Cisco IOS Voice Command Reference

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: loopstart

Command 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.

•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.

•8-digit transmission (KP-npd-nxx-xxxx-st):

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 NPD values greater than 3 are reserved for signifying error conditions.

•10-digit transmission (kp-0-npa-nxx-xxxx-st):

The E.164 number is fully transmitted.

•20-digit transmission (kp-0-npa-nxx-xxxx-st-kp-yyy-yyy-yyyy-st):

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

•kp-2-st transmission (kp-2-st):

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 Emergency 911 calls are not rejected for an ANI mismatch. The call establishes a voice path. The E911 network, however, does not receive the ANI.

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 1/1/1

 signal groundstart

The following example configures a ten-digit ANI transmission:

Router(config)# voice-port 1/0/0

Router(config-voiceport)# signal cama kp-0-npa-nxx-xxxx-st

The 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-ST 

Related 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

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 1/17

Router (config-voiceport)# signal did immediate-start

signal 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

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 10

 signal keepalive 3

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

Related 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:

•If the router detects an oos transmit signaling pattern sent from the PBX, the router transmits the oos transmit signaling pattern to the network.

•If the signal timing oos timeout timer expires and the router receives no signaling packets from the network (network is OOS), the router sends an oos receive signaling pattern to the PBX. (The oos receive pattern is not matched against the signaling packets received from the network; the receive packets indicate an OOS condition directly by setting the AIS alarm indication bit in the packet.)

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 
 exit

dial-peer voice 100 vofr

 voice-class permanent 10

The 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 0001

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

The 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 
 exit

dial-peer voice 100 vofr

 voice-class permanent 10

The 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 0001

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

Related 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 10

 signal-keepalive 3

 signal sequence oos idle-only

 signal timing idle suppress-voice

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

 signal-type transparent

Related 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 [resume-voice [milliseconds]]

no signal timing idle suppress-voice seconds [resume-voice [milliseconds]]

Syntax Description

Command Default

No signal timing idle suppress-voice timer is configured.

Command Modes

Voice-class configuration (config-voice-class)

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.

In some circumstances, the default delay of 500 ms between the detection of incoming seizure and the opening of the audio path may cause a timing issue.

If, during this delay of 500 ms, the near-end originating PBX has already received the acknowledgement from the far-end PBX to begin playing out digits and the audio path is not yet open, the first Dual Tone Multi-Frequency (DTMF) digit might be lost over the permanent trunk.

This loss of the first DTMF digit can occur if a Cisco voice gateway has the following trunk conditioning setting:

!

voice class permanent 1

signal pattern idle transmit 0000

signal pattern idle receive 0000

signal pattern oos transmit 1111

signal pattern oos receive 1111

signal timing idle suppress-voice 10

!

The resume-voice milliseconds option has been added in Release 12.4(15)T10 to modify the delay timer and reduce the wait time. We recommend that you specify a delay of less than 500 ms to avoid the loss of any digits due to the possible discrepancy between the detection of incoming seizure and the opening of the audio path.

The output of the show voice trunk-conditioning supervisory command has been modified in Release 12.4(15)T10 to report values for the suppress-voice and resume-voice keywords (of the signal timing idle suppress-voice command) as the "idle = seconds" and "idle_off = milliseconds" fields, respectively.

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 10

 signal keepalive 3 
 signal pattern idle receive 0101 
 signal pattern idle transmit 0101 
 signal timing idle suppress-voice 5

 exit

 dial-peer voice 100 vofr

 voice-class permanent 10

 signal-type transparent

The 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 10

 signal keepalive 3 
 signal timing idle suppress-voice 5

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

 signal-type transparent

Related 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 10

 signal-keepalive 3 
 signal pattern oos receive 0001 
 signal pattern oos transmit 0001 
 signal timing oos timeout 60

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

Related 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

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 10

 signal-keepalive 3 
 signal pattern oos receive 0001 
 signal pattern oos transmit 0001 
 signal timing oos timeout 60

 signal timing oos restart 30

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

Related 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

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:

•The signal timing oos timeout command is enabled, which controls the start time for the OOS state. The timer for the signal timing oos slave-standby command does not start until the trunk is OOS.

•The voice port is configured as a slave port with the connection trunk digits answer-mode command.

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 1/0/0

 connection trunk 5559262 answer-mode

 exit

 voice-class permanent 10

 signal-keepalive 3 
 signal pattern oos receive 0001 
 signal pattern oos transmit 0001 
 signal timing oos timeout 60

 signal timing oos slave-standby 120

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

Related 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

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 10

 signal-keepalive 3 
 signal pattern oos receive 0001 
 signal pattern oos transmit 0001 
 signal timing oos timeout 60

 signal timing oos suppress-all 60

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

Related 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

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 10

 signal-keepalive 3 
 signal pattern oos receive 0001 
 signal pattern oos transmit 0001 
 signal timing oos timeout 60

 signal timing oos suppress-voice 60

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

Related 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 10

 signal-keepalive 3 
 signal pattern oos receive 0001 
 signal pattern oos transmit 0001 
 signal timing oos timeout 60

 exit

dial-peer voice 100 vofr

 voice-class permanent 10

Related 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 voip

Router(conf-voi-serv)# signaling forward unconditional

Router(conf-voi-serv)# ^Z

Router# show running-config

Building configuration...

Current configuration : 4201 bytes

!

version 12.2

service config

no service single-slot-reload-enable

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

service internal

service udp-small-servers

!

hostname as5300-2

!

no logging buffered

logging rate-limit console 10 except errors

aaa new-model

!

.

.

.

!

voice service voip

 signaling forward unconditional

 h323

!

.

.

.

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 voip

Router(conf-voi-serv)# signaling forward unconditional

Router(conf-voi-serv)# ^Z

Router# show running-config

Building configuration...

Current configuration : 4201 bytes

!

version 12.2

service config

no service single-slot-reload-enable

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

service internal

service udp-small-servers

!

hostname as5300-2

!

no logging buffered

logging rate-limit console 10 except errors

aaa new-model

!

.

.

.

!

voice service voip

 signaling forward unconditional

 h323

!

.

.

.

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 vofr

 signal-type ext-signal

 exit

Related 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

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 pots

 destination-pattern 5551111

 port 1

 no call-waiting

 ring 0

 volume 4

 caller-number 3334444 ring 1

 subaddress 20

 silent-fax

dial-peer voice 2 pots

 destination-pattern 5552222

 port 2

 no call-waiting

 ring 0

 volume 2

 caller-number 3214567 ring 2

 subaddress 10

sip

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 voip

Router(config-voi-srv)# sip

Router(conf-serv-sip)# bind control source-interface FastEthernet 0

Related Commands

sip-header

To specify the Session Initiation Protocol (SIP) header to be sent to the peer call leg, use the sip-header command in voice class configuration mode. To disable the configuration, use the no form of this command.

sip-header {sip-req-uri | header-name}

no sip-header {sip-req-uri | header-name}

Syntax Description

Command Default

SIP header is not sent to the peer call leg.

Command Modes

Voice class configuration (config-class)

Command History

Usage Guidelines

Use the sip-header command to configure Cisco UBE to pass the unsupported parameters present in a mandatory header from one peer call leg to another of a Cisco UBE.

Examples

The following example shows how to configure Cisco UBE to send a "From" header to the peer call leg:

Router(config)# voice class sip-copylist 2

Router(config-class)# sip-header From

Related 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:ipv4-address | ipv6:[ipv6-address][:port-num]}

no sip-server

Syntax Description

Command Default

No network address is configured.

Command Modes

SIP user-agent configuration (conf-serv-sip)

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 a Cisco SIP proxy server (SPS) is present in the network. With an SPS, you can configure the SIP server option and have the interested dial peers use the SPS by default.

To reset this command to a null value, use the default command.

To configure an IPv6 address, the user must enter brackets [ ] around the IPv6 address.

Examples

The following example, beginning in global configuration mode, sets the global SIP server interface to the DNS hostname "3660-2.sip.com." If you also use the session target sip server command, you need not set the DNS hostname for each individual dial peer.

sip-ua

 sip-server dns:3660-2.sip.com

dial-peer voice 29 voip

 session target sip-server

The following example sets the global SIP server interface to an IPv4 address:

sip-ua

 sip-server ipv4:10.0.2.254 

The following example sets the global SIP server interface to an IPv6 address. Note that brackets were entered around the IPv6 address:

sip-ua

 sip-server ipv6:[2001:0DB8:0:0:8:800:200C:417A]

Related 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.

Command Default

If this command is not enabled, no SIP user-agent configuration commands can be entered.

Command Modes

Global configuration (config)

Command History

Usage Guidelines

Use this command to enter SIP user-agent configuration mode. Table 230 lists the SIP user-agent configuration mode commands.

Examples

The following example, beginning in global configuration mode, shows how to enter SIP user-agent configuration mode, configure the SIP user agent, and then return to global configuration mode:

Router# sip-ua

Router(sip-ua)# retry invite 2

Router(sip-ua)# retry response 2

Router(sip-ua)# retry bye 2

Router(sip-ua)# retry cancel 2

Router(sip-ua)# sip-server ipv4:10.0.2.254

Router(sip-ua)# timers invite-wait-100 500

Router(sip-ua)# exit

Router#

Related 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 voip

 snmp enable peer-trap poor-qov

Related Commands

soft-offhook

To enable stepped off-hook resistance during seizure, use the soft-offhook command in voice-port (FXO) 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

Voice-port (FXO) configuration (config-voiceport)

Command History

Usage Guidelines

An off-hook indication into a 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 the potential to deliver high current.

The soft-offhook command applies to the following FXO interface cards (which use the 3050i chipset):

•EM-HDA-3FXS/4FXO (EVM-HD-8FXS/DID, FXO ports only)

•EM-HDA-6FXO (on EVM-HD-8FXS/DID)

•EM2-HDA-4FXO (NM-HDA-4FXS network module only)

•VIC2-4FXO, VIC2-2FXO

Examples

The following example shows a sample configuration session to enable stepped off-hook resistance during seizure on voice port 1/0/0 on a Cisco 3725 router:

Router# configure terminal

Enter configuration commands, one per line.  End with CNTL/Z.

Router(config)# voice-port 1/0/0 

Router(config-voiceport)# soft-offhook

Router(config-voiceport)# shutdown

Router(config-voiceport)#

Nov  3 11:08:53.313 EST: %LINK-3-UPDOWN: Interface Foreign Exchange Office 1/0/0, changed 
state to Administrative Shutdown 

Router(config-voiceport)# no shutdown 

Router(config-voiceport)# 

Nov  3 11:08:58.290 EST: %LINK-3-UPDOWN: Interface Foreign Exchange Office 1/0/0, changed 
state to up

Router(config-voiceport)# ^Z 

Router# 

Nov  3 11:09:01.086 EST: %SYS-5-CONFIG_I: Configured from console by console 

Router#

Related 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 voice

 source carrier-id 4321

Related 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 voice

 source trunk-group-label 21

Related 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

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 7

Router(stcapp-fsd)# speed dial from 7 to 2

To 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-dial

Router(stcapp-fsd)# prefix ##

Router(stcapp-fsd)# speed dial from 2 to 7

Router(stcapp-fsd)# exit

The 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-dial

    prefix ##

    redial ###

    speeddial number of digit(s) 1

    voicemail ##0

    speeddial1 ##2

    speeddial2 ##3

    speeddial3 ##4

    speeddial4 ##5

    speeddial5 ##6

    speeddial6 ##7

The 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-dial

Router(stcapp-fsd)# prefix **

Router(stcapp-fsd)# digit 2

Router(stcapp-fsd)# speed dial from 12 to 17

Router(stcapp-fsd)# exit

Related 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)# srtp

The following example enables secure calls and allows call fallback to nonsecure mode:

Router(config-dial-peer)# srtp fallback

The following example defaults call security to global level SRTP behavior:

Router(config-dial-peer)# srtp system

Related Commands

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)# srtp

The following example enables call fallback to nonsecure mode:

Router(config-voi-serv)# srtp fallback

Related Commands

srv version

To generate Domain Name System Server (DNS SRV) queries with either the 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

Defaults

2 (RFC 2782 style)

Command Modes

SIP UA configurationn (config-sip-ua)

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 configuring 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-ua

Router(config-sip-ua)# srv version 2

Related Commands