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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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Table Of Contents
signal timing idle suppress-voice
signal timing oos slave-standby
signal timing oos suppress-all
signal timing oos suppress-voice
snmp enable peer-trap poor-qov
signal
To specify the type of signaling for a voice port, use the signal command in voice-port configuration mode. To reset to the default, use the no form of this command.
Foreign Exchange Office (FXO) and Foreign Exchange Station (FXS) Voice Ports
signal {groundstart | loopstart [live-feed]}
no signal {groundstart | loopstart}
Ear and mouth (E&M) Voice Ports
signal {delay-dial | immediate | lmr | wink-start}
no signal {delay-dial | immediate | lmr | wink-start}
Centralized Automatic Message Accounting (CAMA) Ports
signal {cama {kp-0-nxx-xxxx-st | kp-0-npa-nxx-xxxx-st | kp-2-st | kp-npd-nxx-xxxx-st | kp-0-npa-nxx-xxxx-st-kp-yyy-yyy-yyyy-st} | groundstart | loopstart}
no signal {cama {kp-0-nxx-xxxx-st | kp-0-npa-nxx-xxxx-st | kp-2-st | kp-npd-nxx-xxxx-st | kp-0-npa-nxx-xxxx-st-kp-yyy-yyy-yyyy-st} | groundstart | loopstart}
Syntax Description
Command Default
FXO and FXS interfaces: loopstart
E&M interfaces: wink-start
CAMA interfaces: loopstartCommand Modes
Voice-port configuration (config-voiceport)
Command History
Usage Guidelines
This command applies to analog voice ports only. A voice port must be shut down and then activated before the configured values take effect.
For an E&M voice port, this command changes only the signal value for the selected voice port.
For an FXO or FXS voice port, this command changes the signal value for both voice ports on a voice port module (VPM). If you change the signal type for an FXO voice port on Cisco 3600 series routers, you need to move the appropriate jumper in the voice interface card of the voice network module. For more information about the physical characteristics of the voice network module, see the installation documentation that came with your voice network module.
Some PBXs miss initial digits if the E&M voice port is configured for immediate start signaling. Immediate start signaling should be used for dial pulse outpulsing only and only on circuits for which the far end is configured to accept digits within a few milliseconds of seizure. Delay dial signaling, which is intended for use on trunks and not lines, relies on the far end to return an off-hook indication on its M-lead as soon as the circuit is seized. When a receiver is attached, the far end removes the off-hook indication to indicate that it is ready to receive digits. Delay dial must be configured on both ends to work properly. 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.
To specify which VIC-2CAMA ports are designated as dedicated CAMA ports for emergency 911 calls, use the signal cama command. No two service areas in the existing North American telephony infrastructure supporting E911 calls have identical service implementations, and many of the factors that drive the design of emergency call handling are matters of local policy and therefore outside the scope of this document. Local policy determines which ANI format is appropriate for the specified Physical Service Access Point (PSAP) location.
The following four types of ANI transmittal schemes are based on the actual number of digits transmitted toward the E911 tandem. In each instance, the actual calling number is proceeded with a key pulse (KP) followed by an information (I) field or a NPD, which is then followed by the ANI calling number, and finally is followed by a start pulse (ST), STP, ST2P, or ST3P, depending on the trunk group type in the PSTN and the traffic mix carried.
The information field is one or two digits, depending on how the circuit was ordered originally. For one-digit information fields, a value of 0 indicates that the calling number is available. A value of 1 indicates that the calling number is not available. A value of 2 indicates an ANI failure. For a complete list of values for two-digit information fields, see SR-2275: Telcordia Notes on the Networks at www.telcordia.com.
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7-digit transmission (kp-0-nxx-xxxx-st):
The calling phone number is transmitted, and the NPA is implied by the trunk group and not transmitted.
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The I field consists of single-digit NPD-to-NPA mapping. When the calling party number of 415-555-0122 places a 911 call, and the Cisco 2600 series or Cisco 3600 series has an NPD (0)-to-NPA (415) mapping, the NPA signaling format is received by the selective router at the central office (CO).
Note
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The E.164 number is fully transmitted.
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Twenty digits support (two 10 digit numbers) on FGD-OS in the following format, KP+II+10 digit ANI+ST+KP+7/10 digit PANI+ ST
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kp-2-st transmission is used if the PBX is unable to out-pulse the ANI. If the ANI received by the Cisco router is not as per configured values, kp-2-st is transmitted. For example, if the voice port is configured for out-pulsing a ten-digit ANI and the 911 call it receives has a seven-digit calling party number, the router transmits kp-2-st.
Note
Examples
The following example configures groundstart signaling on the Cisco 3600 series as the signaling type for a voice port, which means that both sides of a connection can place a call and hang up:
voice-port 1/1/1signal groundstartThe following example configures a ten-digit ANI transmission:
Router(config)# voice-port 1/0/0Router(config-voiceport)# signal cama kp-0-npa-nxx-xxxx-stThe following example configures 20-digit CAMA Signaling with ANI/Pseudo ANI:
Router(config-voiceport)# signal cama KP-0-NPA-NXX-XXXX-ST-KP-YYY-YYY-YYYY-STRelated Commands
signal did
To enable direct inward dialing (DID) on a voice port, use the signal did command in voice-port configuration mode. To disable DID and reset to loop-start signaling, use the no form of this command.
signal did {immediate-start | wink-start | delay-start}
no signal did
Syntax Description
immediate-start
Enables immediate-start signaling on the DID voice port.
wink-start
Enables wink-start signaling on the DID voice port.
delay-start
Enables delay-dial signaling on the DID voice port.
Command Default
No default behavior or values
Command Modes
Voice-port configuration
Command History
Examples
The following example configures a voice port with immediate-start signaling enabled:
Router# voice-port 1/17Router (config-voiceport)# signal did immediate-startsignal keepalive
To configure the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks, use the signal keepalive command in voice-class configuration mode. To reset to the default, use the no form of this command.
signal keepalive {seconds | disabled}
no signal keepalive {seconds | disabled}
Syntax Description
seconds
Keepalive signaling packet interval, in seconds. Range is from 1 to 65535. Default is 5 seconds.
disabled
Specifies that no keepalive signals are sent.
Command Default
seconds: 5 seconds
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring the keepalive signaling interval, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. The voice class must then be assigned to a dial peer using the voice-class permanent (dial peer) command.
To avoid sending keepalive signals to a multicasting network with no specified destination, we recommend that you use the disabled keyword when configuring this command for use in networks that use connection trunk connections and multicasting.
Examples
The following example shows the keepalive signaling interval set to 3 seconds for voice class 10:
voice class permanent 10signal keepalive 3exitdial-peer voice 100 vofrvoice-class permanent 10Related Commands
signal pattern
To define the ABCD bit patterns that identify the idle and out-of-service (OOS) states for Cisco trunks and FRF.11 trunks, use the signal pattern command in voice-class configuration mode. To remove the patterns from the voice class, use the no form of this command.
signal pattern {idle receive | idle transmit | oos receive | oos transmit} bit-pattern
no signal pattern {idle receive | idle transmit | oos receive | oos transmit} bit-pattern
Syntax Description
Command Default
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring the signaling pattern, you must use the voice-class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you define the voice class, you assign it to a dial peer.
Idle Patterns
An idle state is generated if the router detects an idle signaling pattern coming from either direction. If an idle pattern is configured for only one direction (transmit or receive), an idle state can be detected only in the configured direction. Therefore, you should normally enter both the idle receive and the idle transmit keywords.
To suppress voice packets whenever the transmit or receive trunk is in the idle state, use the idle receive and idle transmit keywords in conjunction with the signal timing idle suppress-voice command.
OOS Patterns
An OOS state is generated differently in each direction under the following conditions:
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To suppress voice packets whenever the transmit or receive trunk is in the OOS state, use the oos receive and oos transmit keywords in conjunction with the signal timing oos suppress-voice command.
To suppress voice and signaling packets whenever the transmit or receive trunk is in the OOS state, use the oos receive and oos transmit keywords in conjunction with the signal timing oos suppress-all command.
PBX Busyout
To "busy out" a PBX if the network connection fails, set the oos receive pattern to match the seized state (busy), and set the signal timing oos timeout value. When the timeout value expires and no signaling packets are received, the router sends the oos receive pattern to the PBX.
Use the busy seized pattern only if the PBX does not have a specified pattern for indicating an OOS state. If the PBX has a specific OOS pattern, use that pattern instead.
Examples
The following example, beginning in global configuration mode, configures the signaling bit pattern for the idle receive and transmit states:
voice class permanent 10 signal keepalive 3 signal pattern idle receive 0101 signal pattern idle transmit 0101 exitdial-peer voice 100 vofrvoice-class permanent 10The following example, beginning in global configuration mode, configures the signaling bit pattern for the out-of-service receive and transmit states:
voice class permanent 10 signal keepalive 3 signal pattern oos receive 0001 signal pattern oos transmit 0001exitdial-peer voice 100 vofrvoice-class permanent 10The following example restores default signaling bit patterns for the receive and transmit idle states:
voice class permanent 10 signal keepalive 3 signal timing idle suppress-voice no signal pattern idle receive no signal pattern idle transmit exitdial-peer voice 100 vofrvoice-class permanent 10The following example configures nondefault signaling bit patterns for the receive and transmit out-of-service states:
voice class permanent 10 signal keepalive 3 signal pattern oos receive 0001 signal pattern oos transmit 0001exitdial-peer voice 100 vofrvoice-class permanent 10Related Commands
signal sequence oos
To specify which signaling pattern is sent to the PBX when the far-end keepalive message is lost or an alarm indication signal (AIS) is received from the far end, use the signal sequence oos command in voice-class configuration mode. To reset to the default, use the no form of this command.
signal sequence oos {no-action | idle-only | oos-only | both}
no signal sequence oos
Syntax Description
Command Default
Both idle and OOS signaling patterns are sent.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring the idle or OOS signaling patterns to be sent, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you assign it to a dial peer.
Use the signal sequence oos command to specify which signaling pattern) to send. Use the signal pattern idle receive or the signal pattern oos receive command to define the bit patterns of the signaling patterns if other than the defaults.
Examples
The following example, beginning in global configuration mode, defines voice class 10, sets the signal sequence oos command to send only the idle signal pattern to the PBX, and applies the voice class configuration to VoFR dial peer 100.
voice-class permanent 10signal-keepalive 3signal sequence oos idle-onlysignal timing idle suppress-voiceexitdial-peer voice 100 vofrvoice-class permanent 10signal-type transparentRelated Commands
signal timing idle suppress-voice
To configure the signal timing parameter for the idle state of a call, use the signal timing idle suppress-voice command in voice-class configuration mode. To reset to the default, use the no form of this command.
signal timing idle suppress-voice seconds
no signal timing idle suppress-voice seconds
Syntax Description
seconds
Duration of the idle state, in seconds, before the voice traffic is stopped. Range is from 0 to 65535.
Command Default
No signal timing idle suppress-voice timer is configured.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring the signal timing idle suppress-voice timer, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. The voice class must then be assigned to a dial peer.
The signal timing idle suppress-voice command is used when the signal-type command is set to transparent in the dial peer for the Cisco trunk or FRF.11 trunk connection. The router stops sending voice packets when the timer expires. Signaling packets are still sent.
To detect an idle trunk state, the router or concentrator monitors both transmit and receive signaling for the idle transmit and idle receive signaling patterns. These can be configured by the signal pattern idle transmit or signal pattern idle receive command, or they can be the defaults. The default idle receive pattern is the idle pattern of the local voice port. The default idle transmit pattern is the idle pattern of the far-end voice port.
Examples
The following example, beginning in global configuration mode, sets the signal timing idle suppress-voice timer to 5 seconds for the idle state on voice class 10.
voice class permanent 10signal keepalive 3 signal pattern idle receive 0101 signal pattern idle transmit 0101 signal timing idle suppress-voice 5exitdial-peer voice 100 vofrvoice-class permanent 10signal-type transparentThe following example defines voice class 10, sets the idle detection time to 5 seconds, configures the trunk to use the default transmit and receive idle signal patterns, and applies the voice class configuration to VoFR dial peer 100.
voice class permanent 10signal keepalive 3 signal timing idle suppress-voice 5exitdial-peer voice 100 vofrvoice-class permanent 10signal-type transparentRelated Commands
signal timing oos
To configure the signal timing parameter for the out-of-service (OOS) state of the call, use the signal timing oos command in voice-class configuration mode. To reset to the default, use the no form of this command.
signal timing oos {restart | slave-standby | suppress-all | suppress-voice | timeout} seconds
no signal timing oos {restart | slave-standby | suppress-all | suppress-voice | timeout} seconds
Syntax Description
Command Default
No signal timing OOS pattern parameters are configured.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. The voice class must then be assigned to a dial peer.
You can enter several values for this command. However, the suppress-all and suppress-voice options are mutually exclusive.
Examples
The following example, beginning in global configuration mode, configures the signal timeout parameter for the OOS state on voice class 10. The signal timing oos timeout command is set to 60 seconds.
voice-class permanent 10signal-keepalive 3 signal pattern oos receive 0001 signal pattern oos transmit 0001 signal timing oos timeout 60exitdial-peer voice 100 vofrvoice-class permanent 10Related Commands
signal timing oos restart
To specify that a permanent voice connection be torn down and restarted after the trunk has been out-of-service (OOS) for a specified time, use the signal timing oos restart command in voice-class configuration mode. To reset to the default, use the no form of this command.
signal timing oos restart seconds
no signal timing oos restart
Syntax Description
seconds
Delay duration, in seconds, for the restart attempt. Range is from 0 to 65535. There is no default.
Command Default
No restart attempt is made if the trunk becomes OOS.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. You then assign the voice class to a dial peer.
The signal timing oos restart command is valid only if the signal timing oos timeout command is enabled, which controls the start time for the OOS state. The timer for the signal timing oos restart command does not start until the trunk is OOS.
Examples
The following example, beginning in global configuration mode, creates voice class 10, sets the OOS timeout time to 60 seconds and sets the restart time to 30 seconds:
voice-class permanent 10signal-keepalive 3 signal pattern oos receive 0001 signal pattern oos transmit 0001 signal timing oos timeout 60signal timing oos restart 30exitdial-peer voice 100 vofrvoice-class permanent 10Related Commands
signal timing oos slave-standby
To configure a slave port to return to its initial standby state after the trunk has been out-of-service (OOS) for a specified time, use the signal timing oos slave-standby command in voice-class configuration mode. To reset to the default, use the no form of this command.
signal timing oos slave-standby seconds
no signal timing oos slave-standby
Syntax Description
seconds
Delay duration, in seconds. If no signaling packets are received for this period, the slave port returns to its initial standby state. Range is from 0 to 65535. There is no default.
Command Default
The slave port does not return to its standby state if the trunk becomes OOS.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you assign it to a dial peer.
If no signaling packets are received for the specified delay period, the slave port returns to its initial standby state. The signal timing oos slave-standby command is valid only if both of the following conditions are true:
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Examples
The following example, beginning in global configuration mode, creates a voice port as a slave voice port, creates voice class 10, sets the OOS timeout time to 60 seconds, and sets the return-to-slave-standby time to 120 seconds:
voice-port 1/0/0connection trunk 5559262 answer-modeexitvoice-class permanent 10signal-keepalive 3 signal pattern oos receive 0001 signal pattern oos transmit 0001 signal timing oos timeout 60signal timing oos slave-standby 120exitdial-peer voice 100 vofrvoice-class permanent 10Related Commands
signal timing oos suppress-all
To configure the router or concentrator to stop sending voice and signaling packets to the network if it detects a transmit out-of-service (OOS) signaling pattern from the PBX for a specified time, use the signal timing oos suppress-all command in voice-class configuration mode. To reset to the default, use the no form of this command.
signal timing oos suppress-all seconds
no signal timing oos suppress-all
Syntax Description
seconds
Delay duration, in seconds, before packet transmission is stopped. Range is from 0 to 65535. There is no default.
Command Default
The router or concentrator does not stop sending packets to the network if it detects a transmit OOS signaling pattern from the PBX.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you assign it to a dial peer.
The signal timing oos suppress-all command is valid only if you configure an OOS transmit signaling pattern with the signal pattern oos transmit command. (There is no default oos transmit signaling pattern.)
The signal timing oos suppress-all command is valid whether or not the signal timing oos timeout command is enabled, which controls the start time for the OOS state. The timer for the signal timing oos suppress-all command starts immediately when the OOS transmit signaling pattern is matched.
Examples
The following example, beginning in global configuration mode, creates voice class 10, sets the OOS timeout time to 60 seconds, and sets the packet suppression time to 60 seconds:
voice-class permanent 10signal-keepalive 3 signal pattern oos receive 0001 signal pattern oos transmit 0001 signal timing oos timeout 60signal timing oos suppress-all 60exitdial-peer voice 100 vofrvoice-class permanent 10Related Commands
signal timing oos suppress-voice
To configure the router or concentrator to stop sending voice packets to the network if it detects a transmit out-of-service (OOS) signaling pattern from the PBX for a specified time, use the signal timing oos suppress-voice command in voice-class configuration mode. To reset to the default, use the no form of this command.
signal timing oos suppress-voice seconds
no signal timing oos suppress-voice
Syntax Description
seconds
Delay duration, in seconds, before voice-packet transmission is stopped. Range is from 0 to 65535. There is no default.
Command Default
The router or concentrator does not stop sending voice packets to the network if it detects a transmit OOS signaling pattern from the PBX.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you assign it to a dial peer.
The signal timing oos suppress-voice command is valid only if you configure an OOS transmit signaling pattern with the signal pattern oos transmit command. (There is no default oos transmit signaling pattern.)
The signal timing oos suppress-voice s command is valid whether or not the signal timing oos timeout command is enabled, which controls the start time for the OOS state. The timer for the signal timing oos suppress-voice command starts immediately when the OOS transmit signaling pattern is matched.
Examples
The following example, beginning in global configuration mode, creates voice class 10, sets the OOS timeout time to 60 seconds, and sets the packet suppression time to 60 seconds:
voice-class permanent 10signal-keepalive 3 signal pattern oos receive 0001 signal pattern oos transmit 0001 signal timing oos timeout 60signal timing oos suppress-voice 60exitdial-peer voice 100 vofrvoice-class permanent 10Related Commands
signal timing oos timeout
To change the delay time between the loss of signaling packets from the network and the start time for the out-of-service (OOS) state, use the signal timing oos timeout command in voice-class configuration mode. To reset to the default, use the no form of this command.
signal timing oos timeout [seconds | disabled]
no signal timing oos timeout
Syntax Description
Command Default
No signal timing OOS pattern parameters are configured.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring signal timing OOS parameters, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you assign it to a dial peer.
You can use the signal timing oos timeout command to enable busyout to the PBX.
The signal timing oos timeout command controls the starting time for the signal timing oos restart and signal timing oos slave-standby commands. If this command is entered with the disabled keyword, the signal timing oos restart and signal timing oos slave-standby commands are ineffective.
Examples
The following example, beginning in global configuration mode, creates voice class 10 and sets the OOS timeout time to 60 seconds:
voice-class permanent 10signal-keepalive 3 signal pattern oos receive 0001 signal pattern oos transmit 0001 signal timing oos timeout 60exitdial-peer voice 100 vofrvoice-class permanent 10Related Commands
signaling forward
To enable a Cisco IOS gateway to forward the Generic Transparency Descriptor (GTD) payload to another gateway or gatekeeper system-wide, use the signaling forward command in global configuration mode. To disable forwarding, use the no form of this command.
signaling forward {conditional | unconditional | none}
no signaling forward
Syntax Description
Command Default
Signaling forwarding is conditional.
Command Modes
Global configuration
Command History
Usage Guidelines
This command is used with the Cisco PGW 2200 in the Cisco SS7 Interconnect for Voice Gateways solution. You must configure the Cisco PGW 2200 to encapsulate SS7 ISUP messages in GTD format before using this command on the Cisco gateway.
If the target is a RAS target, for a non-GTD signaling payload, the original payload is forwarded. For a GTD signaling payload, the payload is encapsulated in an admission request (ARQ)/disengage request (DRQ) message and sent to the originating gatekeeper. The gatekeeper conveys the payload to the Gatekeeper Transaction Message Protocol (GKTMP) and external route server for a flexible route decision based upon the ISDN User Part (ISUP) GTD parameters. The gateway then conditionally forwards the GTD payload on the basis of the instruction from the route server.
This command does not prevent sending the GTD to a gatekeeper. Any GTD on the originating gateway is sent to the gatekeeper for use in routing decisions. To prevent GTD creation, the signal-end-to-end command-line interface (CLI) option on the R2 interfaces should be disabled, and the Cisco PGW 2200 should be configured not to send GTD to the gateway.
Examples
The following example sets unconditional signal forwarding on a system-wide basis, where the GTD payload is tunneled in H.225 SETUP messages to endpoints:
Router(config)# voice service voipRouter(conf-voi-serv)# signaling forward unconditionalRouter(conf-voi-serv)# ^ZRouter# show running-configBuilding configuration...Current configuration : 4201 bytes!version 12.2service configno service single-slot-reload-enableno service padservice timestamps debug uptimeservice timestamps log uptimeno service password-encryptionservice internalservice udp-small-servers!hostname as5300-2!no logging bufferedlogging rate-limit console 10 except errorsaaa new-model!...!voice service voipsignaling forward unconditionalh323!...Related Commands
signaling forward (dial peer)
To enable a Cisco IOS gateway to forward the Generic Transparency Descriptor (GTD) payload to another gateway or gatekeeper for an individual dial peer, use the signaling forward command in dial peer configuration mode. To disable forwarding, use the no form of this command.
signaling forward {conditional | unconditional | none}
no signaling forward
Syntax Description
Command Default
The default is the value that is configured system-wide, or conditional if signaling forward is not configured system-wide.
Command Modes
Dial peer configuration
Command History
Usage Guidelines
This command is used with the Cisco PGW 2200 Signaling Controller in the Cisco SS7 Interconnect for Voice Gateways solution. You must configure the Cisco PGW 2200 to encapsulate SS7 ISUP messages in GTD format before using this command on the Cisco gateway.
If the target is a RAS target, for a non-GTD signaling payload, the original payload is forwarded. For a GTD signaling payload, the payload is encapsulated in an admission request (ARQ)/disengage request (DRQ) message and sent to the originating gatekeeper. The gatekeeper conveys the payload to the Gatekeeper Transaction Message Protocol (GKTMP) and external route server for a flexible route decision based upon the ISDN User Part (ISUP) GTD parameters. The gateway then conditionally forwards the GTD payload on the basis of the instruction from the route server.
This command does not prevent sending the GTD to a gatekeeper. Any GTD on the originating gateway is sent to the gatekeeper for use in routing decisions. To prevent GTD creation, the signal-end-to-end command-line interface (CLI) option on the R2 interfaces should be disabled, and the Cisco PGW 2200 should be configured not to send GTD to the gateway.
Examples
The following example sets unconditional signal forwarding on a system-wide basis, where the GTD payload is tunneled in H.225 SETUP messages to endpoints:
Router(config)# voice service voipRouter(conf-voi-serv)# signaling forward unconditionalRouter(conf-voi-serv)# ^ZRouter# show running-configBuilding configuration...Current configuration : 4201 bytes!version 12.2service configno service single-slot-reload-enableno service padservice timestamps debug uptimeservice timestamps log uptimeno service password-encryptionservice internalservice udp-small-servers!hostname as5300-2!no logging bufferedlogging rate-limit console 10 except errorsaaa new-model!...!voice service voipsignaling forward unconditionalh323!...Related Commands
signal-type
To set the signaling type to be used when connecting to a dial peer, use the signal-type command in dial peer configuration mode. To reset to the default, use the no form of this command.
signal-type {cas | cept | ext-signal | transparent}
no signal-type
Syntax Description
Command Default
cas
Command Modes
Dial peer configuration
Command History
Usage Guidelines
This command applies to Voice over Frame Relay (VoFR) and Voice over ATM (VoATM) dial peers. It is used with permanent connections only (Cisco trunks and FRF.11 trunks), not with switched calls.
This command is used to inform the local telephony interface of the type of signaling it should expect to receive from the far-end dial peer. To turn signaling off at this dial peer, select the ext-signal option. If signaling is turned off and there are no external signaling channels, a "hot" line exists, enabling this dial peer to connect to anything at the far end.
When you connect an FXS to another FXS, or if you have anything other than an FXS/FXO or E&M/E&M pair, the appropriate signaling type on Cisco 2600 and Cisco 3600 series routers is ext-signal (disabled).
If you have a digital E1 connection at the remote end that is running cept/MELCAS signaling and you then trunk that across to an analog port, you should make sure that you configure both ends for the cept signal type.
If you have a T1 or E1 connection at both ends and the T1/E1 is running a signaling protocol that is neither EIA-464, or cept/MELCAS, you might want to configure the signal type for the transparent option in order to pass through the signaling.
Examples
The following example disables signaling for VoFR dial peer 200:
dial-peer voice 200 vofrsignal-type ext-signalexitRelated Commands
silent-fax
To configure the voice dial peer for a Type 2 silent fax machine, use the silent-fax command in dial peer voice configuration mode. To disable a silent fax call to any POTS ports, use the no form of this command.
silent-fax
no silent-fax
Syntax Description
This command has no arguments or keywords.
Command Default
Silent fax is not configured.
Command Modes
Dial peer voice configuration
Command History
12.2(8)T
This command was introduced on the Cisco 803, Cisco 804, and Cisco 813.
Usage Guidelines
Use this command to configure the router to send a no ring alert tone to a Type 2 silent fax machine that is connected to any of the POTS ports. To check the status of the silent-fax configuration, use the show running-config command.
Examples
The following example shows that the silent-fax command has been configured on POTS port 1 but not on POTS port 2.
dial-peer voice 1 potsdestination-pattern 5551111port 1no call-waitingring 0volume 4caller-number 3334444 ring 1subaddress 20silent-faxdial-peer voice 2 potsdestination-pattern 5552222port 2no call-waitingring 0volume 2caller-number 3214567 ring 2subaddress 10sip
To enter SIP configuration mode, use the sip command in voice-service VoIP configuration mode.
sip
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values.
Command Modes
Voice-service VoIP configuration
Command History
Usage Guidelines
From the voice-service VoIP configuration mode, this command enables you to enter SIP configuration mode. From this mode, several SIP commands are available, such as the bind, session transport, and url commands.
Examples
The following example enters SIP configuration mode and sets the bind command on the SIP network:
Router(config)# voice service voipRouter(config-voi-srv)# sipRouter(conf-serv-sip)# bind control source-interface FastEthernet 0Related Commands
sip-server
To configure a network address for the Session Initiation Protocol (SIP) server interface, use the sip-server command in SIP UA configuration mode. To remove a network address configured for SIP, use the no form of this command.
sip-server {dns:[host-name] | ipv4:ip-addr [:port-num]}
no sip-server {dns:[host-name] | ipv4:ip-addr [:port-num]}
Syntax Description
Command Default
Null value
Command Modes
SIP UA configuration
Command History
Usage Guidelines
If you use this command, you can also use the session target {sip-server} command on each dial peer instead of repeatedly entering the SIP server interface address for each dial peer. Configuring a SIP server as a session target is useful if there is a Cisco SIP proxy server (CSPS) present in the network. With a CSPS, you can configure the SIP server option and have the interested dial peers use the CSPS by default.
To reset this command to a null value, use the default command.
Examples
The following example, beginning in global configuration mode, sets the global SIP server interface to the DNS host name "3660-2.sip.com." If you also use the command session target {sip server}, you do not have to set the DNS host name for each individual dial peer.
sip-uasip-server dns:3660-2.sip.comdial-peer voice 29 voipsession target sip-serverThe following example sets the global SIP server interface to an IP address:
sip-uasip-server ipv4:10.0.2.254Related Commands
sip-ua
To enable Session Initiation Protocol (SIP) UA configuration commands, in order to configure the user agent, use the sip-ua command in global configuration mode. To reset all SIP UA configuration commands to their default values, use the no form of this command.
sip-ua
no sip-ua
Syntax Description
This command has no arguments or keywords.
Command Default
No default behaviors or values
Command Modes
Global configuration
Command History
Usage Guidelines
Use this command to enter SIP UA configuration mode. Table 226 lists the SIP UA configuration mode commands:
Examples
The following example, beginning in global configuration mode, enters SIP UA configuration mode, configures the SIP user agent, and then returns to global configuration mode:
sip-uaretry invite 2retry response 2retry bye 2retry cancel 2sip-server ipv4:10.0.2.254timers invite-wait-100 500exitRelated Commands
snmp enable peer-trap poor-qov
To generate poor-quality-of-voice notifications for applicable calls associated with VoIP dial peers, use the snmp enable peer-trap poor-qov command in dial peer configuration mode. To disable notification, use the no form of this command.
snmp enable peer-trap poor-qov
no snmp enable peer-trap poor-qov
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
Dial peer configuration
Command History
Usage Guidelines
Use this command to generate poor-quality-of-voice notification for applicable calls associated with a dial peer. If you have a Simple Network Management Protocol (SNMP) manager that uses SNMP messages when voice quality drops, you might want to enable this command. Otherwise, you should disable this command to reduce unnecessary network traffic.
Examples
The following example enables poor-quality-of-voice notification for calls associated with VoIP dial peer 10:
dial-peer voice 10 voipsnmp enable peer-trap poor-qovRelated Commands
soft-offhook
To enable stepped offhook resistance during seizure, use the soft-offhook command in SCCP PLAR configuration mode. To disable this command, use the no form of this command.
soft-offhook
no soft-offhook
Syntax Description
This command has no arguments or keywords.
Command Default
This command is disabled by default, which means there is no stepped off-hook resistance during seizure.
Command Modes
SCCP PLAR configuration
Command History
Usage Guidelines
An off-hook indication into far-end ringing cadence ON condition can occur during glare conditions (outgoing seizure occurring at the same time as an incoming ring). This condition can also occur when the interface configuration includes the connection plar-opx command. If the connection plar-opx command is not configured, the FXO software waits for a ringing cadence to transition from ON to OFF prior to transitioning to the off-hook condition. (Glare can be minimized by configuring ground-start signaling.)
When the soft-offhook command is entered, the FXO hookswitch off-hook resistance is initially set to a midresistance value for outgoing or incoming seizure. This resistance limits the ringing current that occurs during seizure into ringing signals prior to far-end ring-trip. When ringing is no longer detected, hookswitch resistance is returned to its normal lower value. This prevents damage to the FXO line interface that may occur in locations with short loops and conventional ringing sources with low output impedance ringing sources that have potential to deliver high current.
The soft-offhook command applies to the following FXO interface cards:
•
•
•
•
Examples
The following example enables stepped off-hook resistance during seizure on voice port 2/0:
Router(config)# sccp plarRouter(config-sccp-plar)# voiceport 2/0 dial 3660 digit 1234 wait-connect 500 interval 200Router(config-sccp-plar)# soft-offhookRouter(config-sccp-plar)# shutdownRouter(config-sccp-plar)# no shutdownRelated Commands
source carrier-id
To configure debug filtering for the source carrier ID, use the source carrier-id command in call filter match list configuration mode. To disable, use the no form of this command.
source carrier-id string
no source carrier-id string
Syntax Description
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Examples
The following example shows the voice call debug filter set to match source carrier ID 4321:
call filter match-list 1 voicesource carrier-id 4321Related Commands
source trunk-group-label
To configure debug filtering for a source trunk group, use the source trunk-group-label command in call filter match list configuration mode. To disable, use the no form of this command.
source trunk-group-label group_number
no source trunk-group-label group_number
Syntax Description
Command Default
No default behavior or values
Command Modes
Call filter match list configuration
Command History
Examples
The following example shows the voice call debug filter set to match source trunk group 21:
call filter match-list 1 voicesource trunk-group-label 21Related Commands
speed dial
To designate a range of digits for SCCP telephony control (STC) application feature speed-dial codes, use the speed dial command in STC application feature speed-dial configuration mode. To return the range to its default, use the no form of this command.
speed dial from digit to digit
no speed dial
Syntax Description
Command Default
The default speed-dial codes are 1 to 9 for one-digit codes; 01 to 99 for two-digit codes.
Command Modes
STC application feature speed-dial configuration
Command History
12.4(2)T
This command was introduced.
12.4(6)T
The digit argument was modified to allow two-digit codes.
Usage Guidelines
This command is used with the STC application, which enables features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control.
Use this command to set the range of speed-dial codes only if you want to change the range from its default. The digit command determines whether speed-dial codes are one-digit or two-digit.
A maximum of nine one-digit or 99 two-digit speed-dial codes are supported. If you set the starting number to 0, the highest number you can set for the ending number is 8 for one-digit codes, or 98 for two-digit codes.
Note that the actual telephone numbers that are speed dialed are stored on Cisco CallManager or the Cisco CallManager Express system. The speed-dial codes that you set with this command are mapped to speed-dial positions on the call-control device. For example, if you set the starting number to 2 and the ending number to 7, the system maps 2 to speed-dial 1 and maps 7 to speed-dial 6.
You can enter numbers in this command in ascending or descending order. For example, the following commands are both valid:
Router(stcapp-fsd)# speed dial from 2 to 7Router(stcapp-fsd)# speed dial from 7 to 2To use the speed-dial feature on a phone, dial the STC application feature speed-dial (FSD) prefix and one of the speed-dial codes that has been configured with this command (or the default if this command was not used). For example, if the FSD prefix is * (the default) and the speed-dial codes are 1 to 9 (the default), dial *3 to dial the telephone number stored with speed-dial 3.
This command resets to its default range if you modify the value of the digit command. For example, if you set the digit command to 2, then change the digit command back to its default of 1, the speed-dial codes are reset to 1 to 9.
If the digit command is set to 2 and you configure a single-digit speed-dial code, the system converts the speed-dial code to two digits. For example, if you enter the range 1 to 5 in a two-digit configuration, the system converts the speed-dial codes to 11 to 15.
If you set any of the FSD codes in this range to a value that is already in use for another FSD 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 FSD codes only. The show stcapp feature codes command displays all FSD codes.
Examples
The following example sets an FSD code prefix of two pound signs (##) and a speed-dial code range of 2 to 7. After these values are configured, a phone user presses ##2 to dial the number that is stored with speed-dial 1 on the call-control system (Cisco CallManager or Cisco CallManager Express).
Router(config)# stcapp feature speed-dialRouter(stcapp-fsd)# prefix ##Router(stcapp-fsd)# speed dial from 2 to 7Router(stcapp-fsd)# exitThe following example shows how the speed-dial range that is set in the example above is mapped to the speed-dial positions on the call-control system. Note that the range from 2 to 7 is mapped to speed-dial 1 to 6.
Router# show stcapp feature codes...stcapp feature speed-dialprefix ##redial ###speeddial number of digit(s) 1voicemail ##0speeddial1 ##2speeddial2 ##3speeddial3 ##4speeddial4 ##5speeddial5 ##6speeddial6 ##7The following example sets a FSD code prefix of two asterisks (**) and a speed-dial code range of 12 to 17.
Router(config)# stcapp feature speed-dialRouter(stcapp-fsd)# prefix **Router(stcapp-fsd)# digit 2Router(stcapp-fsd)# speed dial from 12 to 17Router(stcapp-fsd)# exitRelated Commands
srtp (dial peer)
To specify that Secure Real-Time Transport Protocol (SRTP) be used to enable secure calls for a specific VoIP dial peer, to enable fallback, and to override global SRTP configuration, use the srtp command in dial peer voice configuration mode. To disable secure calls, to disable fallback, and to override global SRTP configuration, use the no form of this command.
srtp [fallback | system]
no srtp [fallback | system]
Syntax Description
Command Default
Global SRTP configuration set in voice service voip configuration mode is enabled.
Command Modes
Dial peer voice configuration
Command History
Usage Guidelines
You can enable secure calls using the srtp command either at the dial peer level, or at the global level. The srtp command in dial peer voice mode configures call security at the dial-peer level and takes precedence over the global srtp command. Use the srtp command in dial peer voice configuration mode to enable secure calls for a specific dial peer. Use the no form of this command to disable secure calls.
Use the srtp fallback command to enable secure calls and allow calls to fallback to nonsecure mode for a specific dial peer. This security policy applies to all calls going through the dial peer and is not configurable on a per-call basis. Using the srtp fallback command to configure call fallback at the dial-peer level takes precedence over the global srtp fallback command. The no form of this command disables SRTP and fallback. If you disallow fallback using the no srtp fallback command, a call cannot fall back to nonsecure mode.
Use the srtp system command to apply global level security settings to dial peers.
Examples
The following example enables secure calls and disallows fallback for a specific dial peer:
Router(config-dial-peer)# srtpThe following example enables secure calls and allows call fallback to nonsecure mode:
Router(config-dial-peer)# srtp fallbackThe following example defaults call security to global level SRTP behavior:
Router(config-dial-peer)# srtp systemRelated Commands
srtp (voice)
Enables secure calls globally in voice service voip configuration mode.
srtp fallback (voice)
Enables SRTP and fallback globally.
srtp (voice)
To specify that Secure Real-Time Transport Protocol (SRTP) be used to enable secure calls and call fallback, use the srtp command in voice service voip configuration mode. To disable secure calls and disallow fallback, use the no form of this command.
srtp [fallback]
no srtp [fallback]
Syntax Description
Command Default
Voice call security and fallback are disabled.
Command Modes
Voice service voip configuration
Command History
Usage Guidelines
Use the srtp command in voice service voip configuration mode to globally enable secure calls using SRTP media authentication and encryption. This security policy applies to all calls going through the gateway and is not configurable on a per-call basis. To enable secure calls for a specific dial peer, use the srtp command in dial peer voice configuration mode. Using the srtp command to configure call security at the dial-peer level takes precedence over the global srtp command.
Use the srtp fallback command to globally enable secure calls and allow calls to fall back to RTP (nonsecure) mode. This security policy applies to all calls going through the gateway and is not configurable on a per-call basis. To enable secure calls for a specific dial peer, use the srtp command in dial peer voice configuration mode. Using the srtp fallback command in dial peer voice configuration mode to configure call security takes precedence over the srtp fallback global command in voice service voip configuration mode. If you use the no srtp fallback command, fallback from SRTP to RTP (secure to nonsecure) is disallowed.
Examples
The following example enables secure calls:
Router(config-voi-serv)# srtpThe following example enables call fallback to nonsecure mode:
Router(config-voi-serv)# srtp fallbackRelated Commands
srv version
To generate Domain Name System Server (DNS SRV) queries with either RFC 2052 or RFC 2782 format, use the srv version command in SIP UA configuration mode. To reset to the default, use the no form of this command.
srv version {1 | 2}
no srv version
Syntax Description
1
Domain-name prefix of format protocol.transport. (RFC 2052 style).
2
Domain-name prefix of format _protocol._transport. (RFC 2782 style).
Command Default
2 (RFC 2782 style)
Command Modes
SIP UA configuration
Command History
Usage Guidelines
Session Initiation Protocol (SIP) on Cisco VoIP gateways uses DNS SRV queries to determine the IP address of the user endpoint. The query string has a prefix in the form of "protocol.transport." (RFC 2052) or "_protocol._transport." (RFC 2782). The selected string is then attached to the fully qualified domain name (FQDN) of the next hop SIP server.
By choosing the value of 1, this command provides compatibility with older equipment that supports only RFC 2052.
Examples
The following example sets up the srv version command in the RFC 2782 style (underscores surrounding the protocol):
Router(config)# sip-uaRouter(config-sip-ua)# srv version 2Related Commands
