13-Voice Command Reference

HomeSupportResource CenterReference GuidesCommand ReferencesH3C MSR810[2600][3600] Routers Command Reference(V7)-R0809-6W40013-Voice Command Reference
02-Voice entity commands
Title Size Download
02-Voice entity commands 243.91 KB

Voice entity commands

The following compatibility matrixes show the support of hardware platforms for voice entities:

 

Hardware

Voice entity compatibility

MSR810, MSR810-W, MSR810-W-DB, MSR810-LM, MSR810-W-LM, MSR810-10-PoE, MSR810-LM-HK, MSR810-W-LM-HK, MSR810-LM-CNDE-SJK, MSR810-CNDE-SJK

Yes for only VoIP entities and IVR entities

MSR810-LMS, MSR810-LUS

No

MSR810-LMS-EA, MSR810-LME

No

MSR2600-6-X1

No

MSR2600-10-X1

Yes

MSR 2630

Yes

MSR3600-28, MSR3600-51

Yes

MSR3600-28-SI, MSR3600-51-SI

No

MSR3600-28-X1, MSR3600-28-X1-DP, MSR3600-51-X1, MSR3600-51-X1-DP

No

MSR3610-I-DP, MSR3610-IE-DP, MSR3610-IE-ES, MSR3610-IE-EAD

No

MSR3610-X1, MSR3610-X1-DP, MSR3610-X1-DC, MSR3610-X1-DP-DC

Yes

MSR 3610, MSR 3620, MSR 3620-DP, MSR 3640, MSR 3660

Yes

MSR3610-G, MSR3620-G

No

Hardware

Voice entity compatibility

MSR810-W-WiNet, MSR810-LM-WiNet

Yes for only VoIP entities and IVR entities

MSR830-4LM-WiNet

No

MSR830-5BEI-WiNet, MSR830-6EI-WiNet, MSR830-10BEI-WiNet

No

MSR830-6BHI-WiNet, MSR830-10BHI-WiNet

No

MSR2600-6-WiNet

No

MSR2600-10-X1-WiNet

Yes

MSR2630-WiNet

Yes

MSR3600-28-WiNet

Yes

MSR3610-X1-WiNet

Yes

MSR3610-WiNet, MSR3620-10-WiNet, MSR3620-DP-WiNet, MSR3620-WiNet, MSR3660-WiNet

Yes

Hardware

Voice entity compatibility

MSR2630-XS

No

MSR3600-28-XS

No

MSR3610-XS

Yes

MSR3620-XS

Yes

MSR3610-I-XS

No

MSR3610-IE-XS

No

 

Hardware

Command compatibility

MSR810-LM-GL

Yes for only VoIP entities and IVR entities

MSR810-W-LM-GL

Yes for only VoIP entities and IVR entities

MSR830-6EI-GL

No

MSR830-10EI-GL

No

MSR830-6HI-GL

No

MSR830-10HI-GL

No

MSR2600-6-X1-GL

No

MSR3600-28-SI-GL

No

To support voice entities, some device models require the Voice Software License. For more information, see license management in Fundamentals Configuration Guide.

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

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:

¡     The number length threshold is exceeded.

¡     The subscriber enters the terminator.

¡     The timer expires.

·     regular-expression: Specifies a matching pattern of characters. Table 1 lists the available characters.

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 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

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

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

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

A call can be established only when 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

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

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

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

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

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

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 LegCall leg. A voice call has two call legs: an inbound call leg and an outbound call leg.

·     Temp LegTemporary leg. This leg type exists on a device operating as a SIP trunk device.

·     MOH LegMusic on hold leg.

Status

Leg status:

·     Call leg status:

¡     Finding-routeThe leg is waiting for a route lookup response.

¡     Incoming_ACKThe leg received a call.

¡     Outgoing_ACKThe leg sent out a call.

¡     ConnectedA call was connected.

·     MOH leg status:

¡     Waiting-music-responseThe leg is waiting for a response from the MOH server.

¡     MOH_connectedThe 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:

·     CHCall hold.

·     CWCall waiting.

·     MCHMultiparty call hold.

·     MOHMusic on hold.

·     CTCall transfer for SIP-to-SIP calls.

·     CFCall forwarding for SIP-to-SIP calls.

·     CBCall backup.

·     CFOCall forwarding originator.

·     CTOCall transfer originator.

·     CTRCall transfer recipient.

·     CTTCall transfer target.

·     ConferenceThree-party conference.

 

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

Parameters

tag: Specifies a call in progress by its number in the range of 0 to 2147483647.

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.

 

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

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.

 

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

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

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

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

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

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:

¡     The number length threshold is exceeded.

¡     The subscriber enters the terminator.

¡     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, 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

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

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

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)

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

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

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

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:

¡     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.

¡     Because the DSS1 signaling uses ISDN transmission, this problem does not exist.

¡     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:

¡     The number length threshold is exceeded.

¡     The subscriber enters the terminator.

¡     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

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

Usage guidelines

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

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

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

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

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

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

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

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

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

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

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

send-ring

Use send-ring to configure the originating side to play ringback tones.

Use undo send-ring to restore the default.

Syntax

send-ring

undo send-ring

Default

The originating side cannot play ringback tones.

Views

POTS entity view

VoIP entity view

Predefined user roles

network-admin

Usage guidelines

If the terminating side of a call cannot play ringback tones, configure the POTS or VoIP entity on the originating side to play ringback tones.

This feature does not take effect on a POTS entity if the POTS entity is bound to an FXS or FXO interface.

Examples

# Configure the originating side to play ringback tones.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] dial-program

[Sysname-voice-dial] entity 1 voip

[Sysname-voice-dial-entity1] send-ring

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

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

sip log enable

Use sip log enable to enable SIP logging.

Use undo sip log enable to disable SIP logging.

Syntax

sip log enable

undo sip log enable

Default

SIP logging is disabled.

Views

Voice view

Predefined user roles

network-admin

Usage guidelines

SIP logging enables the device to log SIP call events and send the log messages to the information center. With the information center, you can set log message filtering and output rules, including output destinations. For more information about using the information center, see Network Management and Monitoring Configuration Guide.

Examples

# Enable SIP logging.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] sip log enable

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

Parameters

g711: Enables VAD for the G.711 codec. The G.711 codec is supported only on the following interface modules: HMIM-1VE1, HMIM-1VT1, HMIM-2VE1, HMIM-2VT1, SIC-1BSV, SIC-1VE1T1, SIC-2BSV, SIC-2FXS1FXO. The G.711 codec is supported only if the DSP image is configured as the Microsoft-verified version.

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 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

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-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

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-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

Examples

# Enter voice view and enable voice services.

<Sysname> system-view

[Sysname] voice-setup

vqa dsp-buffer maximum-time

 

NOTE:

This command takes effect only on E&M interface modules.

 

Use vqa dsp-buffer maximum-time to set the maximum duration of DSP-buffered data.

Use undo vqa dsp-buffer maximum-time to restore the default.

Syntax

vqa dsp-buffer maximum-time time

undo vqa dsp-buffer maximum-time

The following compatibility matrixes show the support of hardware platforms for this command:

 

Hardware

Command compatibility

MSR810, MSR810-W, MSR810-W-DB, MSR810-LM, MSR810-W-LM, MSR810-10-PoE, MSR810-LM-HK, MSR810-W-LM-HK, MSR810-LM-CNDE-SJK, MSR810-CNDE-SJK

No

MSR810-LMS, MSR810-LUS

No

MSR2600-6-X1, MSR2600-10-X1

No

MSR 2630

No

MSR3600-28, MSR3600-51

No

MSR3600-28-SI, MSR3600-51-SI

No

MSR3600-28-X1, MSR3600-28-X1-DP, MSR3600-51-X1, MSR3600-51-X1-DP

No

MSR3610-I-DP, MSR3610-IE-DP, MSR3610-IE-ES, MSR3610-IE-EAD

No

MSR3610-X1, MSR3610-X1-DP, MSR3610-X1-DC, MSR3610-X1-DP-DC

No

MSR 3610, MSR 3620, MSR 3620-DP, MSR 3640, MSR 3660

Yes

MSR3610-G, MSR3620-G

No

Hardware

Command compatibility

MSR2630-XS

No

MSR3600-28-XS

No

MSR3610-XS

No

MSR3620-XS

Yes

MSR3610-I-XS

No

MSR3610-IE-XS

No

 

Hardware

Command compatibility

MSR810-LM-GL

No

MSR810-W-LM-GL

No

MSR830-6EI-GL

No

MSR830-10EI-GL

No

MSR830-6HI-GL

No

MSR830-10HI-GL

No

MSR2600-6-X1-GL

No

MSR3600-28-SI-GL

No

Default

The maximum duration of DSP-buffered data is 270 milliseconds.

Views

Voice view

Predefined user roles

network-admin

Parameters

time: Specifies the maximum duration of DSP-buffered data, in milliseconds. The value range for this argument is 0 and 10 to 480. If you set the maximum duration of DSP-buffered data to 0 milliseconds, the device does not clear the DSP buffer.

Usage guidelines

VoIP voice data is buffered in the DSP buffer if a network latency or jitter exists. When the maximum duration of DSP-buffered data is reached, the device clears the DSP buffer to improve the quality of VoIP calls. You can use this command to adjust the maximum duration of DSP-buffered data.

When PCM pass-through is enabled, the set maximum duration of DSP-buffered data takes effect. 

When PCM pass-through is disabled, one of the following rules applies:

·     If the set maximum duration of DSP-buffered data is in the range of 10 to 179 milliseconds, the default value (270 milliseconds) takes effect.

·     If the set maximum duration of DSP-buffered data is 0 or in the range of 180 to 480 milliseconds, the set value takes effect.

For more information about PCM pass-through, see voice interface configuration in Voice Configuration Guide.

Examples

# Set the maximum duration of DSP-buffered data to 300 milliseconds.

<Sysname> system-view

[Sysname] voice-setup

[Sysname-voice] vqa dsp-buffer maximum-time 300

Related commands

pcm-passthrough