Cisco IOS Voice Command Reference - A through C - clid through credentials (sip-ua) [Support]

clid

To preauthenticate calls on the basis of the Calling Line IDentification (CLID) number, use the clid command in AAA preauthentication configuration mode. To remove the clid command from your configuration, use the no form of this command.

clid [ if-avail | required ] [accept-stop] [ password password ]

no clid [ if-avail | required ] [accept-stop] [ password password ]

Syntax Description

if-avail	(Optional) Implies that if the switch provides the data, RADIUS must be reachable and must accept the string in order for preauthentication to pass. If the switch does not provide the data, preauthentication passes.
required	(Optional) Implies that the switch must provide the associated data, that RADIUS must be reachable, and that RADIUS must accept the string in order for preauthentication to pass. If these three conditions are not met, preauthentication fails.
accept-stop	(Optional) Prevents subsequent preauthentication elements such as ctype or dnis from being tried once preauthentication has succeeded for a call element.
password password	(Optional) Defines the password for the preauthentication element. The default password string is cisco.

Command Default

The if-avail and required keywords are mutually exclusive. If the if-avail keyword is not configured, the preauthentication setting defaults to required.

Command Modes

AAA preauthentication configuration

Command History

Release	Modification
12.1(2)T	This command was introduced.

Usage Guidelines

You may configure more than one of the authentication, authorization and accounting (AAA) preauthentication commands (clid, ctype, dnis) to set conditions for preauthentication. The sequence of the command configuration decides the sequence of the preauthentication conditions. For example, if you configure dnis, then clid, then ctype, in this order, then this is the order of the conditions considered in the preauthentication process.

In addition to using the preauthentication commands to configure preauthentication on the Cisco router, you must set up the preauthentication profiles on the RADIUS server.

Examples

The following example specifies that incoming calls be preauthenticated on the basis of the CLID number:

aaa preauth
 group radius
 clid required

Related Commands

Command	Description
ctype	Preauthenticates calls on the basis of the call type.
dnis (RADIUS)	Preauthenticates calls on the basis of the DNIS number.
dnis bypass (AAA preauthentication configuration)	Specifies a group of DNIS numbers that will be bypassed for preauthentication.
group (RADIUS)	Specifies the AAA RADIUS server group to use for preauthentication.

clid (dial peer)

To control the presentation and use of calling-line ID (CLID) information, use the clid command in dial peer configuration mode. To remove CLID controls, use the no form of this command.

clid { network-number number [ second-number strip ] | network-provided | override rdnis | restrict | strip [ name | pi-restrict [all] ] | substitute name }

no clid { network-number number [ second-number strip ] | network-provided | override rdnis | restrict | strip [ name | pi-restrict [all] ] | substitute name }

Syntax Description

network-number number	Network number. Establishes the calling-party network number in the CLID for this router.
network-provided	Allows you to set the screening indicator to reflect the number that was provided by the network.
override rdnis	Supported for POTS dial peers only Overrides the CLID with the redirected dialed number identification service (RDNIS) if available.
pi-restrict	Restricted progress indicator (PI). Causes removal of the calling-party number from the CLID when the PI is restricted.
restrict	Restricts presentation of the caller ID in the CLID.
second-number strip	(Optional) Removes a previously configured second network number from the CLID.
strip	Strips the calling-party number from the CLID. name --(Optional) Calling-party name. Causes removal of the calling-party name from the CLID. pi-restrict [all]--(Optional) Restricted PI. Causes removal of all calling-party names and numbers from the CLID when the PI is restricted.
substitute name	Copies the calling number into the display name if PI allows it (and the calling name is empty).

Command Default

No default behavior or values

Command Modes

Dial Peer configuration (config-dial-peer)

Command History

Release	Modification
12.2(11)T	This command was introduced.
12.2(13)T	The overriderdnis keywords were added.
12.4(4)T	The following keywords were added: network-provided, pi-restrictall, and substitutename.

Usage Guidelines

The overriderdnis keywords are supported only for POTS dial peers.

CLID is the collection of information about the billing telephone number from which a call originated. The CLID value might be the entire phone number, the area code, or the area code plus the local exchange. It is also known as caller ID. The various keywords to this command manage the presentation, restriction, or stripping of the various CLID elements.

The clidnetwork-number command sets the presentation indicator to "y" and the screening indicator to "network-provided." The second-numberstrip keyword strips from the H.225 source-address field the original calling-party number, and is valid only if a network number was previously configured.

The clidoverriderdnis command overrides the CLID with the RDNIS if it is available.

The clidrestrict command causes the calling-party number to be present in the information element, but the presentation indicator is set to "n" to prevent its presentation to the called party.

The clidstrip command causes the calling-party number to be null in the information element, and the presentation indicator is set to "n" to prevent its presentation to the called party.

Examples

The following example sets the calling-party network number to 98765 for POTS dial peer 4321:

Router(config)# dial-peer voice 4321 pots
Router(config-dial-peer)# clid network-number 98765

An alternative method of accomplishing this result is to enter the second-numberstrip keywords as part of the clidnetwork-number command. The following example sets the calling-party network number to 56789 for VoIP dial peer 1234 and also prevents the second network number from being sent:

Router(config)# dial-peer voice 1234 voip
Router(config-dial-peer)# clid network-number 56789 second-number strip

The following example overrides the calling-party number with RDNIS if available:

Router(config-dial-peer)# clid override rdnis

The following example prevents the calling-party number from being presented:

Router(config-dial-peer)# clid restrict

The following example removes the calling-party number from the CLID information and prevents the calling-party number from being presented:

Router(config-dial-peer)# clid strip

The following example strips the name from the CLID information and prevents the name from being presented:

Router(config-dial-peer)# clid strip name

The following example strips the calling party number when PI is set to restrict clid strip from the CLID information and prevents the calling party number from being presented:

Router(config-dial-peer)# clid strip pi-restrict

The following example strips calling party name and number when the PI is set to the restrict all clid strip from the CLID information and prevents the calling party name and number from being presented:

Router(config-dial-peer)# clid strip pi-restrict all

The following example substitutes the calling party number into the display name:

Router(config-dial-peer)# clid substitute name

The following example allows you to set the screening indicator to reflect that the number was provided by the network:

Router(config-dial-peer)# clid network-provided

Related Commands

Command	Description
clid (voice-service-voip)	Passes the network provided ISDN numbers in an ISDN calling party information element screening indicator field, removes the calling party name and number from the calling-line identifier in voice service voip configuration mode, or allows a presentation of the calling number by substituting for the missing Display Name field in the Remote-Party-ID and From headers.

clid (voice service voip)

To pass the network-provided ISDN numbers in an ISDN calling party information element screening indicator field, and remove the calling party name and number from the calling-line identifier in voice service voip configuration mode, or allow a presentation of the calling number by substituting for the missing Display Name field in the Remote-Party-ID and From headers use the clid command in voice service voip configuration mode. To return to the default configuration, use the no form of this command.

clid { network-provided | strip pi-restrict all | substitute name }

no clid { network-provided | strip pi-restrict all | substitute name }

Syntax Description

network -provided	Sets the screen indicator as network-provided.
strip pi -restrictall	Removes the CLID when the progress indicator (PI) is restricted for PSTN to SIP operations and removes the calling party name and number when the PI is restricted for PSTN to SIP operations.
substitute name	Copies the calling number to the display name if unavailable for PSTN to SIP operations.

Command Default

The clid command passes along user-provided ISDN numbers in an ISDN calling party information element screening indicator field.

Command Modes

Voice service VoIP configuration (config-voi-srv)

Command History

Release	Modification
12.4(4)T	This command was introduced.

Usage Guidelines

Use the clidnetwork-provided keyword to pass along network-provided ISDN numbers in an ISDN calling party information element screening indicator field.

Use the clidstrippi-restrictall keyword to remove the Calling Party Name and Calling Party Number from the CLID.

Use the clidsubstitutename keyword to allow a presentation of the Display Name field in the Remote-Party-ID and From headers. The Calling Number is substituted for the Display Name field.

Examples

The following example passes along network-provided ISDN numbers in an ISDN calling party information element screening indicator field:

Router(conf-voi-serv)# clid network-provided

The following example passes along user-provided ISDN numbers in an ISDN calling party information element screening indicator field:

Router(conf-voi-serv)# no clid network-provided

The following example removes the calling party name and number from the calling-line identifier (CLID):

Router(conf-voi-serv)# clid strip pi-restrict all

The following example does not remove the calling party name and number from the CLID:

Router(conf-voi-serv)# no clid strip pi-restrict all

The following example allows the presentation of the calling number to be substituted for the missing Display Name field in the Remote-Party-ID and From headers:

Router(conf-voi-serv)# clid substitute name

The following example disallows the presentation of the calling number to be substituted for the missing Display Name field in the Remote-Party-ID and From headers:

Router(conf-voi-serv)# no clid substitute name

Related Commands

Command	Description
clid (dial-peer)	Controls the presentation and use of CLID information in dial peer configuration mode.

clid strip

To remove the calling-party number from calling-line-ID (CLID) information and to prevent the calling-party number from being presented to the called party, use the clidstrip command in dial-peer configuration mode. To remove the restriction, use the no form of this command.

clid strip [name]

no clid strip [name]

Syntax Description

name	(Optional) Removes the calling-party name for both incoming and outgoing calls.

Command Default

Calling-party number and name are included in the CLID information.

Command Modes

Dial-peer configuration (config-dial-peer)

Command History

Cisco IOS Release	Cisco CME Version	Modification
12.2(11)T	2.01	This command was introduced.
12.2(15)ZJ1	3.0	This command was modified. The name keyword was added.
12.3(4)T	3.0	This command was integrated into Cisco IOS Release 12.3(4)T.

Usage Guidelines

If the clidstrip command is issued, the calling-party number is null in the information element, and the presentation indicator is set to "n" to prevent the presentation of the number to the called party.

If you want to remove both the number and the name, you must issue the command twice, once with the name keyword.

Examples

The following example removes the calling-party number from the CLID information and prevents the calling-party number from being presented:

Router(config-dial-peer)# clid strip

The following example removes both the calling-party number and the calling-party name from the caller-ID display:

Router(config-dial-peer)# clid strip
Router(config-dial-peer)# clid strip name

Related Commands

Command	Description
clid network-number	Configures a network number in the router for CLID and uses it as the calling-party number.
clid restrict	Prevents the calling-party number from being presented by CLID.
clid second-number strip	Prevents the second network number from being sent in the CLID information.

clid strip reason

To remove the calling-line ID (CLID) reason code and to prevent it from being displayed on the phone, use the clidstripreason command in dial peer voice configuration mode. To disable the configuration, use the no form of this command.

clid strip reason

no clid strip reason

Syntax Description

This command has no arguments or keywords.

Command Default

The CLID reason code is not removed.

Command Modes

Dial peer voice configuration (config-dial-peer)

Command History

Release	Modification
12.4(15)T	This command was introduced.

Usage Guidelines

When the caller-idenablecommand is enabled on the gateway so that the gateway forwards information depending on the preference of the caller, client layer interface port (CLIP), or calling line identification restriction (CLIR), an "unavailable" message is displayed on the terminating phone. An "unavailable" message is a standard message that indicates the reason for the absence of calling party name.

You can use the clidstripreason command to remove the message and have only the call parameters forwarded.

Examples

The following example shows how to remove the CLID reason code:

Router# configure terminal
Router(config)# dial-peer voice 88 voip
Router(config-dial-peer)# clid strip reason

Related Commands

Command	Description
caller-id enable	Allows the sending or receiving of caller-ID information.
clid strip	Removes the calling-party number from CLID information and prevents the calling-party number from being presented to the called party.
dial-peer voice	Defines a particular dial peer, specifies the method of voice encapsulation, and enters dial peer configuration mode.

clock-rate (codec-profile)

To set the clock rate, in Hz, for the codec, use the clock-rate command in codec-profile configuration mode. To return to the default value, use the no form of this command.

clock-rate clock-rate

no clock-rate

Syntax Description

clock-rate

Number in the range of 1 to 1000000.

Command Default

The default clock rate is 0.

Command Modes

Codec-profile configuration (config-codec-profile)

Command History

Release	Modification
12.4(22)T	This command was introduced.

Usage Guidelines

The clock-rate must be set to 90000 for H.263/H.264.

Examples

The following example shows:

codec profile 116 h263
	clock-rate 500000
	fmtp "fmtp "fmtp:120 SQCIF=1;QCIF=1;CIF=1;CIF4=2;MAXBR=3840;I=1""
!

Related Commands

Command	Description
codec profile	Defines video capabilities needed for video endpoints.

clock-select

To establish the sources and priorities of the requisite clocking signals for the OC-3/STM-1 ATM Circuit Emulation Service network module, use the clock-select command in CES configuration mode.

clock-select priority-numberinterface slot/port

Syntax Description

priority-number	Priority of the clock source. Range is from 1 (high priority) to 4 (low priority). There is no default value.
interface	Specifies the interface to supply the clock source.
slot /port	Backplane slot number and port number on the interface.

Command Default

No default behavior or values

Command Modes

CES configuration (config-ces)

Command History

Release	Modification
12.1(2)T	This command was introduced on the Cisco 3600 series.

Usage Guidelines

This command is used on Cisco 3600 series routers that have OC-3/STM-1 ATM CES network modules.

To support synchronous or synchronous residual time stamp (SRTS) clocking modes, you must specify a primary reference source to synchronize the flow of constant bit rate (CBR) data from its source to its destination.

You can specify up to four clock priorities. The highest priority active interface in the router supplies primary reference source to all other interfaces that require network clock synchronization services. The fifth priority is the local oscillator on the network module.

Use the showcesclock-selectcommand to display the currently configured clock priorities on the router.

Examples

The following example defines two clock priorities on the router:

clock-select 1 cbr 2/0
clock-select 2 atm 2/0

Related Commands

Command	Description
channel-group	Configures the timing recovery clock for the CES interface.
clock source	Configures a transmit clock source for the CES interface.
show ces clock	Displays which ports are designated as network clock sources.

cm-current-enhance

To improve immunity to extreme levels of longitudinal noise present in wiring that includes long cable lengths, use the cm-current-enhance command in Voice-port configuration mode. To return to the default configuration, use the no form of this command.

cm- current- enhance

no cm- current- enhance

Syntax Description

This command has no arguments or keywords.

Command Default

The cm-current-enhance command is not configured.

Command Modes

Voice-port configuration (config-voiceport)

Command History

Release	Modification
15.2(1)T	This command was introduced.

Usage Guidelines

This command should not be used under normal conditions. It should be used only to improve immunity to noise in cases of extreme levels of longitudinal noise on the wiring.

The command is available on the following platforms, in the modes indicated:

VIC3-2FXS-E/DID (FXS and DID mode)
VIC3-2FXS/DID, VIC3-4FXS/DID, and EM3-HDA-8FXS/DID (DID mode only)

Mode of action: When the cm-current-enhance mode is activated, REG 73 of the Silab chip (Si324x) is programmed to 1 to enhance the immunity to common-mode current noise.

Change of signaling type: The command is effective for the current signaling type value. The command state is not saved and applied after a change of signaling type.

Examples

The following example indicates the usage:

Device# config t
Enter configuration commands, one per line.  End with CNTL/Z.
Device(config)# voice-port 0/1/0
Device(config-voiceport)# cm-current-enhance

codec (dial peer)

To specify the voice coder rate of speech for a dial peer, use the codec command in dial peer configuration mode. To reset command settings to the default value, use the no form of this command.

Cisco 1750 and Cisco 1751 Modular Access Routers, Cisco AS5300 and AS5800 Universal Access Servers, and Cisco MC3810 Multiservice Concentrators

codec codec [ bytes payload-size ] [fixed-bytes] [ mode { independent | adaptive } ] [ bit-rate value ] [ framesize { 30 | 60 } [fixed] ]

no codec codec [ bytes payload-size ] [fixed-bytes] [ mode { independent | adaptive } ] [ bit-rate value ] [ framesize { 30 | 60 } [fixed] ]

Cisco 2600, 3600, 7200, and 7500 Series Routers

codec { codec [ bytes payload-size ] | transparent } [fixed-bytes] [ mode { independent | adaptive } ] [ bit-rate value ] [ framesize { 30 | 60 } [fixed] ]

no codec { codec [ bytes payload-size ] | transparent } [fixed-bytes] [ mode { independent | adaptive } ] [ bit-rate value ] [ framesize { 30 | 60 } [fixed] ]

Syntax Description

codec

Specifies the voice coder rate for speech. Codec options available for various platforms are described in the first table below.

bytes

(Optional) Precedes the argument that specifies the number of bytes in the voice payload of each frame.

payload-size

(Optional) Number of bytes in the voice payload of each frame. See the second table below for valid entries and default values.

transparent

Enables codec capabilities to be passed transparently between endpoints in a Cisco Unified Border Element.

Note

The transparent keyword is available only on the Cisco 2600, 3600, 7200, and 7500 series router platforms.

fixed-bytes

(Optional) Indicates that the codec byte size is fixed and nonnegotiable.

mode

(Optional) For Cisco internet Speech Audio Codec (iSAC) codec only. Specifies the iSAC operating frame mode that is encapsulated in each packet.

independent

(Optional) For iSAC codec only. Specifies that the configuration mode variable bit rate is independent (value 1).

adaptive

(Optional) For iSAC codec only. Specifies that the configuration mode variable bit rate is adaptive (value 0).

bit rate value

(Optional) For iSAC codec only. Configures the target bit rate in kilobits per second. Range is 10 to 32.

frame-size

(Optional) For iSAC codec only. Specifies the operating frame in milliseconds (ms). Valid entries are:

30 --30-ms frames
60 --60-ms frames
fixed --This keyword is applicable only for adaptive mode.

Command Default

g729r8, 30-byte payload for Voice over Frame Relay (VoFR) and Voice over ATM (VoATM). g729r8, 20-byte payload for Voice over IP (VoIP). See the second table below for valid entries and default values for codecs.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Release	Modification
11.3(1)T	This command was introduced on the Cisco 3600 series.
11.3(3)T	This command was implemented on the Cisco 2600 series.
12.0(3)T	This command was implemented on the Cisco AS5300. This release does not support the clear-channel keyword.
12.0(4)T	This command was implemented on the Cisco 3600 series, Cisco 7200 series, and Cisco MC3810, and the command was modified for VoFR dial peers.
12.0(5)XE	Additional codec choices and other options were implemented.
12.0(5)XK	The g729br8 and pre-ietf codec keywords were added for the Cisco 2600 and Cisco 3600 series.
12.0(7)T	This command was integrated into Cisco IOS Release 12.0.(7)T and implemented on the Cisco AS5800. Voice coder rates of speech were added. This release does not support the clear-channel keyword, so it is no longer available in the command syntax.
12.0(7)XK	Theg729abr8 and g729ar8 codec keywords were added for the Cisco MC3810, and the pre-ietf keyword was deleted.
12.1(1)T	This command was integrated in Cisco IOS Release 12.1(1)T.
12.1(5)T	Thegsmefr and gsmfr codec keywords were added.
12.2(8)T	The command was implemented on the Cisco 1750 and Cisco 1751.
12.2(13)T3	The transparent keyword was added for use with H.323 to H.323 connections. This keyword is available only in js2 images.
12.4(11)XJ2	Thegsmefrand gsmfrkeywords were removed as configurable codec options for all platforms with the exception of the gsmfr codec on the Cisco AS5400 and AS5350 with MSAv6 DSPs. The transparent keyword now supports H.323 to SIP connections.
12.4(15)T	This command was integrated into Cisco IOS Release 12.4(15)T.
12.4(15)XY	The g722-64 keyword was added.
12.4(20)T	This command was integrated into Cisco IOS Release 12.4(20)T.
15.0(1)M	The fixed-bytes keyword was added.
15.1(1)T	This command was modified. Theisac keyword was added as a codec type, and the mode, independent, adaptive, bitrate, and fixed keywords were added as configurable parameters.

Usage Guidelines

Use thiscommand to define a specific voice coder rate of speech and payload size for a VoIP or VoFR dial peer. This command is also used for VoATM.

Table 1 Codec Support by Platform
Codec	Cisco 1750 and Cisco 1751 Modular Access Routers	Cisco 2600 and 3600 Series Routers and Cisco 7200 and 7500 Series Routers	Cisco AS5300 and AS5800 Universal Access Servers	Cisco MC3810 Multiservice Concentrators
clear-channel --Clear channel at 64,000 bits per second (bps)	Yes	Yes	--	Yes
g711alaw --G.711 A-Law at 64,000 bps	Yes	Yes	Yes	Yes
g711ulaw --G.711 mu-Law at 64,000 bps	Yes	Yes	Yes	Yes
g722-64 --G.722-64 at 64,000 bps	Yes	Yes	Yes	--
g723ar53 --G.723.1 Annex A at 5300 bps	--	Yes	Yes	Yes
g723ar63 --G.723.1 Annex A at 6300 bps	--	Yes	Yes	Yes
g723r53 --G.723.1 at 5300 bps	--	Yes	Yes	Yes
g723r63 --G.723.1 at 6300 bps	--	Yes	Yes	Yes
g726r16 --G.726 at 16,000 bps	Yes	Yes	Yes	Yes
g726r24 --G.726 at 24,000 bps	Yes	Yes	Yes	Yes
g726r32 --G.726 at 32,000 bps	Yes	Yes	Yes	Yes
g726r53 --G.726 at 53,000 bps	Yes	Yes	Yes	--
g726r63 --G.726 at 63,000 bps	Yes	Yes	Yes	--
g728 --G.728 at 16,000 bps	--	Yes	Yes	Yes
g729abr8 --G.729 Annex A and B at 8000 bps	Yes	Yes	Yes	Yes
g729ar8 --G.729 Annex A at 8000 bps	Yes	Yes	Yes	Yes
g729br8 --G.729 Annex B at 8000 bps	Yes	Yes	Yes	Yes
g729r8 --G.729 at 8000 bps. This is the default codec.	Yes	Yes	Yes	Yes
isac --Cisco internet Speech Audio Codec (iSAC) codec.	Yes	Yes	Yes	Yes

A specific codec type can be configured on the dial peer as long as the codec is supported by the setting used with the codeccomplexity voice-card configuration command. The codeccomplexity command is voice-card specific and platform specific. The codeccomplexity voice-card configuration command is set to either high or medium.

If the codeccomplexity command is set to high, the following keywords are available: g711alaw, g711ulaw,g722-64, g723ar53, g723ar63, g723r53, g723r63, g726r16, g726r24, g726r32, g728, g729r8, and g729br8.

If the codeccomplexity command is set to medium, the following keywords are available: g711alaw, g711ulaw, g726r16, g726r24, g726r32, g729r8, and g729br8.

The codec dial peer configuration command is particularly useful when you must change to a small-bandwidth codec. Large-bandwidth codecs, such as G.711, do not fit in a small-bandwidth link. However, the g711alaw and g711ulaw codecs provide higher quality voice transmission than other codecs. The g729r8 codec provides near-toll quality with considerable bandwidth savings.

The transparent keyword is available with H.323 to H.323 call connections beginning in Cisco IOS Release 12.2(13)T3. Support for the keyword in H.32 to SIP call connections begins in Cisco IOS Release 12.4(11)XJ2.

If codec values for the dial peers of a connection do not match, the call fails.

You can change the payload of each VoIP frame by using the byteskeyword; you can change the payload of each VoFR frame by using the bytes keyword with the payload-size argument. However, increasing the payload size can add processing delay for each voice packet.

The table below describes the voice payload options and default values for the codecs and packet voice protocols.

Table 2 Voice Payload-per-Frame Options and Defaults
Codec	Protocol	Voice Payload Options (in Bytes)	Default Voice Payload (in Bytes)
g711alaw g711ulaw	VoIP VoFR VoATM	80, 160 40 to 240 in multiples of 40 40 to 240 in multiples of 40	160 240 240
g722-64	VoIP	80, 160, 240	160
g723ar53 g723r53	VoIP VoFR VoATM	20 to 220 in multiples of 20 20 to 240 in multiples of 20 20 to 240 in multiples of 20	20 20 20
g723ar63 g723r63	VoIP VoFR VoATM	24 to 216 in multiples of 24 24 to 240 in multiples of 24 24 to 240 in multiples of 24	24 24 24
g726r16	VoIP VoFR VoATM	20 to 220 in multiples of 20 10 to 240 in multiples of 10 10 to 240 in multiples of 10	40 60 60
g726r24	VoIP VoFR VoATM	30 to 210 in multiples of 30 15 to 240 in multiples of 15 30 to 240 in multiples of 15	60 90 90
g726r32	VoIP VoFR VoATM	40 to 200 in multiples of 40 20 to 240 in multiples of 20 40 to 240 in multiples of 20	80 120 120
g728	VoIP VoFR VoATM	10 to 230 in multiples of 10 10 to 240 in multiples of 10 10 to 240 in multiples of 10	40 60 60
g729abr8 g729ar8 g729br8 g729r8	VoIP VoFR VoATM	10 to 230 in multiples of 10 10 to 240 in multiples of 10 10 to 240 in multiples of 10	20 30 30
isac	VoIP	10 to 230 in multiples of 10	30 60

Note

If you are configuring G.729r8 or G.723 as the codec-type, the maximum value for the payload-size argument is 60 bytes.

For toll quality, use the g711alaw or g711ulawkeyword. These values provide high-quality voice transmission but use a significant amount of bandwidth. For nearly toll quality (and a significant savings in bandwidth), use the g729r8keyword.

Note

The G.723 and G.728 codecs are not supported on the Cisco 1700 platform for Cisco Hoot and Holler applications.

Note

The clear-channel keyword is not supported on the Cisco AS5300.

Note

The G.722-64 codec is supported only for H.323 and SIP.

Examples

The following example shows how to configure a voice coder rate that provides toll quality voice with a payload of 120 bytes per voice frame on a router that acts as a terminating node. The sample configuration begins in global configuration mode and is for VoFR dial peer 200.

dial-peer voice 200 vofr
 codec g711ulaw bytes 240

The following example shows how to configure a voice coder rate for VoIP dial peer 10 that provides toll quality but uses a relatively high amount of bandwidth:

dial-peer voice 10 voip
 codec g711alaw

The following example shows how to configure the transparent codec used by the Cisco Unified Border Element:

dial-peer voice 1 voip
 incoming called-number .T
 destination-pattern .T
 session target ras
 codec transparent

Related Commands

Command	Description
codec (dsp farm profile)	Specifies call density and codec complexity.
codec (voice port)	Specifies voice compression.
codec complexity	Specifies call density and codec complexity based on the codec used.
show dial peer voice	Displays the codec setting for dial peers.

codec (dsp)

To specify call density and codec complexity based on a particular codec standard, use the codec command in DSP interface DSP farm configuration mode. To reset the card type to the default, use the no form of the command.

codec { high | med }

no codec { high | med }

Syntax Description

high	Specifies high complexity: two channels of any mix of codec.
med	Specifies medium complexity: four channels of g711/g726/g729a/fax.

Command Default

Medium complexity

Command Modes

DSP interface DSP farm

Command History

Release	Modification
12.0(5)XE	This command was introduced on the Cisco 7200 series.
12.1(1)T	This command was integrated into Cisco Release 12.1(1)T.
12.1(3)T	This command was implemented on the Cisco 7500 series.

Usage Guidelines

This command is supported on only the Cisco 7200 series and Cisco 7500 series routers.

Codec complexity refers to the amount of processing required to perform compression. Codec complexity affects the number of calls, referred to as call density, that can take place on the DSPfarm interfaces. The greater the codec complexity, the fewer the calls that are handled. For example, G.711 requires less DSP processing than G.728, so as long as the bandwidth is available, more calls can be handled simultaneously by using the G.711 standard than by using G.728.

The DSPinterface dspfarm codec complexity setting affects the options available for the codecdialpeerconfiguration command.

To change codec complexity, you must first remove any configured-channel associated signaling (CAS) or DS0 groups and then reinstate them after the change.

Note

On the Cisco 2600 series routers, and 3600 series codec-complexity is configured using the codeccomplexity command in voice-card configuration mode.

Examples

The following example configures the DSPfarm interface 1/0 on the Cisco 7200 series routers to support high compression:

dspint DSPFarm 1/0 
 codec high

Related Commands

Command	Description
codec (dial peer)	Specifies the voice codec rate of speech for a dial peer.
codec complexity	Specifies call density and codec complexity based on the codec standard you are using.

codec (DSP farm profile)

To specify the codecs that are supported by a digital signal processor (DSP) farm profile, use the codec command in DSP farm profile configuration mode. To remove the codec, use the no form of this command.

codec { codec-type [resolution] | [ frame-rate framerate ] | [ bitrate bitrate ] | [rfc-2190] | pass-through }

no codec { codec-type [resolution] | [ frame-rate framerate ] | [ bitrate bitrate ] | [rfc-2190] | pass-through }

Syntax Description

codec-type

Specifies the codec preferred.

g711alaw --G.711 a-law 64,000 bits per second (bps)
g711ulaw --G.711 mu-law 64,000 bps
g722r-64 --G.722-64 at 64,000 bps
g729abr8 --G.729 ANNEX A and B 8000 bps
g729ar8 --G.729 ANNEX A 8000 bps
g729br8 --G.729 ANNEX B 8000 bps
g729r8 --G.729 8000 bps
h263 --H.263 video codec
h264 --H.264 video codec
ilbc --Internet Low Bitrate Codec (iLBC)
isac --Cisco internet Speech Audio Codec (iSAC) codec

resolution

Specifies the supported video resolution. The valid entries are:

For H.263--qcifandcif
For H.264--qcif, cif, vga, w360p, w448p, 4cif, and 720p

Note

720p option applies only to homogeneous video conferences.

frame-rate framerate

Specifies the frame rate. The valid entries are 15 fps or 30 fps.

This option applies to homogeneous conferences only.

bitrate bitrate

Specifies the bitrate.

This option applies to homogeneous conferences only.

rfc-2190

Specifies the payload format follow RFC-2190.

pass-through

Enables codec pass-through. Supported for transcoding and media termination point (MTP) profiles.

Transcoding

The following transcoding default apply when you are configuring audio profiles only. When you configure video transcoding, you must specify the audio codecs.

g711alaw
g711ulaw
g729abr8
g729ar8

g711alaw
g711ulaw
g729abr8
g729ar8
g729br8
g729r8

g711ulaw

Command Modes

DSP farm profile configuration (config-dspfarm-profile)

Command History

Release	Modification
12.3(8)T	This command was introduced.
12.4(4)T	The pass-through keyword was added.
12.4(11)XJ2	The gsmefrand gsmfrkeywords were removed as configurable codec options for all platforms.
12.4(15)T	This command was integrated into Cisco IOS Release 12.4(15)T.
12.4(15)XY	The g722r-64 keyword was added.
12.4(20)T	This command was integrated into Cisco IOS Release 12.4(20)T.
12.4(22)T	Support for IPv6 was added.
15.1(1))T	This command was modified. The isac keyword was added.
15.1(4)M	This command was modified. The frame-rate, bitrate, rfc-2190, and pass-throughkeywords were added and codec support was added for ilbc, h.263and h.264.

Usage Guidelines

Only one codec is supported for each MTP profile. To support multiple codecs, you must define a separate MTP profile for each codec.

For homogeneous video profiles, only one video format is supported

For heterogeneous and heterogeneous guaranteed-audio video profiles, multiple video formats and audio codecs are supported.

To change the configured codec in the profile, you must first enter a nomaximumsessioncommand.

The table below shows the relationship between DSP farm functions and codecs.

Table 3 DSP Farm Functions and Codec Relationships
DSP Farm Function	Supported Codec
Transcoding	g711alaw g711ulaw g729abr8 g729ar8 iSAC h263 h264
Conferencing	g711alaw g711ulaw g722r-64 g729abr8 g729ar8 g729br8 g729r8 h263 h264 ilbc
MTP	g711ulaw iSAC

Hardware MTPs support only G.711 a-law and G.711 mu-law. If you configure a profile as a hardware MTP and you want to change the codec to other than G.711, you must first remove the hardware MTP by using thenomaximumsessionshardware command.

The pass-through keyword is supported for transcoding and MTP profiles only; the keyword is not supported for conferencing profiles. To support the Resource Reservation Protocol (RSVP) agent on a Skinny Client Control Protocol (SCCP) device, you must use the codecpass-through command. In the pass-through mode, the SCCP device processes the media stream by using a pure software MTP, regardless of the nature of the stream, which enables video and data streams to be processed in addition to audio streams. When the pass-through mode is set in a transcoding profile, no transcoding is done for the session; the transcoding device performs a pure software MTP function. The pass-through mode can be used for secure Real-Time Transport Protocol (RTP) sessions.

Examples

The following example shows how to set the call density and codec complexity to g729abr8:

Router(config)# dspfarm profile 123 transcode
Router(config-dspfarm-profile)# codec g729abr8
The following example shows how to set up a video conference with guaranteed-audio.
Router(config)# dspfarm profile 99 conference video guaranteed-audio
Router(config-dspfarm-profile)# codec h264 4cif
Router(config-dspfarm-profile)# codec h264 cif
Router(config-dspfarm-profile)# maximum conference-participants 8

Related Commands

Command	Description
associate application	Associates the SCCP protocol to the DSP farm profile.
dspfarm profile	Enters DSP farm profile configuration mode and defines a profile for DSP farm services.
maximum sessions (DSP Farm profile)	Specifies the maximum number of sessions that are supported by the profile.
rsvp	Enables RSVP support on a transcoding or MTP device.
maximum conference-participants (DSP Farm profile)	Specifies the maximum number of conference participants that are supported by this profile.
shutdown (DSP Farm profile)	Disables a DSP farm profile.

codec (voice-card)

To specify call density and codec complexity according to the codec standard that is being used or to increase processing frequency for the G.711 codec, use the codeccommand in voice-card configuration mode. To reset the flex complexity default or to disable configured values, use the no form of this command.

codec { complexity { flex [ reservation-fixed { high | medium } ] | high | medium | secure } | sub-sample }

no codec complexity

Syntax Description

complexity

Manages the complexity and density of codecs used in voice processing.

flex

When the flex keyword is used, up to 16 calls can be completed per digital signal processor (DSP). The number of supported calls varies from 6 to 16, depending on the codec used for a call. In this mode, reservation for analog voice interface cards (VICs) may be needed for certain applications such as Central Automatic Message Accounting (CAMA) E-911 calls because oversubscription of DSPs is possible. If this is true, enable the reservation-fixed keyword. There is no reservation by default.

reservation-fixed

(Optional) If you have specified the flex keyword, the reservation-fixed keyword ensures that sufficient DSP resources are available to handle a call. If you enter the reservation-fixed keyword, set the complexity for high or medium. (See the guidelines following to understand the effects of the keywords.) This option appears only when there is an analog VIC present.

high

If you specify the high keyword to define the complexity, each DSP supports two voice channels encoded in any of the following formats:

g711alaw--G.711 a-law 64,000 bps.
g711ulaw--G.711 mu-law 64,000 bps.
g723ar53--G.723.1 Annex A 5300 bps.
g723ar63--G.723.1 Annex A 6300 bps.
g723r53--G.723.1 5300 bps.
g723r63--G.723.1 6300 bps.
g726r16--G.726 16,000 bps.
g726r24--G726 24,000 bps.
g726r32--G.726 32,000 bps.
g728--G.728 16,000 bps.
g729r8--G.729 8000 bps. This is the default.
g729br8--G.729 Annex B 8000 bps.
fax relay--2400 bps,4800bps,7200bps,9600bps,12kbps,and14.4kbps.

Note

Codecs G.723.1 and G.728 are not supported on Cisco 1750 and Cisco 1751 modular access routers for Cisco Hoot and Holler over IP applications.

medium

If you specify the mediumkeyword to define the complexity, each DSP supports four voice channels encoded in any of the following formats:

g711alaw--G.711 a-law 64,000 bps.
g711ulaw--G.711 mu-law 64,000 bps.
g726r16--G.726 16,000 bps.
g726r24--G.726 24,000 bps.
g726r32--G.726 32,000 bps.
g729r8--G.729 Annex A 8000 bps.
g729br8--G.729 Annex B with Annex A 8000 bps.
fax relay--2400 bps,4800bps,7200bps,9600bps,12kbps,and14.4kbps.Faxrelayisthedefault.

secure

If you specify the securekeyword to define complexity, each DSP on an NM-HDV network module supports two voice channels encoded in any of the following formats:

g711alaw--G.711 a-law 64,000 bps.
g711ulaw--G.711 mu-law 64,000 bps.
g729--G.729 8000 bps.
g729A--G.729 8000 bps.

sub-sample

Increases the processing frequency for the G.711 codec with reduced 5510 DSP density.

Command Default

The default type of codec complexity is flex. The default value for the G.711 codec is 10 milliseconds (ms).

Command Modes

Voice-card configuration (config-voice-card)

Command History

Release	Modification
12.0(5)XK	This command was introduced as the codec complexity on the Cisco 2600 and Cisco 3600 series.
12.0(7)T	This command was integrated into Cisco IOS Release 12.0(7)T.
12.0(7)XK	This command was implemented on the Cisco MC3810 for use with the high-performance compression module (HCM).
12.1(2)T	This command was integrated into Cisco IOS Release 12.1(2)T.
12.2(8)T	This command was implemented on the Cisco 1750 and Cisco 1751.
12.2(13)T	The ecan-extendedkeyword was added.
12.2(15)T	This command was integrated into Cisco IOS Release 12.2(15)T with support for the Cisco 2600 series, Cisco 2600XM, Cisco 3660, Cisco 3725, and Cisco 3745 routers. High codec complexity is supported for DSP processing on these platforms.
12.2(15)ZJ	This command was integrated into Cisco IOS Release 12.2(15)ZJ and the flex keyword was added. The ecan-extended keyword was removed and G.168 echo-cancellation compliance became the default.
12.3(4)T	This command was integrated into Cisco IOS Release 12.3(4)T.
12.3(7)T	This command was integrated into Cisco IOS Release 12.3(7)T and the reservation-fixed keyword was added.
12.3(14)T	This command was integrated into Cisco IOS Release 12.3(14)T and the secure keyword was added to provide secure codec complexity for TI-549 DSP processing on the NM-HDV network module.
12.4(22)T1	The codeccomplexity command was changed to the codec(voice-card) command and the sub-sample keyword was added for the 5510 DSP.

Usage Guidelines

Codec complexity refers to the amount of processing required to perform voice compression. Codec complexity affects the call density--the number of calls reconciled on the DSPs. With higher codec complexity, fewer calls can be handled. Select a higher codec complexity if that is required to support a particular codec or combination of codecs. Select a lower codec complexity to support the greatest number of voice channels, provided that the lower complexity is compatible with the particular codecs in use.

For codec complexity to change, all of the DSP voice channels must be in the idle state.

When you have specified the flexkeyword, you can connect (or configure in the case of DS0 groups and PRI groups) more voice channels to the module than the DSPs can accommodate. If all voice channels should go active simultaneously, the DSPs become oversubscribed, and calls that are unable to allocate a DSP resource fail to connect. The flex keyword allows the DSP to process up to 16 channels. In addition to continuing support for configuring a fixed number of channels per DSP, theflex keyword enables the DSP to handle a flexible number of channels. The total number of supported channels varies from 6 to 16, depending on which codec is used for a call. Therefore, the channel density varies from 6 per DSP (high-complexity codec) to 16 per DSP (g.711 codec).

The high keyword selects a higher codec complexity if that is required to support a particular codec or combination of codecs. When you use the codeccomplexityhigh command to change codec complexity, the system prompts you to remove all existing DS0 or PRI groups using the specified voice card, then all DSPs are reset, loaded with the specified firmware image, and released.

The medium keyword selects a lower codec complexity to support the greatest number of voice channels, provided that the lower complexity is compatible with the particular codecs in use.

The secure keyword restricts the number of TI-549 DSP channels to 2, which is the lower codec complexity required to support Secure Real-Time Transport Protocol (SRTP) package capability on the NM-HDV and enable media authentication and encryption. If the secure command is not configured then the gateway will not advertise secure capability to Cisco CallManager, resulting in nonsecure calls. You do not need to use any command to specify secure codec complexity for TI-5510 DSPs, which support SRTP capability in all modes. Use the mgcppackage-capabilitysrtp-packagecommand to enable MGCP gateway capability to process SRTP packages. Use the showvoicedsp command to display codec complexity status.

Voice quality issues may occur when there are more than 15 G.711 channels on one 5510 DSP. To resolve the voice-quality issue, change the processing period (or segment size) of the G.711 codec from 5 ms to 10 ms. (The segment size of most voice codecs is 10 ms.) However, a voice call with 10-ms segment size has longer end-to-end delay (+ 5ms to 10 ms) than a call with 5-ms segment size.

Beginning in Cisco IOS Release 12.4(22)T1, the sub-sample keyword is added for applications that have strict requirements for round-trip delay times for VoIP. You can now accept the default G.711 (10 ms with lower MIPS) or enter the codecsub-sample command to select 5-ms G.711 (lower delay with higher MIPS). The sub-sample keyword is enabled only for the 5510 DSP.

The codecsub-sample command enables 5-ms processing for the G.711 codec inside the DSP to reduce the delay. However, this reduces the channel density of G.711 channels from 16 to 14. There is no difference in secure channel density when this mode is enabled.

Examples

The following example sets the codec complexity to high on voice card 1 installed on a router, and configures local calls to bypass the DSP:

voice-card 1
 codec complexity high
local-bypass

The following example sets the codec complexity to secure on voice card 1 installed on the NM-HDV, and configures it to support SRTP package processing, media authentication, and encryption:

voice-card 1
 codec complexity secure

The following example shows how to enable 5-ms processing for the G.711 codec inside the 5510 DSP:

voice-card 1
 codec sub-sample

Related Commands

Command	Description
ds0-group	Defines T1/E1 channels for compressed voice calls and the CAS method by which the router connects to the PBX or PSTN.
mgcp package-capability	Enables MGCP gateway capability to process SRTP packages.
show voice dsp	Displays the current status of all DSP voice channels.

codec aal2-profile

To set the codec profile for a digital signal processor (DSP) on a per-call basis, use the codecaal2-profile command in dial peer configuration mode. To restore the default codec profile, use the no form of this command.

codec aal2-profile { itut | custom | atmf } profile-number codec

no codec aal2-profile

Syntax Description

itut	Theprofile-number as an ITU-T type.
custom	The profile-number as a custom type.
atmf	The profile-number as an Asynchronous Transfer Mode Forum (ATMF) type.
profile -number	The availableprofile-number selections depend on the profile type. For ITU-T: 1 = G.711 mu-law 2 = G.711 mu-law with silence insertion descriptor (SID) 7 = G.711 mu-law and G.729ar8 For ATMF: 9 = Broadband Loop Emulation Services (BLES) support for VoAAL2 For custom: 100 = G.711 mu-law and G.726r32 110 = G.711 mu-law, G.726r32, and G.729ar8
codec	Enter one codec for the DSP. The possible codecentries depend on the profile-numbervalue.The valid entries are as follows: For ITU 1--g711mu-law For ITU 2--g711mu-law For ITU 7--g711mu-law or g729ar8 For ATMF--g711mu-law For custom 100--g711mu-law or g726r32 For custom 110--g711mu-law or g726r32 or g729ar8 For lossless compression--llcc

Command Default

ITU-T profile 1 (G.711 mu-law)

Command Modes

Dial peer configuration

Command History

Release	Modification
12.1(1)XA	This command was introduced on the Cisco MC3810.
12.1(2)T	This command was integrated into Cisco IOS Release 12.1(2)T.
12.2(2)T	This command was implemented on the Cisco 7200 series.
12.2(11)T	This command was implemented on the Cisco IAD2420 series.
12.3(4)XD	The lossless compression codec (llcc) keyword was added.
12.3(7)T	This command was integrated into Cisco IOS Release 12.3(7)T.

Usage Guidelines

Use this command to configure the DSP to operate with a specified profile type and codecs.

You must enter the sessionprotocolaal2-trunk command before configuring the codec ATM adaptation Layer 2 (AAL2) profile.

This command is used instead of the codec(dialpeer) command for AAL2 trunk applications.

Examples

The following example sets the codec AAL2 profile type to ITU-T and configures a profile number of 7, enabling codec G.729ar8:

dial-peer voice 100 voatm
 session protocol aal2-trunk
 codec aal2-profile itut 7 g729ar8

The following example sets the codec AAL2 profile type to custom and configures a profile number of 100, enabling codec G.726r32:

dial-peer voice 200 voatm
 session protocol aal2-trunk
 codec aal2-profile custom 100 g726r32

Related Commands

Command	Description
session protocol (dial peer)	Establishes a session protocol for calls between the local and remote routers via the packet network.

codec gsmamr-nb

To specify the Global System for Mobile Adaptive Multi-Rate Narrow Band (GSMAMR-NB) codec for a dial peer, use the codecgsmamr-nbcommand in dial peer voice configuration mode. To disable the GSMAMR-NB codec, use the no form of this command.

codec gsmamr-nb [ packetization-period 20 ] [ encap rfc3267 ] [ frame-format { bandwidth-efficient | octet-aligned [ crc | no-crc ] } ] [ modes modes-value ]

no codec gsmamr-nb

Syntax Description

packetization-period 20	(Optional) Sets the packetization period at 20 ms.
encap rfc3267	(Optional) Sets the encapsulation value to comply with RFC 3267.
frame-format	(Optional) Specifies a frame format. Supported values are octet-aligned and bandwidth-efficient. The default is octet-aligned.
crc \| no-crc	(Optional) CRC is applicable only for octet-aligned frame format. If you enter bandwidth-efficient frame format, the crc \| no-crc options will not be available because they are inapplicable.
modes	(Optional) The eight speech-encoding modes (bit rates between 4.75 and 12.2 kbps) available in the GSMAMR-NB codec.
modes-value	(Optional) Valid values are from 0 to 7. You can specify modes as a range (for example, 0-2), or individual modes separated by commas (for example, 2,4,6), or a combination of the two (for example, 0-2,4,6-7).

Command Default

Packetization period is 20 ms. Encapsulation is rfc3267. Frame format is octet-aligned. CRC is no-crc. Modes value is 0-7.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Release	Modification
12.4(4)XC	This command was introduced.
12.4(9)T	This command was integrated into Cisco IOS Release 12.4(9)T.

Usage Guidelines

The codecgsmamr-nb command configures the GSMAMR-NB codec and its parameters on the Cisco AS5350XM and Cisco AS5400XM platforms.

Examples

The following example sets the codec to gsmamr-nb and sets parameters:

Router(config-dial-peer)# codec gsmamr-nb packetization-period 20 encap rfc3267 frame-format octet-aligned crc

Related Commands

Command	Description
codec complexity	Specifies call density and codec complexity based on the codec used.
show dial peer voice	Displays the codec setting for dial peers.

codec ilbc

To specify the voice coder rate of speech for a dial peer using the internet Low Bandwidth Codec (iLBC), use the codecilbccommand in dial-peer configuration mode. To reset the default value, use the no form of this command.

codec ilbc [ mode frame_size [ bytes payload_size ] ]

no codec ilbc [ mode frame_size [ bytes payload_size ] ]

Syntax Description

mode	(Optional) Specifies the iLBC operating frame mode that is encapsulated in each packet.
frame_size	(Optional) iLBC operating frame in milliseconds (ms). Valid entries are: 20--20ms frames for 15.2kbps bit rate 30--30ms frames for 13.33 kbps bit rate Default is 20.
bytes	(Optional) Specifies the number of bytes in the voice payload of each frame.
payload_size	(Optional) Number of bytes in the voice payload of each frame. Valid entries are: For mode20--38, 76, 114, 152, 190, 228. Default is 38. For mode30--50, 100, 150, 200. Default is 50.

Command Default

20ms frames with a 15.2kbps bit rate.

Command Modes

Dial-peer configuration

Command History

Release	Modification
12.4(11)T	This command was introduced.
IOS Release XE 2.5	This command was integrated into Cisco IOS XE Release 2.5.

Usage Guidelines

Use thiscommand to define a specific voice coder rate of speech and payload size for a VoIP dial peer using an iLBC codec.

If codec values for the dial peers of a connection do not match, the call fails.

You can change the payload of each VoIP frame by using the byteskeyword. However, increasing the payload size can add processing delay for each voice packet.

Examples

The following example shows how to configure the iLBC codec on an IP-to-IP Gateway:

dial-peer voice 1 voip
 rtp payload-type cisco-codec-ilbc 100
 codec ilbc mode 30 bytes 200

Related Commands

Command	Description
show dial peer voice	Displays the codec setting for dial peers.

codec preference

To specify a list of preferred codecs to use on a dial peer, use the codecpreference command in voice-class configuration mode. To disable this functionality, use the no form of this command.

codec preference value codec-type [ mode { independent | adaptive } ] [ frame-size { 20 | 30 | 60 | fixed } ] [ bit rate value ] [ bytes payload-size ] [ packetization-period 20 ] [ encap rfc3267 ] [ frame-format { bandwidth-efficient | octet-aligned [ crc | no-crc ] } ] [ modes modes-value ]

no codec preference value codec-type

Syntax Description

value

The order of preference; 1 is the most preferred and 14 is the least preferred.

codec-type

The codec preferred. Values are as follows:

clear -channel--Clear Channel 64,000 bps.
g711alaw --G.711 a-law 64,000 bps.
g711ulaw --G.711 mu-law 64,000 bps.
g722r-64 --G.722-64 at 64,000 bps.
g723ar53 --G.723.1 Annex-A 5300 bps.
g723ar63 --G.723.1 Annex-A 6300 bps.
g723r53 --G.723.1 5300 bps.
g723r63 --G.723.1 6300 bps.
g726r16 --G.726 16,000 bps
g726r24 --G.726 24,000 bps
g726r32 --G.726 32,000 bps.
g728 --G.728 16,000 bps.
g729abr8 --G.729 ANNEX-A and B 8000 bps.
g729br8 --G.729 ANNEX-B 8000 bps.
g729r8 --G.729 8000 bps.
gsmamr-nb --Enables GSMAMR-NB codec capability.
gsmfr --Global System for Mobile Communications Full Rate (GSMFR) 13,200 bps.

Note

The gsmfr keyword is configurable only on the Cisco AS5350 and AS5400 with MSAv6 digital signal processors (DSPs).

ilbc --internet Low Bitrate Codec (iLBC) at 13,330 bps or 15,200 bps.
isac --Cisco internet Speech Audio Codec (iSAC) codec.
transparent --Enables codec capabilities to be passed transparently between endpoints.

Note

The transparent keyword is not supported when the call-start command is configured.

mode

(Optional) For iLBC and iSAC codecs only. Specifies the iLBC or iSAC operating frame mode that is encapsulated in each packet.

independent

(Optional) For iSAC codec only. Specifies that the configuration mode variable bit rate (VBR) is independent (value 1).

adaptive

(Optional) For iSAC codec only. Specifies that the configuration mode VBR is adaptive (value 0).

frame-size

(Optional) For iLBC and iSAC codecs only. Specifies the operating frame in milliseconds (ms). Valid entries are:

20 --20-ms frames (iLBC only)
30 --30-ms frames (iLBC or iSAC)
60 --60-ms frames (iLBC or iSAC)
fixed --This keyword is applicable only for adaptive mode.

bit rate value

(Optional) Configures the target bit rate in kilobits per second. The range is 10 to 32.

bytes

(Optional) Specifies that the size of the voice frame is in bytes.

payload-size

(Optional) Number of bytes that you specify as the voice payload of each frame. Values depend on the codec type and the packet voice protocol.

packetization-period 20

(Optional) Sets the packetization period at 20 ms. This keyword is applicable only to GSMAMR-NB codec support.

encap rfc3267

(Optional) Sets the encapsulation value to comply with RFC 3267. This keyword is applicable only to GSMAMR-NB codec support.

frame-format

(Optional) Specifies a frame format. Supported values are octet-aligned and bandwidth-efficient. The default is octet-aligned. This keyword is applicable only to GSMAMR-NB codec support.

crc | no-crc

(Optional) Cyclic Redundancy Check (CRC) is applicable only for octet-aligned frame format. If you enter bandwidth-efficient frame format, the crc | no-crcoptions are not available because they are inapplicable. This keyword is applicable only to GSMAMR-NB codec support.

modes modes-values

(Optional) Valid values are from 0 to 7. You can specify modes as a range (for example, 0-2), or individual modes separated by commas (for example, 2,4,6), or a combination of the two (for example, 0-2,4,6-7). This argument is applicable only to GSMAMR-NB codec support.

Command Default

If this command is not entered, no specific types of codecs are identified with preference.

If you enter the gsmamr-nb keyword, the default values are as follows:

Packetization period is 20 ms. Encap is rfc3267. Frame format is octet-aligned. CRC is no-crc. Modes value is 0-7.

If you enter the isac keyword, the default values are as follows:

Mode is independent. Target bit-rate is 32000bps. Framesize is 30ms.

Command Modes

Voice-class configuration (config-voice-class)

Command History

Release	Modification
12.0(2)XH	This command was introduced on the Cisco AS5300.
12.0(7)T	This command was implemented on the Cisco 2600 series and Cisco 3600 series.
12.0(7)XK	This command was implemented on the Cisco MC3810.
12.1(2)T	This command was integrated into Cisco Release IOS Release 12.1(2)T.
12.1(5)T	This command was modified. The gsmefr and gsmfr keywords were added.
12.2(13)T3	This command was modified.The transparent keyword was added.
12.4(4)XC	This command was extended to include GSMAMR-NB codec parameters on the Cisco AS5350XM and Cisco AS5400XM platforms.
12.4(9)T	This command was integrated into Cisco IOS Release 12.4(9)T.
12.4(11)T	This command was modified. The ilbc and mode keywords were added.
12.4(11)XJ2	This command was modified. The gsmefrand gsmfrkeywords were removed as configurable codec options for all platforms with the exception of the gsmfr codec on the Cisco AS5400 and AS5350 with MSAv6 dsps.
12.4(15)T	This command was integrated into Cisco IOS Release 12.4(15)T.
12.4(15)XY	This command was modified. The g722r-64 keyword was added.
12.4(20)T	This command was integrated into Cisco IOS Release 12.4(20)T.
IOS Release XE 2.5	This command was integrated into Cisco IOS XE Release 2.5.
15.1(1)T	This command was modified. Theisac keyword was added as a codec type, and the independent, adaptive, bitrate, and fixed keywords were added as configurable parameters.

Usage Guidelines

The routers at opposite ends of the WAN may have to negotiate the codec selection for the network dial peers. Thecodecpreference command specifies the order of preference for selecting a negotiated codec for the connection. The table below describes the voice payload options and default values for the codecs and packet voice protocols.

Note

The transparent keyword is not supported when the callstart command is configured.

Table 4 Voice Payload-per-Frame Options and Defaults
Codec	Protocol	Voice Payload Options (in Bytes)	Default Voice Payload (in Bytes)
g711alaw g711ulaw	VoIP VoFR VoATM	80, 160 40 to 240 in multiples of 40 40 to 240 in multiples of 40	160 240 240
g722r-64	VoIP	80, 160, 240	160
g723ar53 g723r53	VoIP VoFR VoATM	20 to 220 in multiples of 20 20 to 240 in multiples of 20 20 to 240 in multiples of 20	20 20 20
g723ar63 g723r63	VoIP VoFR VoATM	24 to 216 in multiples of 24 24 to 240 in multiples of 24 24 to 240 in multiples of 24	24 24 24
g726r16	VoIP VoFR VoATM	20 to 220 in multiples of 20 10 to 240 in multiples of 10 10 to 240 in multiples of 10	40 60 60
g726r24	VoIP VoFR VoATM	30 to 210 in multiples of 30 15 to 240 in multiples of 15 30 to 240 in multiples of 15	60 90 90
g726r32	VoIP VoFR VoATM	40 to 200 in multiples of 40 20 to 240 in multiples of 20 40 to 240 in multiples of 20	80 120 120
g728	VoIP VoFR VoATM	10 to 230 in multiples of 10 10 to 240 in multiples of 10 10 to 240 in multiples of 10	40 60 60
g729abr8 g729ar8 g729br8 g729r8	VoIP VoFR VoATM	10 to 230 in multiples of 10 10 to 240 in multiples of 10 10 to 240 in multiples of 10	20 30 30
ilbc	VoIP	For the mode20 keyword, 38,76, 114, 152, 190, 228 For the mode30 keyword, 50, 100, 150, 200	38 50
iSAC	VoIP	--	--

Examples

The following example show how to set the codec preference to the GSMAMR-NB codec and specify parameters:

Router(config-voice-class)# codec preference 1 gsmamr-nb packetization-period 20 encap rfc3267 frame-format octet-aligned crc

The following example shows how to create codec preference list 99 and applies it to dial peer 1919:

voice class codec 99
codec preference 1 g711alaw 
codec preference 2 g711ulaw bytes 80 
codec preference 3 g723ar53 
codec preference 4 g723ar63 bytes 144 
codec preference 5 g723r53 
codec preference 6 g723r63 bytes 120 
codec preference 7 g726r16 
codec preference 8 g726r24 
codec preference 9 g726r32 bytes 80 
codec preference 10 g729br8 
codec preference 11 g729r8 bytes 50
end 
dial-peer voice 1919 voip 
 voice-class codec 99

The following example shows how to configure the transparent codec used by the Cisco Unified Border Element:

voice class codec 99
codec preference 1 transparent

Note

You can assign a preference value of 1 only to the transparent codec. Additional codecs assigned to other preference values are ignored if the transparent codec is used.

The following example shows how to configure the iLBC codec used by the Cisco Unified Border Element:

voice class codec 99
codec preference 1 ilbc mode 30 bytes 200

Related Commands

Command	Description
call-start	Forces an H.323 Version 2 gateway to use fast connect or slow connect procedures for a dial peer.
voice class codec	Enters voice-class configuration mode and assigns an identification tag number to a codec voice class.
voice-class codec (dial peer)	Assigns a previously configured codec selection preference list to a dial peer.

codec profile

To define video capabilities needed for video endpoints, use the codecprofile command in telephony-service configuration mode. To disable the codec profile, use the no form of this command.

codec profile tag profile

no codec profile

Syntax Description

tag

A number in the range of 1 to 1000000.

profile

The name of the audio or video codec profile:

aacld
h263
h263+
h264

Command Default

No codec profile is configured.

Command Modes

Global configuration (config)

Command History

Release	Modification
12.4(22)T	This command was introduced.

Usage Guidelines

For the Cisco Unified Customer Voice Portal solution, only h263 and h263+ are supported profile options.

Examples

The following example shows the codec tagged 116 assigned to the H263 profile.

codec profile 116 H263
	clockrate 90000
	fmtp "fmtp:120 SQCIF=1;QCIF=1;CIF=1;CIF4=2;MAXBR=3840;I=1"

The codec profile can then be added to a voice class codec list, or the VoIP dial peer:

voice class codec 998
 codec preference 1 g711ulaw
 video codec h263 profile 116

Related Commands

Command	Description
clockrate	Sets the clock rate for the codec.
fmtp	Defines a string for video endpoints.

comfort-noise

To generate background noise to fill silent gaps during calls if voice activity detection (VAD) is activated, use the comfort-noise command in voice-port configuration mode. To provide silence when the remote party is not speaking and VAD is enabled at the remote end of the connection, use the no form of this command.

comfort-noise

no comfort-noise

Syntax Description

This command has no arguments or keywords.

Command Default

Background noise is generated by default.

Command Modes

Voice-port configuration (config-voiceport)

Command History

Release	Modification
11.3(1)T	This command was introduced on the Cisco 3600 series.
12.2(13)T	This command was integrated into Cisco IOS Release 12.2(13)T and was implemented on the Cisco 2600 series, the Cisco 7200 series, and the Cisco 7500 series using the extended echo canceller.

Usage Guidelines

Use the comfort-noisecommand to generate background noise to fill silent gaps during calls if VAD is activated. If the comfort-noise command is not enabled, and VAD is enabled at the remote end of the connection, the user hears dead silence when the remote party is not speaking.

The configuration of the comfort-noise command affects only the silence generated at the local interface; it does not affect the use of VAD on either end of the connection or the silence generated at the remote end of the connection.

Examples

The following example enables background noise on voice port 1/0/0:

voice-port 1/0/0
 comfort-noise

Related Commands

Command	Description
vad (dial peer configuration)	Enables VAD for the calls using a particular dial peer.
vad (voice-port configuration)	Enables VAD for the calls using a particular voice port.

compand-type

To specify the companding standard used to convert between analog and digital signals in pulse code modulation (PCM) systems, use the compand-type command in voice-port configuration mode. To disable the compand type, use the no form of this command.

compand-type { u-law | a-law }

no compand-type { u-law | a-law }

Syntax Description

u -law	Specifies the North American mu-law ITU-T PCM encoding standard.
a -law	Specifies the European a-law ITU-T PCM encoding standard.

Command Default

mu -law (T1 digital)a-law (E1 digital)

Command Modes

Voice-port configuration (config-voiceport)

Command History

Release	Modification
11.3(1)MA	This command was introduced.

Usage Guidelines

The Cisco 2660 and the Cisco 3640 routers do not require configuration of the compand-typea-law command. However, if you request a list of commands, the compand-typea-law command displays.

Note

On the Cisco 3600 series routers router, the mu-law and a-law settings are configured using the codec dial peer configuration command.

Note

This command is not supported on the Cisco AS 5300/5350/5400 and 5850 Universal Gateway series which use the Nextport DSP.

Examples

The following example configures a-law encoding on voice port 1/1:

voice-port 1/1
 compand-type a-law

Related Commands

Command	Description
codec (voice-port configuration)	Configures voice compression.

conference

To define a Feature Access Code (FAC) to initiate a three-party conference in feature mode on analog phones connected to FXS ports, use the conference command in STC application feature-mode call-control configuration mode. To return the code to its default, use the no form of this command.

conference keypad-character

no conference

Syntax Description

keypad-character

Character string of one to four characters that can be dialed on a telephone keypad (0-9, *, #). Default is #3.

Command Default

The default value is #3.

Command Modes

STC application feature-mode call-control configuration (config-stcapp-fmcode)

Command History

Release	Modification
15.0(1)M	This command was introduced.

Usage Guidelines

This command changes the value of the FAC for the Call Conference feature from the default (#3) to the specified value.

If you attempt to configure this command with a value that is already configured for another FAC in feature mode, you receive a message. This message will not prevent you from configuring the feature code. If you configure a duplicate FAC, the system implements the first feature it matches in the order of precedence as determined by the value for each FAC (#1 to #5).

If you attempt to configure this command with a value that precludes or is precluded by another FAC in feature mode, you receive a message. If you configure a FAC to a value that precludes or is precluded by another FAC in feature mode, the system always executes the call feature with the shortest code and ignores the longer code. For example, 1 will always preclude 12 and 123. These messages will not prevent you from configuring the feature code. You must configure a new value for the precluded code in order to enable phone user access to that feature.

Examples

The following example shows how to change the value of the feature code for Call Conference from the default (#3). With this configuration, a phone user presses hook flash to get the first dial tone, then dials an extension number to connect to a second call. When the second call is established, the user presses hook flash to get the feature tone and then dials 33 to initiate a three-party conference.

Router(config)# stcapp call-control mode feature
Router(config-stcapp-fmcode)# conference 33
Router(config-stcapp-fmcode)# exit

Related Commands

Command	Description
drop-last-conferee	Defines FAC in feature mode to use to drop last active call during a three-party conference.
hangup-last-active-call	Defines FAC in feature mode to drop last active call during a three-party conferencee.
toggle-between-two-calls	Defines FAC in feature mode to toggle between two active calls.
transfer	Defines FAC in feature mode to connect a call to a third party that the phone user dials.

conference-join custom-cptone

To associate a custom call-progress tone to indicate joining a conference with a DSP farm profile, use the conference-joincustom-cptone command in DSP farm profile configuration mode. To remove the custom call-progress tone association and disable the tone for the conference profile, use the no form of this command.

conference-join custom-cptone cptone-name

no conference-join custom-cptone cptone-name

Syntax Description

cptone-name

Descriptive identifier for this custom call-progress tone that indicates joining a conference.

Command Default

No custom call-progress tone to indicate joining a conference is associated with the DSP farm profile.

Command Modes

DSP farm profile configuration (config-dspfarm-profile)

Command History

Cisco IOS Release	Version	Modification
12.4(11)XJ2	Cisco Unified CME 4.1	This command was introduced.
12.4(15)T	Cisco Unified CME 4.1	This command was integrated into Cisco IOS Release 12.4(15)T

Usage Guidelines

To have a tone played when a party joins a conference, define the join tone, then associate it with the DSP farm profile for that conference.

Use the voiceclasscustom-cptone command to create a voice class for defining custom call-progress tones to indicate joining a conference.
Use the cadence and frequency commands to define the characteristics of the join tone.
Use the conference-joincustom-cptone command to associate the join tone to the DSP farm profile for that conference. Use the showdspfarmprofilecommand to display the DSP farm profile.

Examples

The following example defines a custom call-progress tone to indicate joining a conference and associates that join tone to a DSP farm profile defined for conferencing. Note that the custom call-progress tone names in the voiceclasscustom-cptone and conference-joincustom-cptone commands must be the same.

Router(config)# voice class custom-cptone jointone
Router(cfg-cptone)# dualtone conference
Router(cfg-cp-dualtone)# frequency 500 500
Router(cfg-cp-dualtone)# cadence 100 100 100 100 100
!
Router(config)# dspfarm profile 1 conference
Router(config-dspfarm-profile)# conference-join custom-cptone jointone

Related Commands

Command	Description
cadence	Defines the tone-on and tone-off durations for a call-progress tone.
conference-leave	Associates a custom call-progress tone to indicate leaving a conference with a DSP farm profile.
daultone conference	Enters cp-dualtone configuration mode for specifying a custom call-progress tone.
frequency	Defines the frequency components for a call-progress tone.
show dspfarm profile	Display configured digital signal processor (DSP) farm profile information.
voice class custom-cptone	Creates a voice class for defining custom call-progress tones to be detected.

conference-leave custom-cptone

To associate a custom call-progress tone to indicate leaving a conference with a DSP farm profile, use the conference-leavecustom-cptone command in DSP farm profile configuration mode. To remove the custom call-progress tone association and disable the tone for the conference profile, use the no form of this command.

conference-leave custom-cptone cptone-name

no conference-leave custom-cptone cptone-name

Syntax Description

cptone-name

Descriptive identifier for this custom call-progress tone that indicates leaving a conference.

Command Default

No custom call-progress tone to indicate leaving a conference is is associated with the DSP farm profile.

Command Modes

DSP farm profile configuration (config-dspfarm-profile)

Command History

Cisco IOS Release	Version	Modification
12.4(11)XJ2	Cisco Unified CME 4.1	This command was introduced.
12.4(15)T	Cisco Unified CME 4.1	This command was integrated into Cisco IOS Release 12.4(15)T

Usage Guidelines

For a tone to be played when a party leaves a conference, define the leave tone, then associate it with the DSP farm profile for that conference.

Use the voiceclasscustom-cptone command to create a voice class for defining custom call-progress tones to indicate leaving a conference.

Use the cadence and frequency commands to define the characteristics of the leave tone.

Use the conference-joincustom-cptone command to associate the leave tone to the DSP farm profile for that conference. Use the showdspfarmprofilecommand to display the DSP farm profile.

Examples

The following example defines a custom call-progress tone to indicate leaving a conference and associates that leave tone to a DSP farm profile defined for conferencing. Note that the custom call-progress tone names in the voiceclasscustom-cptone and conference-joincustom-cptone commands must be the same.

Router(config)# voice class custom-cptone leavetone
Router(cfg-cptone)# dualtone conference
Router(cfg-cp-dualtone)# frequency 500 500
Router(cfg-cp-dualtone)# cadence 100 100 100 100 100
!
Router(config)# dspfarm profile 1 conference
Router(config-dspfarm-profile)# conference-join custom-cptone leavetone

Related Commands

Command	Description
cadence	Defines the tone-on and tone-off durations for a call-progress tone.
conference-join	Associates a custom call-progress tone to indicate joining a conference with a DSP farm profile.
dualtone conference	Enters cp-dualtone configuration mode for specifying a custom call-progress tone.
frequency	Defines the frequency components for a call-progress tone.
show dspfarm profile	Display configured digital signal processor (DSP) farm profile information.
voice class custom-cptone	Creates a voice class for defining custom call-progress tones to be detected.

condition

To manipulate the signaling format bit-pattern for all voice signaling types, use the condition command in voice-port configuration mode. To turn off conditioning on the voice port, use the no form of this command.

condition { tx-a-bit | tx-b-bit | tx-c-bit | tx-d-bit } { rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit } { on | off | invert }

no condition { tx-a-bit | tx-b-bit | tx-c-bit | tx-d-bit } { rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit } { on | off | invert }

Syntax Description

tx -a-bit	Sends A bit.
tx -b-bit	Sends B bit.
tx -c-bit	Sends C bit.
tx -d-bit	Sends D bit.
rx -a-bit	Receives A bit.
rx -b-bit	Receives B bit.
rx -c-bit	Receives C bit.
rx -d-bit	Receives D bit.
on	Forces the bit state to 1.
off	Forces the bit state to 0.
invert	Inverts the bit state.

Command Default

The signaling format is not manipulated (for all sent or received A, B, C, and D bits).

Command Modes

Voice-port configuration (config-voiceport)

Command History

Release	Modification
11.3(1)MA	This command was introduced on the Cisco MC3810.
12.0(7)XK	This command was implemented on the Cisco 2600 series and 3600 series.
12.1(2)T	This command was integrated into Cisco IOS Release 12.1(2)T.

Usage Guidelines

Use the condition command to manipulate the sent or received bit patterns to match expected patterns on a connected device. Be careful not to destroy the information content of the bit pattern. For example, forcing the a-bit on or off prevents Foreign Exchange Office (FXO) interfaces from being able to generate both an on-hook and off-hook state.

The condition command is applicable to digital voice ports only.

Examples

The following example manipulates the signaling format bit pattern on digital voice port 0:5:

voice-port 0:5
 condition tx-a-bit invert
 condition rx-a-bit invert

The following example manipulates the signaling format bit pattern on voice port 1/0:0:

voice-port 1/0:0
 condition tx-a-bit invert
 condition rx-a-bit invert

Related Commands

Command	Description
define	Defines the transmit and receive bits for North American E&M and E&M MELCAS voice signaling.
ignore	Configures the North American E&M or E&M MELCAS voice port to ignore specific receive bits.

connect (channel bank)

To define connections between T1 or E1 controller ports for the channel bank feature, use the connectcommand in global configuration mode. To restore default values, use the no form of this command.

connect connection-id voice-port voice-port-number { t1 | e1 } controller-number ds0-group-number

no connect connection-id voice-port voice-port-number { t1 | e1 } controller-number ds0-group-number

Syntax Description

connection-id	A name for this connection.
voice-port	Specifies that a voice port is used in the connection.
voice-port-number	The voice port slot number and port number.
t1	Specifies a T1 port.
e1	Specifies an E1 port.
controller-number	The location of the first T1 or E1 controller to be connected. Valid values for the slot and port are 0 and 1.
ds0-group-number	The number identifier of the DS0 group associated with the first T1 or E1 controller port. The number is created by using the ds0-group command. Valid values are from 0 to 23 for T1 and from 0 to 30 for E1.

Command Default

There is no drop-and-insert connection between the ports.

Command Modes

Global configuration (config)

Command History

Release	Modification
12.0(5)XK	This command was introduced.
12.0(7)T	This command was integrated into Cisco IOS Release 12.0(7)T.
12.2(15)ZJ	The voice-port keyword was added.
12.3(4)T	This command was integrated into Cisco IOS Release 12.3(4)T.

Usage Guidelines

The connect command creates a named connection between two DS0 groups associated with voice ports on T1 or E1 interfaces where the groups have been defined by the ds0-group command.

Examples

The following example shows how to configure a channel bank connection for FXS loop-start signaling:

Router(config)# controller t1 1/0 
Router(config-controller)# ds0-group 1 timeslot 0 type fxo-loop-start
Router(config-controller)# exit
Router(config)# voice-port 1/1/0
Router(config-voiceport)# signal-type fxs-loop-start
Router(config-voiceport)# exit
Router(config)# connect connection1 voice-port 1/1/0 t1 1/0 0

Related Commands

Command	Description
ds0-group	Specifies the DS0 time slots that make up a logical voice port on a T1 or E1 controller and the signaling type by which the router communicates with the PBX or PSTN.
show connect	Displays configuration information about drop-and-insert connections that have been configured on a router.

connect (drop-and-insert)

To define connections among T1 or E1 controller ports for drop-and-insert (also called TDM cross-connect), use the connect command inglobal configuration mode. To restore default values, use the no form of this command.

connect connection-id { t1 | e1 } slotport-1 tdm-group-no-1 { t1 | e1 } slotport-2 tdm-group-no-2

no connect connection-id { t1 | e1 } slotport-1 tdm-group-no-1 { t1 | e1 } slotport-2 tdm-group-no-2

Syntax Description

connection-id	A name for this connection.
t1	Specifies a T1 port.
e1	Specifies an E1 port.
slotport -1	The location of the first T1 or E1 controller to be connected. Range for slot andport is 0 and 1.
tdm -group-no-1	The number identifier of the TDM) group associated with the first T1 or E1 controller port and created by using the tdm-group command. Range is from 0 to 23 for T1 and from 0 to 30 for E1.
slotport -2	The location of the second T1 or E1 controller port to be connected. Range for slot is from 0 to 5, depending on the platform. Range for port is from 0 to 3, depending on the platform and the presence of a network module.
tdm-group-no-2	The number identifier of the TDM group associated with the second T1 or E1 controller and created by using the tdm-group command. Range is from 0 to 23 for T1 and from 0 to 30 for E1.

Command Default

There is no drop-and-insert connection between the ports.

Command Modes

Global configuration (config)

Command History

Release	Modification
12.0(5)XK	The command was introduced on the Cisco 2600 series and Cisco 3600 series.
12.0(7)T	This command was integrated into Cisco IOS Release 12.0(7)T.
12.1(1)T	The command was modified to accommodate two channel groups on a port for 1- and 2-port T1/E1 multiflex voice/WAN interface cards (VWICs) on the Cisco 3600 series.

Usage Guidelines

The connect command creates a named connection between two TDM groups associated with drop-and-insert ports on T1 or E1 interfaces where you have already defined the groups by using the tdm-group command.

Once TDM groups are created on two different physical ports, use the connect command to start the passage of data between the ports. If a crosspoint switch is provided in the AIM slot, the connections can extend between ports on different cards. Otherwise, the connection is restricted to ports on the same VWIC.

The VWIC can make a connection only if the number of time slots at the source and destination are the same. For the connection to be error-free, the two ports must be driven by the same clock source; otherwise, slips occur.

Examples

The following example shows a fractional T1 terminated on port 0 using time slots 1 through 8, a fractional T1 is terminated on port 1 using time slots 2 through 12, and time slots 13 through 20 from port 0 are connected to time slots 14 through 21 on port 1 by using the connect command:

controller t1 0/0
 channel-group 1 timeslots 1-8
 tdm-group 1 timeslots 13-20
 exit
controller t1 0/1
 channel-group 1 timeslots 2-12
 tdm-group 2 timeslot 14-21
 exit
 connect exampleconnection t1 0/0 1 t1 0/1 2

Related Commands

Command	Description
show connect	Displays configuration information about drop-and-insert connections that have been configured on a router.
tdm -group	Configures a list of time slots for creating clear channel groups (pass-through) for TDM cross-connect.

connect atm

To define connections between T1 or E1 controller ports and the ATM interface, enter the connectatmcommand in global configuration mode. Use the no form of this command to restore the default values.

connect connection-id atm slot/port-1 { virtual-circuit-namevpi/vci { atm | T1 | E1 } } slot/port-2 TDM-group-number { virtual-circuit-name | vpi/vci }

Syntax Description

connection-id	A name for this connection.
atm	Specifies the first ATM interface.
slot/port-1	The location of the ATM controller to be connected.
virtual-circuit- name	Specifies the permanent virtual circuit (PVC) or switched virtual circuit (SVC).
vpi / vci	Specifies a virtual path identifier (VPI) and virtual channel identifier (VCI).
atm	Specifies the second ATM interface.
T1	Specifies a T1 port.
E1	Specifies an E1 port.
slot/port-2	The location of the T1 or E1 controller to be connected.
TDM-group-number	The number identifier of the time-division multiplexing (TDM) group associated with the T1 or E1 controller port and created by using the tdm-group command. Range is 0 to 23 for T1 and 0 to 30 for E1.

Command Default

No default behavior or values

Command Modes

Global configuration (config)

Command History

Release	Modification
12.1(2)T	This command was introduced for ATM interfaces on the Cisco 2600 series and Cisco 3600 series.
12.3(4)XD	ATM-to-ATM connections are allowed.
12.3(7)T	Support for ATM-to-ATM connections was integrated into Cisco IOS Release 12.3(7)T.

Usage Guidelines

This command is used on Cisco 2600, Cisco 3600, and Cisco 3700 series routers to provide connections between T1/E1 and ATM interfaces. This command is used after all interfaces are configured.

After TDM groups are created on two different physical ports, you can use the connectatmcommand to start the passage of data between the ports. If a crosspoint switch is provided in the advanced integration module (AIM) slot, the connections can extend between ports on different cards. Otherwise, the connection is restricted to ports on the same VWIC card.

The VWIC can make a connection only if the number of time slots at the source and destination are the same. For the connection to be error free, the two ports must be driven by the same clock source; otherwise, slips occur.

Examples

The following example shows how the ATM permanent virtual circuit (PVC) and T1 TDM group are set up and then connected:

interface atm 1/0
 pvc pvc1 10/100 ces
 exit
controller T1 1/1
 tdm-group 3 timeslots 13-24 type e&m
 exitconnect tdm1 atm 1/0 pvc1 10/100 T1 1/1 3

Related Commands

Command	Description
tdm-group	Creates TDM groups that can be connected.
pvc	Creates a private virtual circuit.

connect interval

To specify the amount of time that a given digital signal processor (DSP) farm profile waits before attempting to connect to a Cisco Unified CallManager when the current Cisco Unified CallManager fails to connect, use the connectintervalcommand in SCCP Cisco Unified CallManager configuration mode. To reset to the default value, use the no form of this command.

connect interval seconds

no connect interval

Syntax Description

seconds

Timer value, in seconds. Range is 1 to 3600. Default is 60.

Command Default

60 seconds

Command Modes

SCCP Cisco Unified CallManager configuration (config-sccp-ccm)

Command History

Release	Modification
12.3(8)T	This command was introduced.

Usage Guidelines

The optimum setting for this command depends on the platform and your individual network characteristics. Adjust the connect interval value to meet your needs.

Examples

The following example specifies that the profile attempts to connect to another Cisco Unified CallManager after 1200 seconds (20 minutes) when the current Cisco Unified CallManager connection fails:

Router(config-sccp-ccm)# connect interval 1200

Related Commands

Command	Description
associate ccm	Associates a Cisco Unified CallManager with a Cisco Unified CallManager group and establishes its priority within the group.
associate profile	Associates a DSP farm profile with a Cisco Unified CallManager group.
bind interface	Binds an interface to a Cisco Unified CallManager group.
connect retries	Specifies the number of times that a DSP farm attempts to connect to a Cisco Unified CallManager when the current Cisco Unified CallManager connections fails.
sccp ccm group	Creates a Cisco Unified CallManager group and enters SCCP Cisco Unified CallManager configuration mode.

connect retries

To specify the number of times that a digital signal processor (DSP) farm attempts to connect to a Cisco Unified CallManager when the current Cisco Unified CallManager connections fails, use the connectretriescommand in SCCP Cisco Unified CallManager configuration mode. To reset this number to the default value, use the no form of this command.

connect retries number

no connect retries

Syntax Description

number

Number of connection attempts. Range is 1 to 32. Default is 3.

Command Default

3 connection attempts

Command Modes

SCCP Cisco Unified CallManager configuration (config-sccp-ccm)

Command History

Release	Modification
12.3(8)T	This command was introduced.

Usage Guidelines

The value of this command specifies the number of times that the given DSP farm attempts to connect to the higher-priority Cisco Unified CallManager before it gives up and attempts to connect to the next Cisco Unified CallManager.

The optimum setting for this command depends on the platform and your individual network characteristics. Adjust the connect retries value to meet your needs.

Examples

The following example allows a DSP farm to make five attempts to connect to the Cisco Unified CallManager before giving up and attempting to connect to the next Cisco Unified CallManager specified in the group:

Router(config-sccp-ccm)# connect retries 5

Related Commands

Command	Description
associate ccm	Associates a Cisco Unified CallManager with a Cisco Unified CallManager group and establishes its priority within the group.
associate profile	Associates a DSP farm profile with a Cisco Unified CallManager group.
bind interface	Binds an interface to a Cisco Unified CallManager group.
connect interval	Specifies how many times a given profile attempts to connect to the specific Cisco Unified CallManager.
sccp ccm group	Creates a Cisco Unified CallManager group and enters SCCP Cisco Unified CallManager configuration mode.

connection

To specify a connection mode for a voice port, use the connection command in voice-port configuration mode. To disable the selected connection mode, use the no form of this command.

{ connection { plar | tie-line | plar opx [ cut-through-wait | immediate ] } phone-number | trunk phone-number [answer-mode] }

no { connection { plar | tie-line | plar opx [ cut-through-wait | immediate ] } phone-number | trunk phone-number [answer-mode] }

Syntax Description

plar

Specifies a private line automatic ringdown (PLAR) connection. PLAR is an autodialing mechanism that permanently associates a voice interface with a far-end voice interface, allowing call completion to a specific telephone number or PBX without dialing. When the calling telephone goes off-hook, a predefined network dial peer is automatically matched, which sets up a call to the destination telephone or PBX.

tie -line

Specifies a connection that emulates a temporary tie-line trunk to a private branch exchange (PBX). A tie-line connection is automatically set up for each call and torn down when the call ends.

plar opx

Specifies a PLAR off-premises extension (OPX) connection. Using this option, the local voice port provides a local response before the remote voice port receives an answer. On Foreign Exchange Office (FXO) interfaces, the voice port does not answer until the remote side has answered.

cut-through-wait

(Optional) Specifies that the router waits for the off-hook signal before cutting through the audio path.

Note

This keyword suppresses the subtle clicking sound that is heard when a phone goes off-hook. Users may have difficulty perceiving when the local FXO port has gone off-hook.

immediate

(Optional) Configures the FXO port to set up calls immediately (without waiting for Caller ID information) so the ring-cycle perception is identical for the caller and the called party. When the Caller ID is available, it is forwarded to the called number if the called party has not already answered the call.

Note

This option cannot be configured on an FXO port that is configured as a Centralized Automatic Message Accounting (CAMA) port.

phone-number

Specifies the destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number.

trunk

Specifies a connection that emulates a permanent trunk connection to a PBX. A trunk connection remains permanent in the absence of any active calls.

answer -mode

(Optional) Specifies that the router does not initiate a trunk connection but waits for an incoming call before establishing the trunk. Use only with the trunk keyword.

Command Default

No connection mode is specified, and the standard session application outputs a dial tone when the interface goes off-hook until enough digits are collected to match a dial peer and complete the call.

Command Modes

Voice-port configuration Router (config-voiceport)

Command History

Release	Modification
11.3(1)T	This command was introduced on the Cisco 3600 series.
11.3(1)MA1	This command was implemented on the Cisco MC3810, and the tie-line keyword added.
11.3(1)MA5	This command was modified. Theplaropx keyword was implemented on the Cisco MC3810 as the plar-opx-ringrelay keyword. The keyword was shortened in a subsequent release.
12.0(2)T	This command was integrated into Cisco IOS Release 12.0(2)T.
12.0(3)XG	This command was modified. The trunk keyword was implemented on the Cisco MC3810. The trunkanswer-mode option was added.
12.0(4)T	This command was integrated in Cisco IOS Release 12.0(4)T.
12.0(7)XK	This command was unified across the Cisco 2600, Cisco 3600, and Cisco MC3810.
12.1(2)T	This command was integrated into Cisco IOS Release 12.1(2)T.
12.3(8)T	This command was modified. The cut-through-wait keyword was added.
12.4(11)XW	This command was modified. The immediatekeyword was added.
12.4(20)T	This command was integrated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

Use the connection command to specify a connection mode for a specific interface. For example, use the connectionplar command to specify a PLAR interface. The string you configure for this command is used as the called number for all incoming calls over this connection. The destination peer is determined by the called number.

The connectionplaropximmediate option enables FXO ports to set up calls with no ring discrepancy for Caller ID between the caller and the called party. To implement the FXO Delayed Caller ID Delivery feature, you must have a configured network with a Cisco 2800 or Cisco 3800 series integrated services router running Cisco IOS Release 12.4(11)XW. The integrated services router must have at least one voice interface card. Cisco CallManager Release 4.2.3 SR1 or later releases must be installed on the network to support this feature.

The two figures below show the network topology and call flow for the FXO Delayed Caller ID feature. The caller is in the PSTN, and the call arrives via an FXO port at the gateway. In the figure below, the gateway is connected via H.323 to Cisco CallManager. Cisco CallManager extends the call to the called party which is a SCCP-based IP phone (Cisco 7941).

In the figure below, the gateway is on the same router as the figure above, and Survivable Remote Site Telephony (SRST) is active. SRST extends the call to the called party, which is a Skinny Client Control Protocol (SCCP)-based IP phone (Cisco 7941).

Use the connectiontrunk command to specify a permanent tie-line connection to a PBX. VoIP simulates a trunk connection by creating virtual trunk tie lines between PBXs connected to Cisco devices on each side of a VoIP connection (see Virtual Trunk Connection Figure). In this example, two PBXs are connected using a virtual trunk. PBX-A is connected to Router A via an E&M voice port; PBX-B is connected to Router B via an E&M voice port. The Cisco routers spoof the connected PBXs into believing that a permanent trunk tie line exists between them.

When configuring virtual trunk connections in VoIP, the following restrictions apply:

You can use the following voice port combinations:
- E&M to E&M (same type)
- Foreign Exchange Station (FXS) to Foreign Exchange Office (FXO)
- FXS to FXS (with no signaling)
Do not perform number expansion on the destination pattern telephone numbers configured for trunk connection.
Configure both end routers for trunk connections.

Note

Because virtual trunk connections do not support number expansion, the destination patterns on each side of the trunk connection must match exactly.

To configure one of the devices in the trunk connection to act as slave and only receive calls, use the answer-mode option with the connectiontrunk command when configuring that device.

Note

When using the connectiontrunk command, you must enter the shutdown command followed by the noshutdown command on the voice port.

VoIP establishes the trunk connection immediately after configuration. Both ports on either end of the connection are dedicated until you disable trunking for that connection. If for some reason the link between the two switching systems goes down, the virtual trunk reestablishes itself after the link comes back up.

Use the connectiontie-line command when the dial plan requires you to add digits in front of any digits dialed by the PBX, and the combined set of digits is used to route the call onto the network. The operation is similar to the connectionplar command operation, but in this case, the tie-line port waits to collect thedigits from the PBX. Tie-line digits are automatically stripped by a terminating port.

Examples

The following example shows PLAR as the connection mode with a destination telephone number of 555-0100:

voice-port 1/0/0
 connection trunk 5550100

The following example shows the tie-line as the connection mode with a destination telephone number of 555-0100:

voice-port 1/1
 connection tie-line 5550100

The following example shows a PLAR off-premises extension connection with a destination telephone number of 555-0100:

voice-port 1/0/0
 connection plar-opx 5550100

The following example shows a trunk connection configuration that is established only when the trunk receives an incoming call:

voice-port 1/0/0
 connection trunk 5550100 answer-mode

The following example shows a PLAR off-premises extension connection with a destination telephone number of 0199. The router waits for the off-hook signal before cutting through the audio path:

voice-port 2/0/0
 connection plar opx 0199 cut-through-wait

The following examples show configuration of the routers on both sides of a VoIP connection (as illustrated in the figure above) to support trunk connections.

Examples

voice-port 1/0/0
 connection trunk +15105550190
dial-peer voice 10 pots
 destination-pattern +13085550181
 port 1/0/0
dial-peer voice 100 voip
 session-target ipv4:172.20.10.10
 destination-pattern +15105550190

Examples

voice-port 1/0/0
 connection trunk +13085550180
dial-peer voice 20 pots
 destination-pattern +15105550191
 port 1/0/0
dial-peer voice 200 voip
 session-target ipv4:172.19.10.10
 destination-pattern +13085550180

Related Commands

Command	Description
destination-pattern	Specifies the prefix or the full E.164 telephone number for a dial peer.
dial peer voice	Enters dial peer configuration mode and specifies the voice encapsulation type.
session-protocol	Establishes a session protocol for calls between the local and remote routers via the packet network.
session-target	Configures a network-specific address for a dial peer.
shutdown	Takes a specific voice port or voice interface card offline.
voice-port	Enters voice-port configuration mode.

connection-timeout

To configure the time in seconds for which a connection is maintained after completion of a communication exchange, use the connection-timeout command in settlement configuration mode. To return to the default value, use the no form of this command.

connection-timeout seconds

no connection-timeout seconds

Syntax Description

seconds

Time, in seconds, for which a connection is maintained after the communication exchange is completed. Range is from 0 to 86400; 0 means that the connection does not time out. The default is 3600 (1 hour).

Command Default

3600 seconds (1 hour)

Command Modes

Settlement configuration (config-settlement)

Command History

Release	Modification
12.0(4)XH1	This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.0(7)T	This command was integrated into Cisco IOS Release 12.0(7)T.

Usage Guidelines

The router maintains the connection for the configured period in anticipation of future communication exchanges to the same server.

Examples

The following example shows a connection configured to be maintained for 3600 seconds after completion of a communications exchange:

settlement 0
 connection-timeout 3600

Related Commands

Command	Description
customer-id	Sets the customer identification.
device-id	Sets the device identification.
encryption	Specifies the encryption method.
max-connection	Sets the maximum simultaneous connections.
response-timeout	Sets the response timeout.
retry-delay	Sets the retry delay.
retry-limit	Sets the connection retry limit.
session-timeout	Sets the session timeout.
settlement	Enters settlement configuration mode.
show settlement	Displays the configuration for all settlement server transactions.
shutdown	Brings up or shuts down the settlement provider.
type	Specifies the provider type.
url	Specifies the Internet service provider address.

copy flash vfc

To copy a new version of VCWare from the Cisco AS5300 universal access server motherboard to voice feature card (VFC) flash memory, use the copyflashvfccommand inprivileged EXEC mode.

copy flash vfc slot-number

Syntax Description

slot -number

Slot on the Cisco AS5300 in which the VFC is installed. Range is from 0 to 2.

Command Modes

Privileged EXEC (#)

Command History

Release	Modification
11.3NA	This command was introduced on the Cisco AS5300.

Usage Guidelines

Use the copyflashvfccommand to use the standard copy user interface in order to copy a new version of VCWare from the Cisco AS5300 universal access server motherboard to VFC flash memory. The VFC is a plug-in feature card for the Cisco AS5300 universal access server and has its own Flash memory storage for embedded firmware. For more information about VFCs, refer to Voice-over-IP Card.

Once the VCWare file has been copied, use the unbundlevfc command to uncompress and install VCWare.

Examples

The following example copies a new version of VCWare from the Cisco AS5300 universal access server motherboard to VFC flash memory:

Router# copy flash vfc 0

Related Commands

Command	Description
copy tftp vfc	Copies a new version of VCWare from a TFTP server to VFC flash memory.
unbundle vfc	Unbundles the current running image of VCWare or DSPWare into separate files.

copy tftp vfc

To copy a new version of VCWare from a TFTP server to voice feature card (VFC) flash memory, use the copytftpvfccommand in privileged EXEC mode.

copy tftp vfc slot-number

Syntax Description

slot -number

Slot on the Cisco AS5300 in which the VFC is installed. Range is from 0 to 2. There is no default.

Command Default

No default behavior or values

Command Modes

Privileged EXEC (#)

Command History

Release	Modification
11.3NA	This command was introduced on the Cisco AS5300.

Usage Guidelines

Use the copytftpvfccommand to copy a new version of VCWare from a TFTP server to VFC flash memory. The VFC is a plug-in feature card for the Cisco AS5300 universal access server and has its own flash storage for embedded firmware. For more information about VFCs, refer to Voice-over-IP Card.

Once the VCWare file has been copied, use the unbundlevfc command to uncompress and install VCWare.

Examples

The following example copies a file from the TFTP server to VFC flash memory:

Router# copy tftp vfc 0

Related Commands

Command	Description
copy flash vfc	Copies a new version of VCWare from the Cisco AS5300 motherboard to VFC flash memory.
unbundle vfc	Unbundles the current running image of VCWare or DSPWare into separate files.

corlist incoming

To specify the class of restrictions (COR) list to be used when a specified dial peer acts as the incoming dial peer, use the corlistincoming command in dial peer configuration mode. To clear the previously defined incoming COR list in preparation for redefining the incoming COR list, use the no form of this command.

corlist incoming cor-list-name

no corlist incoming cor-list-name

Syntax Description

cor-list-name

Name of the dial peer COR list that defines the capabilities that the specified dial peer has when it is used as an incoming dial peer.

Command Default

No default behavior or values.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Release	Modification
12.1(3)T	This command was introduced.

Usage Guidelines

The dial-peercorlist and member commands define a set of capabilities (a COR list). These lists are used in dial peers to indicate the capability set that a dial peer has when it is used as an incoming dial peer (the corlistincoming command) or to indicate the capability set that is required for an incoming dial peer to make an outgoing call through the dial peer (the corlistoutgoing command). For example, if dial peer 100 is the incoming dial peer and its incoming COR list name is list100, dial peer 200 has list200 as the outgoing COR list name. If list100 does not include all the members of list200 (that is, if list100 is not a superset of list200), it is not possible to have a call from dial peer 100 that uses dial peer 200 as the outgoing dial peer.

Examples

In the following example, incoming calls from 526.... are blocked from being switched to outgoing calls to 1900.... because the COR list for the incoming dial peer (list2) is not a superset of the COR list for the outgoing dial peer (list1):

dial-peer list list1
 member 900call
dial-peer list list2
 member 800call
 member othercall
dial-peer voice 526 pots
 answer-address 408555....
 corlist incoming list2
 direct-inward-dial
dial-peer voice 900 pots
 destination pattern 1900.......
 direct-inward-dial
 trunkgroup 101
 prefix 333
 corlist outgoing list1

Related Commands

Command	Description
corlist outgoing	Specifies the COR list to be used by outgoing dial peers.
dial-peer cor list	Defines a COR list name.
member	Adds a member to a dial peer COR list.

corlist outgoing

To specify the class of restrictions (COR) list to be used by outgoing dial peers, use the corlistoutgoingcommand in dial peer configuration mode. To clear the previously defined outgoing COR list in preparation for redefining the outgoing COR list, use the no form of this command.

corlist outgoing cor-list-name

no corlist outgoing cor-list-name

Syntax Description

cor-list-name

Required name of the dial peer COR list for outgoing calls to the configured number using this dial peer.

Command Default

No default behavior or values.

Command Modes

Dial peer configuration (config-dial-peer)

Command History

Release	Modification
12.1(3)T	This command was introduced.

Usage Guidelines

If the COR list for the incoming dial peer is not a superset of the COR list for the outgoing dial peer, calls from the incoming dial peer cannot use that outgoing dial peer.

Examples

In the following example, incoming calls from 526.... are blocked from being switched to outgoing calls to 1900.... because the COR list for the incoming dial peer (list2) is not a superset of the COR list for the outgoing dial peer (list1):

dial-peer list list1
member 900call
dial-peer list list2
 member 800call
 member othercall
dial-peer voice 526 pots
 answer-address 408555....
 corlist incoming list2
 direct-inward-dial
dial-peer voice 900 pots
 destination pattern 1900.......
 direct-inward-dial
 trunk group 101
 prefix 333
 corlist outgoing list1

cpa

To enable the call progress analysis (CPA) algorithm for outbound VoIP calls and to set CPA parameters, use the cpa command in voice service configuration mode. To disable the CPA algorithm, use the no form of this command.

cpa [ threshold { active-signal { 9db | 12db | 15db | 18db | 21db } | noise-level { max { -45dBm0 | -50dBm0 | -55dBm0 | -60dBm0 } | min { -55dBm0 | -60dBm0 | -65dBm0 | -70dBm0 } } } | timing { live-person max-duration | noise-period max-duration | silent min-duration | term-tone max-duration | timeout max-duration | valid-speech min-duration } ]

no cpa

Syntax Description

threshold	(Optional) Sets the CPA thresholds, in decibels (dB).
active-signal	(Optional) Sets the active signal threshold that is related to the measured noise floor level.
9dB \| 12dB \| 15dB \| 18dB \| 21dB	(Optional) Specifies active signal thresholds above the measured noise floor level (in dB). The default value is 15 dB.
noise-level	(Optional) Sets the CPA noise floor level limits.
max	(Optional) Sets the maximum noise floor level.
-45dBm0 \| -50dBm0 \| -55dBm0 \| -60dBm0	(Optional) Specifies maximum noise floor level values (root mean square), in dBm0, where dBm0 is decibels referred to one milliwatt and corrected to a 0-dBm effective power level. The default value is -50 dBm0.
min	(Optional) Sets the minimum noise floor level.
-55dBm0 \| -60dBm0 \| -65dBm0 \| -70dBm0	(Optional) Minimum noise floor level values, in dBm0, where dBm0 is decibels referred to one milliwatt and corrected to a 0-dBm effective power level. The default value is -60 dBm0. Note that this value must be less than or equal to the value configured by the cpa threshold noise-level max command.
timing	(Optional) Sets the CPA timing parameters.
live-person max-duration	(Optional) Sets the maximum waiting time (in milliseconds) that the CPA algorithm uses to determine if the call is answered by a living person. The range is from 1 to 60000. The default value is 2500.
noise-period max-duration	(Optional) Sets the maximum waiting time (in milliseconds) that the CPA algorithm uses to measure the noise floor level at the beginning of the call. The range is from 1 to 60000. The default value is 100.
silent min-duration	(Optional) Sets the minimum silent duration (in milliseconds) afer active speech is detected for the CPA algorithm to declare that the call is answered by a live human. The range is from 1 to 60000. The default value is 375.
term-tone max-duration	(Optional) Sets the maximum waiting time (in milliseconds) that the CPA algorithm uses to wait for the answering machine termination tone after the answering machine is detected. The range is from 1 to 60000. The default value is 15000.
timeout max-duration	(Optional) Sets the maximum waiting time (in milliseconds) that the CPA algorithm uses to timeout if it does not detect any voice signal. The range is from 1 to 60000. The default value is 3000.
valid-speech min-duration	(Optional) Sets the minimum voice duration (in milliseconds) for the CPA algorithm to consider it as a valid speech signal. The range is from 1 to 60000. The default value is 112.

Command Default

The CPA algorithm is enabled for outbound VoIP calls.

Command Modes

Voice service configuration (conf-voi-serv)

Command History

Release	Modification
12.4(24)T	This command was introduced.
Cisco IOS XE Release 3.9S	This command was integrated into Cisco IOS XE Release 3.9S.

Usage Guidelines

Use the cpa command to enable the call progress analysis algorithm for outbound VoIP calls. You must activate the CPA capability using the call-progress-analysis command in digital signal processor (DSP) farm profile configuration mode before you use the cpa command to configure values for threshold and timing parameters.

Examples

The following example shows how to enable CPA and configure the timing and threshold parameters:

Device> enable
Device# configure terminal
Device(config)# voice service voip
Device(conf-voi-serv)# cpa
Device(conf-voi-serv)# cpa threshold active-signal 12dB
Device(conf-voi-serv)# cpa threshold noise-level max -55dBm0
Device(conf-voi-serv)# cpa threshold noise-level min -65dBm0
Device(conf-voi-serv)# cpa timing live-person 5000
Device(conf-voi-serv)# cpa timing timeout 2000
Device(conf-voi-serv)# cpa timing term-tone 7000
Device(conf-voi-serv)# cpa timing silent 380
Device(conf-voi-serv)# cpa timing valid-speech 113
Device(conf-voi-serv)# cpa timing noise-period 101
Device(conf-voi-serv)# exit

Related Commands

Command	Description
call-progress-analysis	Activates CPA for a DSP farm profile on the Cisco UBE.
dspfarm profile	Enters DSP farm profile configuration mode and defines a profile for DSP farm services.
noisefloor	Configures the noise level, in dBm, above which noise reduction (NR) will operate.

cptone

To specify a regional analog voice-interface-related tone, ring, and cadence setting for a voice port, use the cptone command in voice-port configuration mode. To disable the selected tone, use the no form of this command.

cptone locale

no cptone locale

Syntax Description

locale

Country-specific voice-interface-related default tone, ring, and cadence setting (for ISDN PRI and E1 R2 signaling). Keywords are shown in the table below. The default keyword is us in Cisco IOS Release 12.0(4)T and later releases.

Command Default

The default keyword is us for all supported gateways and interfaces in Cisco IOS Release 12.0(4)T and later releases.

Command Modes

Voice-port configuration (config-voiceport)

Command History

Release	Modification
11.3(1)T	This command was introduced on the Cisco 3600 series.
11.3(1)MA	This command was modified. The full keyword names for the countries were first added on the Cisco MC3810.
12.0(4)T	This command was modified. ISO 3166 two-letter country codes were added on the Cisco MC3810.
12.1(5)XM	This command was modified. The following keywords were added: eg, gh, jo, ke, lb, ng,np, pa,pk, sa, and zw.
12.2(2)T	This command was implemented on the Cisco 1750 and integrated into Cisco IOS Release 12.2(2)T.
12.2(15)ZJ	This command was modified. The c1 and c2 keywords were added for the following platforms: Cisco 2610XM, Cisco 2611XM, Cisco 2620XM, Cisco 2621XM, Cisco 2650XM, Cisco 2651XM, Cisco 2691, Cisco 3640A, Cisco 3660, Cisco 3725, and Cisco 3745.
12.3(4)T	This command was integrated into Cisco IOS Release 12.3(4)T.
12.4(15)T	This command was modified. The following keywords were added: ae, kw, and om.
15.0(1)M	This command was modified. The cl keyword was added.
15.1(3)T	This command was modified. The mt keyword was added.

Usage Guidelines

This command defines the detection of call-progress tones generated at the local interface. It does not affect any information passed to the remote end of a connection, and it does not define the detection of tones generated at the remote end of a connection. Use the cptone command to specify a regional analog voice interface-related default tone, ring, and cadence setting for a specified voice port.

If your device is configured to support E1 R2 signaling, the E1 R2 signaling type (whether ITU, ITU variant, or local variant as defined by the cas-customcommand) must match the appropriate pulse code modulation (PCM) encoding type as defined by the cptone command. For countries for which a cptone value has not yet been defined, you can try the following:

If the country uses a-law E1 R2 signaling, use the gb value for the cptone command.
If the country uses mu-law E1 R2 signaling, use the us value for the cptone command.

The table below lists valid entries for the locale argument.

Table 5 Valid Command Entries for locale Argument
Country	cptone locale Command Entry	Country	cptone locale Command Entry
Argentina	ar	Lebanon	lb
Australia	au	Luxembourg	lu
Austria	at	Malaysia	my
Belgium	be	Malta	mt
Brazil	br	Mexico	mx
Canada	ca	Nepal	np
Chile	cl	Netherlands	nl
China	cn	New Zealand	nz
Colombia	co	Nigeria	ng
Custom 1 ¹	c1	Norway	no
Custom 2 ²	c2	Oman	om
Czech Republic	cz	Pakistan	pk
Denmark	dk	Panama	pa
Egypt	eg	Peru	pe
Finland	fi	Philippines	ph
France	fr	Poland	pl
Germany	de	Portugal	pt
Ghana	gh	Russian Federation	ru
Great Britain	gb	Saudi Arabia	sa
Greece	gr	Singapore	sg
Hong Kong	hk	Slovakia	sk
Hungary	hu	Slovenia	si
Iceland	is	South Africa	za
India	in	Spain	es
Indonesia	id	Sweden	se
Ireland	ie	Switzerland	ch
Israel	il	Taiwan	tw
Italy	it	Thailand	th
Japan	jp	Turkey	tr
Jordan	jo	United Arab Emirates	ae
Kenya	ke	United States	us
Korea Republic	kr	Venezuela	ve
Kuwait	kw	Zimbabwe	zw

¹ Automatically configured the first time the XML file is downloaded to the gateway.

² Automatically configured the first time the XML file is downloaded to the gateway.

Examples

The following example configures United States as the call-progress tone locale:

voice-port 1/0/0
 cptone us

The following example configures Brazil as the call-progress tone locale on a Cisco universal access server:

voice-port 1:0
 cptone br
 description Brasil Tone

Related Commands

Command	Description
voice-port	Enters voice-port configuration mode.
cas-custom	Customizes signaling parameters for a particular E1 or T1 channel group on a channelized line.

cptone call-waiting repetition interval

To set the call-waiting alert pattern on analog endpoints that are connected to Foreign Exchange Station (FXS) ports, use the cptonecall-waitingrepetitioninterval command in supplementary-service voice-port configuration mode. To return to the default behavior, use the no form of this command.

cptone call-waiting repetition interval second

no cptone call-waiting repetition interval

Syntax Description

second

Length of time, in seconds for the tone repetition interval. Range: 0 to 30. Default: 0.

Command Default

A single-beep tone is the default behavior.

Command Modes

Supplementary-service voice-port configuration (config-stcapp-suppl-serv-port)

Command History

Release	Modification
15.1(3)T	This command was introduced.

Usage Guidelines

Use the cptonecall-waitingrepetitioninterval command to set the call-waiting alert pattern on analog endpoints that are connected to FXS ports on a Cisco IOS voice gateway, such as a Cisco Integrated Services Router (ISR) or Cisco VG224 Analog Phone Gateway.

When configured, the ringtone periodically repeats with configured interval until either the user switches to the new call or the calling party hangs up.

Examples

The following example shows how to set the call-waiting alert pattern on analog endpoints connected to port 2/0 on a Cisco VG224:

Router(config)# stcapp supplementary-services
Router(config-stcapp-suppl-serv)# port 2/0
Router(config-stcapp-suppl-serv-port)# cptone call-waiting repetition interval 20
Router(config-stcapp-suppl-serv-port)# end

Related Commands

Command	Description
stcapp supplementary-services	Enters supplementary-service configuration mode for configuring STCAPP supplementary-service features on an FXS port.

credential load

To reload a credential file into flash memory, use the credentialload command in privileged EXEC mode.

credential load tag

Syntax Description

tag	Number that identifies the credential (.csv) file to load. Range: 1 to 5. This is the number that was defined with the authenticatecredential command.

Command Default

The credential file is not reloaded.

Command Modes

Privileged EXEC (#)

Command History

Release	Modification
12.4(11)XJ	This command was introduced.
12.4(15)T	This command was integrated into Cisco IOS Release 12.4(15)T.

Usage Guidelines

This command provides a shortcut to reload credential files that were defined with the authenticatecredential command.

Up to five .csv files can be configured and loaded into the system. The contents of these five files are mutually exclusive, that is, the username/password pairs must be unique across all the files. For Cisco Unified CME, these username/password pairs cannot be the same ones defined for SCCP or SIP phones with the usernamecommand.

Examples

The following example shows how to reload credential file 3:

credential load 3

Related Commands

Command	Description
authenticate (voice register global)	Defines the authenticate mode for SIP phones in a Cisco Unified CME or Cisco Unified SRST system.
username (ephone)	Defines a username and password for SCCP phones.
username (voice register pool)	Defines a username and password for authenticating SIP phones.

credentials (SIP UA)

To configure a Cisco IOS Session Initiation Protocol (SIP) time-division multiplexing (TDM) gateway, a Cisco Unified Border Element (Cisco UBE), or Cisco Unified Communications Manager Express (Cisco Unified CME) to send a SIP registration message when in the UP state, use the credentials command in SIP UA configuration mode. To disable SIP digest credentials, use the no form of this command.

credentials { dhcp | number number username username } password [ 0 | 7 ] password realm realm

no credentials { dhcp | number number username username } password [ 0 | 7 ] password realm realm

Syntax Description

dhcp	(Optional) Specifies the Dynamic Host Configuration Protocol (DHCP) is to be used to send the SIP message.
number number	(Optional) A string representing the registrar with which the SIP trunk will register (must be at least four characters).
username username	A string representing the username for the user who is providing authentication (must be at least four characters). This option is only valid when configuring a specific registrar using the number keyword.
password	Specifies password settings for authentication.
0	(Optional) Specifies the encryption type as cleartext (no encryption). This is the default.
7	(Optional) Specifies the encryption type as encrypted.
password	A string representing the password for authentication. If no encryption type is specified, the password will be cleartext format. The string must be between 4 and 128 characters.
realm realm	(Optional) A string representing the domain where the credentials are applicable.

Command Default

SIP digest credentials are disabled.

Command Modes

SIP UA configuration (sip-ua)

Command History

Release	Modification
12.3(8)T	This command was introduced.
12.4(22)T	This command was integrated into Cisco IOS Release 12.4(22)T.
12.4(22)YB	This command was modified. The dhcp keyword was added and the username keyword and username argument were removed.
15.0(1)M	This command was integrated into Cisco IOS Release 15.0(1)M.
15.0(1)XA	This command was modified. The number keyword and number argument were added and the username keyword and username argument reintroduced to configure credentials for a given registrar when multiple registrars are configured.
15.1(1)T	This command was integrated into Cisco IOS Release 15.1(1)T.

Usage Guidelines

The following configuration rules are applicable when credentials are enabled:

Only one password is valid for all domain names. A new configured password overwrites any previously configured password.
The password will always be displayed in encrypted format when the credentials command is configured and the showrunning-config command is used.

The dhcp keyword in the command signifies that the primary number is obtained via DHCP and the Cisco IOS SIP TDM gateway, Cisco UBE, or Cisco Unified CME on which the command is enabled uses this number to register or unregister the received primary number.

Examples

The following example shows how to configure SIP digest credentials without specifying the password encryption type:

Router> enable
Router# configure terminal
Router(config)# sip-ua
Router(config-sip-ua)# credentials dhcp password MyPassword realm example.com

The following example shows how to configure SIP digest credentials using the encrypted format:

Router> enable
Router# configure terminal
Router(config)# sip-ua
Router(config-sip-ua)# credentials dhcp password 7 095FB01AA000401 realm example.com

The following example shows how to disable SIP digest credentials where the encryption type was specified:

Router> enable
Router# configure terminal
Router(config)# sip-ua
Router(config-sip-ua)# no credentials dhcp password 7 095FB01AA000401 realm example.com

The following example shows how to configure SIP digest credentials for two different realms without specifying the encryption type:

Router> enable
Router# configure terminal
Router(config)# sip-ua
Router(config-sip-ua)# credentials number 1111 username MyUser password MyPassword realm MyLocation1.example.com
Router(config-sip-ua)# credentials number 1111 username MyUser password MyPassword realm MyLocation2.example.com

Related Commands

Command	Description
authentication (dial peer)	Enables SIP digest authentication on an individual dial peer.
authentication (SIP UA)	Enables SIP digest authentication.
localhost	Configures global settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages.
registrar	Enables Cisco IOS SIP TDM gateways to register E.164 numbers for FXS, EFXS, and SCCP phones on an external SIP proxy or SIP registrar.
voice-class sip localhost	Configures settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages on an individual dial peer, overriding the global setting.