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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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New and Modified Commands
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Commands: A
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Commands: B
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Commands: cache --> call fallback jitter-probe num-packets
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Commands: call fallback jitter-probe precedence --> ces connect
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Commands: ces initial-delay --> codec preference
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Commands: comfort-noise --> customer-id
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Commands: default (MGCP profile) through direct-inward-dial
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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: I
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Commands: K
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Commands: L
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Commands: M
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Commands: N
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Commands: O
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Commands: P through Q
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Commands: R
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Commands: SCCP through sgcp tse payload
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Commands: show aal2 profile through show controllers timeslots
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Commands: show controllers voice through show mgcp
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Commands: show mgcp connection through show ss7 sm session
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Commands: show ss7 sm set through shutdown (voice-port)
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Commands: signal through suppress
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Commands: T
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Commands: U
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Commands: V through W
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Commands: Z
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Feedback
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
supervisory disconnect anytone
supervisory disconnect dualtone
supervisory disconnect dualtone voice-class
supervisory dualtone-detect-params
Cisco IOS Voice Commands:
Si
This chapter contains commands to configure and maintain Cisco IOS voice applications. The commands are presented in alphabetical order. Some commands required for configuring voice may be found in other Cisco IOS command references. Use the command reference master index or search online to find these commands.
For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice Configuration Guide.
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.
FXO and FXS Voice Ports
signal {loop-start | ground-start}
no signal {loop-start | ground-start}
E&M Voice Ports
signal {wink-start | immediate | delay-dial}
no signal {wink-start | immediate | delay-dial}
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}
| groundstart | loopstart}no signal {cama {kp-0-nxx-xxxx-st | kp-0-npa-nxx-xxxx-st | kp-2-st | kp-npd-nxx-xxxx-st} | groundstart | loopstart}
Syntax Description
Defaults
FXO and FXS interfaces: loop-start
E&M interfaces: wink-start
CAMA interfaces: loop-startCommand Modes
Voice-port configuration
Command History
11.3(1)T
This command was introduced on the Cisco 3600 series.
12.2(11)T
This command was modified to support ANI transmission.
Usage Guidelines
This command applies to analog voice ports only.
Using the signal command for an FXO or FXS voice port changes the signal value for both voice ports on a voice port module (VPM) card.
Note
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, refer to the installation documentation, Voice Network Module and Voice Interface Card Configuration Note, that came with your voice network module.
Configuring this command for an E&M voice port changes only the signal value for the selected voice port. In either case, the voice port must be shut down and then activated before the configured values take effect.
Some PBXs miss initial digits if the E&M voice port is configured for immediate signaling. If this occurs, use delay-dial signaling instead. Some non-Cisco devices have a limited number of DTMF receivers. This type of equipment must delay the calling side until a DTMF receiver is available.
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 PSAP location.
The following four types of ANI transmittal schemes are based on the actual number of digits transmitted toward the E911 tandem or the PSAP. In each instance, the actual calling number is proceeded with a key pulse (KP) followed by an information field (I) or a Numbering Plan Digit (NPD), which is then followed by the Automatic Number Identification (ANI) calling number, and finally followed by a start pulse (ST). The value 0 indicates that the single digit information field (I). ANI was provided by the private branch exchange (PBX). The value 2 indicates that ANI was not provided by the PBX.
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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 415-555-2222 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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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 outpulsing a 10-digit ANI and the 911 call it receives has a 7-digit calling party number, the router transmits kp-2-st.
Note
Examples
The following example configures ground start 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 ground-startThe following example configures a 10-digit ANI transmission:
Router(config)# voice-port 1/0/0Router(config-voiceport)# signal cama kp-0-npa-nxxx-xxxx-stRelated Commands
ani mapping
Preprograms the NPA, or area code, into a single MF digit.
voice-port
Enters voice-port configuration mode.
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.
Defaults
No default behavior or values
Command Modes
Voice-port configuration
Command History
Examples
The following example configures a voice port on a Cisco IAD2420 series IAD 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 number
no signal keepalive number
Syntax Description
Defaults
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.
Examples
The following example, beginning in global configuration mode, sets the keepalive signaling interval 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
Defaults
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
Defaults
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.
Defaults
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
Defaults
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.
Defaults
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.
Defaults
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.
Defaults
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.
Defaults
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
Defaults
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
Defaults
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
Defaults
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
Defaults
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 on a Cisco 2600series or Cisco 3600 series router or on a Cisco MC3810 for VoFR dial peer 200, starting from global configuration mode:
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.
Defaults
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.
Defaults
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 user-agent 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
Defaults
Null value
Command Modes
SIP user-agent 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) user-agent configuration commands, in order to configure the user agent, use the sip-ua command in global configuration mode. To reset all SIP user-agent 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.
Defaults
No default behaviors or values
Command Modes
Global configuration
Command History
Usage Guidelines
Use this command to enter SIP user-agent configuration mode. Table 153 lists the SIP user-agent configuration mode commands:
Examples
The following example, beginning in global configuration mode, enters SIP user-agent 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.
Defaults
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
speed-dial (ephone)
To set speed-dial buttons on a Cisco IP phone, use the speed-dial command in ephone configuration mode. To disable the buttons, use the no form of this command.
speed-dial button-number directory-number
no speed-dial button-number directory-number
Syntax Description
button-number
Speed-dial string tag for the Cisco IP phone speed-dial button number. Range is from 1 to 4.
directory-number
Directory number on a phone.
Defaults
No default behavior or values
Command Modes
Ephone configuration
Command History
Usage Guidelines
If more speed-dial buttons are defined than are actually supported by the physical phone, the extra speed-dial configurations are ignored.
Examples
The following example sets speed-dial button 1 for directory number 5001:
Router(config)# ephone 1Router(config-ephone)# speed-dial 1 5001Related 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 user-agent 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).
Defaults
2 (RFC 2782 style)
Command Modes
SIP user-agent 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
ss7 mtp2-variant bellcore
To configure the router for Telcordia Technologies (formerly Bellcore) standards, use the ss7 mtp2-variant bellcore command in global configuration mode.
ss7 mtp2-variant bellcore [channel] [parameters]
Syntax Description
channel
(Optional) Channel. Range is from 0 to 3.
parameters
(Optional) Particular Bellcore standard. See Table 154 for descriptions, defaults, and ranges.
Defaults
Bellcore is the default variant if no other is configured.
See Table 154 for default parameters.Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
This MTP2 variant has timers and parameters that can be configured using the values listed in Table 154. To restore the designated default, use the no or the default form of the command (see example below).
Note
Examples
The following example sets the aligned/ready timer duration on channel 0 to 30,000 ms:
ss7 mtp2-variant bellcore 0 T1 30000The following example restores the aligned/ready timer default value of 13,000 ms:
ss7 mtp2-variant bellcore 0 no T1Related Commands
ss7 mtp2-variant itu
Specifies the MTP2-variant as ITU.
ss7 mtp2-variant ntt
Specifies the MTP2-variant as NTT.
ss7 mtp2-variant ttc
Specifies the MTP2-variant as TTC.
ss7 mtp2-variant itu
To configure the router for ITU (International Telecom United) standards, use the ss7 mtp2-variant itu command in global configuration mode.
ss7 mtp-variant itu [channel] [parameters]
Syntax Description
channel
Channel. Range is from 0 to 3.
parameters
(Optional) Particular Bellcore standard. See Table 155 for descriptions, defaults, and ranges.
Defaults
Bellcore is the default variant if no other is configured.
See Table 155 for ITU default parameters.Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
The ITU MTP2 variant has timers and parameters that can be configured using the values listed in Table 155. To restore the designated default, use the no or the default form of the command (see the example below).
Examples
The following example sets the emergency proving period on channel 1 to 10,000 ms:
ss7 mtp2-variant itu 1t4-Emergency-Proving 10000The following example restores the emergency proving period default value of 5,000 ms:
ss7 mtp2-variant itu 1default t4-Emergency-ProvingRelated Commands
ss7 mtp2-variant bellcore
Specifies the MTP2-variant as Bellcore.
ss7 mtp2-variant ntt
Specifies the MTP2-variant as NTT.
ss7 mtp2-variant ttc
Specifies the MTP2-variant as TTC.
ss7 mtp2-variant ntt
To configure the router for NTT (Japan) standards, use the ss7 mtp2-variant ntt command in global configuration mode.
ss7 mtp-variant ntt [channel] [parameters]
Syntax Description
channel
Channel. Range is from 0 to 3.
parameters
(Optional) Particular Bellcore standard. See Table 156 for descriptions, defaults, and ranges.
Defaults
Bellcore is the default variant if no other is configured.
See Table 156 for NTT default parameters.Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
The NTT MTP2 variant has timers and parameters that can be configured using the values listed in Table 156. To restore the designated default, use the no or the default form of the command (see the example below).
Examples
The following example sets the SUERM error rate threshold on channel 2 to 100:
ss7 mtp2-variant ntt 2SUERM-threshold 100The following example restores the SUERM error rate threshold default value of 64:
ss7 mtp2-variant ntt 2no SUERM-thresholdRelated Commands
ss7 mtp2-variant bellcore
Specifies the MTP2-variant as Bellcore.
ss7 mtp2-variant itu
Specifies the MTP2-variant as ITU.
ss7 mtp2-variant ttc
Specifies the MTP2-variant as TTC.
ss7 mtp2-variant ttc
To configure the router for TTC (Japan Telecom) standards, use the ss7 mtp2-variant ttc command in global configuration mode.
ss7 mtp-variant ttc [channel] [parameters]
Syntax Description
channel
Channel. Range is from 0 to 3.
parameters
(Optional) Particular Bellcore standard. See Table 157 for descriptions, defaults, and ranges.
Defaults
Bellcore is the default variant if no other is configured.
See Table 157 for TTC default parameters.Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
The TTC MTP2 variant has timers and parameters that can be configured using the values listed in Table 157. To restore the designated default, use the no or the default form of the command (see the example below).
Examples
The following example sets the maximum number of proving attempts for channel 3 to 3:
ss7 mtp2-variant ttc 3proving-attempts 3The following example restores the maximum number of proving attempts to the default value:
ss7 mtp2-variant ttc 3default proving-attemptsRelated Commands
ss7 mtp2-variant bellcore
Specifies the MTP2-variant as Bellcore.
ss7 mtp2-variant itu
Specifies the MTP2-variant as ITU.
ss7 mtp2-variant ntt
Specifies the MTP2-variant as NTT.
ss7 session
To create a Reliable User Datagram Protocol (RUDP) session and explicitly add an RUDP session to a Signaling System 7 (SS7) session set, use the ss7 session command in global configuration mode. To delete the session, use the no form of this command.
ss7 session session-id address destination-address destinaion-port local-address local-port [session-set session-number]
no ss7 session session-id
Syntax Description
Defaults
No session is configured.
Command Modes
Global configuration
Command History
Usage Guidelines
For the Cisco 2600-based SLT, you can configure a maximum of four sessions, two for each Cisco SLT. In a redundant VSC configuration, session 0 and session 2 are configured to one VSC, and session 1 and session 3 are configured to the other. Session 0/1 and session 2/3 run to the Cisco SLT.
The VSC must be configured to send messages to the local port, and it must be configured to listen on the remote port. You must also reload the router whenever you remove a session or change the parameters of a session.
This command replaces the ss7 session-0 address and ss7 session-1 address commands, which contain hard-coded session numbers. The new command is used for the new dual Ethernet capability.
The new CLI supports both single and dual Ethernet configuration by being backward compatible with the previous session-0 and session-1 commands so that you can configure a single Ethernet instead of two, if needed.
For the Cisco AS5350 and Cisco AS5400-based SLT, you can configure a maximum of two sessions, one for each signaling link. In a redundant MGC configuration, session 0 is configured to one MGC and session 1 is configured to the other.
The MGC must be configured to send messages to the local port, and the MGC must be configured to listen on the remote port.
You must reload the router whenever you remove a session or change the parameters of a session.
By default, each RUDP session must belong to SS7 session set 0. This allows backward compatibility with existing SS7 configurations.
If the session-set keyword is omitted, the session is added to the default SS7 session set 0. This allows backward compatibility with older configurations. Entering the no form of the command is still sufficient to remove the session ID for that RUDP session.
If you want to change the SS7 session set to which a session belongs, you have to remove the entire session first. This is intended to preserve connection and recovery logic.
Examples
The following example sets up two sessions on a Cisco 2611 and creates session set 2:
ss7 session-0 address 172.16.1.0 7000 172.16.0.0 7000 session-set 2ss7 session-1 address 172.17.1.0 7002 172.16.0.0 7001 session-set 2
Note
Related Commands
ss7 session cumack_t
To set the Reliable User Datagram Protocol (RUDP) cumulative acknowledgment timer for a specific SS7 signaling link session, use the ss7 session cumack_t command in global configuration mode. To reset to the default, use the no form of this command.
ss7 session-session number cumack_t milliseconds
no ss7 session-session number cumack_t
Caution
Syntax Description
Defaults
300 ms
Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
The cumulative acknowledgment timer determines when the receiver sends an acknowledgment. If the timer is not already running, it is initialized when a valid data, null, or reset segment is received. When the cumulative acknowledgment timer expires, the last in-sequence segment is acknowledged. The RUDP typically tries to "piggyback" acknowledgments on data segments being sent. However, if no data segment is sent in this period of time, it sends a standalone acknowledgment.
Examples
The following example sets up two sessions and sets the cumulative acknowledgment timer to 320 ms for each one:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7000ss7 session-0 cumack_t 320ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7001ss7 session-1 cumack_t 320Related Commands
ss7 session kp_t
To set the null segment (keepalive) timer for a specific SS7 signaling link session, use the ss7 session kp_t command in global configuration mode. To reset to the default, use the no form of this command.
ss7 session-session number kp_t milliseconds
no ss7 session-session number kp_t
Caution
Syntax Description
Defaults
2000 ms
Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
The null segment timer determines when a null segment (keepalive) is sent by the client Cisco 2600 series router. On the client, the timer starts when the connection is established and is reset each time a data segment is sent. If the null segment timer expires, the client sends a keepalive to the server to verify that the connection is still functional. On the server, the timer restarts each time a data or null segment is received from the client.
The value of the server's null segment timer is twice the value configured for the client. If no segments are received by the server in this period of time, the connection is no longer valid.
To disable keepalive, set this parameter to 0.
Examples
The following example sets up two sessions and sets a keepalive of 1,800 ms for each one:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7000ss7 session-0 kp_t 1800ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7001ss7 session-1 kp_t 1800Related Commands
ss7 session m_cumack
To set the maximum number of segments that can be received before the Reliable User Datagram Protocol (RUDP) sends an acknowledgment in a specific SS7 signaling link session, use the ss7 session m_cumack command in global configuration mode. To reset to the default, use the no form of this command.
ss7 session-session number m_cumack segments
no ss7 session-session number m_cumack
Caution
Syntax Description
Defaults
3 segments
Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
The cumulative acknowledgment counter records the number of unacknowledged, in-sequence data, null, or reset segments received without a data, null, or reset segment being sent to the transmitter. If this counter reaches the configured maximum, the receiver sends a standalone acknowledgment (a standalone acknowledgment is a segment that contains only acknowledgment information). The standalone acknowledgment contains the sequence number of the last data, null, or reset segment received.
If you set this parameter to 0, an acknowledgment is sent immediately after a data, null, or reset segment is received.
Examples
The following example sets up two sessions and in each session sets a maximum of two segments for receipt before acknowledgment:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001ss7 session-0 m_cumack 2ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000ss7 session-1 m_cumack 2Related Commands
ss7 session m_outseq
To set the maximum number of out-of-sequence segments that can be received before the Reliable User Datagram Protocol (RUDP) sends an extended acknowledgment in a specific SS7 signaling link session, use the ss7 session m_outseq command in global configuration mode. To reset to the default, use the no form of this command.
ss7 session-session number m_outseq segments
no ss7 session-session number m_outseq
Caution
Syntax Description
Defaults
3 segments
Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
The out-of-sequence acknowledgment counter records the number of data segments that have arrived out of sequence. If this counter reaches the configured maximum, the receiver sends an extended acknowledgment segment that contains the sequence numbers of the out-of-sequence data, null, and reset segments received. When the transmitter receives the extended acknowledgment segment, it retransmits the missing data segments.
If you set this parameter to 0, an acknowledgment is sent immediately after an out-of-sequence segment is received.
Examples
The following example sets up two sessions and sets a maximum number of four out-of-sequence segments for each session:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001ss7 session-0 m_outseq 4ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000ss7 session-1 m_outseq 4Related Commands
ss7 session m_rcvnum
To set the maximum number of segments that the remote end can send before receiving an acknowledgment in a specific SS7 signaling link session, use the ss7 session m_rcvnum command in global configuration mode. To reset to the default, use the no form of this command.
ss7 session-session number m_rcvnum segments
no ss7 session-session number m_rcvnum
Caution
Syntax Description
Defaults
32 segments
Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
The outstanding segments counter is the maximum number of segments that the Cisco IOS software end of the connection can send without getting an acknowledgment from the receiver. The receiver uses the counter for flow control.
Examples
The following example sets up two sessions and for each session sets a maximum of 36 segments for receipt before an acknowledgment:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001ss7 session-0 m_rcvnum 36ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000ss7 session-1 m_rcvnum 36Related Commands
ss7 session m_retrans
To set the maximum number of times that the Reliable User Datagram Protocol (RUDP) attempts to resend a segment before declaring the connection invalid in a specific SS7 signaling link session, use the ss7 session m_retrans command in global configuration mode. To reset to the default, use the no form of this command.
ss7 session-session number m_retrans number
no ss7 session-session number m_retrans
Caution
Syntax Description
Defaults
2 times
Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
The retransmission counter is the number of times a segment has been retransmitted. If this counter reaches the configured maximum, the transmitter resets the connection and informs the upper-layer protocol.
If you set this parameter to 0, the RUDP attempts to resend the segment continuously.
Examples
The following example sets up two sessions and for each session sets a maximum number of three times to resend before a session becomes invalid:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001ss7 session-0 m_retrans 3ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000ss7 session-1 m_retrans 3Related Commands
ss7 session retrans_t
To set the amount of time that the Reliable User Datagram Protocol (RUDP) waits to receive an acknowledgment for a segment in a specific SS7 signaling link session, use the ss7 session retrans_t command in global configuration mode. If the RUDP does not receive the acknowledgment in this time period, the RUDP retransmits the segment. To reset to the default, use the no form of this command.
ss7 session-session number retrans_t milliseconds
no ss7 session-session number retrans_t
Caution
Syntax Description
Defaults
600 ms
Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
The retransmission timer is used to determine whether a packet must be retransmitted and is initialized each time a data, null, or reset segment is sent. If an acknowledgment for the segment is not received by the time the retransmission timer expires, all segments that have been transmitted—but not acknowledged—are retransmitted.
This value should be greater than the value configured for the cumulative acknowledgment timer by using the ss7 session cumack_t command.
Examples
The following example sets up two sessions and specifies 550 ms as the time to wait for an acknowledgment for each session:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001ss7 session-0 retrans_t 550ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000ss7 session-1 retrans_t 550Related Commands
ss7 set
Note
To independently select failover-timer values for each session set and to specify the amount of time that the SS7 Session Manager waits for the active session to recover or for the standby media gateway controller (MGC) to indicate that the Cisco Signaling Link Terminal (SLT) should switch traffic to the standby session, use the ss7 set command in global configuration mode. To restore the restore the failover timer to its default value of 5, use the no form of this command.
ss7 set [session-set session-id] failover-timer ft-value
no ss7 set [session-set session-id] failover-timer
Syntax Description
Defaults
The failover timer is not set.
Command Modes
Global configuration
Command History
12.2(15)T
This command was introduced. This command replaces the ss7 set failover-timer command.
Usage Guidelines
The failover-timer keyword and the ft-value argument specify the number of seconds that the Session Manager waits for the active session to recover or for the standby MGC to indicate that the SLT should switch traffic to the standby session and to make that session the active session. If the failover timer expires without recovery of the original session or if the system fails to get an active message from the standby MGC, the signaling links are taken out of service.
The no form of this command restores the failover timer to its default value of 5. Omitting the optional session-set keyword implicitly selects SS7 session set 0, which is the default.
Examples
The following example sets the failover timer to four seconds without using the session-set option:
ss7 set failover-timer 4The following example sets the failover timer to 10 seconds and sets the SS7 session set value to 5:
ss7 set session-set 5 failover-timer 10Related Commands
ss7 set failover-timer
To specify the amount of time that the SS7 Session Manager waits for the active session to recover or for the standby Media Gateway Controller to indicate that the SLT should switch traffic to the standby session, use the ss7 set failover-timer command in global configuration mode. To reset ti the default, use the no form of this command.
ss7 set failover-timer [seconds]
no ss7 set failover-timer
Syntax Description
seconds
Time, in seconds, that the session manager waits for a session to recover. Range is from 1 to 10. Default is 3.
Defaults
3 seconds
Command Modes
Global configuration
Command History
12.0(7)XR
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
Usage Guidelines
This command specifies the number of seconds that the session manager waits for the active session to recover or for the standby media gateway controller to indicate that the SLT should switch traffic to the standby session and to make that session the active session. If the timer expires without a recovery of the original session or an active message from the standby media gateway controller, the signaling links are taken out of service.
Examples
The following example sets the failover timer to 4 seconds:
ss7 set failover-timer 4Related Commands
show ss7 sm set
Displays the current failover timer setting.
ss7 session
Establishes a session.
station name
To specify the name that is to be sent as caller ID information and to enable caller ID, use the station name command in voice-port configuration mode at the sending Foreign Exchange Station (FXS) voice port or at a Foreign Exchange Office (FXO) port through which routed caller ID calls pass. To remove the name, use the no form of this command.
station name name
no station name name
Syntax Description
Defaults
The default is no station name.
Command Modes
Voice-port configuration
Command History
12.1(2)XH
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
Usage Guidelines
This optional command is configured on FXS voice ports that are used to originate on-net calls. The information entered is displayed by the telephone attached to the FXS port at the far end of the on-net call. It can also be configured on the FXO port of a router on which caller ID information is expected to be received from the Central Office (CO), to suit situations in which a call is placed from the CO, then goes through the FXO interface, and continues to a far-end FXS port through an on-net call. In this case, if no caller ID information is received from the CO telephone line, the far-end call recipient receives the information configured on the FXO port.
Note
Do not use this command when the caller ID standard is dual-tone multifrequency (DTMF). DTMF caller ID can carry only the calling number.
If the station name, station number, or a caller-id alerting command is configured on the voice port, caller ID is automatically enabled, and the caller-id enable command is not necessary.
Examples
The following example configures a Cisco 2600 series or Cisco 3600 series router voice port from which caller ID information is sent:
voice-port 1/0/1cptone USstation name A. Personstation number 4085551111The following example configures a Cisco MC3810 voice port from which caller ID information is sent:
voice-port 1/0cptone northamericastation name A. Personstation number 4085551111caller-id alerting ring 1Related Commands
station number
To specify the telephone or extension number that is to be sent as caller ID information and to enable caller ID, use the station number command in voice-port configuration mode at the sending Foreign Exchange Station (FXS) voice port or at a Foreign Exchange Office (FXO) port through which routed caller ID calls pass. To remove the number, use the no form of this command.
station number number
no station number number
Syntax Description
Defaults
The default is no station number.
Command Modes
Voice-port configuration
Command History
12.1(2)XH
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.1(3)T
This command was integrated into Cisco IOS Release 12.1(3)T.
Usage Guidelines
This optional command is configured on FXS voice ports that are used to originate on-net calls. The information entered is displayed by the telephone attached to the FXS port at the far end of the on-net call. It can also be configured on the FXO port of a router on which caller ID information is expected to be received from the Central Office (CO), to suit situations in which a call is placed from the CO, then goes through the FXO interface, and continues to a far-end FXS port through an on-net call. In this case, if no caller ID information is received from the CO telephone line, the far-end call recipient receives the information configured on the FXO port.
Within the network, if an originating station does not include configured number information, Cisco IOS software determines the number by using reverse dial-peer search.
Note
If the station name, station number, or a caller-id alerting command is configured on the voice port, caller ID is automatically enabled, and the caller-id enable command is not necessary.
Examples
The following example configures a Cisco 2600 or Cisco 3600 series router voice port from which caller ID information is sent:
voice-port 1/0/1cptone USstation name A. Personstation number 4085551111The following example configures a Cisco MC3810 voice port from which caller ID information is sent:
voice-port 1/0cptone northamericastation name A. Personstation number 4085551111caller-id alerting ring 1Related Commands
subaddress
To configure a subaddress for a POTS port, use the subaddress command in dial-peer voice configuration mode. To disable the subaddress, use the no form of this command.
subaddress number
no subaddress number
Syntax Description
Defaults
No subaddress is available for a POTS port.
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
You can use this command for any dial-peer voice POTS port. You can configure only one subaddress for each of the POTS ports. The latest entered subaddress on the dial-peer voice port is stored. To check the status of the subaddress configuration, use the show running-config command.
Examples
The following examples show that a subaddress of 20 has been set for POTS port 1 and that a subaddress of 10 has been set for POTS port 2:
dial-peer voice 1 potsdestination-pattern 5555555port 1no call-waitingring 0volume 4caller-number 1111111 ring 3caller-number 2222222 ring 1caller-number 3333333 ring 1subaddress 20dial-peer voice 2 potsdestination-pattern 4444444port 2no call-waitingring 0volume 2caller-number 6666666 ring 2caller-number 7777777 ring 3subaddress 10subcell-mux
To enable ATM adaption layer 2 (AAL2) common part sublayer (CPS) subcell multiplexing on a Cisco router, use the subcell-mux command in voice-service configuration mode. To reset to the default, use the no form of this command.
subcell-mux time
no subcell-mux time
Syntax Description
time
Timer value, in milliseconds. Range is from 5 to 1000 (1 second). Default is 10. The time argument is implemented for Cisco 3600 routers.
Defaults
10 ms
Subcell multiplexing is offCommand Modes
Voice-service configuration
Command History
Usage Guidelines
Use this command to enable ATM adaption layer 2 (AAL2) common part sublayer (CPS) subcell multiplexing when the Cisco router interoperates with other equipment that uses subcell multiplexing.
Examples
The following example sets AAL2 CPS subcell multiplexing to 15 ms:
Router(conf-voi-serv-sess)# subcell-mux 15Related Commands
voice-service
Specifies the voice encapsulation type and enters voice-service configuration mode.
supervisory answer dualtone
To enable answer supervision on a Foreign Exchange Office (FXO) voice port, use the supervisory answer dualtone command in voice-port configuration mode. To disable answer supervision on a voice port, use the no form of this command.
supervisory answer dualtone [sensitivity {high | medium | low}]
no supervisory answer dualtone
Syntax Description
Defaults
Answer supervision is not enabled on voice ports.
Command Modes
Voice-port configuration
Command History
12.2(2)T
This command was introduced on the following platforms: Cisco 1750, Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
Usage Guidelines
This command configures the FXO voice port to detect voice, fax, and modem traffic when calls are answered. If answer supervision is enabled, calls are not recorded as connected until answer supervision is triggered.
This command enables a ring-no-answer timeout that drops calls after a specified period of ringback. The period of ringback can be configured using the timeouts ringing command.
This command automatically enables disconnect supervision in the preconnect mode on the voice port if disconnect supervision is not already enabled with the supervisory disconnect dualtone command.
This command is applicable to analog FXO voice ports with loop-start signaling.
If false answering is detected, decrease the sensitivity setting. If answering detection is failing, increase the sensitivity setting.
Examples
The following example enables answer supervision on voice port 1/5:
voice-port 1/5supervisory answer dualtoneThe following example enables answer supervision on voice port 0/1/1:
voice-port 0/1/1supervisory answer dualtoneRelated Commands
supervisory custom-cptone
To associate a class of custom call-progress tones with a voice port, use the supervisory custom-cptone command in voice-port configuration mode. To reset to the default, use the no form of this command.
supervisory custom-cptone cptone-name
no supervisory custom-cptone
Syntax Description
Defaults
U.S. standard call-progress tones are associated with a voice port.
Command Modes
Voice-port configuration
Command History
12.1(5)XM
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.2(2)T
This command was implemented on the Cisco 1750.
Usage Guidelines
This command associates a class of custom call-progress tones, defined by the voice class custom-cptone command, with a voice port.
You can associate the same custom call-progress tones to multiple voice ports.
You can associate only one class of custom call-progress tones with a voice port. If you associate a second class of custom call-progress tones with a voice port, the second class of custom tones replaces the one previously assigned.
This command is applicable to analog Foreign Exchange Office (FXO) voice ports with loop-start signaling.
Examples
The following example associates the class of custom call-progress tones named country-x with voice ports 1/4 and 1/5:
voice-port 1/4supervisory custom-cptone country-xexitvoice-port 1/5supervisory custom-cptone country-xexitRelated Commands
supervisory disconnect
To enable a supervisory disconnect signal on Foreign Exchange Office (FXO) ports, use the supervisory disconnect command in voice-port configuration mode. To disable the signal, use the no form of this command.
supervisory disconnect
no supervisory disconnect
Syntax Description
This command has no arguments or keywords.
Defaults
Enabled
Command Modes
Voice-port configuration
Command History
Usage Guidelines
This command indicates whether supervisory disconnect signaling is available on the FXO port. Supervisory disconnect signaling is a power denial from the switch lasting at least 350 ms. When this condition is detected, the system interprets this as a disconnect indication from the switch and clears the call.
You should configure no supervisory disconnect on the voice port if there is no supervisory disconnect available from the switch.
Note
Examples
The following example configures supervisory disconnect on a Cisco 3600 series voice port:
voice-port 2/1/0supervisory disconnectThe following example configures supervisory disconnect on a Cisco MC3810 voice port:
voice-port 1/1supervisory disconnectsupervisory disconnect anytone
To configure a Foreign Exchange Office (FXO) voice port to go on-hook if the router detects any tone from a PBX or the PSTN before an outgoing call is answered, use the supervisory disconnect anytone command in voice-port configuration mode. To disable the supervisory disconnect function, use the no form of this command.
supervisory disconnect anytone
no supervisory disconnect anytone
Syntax Description
This command has no arguments or keywords.
Defaults
The supervisory disconnect function is not enabled on voice ports.
Command Modes
Voice-port configuration
Command History
Usage Guidelines
Use this command to provide disconnect if the PBX or PSTN does not provide a supervisory tone. Examples of tones that trigger a disconnect include busy tone, fast busy tone, and dial tone.
This command is enabled only during call setup (before the call is answered).
You must enable echo cancellation; otherwise, ringback tone from the router can trigger a disconnect.
This command replaces the no supervisory disconnect signal command. If you enter this command, the supervisory disconnect anytone feature is enabled, and the message supervisory disconnect anytone is displayed when show commands are entered.
If you enter either the supervisory disconnect anytone command or the no supervisory disconnect signal command, answer supervision is automatically disabled.
Examples
The following example configures voice ports 1/4 and 1/5 to go on-hook if any tone from the PBX or PSTN is detected before the call is answered:
voice-port 1/4supervisory disconnect anytoneexitvoice-port 1/5supervisory disconnect anytoneexitThe following example disables the disconnect function on voice port 1/5:
voice-port 1/5no supervisory disconnect anytoneexitRelated Commands
supervisory disconnect dualtone
To enable disconnect supervision on a Foreign Exchange Office (FXO) voice port, use the supervisory disconnect dualtone command in voice-port configuration mode. To disable the supervisory disconnect function, use the no form of this command.
supervisory disconnect dualtone {mid-call | pre-connect}
no supervisory disconnect dualtone
Syntax Description
mid-call
Disconnect supervision operates throughout the duration of the call.
pre-connect
Disconnect supervision operates during call setup and stops when the called telephone goes off-hook.
Defaults
Disconnect supervision is not enabled on voice ports.
Command Modes
Voice-port configuration
Command History
12.1(5)XM
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.2(2)T
This command was implemented on the Cisco 1750.
Usage Guidelines
This command configures an FXO voice port to disconnect calls when the router detects call-progress tones from a PBX or the PSTN. Disconnection occurs after the wait-release time specified on the voice port.
Disconnect supervision is automatically enabled in the preconnect mode on the voice port if the supervisory answer dualtone command is entered.
This feature is applicable to analog FXO voice ports with loop-start signaling.
Examples
The following example specifies tone detection during the entire call duration:
voice-port 1/5supervisory disconnect dualtone mid-callexitThe following example specifies tone detection only during call setup:
voice-port 0/1/1supervisory disconnect dualtone pre-connectexitRelated Commands
supervisory disconnect dualtone voice-class
To assign a previously configured voice class for Foreign Exchange Office (FXO) supervisory disconnect tone to a voice port, use the supervisory disconnect dualtone voice-class command in voice port configuration mode. To remove a voice class from a voice-port, use the no form of this command.
supervisory disconnect dualtone {mid-call | pre-connect} voice-class tag
no supervisory disconnect dualtone voice-class tag
Syntax Description
Defaults
No voice class is assigned to a voice port.
Command Modes
Voice-port configuration
Command History
12.1(3)T
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
Usage Guidelines
You can apply an FXO supervisory disconnect tone voice class to multiple voice ports. You can assign only one FXO supervisory disconnect tone voice class to a voice port. If a second voice class is assigned to a voice port, the second voice class replaces the one previously assigned. You cannot assign separate FXO supervisory disconnect tone commands directly to the voice port.
This feature is applicable to analog FXO voice ports with loop-start signaling.
Examples
The following example assigns voice class 70 to FXO voice port 1/5 of a Cisco MC3810 series concentrator and specifies tone detection during the entire call duration:
voice-port 1/5no echo-cancel enablesupervisory disconnect dualtone mid-call voice-class 70The following example assigns voice class 80 to FXO voice port 0/1/1 of a Cisco 3600 series router and specifies tone detection only during call setup:
voice-port 0/1/1no echo-cancel enablesupervisory disconnect dualtone pre-connect voice-class 80Related Commands
supervisory dualtone-detect-params
To associate a class of modified tone-detection tolerance limits with a voice port, use the supervisory dualtone-detect-params command in voice-port configuration mode. To reset to the default, use the no form of this command.
supervisory dualtone-detect-params tag
no supervisory dualtone-detect-params
Syntax Description
Defaults
The default tone-detection tolerance limits are associated with voice ports.
Command Modes
Voice-port configuration
Command History
12.1(5)XM
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
12.2(2)T
This command was implemented on the Cisco 1750.
Usage Guidelines
This command associates a specific set of modified tone-detection tolerance limits, defined by the voice class dualtone-detect-params command, with a voice port.
You can associate the same class of modified tone-detection tolerance limits to multiple voice ports.
You can associate only one class of modified tone-detection tolerance limits to a voice port. If you associate a second class of modified tone-detection tolerance limits with a voice port, the second class replaces the one previously assigned.
This command is applicable to analog Foreign Exchange Office (FXO) voice ports with loop-start signaling.
Examples
The following example associates the class of modified tone-detection tolerance limits that has tag 70 with voice ports 1/5 and 1/6.
voice-port 1/5supervisory dualtone-detect-params 70exitvoice-port 1/6supervisory dualtone-detect-params 70exitThe following example restores the default tone-detection parameters to voice port 1/5.
voice-port 1/5no supervisory dualtone-detect-paramsexitRelated Commands
suppress
To suppress accounting for a specific call leg, use the suppress command in gateway accounting AAA configuration mode. To reenable accounting for that leg, use the no form of this command.
suppress [pots | rotary | voip]
no suppress [pots | rotary | voip]
Syntax Description
Defaults
Accounting is enabled.
Command Modes
Gateway accounting AAA configuration
Command History
Usage Guidelines
Use this command to turn off accounting for a specific call leg.
If both incoming and outgoing call legs are of the same type, no accounting packets are generated.
Use the rotary keyword to suppress excess start and stop accounting records. This causes only one pair of records to be generated for every connection attempt through a dial peer.
Examples
The following example suppresses accounting for the POTS call leg.
suppress potsRelated Commands
debug suppress rotary
Displays connection attempt statistics.
gw-accounting aaa
Enables VoIP gateway accounting.
suspend-resume (SIP)
To enable SIP Suspend and Resume functionality, use the suspend-resume command in SIP user agent configuration mode. To disable SIP Suspend and Resume functionality, use the no form of this command.
suspend-resume
no suspend-resume
Syntax Description
This command has no arguments or keywords.
Defaults
Enabled
Command Modes
SIP user agent configuration
Command History
Usage Guidelines
Session Initiation Protocol (SIP) gateways are now enabled to use Suspend and Resume. Suspend and Resume are basic functions of ISDN and ISDN User Part (ISUP) signaling procedures. A Suspend message temporarily halts communication (call hold), and a Resume message is received after a Suspend message and continues the communication.
Examples
The following example disables Suspend and Resume functionality:
Router(config)# sip-uaRouter(config-sip-ua)# no suspend-resumeRelated Commands
show sip-ua status
Displays SIP UA status.
sip-ua
Enables the SIP user-agent configuration commands.
