial peer designated as 10:
dial-peer video 10 videocodecport signal 1/0Related Commands
port media
Specifies the serial interface to which the local video codec is connected.
show dial-peer video
Displays dial peer configuration.
pots call-waiting
To enable the local call-waiting feature, use the global configuration pots call-waiting command in global configuration mode. To disable the local call-waiting feature, use the no form of this command.
pots call-waiting {local | remote}
no pots call-waiting {local | remote}
Syntax Description
local
Enable call waiting on a local basis for the routers.
remote
Rely on the network provider service instead of the router to hold calls.
Command Default
Remote, in which case the call- holding pattern follows the settings of the service provider rather than those of the router.
Command Modes
Global configuration
Command History
Usage Guidelines
To display the call-waiting setting, use the show running-config or show pots status command. The ISDN call waiting service is used if it is available on the ISDN line connected to the router even if local call waiting is configured on the router. That is, if the ISDN line supports call waiting, the local call waiting configuration on the router is ignored.
Examples
The following example enables local call waiting on a router:
pots call-waiting localRelated Commands
pots country
To configure your connected telephones, fax machines, or modems to use country-specific default settings for each physical characteristic, use the pots country command in global configuration mode. To disable the use of country-specific default settings, use the no form of this command.
pots country country
no pots country country
Syntax Description
Command Default
A default country is not defined.
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to the Cisco 800 series routers.
If you need to change a country-specific default setting of a physical characteristic, you can use the associated command listed in the "Related Commands" section. Enter the pots country ? command to get a list of supported countries and the code you must enter to indicate a particular country.
Examples
The following example specifies that the devices connected to the telephone ports use default settings specific to Germany for the physical characteristics:
pots country deRelated Commands
pots dialing-method
To specify how the router collects and sends digits dialed on your connected telephones, fax machines, or modems, use the pots dialing-method command in global configuration mode. To disable the specified dialing method, use the no form of this command.
pots dialing-method {overlap | enblock}
no pots dialing-method {overlap | enblock}
Syntax Description
overlap
The router sends each digit dialed in a separate message.
enblock
The router collects all digits dialed and sends the digits in one message.
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to Cisco 800 series routers.
To interrupt the collection and transmission of dialed digits, enter a pound sign (#), or stop dialing digits until the interdigit timer runs out (10 seconds).
Examples
The following example specifies that the router uses the enblock dialing method:
pots dialing-method enblockRelated Commands
pots disconnect-supervision
To specify how a router notifies the connected telephones, fax machines, or modems when the calling party has disconnected, use the pots disconnect-supervision command in global configuration mode. To disable the specified disconnect method, use the no form of this command.
pots disconnect-supervision {osi | reversal}
no pots disconnect-supervision {osi | reversal}
Syntax Description
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to Cisco 800 series routers.
Most countries except Japan typically use the osi option. Japan typically uses the reversal option.
Examples
The following example specifies that the router uses the OSI disconnect method:
pots disconnect-supervision osiRelated Commands
pots disconnect-time
To specify the interval in which the disconnect method is applied if your connected telephones, fax machines, or modems fail to detect that a calling party has disconnected, use the pots disconnect-time command in global configuration mode. To disable the specified disconnect interval, use the no form of this command.
pots disconnect-time interval
no pots disconnect-time interval
Syntax Description
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to Cisco 800 series routers.
The pots disconnect-supervision command configures the disconnect method.
Examples
The following example specifies that the connected devices apply the configured disconnect method for 100 ms after a calling party disconnects:
pots disconnect-time 100Related Commands
pots distinctive-ring-guard-time
To specify the delay in which a telephone port can be rung after a previous call is disconnected, use the pots distinctive-ring-guard-time command in global configuration mode. To disable the specified delay, use the no form of this command.
pots distinctive-ring-guard-time milliseconds
no pots distinctive-ring-guard-time milliseconds
Syntax Description
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example specifies that a telephone port can be rung 100 ms after a previous call is disconnected:
pots distinctive-ring-guard-time 100Related Commands
pots encoding
To specify the pulse code modulation (PCM) encoding scheme for your connected telephones, fax machines, or modems, use the pots encoding command in global configuration mode. To disable the specified scheme, use the no form of this command.
pots encoding {alaw | ulaw}
no pots encoding {alaw | ulaw}
Syntax Description
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to Cisco 800 series routers.
Europe typically uses a-law. North America typically uses u-law.
Examples
The following example specifies a-law as the PCM encoding scheme:
pots encoding alawRelated Commands
pots forwarding-method
To configure the type of call-forwarding method to be used for Euro-ISDN (formerly NET3) switches, use the pots forwarding-method command in global configuration mode. To turn forwarding off, use the no form of this command.
pots forwarding-method {keypad | functional}
no pots forwarding-method {keypad | functional}
Syntax Description
keypad
Gives forwarding control to the Euro-ISDN switch.
functional
Gives forwarding control to the router. If you select this method, use the dual-tone multifrequency (DTMF) keypad commands listed in Table 29 to configure call-forwarding service.
Command Default
Forwarding is off
Command Modes
Global configuration
Command History
Usage Guidelines
Use this command to select the type of forwarding method to be used for Euro-ISDN switches. This command does not affect any other switch types.
You can select one or more call-forwarding services at a time, but keep the following Euro-ISDN switch characteristics in mind:
•
•
•
If all three call-forwarding services are enabled, CFU overrides CFB and CFNR. The default is that no call-forwarding service is selected.
If you select the functional forwarding method, use the DTMF keypad commands in Table 29 to configure the call-forwarding service.
Table 29 DTMF Keypad Commands for Call-Forwarding Service
Activate CFU
**21*number#
Deactivate CFU
#21#
Activate CFNR
**61*number#
Deactivate CFNR
#61#
Activate CFB
**67*number#
Deactivate CFB
#67#
1 Where number is the telephone number to which your calls are forwarded.
When you enable or disable the call-forwarding service, it is enabled or disabled for four basic services: speech, audio at 3.1 kilohertz (kHz), telephony at 3.1 kHz, and telephony at 7 kHz. You should hear a dial tone after you enter the DTMF keypad command when the call-forwarding service is successfully enabled for at least one of the four basic services. If you hear a busy tone, the command is invalid or the switch does not support that service.
Examples
The following example gives forwarding control to the router:
pots forwarding-method functionalRelated Commands
pots line-type
To specify the impedance of your connected telephones, fax machines, or modems, use the pots line-type command in global configuration mode. To disable the specified line type, use the no form of this command.
pots line-type {type1 | type2 | type3}
no pots line-type {type1 | type2 | type3}
Syntax Description
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example sets the line type to type1:
pots line-type type1Related Commands
pots prefix filter
To set a filter that prevents a dial prefix from being added to a dialed number when the digits in the dialed number match the filter, use the pots prefix filter command in global configuration mode. To remove the filter, use the no form of this command.
pots prefix filter number
no pots prefix filter number
Syntax Description
Command Default
No default filter is set.
Command Modes
Global configuration
Command History
Usage Guidelines
The pots prefix filter command is used to set a filter for prefix dialing. A maximum of ten filters can be set. Once the maximum number of filters have been configured, an additional filter is not accepted nor does it overwrite any of the existing filters.
To configure a new filter, remove at least one filter using the no pots prefix filter command.
You can set matching criteria for the filter using the * wildcard character. For example, if you configure the filter 1* and a dialed number starts with 1, the called number is not prefixed. Prefix filters can be of variable length. All configured prefix filters are compared to the number dialed, up to the length of the prefix filter. If there is a match, no prefix is added to the dialed number.
Examples
The following example configures five filters that prevent dial prefixes from being added to dialed numbers:
pots prefix filter 192pots prefix filter 1pots prefix filter 9pots prefix filter 0800pots prefix filter 08456With these filters configured, a prefix is not added to the following dialed numbers:
192 Directory calls
100 Operator services
999 Emergency services
0800... Toll-free calls
08456...Calls on an Energis network information controller
Related Commands
pots prefix number
To set a prefix to be added to a called telephone number for analog or modem calls, use the pots prefix number command in global configuration mode. To remove the prefix, use the no form of this command.
pots prefix number number
no pots prefix number number
Syntax Description
Command Default
No prefix is associated with the called number for analog or modem calls
Command Modes
Global configuration
Command History
Usage Guidelines
Only one prefix can be configured using this command. If a prefix already exists, the next prefix configured with this command overwrites the old prefix. Prefixes can be of variable length, up to five digits. The no pots prefix number command removes the prefix.
As numbers are dialed on the keypad, a comparison is made to the configured prefix filter. When a match is determined, the number is dialed without adding the prefix. In the unlikely event that the prefix filter has more digits than the dialed number, and the dialed number matches the first digits of the prefix filter, the prefix is not added to the dialed number. For example, if the prefix filter is 5554000 and you dial 555 and stop, the router considers the called number to be 555 and does not add a prefix to the number. This event is unlikely to occur because the number of digits in dialed numbers is typically greater than the number of digits in prefix filters.
Examples
The following example sets the prefix to 12345:
pots prefix number 12345This prefix is added to any number dialed for analog or modem calls that do not match the prefix filter.
Related Commands
pots prefix filter
Sets a filter that prevents a dial prefix from being added to a dialed number when the digits in the dialed number match the filter.
pots ringing-freq
To specify the frequency on the Cisco 800 series router at which connected telephones, fax machines, or modems ring, use the pots ringing-freq command in global configuration mode. To disable the specified frequency, use the no form of this command.
pots ringing-freq {20Hz | 25Hz | 50Hz}
no pots ringing-freq {20Hz | 25Hz | 50Hz}
Syntax Description
20Hz
Connected devices ring at 20 Hz.
25Hz
Connected devices ring at 25 Hz.
50Hz
Connected devices ring at 50 Hz.
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example sets the ringing frequency to 50 Hz:
pots ringing-freq 50HzRelated Commands
pots silence-time
To specify the interval of silence after a calling party disconnects, use the pots silence-time command in global configuration mode. To disable the specified silence time, use the no form of this command.
pots silence-time interval
no pots silence-time interval
Syntax Description
Command Default
The default depends on the setting of the pots country command. For more information, see the pots country command.
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to Cisco 800 series routers.
Examples
The following example sets the interval of silence to 10 seconds:
pots silence-time 10Related Commands
pots tone-source
To specify the source of dial, ringback, and busy tones for your connected telephones, fax machines, or modems, use the pots tone-source command in global configuration mode. To disable the specified source, use the no form of this command.
pots tone-source {local | remote}
no pots tone-source {local | remote}
Syntax Description
Command Default
Local (router supplies the tones)
Command Modes
Global configuration
Command History
Usage Guidelines
This command applies to Cisco 800 series routers.
This command applies only to ISDN lines connected to a EURO-ISDN (NET3) switch.
Examples
The following example sets the tone source to remote:
pots tone-source remoteRelated Commands
pre-dial delay
To configure a delay on an Foreign Exchange Office (FXO) interface between the beginning of the off-hook state and the initiation of dual-tone multifrequency (DTMF) signaling, use the pre-dial delay command in voice-port configuration mode. To reset to the default, use the no form of the command.
pre-dial delay seconds
no pre-dial delay
Syntax Description
Command Default
1 second
Command Modes
Voice-port configuration
Command History
11.(7)T
This command was introduced on the Cisco 3600 series.
12.0(2)T
This command was integrated into Cisco IOS Release 12.0(2)T.
Usage Guidelines
To disable the command, set the delay to 0. When an FXO interface begins to draw loop current (off-hook state), a delay is required between the initial flow of loop current and the beginning of signaling. Some devices initiate signaling too quickly, resulting in redial attempts. This command allows a signaling delay.
Examples
The following example sets a predial delay value of 3 seconds on the FXO port:
voice-port 1/0/0pre-dial delay 3Related Commands
timeouts initial
Configures the initial digit timeout value for a specified voice port.
timing delay-duration
Configures delay dial signal duration for a specified voice port.
preference (dial peer)
To indicate the preferred order of a dial peer within a hunt group, use the preference command in dial peer configuration mode. To remove the preference, use the no form of this command.
preference value
no preference
Syntax Description
value
Integer from 0 to 10, where the lower the number, the higher the preference. Default is 0 (highest preference).
Command Default
0 (highest preference)
Command Modes
Dial peer configuration
Command History
Usage Guidelines
This command applies to POTS, VoIP, VoFR, and VoATM dial peers.
Use this command to indicate the preference order for matching dial peers in a rotary group. Setting the preference enables the desired dial peer to be selected when multiple dial peers within a hunt group are matched for a dial string.
Note
Use this command with the Rotary Calling Pattern feature described in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2 chapter "Configuring H.323 Gateways."
The hunting algorithm precedence is configurable. For example, if you wish a call processing sequence to go to destination A first, to destination B second, and to destination C third, you would assign preference (0 being the highest priority) to the destinations in the following order:
•
•
•
Examples
The following example sets POTS dial peer 10 to a preference of 1, POTS dial peer 20 to a preference of 2, and VoFR dial peer 30 to a preference of 3:
dial-peer voice 10 potsdestination pattern 5550150preference 1exitdial-peer voice 20 potsdestination pattern 5550150preference 2exitdial-peer voice 30 vofrdestination pattern 5550150preference 3exitThe following examples show different dial peer configurations:
Dialpeer destpat preference session-target1 4085550148 0 (highest) jmmurphy-voip2 408555 0 sj-voip3 408555 1 (lower) backup-sj-voip4 .......... 1 0:D (interface)5 .......... 0 anywhere-voipIf the destination number is 4085550148, the order of attempts is 1, 2, 3, 5, 4:
Dialpeer destpat preference1 408555 02 4085550148 13 4085550 04 ..............4085550.........0If the number dialed is 4085550148, the order is 2, 3, 4, 1.
Note
Related Commands
preemption enable
To enable preemption capability on a trunk group, use the preemption enable command in trunk group configuration mode. To disable preemption capabilities, use the no form of this command.
preemption enable
no preemption enable
Syntax Description
This command has no arguments or keywords.
Command Default
Preemption is disabled on the trunk group.
Command Modes
Trunk group configuration
Command History
12.4(4)XC
This command was introduced.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
Examples
The following command example enables preemption capabilities on trunk group test:
Router(config)# trunk group testRouter(config-trunk-group)# preemption enableRelated Commands
preemption guard timer
To define the time for a DDR call and to allow time to clear the last call from the channel, use the preemption guard timer command in trunk group configuration mode. To disable the preemption guard time, use the no form of this command.
preemption guard timer value
no preemption guard timer
Syntax Description
value
Number, in milliseconds for the preemption guard timer. The range is 60 to 500. The default is 60.
Command Default
No preemption guard timer is configured.
Command Modes
Trunk group configuration
Command History
12.4(4)XC
This command was introduced.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
Examples
The following set of commands configures a 60-millisecond preemption guard timer on the trunk group dial2.
Router(config)# trunk group dial2Router(config-trunk-group)# preemption enableRouter(config-trunk-group)# preemption guard timer 60Related Commands
preemption level
To set the precedence for voice calls to be preempted by a dial-on demand routing (DDR) call for the trunk group, use the preemption level command in dial peer configuration mode. To restore the default preemption level setting, use the no form of this command
preemption level {flash-override | flash | immediate | priority | routine}
no preemption level
Syntax Description
Command Default
The preemption level default is routine (lowest).
Command Modes
Dial peer configuration
Command History
12.4(4)XC
This command was introduced.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
Examples
The following command example sets a preemption level of flash (level 1) on POTS dial-peer 20:
Router(config)# dial-peer voice 20 potsRouter(config-dial-peer)# preemption level flashRelated Commands
preemption tone timer
To set the expiry time for the preemption tone for the outgoing call being preempted by a DDR backup call, use the preemption tone timer command in trunk group configuration mode. To clear the expiry time, use the no form of this command.
preemption tone timer seconds
no preemption tone timer
Syntax Description
Command Default
No preemption tone timer is configured.
Command Modes
Trunk group configuration
Command History
12.4(4)XC
This command was introduced.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
Examples
The following set of commands configures a 20-second preemption tone timer on trunk group dial2.
Router(config)# trunk group dial2Router(config-trunk-group)# preemption enableRouter(config-trunk-group)# preemption tone timer 20Related Commands
prefix
To specify the prefix of the dialed digits for a dial peer, use the prefix command in dial peer configuration mode. To disable this feature, use the no form of this command.
prefix string
no prefix
Syntax Description
string
Integers that represent the prefix of the telephone number associated with the specified dial peer. Valid values are 0 through 9 and a comma (,). Use a comma to include a pause in the prefix.
Command Default
Null string
Command Modes
Dial peer configuration
Command History
Usage Guidelines
Use this 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 plain old telephone service (POTS) dial peers. This command applies to off-ramp store-and-forward fax functions.
Examples
The following example specifies a prefix of 9 and then a pause:
dial-peer voice 10 potsprefix 9,The following example specifies a prefix of 5120002:
Router(config-dial-peer)# prefix 5120002Related Commands
prefix (Annex G)
To restrict the prefixes for which the gatekeeper should query the Annex G border element (BE), use the prefix command in gatekeeper border element configuration mode.
prefix prefix* [seq | blast]
Syntax Description
prefix*
Prefix for which BEs should be queried.
seq
(Optional) Queries are sent out to the neighboring BEs sequentially.
blast
(Optional) Queries are sent out to the neighboring BEs simultaneously.
Command Default
Any time a remote zone query occurs, the BE is also queried.
Command Modes
Gatekeeper border element configuration
Command History
Usage Guidelines
By default, the gatekeeper sends all remote zone requests to the BE. Use this command only if you want to restrict the queries to the BE to a specific prefix or set of prefixes.
Examples
The following example directs the gatekeeper to query the BE using a prefix of 408.
Router(config-gk-annexg)# prefix 408* seqRelated Commands
h323-annexg
Enables the BE on the gatekeeper and enters border element configuration mode.
prefix (stcapp-fac)
To designate a prefix string to precede the dialing of SCCP telephony control (STC) feature access codes, use the prefix command in STC application feature access-code configuration mode. To return the prefix to its default, use the no form of this command.
prefix prefix-string
no prefix
Syntax Description
prefix-string
String of one to ten characters that can be dialed on a telephone keypad. String must start with * (asterisk) or # (pound sign). Default is **.
Command Default
The default prefix is ** (two asterisks).
Command Modes
STC application feature access-code configuration
Command History
Usage Guidelines
This command is used with the STC application, which enables certain features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control. Phone users dial the feature access code (FAC) prefix string before dialing a FAC that activates a feature. For example, to set call forwarding for all calls using the default prefix and FAC, a phone user dials **1.
Use this command only if you want to change the prefix from its default (**).
The show running-config command displays nondefault FACs and prefixes only. The show stcapp feature codes command displays all FACs and prefixes.
Examples
The following example sets a FAC prefix of two pound signs (##). After this value is configured, a phone user dials ##2 on the keypad to forward all calls for that extension.
Router(config)# stcapp feature access-codeRouter(stcapp-fac)# prefix ##Router(stcapp-fac)# call forward all 2Router(stcapp-fac)# call forward cancel 3Router(stcapp-fac)# pickup local 6Router(stcapp-fac)# pickup group 5Router(stcapp-fac)# pickup direct 4Router(stcapp-fac)# exitRelated Commands
prefix (stcapp-fsd)
To designate a prefix string to precede the dialing of SCCP telephony control (STC) application feature speed-dial codes, use the prefix command in STC application feature speed-dial configuration mode. To return the prefix to its default, use the no form of this command.
prefix prefix-string
no prefix
Syntax Description
prefix-string
String of one to ten characters that can be dialed on a telephone keypad. String must start with * (asterisk) or # (pound sign). Default is *.
Command Default
The default prefix is * (one asterisk).
Command Modes
STC application feature speed-dial configuration
Command History
Usage Guidelines
This command is used with the STC application, which enables certain features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control. Phone users dial the feature speed-dial (FSD) prefix string before dialing an FSD code that dials a telephone number. For example, to dial the telephone number that is stored in speed-dial position 2, a phone user dials *2.
Use this command only if you want to change the prefix from its default (*).
The show running-config command displays nondefault FSDs and prefixes only. The show stcapp feature codes command displays all feature speed-dial FSDs and prefixes.
Examples
The following example sets an FSD prefix of three asterisks (***). After this value is configured, a phone user presses ***2 on the keypad to dial speed-dial number 2.
Router(config)# stcapp feature speed-dialRouter(stcapp-fsd)# prefix ***Router(stcapp-fsd)# speed dial from 2 to 7Router(stcapp-fsd)# redial 9Router(stcapp-fsd)# voicemail 8Router(stcapp-fsd)# exitRelated Commands
presence
To enable presence service and enter presence configuration mode, use the presence command in global configuration mode. To disable presence service, use the no form of this command.
presence
no presence
Syntax Description
This command has no arguments or keywords.
Command Default
Presence service is disabled.
Command Modes
Global configuration
Command History
Usage Guidelines
This command enables the router to perform the following presence functions:
•
•
•
Examples
The following example shows how to enable presence and enter presence configuration mode to set the maximum subscriptions to 150:
Router(config)# presenceRouter(config-presence)# max-subscription 150Related Commands
presence call-list
To enable Busy Lamp Field (BLF) monitoring for call lists and directories on phones registered to the Cisco Unified CME router, use the presence call-list command in ephone, presence, or voice register pool configuration mode. To disable BLF indicators for call lists, use the no form of this command.
presence call-list
no presence call-list
Syntax Description
This command has no arguments or keywords.
Command Default
BLF monitoring for call lists is disabled.
Command Modes
Ephone configuration
Presence configuration
Voice register pool configurationCommand History
12.4(11)XJ
Cisco Unified CME 4.1
This command was introduced.
Usage Guidelines
This command enables a phone to monitor the line status of directory numbers listed in a directory or call list, such as a missed calls, placed calls, or received calls list. Using this command in presence mode enables the BLF call-list feature for all phones. To enable the feature for an individual SCCP phone, use this command in ephone configuration mode. To enable the feature for an individual SIP phone, use this command in voice register pool configuration mode.
If this command is disabled globally and enabled in voice register pool or ephone configuration mode, the feature is enabled for that voice register pool or ephone.
If this command is enabled globally, the feature is enabled for all voice register pools and ephones regardless of whether it is enabled or disabled on a specific voice register pool or ephone.
To display a BLF status indicator, the directory number associated with a telephone number or extension must have presence enabled with the allow watch command.
For information on the BLF status indicators that display on specific types of phones, see the Cisco Unified IP Phone documentation for your phone model.
Examples
The following example shows the BLF call-list feature enabled for ephone 1. The line status of a directory number that appears in a call list or directory is displayed on phone 1 if the directory number has presence enabled.
Router(config)# ephone 1Router(config-ephone)# presence call-listRelated Commands
presence enable
To allow incoming presence requests, use the presence enable command in SIP UA configuration mode. To block incoming requests, use the no form of this command.
presence enable
no presence enable
Syntax Description
This command has no arguments or keywords.
Command Default
Incoming presence requests are blocked.
Command Modes
SIP UA configuration
Command History
Usage Guidelines
This command allows the router to accept incoming presence requests (SUBSCRIBE messages) from internal watchers and SIP trunks. It does not impact outgoing presence requests.
Examples
The following example shows how to allow incoming presence requests:
Router(config)# sip-uaRouter(config-sip-ua)# presence enableRelated Commands
pri-group (pri-slt)
To specify an ISDN PRI on a channelized T1 or E1 controller, use the pri-group (pri-slt) command in controller configuration mode. To remove the ISDN PRI configuration, use the no form of this command.
pri-group [timeslots timeslot-range [nfas_d [backup | none | primary [nfas_int number]] [nfas-group number [iua as-name]]]
no pri-group
Syntax Description
Command Default
No ISDN-PRI group is configured.
Command Modes
Controller configuration
Command History
Usage Guidelines
The pri-group (pri-slt) command provides another way to bind a D channel to a specific IUA AS. This option allows the RLM group to be configured at the pri-group level instead of in the D channel configuration. For example, a typical configuration would look like the following:
controller t1 1/0/0pri-group timeslots 1-24 nfas_d pri nfas_int 0 nfas_group 1 iua asnameBefore you enter the pri-group command, you must specify an ISDN-PRI switch type and an E1 or T1 controller.
When configuring NFAS, you use an extended version of the pri-group command to specify the following values for the associated channelized T1 controllers configured for ISDN:
•
•
•
The iua keyword is used to bind an NFAS group to the IUA AS.
When binding the D channel to an IUA AS, the as-name must match the name of an AS set up during IUA configuration.
Before you can modify a PRI group on a Media Gateway Controller (MGC), you must first shut down the D channel.
The following shows how to shut down the D channel:
Router# configure terminalEnter configuration commands, one per line. End with CNTL/Z.Router(config)# interface Dchannel3/0:1Router(config-if)# shutdownExamples
The following example configures the NFAS primary D channel on one channelized T1 controller, and binds the D channel to an IUA AS. This example uses the Cisco AS5400 and applies to T1, which has 24 timeslots and is used mainly in North America and Japan:
Router(config-controller)# pri-group timeslots 1-23 nfas-d primary nfas-int 0 nfas-group 1 iua as5400-4-1The following example applies to E1, which has 32 timeslots and is used by the rest of the world:
Router(config-controller)# pri-group timeslots 1-31 nfas-d primary nfas-int 0 nfas-group 1 iua as5400-4-1The following example configures ISDN-PRI on all time slots of controller E1:
Router(config)# controller E1 4/1Router(config-controller)# pri-group timeslots 1-7,16In the following example, the rlm-timeslot keyword automatically creates interface serial 4/7:11 (4/7:0:11 if you are using the CT3 card) for the D channel object on a Cisco AS5350. You can choose any timeslot other than 24 to be the virtual container for the D channel parameters for ISDN.
Router(config-controller)# pri-group timeslots 1-23 nfas-d primary nfas-int 0 nfas-group 0 rlm-timeslot 3Related Commands
isdn switch-type
Configures the Cisco 2600 series router PRI interface to support QSIG signaling.
pri-group nec-fusion
To configure your NEC PBX to support Fusion Call Control Signaling (FCCS), use the pri-group nec-fusion command in controller configuration mode. To disable FCCS, use the no form of this command.
pri-group nec-fusion {pbx-ip-address | pbx-ip-host-name} pbx-port number
no pri-group nec-fusion {pbx-ip-address | pbx-ip-host-name} pbx-port number
Syntax Description
Command Default
PBX port number: 55000
Command Modes
Controller configuration
Command History
12.0(7)T
This command was introduced on the Cisco AS5300.
12.2(1)
This command was modified to add support for setup messages from a POTS dial peer.
Usage Guidelines
This command is used only if the PBX in your configuration is an NEC PBX, and if you are configuring it to run FCCS and not QSIG signaling.
Examples
The following example directs this NEC PBX to use FCCS:
pri-group nec-fusion 172.31.255.255 pbx-port 60000Related Commands
pri-group timeslots
To specify an ISDN PRI group on a channelized T1 or E1 controller, and to release the ISDN PRI signaling time slot, use the pri-group timeslots command in controller configuration mode. To remove or change the ISDN PRI configuration, use the no form of this command.
pri-group timeslots timeslot-range [nfas_d {backup | none | primary {nfas_int number | nfas_group number | rlm-group number}} | service]
no pri-group timeslots timeslot-range [nfas_d {backup | none | primary {nfas_int number | nfas_group number | rlm-group number}} | service]
Syntax Description
Command Default
No ISDN PRI group is configured. The switch type is automatically set to the National ISDN switch type (primary-ni keyword) when the pri-group timeslots command is configured with the rlm-group subkeyword.
Command Modes
Controller configuration
Command History
Usage Guidelines
The pri-group command supports the use of DS0 time slots for Signaling System 7 (SS7) links, and therefore the coexistence of SS7 links and PRI voice and data bearer channels on the same T1 or E1 span. In these configurations, the command applies to voice applications.
In SS7-enabled Voice over IP (VoIP) configurations when an RLM group is configured, High-Level Data Link Control (HDLC) resources allocated for ISDN signaling on a digital subscriber line (DSL) interface are released and the signaling slot is converted to a bearer channel (B24). The D channel will be running on IP. The chosen D-channel time slot can still be used as a B channel by using the isdn rlm-group interface configuration command to configure the NFAS groups.
NFAS allows a single D channel to control multiple PRI interfaces. Use of a single D channel to control multiple PRI interfaces frees one B channel on each interface to carry other traffic. A backup D channel can also be configured for use when the primary NFAS D channel fails. When a backup D channel is configured, any hard system failure causes a switchover to the backup D channel and currently connected calls remain connected.
NFAS is supported only with a channelized T1 controller and, as a result, must be ISDN PRI capable. Once the channelized T1 controllers are configured for ISDN PRI, only the NFAS primary D channel must be configured; its configuration is distributed to all members of the associated NFAS group. Any configuration changes made to the primary D channel will be propagated to all NFAS group members. The primary D channel interface is the only interface shown after the configuration is written to memory.
The channelized T1 controllers on the router must also be configured for ISDN. The router must connect to either an AT&T 4ESS, Northern Telecom DMS-100 or DMS-250, or National ISDN switch type.
The ISDN switch must be provisioned for NFAS. The primary and backup D channels should be configured on separate T1 controllers. The primary, backup, and B-channel members on the respective controllers should be the same configuration as that configured on the router and ISDN switch. The interface ID assigned to the controllers must match that of the ISDN switch.
You can disable a specified channel or an entire PRI interface, thereby taking it out of service or placing it into one of the other states that is passed in to the switch using the isdn service interface configuration command.
In the event that a controller belonging to an NFAS group is shut down, all active calls on the controller that is shut down will be cleared (regardless of whether the controller is set to primary, backup, or none), and one of the following events will occur:
•
•
•
•
The expected behavior in NFAS when an ISDN D channel (serial interface) is shut down is that ISDN Layer 2 should go down but keep ISDN Layer 1 up, and that the entire interface will go down after the amount of seconds specified for timer T309.
Note
Examples
The following example configures T1 controller 1/0 for PRI and for the NFAS primary D channel. This primary D channel controls all the B channels in NFAS group 1.
controller t1 1/0framing esflinecode b8zspri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 1The following example specifies ISDN PRI on T1 slot 1, port 0, and configures voice and data bearer capability on time slots 2 through 6:
isdn switch-type primary-4esscontroller t1 1/0framing esflinecode b8zspri-group timeslots 2-6The following example configures a standard ISDN PRI interface:
! Standard PRI configuration:controller t1 1pri-group timeslots 1-23 nfas_d primary nfas_int 0 nfas_group 0exit! Standard ISDN serial configuration:interface serial1:23! Set ISDN parameters:isdn T309 4000exitThe following example configures a dedicated T1 link for SS7-enabled VoIP:
controller T1 1pri-group timeslots 1-23 nfas_d primary nfas_int 0 nfas_group 0exit! In a dedicated configuration, we assume the 24th timeslot will be used by ISDN.! Serial interface 0:23 is created for configuring ISDN parameters.interface Serial:24! The D channel is on the RLM.isdn rlm 0isdn T309 4000exitThe following example configures a shared T1 link for SS7-enabled VoIP. The rlm-group 0 portion of the pri-group timeslots command releases the ISDN PRI signaling channel.
controller T1 1pri-group timeslots 1-3 nfas_d primary nfas_int 0 nfas_group 0 rlm-group 0channel group 23 timeslot 24end! D-channel interface is created for configuration of ISDN parameters:interface Dchannel1isdn T309 4000endRelated Commands
privacy
To set privacy support as defined in RFC 3323, use the privacy command in voice-service voip-sip configuration mode or on a dial peer. To remove privacy support as defined in RFC 3323, use the no form of this command.
privacy [[[header | session | user | id] | critical] | pstn | system | disable]
no privacy
Syntax Description
Command Default
The command is disabled.
Command Modes
Voice-service sip configuration (conf-serv-sip)
Dial-peer configuration (config-dial-peer)Command History
Usage Guidelines
Use the privacy command to instruct the gateway to add a Proxy-Require header set to a value supported by RFC 3323 in outgoing SIP request messages.
Use the privacy critical command to instruct the gateway to add a Proxy-Require header with the value set to critical. In the unlikely event a user agent sends a request to an intermediary that does not support privacy extensions, the request fails.
Examples
The following example shows how to set the privacy to PSTN.
Router> enableRouter# configure terminalRouter(config)# voice service voipRouter(conf-voi-serv)# sipRouter(conf-serv-sip)# privacy pstnRelated Commands
progress_ind
To set a specific progress indicator (PI) in call setup, progress, or connect messages from an H.323 VoIP gateway, use the progress_ind command in dial peer configuration mode. To reset to the default, use the no or disable forms of this command.
progress_ind {setup | connect | progress | alert} {enable pi-number | disable}
no progress_ind {setup | connect | progress | alert}
Note
Syntax Description
Command Default
The default progress indicator from the switch is not intercepted or modified.
Command Modes
Dial peer configuration
Command History
Usage Guidelines
This command overrides the default progress indicator that is sent by the switch. This enables you to set the progress indicator at the H.323 gateway, if necessary, to ensure the proper end-to-end signaling for VoIP calls. This command sets the progress indicator only in messages from outbound dial peers that have a set destination pattern, configured by using the destination-pattern command. If a message contains multiple progress indicators, the progress_ind command overrides only the first progress indicator in the message.
The disable and no forms of the progress_ind command have the same result: The call messages are not intercepted by the session application, and the default progress indicator, if any, is forwarded unmodified.
Note
Examples
The following example sets the progress indicator to 1 in progress and connect messages from POTS dial peer 3:
dial-peer voice 3 potsdestination-pattern 55275progress_ind progress enable 1progress_ind connect enable 1Related Commands
protocol rlm port
To configure the RLM port number, use the protocol rlm port RLM configuration command. To disable this function, use the no form of this command.
protocol rlm port port-number
no protocol rlm port port-number
Syntax Description
port-number
RLM port number. See Table 30 for the port number choices.
Command Default
3000
Command Modes
RLM configuration
Command History
Usage Guidelines
The port number for the basic RLM connection can be reconfigured for the entire RLM group. Table 30 lists the default RLM port numbers.
Related Commands
proxy h323
To enable the proxy feature on your router, use the proxy h323 command in global configuration mode. To disable the proxy feature, use the no form of this command.
proxy h323
no proxy h323
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
Global configuration
Command History
11.3(2)NA
This command was introduced on the Cisco 2500 series and Cisco 3600 series.
Usage Guidelines
If the multimedia interface is not enabled using this command or if no gatekeeper is available, starting the proxy allows it to attempt to locate these resources. No calls are accepted until the multimedia interface and the gatekeeper are found.
Examples
The following example turns on the proxy feature:
proxy h323
