Voice entity commands -- answer-address

Use answer-address to configure a calling number string for a voice entity to match incoming calls.

Use undo answer-address to restore the default.

Syntax

answer-address calling-number-string

undo answer-address

Default

No calling number string is configured for a voice entity to match incoming calls.

Views

POTS entity view

VoIP entity view

Predefined user roles

network-admin

mdc-admin

Parameters

calling-number-string: Specifies a calling number string of 1 to 31 characters that is in the format of [ + ] { regular-expression [ T ] [ $ ] | T }. The following describe the characters:

· Plus sign (+): If the plus sign (+) is at the beginning of the string, the string indicates an E.164 standard number. For example, +110022 indicates that 110022 is an E.164 standard number.

· Dollar sign ($): Can be used only at the end of the string. The number must exactly match the string before the dollar sign. If the string has no dollar sign, the number template matches all numbers starting with the string. For example, the answer-address 20 command matches all numbers starting with 20.

· T: Indicates the timer. The system waits for the subscriber to dial any number until one of the following events occurs:

?span style='font:7.0pt "Times New Roman"'> The number length threshold is exceeded.

?span style='font:7.0pt "Times New Roman"'> The subscriber enters the terminator.

?span style='font:7.0pt "Times New Roman"'> The timer expires.

· regular-expression: Specifies a matching pattern of characters. Table 1 lists the available characters. Brackets ([ ]) or parentheses (( )) each are two characters, and any other symbol is one character.

Table 1 Description of the characters in a regular-expression

Character | Description
0-9 | Digits 0 through 9.
Pound sign (#) or asterisk (*) | Indicates a valid digit.
Dot (.) | Wildcard, which can match any valid digit. For example, 555…. can match any 7-digit number beginning with 555.
Exclamation point (!) | Indicates that the preceding subexpression appears zero or one time. For example, 56!1234 can match 51234 and 561234.
Plus sign (+) | Indicates that the preceding subexpression appears one or more times. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on.
Percent sign (%) | Indicates that the preceding subexpression appears zero or more times. For example, 9876(54)% can match 9876, 987654, 98765454, 9876545454, and so on.
Hyphen (-) | Connects two digits to indicate a range of numbers, for example, [1-9] indicates 1 to 9, inclusive. The hyphen (-) can appear only in brackets ([ ]).
Brackets ([ ]) | Indicates a range. Only numbers 0 through 9 are allowed in the range. For example, [1-36] matches 1, 2, 3, or 6.
Parentheses (( )) | Indicates a string of characters. For example, (123) indicates the character string 123. It is usually used together with signs such as exclamation point (!), percent sign (%), and plus sign (+). For example, 408(12)+ can match the character string 40812 or 408121212, but not 408 (that is, the string 12 must appear a minimum of one time).

| NOTE: · An exclamation point (!), plus sign (+), or percent sign (%) must follow a valid digit or digit string. · To use brackets ([ ]) and parentheses (( )) together, use them in the form of "( [ ] )". The "( ( ) )", "[ [ ] ]", and "[ ( ) ]" forms are not allowed.

Usage guidelines

If the calling number of an incoming call matches the calling number string for a voice entity, the voice entity becomes the incoming voice entity of the call.

Examples

# Configure the calling number string as 456 for VoIP entity 1 to match incoming calls.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1 voip

[Sysname-voice-dial-entity1] answer-address 456

Voice entity commands -- codec

Use codec to configure a codec for a voice entity.

Use undo codec to restore the default.

Syntax

codec { g711alaw | g711ulaw | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r40 | g729a | g729br8 | g729r8 } [ bytes payload-size ]

undo codec

Default

No codec is configured for a voice entity.

Views

POTS entity view

VoIP entity view

IVR entity view

Predefined user roles

network-admin

mdc-admin

Parameters

g711alaw: Specifies the G.711 A-Law codec at 64 kbps (without compression), which is typically used in Europe.

g711ulaw: Specifies the G.711 μ-Law codec at 64 kbps (without compression), which is typically used in North America and Japan.

g723r53: Specifies the G.723.1 Annex A codec at 5.3 kbps.

g723r63: Specifies the G.723.1 Annex A codec at 6.3 kbps.

g726r16: Specifies the G.726 Annex A codec at 16 kbps. Support for this keyword depends on the line card.

g726r24: Specifies the G.726 Annex A codec at 24 kbps. Support for this keyword depends on the line card.

g726r32: Specifies the G.726 Annex A codec at 32 kbps. Support for this keyword depends on the line card.

g726r40: Specifies the G.726 Annex A codec at 40 kbps. Support for this keyword depends on the line card.

g729a: Specifies the G.729 Annex A codec (a simplified version of G.729) at 8 kbps.

g729br8: Specifies the G.729 Annex B codec at 8 kbps.

g729r8: Specifies the G.729 codec at 8 kbps.

bytes payload-size: Specifies the number of bytes sent per second.

Table 2 Value range and default of payload-size for codecs

Codec | Value range (in bytes) | Default (in bytes)
g711alaw g711ulaw | 16 to 80 in multiples of 8, 80 to 240 in multiples of 80 | 160
g723r53 | 20 to 120 in multiples of 20 | 20
g723r63 | 24 to 144 in multiples of 24 | 24
g726r16 | 20 to 220 in multiples of 20 | 60
g726r24 | 30 to 210 in multiples of 30 | 90
g726r32 | 40 to 200 in multiples of 40 | 120
g726r40 | 50 to 200 in multiples of 50 | 150
g729a g729br8 g729r8 | 10 to 180 in multiples of 10 | 30

Usage guidelines

To establish a call, you must make sure the calling party and the called party use the same codec.

You can use this command to directly configure a codec for a voice entity, or use the voice-class codec command to bind a codec template to a voice entity.

The g711alaw and g711ulaw codecs provide high-quality voice transmission but consume high bandwidth.

The g723r53 and g723r63 codecs provide silence suppression technology and comfortable noise as follows:

· The relatively high speed output is based on multipulse multiquantitative level technology and provides relatively high voice quality.

· The relatively low speed output is based on the Algebraic-Code-Excited Linear-Prediction technology and provides greater flexibility for applications.

The g729r8 and g729a codecs provide a voice quality (nearly toll quality) similar to the 32-kbps adaptive differential pulse code modulation (ADPCM). These two codecs feature low bandwidth, short delay, and medium processing complexity.

Table 3 Voice quality for codecs

Codec | Voice quality
g711alaw g711ulaw | Excellent
g726r16 g726r24 g726r32 g726r40 | Good
g729a g729br8 g729r8 | Good
g723r53 g723r63 | Average

If you execute this command multiple times, the most recent configuration takes effect.

Examples

# Configure the codec as g711alaw for VoIP entity 10.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 voip

[Sysname-voice-dial-entity10] codec g711alaw

Voice entity commands -- codec preference

Use codec preference to assign a priority to a codec in a codec template.

Use undo codec preference to delete the priority for a codec.

Syntax

codec preference priority { g711alaw | g711ulaw | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r40 | g729a | g729br8 | g729r8 } [ bytes payload-size ]

undo codec preference priority

Default

No codecs exist in a codec template.

Views

Codec template view

Predefined user roles

network-admin

mdc-admin

Parameters

priority: Specifies the priority of a codec, in the range of 1 to 4. The smaller the value, the higher the priority.

g711alaw: Specifies the G.711 A-Law codec at 64 kbps (without compression), which is typically used in Europe.

g711ulaw: Specifies the G.711 μ-Law codec at 64 kbps (without compression), which is typically used in North America and Japan.

g723r53: Specifies the G.723.1 Annex A codec at 5.3 kbps.

g723r63: Specifies the G.723.1 Annex A codec at 6.3 kbps.

g726r16: Specifies the G.726 Annex A codec at 16 kbps. Support for this keyword depends on the line card.

g726r24: Specifies the G.726 Annex A codec at 24 kbps. Support for this keyword depends on the line card.

g726r32: Specifies the G.726 Annex A codec at 32 kbps. Support for this keyword depends on the line card.

g726r40: Specifies the G.726 Annex A codec at 40 kbps. Support for this keyword depends on the line card.

g729a: Specifies the G.729 Annex A codec (a simplified version of G.729) at 8 kbps.

g729br8: Specifies the G.729 Annex B codec at 8 kbps.

g729r8: Specifies the G.729 codec at 8 kbps.

bytes payload-size: Specifies the number of bytes sent per second.

Table 4 Value range and default of payload-size for codecs

Codec | Value range (in bytes) | Default (in bytes)
g711alaw g711ulaw | 16 to 80 in multiples of 8, 80 to 240 in multiples of 80 | 160
g723r53 | 20 to 120 in multiples of 20 | 20
g723r63 | 24 to 144 in multiples of 24 | 24
g726r16 | 20 to 220 in multiples of 20 | 60
g726r24 | 30 to 210 in multiples of 30 | 90
g726r32 | 40 to 200 in multiples of 40 | 120
g726r40 | 50 to 200 in multiples of 50 | 150
g729a g729br8 g729r8 | 10 to 180 in multiples of 10 | 30

Usage guidelines

For information about the codecs, see the codec command.

Examples

# Configure the g711alaw codec to have the highest priority 1.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] voice class codec 1

[Sysname-voice-class-codec1] codec preference 1 g711alaw

Voice entity commands -- description

Use description to configure a description for a voice entity.

Use undo description to restore the default.

Syntax

description text

undo description

Default

No description is configured for a voice entity.

Views

POTS entity view

VoIP entity view

IVR entity view

Predefined user roles

network-admin

mdc-admin

Parameters

text: Specifies a description, a case-sensitive string of 1 to 80 characters.

Examples

# Configure the description as room10 for POTS entity 10.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 pots

[Sysname-voice-dial-entity10] description room10

Voice entity commands -- display voice call

Use display voice call to display control information for voice calls in progress.

Syntax

display voice call

Views

Any view

Predefined user roles

network-admin

network-operator

mdc-admin

mdc-operator

Examples

# As shown in Figure 1, after Telephone 2222 (the calling party) establishes a call with Telephone 1111, display control information for voice calls in progress.

Figure 1 Network diagram

<RouterB> display voice call

Voice call information:

Call1

CallID : 6

Calling number : 2222

Called number : 1111

Call info-table index : 0

Total call-legs : 2

Leg 1

LegID : 10

Leg type : Call-Leg

Status : Connected

Call reference ID : 3

Signal protocol : LGS

Voice line : 2/1/2

Leg 2

LegID : 11

Leg type : Call-Leg

Status : Connected

Call reference ID : 4

Signal protocol : SIP

Target SIP address : 192.168.2.1:5060

# As shown in Figure 1, after Telephone 1111 (the calling party) establishes a call with Telephone 2222 and then Telephone 2222 presses hookflash to place the call on hold, display control information for voice calls in progress.

<RouterB> display voice call

Voice call information:

Call1

CallID : 7

Calling number : 1111

Called number : 2222

Call info-table index : 0

Total call-legs : 2

Leg 1

LegID : 17

Leg type : Call-Leg

Status : Connected

Call reference ID : 7

Signal protocol : SIP

Target SIP address : 192.168.2.1:5060

Leg 2

LegID : 18

Leg type : Call-Leg

Status : Connected

Call reference ID : 14

Signal protocol : LGS

Voice line : 2/1/2

Number of services : 1

Service name : CH

Table 5 Command output

Field | Description
CallID | A call ID in the range of 0 to 999 uniquely identifies a call.
Total call-legs | Total number of call legs, in the range of 0 to 3.
LegID | A leg ID in the range of 0 to 2999 uniquely identifies a leg.
Leg type | Leg type: · Call Leg—Call leg. A voice call has two call legs: an inbound call leg and an outbound call leg. · Temp Leg—Temporary leg. This leg type exists on a device operating as a SIP trunk device. · MOH Leg—Music on hold leg.
Status | Leg status: · Call leg status: ?span style='font:7.0pt "Times New Roman"'> Finding-route—The leg is waiting for a route lookup response. ?span style='font:7.0pt "Times New Roman"'> Incoming_ACK—The leg received a call. ?span style='font:7.0pt "Times New Roman"'> Outgoing_ACK—The leg sent out a call. ?span style='font:7.0pt "Times New Roman"'> Connected—A call was connected. · MOH leg status: ?span style='font:7.0pt "Times New Roman"'> Waiting-music-response—The leg is waiting for a response from the MOH server. ?span style='font:7.0pt "Times New Roman"'> MOH_connected—The leg established a connection with the MOH server. This field displays -NA- for temporary legs, because temporary legs have no status.
Call reference ID | Control block ID for the leg.
Signal protocol | Signaling type for the leg: · SIP. · LGS. · R2. · E&M. · IVA.
Voice line | Voice interface used by the leg.
Number of services | Number of services on the leg.
Service name | Service name: · CH—Call hold. · CW—Call waiting. · MCH—Multiparty call hold. · MOH—Music on hold. · CT—Call transfer for SIP-to-SIP calls. · CF—Call forwarding for SIP-to-SIP calls. · CB—Call backup. · CFO—Call forwarding originator. · CTO—Call transfer originator. · CTR—Call transfer recipient. · CTT—Call transfer target. · Conference—Three-party conference.

Voice entity commands -- display voice call-info

Use display voice call-info to display information about calls in progress.

Syntax

display voice call-info { tag | all }

Views

Any view

Predefined user roles

network-admin

network-operator

mdc-admin

mdc-operator

Parameters

tag: Specifies a call in progress by its number in the range of 0 to 511.

all: Specifies all calls in progress.

Examples

# Display information about all calls in progress.

<Sysname> display voice call-info all

Call tag 0

Caller number : 5000

Called number : 1000

Call direction : From packet switch

Voice interface index : 0x00000000

Voice entity currently used : 1

Voice entities offered : 1

Table 6 Command output

Field | Description
Call direction | · From packet switch—The call is initiated from the IP side. · From circuit switch—The call is initiated from the PSTN side.
Voice interface index | Index of the voice interface that initiates the call.
Voice entities offered | Number of voice entities that can be used for the call.

Voice entity commands -- display voice entity

Use display voice entity to display the configuration of voice entities.

Syntax

display voice entity { entity-tag | all | ivr | pots | voip }

Views

Any view

Predefined user roles

network-admin

network-operator

mdc-admin

mdc-operator

Parameters

entity-tag: Specifies a voice entity by its number in the range of 1 to 2147483647.

all: Specifies all voice entities.

ivr: Specifies IVR entities.

pots: Specifies POTS entities.

voip: Specifies VoIP entities.

Examples

# Display the configuration of all voice entities.

<Sysname> display voice entity all

POTS 9999

Current state: Up

Description: entity9999

Priority level: 0

Match template: 9999

Voice line: 2/2/1

Dial prefix: Not configured

Send number: All

Max connections: 50

Codec: g723r53; bytes: 80; vad: Disabled

Caller permit: 1

Caller group: permit group 1

Substitute called: 9999

Substitute calling: 9999

DTMF relay: Outband-NTE

RTP payload-type for NTE: 113

Playout mode: adaptive

Playout initial delay: 30 ms

Playout minimum delay: 10 ms

Playout maximum delay: 160 ms

IP media DSCP: ef

IP signaling DSCP: ef

Register number: Enabled

Call-forwarding no-reply number: 5555

Call-forwarding on-busy number: 6666

Call-forwarding unavailable number: 7777

Call-forwarding unconditional number: 8888

Authentication info:

Username: 1000

Password: ******

Realm: abc.com

VoIP 8888

Current state: Up

Description: Not configured

Priority level: 0

Match template: 8888

Target SIP address: 1.1.1.1

Max connections: 10

Caller permit: 1

Caller group: permit group 1

Substitute called: 9999

Substitute calling: 9999

DTMF relay: Outband-SIP

Playout mode: adaptive

Playout initial delay: 30 ms

Playout minimum delay: 10 ms

Playout maximum delay: 160 ms

IP media DSCP: ef

Codec transparent: Disabled

Media flow-around: Enabled

Voice class SIP early-offer forced: Disabled

Voice class SIP URI scheme: Global

Voice class SIP bind media source-interface: GigabitEthernet2/1/1

Voice class SIP bind control source-interface: GigabitEthernet2/1/1

Voice class SIP keepalive state: Available

Voice class SIP keepalive up-interval: 60 s

Voice class SIP keepalive down-interval: 30 s

Voice class SIP keepalive retry: 5

Fax protocol: standard-t38; ls-redundancy: 0; hs-redundancy: 0

Fax cng-switch: Disabled

Fax level: -15

Fax local-train threshold: 10

Fax nsf: 0x000000

Fax rate: Voice

Fax train-mode: PPP

Fax ecm: Disabled

Table 7 Command output

Field | Description
VoIP entity-number | Voice entity type and number. The voice entity type can be VoIP, POTS, or IVR.
Match template | Number template of the voice entity.
Target SIP address | Call destination IP address of the voice entity.
Voice line | Voice interface bound to the voice entity.
Send number | Number sending mode: · All—Sends all digits of a called number. · Truncate—Sends a truncated called number. · number—Number of digits (that are extracted from the end of a number) to be sent.
bytes: 80 | Number of bytes sent per second.
Caller permit | Calling number permitted to originate calls to the voice entity.
Caller group | Subscriber group bound to the voice entity.
Substitute called | Number substitution rule list bound to the voice entity and applied to the called number.
Substitute calling | Number substitution rule list bound to the voice entity and applied to the calling number.
DTMF relay | · Outband-SIP—DTMF tones are transmitted in SIP packets. · Outband-NTE—DTMF tones are transmitted in RTP packets compliant with RFC 2833. · Inband-voice—DTMF tones are transmitted in RTP packets.
Playout mode | Playout delay mode: · adaptive. · fixed.
Playout initial delay | Initial playout delay time.
Playout minimum delay | Minimum playout delay time.
Playout maximum delay | Maximum playout delay time.
IP media DSCP | DSCP value of IP packets carrying streaming media.
Codec transparent | State of transparent transmission of codecs: Enabled or Disabled.
Media flow-around | State of the media flow-around feature: Enabled or Disabled.
Voice class SIP early-offer forced | State of DO-EO conversion: Enabled or Disabled.
Voice class SIP URI scheme | URL scheme used for SIP calls: · Global—The SIP scheme is used globally. · SIP—The voice entity uses the SIP scheme. · SIPS—The voice entity uses the SIPS scheme.
Voice class SIP bind media | Source interface of outgoing media streams.
Voice class SIP bind control | Source interface of outgoing SIP messages.
Voice class codec | Codec template bound to the voice entity.
Call-forwarding no-reply number | Destination number to which incoming calls will be forwarded when the voice interface is not answered within a period of time.
Call-forwarding on-busy number | Destination number to which incoming calls will be forwarded when the voice interface is busy.
Call-forwarding unavailable number | Destination number to which incoming calls will be forwarded when the voice interface is shut down by executing the shutdown command.
Call-forwarding unconditional number | Destination number to which incoming calls will be forwarded, whether or not the voice interface is available.
Voice class SIP keepalive state | Status of the VoIP entity: · Available. · Unavailable.
Voice class SIP keepalive up-interval | Interval for the local end to send OPTIONS messages before marking the voice entity unavailable.
Voice class SIP keepalive down-interval | Interval for the local end to send OPTIONS messages before marking the voice entity available.
Voice class SIP keepalive retry | Number of keepalives sent before the status of the voice entity is changed.
Fax cng-switch | CNG fax switchover status: · Enabled. · Disabled.
Fax level | Transmit energy level.
Fax local-train threshold | Threshold percentage of local training.
Fax nsf | NSF code for nonstandard capabilities negotiation.
Fax rate | Maximum fax rate for rate training.
Fax train-mode | Rate training mode: · Local—Local training. · PPP—Point-to-point training.
Fax ecm | Error Correction Mode status: Enabled or Disabled.

Voice entity commands -- dsp-image

Use dsp-image to set the type of the DSP image.

Syntax

dsp-image { ms | general }

Default

The DSP image is a general version.

Views

Voice view

Predefined user roles

network-admin

mdc-admin

Parameters

ms: Specifies a Microsoft-verified version. This version can meet the voice quality requirements of Microsoft but does not support the G.723 codec.

general: Specifies a general version.

Usage guidelines

After you execute this command, you must reboot the device to apply the new configuration.

When the device interoperates with Microsoft Lync Server, you must use the Microsoft-verified version. In other situations, use the general version.

Examples

# Configure the DSP image as a Microsoft-verified version.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dsp-image ms

Voice entity commands -- entity

Use entity to create a voice entity and enter its view, or enter the view of an existing voice entity.

Use undo entity to delete voice entities.

Syntax

entity entity-number [ ivr | pots | voip ]

undo entity { entity-number | all | ivr | pots | voip }

Default

No voice entities exist.

Views

Voice dial program view

Predefined user roles

network-admin

mdc-admin

Parameters

entity-number: Specifies the number of a voice entity, in the range of 1 to 2147483647.

all: Specifies all voice entities.

ivr: Specifies an IVR entity.

pots: Specifies a POTS entity.

voip: Specifies a VoIP entity.

Usage guidelines

If you create a new voice entity, you must specify the voice entity type. If you enter the view of an existing voice entity, you can optionally specify the voice entity type.

You can create a maximum of 1000 voice entities.

Examples

# Create POTS entity 10 and enter its view.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 pots

Voice entity commands -- incoming called-number

Use incoming called-number to configure a called number string for a voice entity to match incoming calls.

Use undo incoming called-number to restore the default.

Syntax

incoming called-number called-number-string

undo incoming called-number

Default

No called number string is configured for a voice entity to match incoming calls.

Views

POTS entity view

VoIP entity view

Predefined user roles

network-admin

mdc-admin

Parameters

called-number-string: Specifies a called number string of 1 to 31 characters that is in the format of [ + ] { regular-expression [ T ] [ $ ] | T }. The following describe the characters:

· Plus sign (+): If the plus sign (+) is at the beginning of the string, the string indicates an E.164 standard number. For example, +110022 indicates that 110022 is an E.164 standard number.

· Dollar sign ($): Can be used only at the end of the string. The number must exactly match the string before the dollar sign. If the string has no dollar sign, the number template matches all numbers starting with the string. For example, the incoming called-number 20 command matches all numbers starting with 20.

· T: Indicates the timer. The system waits for the subscriber to dial any number until one of the following events occurs:

?span style='font:7.0pt "Times New Roman"'> The number length threshold is exceeded.

?span style='font:7.0pt "Times New Roman"'> The subscriber enters the terminator.

?span style='font:7.0pt "Times New Roman"'> The timer expires.

· regular-expression: Specifies a matching pattern of characters. Table 8 lists the available characters.

Table 8 Description of the characters in a regular-expression

Character | Description
0-9 | Digits 0 through 9.
Pound sign (#) or asterisk (*) | Indicates a valid digit.
Dot (.) | Wildcard, which can match any valid digit. For example, 555…. can match any 7-digit numbers beginning with 555.
Exclamation point (!) | Indicates that the preceding subexpression appears zero or one time. For example, 56!1234 can match 51234 and 561234.
Plus sign (+) | Indicates that the preceding subexpression appears one or more times. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on.
Percent sign (%) | Indicates that the preceding subexpression appears zero or more times. For example, 9876(54)% can match 9876, 987654, 98765454, 9876545454, and so on.
Hyphen (-) | Connects two digits to indicate a range of numbers, for example, [1-9] indicates 1 to 9, inclusive. The hyphen (-) can appear only in brackets ([ ]).
Brackets ([ ]) | Indicates a range. Only numbers 0 through 9 are allowed in the range. For example, [1-36] matches 1, 2, 3, or 6.
Parentheses (( )) | Indicates a string of characters. For example, (123) indicates a character string of 123. It is usually used together with signs such as exclamation point (!), percent sign (%), and plus sign (+). For example, 408(12)+ can match the character string 40812 or 408121212. In this pattern, 408 must be followed by one string of 12 at a minimum.

| NOTE: · An exclamation point (!), plus sign (+), or percent sign (%) must follow a valid digit or digit string. · To use brackets ([ ]) and parentheses (( )) together, use them in the form of "( [ ] )". The "( ( ) )", "[ [ ] ]", and "[ ( ) ]" forms are not allowed.

Usage guidelines

If the called number of an incoming call matches the called number string for a voice entity, the voice entity becomes the incoming voice entity of the call.

Examples

# Configure the called number string as 456 for VoIP entity 1 to match incoming calls.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1 voip

[Sysname-voice-dial-entity1] incoming called-number 456

Voice entity commands -- ip qos dscp

Use ip qos dscp to set the DSCP value for IP packets carrying media streams.

Use undo ip qos dscp to restore the default.

Syntax

ip qos dscp { dscp-value | dscp-value-set } media

undo ip qos dscp { dscp-value | dscp-value-set } media

Default

The DSCP value for IP packets is ef (101110), the global default value.

Views

POTS entity view

VoIP entity view

IVR entity view

Predefined user roles

network-admin

mdc-admin

Parameters

dscp-value: Specifies a DSCP value in the range of 0 to 63.

dscp-value-set: DSCP value, which can be the keyword af11, af12, af13, af21, af22, af23, af31, af32, af33, af41, af42, af43, cs1, cs2, cs3, cs4, cs5, cs6, cs7, or ef.

Table 9 DSCP values

Keyword | DSCP value in binary | DSCP value in decimal
af11 | 001010 | 10
af12 | 001100 | 12
af13 | 001110 | 14
af21 | 010010 | 18
af22 | 010100 | 20
af23 | 010110 | 22
af31 | 011010 | 26
af32 | 011100 | 28
af33 | 011110 | 30
af41 | 100010 | 34
af42 | 100100 | 36
af43 | 100110 | 38
cs1 | 001000 | 8
cs2 | 010000 | 16
cs3 | 011000 | 24
cs4 | 100000 | 32
cs5 | 101000 | 40
cs6 | 110000 | 48
cs7 | 111000 | 56
ef | 101110 | 46

Usage guidelines

You can set the DSCP value for IP packets carrying media streams both globally (in SIP view) and for a specific voice entity (in POTS/VoIP entity view). The configuration in POTS/VoIP entity view takes precedence over the global configuration. A voice entity uses the global configuration only when the ip qos dscp command is not configured in POTS/VoIP entity view.

Examples

# Configure DSCP value af41 for IP packets carrying media streams.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 pots

[Sysname-voice-dial-entity10] ip qos dscp af41 media

Related commands

ip qos dscp (SIP view)

Voice entity commands -- line

Use line to bind a voice interface to a POTS entity.

Use undo line to restore the default.

Syntax

line line-number

undo line

Default

No voice interface is bound to a POTS entity.

Views

POTS entity view

Predefined user roles

network-admin

mdc-admin

Parameters

line-number: Specifies a voice interface by its number.

Examples

# Bind voice interface 1/0 to POTS entity 10.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 pots

[Sysname-voice-dial-entity10] line 1/0

Voice entity commands -- match-template

Use match-template to configure a number template for a voice entity.

Use undo match-template to restore the default.

Syntax

match-template match-string

undo match-template

Default

No number template exists.

Views

POTS entity view

VoIP entity view

IVR entity view

Predefined user roles

network-admin

mdc-admin

Parameters

match-string: Specifies a number template, a string of 1 to 31 characters that is in the format of [ + ] { string [ T ] [ $ ] | T }. The following describe the characters:

· Plus sign (+): If the plus sign (+) is at the beginning of the string, the string indicates an E.164 standard number. For example, +110022 indicates that 110022 is an E.164 standard number.

If a number starts with the plus sign (+), note the following when you use it on a trunk:

?span style='font:7.0pt "Times New Roman"'> The E&M, R2, and LGS signaling methods use DTMF transmission. Because the plus sign (+) does not have a DTMF tone, the number cannot be transmitted to the called side successfully.

?span style='font:7.0pt "Times New Roman"'> Because the DSS1 signaling uses ISDN transmission, this problem does not exist.

?span style='font:7.0pt "Times New Roman"'> You should avoid using a number that cannot be identified by the signaling itself. Otherwise, the call will fail.

· Dollar sign ($): Can be only at the end of the string. The number must exactly match the string before the dollar sign. If the string has no dollar sign, the number template matches all numbers starting with the string. For example, the match-template 20 command matches all numbers starting with 20.

· T: Indicates the timer. The system waits for the subscriber to dial any number until one of the following conditions occurs:

?span style='font:7.0pt "Times New Roman"'> The number length threshold is exceeded.

?span style='font:7.0pt "Times New Roman"'> The subscriber enters the terminator.

?span style='font:7.0pt "Times New Roman"'> The timer expires.

· string: Specifies a matching pattern of characters. Table 10 lists the available characters.

Table 10 Description of the characters in a string

Character | Description
0-9 | Digits 0 through 9.
Pound sign (#) or asterisk (*) | Indicates a valid digit.
Dot (.) | Wildcard, which can match any valid digit. For example, 555…. can match any 7-digit numbers beginning with 555.
Exclamation point (!) | Indicates that the preceding subexpression appears zero or one time. For example, 56!1234 can match 51234 and 561234.
Plus sign (+) | Indicates that the preceding subexpression appears one or more times. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on.
Percent sign (%) | Indicates that the preceding subexpression appears zero or more times. For example, 9876(54)% can match 9876, 987654, 98765454, 9876545454, and so on.
Hyphen (-) | Connects two digits to indicate a range of numbers, for example, [1-9] indicates 1 to 9, inclusive. The hyphen (-) can appear only in brackets ([ ]).
Brackets ([ ]) | Indicates a range. Only numbers 0 through 9 are allowed in the range. For example, [1-36] matches 1, 2, 3, or 6.
Parentheses (( )) | Indicates a string of characters. For example, (123) indicates a character string of 123. It is usually used together with signs such as exclamation point (!), percent sign (%), and plus sign (+). For example, 408(12)+ can match the character string 40812 or 408121212, but not 408. In this pattern, 408 must be followed by one string of 12 at a minimum.

| NOTE: · An exclamation point (!), plus sign (+), or percent sign (%) must follow a valid digit or digit string. · To use brackets ([ ]) and parentheses (( )) together, use them in the form of "( [ ] )". The "( ( ) )", "[ [ ] ]", and "[ ( ) ]" forms are not allowed.

Usage guidelines

For a local POTS entity, this command defines a local number template to be bound to the local voice interface.

For a trunk POTS entity or a VoIP entity, this command defines a called number template.

Examples

# Configure the number template as 1000 for POTS entity 1000.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1000 pots

[Sysname-voice-dial-entity1000] match-template 1000

# Configure the number template as 2000 for VoIP entity 2000.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 2000 voip

[Sysname-voice-dial-entity2000] match-template 2000

Voice entity commands -- outband nte

Use outband nte to enable NTE mode for out-of-band DTMF signaling.

Use undo outband to restore the default.

Syntax

outband nte

undo outband

Default

Inband DTMF signaling is used.

Views

POTS entity view

VoIP entity view

IVR entity view

Predefined user roles

network-admin

mdc-admin

Usage guidelines

As a best practice, configure the outband nte command and the same payload type value on the originating and terminating devices to avoid DTMF tone transmission failure.

Examples

# Enable NTE mode for out-of-band DTMF signaling.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 voip

[Sysname-voice-dial-entity10] outband nte

Related commands

rtp payload-type nte

Voice entity commands -- playout delay

Use playout delay to set the playout delay time for voice packets.

Use undo playout delay to restore the default.

Syntax

playout-delay { initial milliseconds | maximum milliseconds | minimum milliseconds }

undo playout-delay { initial | maximum | minimum }

Default

The initial playout delay time for voice packets is 30 milliseconds. The maximum playout delay time is 160 milliseconds. The minimum playout delay time is 10 milliseconds.

Views

POTS entity view

VoIP entity view

Predefined user roles

network-admin

mdc-admin

Parameters

initial milliseconds: Specifies the initial playout delay time in adaptive mode or the fixed playout delay time in fixed mode. The value range for milliseconds is 5 to 300 milliseconds.

maximum milliseconds: (Adaptive mode only) Specifies the maximum playout delay time for voice packets. The value range is 60 to 300 milliseconds.

minimum milliseconds: (Adaptive mode only) Specifies the minimum playout delay time for voice packets. The value range is 0 to 40 milliseconds.

Examples

# Configure the playout delay mode as adaptive, and set the minimum playout delay time to 30 milliseconds.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 voip

[Sysname-voice-dial-entity10] playout-delay mode adaptive

[Sysname-voice-dial-entity10] playout-delay minimum 30

Voice entity commands -- playout delay mode

Use playout delay mode to configure the playout delay mode for voice packets.

Use undo playout delay mode to restore the default.

Syntax

playout-delay mode { adaptive | fixed }

undo playout-delay mode

Default

The playout delay mode for voice packets is fixed.

Views

POTS entity view

VoIP entity view

Predefined user roles

network-admin

mdc-admin

Parameters

adaptive: Specifies the playout delay mode as adaptive. In adaptive mode, the buffer size is automatically adjusted based on network conditions.

fixed: Specifies the playout delay mode as fixed. In fixed mode, the buffer size is fixed.

Usage guidelines

In an ideal voice network environment, the delay of each voice packet (time for each voice packet to travel from the sender to the receiver) is fixed. That is, the jitter is 0. In an actual voice network, the delay might vary from packet to packet.

To smoothly play out voice packets received with different delay times, the receiver can buffer the voice packets for a period of time (playout delay time). By configuring playout delay, you can prevent delay variation (jitter) from affecting voice quality.

Examples

# Configure the playout delay mode as adaptive.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 voip

[Sysname-voice-dial-entity10] playout-delay mode adaptive

Voice entity commands -- rtp payload-type nte

Use rtp payload-type nte to set the NTE payload type value in RTP packets.

Use undo rtp payload-type nte to restore the default.

Syntax

rtp payload-type nte value

undo rtp payload-type nte

Default

The NTE payload type value in RTP packets is 101.

Views

POTS entity view

VoIP entity view

IVR entity view

Predefined user roles

network-admin

mdc-admin

Parameters

value: Specifies the value of the NTE payload type in RTP packets, in the range of 96 to 127. The value 98 is reserved for identifying nonstandard T38 fax packets.

Usage guidelines

| CAUTION: To avoid negotiation failure, do not set the payload type field to a value forbidden by an interconnected device from another vendor.

As a best practice, configure the outband nte command and the same payload type value on the originating and terminating devices to avoid DTMF tone transmission failure.

Examples

# Set the NTE payload value of RTP packets to 102 for VoIP entity 10.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 voip

[Sysname-voice-dial-entity10] outband nte

[Sysname-voice-dial-entity10] rtp payload-type nte 102

Related commands

outband nte

Voice entity commands -- rtp-detect timeout

Use rtp-detect timeout to set the RTP timeout period.

Use undo rtp-detect timeout to restore the default.

Syntax

rtp-detect timeout value

undo rtp-detect timeout

Default

The RTP timeout period is 120 seconds.

Views

Voice view

Predefined user roles

network-admin

Parameters

value: Specifies the RTP timeout period in the range of 2 to 300 seconds.

Usage guidelines

This command enables the device to disconnect a call if it does not receive RTP traffic during the set timeout period.

Examples

# Set the RTP timeout period to 60 seconds.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] rtp-detect timeout 60

Voice entity commands -- shutdown

Use shutdown to shut down a voice entity.

Use undo shutdown to bring up a voice entity.

Syntax

shutdown

undo shutdown

Default

A voice entity is up.

Views

POTS entity view

VoIP entity view

IVR entity view

Predefined user roles

network-admin

mdc-admin

Examples

# Shut down POTS entity 10.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 pots

[Sysname-voice-dial-entity10] shutdown

Voice entity commands -- vad-on

Use vad-on to enable VAD.

Use undo vad-on to disable VAD.

Syntax

vad-on [ g711 | g723r53 | g723r63 | g729a | g729r8 ] *

undo vad-on [ g711 | g723r53 | g723r63 | g729a | g729r8 ] *

Default

VAD is disabled.

Views

POTS entity view

VoIP entity view

IVR entity view

Predefined user roles

network-admin

mdc-admin

Parameters

g711: Enables VAD for the G.711 codec.

g723r53: Enables VAD for the G.723.1 Annex A codec at 5.3 kbps.

g723r63: Enables VAD for the G.723.1 Annex A codec at 6.3 kbps.

g729a: Enables VAD for the G.729 Annex A codec at 8 kbps.

g729r8: Enables VAD for the G.729 codec at 8 kbps.

Usage guidelines

If you execute the vad-on or undo vad-on command without specifying a codec, VAD is enabled or disabled for all codecs.

The G.726 codec does not support VAD. The G.729br8 codec always supports VAD.

Examples

# Enable VAD for the g723r53 codec on POTS entity 10.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 pots

[Sysname-voice-dial-entity10] vad-on g723r53

Voice entity commands -- voice class codec

Use voice class codec to create a codec template and enter its view, or enter the view of an existing codec template.

Use undo voice class codec to delete a codec template.

Syntax

voice class codec tag

undo voice class codec tag

Default

No codec templates exist.

Views

Voice view

Predefined user roles

network-admin

mdc-admin

Parameters

tag: Specifies the number of the codec template, in the range of 1 to 2147483647.

Usage guidelines

You can create a maximum of 16 codec templates.

Examples

# Create codec template 1.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] voice class codec 1

[Sysname-voice-class-codec1]

Voice entity commands -- voice-class codec

Use voice-class codec to bind a codec template to a voice entity.

Use undo voice-class codec to restore the default.

Syntax

voice-class codec tag

undo voice-class codec

Default

No codec template is bound to a voice entity.

Views

POTS entity view

VoIP entity view

IVR entity view

Predefined user roles

network-admin

mdc-admin

Parameters

tag: Specifies a codec template by its number in the range of 1 to 2147483647.

Usage guidelines

You can bind a nonexistent codec template to a voice entity. The codec template takes effect only after you assign priorities to the codecs in the template by using the codec preference command.

A call can be established only when the calling party and the called party use the same codec.

Only one codec template can be bound to a voice entity. If you configure the voice-class codec command multiple times, the most recent configuration takes effect.

You can use this command to bind a codec template to a voice entity, or use the codec command to directly configure a codec for a voice entity.

Examples

# Bind codec template 1 to VoIP entity 10.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 10 voip

[Sysname-voice-dial-entity10] voice-class codec 1

Related commands

codec preference

voice class codec

Voice entity commands -- voice-setup

Use voice-setup to enter voice view and enable voice services.

Use undo voice-setup to disable voice services, delete all voice settings, and exit voice view.

Syntax

voice-setup

undo voice-setup

Default

Voice services are disabled.

Views

System view

Predefined user roles

network-admin

mdc-admin

Examples

# Enter voice view and enable voice services.

<Sysname> system-view

[Sysname] voice-setup