Name

Enter a unique name for the analog interface configuration. The name can contain any characters.

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Description

Enter a description of the analog interface configuration.

description string

Choose a group of voice interface FXO or FXS analog ports to be provisioned.

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Use DSP

Check this check box if you want to allow local calls between analog ports on the same device to use the built-in DSPs.

Default: Unchecked

no local-bypass

Module Location

Choose the slot and sub-slot location for the group of analog ports to be provisioned.

For a list of supported modules, see Supported Devices for Cisco Unified Voice Services using the Workflow Library or Configuration Groups.

voice-card slot/subslot

Add Basic

Click to configure the basic options for the group of analog ports.

You can add multiple instances of these options so that you can configure different basic options for different ports.

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Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

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Signal Type

Choose the signal type that indicates an on-hook or off-hook condition for calls that the ports receive.

Options are LoopStart, GroundStart, and DID. The DID option is available only for FXS voice interface templates.

signal {groundstart | loopstart}

DID Signal Mode

Applies only if you choose DID for an FXS voice interface template.

Choose the mode for the DID signal type

Options are Delay Dial, Immediate, and Wink Start.

signal did {delay-dial | immediate | wink-start}

Shutdown

Enable this option to shut down ports that are not being used.

shutdown

Description

Enter a description of this basic configuration.

description string

Click the Recycle Bin icon to delete the corresponding Basic options instance.

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Add Station ID

Click to configure the station name and station number from which caller ID information is sent.

You can add multiple instances of these options so that you can configure different station ID options for different ports.

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Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

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Station Name

Enter the name of the station.

The station name can contain up to 50 letters, numbers, spaces, dashes (-), and underscores (_).

station-id name name

Station Number

Enter the phone number of the station in E.164 format.

For example: 4085550111

The station number can contain up to 15 numbers.

station-id number number

Click the Recycle Bin icon to delete the corresponding Station ID options instance.

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Line Params

Click and configure options for adjusting voice and tone parameters for the port or ports.

You can add multiple instances of these options so that you can configure different line parameters options for different ports.

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Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

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Gain

Enter the amount of gain, in decibels (dB), for voice input.

Range: Integers –6 through 14

Default: 0

input gain decibels

Attenuation

Enter the amount of attenuation, in dB, for transmitted voice output.

Range: Integers –6 through 14

Default: 0

output attenuation decibels

Echo Canceller

Choose Enable to apply echo cancellation to voice traffic.

This option is disabled by default.

echo-cancel enable

Voice Activity Detection (VAD)

Choose Enable to apply VAD to voice traffic.

This option is disabled by default.

vad

Compand Type

Choose the companding standard to be used to convert between analog and digital signals in PCM systems.

Options are U-law and A-law.

compand-type {u-law | a-law}

Impedance

Choose the terminating impedance for calls.

Default: 600r

impedance {600c | 600r 900c | 900r | complex1 | complex2 | complex3 | complex4 | complex5 | complex6}

Call Progress Tone

Choose the locale for the call progress tone.

cptone locale

Click the Recycle Bin icon to delete the corresponding Line Params options instance.

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Tuning Params

Appears only when the Signal Type option in the Basic tab is configured as LoopStart or GroundStart.

Click to configure the options for various tuning parameters.

You can add multiple instances of these options so that you can configure different tuning parameter options for different ports.

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Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

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Pre Dial Delay

Applies only to FXO voice interface templates.

Enter the time, in seconds, of the delay on the FXO interface between the beginning of the off-hook state and the initiation of DTMF signaling.

Range: Integers 0 through 10

Default: 1

pre-dial-delay seconds

Supervisory Disconnect

Applies only to FXO voice interface templates.

Choose the type of tone that indicates that a call has been released and that a connection should be disconnected:

• Signal: A disconnect signal indicates a supervisory disconnect

• Anytone: Any tone indicates a supervisory disconnect

• Dualtone: A dual tone indicates a supervisory disconnect

Default: signal

• Anytone:

  supervisory disconnect anytone

• Signal:

  supervisory disconnect

• Dualtone:

  supervisory disconnect dualtone {mid-call | pre-connect}

Dial Type

Applies only to FXO voice interface templates.

Choose the dialing method for outgoing calls:

• dtmf: Dual-tone multifrequency dialer

• pulse: Pulse dialer

• mf: Multifrequency dialer

Default: dtmf

dial-type {dtmf | pulse | mf}

Timing Sup-Disconnect

Applies only to FXO voice interface templates.

Enter the minimum time, in milliseconds (ms), that is required to ensure that an on-hook indication is intentional, and not an electrical transient on the line, before a supervisory disconnect occurs.

Range: Integers 50 through 1500

Default: 350

timing sup-disconnect milliseconds

Battery Reversal

Applies only to FXO voice interface templates.

Battery reversal reverses the battery polarity on a PBX when a call connects, then changes the battery polarity back to normal when the far-end disconnects. Choose one of the following options. If you choose Detection Delay or Both, enter a value, in ms, of the delay time after which the port acknowledges a battery-reversal signal.

• Answer: Configures the port to support answer supervision by detection of battery reversal

• Detection Delay: Configures the delay time after which the card acknowledges a battery-reversal signal

• Both: Configures answer and detection delay behavior

Detection delay range: Integers 0 through 800

Detection delay default: 0 (no delay)

battery-reversal [answer]

battery-reversal-detection-delay milliseconds

Timing Hookflash Out

Applies only to FXO voice interface templates.

Enter the duration, in ms, of the hookflash indications that the gateway generates on the FXO interface.

Range: Integers 50 through 1550

Default: 4000

timing hookflash-out milliseconds

Timing Guard Out

Applies only to FXO voice interface templates.

Enter the time, in ms, after a call disconnects before another outgoing call is allowed.

Range: Integers 300 through 3000

Default: 2000

timing guard-out milliseconds

Timing Hookflash In

Applies only to FXS voice interface templates.

Enter the minimum and maximum duration, in ms, for an on-hook condition to be interpreted as a hookflash by the FXS card.

Range for minimum duration: 0 through 400

Default minimum range value: 50

Range for maximum duration: 50 through 1500

Default maximum range value: 1000

timing hookflash-in maximum-milliseconds minimum-milliseconds

Pulse Digit Detection

Applies only to FXS voice interface templates.

Enable this option to enable pulse digit detection at the beginning of a call.

Default: Enabled

pulse-digit-detection

Loop Length

Applies only to FXS voice interface templates.

Choose the length for signaling on FXS ports (Long or Short).

Default: Short

loop-length [long | short]

Ring Frequency

Applies only to FXS voice interface templates.

Choose the frequency, in Hz, of the alternating current that, when applied, rings a connected device.

Default: 23

ring frequency number

DC Offset

Applies only to FXS voice interface templates when Loop Length is set to Long.

Choose the voltage threshold below which a ring does not sound on devices.

Options are 10-volts, 20-volts, 24-volts, 30-volts, and 35-volts.

ring dc-offset number

Ringer Equivalence Number (REN)

Applies only to FXS voice interface templates.

Choose the REN for calls that the port processes. This number specifies the loading effect of a telephone ringer on a line.

Range: Integers 1 through 5

Default: 1

ren number

Click the Recycle Bin icon to delete the corresponding Tuning options instance.

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Add DID Timer

Appears only to FXS voice interface templates when the Signal Type option in the Basic tab is configured as DID.

Click to configure the options for timers for DID calls.

You can add as multiple instances of these options so that you can configure different DID timer options for different ports.

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Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

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Wait before Wink

Enter the amount of time, in ms, that the port waits after receiving a call before sending a wink signal to notify the remote side that it can send DNIS information.

Range: Integers 100 through 6500

Default: 550

timing wait-wink milliseconds

Wink Duration

Enter the maximum amount of time, in ms, of the wink signal for the port.

Range: Integers 50 through 3000

Default: 200

timing wait-duration milliseconds

Clear Wait

Enter the minimum amount of time, in ms, between an inactive seizure signal and the call being cleared for the port.

Range: Integers 200 through 2000

Default: 400

Dial Pulse Min Delay

Enter the amount of time, in ms, between wink-like pulses for the port.

Range: Integers 0, or 140 through 2000

Default: 140

timing dial-pulse min-delay milliseconds

Answer Winkwidth

Enter the minimum delay time, in ms, between the start of an incoming seizure and the wink signal.

Range: Integers 110 through 290

Default: 210

timing answer-winkwidth milliseconds

Click the Recycle Bin icon to delete the corresponding DID Timer options instance.

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Caller ID

Click to configure the options for enabling caller ID for the port or ports.

Caller ID is an analog service by which a telephone central office switch sends digital information about an incoming call. The Caller ID feature for analog FXS ports is configurable on a per-port basis to phones that are connected to analog FXS voice ports. Caller ID also is available on analog FXO ports. Caller ID-related features are based on the identity of the calling party.

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Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

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Caller ID Mode

Choose a noncountry, standard caller ID mode for a receiving FXO or a sending FXS voice port:

• BT: Frequency-Shift Keying (FSK) with Dual Tone Alerting Signal (DTAS) used by British Telecom

• FSK: FSK before or during a call

• DTMF: DTMF digits with the start and end digit codes

caller-id mode {BT | FSK | DTMF}

DTMF Start

Applies only if you choose DTMF for the caller ID mode.

Choose the character that indicates the start of a DTMF string.

caller-id mode {dtmf {start | end}{# | * | A | B | C | D}}

DTMF End

Applies only if you choose DTMF for the caller ID mode.

Choose the character that indicates the end of a DTMF string.

caller-id mode {dtmf {start | end}{# | * | A | B | C | D}}

Alerting Options

Choose the alerting method for on-hook caller ID information:

• Line-Reversal: Sets the line-reversal alerting method for caller ID information for an on-hook (Type 1) caller ID at a sending FXS voice port and for an on-hook caller ID at a receiving FXO voice port.

• Pre-ring: Sets a 250 ms pre-ring alerting method for caller ID information for an on-hook (Type 1) caller ID at a sending FXS and a receiving FXO voice port.

• Ring 1, Ring 2, Ring 3, or Ring 4: Sets the ring-cycle method for receiving caller ID information for an on-hook (Type 1) caller ID at a receiving FXO or a sending FXS voice port.

caller-id alerting {line-reversal | pre-ring | ring {1 | 2 | 3 | 4}}

DSP Pre-Allocate Alerting

Applies only to FXO voice interface templates.

Enable this option to statically allocate a DSP voice channel for receiving caller ID information for an on-hook (Type 1) caller ID at a receiving FXO voice port.

caller-id alerting dsp pre-allocate

Caller ID Block

Applies only to FXS voice interface templates.

Enable this option to request blocking of caller ID information display at the far end of a call that originates from an FXS port.

caller-id block

Caller ID Format E911

Applies only to FXS voice interface templates.

Enable this option to use the enhanced 911 format for calls that are sent on the FXS port.

caller-id format e911

Click the Recycle Bin icon to delete the corresponding Caller ID options instance.

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Connection Plar

Click to configure the options for the connection Private Line Automatic Ringdown (PLAR).

You can add multiple instances of these options so that you can configure different connection PLAR options for different ports.

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Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

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Connection Plar Pattern

Enter the PLAR extension to which the selected ports forward inbound calls.

connection plar digits

OPX

Applies only to FXO voice interface templates.

Check this check box to enable Off-Premises Extension for the PLAR extension.

connection plar opx digits

Click the Recycle Bin icon to delete the corresponding Connection Plar options instance.

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Association

Click to configure options for associating other configured UC voice features with the port or ports. When you associate a feature in this way, the configuration options in that feature are applied to the designated ports.

You can add multiple instances of these options so that you can configure different association options for different ports.

Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port 1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 to specify ports 1 through 5.

Trunk Group

Choose a configured Trunk Group feature to associate with the port.

Trunk Group Priority

Enter the priority of the trunk group. The number you enter is the priority of the POTS dial peer in the trunk group for incoming and outgoing calls.

Range: Integers 1 through 64

Translation Profile

Choose a configured Translation Profile feature to associate with the port.

Translation Profile Direction

Choose the direction of the traffic to which to apply the selected Translation Profile feature:

• Incoming: Applies the corresponding Translation Profile feature to traffic that is incoming to the port

• Outgoing: Applies the corresponding Translation Profile feature to traffic that is outgoing from the port

Supervisory Disconnect

Applies only to FXO voice interface templates.

Choose a configured Supervisory Disconnect feature to associate with the port.

Click the Recycle Bin icon to delete the corresponding Association options instance.

Name

Enter a unique name for the call routing configuration. The name can contain any characters.

Description

Enter a description of the call routing configuration.

Enter a unique number to be pretended to a dial peer tag to ensure that the dial peer tag can be uniquely identified across this feature.

Enter a description of the analog or digital interface configuration to which this call routing configuration is to be associated.

Choose the Analog or Digital Interface feature to which the POTS dial peer call routing port-related configuration is to be associated.

Add Dial Peers

Click to add a dial peer to a dial plan. Configure the following options in the Add Dial Peer dialog box, then click Save

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Add Dial Peer Dialog Box Options

Tag

Enter a number to be used to reference the dial peer.

Range: Integers 1 through 214748364

dial-peer voice number {pots | voip}

Choose the type of dial peer that you are creating. Options are pots and sip.

dial-peer voice number {pots | voip}

Choose the direction of traffic on the dial peer. Options are incoming and outgoing.

• Incoming:

  dial-peer voice  number {pots | voip}

  incoming called-number string

• Outgoing:

  dial-peer voice number {pots | voip}

  destination-pattern string

Enter a description of the dial peer.

description

Enter the string that the router uses to match incoming calls to the dial peer.

Enter the string as an E.164 format regular expression in the following form:

(ipv6:\[([0-9A-Fa-f.:])+\](:[0-9]+)?))

• Incoming:

  dial-peer voice  number {pots | voip}

  incoming called-number string

• Outgoing:

  dial-peer voice number {pots | voip}

  destination-pattern string

Applies only when Dial peer type is configured as pots and direction is configured as outgoing.

Choose how the dial peer transmits digits in outgoing numbers:

• all: The dial peer transmits all digits

• none: The dial peer does not transmit digits that do not match the destination pattern

• some: The dial peer transmits the specified number of right-most digits

Default: none

• All:

  dial-peer voice  number pots

  forward-digits all

• None:

  dial-peer voice  number pots

  forward-digits 0

• Some:

  dial-peer voice  number pots

  forward-digits number

Applies only when you choose Some for Forward Digits Type.

Enter the number of right-most digits in the outgoing number to transmit.

For example, if you set this option to 7 and the outgoing number is 1112223333, the dial peer transmits 2223333.

dial-peer voice  number pots

forward-digits number

Applies only when Dial peer type is configured as pots and direction is configured as outgoing.

Enter a string to be pretended to the dial string for outgoing calls.

Valid values: Integers 0 through 9 and comma (,)

dial-peer voice  number pots

prefix string

Applies only when Dial peer type is configured as sip. Choose the transport protocol for SIP control signaling.

Options are tcp and udp.

dial-peer voice number voip

session transport {tcp | udp}

Enter the preference of the dial peer.

If dial peers have the same match criteria, the system uses the one with the highest preference value.

Range: Integers 0 through 10

Default: 0

dial-peer voice number voip

preference value

dial-peer voice number pots

preference value

Applies only when Dial peer type is configured as pots.

Enter the voice port that the router uses to match calls to the dial peer. For an analog port, enter the port you want. For a digital T1 PRI ISDN port, enter a port with the suffix 23. For a digital E1 PRI ISDN port, enter a port with the suffix 15.

For an outgoing dial peer, the router sends the calls that match the dial peer to this port.

For an incoming dial peer, this port serves as an additional match criterion. The dial peer is matched only if a call comes in on this port.

dial-peer voice number pots

• For an analog port:

  port slot/subslot/port

• For a digital port:

  port slot/subslot/port:15

  port slot/subslot/port:23

Destination Address

Applies only when Dial peer type is configured as sip and direction is configured as outgoing.

Enter the network address of the remote voice gateway to which calls are sent after a local outgoing SIP dial peer is matched.

Enter the address in one of these formats:

• dns:hostname.domain

• sip-server

• ipv4:destination-address

• ipv6:destination-address

session target {ipv4:destination-address | ipv6:destination-address |  sip-server | dns:hostname.domain}

Dial Peer File Options

Download Dial Peer List

To create or edit a dial peer CSV file, click this option to download the Cisco provided file named Dial-Peers.csv.

The first time that you download this file, it contains field names but no records. Update this file as needed by using an application such as Microsoft Excel. For detailed information about this file, see Dial Peer CSV File.

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Upload Dial Peer List

To import configuration information from a dial peer CSV file that you have created, click this option, choose the file to upload, then click Save.

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Click Edit to edit the corresponding Dial Peer options instance. Click Delete to delete the corresponding Dial Peer options.

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Add Fax Protocol

Click to configure the options for the fax protocol capability for a SIP dial peer endpoint.

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Dial Peer Range

Enter the tag or tags of the SIP dial peers for which to enable fax options.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify SIP dial peer tag 1; 1,2,3 to specify tags 1, 2, and 3; or 1-5 so specify tags 1 through 5.

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Primary Protocol

Choose a set of fax protocol options. Each option is a bundled set of related fax commands.

For a detailed description of each bundle, see the “Primary Fax Protocol Command Bundles” table in Configure SIP Dial Peers for a Voice Policy.

The descriptions of the bundles include the following components:

• nse: Uses NSEs to switch to T.38 fax relay mode

• force: Unconditionally uses Cisco Network Services Engines (NSE) to switch to T.38 fax relay

• version: Specifies a version for configuring fax speed:

  • 0: Configures version 0, which uses T.38 version 0 (1998–G3 faxing)

  • 3: Configures version 3, which uses T.38 version 3 (2004–V.34 or SG3 faxing)

• none: No fax pass-through or T.38 fax relay is attempted

• Pass-through: The fax stream uses one of the following high-bandwidth codecs:

  • g711ulaw: Uses the G.711 ulaw codec

  • g711alaw: Uses the G.711 alaw codec

fax protocol { none | pass-through {g711ulaw | g711alaw} [fallback none] | t38 [nse [force]] [version {0 | 3}] [ls-redundancy value [hs-redundancy value]] [fallback {none | pass-through {g711ulaw | g711alaw}}]

Fallback Protocol

Available when the primary protocol bundle name that you selected in the Primary Protocol field begins with “T.38” or “Fax Pass-through."

Choose the fallback mode for fax transmissions. This fallback mode is used if the primary fax protocol cannot be negotiated between device endpoints.

For a detailed description of each option, see the "Fallback Protocol Options” table in Configure SIP Dial Peers for a Voice Policy.

fax protocol {none | pass-through {g711ulaw | g711alaw} [fallback none] | t38 [nse [force]] [version {0 | 3}] [ls-redundancy value [hs-redundancy value]] [fallback {none | pass-through {g711ulaw | g711alaw}}]}

Low Speed Redundancy

Available when the primary protocol bundle name that you selected in the Primary Protocol field begins with “T.38.”

Enter the number of redundant T.38 fax packets to be sent for the low-speed V.21-based T.30 fax machine protocol.

Range: Integers 0 (no redundancy) to 5

Default: 0

ls-redundancy value

High Speed Redundancy

Available when the primary protocol bundle name that you selected in the Primary Protocol field begins with “T.38.”

Enter the number of redundant T.38 fax packets to be sent for high-speed V.17, V.27, and V.29 T.4 or T.6 fax machine image data.

Range: Integers 0 (no redundancy) to 2

Default: 0

hs-redundancy value

Click the Recycle Bin icon to delete the corresponding Fax options instance.

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Add Modem Passthrough

Click to configure the modem pass-through feature for a SIP dial peer endpoint.

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Dial Peer Range

Enter the tag or tags of the SIP dial peers for which to enable modem options.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify SIP dial peer tag 1; 1,2,3 to specify tags 1, 2, and 3; or 1-5 so specify tags 1 through 5.

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Protocol

Choose the protocol for the modem pass-through:

• None: Modem pass-through is disabled on the device

• NSE G.711ulaw: Uses named signaling events (NSEs) to communicate G.711 ulaw codec switchover between gateways

• NSE G.711alaw: Uses named NSEs to communicate G.711 alaw codec switchover between gateways

• None:

  no modem passthrough

• NSE G.711ulaw:

  modem passthrough nse codec g711ulaw

  NSE G.711alaw:

  modem passthrough nse codec g711alaw

Click the Recycle Bin icon to delete the corresponding Modem options instance.

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Association

Click to configure the following options for associating other configured UC features with the dial plan. When you associate a feature in this way, the configuration options in that feature are applied to the designated POTS or SIP dial peers.

You can add multiple instances of these options so that you can configure different association options for different ports.

Dial Peer Range

Enter the dial peer or peers to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify dial peer 1; 1,2,3 to specify dial peers 1, 2, and 3; or 1-5 to specify dial peers 1 through 5.

Media Profile Name

Choose a configured Media Profile feature to associate with the SIP dial peer.

Server Group

Choose a configured Server Group feature to associate with the SIP dial peer.

Trunk Group

Choose a configured Trunk Group feature to associate with the POTS dial peer.

Trunk Group Priority

Enter the priority of the trunk group, which specifies the priority of the POTS dial peer in the trunk group for incoming and outgoing calls.

Range: Integers 1 through 64

Translation Profile

Choose a configured Translation Profile feature to associate with the port.

Translation Profile Direction

Choose the direction of the traffic to which to apply the selected Translation Profile feature:

• Incoming: Applies the corresponding Translation Profile feature to traffic that is incoming to the port

• Outgoing: Applies the corresponding Translation Profile feature to traffic that is outgoing from the port

Voice Tenant

Choose a configured Voice Tenant feature to associate with the port.

Click the Recycle Bin icon to delete the corresponding Association options instance.

Name

Enter a unique name for the DSP farm configuration. The name can contain any characters.

Description

Enter a description of the DSP farm configuration.

Services

Click to configure options for a DSP farm service.

DSP farm services are conferencing, transcoding, and media termination point (MTP).

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DSP Services

Enable this option to use hardware DSP resources.

Disable this option if the device does not have any hardware DSP resources and you want to use the software MTP DSP service.

This option is enabled by default.

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Module Location

If Services is enabled, choose the slot and sub-slot location for the hardware DSP.

You can configure as many module locations as needed.

For a list of supported modules, see Configure UC Voice Services Using the Workflow Library or Configuration Groups.

voice-card slot/subslot

dsp service dspfarm

SCCP

Check this check box to enable the SCCP application for provisioning conference, transcoding, and MTP services. Then configure the Profile options as described in the following table.

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CUBE

Check this check box to enable the CUBE application for provisioning inbox transcoding services. Then configure the Profile options as described in the following table.

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Click the Recycle Bin icon to delete the corresponding Services options instance.

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Add Profile

Click and, in the Profile dialog box for a DSP farm profile, configure the options that this table describes. Click Add in the dialog box to add the profile to the table of profiles.

A profile includes options for provisioning a specific DSP farm service type, which can be transcoding, conferencing, or MTP. A profile is associated with either the SCCP application or CUBE, which invokes the resources for a service as needed.

You can add multiple instances of these options so that you can configure different profile options for as needed.

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Profile ID

Displays a unique system-generated identifier for the profile.

profile-identifier

Application

Choose the application with which to associate the profile. Options are sccp and cube.

associate application{sccp | cube}

Profile Type

For the sccp application, choose the service type to provision. Options are transcode, conference, and mtp.

For the cube application, transcode is selected automatically as the service to provision.

dspfarm profile profile-identifier { conference | mtp | transcode}

Transcode Universal Profile Type

For the transcode profile type, check this check box to allow transcoding between codecs of any type.

When this check box is unchecked, transcoding is allowed only between the G.711 codec and other codecs.

dspfarm profile profile-identifier transcode [universal]

MTP Type Hardware

For the mtp profile type, check this check box to have MTP translations and conversions performed by the hardware DSP resources.

maximum session hardware

MTP Type Software

For the mtp profile type, check this check box to have MTP translations and conversions performed by the router CPU.

maximum session software

Profile Name

For the transcode or conference profile type for the sccp application, or for the cube application, enter a unique name that you can use to identify the profile.

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Codec List

Choose the codecs to be available for the DSP farm service that this profile defines.

For the mtp profile type, you can choose pass-through and one other option. To change a codec, remove the current one before choosing a new one.

The following codecs are supported:

• For the transcode profile type:

  • g711alaw

  • g711ulaw

  • g729abr8

  • g729ar8

  • g729br8

  • g729r8

  • g722-64

  • ilbc

  • iSAC

  • opus

  • pass-through

• For the conference profile type:

  • g711alaw

  • g711ulaw

  • g722r-64

  • g729abr8

  • g729ar8

  • g729br8

  • g729r8

• For the mtp profile type when MTP Type Hardware or both MTP Type Hardware and MTP Type Software are chosen:

  • g711ulaw

  • g711alaw

  • pass-through

• For the mtp profile type when MTP Type Software is chosen:

  • g711ulaw

  • g711alaw

  • g722-64

  • g729abr8

  • g729ar8

  • g729br8

  • g729r8

  • ilbc

  • iSAC

  • pass-through

codec codec-name

Feature List

For the cube application, choose the features to enable for in-box transcoding.

—

Maximum Sessions

For the transcode or conference profile type, enter the maximum number of sessions that this profile can support.

This value depends on the maximum number sessions that can be configured with the DSP resources that are available on the router. These resources are based on the type of modules in the router. To determine these resources, you can use the Cisco DSP Calculator.

maximum sessions number

MTP Maximum Hardware Sessions

If you checked MTP Type Hardware, enter the maximum number of hardware sessions that can be used for MPT translations and conversions.

Range: Integers 1 through 4000

maximum session hardware number

MTP Maximum Software Sessions

If you checked MTP Type Software, enter the maximum number of CPU sessions that can be used for MRP translations and conversions.

Range: Integers 1 through 6000

maximum session software number

Shutdown

Enable this option to take this profile out of service.

shutdown

Click Edit to edit the corresponding Profile options instance. Click Delete to delete the corresponding Profile options.

—

CUCM

Click to configure the Cisco Unified Communications Manager servers to which the profiles that you define register.

You can configure up to 12 Cisco Unified Communications Manager servers.

—

Configure Local Interface

Enter the local interface that DSP services that are associated with the SCCP application use to register with Cisco Unified Communications Manager.

Enter the interface in this format:

interface-type/interface-number/port

where:

• interface-type: Type of interface that the services use to register with Cisco Unified Communications Manager. The type can be a Gigabit Ethernet interface or a port channel interface.

• interface-number: Interface number that the services use to register with Cisco Unified Communications Manager.

• port: (Optional) Port on which the interface communicates with Cisco Unified Communications Manager. If you do not specify a port, the default value 2000 is used.

For example: GigabitEthernet0/0/0.

sccp local interface-type interface-number [port port-number]

IP Precedence

Enter the IP precedence value to be used by the SCCP application for IP packets.

Range: 1 (lowest) through 7 (highest)

Default: 5

sccp ip precedence value

Add Configure Server List

Click to display the following options for a Cisco Unified Communications Manager server:

• Server Identifier: Unique system-generated identifier of the Cisco Unified Communications Manager server

• Server IP: Enter the IP address of the Cisco Unified Communications Manager server

• Server identifier:

  identifier-number

• Server IP:

  sccp ccm {ipv4-address | ipv6-address | dns} identifier identifier-number version 7.0+

Click the Recycle Bin icon to delete the corresponding CUCM options instance.

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Add CUCM Group

Click and, in the CUCM Group dialog box, configure a Cisco Unified Communications Manager group by using the options that this table describes. Each group includes up to 4 Cisco Unified Communications Manager servers that control the DSP farm services that, in turn, are associated with the servers. Click Add in the dialog box when you are finished.

You can add multiple Cisco Unified Communications Manager groups.

—

CUCM Media Resource Name

Enter a unique name that is used to register a DSP farm profile to the Cisco Unified Communications Manager servers.

The name must contain from 6 to 15 characters. Characters can be letter, numbers, slashes (/), hyphens (-), and underscores (_).

associate ccm profile-identifier register device-name

Profile Name

Enter the name that you entered for the DSP farm profile that is to be registered to this Cisco Unified Communications Manager group.

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Server Groups Priority Order

Designate the priority in which the Cisco Unified Communications Manager servers in this Cisco Unified Communications Manager group are used.

The drop-down list displays the server identifiers of the Cisco Unified Communications Manager servers that you configured.

Choose the server that you want to be the primary server. This server has the highest priority. Then choose the server that you want to be a redundant server with the next highest priority. Continue in this way to choose other redundant servers.

The servers in the field appear in descending order of priority, with the highest priority server appearing first.

To remove a server from the field, click its X icon. To change the priority order of servers, remove the servers and add them back in the desired order.

associate ccm cisco-unified-communications-manager-id priority priority

CUCM Switchback

Choose the switchback method that the Cisco Unified Communications Manager servers in this Cisco Unified Communications Manager group use to switch back after a failover:

• guard: Switchback occurs when active sessions are terminated gracefully or when the guard timer expires, whichever happens first

• graceful: Switchback occurs after all active sessions terminate gracefully

• immediate: Performs the Cisco Unified Communications Manager switchback to the higher priority Cisco Unified Communications Manager immediately when the timer expires, whether or not there is an active connection

Default: graceful

switchback method{ graceful | guard [timeout-guard-value] | immediate}

Server Switchover

Choose the switchover method that Cisco Unified Communications Manager servers in this Cisco Unified Communications Manager group use when failing over:

• graceful: Switchover occurs after all active sessions terminate gracefully

• immediate: Switchover occurs immediately, whether or not there is an active connection

Default: graceful

switchover method {graceful | immediate}

Keep Alive Retries

Enter the number of keepalive retries from the SCCP application to Cisco Unified Communications Manager.

Range: Integers 1 to 180

Default: 3

keepalive retries number

Keep Alive Time Out

Enter the number of seconds between successive keepalive messages from the SCCP application to Cisco Unified Communications Manager.

Range: Integers 1 to 180

Default: 20

keepalive retries seconds

Bind Interface

Enter the interface to bind with the Cisco Unified Communications Manager group.

bind interface interface-name

Click Edit to edit the corresponding CUCM Group options instance. Click Delete to delete the corresponding CUCM Group options.

—

Add Basic

Click to configure basic options for the group of digital ports.

You can add as multiple instances of these options so that you can configure different basic options for different ports.

—

Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

—

Enter the number of time slots of the interface.

Ranges:

• For T1 PRI: Time slots 1 through 24. The 24th time slot is the D channel.

• For E1 PRI: Time slots 1 through 31. The 16th time slot is the D channel.

controller e1/t1 slot/sub-slot/port

pri-group timeslots timeslots-range [voice-dsp]

Line Termination

Applies to E1 voice interface templates only. Choose the termination type for the interface:

• 75-ohm: 75 ohm unbalanced termination

• 120-ohm: 120 ohm balanced termination (default)

controller e1 slot/sub-slot/port line-termination {75-ohm | 120-ohm}

Cable Length Type

Applies to T1 voice interface templates only. Choose the cable length type for the interface:

• Long: Applies to cables that are longer than 660 feet (201.2 m). Attenuates the pulse from the transmitter by using pulse equalization and line build-out.

  This value is the default

• Short: Applies to cables that are 660 feet (201.2 m) or less in length. Sets transmission attenuation for the cable.

controller t1 slot/sub-slot/port cablelength {short | long}

Cable Length

Applies to T1 voice interface templates only. Choose the length of the cable for the interface:

• For a Long cable length, enter the loss value, decibels (dB).

  Options are –7.5, –15, –22.5, and 0.

  The default value is 0.

• For a Short cable length (up to 660 feet (201.2 m), enter the value that most closely exceeds the length of the cable. For example, if the cable length is 180 feet (55 m) enter 220.

controller t1 slot/subslot/port cablelength {[short [110ft | 220ft | 330ft | 440ft | 550ft | 660ft ]] [long [-15db |-22db |-7.5db | 0db]]}

Line Code

Choose the line code type for the interface.

For a T1 voice interface template:

• ami: Use alternate mark inversion as the line code type

• b8zs: Use binary 8-zero substitution as the line code type (default)

For an E1 voice interface template:

• amiami: Use alternate mark inversion as the line code type

• hdb3: Use high-density bipolar 3-zero as the line code type (default)

linecode {ami | b8zs | hdb3

Framing

Choose the frame type for the interface.

For a T1 voice interface template:

• esf: Extended super frame (default)

• sf: Super frame

For an E1 voice interface template:

• ccr4: CRC4 framing type (default)

• no-crc4: No CRC4 framing type

controller t1 slot/sub-slot/port framing [esf | sf]

controller e1 slot/sub-slot/port framing [crc4 | no-crc4] [australia]

Framing Australia

Applies to E1 voice interface templates only. Enable this option to use the Australia framing type.

controller e1 slot/sub-slot/port framing [crc4 | no-crc4] australia

Network Side

Enable this option to have the device to which this configuration is to be associated use the standard PRI network-side interface.

Default: Disabled

interface serial slot/sub-slot/port:{15| 23}

isdn protocol-emulate [network | user]

Switch Type

Choose the ISDN switch type for this interface:

• primary-qsig: Supports QSIG signaling according to the Q.931 protocol. Network side functionality is assigned with the isdn protocol-emulate command.

• primary-4ess: Lucent (AT&T) 4ESS switch type for the United States.

• primary-5ess: Lucent (AT&T) 5ESS switch type for the United States.

• primary-dms100: Nortel DMS-100 switch type for the United States.

• primary-net5: NET5 ISDN PRI switch types for Asia, Australia, and New Zealand. ETSI-compliant switches for Euro-ISDN E-DSS1 signaling system.

• primary-ni: National ISDN switch type.

• primary-ntt: Japanese NTT ISDN PRI switches.

interface serial slot/sub-slot/port:{15 | 23}

isdn switch-type [primary-4ess | primary-5ess | primary-dms100 |  primary-net5 | primary-ni | primary-ntt | primary-qsig]

Delay Connect Timer

Enter the duration, in ms, to delay connect a PRI ISDN hairpin call.

Range: Integers 0 through 200

Default: 20

voice-port slot/sub-slot/port:{15 | 23} timing delay-connect value

Click the Recycle Bin icon to delete the corresponding Basic options instance.

—

Add ISDN Timer

Click to configure options for the ISDN timer for the interface.

You can add multiple instances of these options so that you can configure different ISDN timer options for different ports.

—

Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

—

Click to configure an ISDN timer. Configure the following fields in the ISDN Timer and Value dialog box, then click Save.

• Port Range: Displays the ports that you chose

• ISDN Timer: Displays the ISDN timers that you can provision.

• Value: Enter the value, in ms, for the corresponding ISDN timer:

  • For the T200 ISDN timer:

    • Range: 400 through 400000

    • Default for all switch types: 1000

  • For the T203 ISDN timer:

    • Range: Integers 400 through 400000

    • Default for QSIG, ETSI Net5, and DMS-100 switch types: 10000

    • Default for 4ESS, 5ESS, NTT, and NI switch types: 30000

  • For the T301 ISDN timer:

    • Range: 180000 through 86400000

    • Default for NTT and ETSI Net5 switch types: 180000

    • Default for other switch types: 300000

  • For the T303 ISDN timer:

    • Range: 400 through 86400000

    • Default for QSIG switch type: 6000

    • Default for other switch types: 4000

  • For the T306 ISDN timer:

    • Range: 400 through 86400000

    • Default for all switch types: 30000

  • For the T309 ISDN timer:

    • Range: 0 through 86400000

    • Default for all switch types when network side configuration is false (User): 90000

    • Default for all switch types when network side configuration is true (Network): 5000

  • For the T310 ISDN timer:

    • Range: 400 through 400000

    • Default for NI , 4ESS and 5ESS switch types when network side configuration is false (User): 30000

    • Default for NI, 4ESS, and 5ESS switch types when network side configuration is true (Network): 10000

    • Default for ETSI Net5 switch types: 4000

    • Default for QSIG switch type: 120000

    • Default for NTT switch type: 3000

    • Default for DMS-100 switch type when network side configuration is false (User): 1000

    • Default for DMS-100 switch type when network side configuration is true (Network): 4000

    • Default for other switch types: 4000

  • For the T321 ISDN timer:

    • Range: 0 through 86400000

    • Default for ETSI Net5 switch type: 30000

    • Default for other switch types: 40000

interface serial slot/sub-slot/port:{15| 23}

isdn timer T200 value

isdn timer T203 value

isdn timer T301 value

isdn timer T303 value

isdn timer T306 value

isdn timer T309 value

isdn timer T310 value

isdn timer T321 value

Click the Recycle Bin icon to delete the corresponding ISDN Timer options instance.

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Add ISDN Map

Click to configure the following options to override with custom values the default ISDN type and plan that the router generates.

You can add multiple instances of these options so that you can configure different ISDN mapping options for different ports.

—

Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

—

Digit Range

Enter a digit or range of digits to map to ISDN telephone numbers that are used internally

isdn map address {{ address | reg-exp} plan plan type type | transparent}

Plan

Choose an ISDN numbering plan:

• data: X.121 data numbering plan

• isdn: E.164 ISDN/Telephony numbering plan

• national: Number called to reach a subscriber in the same country, but outside the local network

• privacy: Private numbering plan

• reserved/extension: Reserved for the extension

isdn map address {{ address | reg-exp} plan plan type type | transparent}

Choose an ISDN number type:

• abbreviated: Abbreviated representation of the complete number as supported by your network

• international: Number called to reach a subscriber in another country

• national: Number called to reach a subscriber in the same country, but outside the local network

• reserved/5: Reserved for the extension

isdn map address {{ address | reg-exp} plan plan type type }

Click the Recycle Bin icon to delete the corresponding ISDN Map options instance.

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Add Shutdown

Click to configure to disable or enable the controller, serial interface, or voice port that is associated with the interface port.

You can add multiple instances of these options so that you can configure different shutdown options for different ports.

—

Port ID

Enter the port or ports to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

—

Controller

Enable this option to shut down a controller.

controller e1/t1 slot/sub-slot/port shutdown

Serial

Check this check box to shut down a serial interface.

interface serial slot/sub-slot/port:{ 15 | 23} shutdown

Check this check box to shut down a voice port.

voice-port slot/sub-slot/port:{ 15 | 23} shutdown

Click the Recycle Bin icon to delete the corresponding Shutdown options instance.

—

Add Line Params

Click to configure options for adjusting various line parameters for the port or ports.

You can add multiple instances of these options so that you can configure different line parameters for different ports .

—

Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

—

Gain

Enter the amount of gain, decibels (dB), for voice input.

Range: Integers –6 through 14

Default: 0

input gain decibels

Attenuation

Enter the amount of attenuation, decibels (dB), for transmitted voice output.

Range: Integers –6 through 14

Default: 3

output attenuation decibels

Echo Canceller

Choose Enable to apply echo cancellation to voice traffic.

This option is disabled by default.

echo-cancel enable

Voice Activity Detection

Choose Enable to apply VAD to voice traffic.

This option is disabled by default.

vad

Compand Type

Choose the companding standard to be used to convert between analog and digital signals in PCM systems (U-law or A-law).

The default is U-law.

compand-type {u-law | a-law}

Call Progress Tone

Choose the locale for the call progress tone.

cptone locale

Click the Recycle Bin icon to delete the corresponding Line Params options instance.

—

Add Outgoing IE

Click to configure the following options for the outgoing Information Element.

You can add multiple instances of these options so that you can configure outgoing Information Element options for different ports.

—

Port Range

Enter the port or ports within the voice interface template to which the following option applies.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 so specify ports 1 through 5.

—

Type

Choose one or more of the following options to specify the Information Elements to pass in outgoing ISDN messages:

To remove an option from the field, click its X icon.

• called-number: Indicates the outgoing call number

• called-subaddr: Indicates the subaddress of the outgoing call

• caller-number: Indicates the incoming call number

• caller-subaddr: Indicates the subaddress of the incoming call

• connected-number: Indicates the number of the remaining caller if a disconnect occurs during a conference

• connected-subaddr: Indicates the subaddress of the remaining caller if a disconnect occurs during a conference

• display: Provides information about the text display

• extended-facility: Provides information about extended facility requests

• facility: Provides information about facility requests

• high-layer-compat: Provides information about higher layer compatibility

• low-layer-compat: Provides information about lower layer compatibility

• network-facility: Provides information about network facility requests

• notify-indicator: Provides information about notifications

• progress-indicator: Provides information about the call in progress

• redirecting-number: Indicates the number that is redirecting the call

• user-user: Provides information about the users at either end of the call

isdn outgoing ie type

Click the Recycle Bin icon to delete the corresponding Outgoing IE options instance.

—

Association

Click to configure options for associating other configured UC voice features with the port or ports. When you associate a feature in this way, the configuration options in that feature are applied to the designated ports.

You can add multiple instances of these options so that you can configure different association options for different ports.

Port Range

Enter the port or ports within the voice interface template to which these options apply.

Enter a number, a comma separated string of numbers, or a range of numbers separated with a hyphen. For example, enter 1 to specify port 1; 1,2,3 to specify ports 1, 2, and 3; or 1-5 to specify ports 1 through 5.

Trunk Group

Choose a configured Trunk Group feature to associate with the port.

Trunk Group Priority

Enter the priority of the trunk group. The number you enter is the priority of the POTS dial peer in the trunk group for incoming and outgoing calls.

Range: Integers 1 through 64

Translation Profile

Choose a configured Translation Profile feature to associate with the port.

Translation Profile Direction

Choose the direction of the traffic to which to apply the selected Translation Profile feature:

• Incoming: Applies the corresponding Translation Profile feature to traffic that is incoming to the port

• Outgoing: Applies the corresponding Translation Profile feature to traffic that is outgoing from the port

Supervisory Disconnect

Applies only to FXO voice interface templates.

Choose a configured Supervisory Disconnect feature to associate with the port.

Click the Recycle Bin icon to delete the corresponding Associations options instance.

Name

Enter a unique name for the media profile configuration. The name can contain any characters.

—

Description

—

Media Profile Number

Enter a number for this SIP media profile.

Range: Integers 1 through 10000

voice class codec tag-number

DTMF Target

Choose the DTMF relay options that you want the system to use for SIP calls:

• rtp-nte: Real-Time Transport Protocol (RTP) Named Telephone Events (NTE). An in-band DTMF relay method, which uses RTP Named Telephony Event (NTE) packets to carry DTMF information instead of voice.

• sip-notify: A Cisco proprietary out-of-band DTMF relay mechanism that transports DTMF signals using SIP NOTIFY messages.

• sip-kpml: Keypad Markup Language (KPML) is used to indicate DTMF tones in SIP messaging. It transmits DTMF tone indications via SIP NOTIFY messages

Choose the option that you want to have the highest priority. Then choose the option that you want to have the next highest priority. Continue in this way to choose a third option.

The options in the field appear in descending order of priority, with the highest priority option appearing first.

To remove an option from the field, click its X icon. To change the priority order of options, remove the options and add them back in the desired order.

dtmf-relay {[[sip-notify] [sip-kpml] [rtp-nte]]}

Codec List

Choose the codecs that you want to be made available for the SIP trunk to use when communicating with the remote dial peer.

Choose the codec that you want to have the highest priority. Then choose the codec that you want to have the next highest priority. Continue in this way to choose other codecs.

The codecs in the field appear in descending order of priority, with the highest priority option appearing first.

To remove a codec from the field, click its X icon. To change the priority order of codecs, remove the codecs and add them back in the desired order.

voice class codec tag-number

codec preference value codec-type

Name

Enter a unique name for the SRST configuration. The name can contain any characters.

Description

Enter a description of the SRST configuration.

Max Phones

Enter the number of phones that the system can register to the local gateway when the gateway is in Cisco Unified SRST mode.

voice register global

max-pool max-voice-register-pools

Max Directory Numbers

Enter the number of directory numbers that the gateway supports when the gateway is in Cisco Unified SRST mode.

The maximum values that you can enter depend on the device that you are configuring.

voice register global

max-dn max-directory-numbers

Music on Hold

—

Music on Hold File

Enter the path and filename of the audio file for music on hold.

The file must be in the system flash and must be in the .au or .wav format. In addition, the file format must contain 8-bit 8-kHz data, for example, CCITT a-law or u-law data format.

call-manager-fallback

moh filename

System Message

Enter a message that displays on endpoints when Cisco Unified SRST mode is in effect.

voice register global

system message string

Add New Phone Pool Profile

Click to configure the options for providing registration permission control and certain dial-peer attributes that are applied to the dynamically created VoIP dial peers when SIP phone registrations match the pool

You can add multiple instances of these options so that you can configure different options for different pool tags.

—

Pool Tag

Enter the unique sequence number of the set of SIP phones to be configured.

Range: Integers 1 to the number of phones that you configured with the Max Phones option.

voice register pool pool-tag

IPv4/6 Network Access

voice register pool pool-tag

id [network address mask mask]

Click the Recycle Bin icon to delete the corresponding Phone Profile options instance.

—

Add New Call Forward

Click to configure the options for forwarding incoming voice calls to SIP phones.

You can add multiple instances of these options so that you can configure different options for different pool tags.

—

Pool Tag

Enter one of the pool tags that you defined for the phone profile to associate with call forwarding actions.

—

Action

Choose the situation that causes a directory number to be forwarded to another directory number when the gateway is in SRST mode:

• busy: Forwards a call to another directory number when a phone is busy

• all: Forwards all incoming calls to another directory number

• noan: Forwards a call to another directory number when no answer is received after a configured timeout

call-forward b2bua all {number | busy number | noan number [timeout seconds ]}

Digit String

Enter the directory number to which forwarded calls are sent.

call-forward b2bua all {number | busy number | noan number [timeout seconds]}

Timeout

For a call forward noan action, enter the number of seconds that a call rings with no answer after which the call is forwarded to the directory number that theDigit String option defines.

Range: Integers 3 to 60000

Default: 20

call-forward b2bua noan {number [ timeout seconds]}

Click the Recycle Bin icon to delete the corresponding Call Forward options instance.

—

Association

Click to configure options for associating other configured UC voice features with the port or ports. When you associate a feature in this way, the configuration options in that feature are applied to the designated set of SIP phones.

You can add as multiple instances of these options so that you can configure different association options for different phone pools.

Pool Tag

Enter the unique sequence number of the set of SIP phones to be configured.

Media Profile

Choose a configured Media Profile feature to associate with the phone pool profile.

Translation Profile

Choose a configured Translation Profile feature to associate with the port.

Translation Profile Direction

Choose the direction of the traffic to which to apply the selected Translation Profile feature:

• Incoming: Applies the corresponding Translation Profile feature to traffic that is incoming to the port

• Outgoing: Applies the corresponding Translation Profile feature to traffic that is outgoing from the port

Click the Recycle Bin icon to delete the corresponding Association options instance.

Name

Enter a unique name for the server group configuration. The name can contain any characters.

Description

Enter a description of the server group configuration.

Server Group ID

Enter a unique identification number for this server group.

Range: Integers 1 through 10000

voice class server-group server-group-id

Description

Enter a description of this server group.

description string

Hunt Scheme

Choose the hunt method for the order of selection of target server IP addresses, which are IP addresses of the servers in the server group, for setting up outgoing calls. (Server addresses are configured as described in the following Address List table.)

Options are:

• none: No hunt scheme defined.

  If a hunt scheme is not defined, an available IP address of the highest Preference value is selected. (The preference is configured as described in the following Address List table.)

• round-robin: Searches IP addresses in turn for the next available server, starting with the server that follows the last used member of the server group.

hunt-scheme round-robin

Shutdown

Enable this option to put this server group in shutdown mode, which causes the outbound SIP dial peers that use this server group to be out of service.

—

Add Address List

Click to configure options for adding a server to the server group.

You can add up to 5 instances of these options that you can add up to 5 servers to the server group.

—

IPv4/6 Address

Enter the IPv4 or IPv6 address of the server.

ipv4|ipv6} address

Port

Enter the number of the server port that is listening for SIP calls.

port port

Preference

Applies only if the Hunt Scheme Basic Configuration option is set to none.

Choose the order of selection preference of the server for the setting up of outgoing calls.

Range: Integers 0 (highest preference) through 5 (lowest preference)

Default: 0

preference preference-order

Click the Recycle Bin icon to delete the corresponding Address List options instance.

—

Add Hunt Stop Rules

Click to configure options for configuring a hunt stop rule. This rule stops hunting for servers in the server group based on configured SIP response codes.

You can add up to 10,000 instances of these options so that you can configure different hunt stop rules for different response codes.

—

Rule ID

Enter the identifier of the hunt stop rule.

Range: Integers 1 through 1000

huntstop rule-tag resp-code from_resp_code to to_resp_code

Response Code Start

Enter the first SIP response code in a range of codes for the hunt stop rule.

Range: Integers 400 through 599

huntstop rule-tag resp-code from_resp_code to to_resp_code

Response Code End

Enter the last SIP response code in a range of codes for the hunt stop rule. For example, huntstop 1 resp-code 401.

Range: Integers 400 through 599

huntstop rule-tag resp-code from_resp_code to to_resp_code

Click the Recycle Bin icon to delete the corresponding Hunt Stop Rules options instance.

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Name

Enter a unique name for the supervisory disconnect configuration. The name can contain any characters.

Description

Enter a description of the supervisory disconnect configuration.

Add Custom CPTone

Click to configure options for custom call progress tones for a supervisory disconnect event.

You can add as multiple instances of these options so that you can configure different dual-tone options for a supervisory name.

—

Supervisory Name

Enter a name for the supervisory disconnect event.

The name can contain up to 32 characters. Valid characters are letters, numbers, dashes (-), and underscores (_).

voice class custom-cptone cptone-name

Dualtone

Choose the type of dual-tone that causes a supervisory disconnect event:

• Busy

• Disconnect

• Number Unobtainable

• Out of Service

• Reorder

• Ringback

dualtone {ringback | busy | reorder | out-of-service | number-unobtainable | disconnect}

Cadence

Enter the cadence interval, in ms, of the dual-tones that cause a supervisory disconnect event.

Enter the cadence as an on/off value pair, separated with a space.

You can enter up to 4 on/off value pairs, separated with spaces.

cadence cycle-1-on-time cycle-1-off-time [cycle-2-on-time cycle-2-off-time [cycle-3-on-time cycle-3-off-time [ cycle-4-on-time cycle-4-off-time ]]]

Dualtone Frequency

Enter the frequency, in Hz, for each tone in the dual tone.

Range for each tone: Integers 300 through 3600

frequency frequency-1 [frequency-2]

Click the Recycle Bin icon to delete the corresponding Custom CPTone options instance.

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Add Dual Tone Detection Params

Click to configure the following options for dual-tone detection parameters for a supervisory disconnect event.

You can add multiple instances of these options.

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Supervisory Number

Enter a unique number to identify dual-tone detection parameters.

Range: Integers 1 through 10000

voice class dualtone-detect-params tag-number

Cadence-Variation

Enter the maximum time, in ms, by which the tone onset can vary from the specified onset time and still be detected. The system multiplies the value that you enter by 10.

Range: Integers 0 through 200 (0 through 2000 ms)

Default: 10 (100 ms)

Frequency Max Delay

Enter the maximum delay, in milliseconds, before a supervisory disconnect occurs after the dual-tone is detected. The system multiplies the value that you enter by 10.

Range: Integers 0 through 100 (0 through 1000 ms)

Default: 10 (100 ms)

freq-max-delay time

Frequency Max Deviation

Enter the maximum deviation, in Hz, by which each tone can deviate from configured frequencies and be detected.

Range: Integers 0 through 125

Default: 10

freq-max-deviation hertz

Frequency Max Power

Enter the power of the dual-tone, in dBm0, above which a supervisory disconnect is not detected.

Range: Integers 0 through 20

Default: 10

freq-max-power dBm0

Frequency Min Power

Enter the power of the dual-tone, in dBm0, below which a supervisory disconnect is not detected.

Range: Integers 0 through 35

Default: 3

freq-min-power dBm0

Frequency Power Twist

Enter the difference, in dBm0, between the minimum power and the maximum power of the dual-tone above which a supervisory disconnect is not detected.

Range: Integers 0 through 15

Default: 6

freq-power-twist dBm0

Click the Recycle Bin icon to delete the corresponding Dual Tone Detection Params options instance.

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Name

Enter a unique name for the translation profile configuration. The name can contain any characters.

Description

Enter a description of the translation profile configuration.

Name

Enter a unique name for the translation profile.

If you do not enter a name, “Translation Profile” is used as the name.

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Click to configure options for mapping rules that are defined by the Translation Rule feature for calling and called numbers.

You can add up to 2 instances of these options, one instance for the calling call type and one for the called call type.

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Select Call Type

Choose the type of call to which to map a translation rule set:

• calling: Maps a translation rule set for the number that is calling in

• called: Maps a translation rule set for the number that is being called

• Calling:

  translate calling translation-rule-number

• Called:

  translate called translation-rule-number

Select Translation Rule

Choose a provisioned Translation Rule feature to associate with to the call type that you chose.

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Click to view the translation rule that you chose.

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Name

Enter a unique name for the translation rule configuration. The name can contain any characters.

Description

Enter a description of the translation rule configuration.

Translation rule set number

Enter a unique number to assign to a translation rule set that you are creating.

voice translation rule number

Import

Click to copy translation rules from a CSV file to Cisco Catalyst SD-WAN Manager.

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Export

Click to save existing translation rules that your created in a CSV file.

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Add Rule

Click to configure the options for the Translation Rule feature.

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Rule number

Displays a number that designates the precedence for this rule.

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Matching pattern

Enter the string that you want the translation rule to affect.

Enter the string in regular expression format beginning and ending with a slash (/). For example, /^9/.

To include the backslash character (\) in a match string, precede the backslash with a backslash.

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Action

Choose one of the following options to designate the action that the system performs for calls that match the string in the Matching pattern field:

• reject: Causes the system to reject the call.

• replace: Causes the system to replace the string in the Matching pattern field with a string that you specify.

voice translation-rule number

• Match and replace rule:

  rule precedence /match-pattern/ / replace-pattern/

• Reject rule:

  rule precedence reject /match-pattern/

Replacement pattern

If you choose the replace action for the rule, enter the string to which to translate the matched string.

Enter the number in regular expression format beginning and ending with a slash (/). For example, //, which indicates a replacement of no string.

To include the backslash character (\) in a replace string, precede the backslash with a backslash.

For example, if you specify a matching pattern of /^9/ and a replacement pattern string of //, the system removes the leading 9 from calls with a number that begins with 9. In this case, the system translates 914085551212 to 14085551212.

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Click the Recycle Bin icon to delete the corresponding Rule options instance.

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Name

Enter a unique name for the trunk group configuration. The name can contain any characters.

Description

Enter a description of the trunk group configuration.

Name

Enter the name of the trunk group.

The name can contain up to 32 characters.

trunk group name

Hunt Scheme

Choose the hunt scheme in the hunt group for outgoing calls.

• least-idle: Searches for an idle channel with the shortest idle time

• least-used: Searches for a trunk group member that has the highest number of available channels (applies only to PRI ISDN cards)

• longest-idle: Searches for an idle channel with the longest idle time

  round-robin: Searches trunk group members in turn for an idle channel, starting with the trunk group member that follows the last used

• sequential: Searches for an idle channel, starting with the trunk group member with the highest preference within the trunk group

• random: Searches for a trunk group member at random and selects a channel from the member at random

hunt-scheme least-idle [even | odd | both]

hunt-scheme least-used [even | odd | both [up | down]

hunt-scheme longest-idle [even | odd | both]

hunt-scheme random

hunt-scheme round-robin [even | odd | both [up | down]

hunt-scheme sequential [even | odd | both [up | down]

Max Calls In

Enter the maximum number of incoming calls that are allowed for the trunk group. If you do not enter a value, there is no limit on the number of incoming calls.

If the maximum number of incoming calls is reached, the trunk group becomes unavailable for more calls.

Range: Integers 0 through 1000

trunk group name

max-calls voice number-of-calls direction in

Max Calls Out

Enter the maximum number of outgoing calls that are allowed for the trunk group. If you do not enter a value, there is no limit on the number of outgoing calls.

If the maximum number of outgoing calls is reached, the trunk group becomes unavailable for more calls.

Range: Integers 0 through 1000

trunk group name

max-calls voice number-of-calls direction out

This option does not appear when the Hunt Scheme option is set to random.

Choose the type of channel that the hunt scheme searches for:

• Both: Searches both even- and odd-numbered channels.

• Even: Searches for an idle even-numbered channel. If no idle even-numbered channels are available, an odd-numbered channel is sought.

• Odd: Searches for an idle odd-numbered channel. If no idle odd-numbered channels are available, an even-numbered channel is sought.

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This option appears when the Hunt Scheme option is set to round-robin or sequential.

Choose the order in which the hunt scheme searches for channels:

• up: Searches channels in ascending order within a trunk group member.

• down: Searches channels in descending order within a trunk group member.

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Max Retry

Enter the maximum number of outgoing call attempts that the trunk group makes if an outgoing call fails.

If you do not enter a value and a call fails, the system does not attempt to make the call again.

Range: Integers 1 through 5

trunk group name

max-retry attempts

Name

Enter a unique name for the voice global configuration. The name can contain any characters.

Description

Enter a description of the voice global configuration.

Trusted IPv4/6 Prefix List

Enter a comma separated list of IPv4 or IPv6 addresses with which the router can communicate through SIP.

Enter each IPv4 address in CIDR format. For example, 10.1.2.3/32.

The router does not communicate with other addresses, which prevents fraudulent calls being placed through the router.

A Trusted IPv4 or IPv66 prefix is required for TDM to IP calls.

voice service voip

ip address trusted list

ipv4 ipv4-address/ipv4-network-mask

Source Interface

Enter the name of the source interface from which the router initiates SIP control and media traffic.

This information defines how the return/response to this traffic should be sent.

voice service voip

sip

bind control source-interface interface-id

bind media source-interface interface-id

Participation

Enable this option to configure all T1 or E1 digital interfaces to participate in the backplane clock.

Disable this option to remove the clock synchronization with the backplane clock for the module.

Default: Enabled

network-clock synchronization participate slot sub-slot

Appears only if you have configured a digital interface and selected either a primary or secondary clock source for the interface.

Designate the priority of up to 6 clock sources for the digital interface.

The drop-down list displays the interface ports for which a primary or secondary clock source is defined and that is configured for network participation.

Choose the port that you want to have the highest priority. Then choose the port that you want to have the next highest priority. Continue in this way to choose other ports.

The ports in the field appear in descending order of priority, with the highest priority port appearing first.

To remove a port from the field, click its X icon. To change the priority order of ports, remove the ports and add them back in the desired order.

We recommend that all ports in the priority list be of the same type, either E1-PRI or T1-PRI.

network-clock-input-source priority controller [t1|e1] slot/sub-slot/port

Automatically Sync

Choose true to enable network synchronization between all modules and the router. Choose false to disable network synchronization between all modules and the router.

Default: False

network-clock synchronization automatic

Wait to restore clock

Enter the amount of time, in ms, that the router waits before including a primary clock source in the clock selection process.

Range: Integers 0 through 86400

Default: 300

network-clock wait-to-restore milliseconds

Name

Enter a unique name for the voice tenant configuration. The name can contain any characters.

Description

Enter a description of the voice tenant configuration.

Tag

Enter a unique name for this voice tenant configuration.

voice class tenant tag

Bind Interface

Choose the type of packets that are bound to network interfaces for advertising the source IP address of the tenant:

• Both: Control and media packets

• Control: Control packets

• Media: Media packets

• Disabled: Bind interface is not configured

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Transport Type

Choose the transport protocol for SIP control signaling for the tenant.

Options are TCP, UDP, and TCP TLS.

session transport {udp | tcp [tls]}

Enter a network interface name for binding control packets.

bind control source-interface interface-id

Enter a network interface name for binding media packets.

bind media source-interface interface-id